From 8ca094795add18733faeb2a1f335bb33f40f9894 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Tue, 7 Dec 2010 11:43:13 +0100 Subject: [PATCH 001/351] pulsesink: don't uncork in _start Don't uncork in the _start method just yet but wait until we have written some samples to pulseaudio. This avoid underruns on pulseaudio and less crackling noises when starting. --- ext/pulse/pulsesink.c | 1 - 1 file changed, 1 deletion(-) diff --git a/ext/pulse/pulsesink.c b/ext/pulse/pulsesink.c index 8bed0c1b29..2a9a108b2e 100644 --- a/ext/pulse/pulsesink.c +++ b/ext/pulse/pulsesink.c @@ -1072,7 +1072,6 @@ gst_pulseringbuffer_start (GstRingBuffer * buf) GST_DEBUG_OBJECT (psink, "starting"); pbuf->paused = FALSE; - gst_pulsering_set_corked (pbuf, FALSE, FALSE); pa_threaded_mainloop_unlock (mainloop); return TRUE; From 6f8ce30c207ed0b0402e6c0f6727f5b2c31c4e10 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Tue, 7 Dec 2010 13:11:48 +0100 Subject: [PATCH 002/351] flvdemux: use aac codec-data to adjust samplerate if needed Based on patch by Fabien Lebaillif-Delamare Fixes #636621. --- gst/flv/Makefile.am | 6 +++--- gst/flv/gstflvdemux.c | 30 ++++++++++++++++++++++++++---- 2 files changed, 29 insertions(+), 7 deletions(-) diff --git a/gst/flv/Makefile.am b/gst/flv/Makefile.am index b2d37ebbb5..ab08cdd9e7 100644 --- a/gst/flv/Makefile.am +++ b/gst/flv/Makefile.am @@ -1,9 +1,9 @@ plugin_LTLIBRARIES = libgstflv.la libgstflv_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) -libgstflv_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) $(GST_BASE_LIBS) $(GST_LIBS) \ - -lgstpbutils-@GST_MAJORMINOR@ -libgstflv_la_LDFLAGS = ${GST_PLUGIN_LDFLAGS} +libgstflv_la_LIBADD = -lgstpbutils-@GST_MAJORMINOR@ \ + $(GST_PLUGINS_BASE_LIBS) $(GST_BASE_LIBS) $(GST_LIBS) +libgstflv_la_LDFLAGS = ${GST_PLUGIN_LDFLAGS} libgstflv_la_SOURCES = gstflvdemux.c gstflvmux.c libgstflv_la_LIBTOOLFLAGS = --tag=disable-static diff --git a/gst/flv/gstflvdemux.c b/gst/flv/gstflvdemux.c index 04d4b5b639..d976fedba2 100644 --- a/gst/flv/gstflvdemux.c +++ b/gst/flv/gstflvdemux.c @@ -40,6 +40,7 @@ #include #include #include +#include static GstStaticPadTemplate flv_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, @@ -575,6 +576,7 @@ gst_flv_demux_audio_negotiate (GstFlvDemux * demux, guint32 codec_tag, GstCaps *caps = NULL; gchar *codec_name = NULL; gboolean ret = FALSE; + guint adjusted_rate = rate; switch (codec_tag) { case 1: @@ -603,9 +605,29 @@ gst_flv_demux_audio_negotiate (GstFlvDemux * demux, guint32 codec_tag, caps = gst_caps_new_simple ("audio/x-nellymoser", NULL); break; case 10: + { + /* use codec-data to extract and verify samplerate */ + if (demux->audio_codec_data && + GST_BUFFER_SIZE (demux->audio_codec_data) >= 2) { + gint freq_index; + + freq_index = + ((GST_READ_UINT16_BE (GST_BUFFER_DATA (demux->audio_codec_data)))); + freq_index = (freq_index & 0x0780) >> 7; + adjusted_rate = + gst_codec_utils_aac_get_sample_rate_from_index (freq_index); + + if (adjusted_rate && (rate != adjusted_rate)) { + GST_LOG_OBJECT (demux, "Ajusting AAC sample rate %d -> %d", rate, + adjusted_rate); + } else { + adjusted_rate = rate; + } + } caps = gst_caps_new_simple ("audio/mpeg", "mpegversion", G_TYPE_INT, 4, "framed", G_TYPE_BOOLEAN, TRUE, NULL); break; + } case 7: caps = gst_caps_new_simple ("audio/x-alaw", NULL); break; @@ -624,8 +646,8 @@ gst_flv_demux_audio_negotiate (GstFlvDemux * demux, guint32 codec_tag, goto beach; } - gst_caps_set_simple (caps, - "rate", G_TYPE_INT, rate, "channels", G_TYPE_INT, channels, NULL); + gst_caps_set_simple (caps, "rate", G_TYPE_INT, adjusted_rate, + "channels", G_TYPE_INT, channels, NULL); if (demux->audio_codec_data) { gst_caps_set_simple (caps, "codec_data", GST_TYPE_BUFFER, @@ -635,7 +657,7 @@ gst_flv_demux_audio_negotiate (GstFlvDemux * demux, guint32 codec_tag, ret = gst_pad_set_caps (demux->audio_pad, caps); if (G_LIKELY (ret)) { - /* Store the caps we have set */ + /* Store the caps we got from tags */ demux->audio_codec_tag = codec_tag; demux->rate = rate; demux->channels = channels; @@ -851,7 +873,7 @@ gst_flv_demux_parse_tag_audio (GstFlvDemux * demux, GstBuffer * buffer) switch (aac_packet_type) { case 0: { - /* AudioSpecificConfic data */ + /* AudioSpecificConfig data */ GST_LOG_OBJECT (demux, "got an AAC codec data packet"); if (demux->audio_codec_data) { gst_buffer_unref (demux->audio_codec_data); From d2948bb259ac070dbbb8ca1e1b0341e39ce1d87a Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Tue, 7 Dec 2010 17:19:00 +0100 Subject: [PATCH 003/351] qtdemux: fragmented support; fix offset handling and relax error raising In particular, accept unknown stream in track fragment, and only error out if that raises problems later on with respect to offset tracking. Fixes #620283. --- gst/qtdemux/qtdemux.c | 61 +++++++++++++++++++++++++++++++------------ 1 file changed, 44 insertions(+), 17 deletions(-) diff --git a/gst/qtdemux/qtdemux.c b/gst/qtdemux/qtdemux.c index 3d5498e80b..b174ac7636 100644 --- a/gst/qtdemux/qtdemux.c +++ b/gst/qtdemux/qtdemux.c @@ -1983,7 +1983,7 @@ static gboolean qtdemux_parse_trun (GstQTDemux * qtdemux, GstByteReader * trun, QtDemuxStream * stream, guint32 d_sample_duration, guint32 d_sample_size, guint32 d_sample_flags, gint64 moof_offset, gint64 moof_length, - gint64 * base_offset) + gint64 * base_offset, gint64 * running_offset) { guint64 timestamp; gint32 data_offset = 0; @@ -2016,7 +2016,7 @@ qtdemux_parse_trun (GstQTDemux * qtdemux, GstByteReader * trun, GST_LOG_OBJECT (qtdemux, "base_offset at moof"); *base_offset = moof_offset; } - *base_offset += data_offset; + *running_offset = *base_offset + data_offset; } else { /* if no offset at all, that would mean data starts at moof start, * which is a bit wrong and is ismv crappy way, so compensate @@ -2026,10 +2026,12 @@ qtdemux_parse_trun (GstQTDemux * qtdemux, GstByteReader * trun, GST_LOG_OBJECT (qtdemux, "base_offset assumed in mdat after moof"); ismv = TRUE; } + if (*running_offset == -1) + *running_offset = *base_offset; } - GST_LOG_OBJECT (qtdemux, "base offset now %" G_GINT64_FORMAT, *base_offset); - + GST_LOG_OBJECT (qtdemux, "running offset now %" G_GINT64_FORMAT, + *running_offset); GST_LOG_OBJECT (qtdemux, "trun offset %d, flags 0x%x, entries %d", data_offset, flags, samples_count); @@ -2135,7 +2137,7 @@ qtdemux_parse_trun (GstQTDemux * qtdemux, GstByteReader * trun, data += entry_size; /* fill the sample information */ - sample->offset = *base_offset; + sample->offset = *running_offset; sample->pts_offset = ct; sample->size = size; sample->timestamp = timestamp; @@ -2144,7 +2146,7 @@ qtdemux_parse_trun (GstQTDemux * qtdemux, GstByteReader * trun, /* ismv seems to use 0x40 for keyframe, 0xc0 for non-keyframe, * now idea how it relates to bitfield other than massive LE/BE confusion */ sample->keyframe = ismv ? ((sflags & 0xff) == 0x40) : !(sflags & 0x10000); - *base_offset += size; + *running_offset += size; timestamp += dur; sample++; } @@ -2220,10 +2222,8 @@ qtdemux_parse_tfhd (GstQTDemux * qtdemux, GstByteReader * tfhd, goto invalid_track; *stream = qtdemux_find_stream (qtdemux, track_id); - if (G_UNLIKELY (!*stream)) { - GST_DEBUG_OBJECT (qtdemux, "unknown stream in tfhd"); - goto invalid_track; - } + if (G_UNLIKELY (!*stream)) + goto unknown_stream; if (flags & TF_BASE_DATA_OFFSET) if (!gst_byte_reader_get_uint64_be (tfhd, (guint64 *) base_offset)) @@ -2254,9 +2254,14 @@ qtdemux_parse_tfhd (GstQTDemux * qtdemux, GstByteReader * tfhd, invalid_track: { - GST_WARNING_OBJECT (qtdemux, "invalid track header"); + GST_WARNING_OBJECT (qtdemux, "invalid track fragment header"); return FALSE; } +unknown_stream: + { + GST_DEBUG_OBJECT (qtdemux, "unknown stream in tfhd"); + return TRUE; + } } static gboolean @@ -2266,7 +2271,7 @@ qtdemux_parse_moof (GstQTDemux * qtdemux, const guint8 * buffer, guint length, GNode *moof_node, *traf_node, *tfhd_node, *trun_node; GstByteReader trun_data, tfhd_data; guint32 ds_size = 0, ds_duration = 0, ds_flags = 0; - gint64 base_offset; + gint64 base_offset, running_offset; /* NOTE @stream ignored */ @@ -2275,7 +2280,7 @@ qtdemux_parse_moof (GstQTDemux * qtdemux, const guint8 * buffer, guint length, qtdemux_node_dump (qtdemux, moof_node); /* unknown base_offset to start with */ - base_offset = -1; + base_offset = running_offset = -1; traf_node = qtdemux_tree_get_child_by_type (moof_node, FOURCC_traf); while (traf_node) { /* Fragment Header node */ @@ -2287,18 +2292,31 @@ qtdemux_parse_moof (GstQTDemux * qtdemux, const guint8 * buffer, guint length, if (!qtdemux_parse_tfhd (qtdemux, &tfhd_data, &stream, &ds_duration, &ds_size, &ds_flags, &base_offset)) goto missing_tfhd; + if (G_UNLIKELY (!stream)) { + /* we lost track of offset, we'll need to regain it, + * but can delay complaining until later or avoid doing so altogether */ + base_offset = -2; + goto next; + } + if (G_UNLIKELY (base_offset < -1)) + goto lost_offset; /* Track Run node */ trun_node = qtdemux_tree_get_child_by_type_full (traf_node, FOURCC_trun, &trun_data); while (trun_node) { qtdemux_parse_trun (qtdemux, &trun_data, stream, - ds_duration, ds_size, ds_flags, moof_offset, length, &base_offset); + ds_duration, ds_size, ds_flags, moof_offset, length, &base_offset, + &running_offset); /* iterate all siblings */ trun_node = qtdemux_tree_get_sibling_by_type_full (trun_node, FOURCC_trun, &trun_data); } - + /* if no new base_offset provided for next traf, + * base is end of current traf */ + base_offset = running_offset; + running_offset = -1; + next: /* iterate all siblings */ traf_node = qtdemux_tree_get_sibling_by_type (traf_node, FOURCC_traf); } @@ -2306,11 +2324,20 @@ qtdemux_parse_moof (GstQTDemux * qtdemux, const guint8 * buffer, guint length, return TRUE; missing_tfhd: + { + GST_DEBUG_OBJECT (qtdemux, "missing tfhd box"); + goto fail; + } +lost_offset: + { + GST_DEBUG_OBJECT (qtdemux, "lost offset"); + goto fail; + } +fail: { g_node_destroy (moof_node); GST_ELEMENT_ERROR (qtdemux, STREAM, DEMUX, - (_("This file is corrupt and cannot be played.")), - ("missing tfhd box")); + (_("This file is corrupt and cannot be played.")), (NULL)); return FALSE; } } From 5239e19e9cb33e9072af4fc700ec9901ae67f601 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Wed, 8 Dec 2010 11:35:33 +0100 Subject: [PATCH 004/351] qtdemux: fix handling near end-of-file corner cases Also, relax some error handling to not bail out completely when something feels amiss, but consider this EOF and continue with was obtained so far. --- gst/qtdemux/qtdemux.c | 31 +++++++++++++++++++------------ 1 file changed, 19 insertions(+), 12 deletions(-) diff --git a/gst/qtdemux/qtdemux.c b/gst/qtdemux/qtdemux.c index b174ac7636..134188e046 100644 --- a/gst/qtdemux/qtdemux.c +++ b/gst/qtdemux/qtdemux.c @@ -1856,9 +1856,10 @@ qtdemux_parse_uuid (GstQTDemux * qtdemux, const guint8 * buffer, gint length) } } +/* caller verifies at least 8 bytes in buf */ static void -extract_initial_length_and_fourcc (const guint8 * data, guint64 * plength, - guint32 * pfourcc) +extract_initial_length_and_fourcc (const guint8 * data, guint size, + guint64 * plength, guint32 * pfourcc) { guint64 length; guint32 fourcc; @@ -1870,7 +1871,7 @@ extract_initial_length_and_fourcc (const guint8 * data, guint64 * plength, if (length == 0) { length = G_MAXUINT32; - } else if (length == 1) { + } else if (length == 1 && size >= 16) { /* this means we have an extended size, which is the 64 bit value of * the next 8 bytes */ length = QT_UINT64 (data + 8); @@ -2537,16 +2538,18 @@ gst_qtdemux_loop_state_header (GstQTDemux * qtdemux) ret = gst_pad_pull_range (qtdemux->sinkpad, cur_offset, 16, &buf); if (G_UNLIKELY (ret != GST_FLOW_OK)) goto beach; - if (G_LIKELY (GST_BUFFER_SIZE (buf) == 16)) - extract_initial_length_and_fourcc (GST_BUFFER_DATA (buf), &length, &fourcc); + if (G_LIKELY (GST_BUFFER_SIZE (buf) >= 8)) + extract_initial_length_and_fourcc (GST_BUFFER_DATA (buf), + GST_BUFFER_SIZE (buf), &length, &fourcc); gst_buffer_unref (buf); + /* maybe we already got most we needed, so only consider this eof */ if (G_UNLIKELY (length == 0)) { - GST_ELEMENT_ERROR (qtdemux, STREAM, DEMUX, - (_("This file is invalid and cannot be played.")), + GST_ELEMENT_WARNING (qtdemux, STREAM, DEMUX, + (_("Invalid atom size.")), ("Header atom '%" GST_FOURCC_FORMAT "' has empty length", GST_FOURCC_ARGS (fourcc))); - ret = GST_FLOW_ERROR; + ret = GST_FLOW_UNEXPECTED; goto beach; } @@ -3887,7 +3890,8 @@ gst_qtdemux_chain (GstPad * sinkpad, GstBuffer * inbuf) data = gst_adapter_peek (demux->adapter, demux->neededbytes); /* get fourcc/length, set neededbytes */ - extract_initial_length_and_fourcc ((guint8 *) data, &size, &fourcc); + extract_initial_length_and_fourcc ((guint8 *) data, demux->neededbytes, + &size, &fourcc); GST_DEBUG_OBJECT (demux, "Peeking found [%" GST_FOURCC_FORMAT "] " "size: %" G_GUINT64_FORMAT, GST_FOURCC_ARGS (fourcc), size); if (size == 0) { @@ -3978,7 +3982,8 @@ gst_qtdemux_chain (GstPad * sinkpad, GstBuffer * inbuf) data = gst_adapter_peek (demux->adapter, demux->neededbytes); /* parse the header */ - extract_initial_length_and_fourcc (data, NULL, &fourcc); + extract_initial_length_and_fourcc (data, demux->neededbytes, NULL, + &fourcc); if (fourcc == FOURCC_moov) { GST_DEBUG_OBJECT (demux, "Parsing [moov]"); @@ -4977,11 +4982,13 @@ qtdemux_find_atom (GstQTDemux * qtdemux, guint64 * offset, if (G_UNLIKELY (ret != GST_FLOW_OK)) goto locate_failed; if (G_LIKELY (GST_BUFFER_SIZE (buf) != 16)) { + /* likely EOF */ + ret = GST_FLOW_UNEXPECTED; gst_buffer_unref (buf); - ret = GST_FLOW_ERROR; goto locate_failed; } - extract_initial_length_and_fourcc (GST_BUFFER_DATA (buf), length, &lfourcc); + extract_initial_length_and_fourcc (GST_BUFFER_DATA (buf), 16, length, + &lfourcc); gst_buffer_unref (buf); if (G_UNLIKELY (*length == 0)) { From df2982ce7da4f3b425fe3e7901b7a9960a39b1eb Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Thu, 9 Dec 2010 13:49:04 +0100 Subject: [PATCH 005/351] qtdemux: fix flow return aggregation --- gst/qtdemux/qtdemux.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/gst/qtdemux/qtdemux.c b/gst/qtdemux/qtdemux.c index 134188e046..a744fbc000 100644 --- a/gst/qtdemux/qtdemux.c +++ b/gst/qtdemux/qtdemux.c @@ -3248,6 +3248,11 @@ gst_qtdemux_combine_flows (GstQTDemux * demux, QtDemuxStream * stream, /* store the value */ stream->last_ret = ret; + /* any other error that is not-linked or eos can be returned right away */ + if (G_LIKELY (ret != GST_FLOW_UNEXPECTED && ret != GST_FLOW_NOT_LINKED)) + goto done; + + /* only return NOT_LINKED if all other pads returned NOT_LINKED */ for (i = 0; i < demux->n_streams; i++) { QtDemuxStream *ostream = demux->streams[i]; From 6729a3b79ccbc108ea27641e460caec090703f4a Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Mon, 6 Dec 2010 12:16:12 +0100 Subject: [PATCH 006/351] j2kdepay: make the depayloader more resilient Use 3 adapters, one to accumulate paketization units, another on to accumulate tiles and a last one to accumulate the final frame. Don't just blindly flush the adapter on DISCONT but only discard the current packetization unit. When we dropped jpeg2000 packets between SOP markers, adjust the SOT header with the new lenght. --- gst/rtp/gstrtpj2kdepay.c | 421 ++++++++++++++++++++++++++++++++++----- gst/rtp/gstrtpj2kdepay.h | 15 +- 2 files changed, 383 insertions(+), 53 deletions(-) diff --git a/gst/rtp/gstrtpj2kdepay.c b/gst/rtp/gstrtpj2kdepay.c index 83ccd9c4a7..82411aa987 100644 --- a/gst/rtp/gstrtpj2kdepay.c +++ b/gst/rtp/gstrtpj2kdepay.c @@ -46,6 +46,16 @@ GST_STATIC_PAD_TEMPLATE ("sink", "clock-rate = (int) 90000, " "encoding-name = (string) \"JPEG2000\"") ); +typedef enum +{ + J2K_MARKER = 0xFF, + J2K_MARKER_SOC = 0x4F, + J2K_MARKER_SOT = 0x90, + J2K_MARKER_SOP = 0x91, + J2K_MARKER_SOD = 0x93, + J2K_MARKER_EOC = 0xD9 +} RtpJ2KMarker; + GST_BOILERPLATE (GstRtpJ2KDepay, gst_rtp_j2k_depay, GstBaseRTPDepayload, GST_TYPE_BASE_RTP_DEPAYLOAD); @@ -102,7 +112,40 @@ static void gst_rtp_j2k_depay_init (GstRtpJ2KDepay * rtpj2kdepay, GstRtpJ2KDepayClass * klass) { - rtpj2kdepay->adapter = gst_adapter_new (); + rtpj2kdepay->pu_adapter = gst_adapter_new (); + rtpj2kdepay->t_adapter = gst_adapter_new (); + rtpj2kdepay->f_adapter = gst_adapter_new (); +} + +static void +store_mheader (GstRtpJ2KDepay * rtpj2kdepay, guint idx, GstBuffer * buf) +{ + GstBuffer *old; + + GST_DEBUG_OBJECT (rtpj2kdepay, "storing main header %p at index %u", buf, + idx); + if ((old = rtpj2kdepay->MH[idx])) + gst_buffer_unref (old); + rtpj2kdepay->MH[idx] = buf; +} + +static void +clear_mheaders (GstRtpJ2KDepay * rtpj2kdepay) +{ + guint i; + + for (i = 0; i < 8; i++) + store_mheader (rtpj2kdepay, i, NULL); +} + +static void +gst_rtp_j2k_depay_reset (GstRtpJ2KDepay * rtpj2kdepay) +{ + clear_mheaders (rtpj2kdepay); + gst_adapter_clear (rtpj2kdepay->pu_adapter); + gst_adapter_clear (rtpj2kdepay->t_adapter); + gst_adapter_clear (rtpj2kdepay->f_adapter); + rtpj2kdepay->next_frag = 0; } static void @@ -112,8 +155,11 @@ gst_rtp_j2k_depay_finalize (GObject * object) rtpj2kdepay = GST_RTP_J2K_DEPAY (object); - g_object_unref (rtpj2kdepay->adapter); - rtpj2kdepay->adapter = NULL; + clear_mheaders (rtpj2kdepay); + + g_object_unref (rtpj2kdepay->pu_adapter); + g_object_unref (rtpj2kdepay->t_adapter); + g_object_unref (rtpj2kdepay->f_adapter); G_OBJECT_CLASS (parent_class)->finalize (object); } @@ -142,65 +188,185 @@ gst_rtp_j2k_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps) return res; } -static GstBuffer * -gst_rtp_j2k_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf) +static void +gst_rtp_j2k_depay_clear_pu (GstRtpJ2KDepay * rtpj2kdepay) +{ + gst_adapter_clear (rtpj2kdepay->pu_adapter); + rtpj2kdepay->have_sync = FALSE; +} + +static GstFlowReturn +gst_rtp_j2k_depay_flush_pu (GstBaseRTPDepayload * depayload) { GstRtpJ2KDepay *rtpj2kdepay; - GstBuffer *outbuf; - guint8 *payload; - guint frag_offset; + GstBuffer *mheader; + guint avail, MHF, mh_id; rtpj2kdepay = GST_RTP_J2K_DEPAY (depayload); - /* flush everything on discont for now */ - if (GST_BUFFER_IS_DISCONT (buf)) { - GST_DEBUG_OBJECT (rtpj2kdepay, "DISCONT, flushing data"); - gst_adapter_clear (rtpj2kdepay->adapter); - rtpj2kdepay->need_header = TRUE; + /* take all available buffers */ + avail = gst_adapter_available (rtpj2kdepay->pu_adapter); + if (avail == 0) + goto done; + + MHF = rtpj2kdepay->pu_MHF; + mh_id = rtpj2kdepay->last_mh_id; + + GST_DEBUG_OBJECT (rtpj2kdepay, "flushing PU of size %u", avail); + + if (MHF == 0) { + GList *packets, *walk; + + packets = gst_adapter_take_list (rtpj2kdepay->pu_adapter, avail); + /* append packets */ + for (walk = packets; walk; walk = g_list_next (walk)) { + GstBuffer *buf = GST_BUFFER_CAST (walk->data); + GST_DEBUG_OBJECT (rtpj2kdepay, "append pu packet of size %u", + GST_BUFFER_SIZE (buf)); + gst_adapter_push (rtpj2kdepay->t_adapter, buf); + } + g_list_free (packets); + } else { + /* we have a header */ + GST_DEBUG_OBJECT (rtpj2kdepay, "keeping header %u", mh_id); + /* we managed to see the start and end of the header, take all from + * adapter and keep in header */ + mheader = gst_adapter_take_buffer (rtpj2kdepay->pu_adapter, avail); + + store_mheader (rtpj2kdepay, mh_id, mheader); } - if (gst_rtp_buffer_get_payload_len (buf) < 8) - goto empty_packet; +done: + rtpj2kdepay->have_sync = FALSE; - payload = gst_rtp_buffer_get_payload (buf); + return GST_FLOW_OK; +} - /* - * 0 1 2 3 - * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 - * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - * |tp |MHF|mh_id|T| priority | tile number | - * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - * |reserved | fragment offset | - * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - */ - frag_offset = (payload[5] << 16) | (payload[6] << 8) | payload[7]; +static GstFlowReturn +gst_rtp_j2k_depay_flush_tile (GstBaseRTPDepayload * depayload) +{ + GstRtpJ2KDepay *rtpj2kdepay; + guint avail, mh_id; + GList *packets, *walk; + guint8 end[2]; + GstFlowReturn ret = GST_FLOW_OK; - GST_DEBUG_OBJECT (rtpj2kdepay, "frag %u", frag_offset); + rtpj2kdepay = GST_RTP_J2K_DEPAY (depayload); - if (rtpj2kdepay->need_header) { - if (frag_offset != 0) + /* flush pending PU */ + gst_rtp_j2k_depay_flush_pu (depayload); + + /* take all available buffers */ + avail = gst_adapter_available (rtpj2kdepay->t_adapter); + if (avail == 0) + goto done; + + mh_id = rtpj2kdepay->last_mh_id; + + GST_DEBUG_OBJECT (rtpj2kdepay, "flushing tile of size %u", avail); + + if (gst_adapter_available (rtpj2kdepay->f_adapter) == 0) { + GstBuffer *mheader; + + /* we need a header now */ + if ((mheader = rtpj2kdepay->MH[mh_id]) == NULL) goto waiting_header; - rtpj2kdepay->need_header = FALSE; + /* push header in the adapter */ + GST_DEBUG_OBJECT (rtpj2kdepay, "pushing header %u", mh_id); + gst_adapter_push (rtpj2kdepay->f_adapter, gst_buffer_ref (mheader)); } - /* take JPEG 2000 data, push in the adapter */ - outbuf = gst_rtp_buffer_get_payload_subbuffer (buf, 8, -1); - gst_adapter_push (rtpj2kdepay->adapter, outbuf); - outbuf = NULL; + /* check for last bytes */ + gst_adapter_copy (rtpj2kdepay->t_adapter, end, avail - 2, 2); - if (gst_rtp_buffer_get_marker (buf)) { - guint avail; - guint8 end[2]; - guint8 *data; + /* now append the tile packets to the frame */ + packets = gst_adapter_take_list (rtpj2kdepay->t_adapter, avail); + for (walk = packets; walk; walk = g_list_next (walk)) { + GstBuffer *buf = GST_BUFFER_CAST (walk->data); - /* last buffer take all data out of the adapter */ - avail = gst_adapter_available (rtpj2kdepay->adapter); - GST_DEBUG_OBJECT (rtpj2kdepay, "marker set, last buffer"); + if (walk == packets) { + guint8 *data; + guint size; + /* first buffer should contain the SOT */ + data = GST_BUFFER_DATA (buf); + size = GST_BUFFER_SIZE (buf); + + if (size < 12) + goto invalid_tile; + + if (data[0] == 0xff && data[1] == J2K_MARKER_SOT) { + guint Psot, nPsot; + + if (end[0] == 0xff && end[1] == J2K_MARKER_EOC) + nPsot = avail - 2; + else + nPsot = avail; + + Psot = GST_READ_UINT32_BE (&data[6]); + if (Psot != nPsot && Psot != 0) { + /* Psot must match the size of the tile */ + GST_DEBUG_OBJECT (rtpj2kdepay, "set Psot from %u to %u", Psot, nPsot); + buf = gst_buffer_make_writable (buf); + data = GST_BUFFER_DATA (buf); + GST_WRITE_UINT32_BE (&data[6], nPsot); + } + } + } + + GST_DEBUG_OBJECT (rtpj2kdepay, "append pu packet of size %u", + GST_BUFFER_SIZE (buf)); + gst_adapter_push (rtpj2kdepay->f_adapter, buf); + } + g_list_free (packets); + +done: + rtpj2kdepay->last_tile = -1; + + return ret; + + /* errors */ +waiting_header: + { + GST_DEBUG_OBJECT (rtpj2kdepay, "waiting for header %u", mh_id); + gst_adapter_clear (rtpj2kdepay->t_adapter); + rtpj2kdepay->last_tile = -1; + return ret; + } +invalid_tile: + { + GST_ELEMENT_WARNING (rtpj2kdepay, STREAM, DECODE, ("Invalid tile"), (NULL)); + gst_adapter_clear (rtpj2kdepay->t_adapter); + rtpj2kdepay->last_tile = -1; + return ret; + } +} + +static GstFlowReturn +gst_rtp_j2k_depay_flush_frame (GstBaseRTPDepayload * depayload) +{ + GstRtpJ2KDepay *rtpj2kdepay; + guint8 end[2]; + guint8 *data; + guint avail; + GstBuffer *outbuf; + GstFlowReturn ret = GST_FLOW_OK; + + rtpj2kdepay = GST_RTP_J2K_DEPAY (depayload); + + /* flush pending tile */ + gst_rtp_j2k_depay_flush_tile (depayload); + + /* last buffer take all data out of the adapter */ + avail = gst_adapter_available (rtpj2kdepay->f_adapter); + if (avail == 0) + goto done; + + if (avail > 2) { /* take the last bytes of the JPEG 2000 data to see if there is an EOC * marker */ - gst_adapter_copy (rtpj2kdepay->adapter, end, avail - 2, 2); + gst_adapter_copy (rtpj2kdepay->f_adapter, end, avail - 2, 2); if (end[0] != 0xff && end[1] != 0xd9) { GST_DEBUG_OBJECT (rtpj2kdepay, "no EOC marker, adding one"); @@ -211,15 +377,166 @@ gst_rtp_j2k_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf) data[0] = 0xff; data[1] = 0xd9; - gst_adapter_push (rtpj2kdepay->adapter, outbuf); + gst_adapter_push (rtpj2kdepay->f_adapter, outbuf); avail += 2; } - outbuf = gst_adapter_take_buffer (rtpj2kdepay->adapter, avail); - GST_DEBUG_OBJECT (rtpj2kdepay, "returning %u bytes", avail); + GST_DEBUG_OBJECT (rtpj2kdepay, "pushing %u bytes", avail); + outbuf = gst_adapter_take_buffer (rtpj2kdepay->f_adapter, avail); + ret = gst_base_rtp_depayload_push (depayload, outbuf); + } else { + GST_WARNING_OBJECT (rtpj2kdepay, "empty packet"); + gst_adapter_clear (rtpj2kdepay->f_adapter); } - return outbuf; + /* we accept any mh_id now */ + rtpj2kdepay->last_mh_id = -1; + + /* reset state */ + rtpj2kdepay->next_frag = 0; + rtpj2kdepay->have_sync = FALSE; + +done: + /* we can't keep headers with mh_id of 0 */ + store_mheader (rtpj2kdepay, 0, NULL); + + return ret; +} + +static GstBuffer * +gst_rtp_j2k_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf) +{ + GstRtpJ2KDepay *rtpj2kdepay; + guint8 *payload; + guint MHF, mh_id, frag_offset, tile, payload_len, j2klen; + gint gap; + guint32 rtptime; + + rtpj2kdepay = GST_RTP_J2K_DEPAY (depayload); + + payload = gst_rtp_buffer_get_payload (buf); + payload_len = gst_rtp_buffer_get_payload_len (buf); + + /* we need at least a header */ + if (payload_len < 8) + goto empty_packet; + + rtptime = gst_rtp_buffer_get_timestamp (buf); + + /* new timestamp marks new frame */ + if (rtpj2kdepay->last_rtptime != rtptime) { + rtpj2kdepay->last_rtptime = rtptime; + /* flush pending frame */ + gst_rtp_j2k_depay_flush_frame (depayload); + } + + /* + * 0 1 2 3 + * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 + * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + * |tp |MHF|mh_id|T| priority | tile number | + * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + * |reserved | fragment offset | + * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + */ + MHF = (payload[0] & 0x30) >> 4; + mh_id = (payload[0] & 0xe) >> 1; + + if (rtpj2kdepay->last_mh_id == -1) + rtpj2kdepay->last_mh_id = mh_id; + else if (rtpj2kdepay->last_mh_id != mh_id) + goto wrong_mh_id; + + tile = (payload[2] << 8) | payload[3]; + frag_offset = (payload[5] << 16) | (payload[6] << 8) | payload[7]; + j2klen = payload_len - 8; + + GST_DEBUG_OBJECT (rtpj2kdepay, "MHF %u, tile %u, frag %u, expected %u", MHF, + tile, frag_offset, rtpj2kdepay->next_frag); + + /* calculate the gap between expected frag */ + gap = frag_offset - rtpj2kdepay->next_frag; + /* calculate next frag */ + rtpj2kdepay->next_frag = frag_offset + j2klen; + + if (gap != 0) { + GST_DEBUG_OBJECT (rtpj2kdepay, "discont of %d, clear PU", gap); + /* discont, clear pu adapter and resync */ + gst_rtp_j2k_depay_clear_pu (rtpj2kdepay); + } + + /* check for sync code */ + if (j2klen > 2 && payload[8] == 0xff) { + guint marker = payload[9]; + + /* packets must start with SOC, SOT or SOP */ + switch (marker) { + case J2K_MARKER_SOC: + GST_DEBUG_OBJECT (rtpj2kdepay, "found SOC packet"); + /* flush the previous frame, should have happened when the timestamp + * changed above. */ + gst_rtp_j2k_depay_flush_frame (depayload); + rtpj2kdepay->have_sync = TRUE; + break; + case J2K_MARKER_SOT: + /* flush the previous tile */ + gst_rtp_j2k_depay_flush_tile (depayload); + GST_DEBUG_OBJECT (rtpj2kdepay, "found SOT packet"); + rtpj2kdepay->have_sync = TRUE; + /* we sync on the tile now */ + rtpj2kdepay->last_tile = tile; + break; + case J2K_MARKER_SOP: + GST_DEBUG_OBJECT (rtpj2kdepay, "found SOP packet"); + /* flush the previous PU */ + gst_rtp_j2k_depay_flush_pu (depayload); + if (rtpj2kdepay->last_tile != tile) { + /* wrong tile, we lose sync and we need a new SOT or SOC to regain + * sync. First flush out the previous tile if we have one. */ + if (rtpj2kdepay->last_tile != -1) + gst_rtp_j2k_depay_flush_tile (depayload); + /* now we have no more valid tile and no sync */ + rtpj2kdepay->last_tile = -1; + rtpj2kdepay->have_sync = FALSE; + } else { + rtpj2kdepay->have_sync = TRUE; + } + break; + default: + GST_DEBUG_OBJECT (rtpj2kdepay, "no sync packet 0x%02d", marker); + break; + } + } + + if (rtpj2kdepay->have_sync) { + GstBuffer *pu_frag; + + if (gst_adapter_available (rtpj2kdepay->pu_adapter) == 0) { + /* first part of pu, record state */ + GST_DEBUG_OBJECT (rtpj2kdepay, "first PU"); + rtpj2kdepay->pu_MHF = MHF; + } + /* and push in pu adapter */ + GST_DEBUG_OBJECT (rtpj2kdepay, "push pu of size %u in adapter", j2klen); + pu_frag = gst_rtp_buffer_get_payload_subbuffer (buf, 8, -1); + gst_adapter_push (rtpj2kdepay->pu_adapter, pu_frag); + + if (MHF & 2) { + /* last part of main header received, we can flush it */ + GST_DEBUG_OBJECT (rtpj2kdepay, "header end, flush pu"); + gst_rtp_j2k_depay_flush_pu (depayload); + } + } else { + GST_DEBUG_OBJECT (rtpj2kdepay, "discard packet, no sync"); + } + + /* marker bit finishes the frame */ + if (gst_rtp_buffer_get_marker (buf)) { + GST_DEBUG_OBJECT (rtpj2kdepay, "marker set, last buffer"); + /* then flush frame */ + gst_rtp_j2k_depay_flush_frame (depayload); + } + return NULL; /* ERRORS */ empty_packet: @@ -228,9 +545,12 @@ empty_packet: ("Empty Payload."), (NULL)); return NULL; } -waiting_header: +wrong_mh_id: { - GST_DEBUG_OBJECT (rtpj2kdepay, "we are waiting for a header"); + GST_ELEMENT_WARNING (rtpj2kdepay, STREAM, DECODE, + ("Invalid mh_id %u, expected %u", mh_id, rtpj2kdepay->last_mh_id), + (NULL)); + gst_rtp_j2k_depay_clear_pu (rtpj2kdepay); return NULL; } } @@ -247,8 +567,7 @@ gst_rtp_j2k_depay_change_state (GstElement * element, GstStateChange transition) case GST_STATE_CHANGE_NULL_TO_READY: break; case GST_STATE_CHANGE_READY_TO_PAUSED: - gst_adapter_clear (rtpj2kdepay->adapter); - rtpj2kdepay->need_header = TRUE; + gst_rtp_j2k_depay_reset (rtpj2kdepay); break; default: break; @@ -258,7 +577,7 @@ gst_rtp_j2k_depay_change_state (GstElement * element, GstStateChange transition) switch (transition) { case GST_STATE_CHANGE_PAUSED_TO_READY: - gst_adapter_clear (rtpj2kdepay->adapter); + gst_rtp_j2k_depay_reset (rtpj2kdepay); break; case GST_STATE_CHANGE_READY_TO_NULL: break; diff --git a/gst/rtp/gstrtpj2kdepay.h b/gst/rtp/gstrtpj2kdepay.h index 41120af42e..e855eef940 100644 --- a/gst/rtp/gstrtpj2kdepay.h +++ b/gst/rtp/gstrtpj2kdepay.h @@ -44,8 +44,19 @@ struct _GstRtpJ2KDepay { GstBaseRTPDepayload depayload; - GstAdapter *adapter; - gboolean need_header; + guint64 last_rtptime; + guint last_mh_id; + guint last_tile; + + GstBuffer *MH[8]; + + guint pu_MHF; + GstAdapter *pu_adapter; + GstAdapter *t_adapter; + GstAdapter *f_adapter; + + guint next_frag; + gboolean have_sync; gint width, height; }; From 52b5929a2b7d4c0822ccd9611de28f7b7e2ffe9e Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Mon, 25 Oct 2010 11:51:06 +0200 Subject: [PATCH 007/351] rtspsrc: add and use auto buffering mode ... which selects BUFFER for a non-live stream, and otherwise SLAVE. Fixes #633088. --- gst/rtsp/gstrtspsrc.c | 52 ++++++++++++++++++++++++++++++++++++++----- 1 file changed, 46 insertions(+), 6 deletions(-) diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c index dc8655f1cf..f746a2a0d9 100644 --- a/gst/rtsp/gstrtspsrc.c +++ b/gst/rtsp/gstrtspsrc.c @@ -129,15 +129,24 @@ enum LAST_SIGNAL }; +enum _GstRtspSrcBufferMode +{ + BUFFER_MODE_NONE, + BUFFER_MODE_SLAVE, + BUFFER_MODE_BUFFER, + BUFFER_MODE_AUTO +}; + #define GST_TYPE_RTSP_SRC_BUFFER_MODE (gst_rtsp_src_buffer_mode_get_type()) static GType gst_rtsp_src_buffer_mode_get_type (void) { static GType buffer_mode_type = 0; static const GEnumValue buffer_modes[] = { - {0, "Only use RTP timestamps", "none"}, - {1, "Slave receiver to sender clock", "slave"}, - {2, "Do low/high watermark buffering", "buffer"}, + {BUFFER_MODE_NONE, "Only use RTP timestamps", "none"}, + {BUFFER_MODE_SLAVE, "Slave receiver to sender clock", "slave"}, + {BUFFER_MODE_BUFFER, "Do low/high watermark buffering", "buffer"}, + {BUFFER_MODE_AUTO, "Choose mode depending on stream live", "auto"}, {0, NULL, NULL}, }; @@ -163,7 +172,7 @@ gst_rtsp_src_buffer_mode_get_type (void) #define DEFAULT_RTP_BLOCKSIZE 0 #define DEFAULT_USER_ID NULL #define DEFAULT_USER_PW NULL -#define DEFAULT_BUFFER_MODE 1 +#define DEFAULT_BUFFER_MODE BUFFER_MODE_AUTO #define DEFAULT_PORT_RANGE NULL enum @@ -2330,8 +2339,39 @@ gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream, g_object_set (src->session, "latency", src->latency, NULL); klass = G_OBJECT_GET_CLASS (G_OBJECT (src->session)); - if (g_object_class_find_property (klass, "buffer-mode")) - g_object_set (src->session, "buffer-mode", src->buffer_mode, NULL); + if (g_object_class_find_property (klass, "buffer-mode")) { + if (src->buffer_mode != BUFFER_MODE_AUTO) { + g_object_set (src->session, "buffer-mode", src->buffer_mode, NULL); + } else { + gboolean need_slave; + GstStructure *s; + const gchar *encoding; + + /* buffer mode pauses are handled by adding offsets to buffer times, + * but some depayloaders may have a hard time syncing output times + * with such input times, e.g. container ones, most notably ASF */ + /* TODO alternatives are having an event that indicates these shifts, + * or having rtsp extensions provide suggestion on buffer mode */ + need_slave = stream->container; + if (stream->caps && (s = gst_caps_get_structure (stream->caps, 0)) && + (encoding = gst_structure_get_string (s, "encoding-name"))) + need_slave = need_slave || (strcmp (encoding, "X-ASF-PF") == 0); + GST_DEBUG_OBJECT (src, "auto buffering mode, need_slave %d", + need_slave); + /* valid duration implies not likely live pipeline, + * so slaving in jitterbuffer does not make much sense + * (and might mess things up due to bursts) */ + if (GST_CLOCK_TIME_IS_VALID (src->segment.duration) && + src->segment.duration && !need_slave) { + GST_DEBUG_OBJECT (src, "selected buffer"); + g_object_set (src->session, "buffer-mode", BUFFER_MODE_BUFFER, + NULL); + } else { + GST_DEBUG_OBJECT (src, "selected slave"); + g_object_set (src->session, "buffer-mode", BUFFER_MODE_SLAVE, NULL); + } + } + } /* connect to signals if we did not already do so */ GST_DEBUG_OBJECT (src, "connect to signals on session manager, stream %p", From c25625c31cd75dc922600f7632334dc1710cc4ea Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Thu, 2 Dec 2010 16:08:34 +0100 Subject: [PATCH 008/351] rtspsrc: degrade gracefully upon failing seek and tweak QUERY_SEEKING response --- gst/rtsp/gstrtspsrc.c | 43 +++++++++++++++++++++++++++++++++++++++++++ gst/rtsp/gstrtspsrc.h | 1 + 2 files changed, 44 insertions(+) diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c index f746a2a0d9..9f8edb8037 100644 --- a/gst/rtsp/gstrtspsrc.c +++ b/gst/rtsp/gstrtspsrc.c @@ -2033,6 +2033,11 @@ gst_rtspsrc_handle_src_query (GstPad * pad, GstQuery * query) gboolean seekable = src->cur_protocols != GST_RTSP_LOWER_TRANS_UDP_MCAST; + /* seeking without duration is unlikely */ + seekable = seekable && src->seekable && src->segment.duration && + GST_CLOCK_TIME_IS_VALID (src->segment.duration); + + /* FIXME ?? should we have 0 and segment.duration here; see demuxers */ gst_query_set_seeking (query, GST_FORMAT_TIME, seekable, src->segment.start, src->segment.stop); res = TRUE; @@ -4604,6 +4609,8 @@ gst_rtspsrc_parse_methods (GstRTSPSrc * src, GstRTSPMessage * response) /* always assume PLAY, FIXME, extensions should be able to override * this */ src->methods |= GST_RTSP_PLAY; + /* also assume it will support Range */ + src->seekable = TRUE; /* we need describe and setup */ if (!(src->methods & GST_RTSP_DESCRIBE)) @@ -5777,6 +5784,20 @@ gen_range_header (GstRTSPSrc * src, GstSegment * segment) return res; } +static void +clear_rtp_base (GstRTSPSrc * src, GstRTSPStream * stream) +{ + stream->timebase = -1; + stream->seqbase = -1; + if (stream->caps) { + GstStructure *s; + + stream->caps = gst_caps_make_writable (stream->caps); + s = gst_caps_get_structure (stream->caps, 0); + gst_structure_remove_fields (s, "clock-base", "seqnum-base", NULL); + } +} + static gboolean gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment) { @@ -5861,6 +5882,28 @@ gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment) if (gst_rtspsrc_send (src, conn, &request, &response, NULL) < 0) goto send_error; + /* seek may have silently failed as it is not supported */ + if (!(src->methods & GST_RTSP_PLAY)) { + GST_DEBUG_OBJECT (src, "PLAY Range not supported; re-enable PLAY"); + /* obviously it is supported as we made it here */ + src->methods |= GST_RTSP_PLAY; + src->seekable = FALSE; + /* but there is nothing to parse in the response, + * so convey we have no idea and not to expect anything particular */ + clear_rtp_base (src, stream); + if (control) { + GList *run; + + /* need to do for all streams */ + for (run = src->streams; run; run = g_list_next (run)) + clear_rtp_base (src, (GstRTSPStream *) run->data); + } + /* NOTE the above also disables npt based eos detection */ + /* and below forces position to 0, + * which is visible feedback we lost the plot */ + segment->start = segment->last_stop = 0; + } + gst_rtsp_message_unset (&request); /* parse RTP npt field. This is the current position in the stream (Normal diff --git a/gst/rtsp/gstrtspsrc.h b/gst/rtsp/gstrtspsrc.h index b1402c2aaa..bdaff49b0d 100644 --- a/gst/rtsp/gstrtspsrc.h +++ b/gst/rtsp/gstrtspsrc.h @@ -230,6 +230,7 @@ struct _GstRTSPSrc { /* supported methods */ gint methods; + gboolean seekable; /* session management */ GstElement *session; From 287894a89a1e075e152d6a337e8021c407127289 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Mon, 8 Feb 2010 21:41:29 +0100 Subject: [PATCH 009/351] rtspsrc: mark DISCONT when resuming PLAY In particular, when streaming interleaved, this arranges for setting a new timestamp on outgoing buffer so downstream can appropriate reset to a change in (rtp)time. --- gst/rtsp/gstrtspsrc.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c index 9f8edb8037..ced8ae5759 100644 --- a/gst/rtsp/gstrtspsrc.c +++ b/gst/rtsp/gstrtspsrc.c @@ -5960,6 +5960,13 @@ gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment) gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, FALSE); gst_task_start (src->task); + /* mark discont */ + GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position"); + for (walk = src->streams; walk; walk = g_list_next (walk)) { + GstRTSPStream *stream = (GstRTSPStream *) walk->data; + stream->discont = TRUE; + } + done: GST_RTSP_STATE_UNLOCK (src); From bead93ab7b0cb24071d0ede8eca368d08834e7f9 Mon Sep 17 00:00:00 2001 From: Jan Schmidt Date: Fri, 29 Oct 2010 22:50:14 +0100 Subject: [PATCH 010/351] pulsemixer: Implement MIXER_FLAG_AUTO_NOTIFICATIONS Add the mixer flag and send notifications when either the volume or muted status changes. https://bugzilla.gnome.org/show_bug.cgi?id=618389 --- ext/pulse/pulsemixerctrl.c | 48 +++++++++++++++++++++++++++++++++++--- ext/pulse/pulsemixerctrl.h | 12 ++++++++++ 2 files changed, 57 insertions(+), 3 deletions(-) diff --git a/ext/pulse/pulsemixerctrl.c b/ext/pulse/pulsemixerctrl.c index be5b536797..220c8d0655 100644 --- a/ext/pulse/pulsemixerctrl.c +++ b/ext/pulse/pulsemixerctrl.c @@ -61,6 +61,8 @@ gst_pulsemixer_ctrl_sink_info_cb (pa_context * context, const pa_sink_info * i, int eol, void *userdata) { GstPulseMixerCtrl *c = userdata; + gboolean vol_chg = FALSE; + gboolean old_mute; /* Called from the background thread! */ @@ -90,8 +92,10 @@ gst_pulsemixer_ctrl_sink_info_cb (pa_context * context, const pa_sink_info * i, c->description = g_strdup (i->description); c->index = i->index; c->channel_map = i->channel_map; + vol_chg = !pa_cvolume_equal (&c->volume, &i->volume); c->volume = i->volume; - c->muted = ! !i->mute; + old_mute = c->muted; + c->muted = !!i->mute; c->type = GST_PULSEMIXER_SINK; if (c->track) { @@ -104,6 +108,19 @@ gst_pulsemixer_ctrl_sink_info_cb (pa_context * context, const pa_sink_info * i, c->operation_success = TRUE; pa_threaded_mainloop_signal (c->mainloop, 0); + + if (vol_chg && c->track) { + gint volumes[PA_CHANNELS_MAX]; + gint i; + for (i = 0; i < c->volume.channels; i++) + volumes[i] = (gint) (c->volume.values[i]); + GST_LOG_OBJECT (c->object, "Sending volume change notification"); + gst_mixer_volume_changed (GST_MIXER (c->object), c->track, volumes); + } + if ((c->muted != old_mute) && c->track) { + GST_LOG_OBJECT (c->object, "Sending mute toggled notification"); + gst_mixer_mute_toggled (GST_MIXER (c->object), c->track, c->muted); + } } static void @@ -111,6 +128,8 @@ gst_pulsemixer_ctrl_source_info_cb (pa_context * context, const pa_source_info * i, int eol, void *userdata) { GstPulseMixerCtrl *c = userdata; + gboolean vol_chg = FALSE; + gboolean old_mute; /* Called from the background thread! */ @@ -140,8 +159,10 @@ gst_pulsemixer_ctrl_source_info_cb (pa_context * context, c->description = g_strdup (i->description); c->index = i->index; c->channel_map = i->channel_map; + vol_chg = !pa_cvolume_equal (&c->volume, &i->volume); c->volume = i->volume; - c->muted = ! !i->mute; + old_mute = c->muted; + c->muted = !!i->mute; c->type = GST_PULSEMIXER_SOURCE; if (c->track) { @@ -154,6 +175,19 @@ gst_pulsemixer_ctrl_source_info_cb (pa_context * context, c->operation_success = TRUE; pa_threaded_mainloop_signal (c->mainloop, 0); + + if (vol_chg && c->track) { + gint volumes[PA_CHANNELS_MAX]; + gint i; + for (i = 0; i < c->volume.channels; i++) + volumes[i] = (gint) (c->volume.values[i]); + GST_LOG_OBJECT (c->object, "Sending volume change notification"); + gst_mixer_volume_changed (GST_MIXER (c->object), c->track, volumes); + } + if ((c->muted != old_mute) && c->track) { + GST_LOG_OBJECT (c->object, "Sending mute toggled notification"); + gst_mixer_mute_toggled (GST_MIXER (c->object), c->track, c->muted); + } } static void @@ -195,7 +229,7 @@ gst_pulsemixer_ctrl_success_cb (pa_context * context, int success, { GstPulseMixerCtrl *c = (GstPulseMixerCtrl *) userdata; - c->operation_success = ! !success; + c->operation_success = !!success; pa_threaded_mainloop_signal (c->mainloop, 0); } @@ -395,6 +429,8 @@ gst_pulsemixer_ctrl_new (GObject * object, const gchar * server, const gchar * device, GstPulseMixerType type) { GstPulseMixerCtrl *c = NULL; + g_return_val_if_fail (G_TYPE_CHECK_INSTANCE_TYPE ((object), + GST_TYPE_MIXER), c); GST_DEBUG_OBJECT (object, "new mixer ctrl for %s", device); c = g_new (GstPulseMixerCtrl, 1); @@ -596,3 +632,9 @@ gst_pulsemixer_ctrl_set_mute (GstPulseMixerCtrl * c, GstMixerTrack * track, pa_threaded_mainloop_unlock (c->mainloop); } + +GstMixerFlags +gst_pulsemixer_ctrl_get_mixer_flags (GstPulseMixerCtrl * mixer) +{ + return GST_MIXER_FLAG_AUTO_NOTIFICATIONS; +} diff --git a/ext/pulse/pulsemixerctrl.h b/ext/pulse/pulsemixerctrl.h index e6b67ad266..c1d1e85285 100644 --- a/ext/pulse/pulsemixerctrl.h +++ b/ext/pulse/pulsemixerctrl.h @@ -90,6 +90,7 @@ void gst_pulsemixer_ctrl_set_mute (GstPulseMixerCtrl * mixer, GstMixerTrack * track, gboolean mute); void gst_pulsemixer_ctrl_set_record (GstPulseMixerCtrl * mixer, GstMixerTrack * track, gboolean record); +GstMixerFlags gst_pulsemixer_ctrl_get_mixer_flags (GstPulseMixerCtrl * mixer); #define GST_IMPLEMENT_PULSEMIXER_CTRL_METHODS(Type, interface_as_function) \ static const GList* \ @@ -146,6 +147,16 @@ interface_as_function ## _set_mute (GstMixer * mixer, GstMixerTrack * track, \ gst_pulsemixer_ctrl_set_mute (this->mixer, track, mute); \ } \ +static GstMixerFlags \ +interface_as_function ## _get_mixer_flags (GstMixer * mixer) \ +{ \ + Type *this = (Type*) mixer; \ + \ + g_return_val_if_fail (this != NULL, GST_MIXER_FLAG_NONE); \ + g_return_val_if_fail (this->mixer != NULL, GST_MIXER_FLAG_NONE); \ + \ + return gst_pulsemixer_ctrl_get_mixer_flags (this->mixer); \ +} \ static void \ interface_as_function ## _mixer_interface_init (GstMixerClass * klass) \ { \ @@ -156,6 +167,7 @@ interface_as_function ## _mixer_interface_init (GstMixerClass * klass) klass->get_volume = interface_as_function ## _get_volume; \ klass->set_mute = interface_as_function ## _set_mute; \ klass->set_record = interface_as_function ## _set_record; \ + klass->get_mixer_flags = interface_as_function ## _get_mixer_flags; \ } G_END_DECLS From d73440cb976d8b2ae51f231f19c10d7eaddb6074 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Thu, 1 Jul 2010 00:22:07 +0100 Subject: [PATCH 011/351] matroska-demux: put unrecognised RIFF format IDs into the unknown caps Extra info can't hurt. Field names aren't necessarily consistent with what's used elsewhere though (e.g. avidemux), but then neither are the caps. https://bugzilla.gnome.org/show_bug.cgi?id=623178 --- gst/matroska/matroska-demux.c | 52 +++++++++++++++++++++++++++-------- 1 file changed, 41 insertions(+), 11 deletions(-) diff --git a/gst/matroska/matroska-demux.c b/gst/matroska/matroska-demux.c index 3bc8cd4f8d..7a0eb3183b 100644 --- a/gst/matroska/matroska-demux.c +++ b/gst/matroska/matroska-demux.c @@ -165,11 +165,11 @@ static GstIndex *gst_matroska_demux_get_index (GstElement * element); /* caps functions */ static GstCaps *gst_matroska_demux_video_caps (GstMatroskaTrackVideoContext - * videocontext, - const gchar * codec_id, guint8 * data, guint size, gchar ** codec_name); + * videocontext, const gchar * codec_id, guint8 * data, guint size, + gchar ** codec_name, guint32 * riff_fourcc); static GstCaps *gst_matroska_demux_audio_caps (GstMatroskaTrackAudioContext - * audiocontext, - const gchar * codec_id, guint8 * data, guint size, gchar ** codec_name); + * audiocontext, const gchar * codec_id, guint8 * data, guint size, + gchar ** codec_name, guint16 * riff_audio_fmt); static GstCaps * gst_matroska_demux_subtitle_caps (GstMatroskaTrackSubtitleContext * subtitlecontext, const gchar * codec_id, gpointer data, guint size); @@ -1173,7 +1173,8 @@ gst_matroska_demux_add_stream (GstMatroskaDemux * demux, GstEbmlRead * ebml) GstCaps *caps = NULL; gchar *padname = NULL; GstFlowReturn ret; - guint32 id; + guint32 id, riff_fourcc = 0; + guint16 riff_audio_fmt = 0; GstTagList *list = NULL; gchar *codec = NULL; @@ -1852,8 +1853,9 @@ gst_matroska_demux_add_stream (GstMatroskaDemux * demux, GstEbmlRead * ebml) padname = g_strdup_printf ("video_%02d", demux->num_v_streams++); templ = gst_element_class_get_pad_template (klass, "video_%02d"); caps = gst_matroska_demux_video_caps (videocontext, - context->codec_id, - (guint8 *) context->codec_priv, context->codec_priv_size, &codec); + context->codec_id, (guint8 *) context->codec_priv, + context->codec_priv_size, &codec, &riff_fourcc); + if (codec) { list = gst_tag_list_new (); gst_tag_list_add (list, GST_TAG_MERGE_REPLACE, @@ -1870,8 +1872,9 @@ gst_matroska_demux_add_stream (GstMatroskaDemux * demux, GstEbmlRead * ebml) padname = g_strdup_printf ("audio_%02d", demux->num_a_streams++); templ = gst_element_class_get_pad_template (klass, "audio_%02d"); caps = gst_matroska_demux_audio_caps (audiocontext, - context->codec_id, - context->codec_priv, context->codec_priv_size, &codec); + context->codec_id, context->codec_priv, context->codec_priv_size, + &codec, &riff_audio_fmt); + if (codec) { list = gst_tag_list_new (); gst_tag_list_add (list, GST_TAG_MERGE_REPLACE, @@ -1940,6 +1943,12 @@ gst_matroska_demux_add_stream (GstMatroskaDemux * demux, GstEbmlRead * ebml) } gst_caps_set_simple (caps, "codec-id", G_TYPE_STRING, context->codec_id, NULL); + + /* add any unrecognised riff fourcc / audio format, but after codec-id */ + if (context->type == GST_MATROSKA_TRACK_TYPE_AUDIO && riff_audio_fmt != 0) + gst_caps_set_simple (caps, "format", G_TYPE_INT, riff_audio_fmt, NULL); + else if (context->type == GST_MATROSKA_TRACK_TYPE_VIDEO && riff_fourcc != 0) + gst_caps_set_simple (caps, "fourcc", GST_TYPE_FOURCC, riff_fourcc, NULL); } /* the pad in here */ @@ -6286,7 +6295,7 @@ gst_matroska_demux_sink_activate_pull (GstPad * sinkpad, gboolean active) static GstCaps * gst_matroska_demux_video_caps (GstMatroskaTrackVideoContext * videocontext, const gchar * codec_id, guint8 * data, guint size, - gchar ** codec_name) + gchar ** codec_name, guint32 * riff_fourcc) { GstMatroskaTrackContext *context = (GstMatroskaTrackContext *) videocontext; GstCaps *caps = NULL; @@ -6298,6 +6307,9 @@ gst_matroska_demux_video_caps (GstMatroskaTrackVideoContext * context->send_flac_headers = FALSE; context->send_speex_headers = FALSE; + if (riff_fourcc) + *riff_fourcc = 0; + /* TODO: check if we have all codec types from matroska-ids.h * check if we have to do more special things with codec_private * @@ -6344,9 +6356,17 @@ gst_matroska_demux_video_caps (GstMatroskaTrackVideoContext * GST_BUFFER_SIZE (buf)); } + if (riff_fourcc) + *riff_fourcc = vids->compression; + caps = gst_riff_create_video_caps (vids->compression, NULL, vids, buf, NULL, codec_name); + if (caps == NULL) { + GST_WARNING ("Unhandled RIFF fourcc %" GST_FOURCC_FORMAT, + GST_FOURCC_ARGS (vids->compression)); + } + if (buf) gst_buffer_unref (buf); @@ -6652,7 +6672,7 @@ aac_profile_idx (const gchar * codec_id) static GstCaps * gst_matroska_demux_audio_caps (GstMatroskaTrackAudioContext * audiocontext, const gchar * codec_id, guint8 * data, guint size, - gchar ** codec_name) + gchar ** codec_name, guint16 * riff_audio_fmt) { GstMatroskaTrackContext *context = (GstMatroskaTrackContext *) audiocontext; GstCaps *caps = NULL; @@ -6660,6 +6680,9 @@ gst_matroska_demux_audio_caps (GstMatroskaTrackAudioContext * g_assert (audiocontext != NULL); g_assert (codec_name != NULL); + if (riff_audio_fmt) + *riff_audio_fmt = 0; + context->send_xiph_headers = FALSE; context->send_flac_headers = FALSE; context->send_speex_headers = FALSE; @@ -6753,9 +6776,16 @@ gst_matroska_demux_audio_caps (GstMatroskaTrackAudioContext * /* 18 is the waveformatex size */ gst_buffer_set_data (codec_data, data + 18, auds.size); + if (riff_audio_fmt) + *riff_audio_fmt = auds.format; + caps = gst_riff_create_audio_caps (auds.format, NULL, &auds, NULL, codec_data, codec_name); gst_buffer_unref (codec_data); + + if (caps == NULL) { + GST_WARNING ("Unhandled RIFF audio format 0x%02x", auds.format); + } } } else if (g_str_has_prefix (codec_id, GST_MATROSKA_CODEC_ID_AUDIO_AAC)) { GstBuffer *priv = NULL; From 5d9c8996da5e86a2faf66c8e94fb9524efbeb7fb Mon Sep 17 00:00:00 2001 From: Kishore Arepalli Date: Sat, 11 Dec 2010 17:46:17 +0100 Subject: [PATCH 012/351] qtdemux: Implement CONVERT and FORMATS query Fixes bug #636784. --- gst/qtdemux/qtdemux.c | 127 +++++++++++++++++++++++++++++++++--------- 1 file changed, 101 insertions(+), 26 deletions(-) diff --git a/gst/qtdemux/qtdemux.c b/gst/qtdemux/qtdemux.c index a744fbc000..5a6fb7e1f7 100644 --- a/gst/qtdemux/qtdemux.c +++ b/gst/qtdemux/qtdemux.c @@ -387,6 +387,13 @@ GST_BOILERPLATE (GstQTDemux, gst_qtdemux, GstQTDemux, GST_TYPE_ELEMENT); static void gst_qtdemux_dispose (GObject * object); +static guint32 +gst_qtdemux_find_index_linear (GstQTDemux * qtdemux, QtDemuxStream * str, + guint64 media_time); +static guint32 +gst_qtdemux_find_index_for_given_media_offset_linear (GstQTDemux * qtdemux, + QtDemuxStream * str, gint64 media_offset); + static void gst_qtdemux_set_index (GstElement * element, GstIndex * index); static GstIndex *gst_qtdemux_get_index (GstElement * element); static GstStateChangeReturn gst_qtdemux_change_state (GstElement * element, @@ -561,47 +568,57 @@ gst_qtdemux_pull_atom (GstQTDemux * qtdemux, guint64 offset, guint64 size, return flow; } -#if 0 +#if 1 static gboolean gst_qtdemux_src_convert (GstPad * pad, GstFormat src_format, gint64 src_value, - GstFormat * dest_format, gint64 * dest_value) + GstFormat dest_format, gint64 * dest_value) { gboolean res = TRUE; QtDemuxStream *stream = gst_pad_get_element_private (pad); + GstQTDemux *qtdemux = GST_QTDEMUX (gst_pad_get_parent (pad)); + gint32 index; - if (stream->subtype == GST_MAKE_FOURCC ('v', 'i', 'd', 'e') && - (src_format == GST_FORMAT_BYTES || *dest_format == GST_FORMAT_BYTES)) + if (stream->subtype != FOURCC_vide) return FALSE; switch (src_format) { case GST_FORMAT_TIME: - switch (*dest_format) { - case GST_FORMAT_BYTES: - *dest_value = src_value * 1; /* FIXME */ - break; - case GST_FORMAT_DEFAULT: - *dest_value = src_value * 1; /* FIXME */ + switch (dest_format) { + case GST_FORMAT_BYTES:{ + index = gst_qtdemux_find_index_linear (qtdemux, stream, src_value); + if (-1 == index) + return FALSE; + + *dest_value = stream->samples[index].offset; + + GST_DEBUG_OBJECT (qtdemux, "Format Conversion Time->Offset :%" + GST_TIME_FORMAT "->%" G_GUINT64_FORMAT, + GST_TIME_ARGS (src_value), *dest_value); break; + } default: res = FALSE; break; } break; case GST_FORMAT_BYTES: - switch (*dest_format) { - case GST_FORMAT_TIME: - *dest_value = src_value * 1; /* FIXME */ - break; - default: - res = FALSE; - break; - } - break; - case GST_FORMAT_DEFAULT: - switch (*dest_format) { - case GST_FORMAT_TIME: - *dest_value = src_value * 1; /* FIXME */ + switch (dest_format) { + case GST_FORMAT_TIME:{ + index = + gst_qtdemux_find_index_for_given_media_offset_linear (qtdemux, + stream, src_value); + + if (-1 == index) + return FALSE; + + *dest_value = + gst_util_uint64_scale (stream->samples[index].timestamp, + GST_SECOND, stream->timescale); + GST_DEBUG_OBJECT (qtdemux, "Format Conversion Offset->Time :%" + G_GUINT64_FORMAT "->%" GST_TIME_FORMAT, + src_value, GST_TIME_ARGS (*dest_value)); break; + } default: res = FALSE; break; @@ -621,6 +638,8 @@ gst_qtdemux_get_src_query_types (GstPad * pad) static const GstQueryType src_types[] = { GST_QUERY_POSITION, GST_QUERY_DURATION, + GST_QUERY_CONVERT, + GST_QUERY_FORMATS, GST_QUERY_SEEKING, 0 }; @@ -675,6 +694,24 @@ gst_qtdemux_handle_src_query (GstPad * pad, GstQuery * query) } break; } + case GST_QUERY_CONVERT:{ + GstFormat src_fmt, dest_fmt; + gint64 src_value, dest_value; + + gst_query_parse_convert (query, &src_fmt, &src_value, &dest_fmt, NULL); + + res = gst_qtdemux_src_convert (pad, + src_fmt, src_value, dest_fmt, &dest_value); + if (res) { + gst_query_set_convert (query, src_fmt, src_value, dest_fmt, dest_value); + res = TRUE; + } + break; + } + case GST_QUERY_FORMATS: + gst_query_set_formats (query, 2, GST_FORMAT_TIME, GST_FORMAT_BYTES); + res = TRUE; + break; case GST_QUERY_SEEKING:{ GstFormat fmt; gboolean seekable; @@ -825,6 +862,44 @@ gst_qtdemux_find_index (GstQTDemux * qtdemux, QtDemuxStream * str, return index; } + + +/* find the index of the sample that includes the data for @media_offset using a + * linear search + * + * Returns the index of the sample. + */ +static guint32 +gst_qtdemux_find_index_for_given_media_offset_linear (GstQTDemux * qtdemux, + QtDemuxStream * str, gint64 media_offset) +{ + QtDemuxSample *result = str->samples; + guint32 index = 0; + + if (media_offset == result->offset) + return index; + + result++; + while (index < str->n_samples - 1) { + if (!qtdemux_parse_samples (qtdemux, str, index + 1)) + goto parse_failed; + + if (media_offset < result->offset) + break; + + index++; + result++; + } + return index; + + /* ERRORS */ +parse_failed: + { + GST_LOG_OBJECT (qtdemux, "Parsing of index %u failed!", index + 1); + return -1; + } +} + /* find the index of the sample that includes the data for @media_time using a * linear search, and keeping in mind that not all samples may have been parsed * yet. If possible, it will delegate to binary search. @@ -5123,7 +5198,7 @@ qtdemux_stbl_init (GstQTDemux * qtdemux, QtDemuxStream * stream, GNode * stbl) /* sync sample atom */ stream->stps_present = FALSE; if ((stream->stss_present = - !!qtdemux_tree_get_child_by_type_full (stbl, FOURCC_stss, + ! !qtdemux_tree_get_child_by_type_full (stbl, FOURCC_stss, &stream->stss) ? TRUE : FALSE) == TRUE) { /* copy atom data into a new buffer for later use */ stream->stss.data = g_memdup (stream->stss.data, stream->stss.size); @@ -5141,7 +5216,7 @@ qtdemux_stbl_init (GstQTDemux * qtdemux, QtDemuxStream * stream, GNode * stbl) /* partial sync sample atom */ if ((stream->stps_present = - !!qtdemux_tree_get_child_by_type_full (stbl, FOURCC_stps, + ! !qtdemux_tree_get_child_by_type_full (stbl, FOURCC_stps, &stream->stps) ? TRUE : FALSE) == TRUE) { /* copy atom data into a new buffer for later use */ stream->stps.data = g_memdup (stream->stps.data, stream->stps.size); @@ -5261,7 +5336,7 @@ qtdemux_stbl_init (GstQTDemux * qtdemux, QtDemuxStream * stream, GNode * stbl) /* composition time-to-sample */ if ((stream->ctts_present = - !!qtdemux_tree_get_child_by_type_full (stbl, FOURCC_ctts, + ! !qtdemux_tree_get_child_by_type_full (stbl, FOURCC_ctts, &stream->ctts) ? TRUE : FALSE) == TRUE) { /* copy atom data into a new buffer for later use */ stream->ctts.data = g_memdup (stream->ctts.data, stream->ctts.size); From e7ac799278398fb5f4f0aa5473449ad534fde422 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Sebastian=20Dr=C3=B6ge?= Date: Sat, 11 Dec 2010 17:49:03 +0100 Subject: [PATCH 013/351] qtdemux: Check if there's actually a seek table before parsing it --- gst/qtdemux/qtdemux.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/gst/qtdemux/qtdemux.c b/gst/qtdemux/qtdemux.c index 5a6fb7e1f7..dac0db10e2 100644 --- a/gst/qtdemux/qtdemux.c +++ b/gst/qtdemux/qtdemux.c @@ -876,6 +876,9 @@ gst_qtdemux_find_index_for_given_media_offset_linear (GstQTDemux * qtdemux, QtDemuxSample *result = str->samples; guint32 index = 0; + if (result == NULL || str->n_samples == 0) + return -1; + if (media_offset == result->offset) return index; From 100673bf14972e9bf67247f449fac92c1e3521ab Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Fri, 28 May 2010 15:27:14 +0100 Subject: [PATCH 014/351] jpegenc: free/malloc instead of realloc, avoids memcpy --- ext/jpeg/gstjpegenc.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/ext/jpeg/gstjpegenc.c b/ext/jpeg/gstjpegenc.c index 8f8196417d..e562867216 100644 --- a/ext/jpeg/gstjpegenc.c +++ b/ext/jpeg/gstjpegenc.c @@ -520,11 +520,12 @@ gst_jpegenc_resync (GstJpegEnc * jpegenc) jpegenc->h_samp[i], jpegenc->v_samp[i]); jpegenc->cinfo.comp_info[i].h_samp_factor = jpegenc->h_samp[i]; jpegenc->cinfo.comp_info[i].v_samp_factor = jpegenc->v_samp[i]; - jpegenc->line[i] = g_realloc (jpegenc->line[i], - jpegenc->v_max_samp * DCTSIZE * sizeof (char *)); + g_free (jpegenc->line[i]); + jpegenc->line[i] = g_new (guchar *, jpegenc->v_max_samp * DCTSIZE); if (!jpegenc->planar) { for (j = 0; j < jpegenc->v_max_samp * DCTSIZE; j++) { - jpegenc->row[i][j] = g_realloc (jpegenc->row[i][j], width); + g_free (jpegenc->row[i][j]); + jpegenc->row[i][j] = g_malloc (width); jpegenc->line[i][j] = jpegenc->row[i][j]; } } From addbc3c4ca29a6869fe21ea5b1228a496ac0f26a Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Sat, 11 Dec 2010 17:39:20 +0000 Subject: [PATCH 015/351] jpegdec: add "max-errors" property to ignore decoding errors Add property to ignore decoding errors. Default is to ignore a few decoding errors if the input is packetized, but error out immediately if the input is not packetized. Ignoring errors for packetized input most likely doesn't work properly yet, so don't do that for now. https://bugzilla.gnome.org/show_bug.cgi?id=623063 --- ext/jpeg/gstjpegdec.c | 147 ++++++++++++++++++++++++++++++++++++------ ext/jpeg/gstjpegdec.h | 12 ++++ 2 files changed, 140 insertions(+), 19 deletions(-) diff --git a/ext/jpeg/gstjpegdec.c b/ext/jpeg/gstjpegdec.c index 0518783e99..71ae4b9aee 100644 --- a/ext/jpeg/gstjpegdec.c +++ b/ext/jpeg/gstjpegdec.c @@ -52,11 +52,13 @@ (((struct GstJpegDecSourceMgr*)((cinfo_ptr)->src))->dec) #define JPEG_DEFAULT_IDCT_METHOD JDCT_FASTEST +#define JPEG_DEFAULT_MAX_ERRORS 0 enum { PROP_0, - PROP_IDCT_METHOD + PROP_IDCT_METHOD, + PROP_MAX_ERRORS }; /* *INDENT-OFF* */ @@ -192,6 +194,21 @@ gst_jpeg_dec_class_init (GstJpegDecClass * klass) JPEG_DEFAULT_IDCT_METHOD, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + /** + * GstJpegDec:max-errors + * + * Error out after receiving N consecutive decoding errors + * (-1 = never error out, 0 = automatic, 1 = fail on first error, etc.) + * + * Since: 0.10.27 + **/ + g_object_class_install_property (gobject_class, PROP_MAX_ERRORS, + g_param_spec_int ("max-errors", "Maximum Consecutive Decoding Errors", + "Error out after receiving N consecutive decoding errors " + "(-1 = never fail, 0 = automatic, 1 = fail on first error)", + -1, G_MAXINT, JPEG_DEFAULT_MAX_ERRORS, + G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_jpeg_dec_change_state); @@ -199,6 +216,81 @@ gst_jpeg_dec_class_init (GstJpegDecClass * klass) GST_DEBUG_CATEGORY_GET (GST_CAT_PERFORMANCE, "GST_PERFORMANCE"); } +static void +gst_jpeg_dec_clear_error (GstJpegDec * dec) +{ + g_free (dec->error_msg); + dec->error_msg = NULL; + dec->error_line = 0; + dec->error_func = NULL; +} + +static void +gst_jpeg_dec_set_error_va (GstJpegDec * dec, const gchar * func, gint line, + const gchar * debug_msg_format, va_list args) +{ +#ifndef GST_DISABLE_GST_DEBUG + gst_debug_log_valist (GST_CAT_DEFAULT, GST_LEVEL_WARNING, __FILE__, func, + line, (GObject *) dec, debug_msg_format, args); +#endif + + g_free (dec->error_msg); + if (debug_msg_format) + dec->error_msg = g_strdup_vprintf (debug_msg_format, args); + else + dec->error_msg = NULL; + + dec->error_line = line; + dec->error_func = func; +} + +static void +gst_jpeg_dec_set_error (GstJpegDec * dec, const gchar * func, gint line, + const gchar * debug_msg_format, ...) +{ + va_list va; + + va_start (va, debug_msg_format); + gst_jpeg_dec_set_error_va (dec, func, line, debug_msg_format, va); + va_end (va); +} + +static GstFlowReturn +gst_jpeg_dec_post_error_or_warning (GstJpegDec * dec) +{ + GstFlowReturn ret; + int max_errors; + + ++dec->error_count; + max_errors = g_atomic_int_get (&dec->max_errors); + + if (max_errors < 0) { + ret = GST_FLOW_OK; + } else if (max_errors == 0) { + /* FIXME: do something more clever in "automatic mode" */ + if (dec->packetized) { + ret = (dec->error_count < 3) ? GST_FLOW_OK : GST_FLOW_ERROR; + } else { + ret = GST_FLOW_ERROR; + } + } else { + ret = (dec->error_count < max_errors) ? GST_FLOW_OK : GST_FLOW_ERROR; + } + + GST_INFO_OBJECT (dec, "decoding error %d/%d (%s)", dec->error_count, + max_errors, (ret == GST_FLOW_OK) ? "ignoring error" : "erroring out"); + + gst_element_message_full (GST_ELEMENT (dec), + (ret == GST_FLOW_OK) ? GST_MESSAGE_WARNING : GST_MESSAGE_ERROR, + GST_STREAM_ERROR, GST_STREAM_ERROR_DECODE, + g_strdup (_("Failed to decode JPEG image")), dec->error_msg, + __FILE__, dec->error_func, dec->error_line); + + dec->error_msg = NULL; + gst_jpeg_dec_clear_error (dec); + return ret; +} + static boolean gst_jpeg_dec_fill_input_buffer (j_decompress_ptr cinfo) { @@ -346,6 +438,7 @@ gst_jpeg_dec_init (GstJpegDec * dec) /* init properties */ dec->idct_method = JPEG_DEFAULT_IDCT_METHOD; + dec->max_errors = JPEG_DEFAULT_MAX_ERRORS; dec->adapter = gst_adapter_new (); } @@ -954,10 +1047,9 @@ gst_jpeg_dec_decode_direct (GstJpegDec * dec, guchar * base[3], format_not_supported: { - GST_ELEMENT_ERROR (dec, STREAM, DECODE, - (_("Failed to decode JPEG image")), - ("Unsupported subsampling schema: v_samp factors: %u %u %u", - v_samp[0], v_samp[1], v_samp[2])); + gst_jpeg_dec_set_error (dec, GST_FUNCTION, __LINE__, + "Unsupported subsampling schema: v_samp factors: %u %u %u", + v_samp[0], v_samp[1], v_samp[2]); return GST_FLOW_ERROR; } } @@ -1440,6 +1532,11 @@ again: goto drop_buffer; } + /* reset error count on successful decode */ + dec->error_count = 0; + + ++dec->good_count; + GST_LOG_OBJECT (dec, "pushing buffer (ts=%" GST_TIME_FORMAT ", dur=%" GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)), GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf))); @@ -1452,6 +1549,11 @@ done: exit: + if (G_UNLIKELY (ret == GST_FLOW_ERROR)) { + jpeg_abort_decompress (&dec->cinfo); + ret = gst_jpeg_dec_post_error_or_warning (dec); + } + return ret; /* special cases */ @@ -1468,9 +1570,8 @@ need_more_data: /* ERRORS */ wrong_size: { - GST_ELEMENT_ERROR (dec, STREAM, DECODE, - ("Picture is too small or too big (%ux%u)", width, height), - ("Picture is too small or too big (%ux%u)", width, height)); + gst_jpeg_dec_set_error (dec, GST_FUNCTION, __LINE__, + "Picture is too small or too big (%ux%u)", width, height); ret = GST_FLOW_ERROR; goto done; } @@ -1480,8 +1581,9 @@ decode_error: dec->jerr.pub.format_message ((j_common_ptr) (&dec->cinfo), err_msg); - GST_ELEMENT_ERROR (dec, STREAM, DECODE, - (_("Failed to decode JPEG image")), ("Error #%u: %s", code, err_msg)); + gst_jpeg_dec_set_error (dec, GST_FUNCTION, __LINE__, + "Decode error #%u: %s", code, err_msg); + if (outbuf) { gst_buffer_unref (outbuf); outbuf = NULL; @@ -1507,9 +1609,8 @@ alloc_failed: jpeg_abort_decompress (&dec->cinfo); if (ret != GST_FLOW_UNEXPECTED && ret != GST_FLOW_WRONG_STATE && ret != GST_FLOW_NOT_LINKED) { - GST_ELEMENT_ERROR (dec, STREAM, DECODE, - ("Buffer allocation failed, reason: %s", reason), - ("Buffer allocation failed, reason: %s", reason)); + gst_jpeg_dec_set_error (dec, GST_FUNCTION, __LINE__, + "Buffer allocation failed, reason: %s", reason); } goto exit; } @@ -1522,22 +1623,22 @@ drop_buffer: } components_not_supported: { - GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), - ("more components than supported: %d > 3", dec->cinfo.num_components)); + gst_jpeg_dec_set_error (dec, GST_FUNCTION, __LINE__, + "more components than supported: %d > 3", dec->cinfo.num_components); ret = GST_FLOW_ERROR; goto done; } unsupported_colorspace: { - GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), - ("Picture has unknown or unsupported colourspace")); + gst_jpeg_dec_set_error (dec, GST_FUNCTION, __LINE__, + "Picture has unknown or unsupported colourspace"); ret = GST_FLOW_ERROR; goto done; } invalid_yuvrgbgrayscale: { - GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), - ("Picture is corrupt or unhandled YUV/RGB/grayscale layout")); + gst_jpeg_dec_set_error (dec, GST_FUNCTION, __LINE__, + "Picture is corrupt or unhandled YUV/RGB/grayscale layout"); ret = GST_FLOW_ERROR; goto done; } @@ -1632,6 +1733,9 @@ gst_jpeg_dec_set_property (GObject * object, guint prop_id, case PROP_IDCT_METHOD: dec->idct_method = g_value_get_enum (value); break; + case PROP_MAX_ERRORS: + g_atomic_int_set (&dec->max_errors, g_value_get_int (value)); + break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); @@ -1651,6 +1755,9 @@ gst_jpeg_dec_get_property (GObject * object, guint prop_id, GValue * value, case PROP_IDCT_METHOD: g_value_set_enum (value, dec->idct_method); break; + case PROP_MAX_ERRORS: + g_value_set_int (value, g_atomic_int_get (&dec->max_errors)); + break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); @@ -1668,6 +1775,8 @@ gst_jpeg_dec_change_state (GstElement * element, GstStateChange transition) switch (transition) { case GST_STATE_CHANGE_READY_TO_PAUSED: + dec->error_count = 0; + dec->good_count = 0; dec->framerate_numerator = 0; dec->framerate_denominator = 1; dec->caps_framerate_numerator = dec->caps_framerate_denominator = 0; diff --git a/ext/jpeg/gstjpegdec.h b/ext/jpeg/gstjpegdec.h index 7f1ea53a81..7daf7b6f23 100644 --- a/ext/jpeg/gstjpegdec.h +++ b/ext/jpeg/gstjpegdec.h @@ -113,6 +113,18 @@ struct _GstJpegDec { /* properties */ gint idct_method; + gint max_errors; /* ATOMIC */ + + /* current error (the message is the debug message) */ + gchar *error_msg; + int error_line; + const gchar *error_func; + + /* number of errors since start or last successfully decoded image */ + guint error_count; + + /* number of successfully decoded images since start */ + guint good_count; struct jpeg_decompress_struct cinfo; struct GstJpegDecErrorMgr jerr; From 9941d3f0a260317e4f555d538b6b0f04296a1be1 Mon Sep 17 00:00:00 2001 From: Jan Schmidt Date: Sun, 12 Dec 2010 01:39:06 +1100 Subject: [PATCH 016/351] deinterlace: Avoid infinite loop draining frames When the pipeline is flushed just as we're draining history, don't loop infinitely, just discard the history and abort. --- gst/deinterlace/gstdeinterlace.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/gst/deinterlace/gstdeinterlace.c b/gst/deinterlace/gstdeinterlace.c index 7d235e443c..37541547e0 100644 --- a/gst/deinterlace/gstdeinterlace.c +++ b/gst/deinterlace/gstdeinterlace.c @@ -593,6 +593,16 @@ gst_deinterlace_reset_history (GstDeinterlace * self, gboolean drop_all) { gint i; + if (!drop_all) { + GST_DEBUG_OBJECT (self, "Flushing history (count %d)", self->history_count); + while (self->history_count > 0) { + if (gst_deinterlace_output_frame (self, TRUE) != GST_FLOW_OK) { + /* Encountered error, or flushing -> skip and drop all remaining */ + drop_all = TRUE; + break; + } + } + } if (drop_all) { GST_DEBUG_OBJECT (self, "Resetting history (count %d)", self->history_count); @@ -603,10 +613,6 @@ gst_deinterlace_reset_history (GstDeinterlace * self, gboolean drop_all) self->field_history[i].buf = NULL; } } - } else { - GST_DEBUG_OBJECT (self, "Flushing history (count %d)", self->history_count); - while (self->history_count > 0) - gst_deinterlace_output_frame (self, TRUE); } memset (self->field_history, 0, GST_DEINTERLACE_MAX_FIELD_HISTORY * sizeof (GstDeinterlaceField)); From acc37e52a73042f9d91cfb949146dcb5c6d7bac4 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Sun, 12 Dec 2010 14:53:17 +0100 Subject: [PATCH 017/351] mp4vpay: we can also accept xvid caps --- gst/rtp/gstrtpmp4vpay.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/gst/rtp/gstrtpmp4vpay.c b/gst/rtp/gstrtpmp4vpay.c index 07fd84a082..b7660123a7 100644 --- a/gst/rtp/gstrtpmp4vpay.c +++ b/gst/rtp/gstrtpmp4vpay.c @@ -31,11 +31,11 @@ GST_DEBUG_CATEGORY_STATIC (rtpmp4vpay_debug); #define GST_CAT_DEFAULT (rtpmp4vpay_debug) static GstStaticPadTemplate gst_rtp_mp4v_pay_sink_template = -GST_STATIC_PAD_TEMPLATE ("sink", + GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("video/mpeg," - "mpegversion=(int) 4," "systemstream=(boolean)false") + "mpegversion=(int) 4," "systemstream=(boolean)false;" "video/x-xvid") ); static GstStaticPadTemplate gst_rtp_mp4v_pay_src_template = From ffc7cd98034ebf1000e3f0cc3659d979977531a5 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Mon, 13 Dec 2010 12:56:12 +0100 Subject: [PATCH 018/351] jitterbuffer: avoid leaking sink events Avoid leaking the newsegment event when it has the wrong format. --- gst/rtpmanager/gstrtpjitterbuffer.c | 1 + 1 file changed, 1 insertion(+) diff --git a/gst/rtpmanager/gstrtpjitterbuffer.c b/gst/rtpmanager/gstrtpjitterbuffer.c index 918238ce7b..5b7e126010 100644 --- a/gst/rtpmanager/gstrtpjitterbuffer.c +++ b/gst/rtpmanager/gstrtpjitterbuffer.c @@ -1118,6 +1118,7 @@ newseg_wrong_format: { GST_DEBUG_OBJECT (jitterbuffer, "received non TIME newsegment"); ret = FALSE; + gst_event_unref (event); goto done; } } From d30f3baffa197e44686225fcd710a1baf91a3a63 Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Mon, 13 Dec 2010 16:23:24 +0200 Subject: [PATCH 019/351] Automatic update of common submodule From 011bcc8 to 20742ae --- common | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/common b/common index 011bcc8a0f..20742aee03 160000 --- a/common +++ b/common @@ -1 +1 @@ -Subproject commit 011bcc8a0fc7f798ee874a7ba899123fb2470e22 +Subproject commit 20742aee033cc7c4aa9a817df005d15d5fa6ba85 From 957eac957952b6f1d3586f6b550e9b0f99f24f90 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Tue, 14 Dec 2010 15:27:52 +0100 Subject: [PATCH 020/351] j2kpay: stop scanning when we reached the end Stop scanning for markers when we reached the end of the data. --- gst/rtp/gstrtpj2kpay.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/gst/rtp/gstrtpj2kpay.c b/gst/rtp/gstrtpj2kpay.c index de4462e45b..6fc129c9cb 100644 --- a/gst/rtp/gstrtpj2kpay.c +++ b/gst/rtp/gstrtpj2kpay.c @@ -437,6 +437,11 @@ gst_rtp_j2k_pay_handle_buffer (GstBaseRTPPayload * basepayload, } pos = end; + + /* exit when finished */ + if (pos == size) + break; + /* scan next packetization unit and fill in the header */ end = find_pu_end (pay, data, size, pos, &state); } while (TRUE); From 7ebd374766fb4aab3b2ef0a3700e51a56e84144d Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Tue, 14 Dec 2010 17:15:23 +0100 Subject: [PATCH 021/351] rtpbin: correctly calculate RTCP packet size --- gst/rtpmanager/rtpsession.c | 20 +++++++++++--------- gst/rtpmanager/rtpstats.c | 9 +++++---- 2 files changed, 16 insertions(+), 13 deletions(-) diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c index 3ca79660d5..c50141d76c 100644 --- a/gst/rtpmanager/rtpsession.c +++ b/gst/rtpmanager/rtpsession.c @@ -74,13 +74,14 @@ enum PROP_LAST }; -/* update average packet size, we keep this scaled by 16 to keep enough - * precision. */ +/* update average packet size */ +#define INIT_AVG(avg, val) \ + (avg) = (val); #define UPDATE_AVG(avg, val) \ if ((avg) == 0) \ - (avg) = (val) << 4; \ + (avg) = (val); \ else \ - (avg) = ((val) + (15 * (avg))); + (avg) = ((val) + (15 * (val))) >> 4; /* The number RTCP intervals after which to timeout entries in the * collision table @@ -2169,8 +2170,7 @@ rtp_session_schedule_bye_locked (RTPSession * sess, const gchar * reason, /* at least one member wants to send a BYE */ g_free (sess->bye_reason); sess->bye_reason = g_strdup (reason); - /* The avg packet size is kept scaled by 16 */ - sess->stats.avg_rtcp_packet_size = 100 * 16; + INIT_AVG (sess->stats.avg_rtcp_packet_size, 100); sess->stats.bye_members = 1; sess->first_rtcp = TRUE; sess->sent_bye = FALSE; @@ -2692,12 +2692,14 @@ rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time, /* close the RTCP packet */ gst_rtcp_buffer_end (data.rtcp); - GST_DEBUG ("sending packet"); if (sess->callbacks.send_rtcp) { UPDATE_AVG (sess->stats.avg_rtcp_packet_size, GST_BUFFER_SIZE (data.rtcp)); - result = sess->callbacks.send_rtcp (sess, own, data.rtcp, - sess->sent_bye, sess->send_rtcp_user_data); + GST_DEBUG ("sending RTCP packet, avg size %u", + sess->stats.avg_rtcp_packet_size); + result = + sess->callbacks.send_rtcp (sess, own, data.rtcp, sess->sent_bye, + sess->send_rtcp_user_data); } else { GST_DEBUG ("freeing packet"); gst_buffer_unref (data.rtcp); diff --git a/gst/rtpmanager/rtpstats.c b/gst/rtpmanager/rtpstats.c index a6de7d880b..a31e7e52ff 100644 --- a/gst/rtpmanager/rtpstats.c +++ b/gst/rtpmanager/rtpstats.c @@ -166,7 +166,7 @@ rtp_stats_calculate_rtcp_interval (RTPSessionStats * stats, gboolean we_send, if (rtcp_bw <= 0.00001) return GST_CLOCK_TIME_NONE; - avg_rtcp_size = stats->avg_rtcp_packet_size / 16.0; + avg_rtcp_size = stats->avg_rtcp_packet_size; /* * The effective number of sites times the average packet size is * the total number of octets sent when each site sends a report. @@ -176,6 +176,7 @@ rtp_stats_calculate_rtcp_interval (RTPSessionStats * stats, gboolean we_send, * time interval we send one report so this time is also our * average time between reports. */ + GST_DEBUG ("avg size %f, n %f, rtcp_bw %f", avg_rtcp_size, n, rtcp_bw); interval = avg_rtcp_size * n / rtcp_bw; if (interval < rtcp_min_time) interval = rtcp_min_time; @@ -245,7 +246,7 @@ rtp_stats_calculate_bye_interval (RTPSessionStats * stats) if (rtcp_bw <= 0.0001) return GST_CLOCK_TIME_NONE; - avg_rtcp_size = stats->avg_rtcp_packet_size / 16.0; + avg_rtcp_size = stats->avg_rtcp_packet_size; /* * The effective number of sites times the average packet size is * the total number of octets sent when each site sends a report. @@ -274,7 +275,7 @@ rtp_stats_calculate_bye_interval (RTPSessionStats * stats) * Returns: total RTP packets lost. */ gint64 -rtp_stats_get_packets_lost (const RTPSourceStats *stats) +rtp_stats_get_packets_lost (const RTPSourceStats * stats) { gint64 lost; guint64 extended_max, expected; @@ -284,4 +285,4 @@ rtp_stats_get_packets_lost (const RTPSourceStats *stats) lost = expected - stats->packets_received; return lost; -} \ No newline at end of file +} From 0c3333da04e24fccaec4683472c739b160e697dc Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Tue, 14 Dec 2010 18:12:43 +0100 Subject: [PATCH 022/351] session: fix average RTCP packet size some more. Fix stupid error in averaging macro. Include udp headers in packet length estimation. --- gst/rtpmanager/rtpsession.c | 17 ++++++++++++----- 1 file changed, 12 insertions(+), 5 deletions(-) diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c index c50141d76c..0a0ee88ad3 100644 --- a/gst/rtpmanager/rtpsession.c +++ b/gst/rtpmanager/rtpsession.c @@ -81,7 +81,8 @@ enum if ((avg) == 0) \ (avg) = (val); \ else \ - (avg) = ((val) + (15 * (val))) >> 4; + (avg) = ((val) + (15 * (avg))) >> 4; + /* The number RTCP intervals after which to timeout entries in the * collision table @@ -368,6 +369,7 @@ rtp_session_init (RTPSession * sess) sess->source->validated = TRUE; sess->source->internal = TRUE; sess->stats.active_sources++; + INIT_AVG (sess->stats.avg_rtcp_packet_size, 100); /* default UDP header length */ sess->header_len = 28; @@ -2001,6 +2003,8 @@ rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer, /* keep track of average packet size */ UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes); } + GST_DEBUG ("%p, received RTCP packet, avg size %u, %u", &sess->stats, + sess->stats.avg_rtcp_packet_size, arrival.bytes); RTP_SESSION_UNLOCK (sess); /* notify caller of sr packets in the callback */ @@ -2693,10 +2697,13 @@ rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time, gst_rtcp_buffer_end (data.rtcp); if (sess->callbacks.send_rtcp) { - UPDATE_AVG (sess->stats.avg_rtcp_packet_size, - GST_BUFFER_SIZE (data.rtcp)); - GST_DEBUG ("sending RTCP packet, avg size %u", - sess->stats.avg_rtcp_packet_size); + guint packet_size; + + packet_size = GST_BUFFER_SIZE (data.rtcp) + sess->header_len; + + UPDATE_AVG (sess->stats.avg_rtcp_packet_size, packet_size); + GST_DEBUG ("%p, sending RTCP packet, avg size %u, %u", &sess->stats, + sess->stats.avg_rtcp_packet_size, packet_size); result = sess->callbacks.send_rtcp (sess, own, data.rtcp, sess->sent_bye, sess->send_rtcp_user_data); From 184c4219a72f033f3668ef83bc6b2f2ad2110d3f Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Wed, 15 Dec 2010 13:12:09 +0100 Subject: [PATCH 023/351] j2kdepay: add support for buffer lists --- gst/rtp/gstrtpj2kdepay.c | 87 ++++++++++++++++++++++++++++++++++++++-- gst/rtp/gstrtpj2kdepay.h | 1 + 2 files changed, 84 insertions(+), 4 deletions(-) diff --git a/gst/rtp/gstrtpj2kdepay.c b/gst/rtp/gstrtpj2kdepay.c index 82411aa987..f864c380f4 100644 --- a/gst/rtp/gstrtpj2kdepay.c +++ b/gst/rtp/gstrtpj2kdepay.c @@ -56,11 +56,25 @@ typedef enum J2K_MARKER_EOC = 0xD9 } RtpJ2KMarker; +#define DEFAULT_BUFFER_LIST TRUE + +enum +{ + PROP_0, + PROP_BUFFER_LIST, + PROP_LAST +}; + GST_BOILERPLATE (GstRtpJ2KDepay, gst_rtp_j2k_depay, GstBaseRTPDepayload, GST_TYPE_BASE_RTP_DEPAYLOAD); static void gst_rtp_j2k_depay_finalize (GObject * object); +static void gst_rtp_j2k_depay_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec); +static void gst_rtp_j2k_depay_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec); + static GstStateChangeReturn gst_rtp_j2k_depay_change_state (GstElement * element, GstStateChange transition); @@ -99,6 +113,14 @@ gst_rtp_j2k_depay_class_init (GstRtpJ2KDepayClass * klass) gobject_class->finalize = gst_rtp_j2k_depay_finalize; + gobject_class->set_property = gst_rtp_j2k_depay_set_property; + gobject_class->get_property = gst_rtp_j2k_depay_get_property; + + g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BUFFER_LIST, + g_param_spec_boolean ("buffer-list", "Buffer List", + "Use Buffer Lists", + DEFAULT_BUFFER_LIST, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + gstelement_class->change_state = gst_rtp_j2k_depay_change_state; gstbasertpdepayload_class->set_caps = gst_rtp_j2k_depay_setcaps; @@ -112,6 +134,8 @@ static void gst_rtp_j2k_depay_init (GstRtpJ2KDepay * rtpj2kdepay, GstRtpJ2KDepayClass * klass) { + rtpj2kdepay->buffer_list = DEFAULT_BUFFER_LIST; + rtpj2kdepay->pu_adapter = gst_adapter_new (); rtpj2kdepay->t_adapter = gst_adapter_new (); rtpj2kdepay->f_adapter = gst_adapter_new (); @@ -350,7 +374,7 @@ gst_rtp_j2k_depay_flush_frame (GstBaseRTPDepayload * depayload) guint8 end[2]; guint8 *data; guint avail; - GstBuffer *outbuf; + GstFlowReturn ret = GST_FLOW_OK; rtpj2kdepay = GST_RTP_J2K_DEPAY (depayload); @@ -364,6 +388,8 @@ gst_rtp_j2k_depay_flush_frame (GstBaseRTPDepayload * depayload) goto done; if (avail > 2) { + GstBuffer *outbuf; + /* take the last bytes of the JPEG 2000 data to see if there is an EOC * marker */ gst_adapter_copy (rtpj2kdepay->f_adapter, end, avail - 2, 2); @@ -381,9 +407,26 @@ gst_rtp_j2k_depay_flush_frame (GstBaseRTPDepayload * depayload) avail += 2; } - GST_DEBUG_OBJECT (rtpj2kdepay, "pushing %u bytes", avail); - outbuf = gst_adapter_take_buffer (rtpj2kdepay->f_adapter, avail); - ret = gst_base_rtp_depayload_push (depayload, outbuf); + if (rtpj2kdepay->buffer_list) { + GList *list; + GstBufferList *buflist; + GstBufferListIterator *it; + + GST_DEBUG_OBJECT (rtpj2kdepay, "pushing buffer list of %u bytes", avail); + list = gst_adapter_take_list (rtpj2kdepay->f_adapter, avail); + + buflist = gst_buffer_list_new (); + it = gst_buffer_list_iterate (buflist); + gst_buffer_list_iterator_add_group (it); + gst_buffer_list_iterator_add_list (it, list); + gst_buffer_list_iterator_free (it); + + ret = gst_base_rtp_depayload_push_list (depayload, buflist); + } else { + GST_DEBUG_OBJECT (rtpj2kdepay, "pushing buffer of %u bytes", avail); + outbuf = gst_adapter_take_buffer (rtpj2kdepay->f_adapter, avail); + ret = gst_base_rtp_depayload_push (depayload, outbuf); + } } else { GST_WARNING_OBJECT (rtpj2kdepay, "empty packet"); gst_adapter_clear (rtpj2kdepay->f_adapter); @@ -555,6 +598,42 @@ wrong_mh_id: } } +static void +gst_rtp_j2k_depay_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec) +{ + GstRtpJ2KDepay *rtpj2kdepay; + + rtpj2kdepay = GST_RTP_J2K_DEPAY (object); + + switch (prop_id) { + case PROP_BUFFER_LIST: + rtpj2kdepay->buffer_list = g_value_get_boolean (value); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static void +gst_rtp_j2k_depay_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec) +{ + GstRtpJ2KDepay *rtpj2kdepay; + + rtpj2kdepay = GST_RTP_J2K_DEPAY (object); + + switch (prop_id) { + case PROP_BUFFER_LIST: + g_value_set_boolean (value, rtpj2kdepay->buffer_list); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + static GstStateChangeReturn gst_rtp_j2k_depay_change_state (GstElement * element, GstStateChange transition) { diff --git a/gst/rtp/gstrtpj2kdepay.h b/gst/rtp/gstrtpj2kdepay.h index e855eef940..40373b264b 100644 --- a/gst/rtp/gstrtpj2kdepay.h +++ b/gst/rtp/gstrtpj2kdepay.h @@ -58,6 +58,7 @@ struct _GstRtpJ2KDepay guint next_frag; gboolean have_sync; + gboolean buffer_list; gint width, height; }; From 3baf936e6dbed34d42493462aac2b57e92b5eec2 Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Wed, 15 Dec 2010 14:55:58 +0200 Subject: [PATCH 024/351] Automatic update of common submodule From 20742ae to 169462a --- common | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/common b/common index 20742aee03..169462a62f 160000 --- a/common +++ b/common @@ -1 +1 @@ -Subproject commit 20742aee033cc7c4aa9a817df005d15d5fa6ba85 +Subproject commit 169462a62f8b3cb91d721092af13530a3f72a462 From 984564150fb0ecd302a14460eb6866558f73d45d Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Wed, 15 Dec 2010 13:30:50 +0000 Subject: [PATCH 025/351] configure: depend on -base from git for new rtp base depayloader features This is ok in this case, since the plan is to release core/base again along with good/ugly/bad in the next cycle. --- configure.ac | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/configure.ac b/configure.ac index 919a852fe3..f510f70673 100644 --- a/configure.ac +++ b/configure.ac @@ -52,8 +52,8 @@ AC_LIBTOOL_WIN32_DLL AM_PROG_LIBTOOL dnl *** required versions of GStreamer stuff *** -GST_REQ=0.10.31 -GSTPB_REQ=0.10.31 +GST_REQ=0.10.31.1 +GSTPB_REQ=0.10.31.1 dnl *** autotools stuff **** From 744472d2add12b115af2f5bcfb95d950c4a75989 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Wed, 15 Dec 2010 16:22:54 +0100 Subject: [PATCH 026/351] j2kpay: cleanup header construction Use a simpler way of constructing the header that doesn't depend on the endianness. --- gst/rtp/gstrtpj2kpay.c | 68 ++++++++++++++++++------------------------ 1 file changed, 29 insertions(+), 39 deletions(-) diff --git a/gst/rtp/gstrtpj2kpay.c b/gst/rtp/gstrtpj2kpay.c index 6fc129c9cb..11e4e24f99 100644 --- a/gst/rtp/gstrtpj2kpay.c +++ b/gst/rtp/gstrtpj2kpay.c @@ -86,47 +86,18 @@ enum PROP_LAST }; -/* - * RtpJ2KHeader: - * @tp: type (0 progressive, 1 odd field, 2 even field) - * @MHF: Main Header Flag - * @mh_id: Main Header Identification - * @T: Tile field invalidation flag - * @priority: priority - * @tile number: the tile number of the payload - * @reserved: set to 0 - * @fragment offset: the byte offset of the current payload - * - * 0 1 2 3 - * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 - * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - * |tp |MHF|mh_id|T| priority | tile number | - * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - * |reserved | fragment offset | - * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - */ typedef struct { -#if G_BYTE_ORDER == G_LITTLE_ENDIAN - guint T:1; - guint mh_id:3; - guint MHF:2; - guint tp:2; -#elif G_BYTE_ORDER == G_BIG_ENDIAN guint tp:2; guint MHF:2; guint mh_id:3; guint T:1; -#else -#error "G_BYTE_ORDER should be big or little endian." -#endif guint priority:8; guint tile:16; - guint reserved:8; guint offset:24; } RtpJ2KHeader; -#define HEADER_SIZE sizeof(RtpJ2KHeader) +#define HEADER_SIZE 8 static void gst_rtp_j2k_pay_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); @@ -388,7 +359,6 @@ gst_rtp_j2k_pay_handle_buffer (GstBaseRTPPayload * basepayload, state.header.T = 1; /* invalid tile */ state.header.priority = 255; /* always 255 for now */ state.header.tile = 0; /* no tile number */ - state.header.reserved = 0; state.bitstream = FALSE; state.n_tiles = 0; state.next_sot = 0; @@ -493,14 +463,34 @@ gst_rtp_j2k_pay_handle_buffer (GstBaseRTPPayload * basepayload, gst_rtp_buffer_set_marker (outbuf, TRUE); } - /* copy the header and push the packet */ -#if (G_BYTE_ORDER == G_LITTLE_ENDIAN) - state.header.offset = ((offset & 0x0000FF) << 16) | - ((offset & 0xFF0000) >> 16) | (offset & 0x00FF00); -#else - state.header.offset = offset; -#endif - memcpy (header, &state.header, HEADER_SIZE); + /* + * RtpJ2KHeader: + * @tp: type (0 progressive, 1 odd field, 2 even field) + * @MHF: Main Header Flag + * @mh_id: Main Header Identification + * @T: Tile field invalidation flag + * @priority: priority + * @tile number: the tile number of the payload + * @reserved: set to 0 + * @fragment offset: the byte offset of the current payload + * + * 0 1 2 3 + * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 + * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + * |tp |MHF|mh_id|T| priority | tile number | + * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + * |reserved | fragment offset | + * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + */ + header[0] = (state.header.tp << 6) | (state.header.MHF << 4) | + (state.header.mh_id << 1) | state.header.T; + header[1] = state.header.priority; + header[2] = state.header.tile >> 8; + header[3] = state.header.tile & 0xff; + header[4] = 0; + header[5] = state.header.offset >> 16; + header[6] = (state.header.offset >> 8) & 0xff; + header[7] = state.header.offset & 0xff; if (pay->buffer_list) { GstBuffer *paybuf; From 34222431aa441de06da0788cb8bbfc65e7d4bbfc Mon Sep 17 00:00:00 2001 From: Edward Hervey Date: Wed, 15 Dec 2010 18:21:34 +0100 Subject: [PATCH 027/351] rtpj2kpay: Initialize all fields Makes sad compliers happy --- gst/rtp/gstrtpj2kpay.c | 1 + 1 file changed, 1 insertion(+) diff --git a/gst/rtp/gstrtpj2kpay.c b/gst/rtp/gstrtpj2kpay.c index 11e4e24f99..7d98185575 100644 --- a/gst/rtp/gstrtpj2kpay.c +++ b/gst/rtp/gstrtpj2kpay.c @@ -359,6 +359,7 @@ gst_rtp_j2k_pay_handle_buffer (GstBaseRTPPayload * basepayload, state.header.T = 1; /* invalid tile */ state.header.priority = 255; /* always 255 for now */ state.header.tile = 0; /* no tile number */ + state.header.offset = 0; /* offset of 0 */ state.bitstream = FALSE; state.n_tiles = 0; state.next_sot = 0; From 76ad0ee09bf6afbc219067abd3be5c7dd1b31dd6 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Sebastian=20Dr=C3=B6ge?= Date: Thu, 16 Dec 2010 10:04:19 +0100 Subject: [PATCH 028/351] deinterlace: Change classification to Filter/Effect/Video/Deinterlace --- gst/deinterlace/gstdeinterlace.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/gst/deinterlace/gstdeinterlace.c b/gst/deinterlace/gstdeinterlace.c index 37541547e0..ed752b20d0 100644 --- a/gst/deinterlace/gstdeinterlace.c +++ b/gst/deinterlace/gstdeinterlace.c @@ -369,7 +369,7 @@ gst_deinterlace_base_init (gpointer klass) gst_element_class_set_details_simple (element_class, "Deinterlacer", - "Filter/Video", + "Filter/Effect/Video/Deinterlace", "Deinterlace Methods ported from DScaler/TvTime", "Martin Eikermann , " "Sebastian Dröge "); From 53b9f87436c583a9d33c9944511760da2065b5b8 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Thu, 16 Dec 2010 15:15:49 +0100 Subject: [PATCH 029/351] pulsesink: flush remaining buffered samples on EOS ... which can make a difference between all or nothing when dealing with short streams and relatively large ringbuffer segment. --- ext/pulse/pulsesink.c | 75 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 75 insertions(+) diff --git a/ext/pulse/pulsesink.c b/ext/pulse/pulsesink.c index 2a9a108b2e..796c7b7234 100644 --- a/ext/pulse/pulsesink.c +++ b/ext/pulse/pulsesink.c @@ -1649,6 +1649,50 @@ write_failed: } } +/* write pending local samples, must be called with the mainloop lock */ +static void +gst_pulsering_flush (GstPulseRingBuffer * pbuf) +{ +#ifdef HAVE_PULSE_0_9_16 + GstPulseSink *psink; + + psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf)); + GST_DEBUG_OBJECT (psink, "entering flush"); + + /* flush the buffer if possible */ + if (pbuf->stream && (pbuf->m_data != NULL) && (pbuf->m_towrite > 0)) { +#ifndef GST_DISABLE_GST_DEBUG + gint bps; + + bps = (GST_RING_BUFFER_CAST (pbuf))->spec.bytes_per_sample; + GST_LOG_OBJECT (psink, + "flushing %u samples at offset %" G_GINT64_FORMAT, + (guint) pbuf->m_towrite / bps, pbuf->m_offset); +#endif + + if (pa_stream_write (pbuf->stream, (uint8_t *) pbuf->m_data, + pbuf->m_towrite, NULL, pbuf->m_offset, PA_SEEK_ABSOLUTE) < 0) { + goto write_failed; + } + + pbuf->m_towrite = 0; + pbuf->m_offset += pbuf->m_towrite; /* keep track of current offset */ + } + +done: + return; + + /* ERRORS */ +write_failed: + { + GST_ELEMENT_ERROR (psink, RESOURCE, FAILED, + ("pa_stream_write() failed: %s", + pa_strerror (pa_context_errno (pbuf->context))), (NULL)); + goto done; + } +#endif +} + static void gst_pulsesink_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_pulsesink_get_property (GObject * object, guint prop_id, @@ -2579,6 +2623,34 @@ update_failed: } #endif +static void +gst_pulsesink_flush_ringbuffer (GstPulseSink * psink) +{ + GstPulseRingBuffer *pbuf; + + pa_threaded_mainloop_lock (mainloop); + + pbuf = GST_PULSERING_BUFFER_CAST (GST_BASE_AUDIO_SINK (psink)->ringbuffer); + + if (pbuf == NULL || pbuf->stream == NULL) + goto no_buffer; + + gst_pulsering_flush (pbuf); + + /* We're not interested if this operation failed or not */ +unlock: + pa_threaded_mainloop_unlock (mainloop); + + return; + + /* ERRORS */ +no_buffer: + { + GST_DEBUG_OBJECT (psink, "we have no ringbuffer"); + goto unlock; + } +} + static gboolean gst_pulsesink_event (GstBaseSink * sink, GstEvent * event) { @@ -2626,6 +2698,9 @@ gst_pulsesink_event (GstBaseSink * sink, GstEvent * event) break; } + case GST_EVENT_EOS: + gst_pulsesink_flush_ringbuffer (pulsesink); + break; default: ; } From b87ec0262ba9df0d57dd3694ac12c51f3fe57e90 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Thu, 16 Dec 2010 18:55:43 +0100 Subject: [PATCH 030/351] j2kpay: handle EOC correctly Don't include the next 2 bytes when we are at the end of the data and there are no more bytes left. --- gst/rtp/gstrtpj2kpay.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/gst/rtp/gstrtpj2kpay.c b/gst/rtp/gstrtpj2kpay.c index 7d98185575..e427227aa1 100644 --- a/gst/rtp/gstrtpj2kpay.c +++ b/gst/rtp/gstrtpj2kpay.c @@ -241,8 +241,8 @@ find_pu_end (GstRtpJ2KPay * pay, const guint8 * data, guint size, GST_LOG_OBJECT (pay, "reached next SOT at %u", offset); state->bitstream = FALSE; state->force_packet = TRUE; - if (marker == J2K_MARKER_EOC) - /* include EOC */ + if (marker == J2K_MARKER_EOC && state->next_sot + 2 <= size) + /* include EOC but never go past the max size */ return state->next_sot + 2; else return state->next_sot; From 4c368242c0a2105bd90c8945680ae23ff31d31fd Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Fri, 17 Dec 2010 15:38:15 +0100 Subject: [PATCH 031/351] rtph264depay: determine output h264 layout using caps negotiation ... thereby (partially) deprecating properties currently controlling whether or not byte-stream output or NAL/AU alignment (though properties still determine fallback if nothing specified in caps). Fixes #606662. --- gst/rtp/gstrtph264depay.c | 74 ++++++++++++++++++++++++++++++++++++--- 1 file changed, 70 insertions(+), 4 deletions(-) diff --git a/gst/rtp/gstrtph264depay.c b/gst/rtp/gstrtph264depay.c index 915c9fdf11..dc333ba8d4 100644 --- a/gst/rtp/gstrtph264depay.c +++ b/gst/rtp/gstrtph264depay.c @@ -130,12 +130,12 @@ gst_rtp_h264_depay_class_init (GstRtpH264DepayClass * klass) g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BYTE_STREAM, g_param_spec_boolean ("byte-stream", "Byte Stream", - "Generate byte stream format of NALU", DEFAULT_BYTE_STREAM, - G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + "Generate byte stream format of NALU (deprecated; use caps)", + DEFAULT_BYTE_STREAM, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_ACCESS_UNIT, g_param_spec_boolean ("access-unit", "Access Unit", - "Merge NALU into AU (picture)", DEFAULT_ACCESS_UNIT, - G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + "Merge NALU into AU (picture) (deprecated; use caps)", + DEFAULT_ACCESS_UNIT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); gstelement_class->change_state = gst_rtp_h264_depay_change_state; @@ -214,6 +214,65 @@ gst_rtp_h264_depay_get_property (GObject * object, guint prop_id, } } +static void +gst_rtp_h264_depay_negotiate (GstRtpH264Depay * rtph264depay) +{ + GstCaps *caps; + gint byte_stream = -1; + gint merge = -1; + + caps = + gst_pad_get_allowed_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (rtph264depay)); + + GST_DEBUG_OBJECT (rtph264depay, "allowed caps: %" GST_PTR_FORMAT, caps); + + if (caps) { + if (gst_caps_get_size (caps) > 0) { + GstStructure *s = gst_caps_get_structure (caps, 0); + const gchar *str = NULL; + + if ((str = gst_structure_get_string (s, "stream-format"))) { + if (strcmp (str, "avc") == 0) { + byte_stream = FALSE; + } else if (strcmp (str, "byte-stream") == 0) { + byte_stream = TRUE; + } else { + GST_DEBUG_OBJECT (rtph264depay, "unknown stream-format: %s", str); + } + } + + if ((str = gst_structure_get_string (s, "alignment"))) { + if (strcmp (str, "au") == 0) { + merge = TRUE; + } else if (strcmp (str, "nal") == 0) { + merge = FALSE; + } else { + GST_DEBUG_OBJECT (rtph264depay, "unknown alignment: %s", str); + } + } + } + gst_caps_unref (caps); + } + + if (byte_stream >= 0) { + GST_DEBUG_OBJECT (rtph264depay, "downstream requires byte-stream %d", + byte_stream); + if (rtph264depay->byte_stream != byte_stream) { + GST_WARNING_OBJECT (rtph264depay, + "overriding property setting based on caps"); + rtph264depay->byte_stream = byte_stream; + } + } + if (merge >= 0) { + GST_DEBUG_OBJECT (rtph264depay, "downstream requires merge %d", merge); + if (rtph264depay->merge != merge) { + GST_WARNING_OBJECT (rtph264depay, + "overriding property setting based on caps"); + rtph264depay->merge = merge; + } + } +} + static gboolean gst_rtp_h264_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps) { @@ -239,6 +298,9 @@ gst_rtp_h264_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps) /* hex: AVCProfileIndication:8 | profile_compat:8 | AVCLevelIndication:8 */ profile = gst_structure_get_string (structure, "profile-level-id"); + /* negotiate with downstream w.r.t. output format and alignment */ + gst_rtp_h264_depay_negotiate (rtph264depay); + if (rtph264depay->byte_stream && ps != NULL) { /* for bytestream we only need the parameter sets but we don't error out * when they are not there, we assume they are in the stream. */ @@ -385,6 +447,10 @@ gst_rtp_h264_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps) "codec_data", GST_TYPE_BUFFER, codec_data, NULL); } + gst_caps_set_simple (srccaps, "stream-format", G_TYPE_STRING, + rtph264depay->byte_stream ? "byte-stream" : "avc", + "alignment", G_TYPE_STRING, rtph264depay->merge ? "au" : "nal", NULL); + res = gst_pad_set_caps (depayload->srcpad, srccaps); gst_caps_unref (srccaps); From 66387181ac6d7608fced2a1e5056d2113f197e1f Mon Sep 17 00:00:00 2001 From: Rob Clark Date: Fri, 17 Dec 2010 19:19:35 -0600 Subject: [PATCH 032/351] fix compile errors on macosx with i686-apple-darwin10-gcc-4.2.1: gstv4l2object.c: In function 'gst_v4l2_object_get_nearest_size': gstv4l2object.c:1988: warning: format '%u' expects type 'unsigned int', but argument 12 has type 'gint *' gstv4l2object.c:1988: warning: format '%u' expects type 'unsigned int', but argument 13 has type 'gint *' --- sys/v4l2/gstv4l2object.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sys/v4l2/gstv4l2object.c b/sys/v4l2/gstv4l2object.c index 8653573f20..4abdb107a3 100644 --- a/sys/v4l2/gstv4l2object.c +++ b/sys/v4l2/gstv4l2object.c @@ -1987,7 +1987,7 @@ gst_v4l2_object_get_nearest_size (GstV4l2Object * v4l2object, default: GST_WARNING_OBJECT (v4l2object->element, "Unsupported field type for %" GST_FOURCC_FORMAT "@%ux%u", - GST_FOURCC_ARGS (pixelformat), width, height); + GST_FOURCC_ARGS (pixelformat), *width, *height); return FALSE; } From fbd9581f87570df1b55c11eda9c84c992a7e783f Mon Sep 17 00:00:00 2001 From: Vincent Penquerc'h Date: Sat, 27 Nov 2010 17:15:32 +0000 Subject: [PATCH 033/351] ximagesrc: fix various width/height calculations being off by one, ximagesrc: fix various width/height calculations being off by one, and make it so a single pixel width/height can be captured (except the top left one, as 0,0,0,0 is reserved for full screen as per the property comments). --- sys/ximage/gstximagesrc.c | 37 +++++++++++++++++++++++-------------- 1 file changed, 23 insertions(+), 14 deletions(-) diff --git a/sys/ximage/gstximagesrc.c b/sys/ximage/gstximagesrc.c index e525fbbc35..a1a9dbd849 100644 --- a/sys/ximage/gstximagesrc.c +++ b/sys/ximage/gstximagesrc.c @@ -465,10 +465,10 @@ gst_ximage_src_ximage_get (GstXImageSrc * ximagesrc) if (ximagesrc->endx > ximagesrc->startx && ximagesrc->endy > ximagesrc->starty) { /* see if damage area intersects */ - if (rects[i].x + rects[i].width < ximagesrc->startx || + if (rects[i].x + rects[i].width - 1 < ximagesrc->startx || rects[i].x > ximagesrc->endx) { /* trivial reject */ - } else if (rects[i].y + rects[i].height < ximagesrc->starty || + } else if (rects[i].y + rects[i].height - 1 < ximagesrc->starty || rects[i].y > ximagesrc->endy) { /* trivial reject */ } else { @@ -479,12 +479,12 @@ gst_ximage_src_ximage_get (GstXImageSrc * ximagesrc) rects[i].x; starty = (rects[i].y < ximagesrc->starty) ? ximagesrc->starty : rects[i].y; - width = (rects[i].x + rects[i].width < ximagesrc->endx) ? + width = (rects[i].x + rects[i].width - 1 < ximagesrc->endx) ? rects[i].x + rects[i].width - startx : - ximagesrc->endx - startx; - height = (rects[i].y + rects[i].height < ximagesrc->endy) ? + ximagesrc->endx - startx + 1; + height = (rects[i].y + rects[i].height - 1 < ximagesrc->endy) ? rects[i].y + rects[i].height - starty : ximagesrc->endy - - starty; + starty + 1; GST_LOG_OBJECT (ximagesrc, "Retrieving damaged sub-region @ %d,%d size %dx%d as intersect region", @@ -943,27 +943,36 @@ gst_ximage_src_get_caps (GstBaseSrc * bs) width = xcontext->width; height = xcontext->height; - if (s->endx > s->startx && s->endy > s->starty) { + + /* property comments say 0 means right/bottom, means we can't capture + the top left pixel alone */ + if (s->endx == 0) + s->endx = width - 1; + if (s->endy == 0) + s->endy = height - 1; + + if (s->endx >= s->startx && s->endy >= s->starty) { /* this means user has put in values */ if (s->startx < xcontext->width && s->endx < xcontext->width && - s->starty < xcontext->height && s->endy < xcontext->height) { + s->starty < xcontext->height && s->endy < xcontext->height && + s->startx >= 0 && s->starty >= 0) { /* values are fine */ - s->width = width = s->endx - s->startx; - s->height = height = s->endy - s->starty; + s->width = width = s->endx - s->startx + 1; + s->height = height = s->endy - s->starty + 1; } else { GST_WARNING ("User put in co-ordinates overshooting the X resolution, setting to full screen"); s->startx = 0; s->starty = 0; - s->endx = 0; - s->endy = 0; + s->endx = width - 1; + s->endy = height - 1; } } else { GST_WARNING ("User put in bogus co-ordinates, setting to full screen"); s->startx = 0; s->starty = 0; - s->endx = 0; - s->endy = 0; + s->endx = width - 1; + s->endy = height - 1; } GST_DEBUG ("width = %d, height=%d", width, height); return gst_caps_new_simple ("video/x-raw-rgb", From ac872f61cf8679f0e1f07c6fe8564e25e9a5f352 Mon Sep 17 00:00:00 2001 From: Vincent Penquerc'h Date: Sun, 28 Nov 2010 16:04:35 +0000 Subject: [PATCH 034/351] ximagesrc: use XGetSubImage instead of XGetImage, works with remote X ximagesrc: use XGetSubImage instead of XGetImage, works with remote X (on my setup anyway...) --- sys/ximage/gstximagesrc.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sys/ximage/gstximagesrc.c b/sys/ximage/gstximagesrc.c index a1a9dbd849..f66a1c9c72 100644 --- a/sys/ximage/gstximagesrc.c +++ b/sys/ximage/gstximagesrc.c @@ -591,9 +591,9 @@ gst_ximage_src_ximage_get (GstXImageSrc * ximagesrc) #endif /* HAVE_XSHM */ { GST_DEBUG_OBJECT (ximagesrc, "Retrieving screen using XGetImage"); - ximage->ximage = XGetImage (ximagesrc->xcontext->disp, ximagesrc->xwindow, + XGetSubImage (ximagesrc->xcontext->disp, ximagesrc->xwindow, ximagesrc->startx, ximagesrc->starty, ximagesrc->width, - ximagesrc->height, AllPlanes, ZPixmap); + ximagesrc->height, AllPlanes, ZPixmap, ximage->ximage, 0, 0); } #ifdef HAVE_XDAMAGE } From 7ac31b3d226feba3d46e48679a8097fb30e86f04 Mon Sep 17 00:00:00 2001 From: Vincent Penquerc'h Date: Mon, 29 Nov 2010 12:36:06 +0000 Subject: [PATCH 035/351] ximagesrc: change from XGetImage to XGetSubImage dependant on a property ximagesrc: change from XGetImage to XGetSubImage dependant on a property to avoid unnecessary performance hits by default. --- sys/ximage/gstximagesrc.c | 36 ++++++++++++++++++++++++++++++++---- sys/ximage/gstximagesrc.h | 3 +++ 2 files changed, 35 insertions(+), 4 deletions(-) diff --git a/sys/ximage/gstximagesrc.c b/sys/ximage/gstximagesrc.c index f66a1c9c72..e6a15a1ebd 100644 --- a/sys/ximage/gstximagesrc.c +++ b/sys/ximage/gstximagesrc.c @@ -69,7 +69,8 @@ enum PROP_STARTX, PROP_STARTY, PROP_ENDX, - PROP_ENDY + PROP_ENDY, + PROP_REMOTE, }; GST_BOILERPLATE (GstXImageSrc, gst_ximage_src, GstPushSrc, GST_TYPE_PUSH_SRC); @@ -591,9 +592,16 @@ gst_ximage_src_ximage_get (GstXImageSrc * ximagesrc) #endif /* HAVE_XSHM */ { GST_DEBUG_OBJECT (ximagesrc, "Retrieving screen using XGetImage"); - XGetSubImage (ximagesrc->xcontext->disp, ximagesrc->xwindow, - ximagesrc->startx, ximagesrc->starty, ximagesrc->width, - ximagesrc->height, AllPlanes, ZPixmap, ximage->ximage, 0, 0); + if (ximagesrc->remote) { + XGetSubImage (ximagesrc->xcontext->disp, ximagesrc->xwindow, + ximagesrc->startx, ximagesrc->starty, ximagesrc->width, + ximagesrc->height, AllPlanes, ZPixmap, ximage->ximage, 0, 0); + } else { + ximage->ximage = + XGetImage (ximagesrc->xcontext->disp, ximagesrc->xwindow, + ximagesrc->startx, ximagesrc->starty, ximagesrc->width, + ximagesrc->height, AllPlanes, ZPixmap); + } } #ifdef HAVE_XDAMAGE } @@ -824,6 +832,9 @@ gst_ximage_src_set_property (GObject * object, guint prop_id, case PROP_ENDY: src->endy = g_value_get_uint (value); break; + case PROP_REMOTE: + src->remote = g_value_get_boolean (value); + break; default: break; } @@ -864,6 +875,9 @@ gst_ximage_src_get_property (GObject * object, guint prop_id, GValue * value, case PROP_ENDY: g_value_set_uint (value, src->endy); break; + case PROP_REMOTE: + g_value_set_uint (value, src->remote); + break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; @@ -1110,6 +1124,19 @@ gst_ximage_src_class_init (GstXImageSrcClass * klass) "Y coordinate of bottom right corner of area to be recorded (0 for bottom right of screen)", 0, G_MAXINT, 0, G_PARAM_READWRITE)); + /** + * GstXImageSrc:remote + * + * Whether the X display is remote. The element will try to use alternate calls + * known to work better with remote displays. + * + * Since: 0.10.26 + **/ + g_object_class_install_property (gc, PROP_REMOTE, + g_param_spec_boolean ("remote", "Remote dispay", + "Whether the display is remote", FALSE, + G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + parent_class = g_type_class_peek_parent (klass); push_class->create = gst_ximage_src_create; @@ -1136,6 +1163,7 @@ gst_ximage_src_init (GstXImageSrc * ximagesrc, GstXImageSrcClass * klass) ximagesrc->starty = 0; ximagesrc->endx = 0; ximagesrc->endy = 0; + ximagesrc->remote = FALSE; } static gboolean diff --git a/sys/ximage/gstximagesrc.h b/sys/ximage/gstximagesrc.h index f223d1e761..f436df68fb 100644 --- a/sys/ximage/gstximagesrc.h +++ b/sys/ximage/gstximagesrc.h @@ -83,6 +83,9 @@ struct _GstXImageSrc guint endx; guint endy; + /* whether to use remote friendly calls */ + gboolean remote; + #ifdef HAVE_XFIXES int fixes_event_base; XFixesCursorImage *cursor_image; From cdc03948cefc8c1a455595f23fe6649b397b78af Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Sebastian=20Dr=C3=B6ge?= Date: Sun, 19 Dec 2010 10:22:29 +0100 Subject: [PATCH 036/351] capsdebug: Don't leak pad templates created from static pad templates --- gst/debugutils/gstcapsdebug.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/gst/debugutils/gstcapsdebug.c b/gst/debugutils/gstcapsdebug.c index f4a1743cee..469408c56c 100644 --- a/gst/debugutils/gstcapsdebug.c +++ b/gst/debugutils/gstcapsdebug.c @@ -103,8 +103,7 @@ gst_caps_debug_init (GstCapsDebug * capsdebug, { capsdebug->srcpad = - gst_pad_new_from_template (gst_static_pad_template_get - (&gst_caps_debug_src_template), "src"); + gst_pad_new_from_static_template (&gst_caps_debug_src_template, "src"); gst_pad_set_getcaps_function (capsdebug->srcpad, GST_DEBUG_FUNCPTR (gst_caps_debug_getcaps)); gst_pad_set_acceptcaps_function (capsdebug->srcpad, @@ -112,8 +111,7 @@ gst_caps_debug_init (GstCapsDebug * capsdebug, gst_element_add_pad (GST_ELEMENT (capsdebug), capsdebug->srcpad); capsdebug->sinkpad = - gst_pad_new_from_template (gst_static_pad_template_get - (&gst_caps_debug_sink_template), "sink"); + gst_pad_new_from_static_template (&gst_caps_debug_sink_template, "sink"); gst_pad_set_chain_function (capsdebug->sinkpad, GST_DEBUG_FUNCPTR (gst_caps_debug_sink_chain)); gst_pad_set_bufferalloc_function (capsdebug->sinkpad, From 9d4f3494542cefbee5389d9f614fb22e81dc33ed Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Sebastian=20Dr=C3=B6ge?= Date: Sun, 19 Dec 2010 10:58:16 +0100 Subject: [PATCH 037/351] radioactv: Prevent use of uninitialized values Fixes bug #618652. --- gst/effectv/gstradioac.c | 6 +++--- gst/effectv/gstradioac.h | 2 +- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/gst/effectv/gstradioac.c b/gst/effectv/gstradioac.c index 85b4ecc400..58c1b7b9e8 100644 --- a/gst/effectv/gstradioac.c +++ b/gst/effectv/gstradioac.c @@ -287,7 +287,7 @@ blurzoomcore (GstRadioacTV * filter) /* Background image is refreshed every frame */ static void -image_bgsubtract_update_y (guint32 * src, guint32 * background, guint8 * diff, +image_bgsubtract_update_y (guint32 * src, gint16 * background, guint8 * diff, gint video_area, gint y_threshold) { gint i; @@ -298,7 +298,7 @@ image_bgsubtract_update_y (guint32 * src, guint32 * background, guint8 * diff, gint v; p = src; - q = (gint16 *) background; + q = background; r = diff; for (i = 0; i < video_area; i++) { R = ((*p) & 0xff0000) >> (16 - 1); @@ -446,7 +446,7 @@ gst_radioactv_set_caps (GstBaseTransform * btrans, GstCaps * incaps, if (filter->background) g_free (filter->background); - filter->background = g_new (guint32, filter->width * filter->height); + filter->background = g_new0 (gint16, filter->width * filter->height); setTable (filter); diff --git a/gst/effectv/gstradioac.h b/gst/effectv/gstradioac.h index c80fa0a01f..34ad8ed721 100644 --- a/gst/effectv/gstradioac.h +++ b/gst/effectv/gstradioac.h @@ -64,7 +64,7 @@ struct _GstRadioacTV guint32 *snapframe; guint8 *blurzoombuf; guint8 *diff; - guint32 *background; + gint16 *background; gint *blurzoomx; gint *blurzoomy; From 4ae9f125f0065121301b4a6c673ac56af801fad9 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Sebastian=20Dr=C3=B6ge?= Date: Sun, 19 Dec 2010 11:02:41 +0100 Subject: [PATCH 038/351] wavenc: Fix memory leaks in the unit test --- tests/check/pipelines/wavenc.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/tests/check/pipelines/wavenc.c b/tests/check/pipelines/wavenc.c index 1db4635dce..ee5a3ffc00 100644 --- a/tests/check/pipelines/wavenc.c +++ b/tests/check/pipelines/wavenc.c @@ -111,6 +111,7 @@ make_n_channel_wav (const gint channels, const GValueArray * arr) g_main_loop_run (loop); gst_element_set_state (pipeline, GST_STATE_NULL); + gst_object_unref (pipeline); g_free (audiotestsrc); } @@ -129,6 +130,7 @@ GST_START_TEST (test_encode_stereo) g_value_unset (&val); make_n_channel_wav (2, arr); + g_value_array_free (arr); } GST_END_TEST; From 210f1c44c7f2bdb07e2586f5ed4c314a4ca7786b Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Mon, 20 Dec 2010 10:49:56 +0100 Subject: [PATCH 039/351] rtpbin: copy buffering stats when we create an aggregate buffering message, copy the buffering stats form the last message. At least we get correct buffering mode then. --- gst/rtpmanager/gstrtpbin.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/gst/rtpmanager/gstrtpbin.c b/gst/rtpmanager/gstrtpbin.c index 560bf5f954..48e533ff78 100644 --- a/gst/rtpmanager/gstrtpbin.c +++ b/gst/rtpmanager/gstrtpbin.c @@ -1927,8 +1927,13 @@ gst_rtp_bin_handle_message (GstBin * bin, GstMessage * message) GstRtpBinStream *stream; gboolean change = FALSE, active = FALSE; GstClockTime min_out_time; + GstBufferingMode mode; + gint avg_in, avg_out; + gint64 buffering_left; gst_message_parse_buffering (message, &percent); + gst_message_parse_buffering_stats (message, &mode, &avg_in, &avg_out, + &buffering_left); stream = g_object_get_data (G_OBJECT (GST_MESSAGE_SRC (message)), @@ -1990,6 +1995,8 @@ gst_rtp_bin_handle_message (GstBin * bin, GstMessage * message) /* make a new buffering message with the min value */ message = gst_message_new_buffering (GST_OBJECT_CAST (bin), min_percent); + gst_message_set_buffering_stats (message, mode, avg_in, avg_out, + buffering_left); if (G_UNLIKELY (change)) { GstClock *clock; From 6cb0efede4f3c208c09855172c362b40252a96f2 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Mon, 20 Dec 2010 11:10:22 +0100 Subject: [PATCH 040/351] jitterbuffer: provide a clock. since we are using the clock for sync, we need to also provide a clock for good measure. The reason is that even if downstream elements provide a clock, we don't want to have that clock selected because it might not be running yet. --- gst/rtpmanager/gstrtpjitterbuffer.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/gst/rtpmanager/gstrtpjitterbuffer.c b/gst/rtpmanager/gstrtpjitterbuffer.c index 5b7e126010..fcaef015a1 100644 --- a/gst/rtpmanager/gstrtpjitterbuffer.c +++ b/gst/rtpmanager/gstrtpjitterbuffer.c @@ -235,6 +235,7 @@ static GstPad *gst_rtp_jitter_buffer_request_new_pad (GstElement * element, GstPadTemplate * templ, const gchar * name); static void gst_rtp_jitter_buffer_release_pad (GstElement * element, GstPad * pad); +static GstClock *gst_rtp_jitter_buffer_provide_clock (GstElement * element); /* pad overrides */ static GstCaps *gst_rtp_jitter_buffer_getcaps (GstPad * pad); @@ -439,6 +440,8 @@ gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass) GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_request_new_pad); gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_release_pad); + gstelement_class->provide_clock = + GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_provide_clock); klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_clear_pt_map); klass->set_active = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_active); @@ -651,6 +654,12 @@ wrong_pad: } } +static GstClock * +gst_rtp_jitter_buffer_provide_clock (GstElement * element) +{ + return gst_system_clock_obtain (); +} + static void gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer) { From eb6d552353f056bbeb9e2086823ae9ee9efaacc3 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Mon, 20 Dec 2010 15:54:45 +0100 Subject: [PATCH 041/351] jitterbuffer: get better buffering level When the jitterbuffer contains -1 timestamps, make sure we still calculate the buffer fill level by skipping the -1 buffers. Try to be more resilient to weird input timestamps. --- gst/rtpmanager/rtpjitterbuffer.c | 32 +++++++++++++++++++++++++++++--- 1 file changed, 29 insertions(+), 3 deletions(-) diff --git a/gst/rtpmanager/rtpjitterbuffer.c b/gst/rtpmanager/rtpjitterbuffer.c index 2dee30c071..59307df358 100644 --- a/gst/rtpmanager/rtpjitterbuffer.c +++ b/gst/rtpmanager/rtpjitterbuffer.c @@ -212,11 +212,30 @@ rtp_jitter_buffer_resync (RTPJitterBuffer * jbuf, GstClockTime time, static guint64 get_buffer_level (RTPJitterBuffer * jbuf) { - GstBuffer *high_buf, *low_buf; + GstBuffer *high_buf = NULL, *low_buf = NULL; guint64 level; + GList *find; - high_buf = g_queue_peek_head (jbuf->packets); - low_buf = g_queue_peek_tail (jbuf->packets); + /* first first buffer with timestamp */ + find = g_queue_peek_head_link (jbuf->packets); + while (find) { + high_buf = find->data; + if (GST_BUFFER_TIMESTAMP (high_buf) != -1) + break; + + high_buf = NULL; + find = g_list_next (find); + } + + find = g_queue_peek_tail_link (jbuf->packets); + while (find) { + low_buf = find->data; + if (GST_BUFFER_TIMESTAMP (low_buf) != -1) + break; + + low_buf = NULL; + find = g_list_previous (find); + } if (!high_buf || !low_buf || high_buf == low_buf) { level = 0; @@ -230,6 +249,11 @@ get_buffer_level (RTPJitterBuffer * jbuf) level = high_ts - low_ts; else level = 0; + + GST_LOG_OBJECT (jbuf, + "low %" GST_TIME_FORMAT " high %" GST_TIME_FORMAT " level %" + G_GUINT64_FORMAT, GST_TIME_ARGS (low_ts), GST_TIME_ARGS (high_ts), + level); } return level; } @@ -394,6 +418,8 @@ calculate_skew (RTPJitterBuffer * jbuf, guint32 rtptime, GstClockTime time, } else { GST_WARNING ("backward timestamps at server but no timestamps"); send_diff = 0; + /* at least try to get a new timestamp.. */ + jbuf->base_time = -1; } GST_DEBUG ("extrtp %" G_GUINT64_FORMAT ", gstrtp %" GST_TIME_FORMAT ", base %" From 984849f8fec0088ec5adf40b0009da2ba86e345a Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Mon, 20 Dec 2010 16:05:36 +0100 Subject: [PATCH 042/351] mpapay: reset payloader on flush Reset the payloader on a flush event. Handle DISCONT better. --- gst/rtp/gstrtpmpapay.c | 59 +++++++++++++++++++++++++++++++++++------- 1 file changed, 50 insertions(+), 9 deletions(-) diff --git a/gst/rtp/gstrtpmpapay.c b/gst/rtp/gstrtpmpapay.c index f57d628ead..ed0200391f 100644 --- a/gst/rtp/gstrtpmpapay.c +++ b/gst/rtp/gstrtpmpapay.c @@ -52,6 +52,7 @@ static void gst_rtp_mpa_pay_finalize (GObject * object); static gboolean gst_rtp_mpa_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps); +static gboolean gst_rtp_mpa_pay_handle_event (GstPad * pad, GstEvent * event); static GstFlowReturn gst_rtp_mpa_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buffer); @@ -85,6 +86,7 @@ gst_rtp_mpa_pay_class_init (GstRtpMPAPayClass * klass) gobject_class->finalize = gst_rtp_mpa_pay_finalize; gstbasertppayload_class->set_caps = gst_rtp_mpa_pay_setcaps; + gstbasertppayload_class->handle_event = gst_rtp_mpa_pay_handle_event; gstbasertppayload_class->handle_buffer = gst_rtp_mpa_pay_handle_buffer; } @@ -107,6 +109,15 @@ gst_rtp_mpa_pay_finalize (GObject * object) G_OBJECT_CLASS (parent_class)->finalize (object); } +static void +gst_rtp_mpa_pay_reset (GstRtpMPAPay * pay) +{ + pay->first_ts = -1; + pay->duration = 0; + gst_adapter_clear (pay->adapter); + GST_DEBUG_OBJECT (pay, "reset depayloader"); +} + static gboolean gst_rtp_mpa_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps) { @@ -118,6 +129,27 @@ gst_rtp_mpa_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps) return res; } +static gboolean +gst_rtp_mpa_pay_handle_event (GstPad * pad, GstEvent * event) +{ + GstRtpMPAPay *rtpmpapay; + + rtpmpapay = GST_RTP_MPA_PAY (gst_pad_get_parent (pad)); + + switch (GST_EVENT_TYPE (event)) { + case GST_EVENT_FLUSH_STOP: + gst_rtp_mpa_pay_reset (rtpmpapay); + break; + default: + break; + } + + gst_object_unref (rtpmpapay); + + /* FALSE to let the parent handle the event as well */ + return FALSE; +} + static GstFlowReturn gst_rtp_mpa_pay_flush (GstRtpMPAPay * rtpmpapay) { @@ -143,7 +175,7 @@ gst_rtp_mpa_pay_flush (GstRtpMPAPay * rtpmpapay) guint payload_len; guint packet_len; - /* this will be the total lenght of the packet */ + /* this will be the total length of the packet */ packet_len = gst_rtp_buffer_calc_packet_len (4 + avail, 0, 0); /* fill one MTU or all available bytes */ @@ -198,18 +230,20 @@ gst_rtp_mpa_pay_handle_buffer (GstBaseRTPPayload * basepayload, GstFlowReturn ret; guint size, avail; guint packet_len; - GstClockTime duration; + GstClockTime duration, timestamp; rtpmpapay = GST_RTP_MPA_PAY (basepayload); size = GST_BUFFER_SIZE (buffer); duration = GST_BUFFER_DURATION (buffer); + timestamp = GST_BUFFER_TIMESTAMP (buffer); + + if (GST_BUFFER_IS_DISCONT (buffer)) { + GST_DEBUG_OBJECT (rtpmpapay, "DISCONT"); + gst_rtp_mpa_pay_reset (rtpmpapay); + } avail = gst_adapter_available (rtpmpapay->adapter); - if (avail == 0) { - rtpmpapay->first_ts = GST_BUFFER_TIMESTAMP (buffer); - rtpmpapay->duration = 0; - } /* get packet length of previous data and this new data, * payload length includes a 4 byte header */ @@ -220,14 +254,21 @@ gst_rtp_mpa_pay_handle_buffer (GstBaseRTPPayload * basepayload, if (gst_basertppayload_is_filled (basepayload, packet_len, rtpmpapay->duration + duration)) { ret = gst_rtp_mpa_pay_flush (rtpmpapay); - rtpmpapay->first_ts = GST_BUFFER_TIMESTAMP (buffer); - rtpmpapay->duration = 0; + avail = 0; } else { ret = GST_FLOW_OK; } + if (avail == 0) { + GST_DEBUG_OBJECT (rtpmpapay, + "first packet, save timestamp %" GST_TIME_FORMAT, + GST_TIME_ARGS (timestamp)); + rtpmpapay->first_ts = timestamp; + rtpmpapay->duration = 0; + } + gst_adapter_push (rtpmpapay->adapter, buffer); - rtpmpapay->duration += duration; + rtpmpapay->duration = duration; return ret; } From 933a17089853681d5316f56d82c37b06dd5139d0 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Mon, 20 Dec 2010 16:51:47 +0100 Subject: [PATCH 043/351] mpapay: reset payloader on state change --- gst/rtp/gstrtpmpapay.c | 35 +++++++++++++++++++++++++++++++++++ 1 file changed, 35 insertions(+) diff --git a/gst/rtp/gstrtpmpapay.c b/gst/rtp/gstrtpmpapay.c index ed0200391f..23efb670f6 100644 --- a/gst/rtp/gstrtpmpapay.c +++ b/gst/rtp/gstrtpmpapay.c @@ -50,6 +50,9 @@ static GstStaticPadTemplate gst_rtp_mpa_pay_src_template = static void gst_rtp_mpa_pay_finalize (GObject * object); +static GstStateChangeReturn gst_rtp_mpa_pay_change_state (GstElement * element, + GstStateChange transition); + static gboolean gst_rtp_mpa_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps); static gboolean gst_rtp_mpa_pay_handle_event (GstPad * pad, GstEvent * event); @@ -78,13 +81,17 @@ static void gst_rtp_mpa_pay_class_init (GstRtpMPAPayClass * klass) { GObjectClass *gobject_class; + GstElementClass *gstelement_class; GstBaseRTPPayloadClass *gstbasertppayload_class; gobject_class = (GObjectClass *) klass; + gstelement_class = (GstElementClass *) klass; gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass; gobject_class->finalize = gst_rtp_mpa_pay_finalize; + gstelement_class->change_state = gst_rtp_mpa_pay_change_state; + gstbasertppayload_class->set_caps = gst_rtp_mpa_pay_setcaps; gstbasertppayload_class->handle_event = gst_rtp_mpa_pay_handle_event; gstbasertppayload_class->handle_buffer = gst_rtp_mpa_pay_handle_buffer; @@ -273,6 +280,34 @@ gst_rtp_mpa_pay_handle_buffer (GstBaseRTPPayload * basepayload, return ret; } +static GstStateChangeReturn +gst_rtp_mpa_pay_change_state (GstElement * element, GstStateChange transition) +{ + GstRtpMPAPay *rtpmpapay; + GstStateChangeReturn ret; + + rtpmpapay = GST_RTP_MPA_PAY (element); + + switch (transition) { + case GST_STATE_CHANGE_READY_TO_PAUSED: + gst_rtp_mpa_pay_reset (rtpmpapay); + break; + default: + break; + } + + ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); + + switch (transition) { + case GST_STATE_CHANGE_PAUSED_TO_READY: + gst_rtp_mpa_pay_reset (rtpmpapay); + break; + default: + break; + } + return ret; +} + gboolean gst_rtp_mpa_pay_plugin_init (GstPlugin * plugin) { From 2b142bb279a08d4ad8c9559860edc9b8ed86ad78 Mon Sep 17 00:00:00 2001 From: Edward Hervey Date: Mon, 20 Dec 2010 17:47:05 +0100 Subject: [PATCH 044/351] Automatic update of common submodule From 169462a to 46445ad --- common | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/common b/common index 169462a62f..46445ad50f 160000 --- a/common +++ b/common @@ -1 +1 @@ -Subproject commit 169462a62f8b3cb91d721092af13530a3f72a462 +Subproject commit 46445ad50f184d2640dd205361e0446e9121ba5f From 6a5e6eac55aab4a71265e09f1af59b959055cf36 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Mon, 20 Dec 2010 18:49:49 +0100 Subject: [PATCH 045/351] mpapay: flush last packets on EOS --- gst/rtp/gstrtpmpapay.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/gst/rtp/gstrtpmpapay.c b/gst/rtp/gstrtpmpapay.c index 23efb670f6..88d71bce89 100644 --- a/gst/rtp/gstrtpmpapay.c +++ b/gst/rtp/gstrtpmpapay.c @@ -56,6 +56,7 @@ static GstStateChangeReturn gst_rtp_mpa_pay_change_state (GstElement * element, static gboolean gst_rtp_mpa_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps); static gboolean gst_rtp_mpa_pay_handle_event (GstPad * pad, GstEvent * event); +static GstFlowReturn gst_rtp_mpa_pay_flush (GstRtpMPAPay * rtpmpapay); static GstFlowReturn gst_rtp_mpa_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buffer); @@ -144,6 +145,10 @@ gst_rtp_mpa_pay_handle_event (GstPad * pad, GstEvent * event) rtpmpapay = GST_RTP_MPA_PAY (gst_pad_get_parent (pad)); switch (GST_EVENT_TYPE (event)) { + case GST_EVENT_EOS: + /* make sure we push the last packets in the adapter on EOS */ + gst_rtp_mpa_pay_flush (rtpmpapay); + break; case GST_EVENT_FLUSH_STOP: gst_rtp_mpa_pay_reset (rtpmpapay); break; From e8b8753c909fcace516a6ae3fd759d78155c8c0a Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Tue, 21 Dec 2010 12:29:58 +0100 Subject: [PATCH 046/351] h264pay: flush adapter on flush-stop --- gst/rtp/gstrtph264pay.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/gst/rtp/gstrtph264pay.c b/gst/rtp/gstrtph264pay.c index d90481e86f..d2d0a86180 100644 --- a/gst/rtp/gstrtph264pay.c +++ b/gst/rtp/gstrtph264pay.c @@ -1114,11 +1114,12 @@ gst_rtp_h264_pay_handle_buffer (GstBaseRTPPayload * basepayload, return ret; caps_rejected: - - GST_WARNING_OBJECT (basepayload, "Could not set outcaps"); - g_array_set_size (nal_queue, 0); - gst_buffer_unref (buffer); - return GST_FLOW_NOT_NEGOTIATED; + { + GST_WARNING_OBJECT (basepayload, "Could not set outcaps"); + g_array_set_size (nal_queue, 0); + gst_buffer_unref (buffer); + return GST_FLOW_NOT_NEGOTIATED; + } } static gboolean @@ -1128,6 +1129,9 @@ gst_rtp_h264_pay_handle_event (GstPad * pad, GstEvent * event) GstRtpH264Pay *rtph264pay = GST_RTP_H264_PAY (GST_PAD_PARENT (pad)); switch (GST_EVENT_TYPE (event)) { + case GST_EVENT_FLUSH_STOP: + gst_adapter_clear (rtph264pay->adapter); + break; case GST_EVENT_CUSTOM_DOWNSTREAM: s = gst_event_get_structure (event); if (gst_structure_has_name (s, "GstForceKeyUnit")) { From 1eb0f65f39a4bf45b2eff5bd7f07424e92177e14 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Tue, 21 Dec 2010 12:31:44 +0100 Subject: [PATCH 047/351] mp4gdepay: reset depayloader state --- gst/rtp/gstrtpmp4gdepay.c | 23 ++++++++++++++--------- 1 file changed, 14 insertions(+), 9 deletions(-) diff --git a/gst/rtp/gstrtpmp4gdepay.c b/gst/rtp/gstrtpmp4gdepay.c index 8307becca6..3938ab3cc8 100644 --- a/gst/rtp/gstrtpmp4gdepay.c +++ b/gst/rtp/gstrtpmp4gdepay.c @@ -315,6 +315,18 @@ gst_rtp_mp4g_depay_clear_queue (GstRtpMP4GDepay * rtpmp4gdepay) gst_buffer_unref (outbuf); } +static void +gst_rtp_mp4g_depay_reset (GstRtpMP4GDepay * rtpmp4gdepay) +{ + gst_adapter_clear (rtpmp4gdepay->adapter); + rtpmp4gdepay->max_AU_index = -1; + rtpmp4gdepay->next_AU_index = -1; + rtpmp4gdepay->prev_AU_index = -1; + rtpmp4gdepay->prev_rtptime = -1; + rtpmp4gdepay->last_AU_index = -1; + gst_rtp_mp4g_depay_clear_queue (rtpmp4gdepay); +} + static void gst_rtp_mp4g_depay_flush_queue (GstRtpMP4GDepay * rtpmp4gdepay) { @@ -710,13 +722,7 @@ gst_rtp_mp4g_depay_change_state (GstElement * element, switch (transition) { case GST_STATE_CHANGE_READY_TO_PAUSED: - gst_adapter_clear (rtpmp4gdepay->adapter); - rtpmp4gdepay->max_AU_index = -1; - rtpmp4gdepay->next_AU_index = -1; - rtpmp4gdepay->prev_AU_index = -1; - rtpmp4gdepay->prev_rtptime = -1; - rtpmp4gdepay->last_AU_index = -1; - rtpmp4gdepay->prev_AU_num = -1; + gst_rtp_mp4g_depay_reset (rtpmp4gdepay); break; default: break; @@ -726,8 +732,7 @@ gst_rtp_mp4g_depay_change_state (GstElement * element, switch (transition) { case GST_STATE_CHANGE_PAUSED_TO_READY: - gst_adapter_clear (rtpmp4gdepay->adapter); - gst_rtp_mp4g_depay_clear_queue (rtpmp4gdepay); + gst_rtp_mp4g_depay_reset (rtpmp4gdepay); break; default: break; From 2c6e198157f8dd2314a6ccb72e214c61b4188934 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Tue, 21 Dec 2010 13:49:41 +0100 Subject: [PATCH 048/351] vorbispay: clear packet on flush-stop --- gst/rtp/gstrtpvorbispay.c | 31 ++++++++++++++++++++++++++++--- 1 file changed, 28 insertions(+), 3 deletions(-) diff --git a/gst/rtp/gstrtpvorbispay.c b/gst/rtp/gstrtpvorbispay.c index ae16253520..5bce56f9ea 100644 --- a/gst/rtp/gstrtpvorbispay.c +++ b/gst/rtp/gstrtpvorbispay.c @@ -67,6 +67,8 @@ static GstStateChangeReturn gst_rtp_vorbis_pay_change_state (GstElement * element, GstStateChange transition); static GstFlowReturn gst_rtp_vorbis_pay_handle_buffer (GstBaseRTPPayload * pad, GstBuffer * buffer); +static gboolean gst_rtp_vorbis_pay_handle_event (GstPad * pad, + GstEvent * event); static void gst_rtp_vorbis_pay_base_init (gpointer klass) @@ -97,6 +99,7 @@ gst_rtp_vorbis_pay_class_init (GstRtpVorbisPayClass * klass) gstbasertppayload_class->set_caps = gst_rtp_vorbis_pay_setcaps; gstbasertppayload_class->handle_buffer = gst_rtp_vorbis_pay_handle_buffer; + gstbasertppayload_class->handle_event = gst_rtp_vorbis_pay_handle_event; GST_DEBUG_CATEGORY_INIT (rtpvorbispay_debug, "rtpvorbispay", 0, "Vorbis RTP Payloader"); @@ -109,6 +112,14 @@ gst_rtp_vorbis_pay_init (GstRtpVorbisPay * rtpvorbispay, /* needed because of GST_BOILERPLATE */ } +static void +gst_rtp_vorbis_pay_clear_packet (GstRtpVorbisPay * rtpvorbispay) +{ + if (rtpvorbispay->packet) + gst_buffer_unref (rtpvorbispay->packet); + rtpvorbispay->packet = NULL; +} + static void gst_rtp_vorbis_pay_cleanup (GstRtpVorbisPay * rtpvorbispay) { @@ -116,9 +127,7 @@ gst_rtp_vorbis_pay_cleanup (GstRtpVorbisPay * rtpvorbispay) g_list_free (rtpvorbispay->headers); rtpvorbispay->headers = NULL; - if (rtpvorbispay->packet) - gst_buffer_unref (rtpvorbispay->packet); - rtpvorbispay->packet = NULL; + gst_rtp_vorbis_pay_clear_packet (rtpvorbispay); } static gboolean @@ -635,6 +644,22 @@ header_error: } } +static gboolean +gst_rtp_vorbis_pay_handle_event (GstPad * pad, GstEvent * event) +{ + GstRtpVorbisPay *rtpvorbispay = GST_RTP_VORBIS_PAY (GST_PAD_PARENT (pad)); + + switch (GST_EVENT_TYPE (event)) { + case GST_EVENT_FLUSH_STOP: + gst_rtp_vorbis_pay_clear_packet (rtpvorbispay); + break; + default: + break; + } + /* false to let parent handle event as well */ + return FALSE; +} + static GstStateChangeReturn gst_rtp_vorbis_pay_change_state (GstElement * element, GstStateChange transition) From e47f4487b40c8b7139887cf66365a2edf033b3ac Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Tue, 21 Dec 2010 13:55:40 +0100 Subject: [PATCH 049/351] theorapay: clear packet on flush-stop --- gst/rtp/gstrtptheorapay.c | 32 +++++++++++++++++++++++++++++--- 1 file changed, 29 insertions(+), 3 deletions(-) diff --git a/gst/rtp/gstrtptheorapay.c b/gst/rtp/gstrtptheorapay.c index 82651eca86..47a156103b 100644 --- a/gst/rtp/gstrtptheorapay.c +++ b/gst/rtp/gstrtptheorapay.c @@ -84,6 +84,9 @@ static GstStateChangeReturn gst_rtp_theora_pay_change_state (GstElement * element, GstStateChange transition); static GstFlowReturn gst_rtp_theora_pay_handle_buffer (GstBaseRTPPayload * pad, GstBuffer * buffer); +static gboolean gst_rtp_theora_pay_handle_event (GstPad * pad, + GstEvent * event); + static void gst_rtp_theora_pay_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); @@ -121,6 +124,7 @@ gst_rtp_theora_pay_class_init (GstRtpTheoraPayClass * klass) gstbasertppayload_class->set_caps = gst_rtp_theora_pay_setcaps; gstbasertppayload_class->handle_buffer = gst_rtp_theora_pay_handle_buffer; + gstbasertppayload_class->handle_event = gst_rtp_theora_pay_handle_event; gobject_class->set_property = gst_rtp_theora_pay_set_property; gobject_class->get_property = gst_rtp_theora_pay_get_property; @@ -144,6 +148,14 @@ gst_rtp_theora_pay_init (GstRtpTheoraPay * rtptheorapay, rtptheorapay->last_config = GST_CLOCK_TIME_NONE; } +static void +gst_rtp_theora_pay_clear_packet (GstRtpTheoraPay * rtptheorapay) +{ + if (rtptheorapay->packet) + gst_buffer_unref (rtptheorapay->packet); + rtptheorapay->packet = NULL; +} + static void gst_rtp_theora_pay_cleanup (GstRtpTheoraPay * rtptheorapay) { @@ -151,9 +163,7 @@ gst_rtp_theora_pay_cleanup (GstRtpTheoraPay * rtptheorapay) g_list_free (rtptheorapay->headers); rtptheorapay->headers = NULL; - if (rtptheorapay->packet) - gst_buffer_unref (rtptheorapay->packet); - rtptheorapay->packet = NULL; + gst_rtp_theora_pay_clear_packet (rtptheorapay); if (rtptheorapay->config_data) g_free (rtptheorapay->config_data); @@ -758,6 +768,22 @@ header_error: } } +static gboolean +gst_rtp_theora_pay_handle_event (GstPad * pad, GstEvent * event) +{ + GstRtpTheoraPay *rtptheorapay = GST_RTP_THEORA_PAY (GST_PAD_PARENT (pad)); + + switch (GST_EVENT_TYPE (event)) { + case GST_EVENT_FLUSH_STOP: + gst_rtp_theora_pay_clear_packet (rtptheorapay); + break; + default: + break; + } + /* false to let parent handle event as well */ + return FALSE; +} + static GstStateChangeReturn gst_rtp_theora_pay_change_state (GstElement * element, GstStateChange transition) From eb99eb551514951c795a4f759d25f5a438920622 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Tue, 21 Dec 2010 15:07:14 +0100 Subject: [PATCH 050/351] mp4pay: use vmethod for intercepting events --- gst/rtp/gstrtpmp4vpay.c | 14 +++++--------- gst/rtp/gstrtpmp4vpay.h | 2 -- 2 files changed, 5 insertions(+), 11 deletions(-) diff --git a/gst/rtp/gstrtpmp4vpay.c b/gst/rtp/gstrtpmp4vpay.c index b7660123a7..2427abed21 100644 --- a/gst/rtp/gstrtpmp4vpay.c +++ b/gst/rtp/gstrtpmp4vpay.c @@ -78,7 +78,7 @@ static gboolean gst_rtp_mp4v_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps); static GstFlowReturn gst_rtp_mp4v_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buffer); -static gboolean gst_rtp_mp4v_pay_event (GstPad * pad, GstEvent * event); +static gboolean gst_rtp_mp4v_pay_handle_event (GstPad * pad, GstEvent * event); GST_BOILERPLATE (GstRtpMP4VPay, gst_rtp_mp4v_pay, GstBaseRTPPayload, GST_TYPE_BASE_RTP_PAYLOAD) @@ -132,6 +132,7 @@ gst_rtp_mp4v_pay_class_init (GstRtpMP4VPayClass * klass) gstbasertppayload_class->set_caps = gst_rtp_mp4v_pay_setcaps; gstbasertppayload_class->handle_buffer = gst_rtp_mp4v_pay_handle_buffer; + gstbasertppayload_class->handle_event = gst_rtp_mp4v_pay_handle_event; GST_DEBUG_CATEGORY_INIT (rtpmp4vpay_debug, "rtpmp4vpay", 0, "MP4 video RTP Payloader"); @@ -154,9 +155,6 @@ gst_rtp_mp4v_pay_init (GstRtpMP4VPay * rtpmp4vpay, GstRtpMP4VPayClass * klass) rtpmp4vpay->config = NULL; sinkpad = GST_BASE_RTP_PAYLOAD_SINKPAD (rtpmp4vpay); - - rtpmp4vpay->old_event_func = sinkpad->eventfunc; - gst_pad_set_event_function (sinkpad, gst_rtp_mp4v_pay_event); } static void @@ -596,10 +594,9 @@ gst_rtp_mp4v_pay_handle_buffer (GstBaseRTPPayload * basepayload, } static gboolean -gst_rtp_mp4v_pay_event (GstPad * pad, GstEvent * event) +gst_rtp_mp4v_pay_handle_event (GstPad * pad, GstEvent * event) { GstRtpMP4VPay *rtpmp4vpay; - gboolean ret; rtpmp4vpay = GST_RTP_MP4V_PAY (gst_pad_get_parent (pad)); @@ -619,11 +616,10 @@ gst_rtp_mp4v_pay_event (GstPad * pad, GstEvent * event) break; } - ret = rtpmp4vpay->old_event_func (pad, event); - g_object_unref (rtpmp4vpay); - return ret; + /* let parent handle event too */ + return FALSE; } static void diff --git a/gst/rtp/gstrtpmp4vpay.h b/gst/rtp/gstrtpmp4vpay.h index 7d8bac2407..0f8ab14f6f 100644 --- a/gst/rtp/gstrtpmp4vpay.h +++ b/gst/rtp/gstrtpmp4vpay.h @@ -60,8 +60,6 @@ struct _GstRtpMP4VPay * payloader */ guint config_interval; GstClockTime last_config; - - GstPadEventFunction old_event_func; }; struct _GstRtpMP4VPayClass From d5c8771b2b6aa26fa1bd015f6deff63042b2b785 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Tue, 21 Dec 2010 15:24:18 +0100 Subject: [PATCH 051/351] vrawdepay: reset depayloader state Reset the depayloader state on flush-stop. --- gst/rtp/gstrtpvrawdepay.c | 43 ++++++++++++++++++++++++++++++++++----- 1 file changed, 38 insertions(+), 5 deletions(-) diff --git a/gst/rtp/gstrtpvrawdepay.c b/gst/rtp/gstrtpvrawdepay.c index 54a4b52875..aa2cbfaa1c 100644 --- a/gst/rtp/gstrtpvrawdepay.c +++ b/gst/rtp/gstrtpvrawdepay.c @@ -58,6 +58,9 @@ static GstBuffer *gst_rtp_vraw_depay_process (GstBaseRTPDepayload * depayload, static GstStateChangeReturn gst_rtp_vraw_depay_change_state (GstElement * element, GstStateChange transition); +static gboolean gst_rtp_vraw_depay_handle_event (GstBaseRTPDepayload * filter, + GstEvent * event); + static void gst_rtp_vraw_depay_base_init (gpointer klass) { @@ -87,6 +90,7 @@ gst_rtp_vraw_depay_class_init (GstRtpVRawDepayClass * klass) gstbasertpdepayload_class->set_caps = gst_rtp_vraw_depay_setcaps; gstbasertpdepayload_class->process = gst_rtp_vraw_depay_process; + gstbasertpdepayload_class->handle_event = gst_rtp_vraw_depay_handle_event; GST_DEBUG_CATEGORY_INIT (rtpvrawdepay_debug, "rtpvrawdepay", 0, "raw video RTP Depayloader"); @@ -99,6 +103,16 @@ gst_rtp_vraw_depay_init (GstRtpVRawDepay * rtpvrawdepay, /* needed because of GST_BOILERPLATE */ } +static void +gst_rtp_vraw_depay_reset (GstRtpVRawDepay * rtpvrawdepay) +{ + if (rtpvrawdepay->outbuf) { + gst_buffer_unref (rtpvrawdepay->outbuf); + rtpvrawdepay->outbuf = NULL; + } + rtpvrawdepay->timestamp = -1; +} + static gboolean gst_rtp_vraw_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps) { @@ -513,6 +527,28 @@ short_packet: } } +static gboolean +gst_rtp_vraw_depay_handle_event (GstBaseRTPDepayload * filter, GstEvent * event) +{ + gboolean ret; + GstRtpVRawDepay *rtpvrawdepay; + + rtpvrawdepay = GST_RTP_VRAW_DEPAY (filter); + + switch (GST_EVENT_TYPE (event)) { + case GST_EVENT_FLUSH_STOP: + gst_rtp_vraw_depay_reset (rtpvrawdepay); + break; + default: + break; + } + + ret = + GST_BASE_RTP_DEPAYLOAD_CLASS (parent_class)->handle_event (filter, event); + + return ret; +} + static GstStateChangeReturn gst_rtp_vraw_depay_change_state (GstElement * element, GstStateChange transition) @@ -526,7 +562,7 @@ gst_rtp_vraw_depay_change_state (GstElement * element, case GST_STATE_CHANGE_NULL_TO_READY: break; case GST_STATE_CHANGE_READY_TO_PAUSED: - rtpvrawdepay->timestamp = -1; + gst_rtp_vraw_depay_reset (rtpvrawdepay); break; default: break; @@ -536,10 +572,7 @@ gst_rtp_vraw_depay_change_state (GstElement * element, switch (transition) { case GST_STATE_CHANGE_PAUSED_TO_READY: - if (rtpvrawdepay->outbuf) { - gst_buffer_unref (rtpvrawdepay->outbuf); - rtpvrawdepay->outbuf = NULL; - } + gst_rtp_vraw_depay_reset (rtpvrawdepay); break; case GST_STATE_CHANGE_READY_TO_NULL: break; From f357e09ac18e995412704cb0ca98da31ffe2aaff Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Tue, 21 Dec 2010 16:49:28 +0100 Subject: [PATCH 052/351] rtp: fix rank of payloaders and depayloaders Set the payloaders and depayloaders to a reasonable rank. --- gst/rtp/gstasteriskh263.c | 2 +- gst/rtp/gstrtpL16depay.c | 2 +- gst/rtp/gstrtpL16pay.c | 2 +- gst/rtp/gstrtpac3depay.c | 2 +- gst/rtp/gstrtpamrdepay.c | 2 +- gst/rtp/gstrtpamrpay.c | 2 +- gst/rtp/gstrtpbvdepay.c | 2 +- gst/rtp/gstrtpbvpay.c | 2 +- gst/rtp/gstrtpceltdepay.c | 2 +- gst/rtp/gstrtpceltpay.c | 2 +- gst/rtp/gstrtpdepay.c | 2 +- gst/rtp/gstrtpdvdepay.c | 2 +- gst/rtp/gstrtpdvpay.c | 2 +- gst/rtp/gstrtpg722depay.c | 2 +- gst/rtp/gstrtpg722pay.c | 2 +- gst/rtp/gstrtpg723depay.c | 2 +- gst/rtp/gstrtpg723pay.c | 2 +- gst/rtp/gstrtpg726depay.c | 2 +- gst/rtp/gstrtpg726pay.c | 2 +- gst/rtp/gstrtpg729depay.c | 2 +- gst/rtp/gstrtpg729pay.c | 2 +- gst/rtp/gstrtpgsmdepay.c | 2 +- gst/rtp/gstrtpgsmpay.c | 2 +- gst/rtp/gstrtph263depay.c | 2 +- gst/rtp/gstrtph263pay.c | 2 +- gst/rtp/gstrtph263pdepay.c | 2 +- gst/rtp/gstrtph263ppay.c | 2 +- gst/rtp/gstrtph264depay.c | 2 +- gst/rtp/gstrtph264pay.c | 2 +- gst/rtp/gstrtpilbcdepay.c | 2 +- gst/rtp/gstrtpilbcpay.c | 2 +- gst/rtp/gstrtpj2kdepay.c | 2 +- gst/rtp/gstrtpj2kpay.c | 2 +- gst/rtp/gstrtpjpegdepay.c | 2 +- gst/rtp/gstrtpjpegpay.c | 2 +- gst/rtp/gstrtpmp1sdepay.c | 2 +- gst/rtp/gstrtpmp2tdepay.c | 2 +- gst/rtp/gstrtpmp2tpay.c | 2 +- gst/rtp/gstrtpmp4adepay.c | 2 +- gst/rtp/gstrtpmp4apay.c | 2 +- gst/rtp/gstrtpmp4gdepay.c | 2 +- gst/rtp/gstrtpmp4gpay.c | 2 +- gst/rtp/gstrtpmp4vdepay.c | 2 +- gst/rtp/gstrtpmp4vpay.c | 2 +- gst/rtp/gstrtpmpadepay.c | 2 +- gst/rtp/gstrtpmpapay.c | 2 +- gst/rtp/gstrtpmparobustdepay.c | 2 +- gst/rtp/gstrtpmpvdepay.c | 2 +- gst/rtp/gstrtpmpvpay.c | 2 +- gst/rtp/gstrtppcmadepay.c | 2 +- gst/rtp/gstrtppcmapay.c | 2 +- gst/rtp/gstrtppcmudepay.c | 2 +- gst/rtp/gstrtppcmupay.c | 2 +- gst/rtp/gstrtpqcelpdepay.c | 2 +- gst/rtp/gstrtpqdmdepay.c | 2 +- gst/rtp/gstrtpsirendepay.c | 2 +- gst/rtp/gstrtpsirenpay.c | 2 +- gst/rtp/gstrtpspeexdepay.c | 2 +- gst/rtp/gstrtpspeexpay.c | 2 +- gst/rtp/gstrtpsv3vdepay.c | 2 +- gst/rtp/gstrtptheoradepay.c | 2 +- gst/rtp/gstrtptheorapay.c | 2 +- gst/rtp/gstrtpvorbisdepay.c | 2 +- gst/rtp/gstrtpvorbispay.c | 2 +- gst/rtp/gstrtpvrawdepay.c | 2 +- gst/rtp/gstrtpvrawpay.c | 2 +- 66 files changed, 66 insertions(+), 66 deletions(-) diff --git a/gst/rtp/gstasteriskh263.c b/gst/rtp/gstasteriskh263.c index a1ad4dee4f..474e4c3a3b 100644 --- a/gst/rtp/gstasteriskh263.c +++ b/gst/rtp/gstasteriskh263.c @@ -231,5 +231,5 @@ gboolean gst_asteriskh263_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "asteriskh263", - GST_RANK_MARGINAL, GST_TYPE_ASTERISK_H263); + GST_RANK_SECONDARY, GST_TYPE_ASTERISK_H263); } diff --git a/gst/rtp/gstrtpL16depay.c b/gst/rtp/gstrtpL16depay.c index ac05d8b2fe..35b9a02061 100644 --- a/gst/rtp/gstrtpL16depay.c +++ b/gst/rtp/gstrtpL16depay.c @@ -261,5 +261,5 @@ gboolean gst_rtp_L16_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpL16depay", - GST_RANK_MARGINAL, GST_TYPE_RTP_L16_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_L16_DEPAY); } diff --git a/gst/rtp/gstrtpL16pay.c b/gst/rtp/gstrtpL16pay.c index 6d5ab352d2..2f3f561b71 100644 --- a/gst/rtp/gstrtpL16pay.c +++ b/gst/rtp/gstrtpL16pay.c @@ -234,5 +234,5 @@ gboolean gst_rtp_L16_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpL16pay", - GST_RANK_NONE, GST_TYPE_RTP_L16_PAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_L16_PAY); } diff --git a/gst/rtp/gstrtpac3depay.c b/gst/rtp/gstrtpac3depay.c index c1136c1f4f..e206156a51 100644 --- a/gst/rtp/gstrtpac3depay.c +++ b/gst/rtp/gstrtpac3depay.c @@ -214,5 +214,5 @@ gboolean gst_rtp_ac3_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpac3depay", - GST_RANK_MARGINAL, GST_TYPE_RTP_AC3_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_AC3_DEPAY); } diff --git a/gst/rtp/gstrtpamrdepay.c b/gst/rtp/gstrtpamrdepay.c index c08c406882..82d94cee44 100644 --- a/gst/rtp/gstrtpamrdepay.c +++ b/gst/rtp/gstrtpamrdepay.c @@ -455,5 +455,5 @@ gboolean gst_rtp_amr_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpamrdepay", - GST_RANK_MARGINAL, GST_TYPE_RTP_AMR_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_AMR_DEPAY); } diff --git a/gst/rtp/gstrtpamrpay.c b/gst/rtp/gstrtpamrpay.c index 57cfb55dfa..52a46a0259 100644 --- a/gst/rtp/gstrtpamrpay.c +++ b/gst/rtp/gstrtpamrpay.c @@ -435,5 +435,5 @@ gboolean gst_rtp_amr_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpamrpay", - GST_RANK_NONE, GST_TYPE_RTP_AMR_PAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_AMR_PAY); } diff --git a/gst/rtp/gstrtpbvdepay.c b/gst/rtp/gstrtpbvdepay.c index f398394357..2939803ea1 100644 --- a/gst/rtp/gstrtpbvdepay.c +++ b/gst/rtp/gstrtpbvdepay.c @@ -179,5 +179,5 @@ gboolean gst_rtp_bv_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpbvdepay", - GST_RANK_MARGINAL, GST_TYPE_RTP_BV_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_BV_DEPAY); } diff --git a/gst/rtp/gstrtpbvpay.c b/gst/rtp/gstrtpbvpay.c index 1ea343a7e7..a0b0173da0 100644 --- a/gst/rtp/gstrtpbvpay.c +++ b/gst/rtp/gstrtpbvpay.c @@ -219,5 +219,5 @@ gboolean gst_rtp_bv_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpbvpay", - GST_RANK_NONE, GST_TYPE_RTP_BV_PAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_BV_PAY); } diff --git a/gst/rtp/gstrtpceltdepay.c b/gst/rtp/gstrtpceltdepay.c index 0bf387b1a5..e84b9dc8a3 100644 --- a/gst/rtp/gstrtpceltdepay.c +++ b/gst/rtp/gstrtpceltdepay.c @@ -271,5 +271,5 @@ gboolean gst_rtp_celt_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpceltdepay", - GST_RANK_MARGINAL, GST_TYPE_RTP_CELT_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_CELT_DEPAY); } diff --git a/gst/rtp/gstrtpceltpay.c b/gst/rtp/gstrtpceltpay.c index 7524a27b00..fa7426b1fa 100644 --- a/gst/rtp/gstrtpceltpay.c +++ b/gst/rtp/gstrtpceltpay.c @@ -473,5 +473,5 @@ gboolean gst_rtp_celt_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpceltpay", - GST_RANK_NONE, GST_TYPE_RTP_CELT_PAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_CELT_PAY); } diff --git a/gst/rtp/gstrtpdepay.c b/gst/rtp/gstrtpdepay.c index b037efd298..1e1ded7b73 100644 --- a/gst/rtp/gstrtpdepay.c +++ b/gst/rtp/gstrtpdepay.c @@ -158,5 +158,5 @@ gboolean gst_rtp_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpdepay", - GST_RANK_NONE, GST_TYPE_RTP_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_DEPAY); } diff --git a/gst/rtp/gstrtpdvdepay.c b/gst/rtp/gstrtpdvdepay.c index ca190fa77a..eccda3e8ef 100644 --- a/gst/rtp/gstrtpdvdepay.c +++ b/gst/rtp/gstrtpdvdepay.c @@ -411,5 +411,5 @@ gboolean gst_rtp_dv_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpdvdepay", - GST_RANK_NONE, GST_TYPE_RTP_DV_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_DV_DEPAY); } diff --git a/gst/rtp/gstrtpdvpay.c b/gst/rtp/gstrtpdvpay.c index 7d0c48dc16..bb4fc5deb2 100644 --- a/gst/rtp/gstrtpdvpay.c +++ b/gst/rtp/gstrtpdvpay.c @@ -371,5 +371,5 @@ gboolean gst_rtp_dv_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpdvpay", - GST_RANK_NONE, GST_TYPE_RTP_DV_PAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_DV_PAY); } diff --git a/gst/rtp/gstrtpg722depay.c b/gst/rtp/gstrtpg722depay.c index ff479c1839..e5554f2476 100644 --- a/gst/rtp/gstrtpg722depay.c +++ b/gst/rtp/gstrtpg722depay.c @@ -256,5 +256,5 @@ gboolean gst_rtp_g722_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpg722depay", - GST_RANK_MARGINAL, GST_TYPE_RTP_G722_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_G722_DEPAY); } diff --git a/gst/rtp/gstrtpg722pay.c b/gst/rtp/gstrtpg722pay.c index 56c4a9f406..f97813b32e 100644 --- a/gst/rtp/gstrtpg722pay.c +++ b/gst/rtp/gstrtpg722pay.c @@ -206,5 +206,5 @@ gboolean gst_rtp_g722_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpg722pay", - GST_RANK_NONE, GST_TYPE_RTP_G722_PAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_G722_PAY); } diff --git a/gst/rtp/gstrtpg723depay.c b/gst/rtp/gstrtpg723depay.c index 38215d44f4..f39fe334ec 100644 --- a/gst/rtp/gstrtpg723depay.c +++ b/gst/rtp/gstrtpg723depay.c @@ -224,5 +224,5 @@ gboolean gst_rtp_g723_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpg723depay", - GST_RANK_MARGINAL, GST_TYPE_RTP_G723_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_G723_DEPAY); } diff --git a/gst/rtp/gstrtpg723pay.c b/gst/rtp/gstrtpg723pay.c index 948a8fa22c..764122ed3f 100644 --- a/gst/rtp/gstrtpg723pay.c +++ b/gst/rtp/gstrtpg723pay.c @@ -311,6 +311,6 @@ gst_rtp_g723_pay_change_state (GstElement * element, GstStateChange transition) gboolean gst_rtp_g723_pay_plugin_init (GstPlugin * plugin) { - return gst_element_register (plugin, "rtpg723pay", GST_RANK_NONE, + return gst_element_register (plugin, "rtpg723pay", GST_RANK_SECONDARY, gst_rtp_g723_pay_get_type ()); } diff --git a/gst/rtp/gstrtpg726depay.c b/gst/rtp/gstrtpg726depay.c index 5f70bc317f..aaeb9c2c35 100644 --- a/gst/rtp/gstrtpg726depay.c +++ b/gst/rtp/gstrtpg726depay.c @@ -377,5 +377,5 @@ gboolean gst_rtp_g726_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpg726depay", - GST_RANK_MARGINAL, GST_TYPE_RTP_G726_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_G726_DEPAY); } diff --git a/gst/rtp/gstrtpg726pay.c b/gst/rtp/gstrtpg726pay.c index bd1cb99dca..a702ccc547 100644 --- a/gst/rtp/gstrtpg726pay.c +++ b/gst/rtp/gstrtpg726pay.c @@ -415,5 +415,5 @@ gboolean gst_rtp_g726_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpg726pay", - GST_RANK_NONE, GST_TYPE_RTP_G726_PAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_G726_PAY); } diff --git a/gst/rtp/gstrtpg729depay.c b/gst/rtp/gstrtpg729depay.c index 6b76a6b63b..ca52579980 100644 --- a/gst/rtp/gstrtpg729depay.c +++ b/gst/rtp/gstrtpg729depay.c @@ -223,5 +223,5 @@ gboolean gst_rtp_g729_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpg729depay", - GST_RANK_MARGINAL, GST_TYPE_RTP_G729_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_G729_DEPAY); } diff --git a/gst/rtp/gstrtpg729pay.c b/gst/rtp/gstrtpg729pay.c index c6a2b403d9..21aebaabb4 100644 --- a/gst/rtp/gstrtpg729pay.c +++ b/gst/rtp/gstrtpg729pay.c @@ -399,5 +399,5 @@ gboolean gst_rtp_g729_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpg729pay", - GST_RANK_NONE, GST_TYPE_RTP_G729_PAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_G729_PAY); } diff --git a/gst/rtp/gstrtpgsmdepay.c b/gst/rtp/gstrtpgsmdepay.c index 261bc0dbdb..8f3d68bf22 100644 --- a/gst/rtp/gstrtpgsmdepay.c +++ b/gst/rtp/gstrtpgsmdepay.c @@ -148,5 +148,5 @@ gboolean gst_rtp_gsm_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpgsmdepay", - GST_RANK_MARGINAL, GST_TYPE_RTP_GSM_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_GSM_DEPAY); } diff --git a/gst/rtp/gstrtpgsmpay.c b/gst/rtp/gstrtpgsmpay.c index a7dd805f95..c207c0854c 100644 --- a/gst/rtp/gstrtpgsmpay.c +++ b/gst/rtp/gstrtpgsmpay.c @@ -173,5 +173,5 @@ gboolean gst_rtp_gsm_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpgsmpay", - GST_RANK_NONE, GST_TYPE_RTP_GSM_PAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_GSM_PAY); } diff --git a/gst/rtp/gstrtph263depay.c b/gst/rtp/gstrtph263depay.c index 8ae5a28862..0728e683bf 100644 --- a/gst/rtp/gstrtph263depay.c +++ b/gst/rtp/gstrtph263depay.c @@ -377,5 +377,5 @@ gboolean gst_rtp_h263_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtph263depay", - GST_RANK_MARGINAL, GST_TYPE_RTP_H263_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_H263_DEPAY); } diff --git a/gst/rtp/gstrtph263pay.c b/gst/rtp/gstrtph263pay.c index 5604ee0ab4..f7701c5f8d 100644 --- a/gst/rtp/gstrtph263pay.c +++ b/gst/rtp/gstrtph263pay.c @@ -1796,5 +1796,5 @@ gboolean gst_rtp_h263_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtph263pay", - GST_RANK_NONE, GST_TYPE_RTP_H263_PAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_H263_PAY); } diff --git a/gst/rtp/gstrtph263pdepay.c b/gst/rtp/gstrtph263pdepay.c index 9be2f366e4..05d9c3c6f5 100644 --- a/gst/rtp/gstrtph263pdepay.c +++ b/gst/rtp/gstrtph263pdepay.c @@ -395,5 +395,5 @@ gboolean gst_rtp_h263p_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtph263pdepay", - GST_RANK_MARGINAL, GST_TYPE_RTP_H263P_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_H263P_DEPAY); } diff --git a/gst/rtp/gstrtph263ppay.c b/gst/rtp/gstrtph263ppay.c index 0a547db7c8..2723092b45 100644 --- a/gst/rtp/gstrtph263ppay.c +++ b/gst/rtp/gstrtph263ppay.c @@ -333,5 +333,5 @@ gboolean gst_rtp_h263p_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtph263ppay", - GST_RANK_NONE, GST_TYPE_RTP_H263P_PAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_H263P_PAY); } diff --git a/gst/rtp/gstrtph264depay.c b/gst/rtp/gstrtph264depay.c index dc333ba8d4..d6ec00ca0a 100644 --- a/gst/rtp/gstrtph264depay.c +++ b/gst/rtp/gstrtph264depay.c @@ -872,5 +872,5 @@ gboolean gst_rtp_h264_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtph264depay", - GST_RANK_MARGINAL, GST_TYPE_RTP_H264_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_H264_DEPAY); } diff --git a/gst/rtp/gstrtph264pay.c b/gst/rtp/gstrtph264pay.c index d2d0a86180..fc3e4fd82b 100644 --- a/gst/rtp/gstrtph264pay.c +++ b/gst/rtp/gstrtph264pay.c @@ -1234,5 +1234,5 @@ gboolean gst_rtp_h264_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtph264pay", - GST_RANK_NONE, GST_TYPE_RTP_H264_PAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_H264_PAY); } diff --git a/gst/rtp/gstrtpilbcdepay.c b/gst/rtp/gstrtpilbcdepay.c index 381abca5fe..723835c743 100644 --- a/gst/rtp/gstrtpilbcdepay.c +++ b/gst/rtp/gstrtpilbcdepay.c @@ -232,5 +232,5 @@ gboolean gst_rtp_ilbc_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpilbcdepay", - GST_RANK_MARGINAL, GST_TYPE_RTP_ILBC_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_ILBC_DEPAY); } diff --git a/gst/rtp/gstrtpilbcpay.c b/gst/rtp/gstrtpilbcpay.c index c9381a48ac..0df5b13008 100644 --- a/gst/rtp/gstrtpilbcpay.c +++ b/gst/rtp/gstrtpilbcpay.c @@ -212,5 +212,5 @@ gboolean gst_rtp_ilbc_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpilbcpay", - GST_RANK_NONE, GST_TYPE_RTP_ILBC_PAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_ILBC_PAY); } diff --git a/gst/rtp/gstrtpj2kdepay.c b/gst/rtp/gstrtpj2kdepay.c index f864c380f4..98eeef2e93 100644 --- a/gst/rtp/gstrtpj2kdepay.c +++ b/gst/rtp/gstrtpj2kdepay.c @@ -670,5 +670,5 @@ gboolean gst_rtp_j2k_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpj2kdepay", - GST_RANK_MARGINAL, GST_TYPE_RTP_J2K_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_J2K_DEPAY); } diff --git a/gst/rtp/gstrtpj2kpay.c b/gst/rtp/gstrtpj2kpay.c index e427227aa1..73ebf25e9a 100644 --- a/gst/rtp/gstrtpj2kpay.c +++ b/gst/rtp/gstrtpj2kpay.c @@ -575,6 +575,6 @@ gst_rtp_j2k_pay_get_property (GObject * object, guint prop_id, gboolean gst_rtp_j2k_pay_plugin_init (GstPlugin * plugin) { - return gst_element_register (plugin, "rtpj2kpay", GST_RANK_NONE, + return gst_element_register (plugin, "rtpj2kpay", GST_RANK_SECONDARY, GST_TYPE_RTP_J2K_PAY); } diff --git a/gst/rtp/gstrtpjpegdepay.c b/gst/rtp/gstrtpjpegdepay.c index f2df87f7b3..dbd2446903 100644 --- a/gst/rtp/gstrtpjpegdepay.c +++ b/gst/rtp/gstrtpjpegdepay.c @@ -747,5 +747,5 @@ gboolean gst_rtp_jpeg_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpjpegdepay", - GST_RANK_MARGINAL, GST_TYPE_RTP_JPEG_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_JPEG_DEPAY); } diff --git a/gst/rtp/gstrtpjpegpay.c b/gst/rtp/gstrtpjpegpay.c index 3d09c698ff..12af2809b4 100644 --- a/gst/rtp/gstrtpjpegpay.c +++ b/gst/rtp/gstrtpjpegpay.c @@ -926,6 +926,6 @@ gst_rtp_jpeg_pay_get_property (GObject * object, guint prop_id, gboolean gst_rtp_jpeg_pay_plugin_init (GstPlugin * plugin) { - return gst_element_register (plugin, "rtpjpegpay", GST_RANK_NONE, + return gst_element_register (plugin, "rtpjpegpay", GST_RANK_SECONDARY, GST_TYPE_RTP_JPEG_PAY); } diff --git a/gst/rtp/gstrtpmp1sdepay.c b/gst/rtp/gstrtpmp1sdepay.c index 190bfca65c..af4f03c657 100644 --- a/gst/rtp/gstrtpmp1sdepay.c +++ b/gst/rtp/gstrtpmp1sdepay.c @@ -141,5 +141,5 @@ gboolean gst_rtp_mp1s_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpmp1sdepay", - GST_RANK_MARGINAL, GST_TYPE_RTP_MP1S_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_MP1S_DEPAY); } diff --git a/gst/rtp/gstrtpmp2tdepay.c b/gst/rtp/gstrtpmp2tdepay.c index 4f9ba8933f..543de0a78e 100644 --- a/gst/rtp/gstrtpmp2tdepay.c +++ b/gst/rtp/gstrtpmp2tdepay.c @@ -223,5 +223,5 @@ gboolean gst_rtp_mp2t_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpmp2tdepay", - GST_RANK_MARGINAL, GST_TYPE_RTP_MP2T_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_MP2T_DEPAY); } diff --git a/gst/rtp/gstrtpmp2tpay.c b/gst/rtp/gstrtpmp2tpay.c index 41b450e3ca..64127772a5 100644 --- a/gst/rtp/gstrtpmp2tpay.c +++ b/gst/rtp/gstrtpmp2tpay.c @@ -202,5 +202,5 @@ gboolean gst_rtp_mp2t_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpmp2tpay", - GST_RANK_NONE, GST_TYPE_RTP_MP2T_PAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_MP2T_PAY); } diff --git a/gst/rtp/gstrtpmp4adepay.c b/gst/rtp/gstrtpmp4adepay.c index f913a67cac..60d30f4d7c 100644 --- a/gst/rtp/gstrtpmp4adepay.c +++ b/gst/rtp/gstrtpmp4adepay.c @@ -374,5 +374,5 @@ gboolean gst_rtp_mp4a_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpmp4adepay", - GST_RANK_MARGINAL, GST_TYPE_RTP_MP4A_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_MP4A_DEPAY); } diff --git a/gst/rtp/gstrtpmp4apay.c b/gst/rtp/gstrtpmp4apay.c index b30b3c9558..1bb34adb4e 100644 --- a/gst/rtp/gstrtpmp4apay.c +++ b/gst/rtp/gstrtpmp4apay.c @@ -433,5 +433,5 @@ gboolean gst_rtp_mp4a_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpmp4apay", - GST_RANK_NONE, GST_TYPE_RTP_MP4A_PAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_MP4A_PAY); } diff --git a/gst/rtp/gstrtpmp4gdepay.c b/gst/rtp/gstrtpmp4gdepay.c index 3938ab3cc8..2c468b7c78 100644 --- a/gst/rtp/gstrtpmp4gdepay.c +++ b/gst/rtp/gstrtpmp4gdepay.c @@ -744,5 +744,5 @@ gboolean gst_rtp_mp4g_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpmp4gdepay", - GST_RANK_MARGINAL, GST_TYPE_RTP_MP4G_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_MP4G_DEPAY); } diff --git a/gst/rtp/gstrtpmp4gpay.c b/gst/rtp/gstrtpmp4gpay.c index 81e6f9b86d..51d8d25d01 100644 --- a/gst/rtp/gstrtpmp4gpay.c +++ b/gst/rtp/gstrtpmp4gpay.c @@ -586,5 +586,5 @@ gboolean gst_rtp_mp4g_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpmp4gpay", - GST_RANK_NONE, GST_TYPE_RTP_MP4G_PAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_MP4G_PAY); } diff --git a/gst/rtp/gstrtpmp4vdepay.c b/gst/rtp/gstrtpmp4vdepay.c index c1c911afe8..447198ad8f 100644 --- a/gst/rtp/gstrtpmp4vdepay.c +++ b/gst/rtp/gstrtpmp4vdepay.c @@ -227,5 +227,5 @@ gboolean gst_rtp_mp4v_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpmp4vdepay", - GST_RANK_MARGINAL, GST_TYPE_RTP_MP4V_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_MP4V_DEPAY); } diff --git a/gst/rtp/gstrtpmp4vpay.c b/gst/rtp/gstrtpmp4vpay.c index 2427abed21..708494f766 100644 --- a/gst/rtp/gstrtpmp4vpay.c +++ b/gst/rtp/gstrtpmp4vpay.c @@ -672,5 +672,5 @@ gboolean gst_rtp_mp4v_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpmp4vpay", - GST_RANK_NONE, GST_TYPE_RTP_MP4V_PAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_MP4V_PAY); } diff --git a/gst/rtp/gstrtpmpadepay.c b/gst/rtp/gstrtpmpadepay.c index 49c97e89c8..f04d36bdf7 100644 --- a/gst/rtp/gstrtpmpadepay.c +++ b/gst/rtp/gstrtpmpadepay.c @@ -179,5 +179,5 @@ gboolean gst_rtp_mpa_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpmpadepay", - GST_RANK_MARGINAL, GST_TYPE_RTP_MPA_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_MPA_DEPAY); } diff --git a/gst/rtp/gstrtpmpapay.c b/gst/rtp/gstrtpmpapay.c index 88d71bce89..78e0b0d63e 100644 --- a/gst/rtp/gstrtpmpapay.c +++ b/gst/rtp/gstrtpmpapay.c @@ -317,5 +317,5 @@ gboolean gst_rtp_mpa_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpmpapay", - GST_RANK_NONE, GST_TYPE_RTP_MPA_PAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_MPA_PAY); } diff --git a/gst/rtp/gstrtpmparobustdepay.c b/gst/rtp/gstrtpmparobustdepay.c index b250f4137d..61102b774b 100644 --- a/gst/rtp/gstrtpmparobustdepay.c +++ b/gst/rtp/gstrtpmparobustdepay.c @@ -788,5 +788,5 @@ gboolean gst_rtp_mpa_robust_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpmparobustdepay", - GST_RANK_MARGINAL, GST_TYPE_RTP_MPA_ROBUST_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_MPA_ROBUST_DEPAY); } diff --git a/gst/rtp/gstrtpmpvdepay.c b/gst/rtp/gstrtpmpvdepay.c index bb377eddfe..e6528d5bb4 100644 --- a/gst/rtp/gstrtpmpvdepay.c +++ b/gst/rtp/gstrtpmpvdepay.c @@ -198,5 +198,5 @@ gboolean gst_rtp_mpv_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpmpvdepay", - GST_RANK_MARGINAL, GST_TYPE_RTP_MPV_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_MPV_DEPAY); } diff --git a/gst/rtp/gstrtpmpvpay.c b/gst/rtp/gstrtpmpvpay.c index 6523dbbedb..5338c87df6 100644 --- a/gst/rtp/gstrtpmpvpay.c +++ b/gst/rtp/gstrtpmpvpay.c @@ -228,5 +228,5 @@ gboolean gst_rtp_mpv_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpmpvpay", - GST_RANK_NONE, GST_TYPE_RTP_MPV_PAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_MPV_PAY); } diff --git a/gst/rtp/gstrtppcmadepay.c b/gst/rtp/gstrtppcmadepay.c index acb5336050..3b9695d7e0 100644 --- a/gst/rtp/gstrtppcmadepay.c +++ b/gst/rtp/gstrtppcmadepay.c @@ -158,5 +158,5 @@ gboolean gst_rtp_pcma_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtppcmadepay", - GST_RANK_MARGINAL, GST_TYPE_RTP_PCMA_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_PCMA_DEPAY); } diff --git a/gst/rtp/gstrtppcmapay.c b/gst/rtp/gstrtppcmapay.c index 270f3663e3..b1531c61ba 100644 --- a/gst/rtp/gstrtppcmapay.c +++ b/gst/rtp/gstrtppcmapay.c @@ -114,5 +114,5 @@ gboolean gst_rtp_pcma_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtppcmapay", - GST_RANK_NONE, GST_TYPE_RTP_PCMA_PAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_PCMA_PAY); } diff --git a/gst/rtp/gstrtppcmudepay.c b/gst/rtp/gstrtppcmudepay.c index 9c47891f80..a94e9991b1 100644 --- a/gst/rtp/gstrtppcmudepay.c +++ b/gst/rtp/gstrtppcmudepay.c @@ -158,5 +158,5 @@ gboolean gst_rtp_pcmu_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtppcmudepay", - GST_RANK_MARGINAL, GST_TYPE_RTP_PCMU_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_PCMU_DEPAY); } diff --git a/gst/rtp/gstrtppcmupay.c b/gst/rtp/gstrtppcmupay.c index 19074ad86f..0537b8433e 100644 --- a/gst/rtp/gstrtppcmupay.c +++ b/gst/rtp/gstrtppcmupay.c @@ -114,5 +114,5 @@ gboolean gst_rtp_pcmu_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtppcmupay", - GST_RANK_NONE, GST_TYPE_RTP_PCMU_PAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_PCMU_PAY); } diff --git a/gst/rtp/gstrtpqcelpdepay.c b/gst/rtp/gstrtpqcelpdepay.c index 5499e307c4..492fe9a24f 100644 --- a/gst/rtp/gstrtpqcelpdepay.c +++ b/gst/rtp/gstrtpqcelpdepay.c @@ -431,5 +431,5 @@ gboolean gst_rtp_qcelp_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpqcelpdepay", - GST_RANK_MARGINAL, GST_TYPE_RTP_QCELP_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_QCELP_DEPAY); } diff --git a/gst/rtp/gstrtpqdmdepay.c b/gst/rtp/gstrtpqdmdepay.c index 1ab4d598f9..001b93a4f1 100644 --- a/gst/rtp/gstrtpqdmdepay.c +++ b/gst/rtp/gstrtpqdmdepay.c @@ -415,5 +415,5 @@ gst_rtp_qdm2_depay_plugin_init (GstPlugin * plugin) "RTP QDM2 depayloader"); return gst_element_register (plugin, "rtpqdm2depay", - GST_RANK_MARGINAL, GST_TYPE_RTP_QDM2_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_QDM2_DEPAY); } diff --git a/gst/rtp/gstrtpsirendepay.c b/gst/rtp/gstrtpsirendepay.c index 736160fdf1..3c46ccb7cb 100644 --- a/gst/rtp/gstrtpsirendepay.c +++ b/gst/rtp/gstrtpsirendepay.c @@ -120,5 +120,5 @@ gboolean gst_rtp_siren_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpsirendepay", - GST_RANK_MARGINAL, GST_TYPE_RTP_SIREN_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_SIREN_DEPAY); } diff --git a/gst/rtp/gstrtpsirenpay.c b/gst/rtp/gstrtpsirenpay.c index dde01f4d6c..0ddf0af5e8 100644 --- a/gst/rtp/gstrtpsirenpay.c +++ b/gst/rtp/gstrtpsirenpay.c @@ -148,5 +148,5 @@ gboolean gst_rtp_siren_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpsirenpay", - GST_RANK_NONE, GST_TYPE_RTP_SIREN_PAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_SIREN_PAY); } diff --git a/gst/rtp/gstrtpspeexdepay.c b/gst/rtp/gstrtpspeexdepay.c index fe6d0045d7..4bb10aaf26 100644 --- a/gst/rtp/gstrtpspeexdepay.c +++ b/gst/rtp/gstrtpspeexdepay.c @@ -221,5 +221,5 @@ gboolean gst_rtp_speex_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpspeexdepay", - GST_RANK_MARGINAL, GST_TYPE_RTP_SPEEX_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_SPEEX_DEPAY); } diff --git a/gst/rtp/gstrtpspeexpay.c b/gst/rtp/gstrtpspeexpay.c index 7dfc6a96d7..424070a9e0 100644 --- a/gst/rtp/gstrtpspeexpay.c +++ b/gst/rtp/gstrtpspeexpay.c @@ -332,5 +332,5 @@ gboolean gst_rtp_speex_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpspeexpay", - GST_RANK_NONE, GST_TYPE_RTP_SPEEX_PAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_SPEEX_PAY); } diff --git a/gst/rtp/gstrtpsv3vdepay.c b/gst/rtp/gstrtpsv3vdepay.c index 248ed45146..f67f9853fa 100644 --- a/gst/rtp/gstrtpsv3vdepay.c +++ b/gst/rtp/gstrtpsv3vdepay.c @@ -320,5 +320,5 @@ gst_rtp_sv3v_depay_plugin_init (GstPlugin * plugin) "RTP SV3V depayloader"); return gst_element_register (plugin, "rtpsv3vdepay", - GST_RANK_MARGINAL, GST_TYPE_RTP_SV3V_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_SV3V_DEPAY); } diff --git a/gst/rtp/gstrtptheoradepay.c b/gst/rtp/gstrtptheoradepay.c index 27ba21ce2e..fa6ef62450 100644 --- a/gst/rtp/gstrtptheoradepay.c +++ b/gst/rtp/gstrtptheoradepay.c @@ -616,5 +616,5 @@ gboolean gst_rtp_theora_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtptheoradepay", - GST_RANK_MARGINAL, GST_TYPE_RTP_THEORA_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_THEORA_DEPAY); } diff --git a/gst/rtp/gstrtptheorapay.c b/gst/rtp/gstrtptheorapay.c index 47a156103b..7fd1182848 100644 --- a/gst/rtp/gstrtptheorapay.c +++ b/gst/rtp/gstrtptheorapay.c @@ -849,5 +849,5 @@ gboolean gst_rtp_theora_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtptheorapay", - GST_RANK_NONE, GST_TYPE_RTP_THEORA_PAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_THEORA_PAY); } diff --git a/gst/rtp/gstrtpvorbisdepay.c b/gst/rtp/gstrtpvorbisdepay.c index 22f62b8f68..5cf5ef8ebc 100644 --- a/gst/rtp/gstrtpvorbisdepay.c +++ b/gst/rtp/gstrtpvorbisdepay.c @@ -696,5 +696,5 @@ gboolean gst_rtp_vorbis_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpvorbisdepay", - GST_RANK_MARGINAL, GST_TYPE_RTP_VORBIS_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_VORBIS_DEPAY); } diff --git a/gst/rtp/gstrtpvorbispay.c b/gst/rtp/gstrtpvorbispay.c index 5bce56f9ea..76f3fe84da 100644 --- a/gst/rtp/gstrtpvorbispay.c +++ b/gst/rtp/gstrtpvorbispay.c @@ -690,5 +690,5 @@ gboolean gst_rtp_vorbis_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpvorbispay", - GST_RANK_NONE, GST_TYPE_RTP_VORBIS_PAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_VORBIS_PAY); } diff --git a/gst/rtp/gstrtpvrawdepay.c b/gst/rtp/gstrtpvrawdepay.c index aa2cbfaa1c..fae1692401 100644 --- a/gst/rtp/gstrtpvrawdepay.c +++ b/gst/rtp/gstrtpvrawdepay.c @@ -586,5 +586,5 @@ gboolean gst_rtp_vraw_depay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpvrawdepay", - GST_RANK_MARGINAL, GST_TYPE_RTP_VRAW_DEPAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_VRAW_DEPAY); } diff --git a/gst/rtp/gstrtpvrawpay.c b/gst/rtp/gstrtpvrawpay.c index aee077f763..911aed34b5 100644 --- a/gst/rtp/gstrtpvrawpay.c +++ b/gst/rtp/gstrtpvrawpay.c @@ -594,5 +594,5 @@ gboolean gst_rtp_vraw_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpvrawpay", - GST_RANK_NONE, GST_TYPE_RTP_VRAW_PAY); + GST_RANK_SECONDARY, GST_TYPE_RTP_VRAW_PAY); } From f4155f3cf348bba3cca7b289c98144ee0f7ec9d7 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Tue, 21 Dec 2010 16:58:47 +0100 Subject: [PATCH 053/351] rtp: add RTP hint to the klass --- gst/rtp/gstasteriskh263.c | 2 +- gst/rtp/gstrtpL16depay.c | 2 +- gst/rtp/gstrtpL16pay.c | 2 +- gst/rtp/gstrtpac3depay.c | 2 +- gst/rtp/gstrtpamrdepay.c | 2 +- gst/rtp/gstrtpamrpay.c | 2 +- gst/rtp/gstrtpbvdepay.c | 2 +- gst/rtp/gstrtpbvpay.c | 2 +- gst/rtp/gstrtpceltdepay.c | 2 +- gst/rtp/gstrtpceltpay.c | 2 +- gst/rtp/gstrtpdepay.c | 2 +- gst/rtp/gstrtpdvdepay.c | 2 +- gst/rtp/gstrtpdvpay.c | 2 +- gst/rtp/gstrtpg722depay.c | 2 +- gst/rtp/gstrtpg722pay.c | 2 +- gst/rtp/gstrtpg723depay.c | 2 +- gst/rtp/gstrtpg723pay.c | 2 +- gst/rtp/gstrtpg726depay.c | 2 +- gst/rtp/gstrtpg726pay.c | 2 +- gst/rtp/gstrtpg729depay.c | 2 +- gst/rtp/gstrtpg729pay.c | 2 +- gst/rtp/gstrtpgsmdepay.c | 2 +- gst/rtp/gstrtpgsmpay.c | 2 +- gst/rtp/gstrtph263depay.c | 2 +- gst/rtp/gstrtph263pay.c | 2 +- gst/rtp/gstrtph263pdepay.c | 2 +- gst/rtp/gstrtph263ppay.c | 2 +- gst/rtp/gstrtph264depay.c | 2 +- gst/rtp/gstrtph264pay.c | 2 +- gst/rtp/gstrtpilbcdepay.c | 2 +- gst/rtp/gstrtpilbcpay.c | 2 +- gst/rtp/gstrtpj2kdepay.c | 2 +- gst/rtp/gstrtpj2kpay.c | 2 +- gst/rtp/gstrtpjpegdepay.c | 2 +- gst/rtp/gstrtpjpegpay.c | 2 +- gst/rtp/gstrtpmp1sdepay.c | 2 +- gst/rtp/gstrtpmp2tdepay.c | 2 +- gst/rtp/gstrtpmp2tpay.c | 2 +- gst/rtp/gstrtpmp4adepay.c | 2 +- gst/rtp/gstrtpmp4apay.c | 2 +- gst/rtp/gstrtpmp4gdepay.c | 2 +- gst/rtp/gstrtpmp4gpay.c | 2 +- gst/rtp/gstrtpmp4vdepay.c | 2 +- gst/rtp/gstrtpmp4vpay.c | 2 +- gst/rtp/gstrtpmpadepay.c | 2 +- gst/rtp/gstrtpmpapay.c | 2 +- gst/rtp/gstrtpmparobustdepay.c | 2 +- gst/rtp/gstrtpmpvdepay.c | 2 +- gst/rtp/gstrtpmpvpay.c | 2 +- gst/rtp/gstrtppcmadepay.c | 2 +- gst/rtp/gstrtppcmapay.c | 2 +- gst/rtp/gstrtppcmudepay.c | 2 +- gst/rtp/gstrtppcmupay.c | 2 +- gst/rtp/gstrtpqcelpdepay.c | 2 +- gst/rtp/gstrtpqdmdepay.c | 2 +- gst/rtp/gstrtpsirendepay.c | 2 +- gst/rtp/gstrtpsirenpay.c | 2 +- gst/rtp/gstrtpspeexdepay.c | 2 +- gst/rtp/gstrtpspeexpay.c | 2 +- gst/rtp/gstrtpsv3vdepay.c | 2 +- gst/rtp/gstrtptheoradepay.c | 2 +- gst/rtp/gstrtptheorapay.c | 2 +- gst/rtp/gstrtpvorbisdepay.c | 2 +- gst/rtp/gstrtpvorbispay.c | 2 +- gst/rtp/gstrtpvrawdepay.c | 2 +- gst/rtp/gstrtpvrawpay.c | 2 +- 66 files changed, 66 insertions(+), 66 deletions(-) diff --git a/gst/rtp/gstasteriskh263.c b/gst/rtp/gstasteriskh263.c index 474e4c3a3b..79aa32be44 100644 --- a/gst/rtp/gstasteriskh263.c +++ b/gst/rtp/gstasteriskh263.c @@ -85,7 +85,7 @@ gst_asteriskh263_base_init (gpointer klass) gst_static_pad_template_get (&gst_asteriskh263_sink_template)); gst_element_class_set_details_simple (element_class, - "RTP Asterisk H263 depayloader", "Codec/Depayloader/Network", + "RTP Asterisk H263 depayloader", "Codec/Depayloader/Network/RTP", "Extracts H263 video from RTP and encodes in Asterisk H263 format", "Neil Stratford "); } diff --git a/gst/rtp/gstrtpL16depay.c b/gst/rtp/gstrtpL16depay.c index 35b9a02061..7943f1b469 100644 --- a/gst/rtp/gstrtpL16depay.c +++ b/gst/rtp/gstrtpL16depay.c @@ -86,7 +86,7 @@ gst_rtp_L16_depay_base_init (gpointer klass) gst_static_pad_template_get (&gst_rtp_L16_depay_sink_template)); gst_element_class_set_details_simple (element_class, "RTP audio depayloader", - "Codec/Depayloader/Network", + "Codec/Depayloader/Network/RTP", "Extracts raw audio from RTP packets", "Zeeshan Ali ," "Wim Taymans "); } diff --git a/gst/rtp/gstrtpL16pay.c b/gst/rtp/gstrtpL16pay.c index 2f3f561b71..a6fbacc4b0 100644 --- a/gst/rtp/gstrtpL16pay.c +++ b/gst/rtp/gstrtpL16pay.c @@ -86,7 +86,7 @@ gst_rtp_L16_pay_base_init (gpointer klass) gst_static_pad_template_get (&gst_rtp_L16_pay_sink_template)); gst_element_class_set_details_simple (element_class, "RTP audio payloader", - "Codec/Payloader/Network", + "Codec/Payloader/Network/RTP", "Payload-encode Raw audio into RTP packets (RFC 3551)", "Wim Taymans "); } diff --git a/gst/rtp/gstrtpac3depay.c b/gst/rtp/gstrtpac3depay.c index e206156a51..80f91b3341 100644 --- a/gst/rtp/gstrtpac3depay.c +++ b/gst/rtp/gstrtpac3depay.c @@ -66,7 +66,7 @@ gst_rtp_ac3_depay_base_init (gpointer klass) gst_static_pad_template_get (&gst_rtp_ac3_depay_sink_template)); gst_element_class_set_details_simple (element_class, "RTP AC3 depayloader", - "Codec/Depayloader/Network", + "Codec/Depayloader/Network/RTP", "Extracts AC3 audio from RTP packets (RFC 4184)", "Wim Taymans "); } diff --git a/gst/rtp/gstrtpamrdepay.c b/gst/rtp/gstrtpamrdepay.c index 82d94cee44..f28904bddb 100644 --- a/gst/rtp/gstrtpamrdepay.c +++ b/gst/rtp/gstrtpamrdepay.c @@ -125,7 +125,7 @@ gst_rtp_amr_depay_base_init (gpointer klass) gst_static_pad_template_get (&gst_rtp_amr_depay_sink_template)); gst_element_class_set_details_simple (element_class, "RTP AMR depayloader", - "Codec/Depayloader/Network", + "Codec/Depayloader/Network/RTP", "Extracts AMR or AMR-WB audio from RTP packets (RFC 3267)", "Wim Taymans "); } diff --git a/gst/rtp/gstrtpamrpay.c b/gst/rtp/gstrtpamrpay.c index 52a46a0259..89a149fe26 100644 --- a/gst/rtp/gstrtpamrpay.c +++ b/gst/rtp/gstrtpamrpay.c @@ -107,7 +107,7 @@ gst_rtp_amr_pay_base_init (gpointer klass) gst_static_pad_template_get (&gst_rtp_amr_pay_sink_template)); gst_element_class_set_details_simple (element_class, "RTP AMR payloader", - "Codec/Payloader/Network", + "Codec/Payloader/Network/RTP", "Payload-encode AMR or AMR-WB audio into RTP packets (RFC 3267)", "Wim Taymans "); } diff --git a/gst/rtp/gstrtpbvdepay.c b/gst/rtp/gstrtpbvdepay.c index 2939803ea1..00c145267f 100644 --- a/gst/rtp/gstrtpbvdepay.c +++ b/gst/rtp/gstrtpbvdepay.c @@ -67,7 +67,7 @@ gst_rtp_bv_depay_base_init (gpointer klass) gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_bv_depay_sink_template)); gst_element_class_set_details_simple (element_class, - "RTP BroadcomVoice depayloader", "Codec/Depayloader/Network", + "RTP BroadcomVoice depayloader", "Codec/Depayloader/Network/RTP", "Extracts BroadcomVoice audio from RTP packets (RFC 4298)", "Wim Taymans "); } diff --git a/gst/rtp/gstrtpbvpay.c b/gst/rtp/gstrtpbvpay.c index a0b0173da0..7ddc5781c9 100644 --- a/gst/rtp/gstrtpbvpay.c +++ b/gst/rtp/gstrtpbvpay.c @@ -71,7 +71,7 @@ gst_rtp_bv_pay_base_init (gpointer klass) gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_bv_pay_src_template)); gst_element_class_set_details_simple (element_class, "RTP BV Payloader", - "Codec/Payloader/Network", + "Codec/Payloader/Network/RTP", "Packetize BroadcomVoice audio streams into RTP packets (RFC 4298)", "Wim Taymans "); } diff --git a/gst/rtp/gstrtpceltdepay.c b/gst/rtp/gstrtpceltdepay.c index e84b9dc8a3..c03a5b4825 100644 --- a/gst/rtp/gstrtpceltdepay.c +++ b/gst/rtp/gstrtpceltdepay.c @@ -83,7 +83,7 @@ gst_rtp_celt_depay_base_init (gpointer klass) gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_celt_depay_sink_template)); gst_element_class_set_details_simple (element_class, "RTP CELT depayloader", - "Codec/Depayloader/Network", + "Codec/Depayloader/Network/RTP", "Extracts CELT audio from RTP packets", "Wim Taymans "); diff --git a/gst/rtp/gstrtpceltpay.c b/gst/rtp/gstrtpceltpay.c index fa7426b1fa..05715c10aa 100644 --- a/gst/rtp/gstrtpceltpay.c +++ b/gst/rtp/gstrtpceltpay.c @@ -75,7 +75,7 @@ gst_rtp_celt_pay_base_init (gpointer klass) gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_celt_pay_src_template)); gst_element_class_set_details_simple (element_class, "RTP CELT payloader", - "Codec/Payloader/Network", + "Codec/Payloader/Network/RTP", "Payload-encodes CELT audio into a RTP packet", "Wim Taymans "); diff --git a/gst/rtp/gstrtpdepay.c b/gst/rtp/gstrtpdepay.c index 1e1ded7b73..d94d9b1964 100644 --- a/gst/rtp/gstrtpdepay.c +++ b/gst/rtp/gstrtpdepay.c @@ -72,7 +72,7 @@ gst_rtp_depay_base_init (gpointer klass) gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&gst_rtp_depay_sink_rtcp_template)); gst_element_class_set_details_simple (gstelement_class, - "Dummy RTP session manager", "Codec/Depayloader/Network", + "Dummy RTP session manager", "Codec/Depayloader/Network/RTP", "Accepts raw RTP and RTCP packets and sends them forward", "Wim Taymans "); } diff --git a/gst/rtp/gstrtpdvdepay.c b/gst/rtp/gstrtpdvdepay.c index eccda3e8ef..883d1152a2 100644 --- a/gst/rtp/gstrtpdvdepay.c +++ b/gst/rtp/gstrtpdvdepay.c @@ -93,7 +93,7 @@ GST_BOILERPLATE (GstRTPDVDepay, gst_rtp_dv_depay, GstBaseRTPDepayload, gst_static_pad_template_get (&sink_factory)); gst_element_class_set_details_simple (element_class, "RTP DV Depayloader", - "Codec/Depayloader/Network", + "Codec/Depayloader/Network/RTP", "Depayloads DV from RTP packets (RFC 3189)", "Marcel Moreaux , Wim Taymans "); } diff --git a/gst/rtp/gstrtpdvpay.c b/gst/rtp/gstrtpdvpay.c index bb4fc5deb2..1d6230ccf9 100644 --- a/gst/rtp/gstrtpdvpay.c +++ b/gst/rtp/gstrtpdvpay.c @@ -106,7 +106,7 @@ GST_BOILERPLATE (GstRTPDVPay, gst_rtp_dv_pay, GstBaseRTPPayload, gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_dv_pay_src_template)); gst_element_class_set_details_simple (element_class, "RTP DV Payloader", - "Codec/Payloader/Network", + "Codec/Payloader/Network/RTP", "Payloads DV into RTP packets (RFC 3189)", "Marcel Moreaux , Wim Taymans "); } diff --git a/gst/rtp/gstrtpg722depay.c b/gst/rtp/gstrtpg722depay.c index e5554f2476..1e892fbc18 100644 --- a/gst/rtp/gstrtpg722depay.c +++ b/gst/rtp/gstrtpg722depay.c @@ -81,7 +81,7 @@ gst_rtp_g722_depay_base_init (gpointer klass) gst_static_pad_template_get (&gst_rtp_g722_depay_sink_template)); gst_element_class_set_details_simple (element_class, "RTP audio depayloader", - "Codec/Depayloader/Network", + "Codec/Depayloader/Network/RTP", "Extracts G722 audio from RTP packets", "Wim Taymans "); } diff --git a/gst/rtp/gstrtpg722pay.c b/gst/rtp/gstrtpg722pay.c index f97813b32e..7346ca149a 100644 --- a/gst/rtp/gstrtpg722pay.c +++ b/gst/rtp/gstrtpg722pay.c @@ -70,7 +70,7 @@ gst_rtp_g722_pay_base_init (gpointer klass) gst_static_pad_template_get (&gst_rtp_g722_pay_sink_template)); gst_element_class_set_details_simple (element_class, "RTP audio payloader", - "Codec/Payloader/Network", + "Codec/Payloader/Network/RTP", "Payload-encode Raw audio into RTP packets (RFC 3551)", "Wim Taymans "); } diff --git a/gst/rtp/gstrtpg723depay.c b/gst/rtp/gstrtpg723depay.c index f39fe334ec..b4aa5fe538 100644 --- a/gst/rtp/gstrtpg723depay.c +++ b/gst/rtp/gstrtpg723depay.c @@ -92,7 +92,7 @@ gst_rtp_g723_depay_base_init (gpointer klass) gst_static_pad_template_get (&gst_rtp_g723_depay_sink_template)); gst_element_class_set_details_simple (element_class, "RTP G.723 depayloader", - "Codec/Depayloader/Network", + "Codec/Depayloader/Network/RTP", "Extracts G.723 audio from RTP packets (RFC 3551)", "Wim Taymans "); diff --git a/gst/rtp/gstrtpg723pay.c b/gst/rtp/gstrtpg723pay.c index 764122ed3f..439ef0284e 100644 --- a/gst/rtp/gstrtpg723pay.c +++ b/gst/rtp/gstrtpg723pay.c @@ -80,7 +80,7 @@ gst_rtp_g723_pay_base_init (gpointer klass) gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_g723_pay_src_template)); gst_element_class_set_details_simple (element_class, "RTP G.723 payloader", - "Codec/Payloader/Network", + "Codec/Payloader/Network/RTP", "Packetize G.723 audio into RTP packets", "Wim Taymans "); } diff --git a/gst/rtp/gstrtpg726depay.c b/gst/rtp/gstrtpg726depay.c index aaeb9c2c35..2b36755e71 100644 --- a/gst/rtp/gstrtpg726depay.c +++ b/gst/rtp/gstrtpg726depay.c @@ -100,7 +100,7 @@ gst_rtp_g726_depay_base_init (gpointer klass) gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_g726_depay_sink_template)); gst_element_class_set_details_simple (element_class, "RTP G.726 depayloader", - "Codec/Depayloader/Network", + "Codec/Depayloader/Network/RTP", "Extracts G.726 audio from RTP packets", "Axis Communications "); } diff --git a/gst/rtp/gstrtpg726pay.c b/gst/rtp/gstrtpg726pay.c index a702ccc547..16086bfd68 100644 --- a/gst/rtp/gstrtpg726pay.c +++ b/gst/rtp/gstrtpg726pay.c @@ -88,7 +88,7 @@ gst_rtp_g726_pay_base_init (gpointer klass) gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_g726_pay_src_template)); gst_element_class_set_details_simple (element_class, "RTP G.726 payloader", - "Codec/Payloader/Network", + "Codec/Payloader/Network/RTP", "Payload-encodes G.726 audio into a RTP packet", "Axis Communications "); } diff --git a/gst/rtp/gstrtpg729depay.c b/gst/rtp/gstrtpg729depay.c index ca52579980..3d22508622 100644 --- a/gst/rtp/gstrtpg729depay.c +++ b/gst/rtp/gstrtpg729depay.c @@ -90,7 +90,7 @@ gst_rtp_g729_depay_base_init (gpointer klass) gst_static_pad_template_get (&gst_rtp_g729_depay_sink_template)); gst_element_class_set_details_simple (element_class, "RTP G.729 depayloader", - "Codec/Depayloader/Network", + "Codec/Depayloader/Network/RTP", "Extracts G.729 audio from RTP packets (RFC 3551)", "Laurent Glayal "); diff --git a/gst/rtp/gstrtpg729pay.c b/gst/rtp/gstrtpg729pay.c index 21aebaabb4..37049d638d 100644 --- a/gst/rtp/gstrtpg729pay.c +++ b/gst/rtp/gstrtpg729pay.c @@ -87,7 +87,7 @@ gst_rtp_g729_pay_base_init (gpointer klass) gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_g729_pay_src_template)); gst_element_class_set_details_simple (element_class, "RTP G.729 payloader", - "Codec/Payloader/Network", + "Codec/Payloader/Network/RTP", "Packetize G.729 audio into RTP packets", "Olivier Crete "); diff --git a/gst/rtp/gstrtpgsmdepay.c b/gst/rtp/gstrtpgsmdepay.c index 8f3d68bf22..bb62c50917 100644 --- a/gst/rtp/gstrtpgsmdepay.c +++ b/gst/rtp/gstrtpgsmdepay.c @@ -75,7 +75,7 @@ gst_rtp_gsm_depay_base_init (gpointer klass) gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_gsm_depay_sink_template)); gst_element_class_set_details_simple (element_class, "RTP GSM depayloader", - "Codec/Depayloader/Network", + "Codec/Depayloader/Network/RTP", "Extracts GSM audio from RTP packets", "Zeeshan Ali "); } diff --git a/gst/rtp/gstrtpgsmpay.c b/gst/rtp/gstrtpgsmpay.c index c207c0854c..ad984bcde1 100644 --- a/gst/rtp/gstrtpgsmpay.c +++ b/gst/rtp/gstrtpgsmpay.c @@ -70,7 +70,7 @@ gst_rtp_gsm_pay_base_init (gpointer klass) gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_gsm_pay_src_template)); gst_element_class_set_details_simple (element_class, "RTP GSM payloader", - "Codec/Payloader/Network", + "Codec/Payloader/Network/RTP", "Payload-encodes GSM audio into a RTP packet", "Zeeshan Ali "); } diff --git a/gst/rtp/gstrtph263depay.c b/gst/rtp/gstrtph263depay.c index 0728e683bf..4bc8704265 100644 --- a/gst/rtp/gstrtph263depay.c +++ b/gst/rtp/gstrtph263depay.c @@ -85,7 +85,7 @@ gst_rtp_h263_depay_base_init (gpointer klass) gst_static_pad_template_get (&gst_rtp_h263_depay_sink_template)); gst_element_class_set_details_simple (element_class, "RTP H263 depayloader", - "Codec/Depayloader/Network", + "Codec/Depayloader/Network/RTP", "Extracts H263 video from RTP packets (RFC 2190)", "Philippe Kalaf , " "Edward Hervey "); diff --git a/gst/rtp/gstrtph263pay.c b/gst/rtp/gstrtph263pay.c index f7701c5f8d..8eb85bc8cd 100644 --- a/gst/rtp/gstrtph263pay.c +++ b/gst/rtp/gstrtph263pay.c @@ -408,7 +408,7 @@ GST_BOILERPLATE (GstRtpH263Pay, gst_rtp_h263_pay, GstBaseRTPPayload, gst_static_pad_template_get (&gst_rtp_h263_pay_sink_template)); gst_element_class_set_details_simple (element_class, - "RTP H263 packet payloader", "Codec/Payloader/Network", + "RTP H263 packet payloader", "Codec/Payloader/Network/RTP", "Payload-encodes H263 video in RTP packets (RFC 2190)", "Neil Stratford " "Dejan Sakelsak "); diff --git a/gst/rtp/gstrtph263pdepay.c b/gst/rtp/gstrtph263pdepay.c index 05d9c3c6f5..5cad4bf91b 100644 --- a/gst/rtp/gstrtph263pdepay.c +++ b/gst/rtp/gstrtph263pdepay.c @@ -103,7 +103,7 @@ gst_rtp_h263p_depay_base_init (gpointer klass) gst_element_class_set_details_simple (element_class, "RTP H263 depayloader", - "Codec/Depayloader/Network", + "Codec/Depayloader/Network/RTP", "Extracts H263/+/++ video from RTP packets (RFC 4629)", "Wim Taymans "); } diff --git a/gst/rtp/gstrtph263ppay.c b/gst/rtp/gstrtph263ppay.c index 2723092b45..fff68ecc4a 100644 --- a/gst/rtp/gstrtph263ppay.c +++ b/gst/rtp/gstrtph263ppay.c @@ -104,7 +104,7 @@ gst_rtp_h263p_pay_base_init (gpointer klass) gst_static_pad_template_get (&gst_rtp_h263p_pay_sink_template)); gst_element_class_set_details_simple (element_class, "RTP H263 payloader", - "Codec/Payloader/Network", + "Codec/Payloader/Network/RTP", "Payload-encodes H263/+/++ video in RTP packets (RFC 4629)", "Wim Taymans "); } diff --git a/gst/rtp/gstrtph264depay.c b/gst/rtp/gstrtph264depay.c index d6ec00ca0a..20261c5803 100644 --- a/gst/rtp/gstrtph264depay.c +++ b/gst/rtp/gstrtph264depay.c @@ -107,7 +107,7 @@ gst_rtp_h264_depay_base_init (gpointer klass) gst_static_pad_template_get (&gst_rtp_h264_depay_sink_template)); gst_element_class_set_details_simple (element_class, "RTP H264 depayloader", - "Codec/Depayloader/Network", + "Codec/Depayloader/Network/RTP", "Extracts H264 video from RTP packets (RFC 3984)", "Wim Taymans "); } diff --git a/gst/rtp/gstrtph264pay.c b/gst/rtp/gstrtph264pay.c index fc3e4fd82b..15b04705d5 100644 --- a/gst/rtp/gstrtph264pay.c +++ b/gst/rtp/gstrtph264pay.c @@ -130,7 +130,7 @@ gst_rtp_h264_pay_base_init (gpointer klass) gst_static_pad_template_get (&gst_rtp_h264_pay_sink_template)); gst_element_class_set_details_simple (element_class, "RTP H264 payloader", - "Codec/Payloader/Network", + "Codec/Payloader/Network/RTP", "Payload-encode H264 video into RTP packets (RFC 3984)", "Laurent Glayal "); } diff --git a/gst/rtp/gstrtpilbcdepay.c b/gst/rtp/gstrtpilbcdepay.c index 723835c743..18ca426a7c 100644 --- a/gst/rtp/gstrtpilbcdepay.c +++ b/gst/rtp/gstrtpilbcdepay.c @@ -101,7 +101,7 @@ gst_rtp_ilbc_depay_base_init (gpointer klass) gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_ilbc_depay_sink_template)); gst_element_class_set_details_simple (element_class, "RTP iLBC depayloader", - "Codec/Depayloader/Network", + "Codec/Depayloader/Network/RTP", "Extracts iLBC audio from RTP packets (RFC 3952)", "Philippe Kalaf "); } diff --git a/gst/rtp/gstrtpilbcpay.c b/gst/rtp/gstrtpilbcpay.c index 0df5b13008..c070d6ca10 100644 --- a/gst/rtp/gstrtpilbcpay.c +++ b/gst/rtp/gstrtpilbcpay.c @@ -66,7 +66,7 @@ gst_rtp_ilbc_pay_base_init (gpointer klass) gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_ilbc_pay_src_template)); gst_element_class_set_details_simple (element_class, "RTP iLBC Payloader", - "Codec/Payloader/Network", + "Codec/Payloader/Network/RTP", "Packetize iLBC audio streams into RTP packets", "Philippe Kalaf "); } diff --git a/gst/rtp/gstrtpj2kdepay.c b/gst/rtp/gstrtpj2kdepay.c index 98eeef2e93..4ac7c34d5f 100644 --- a/gst/rtp/gstrtpj2kdepay.c +++ b/gst/rtp/gstrtpj2kdepay.c @@ -95,7 +95,7 @@ gst_rtp_j2k_depay_base_init (gpointer klass) gst_static_pad_template_get (&gst_rtp_j2k_depay_sink_template)); gst_element_class_set_details_simple (element_class, - "RTP JPEG 2000 depayloader", "Codec/Depayloader/Network", + "RTP JPEG 2000 depayloader", "Codec/Depayloader/Network/RTP", "Extracts JPEG 2000 video from RTP packets (RFC 5371)", "Wim Taymans "); } diff --git a/gst/rtp/gstrtpj2kpay.c b/gst/rtp/gstrtpj2kpay.c index 73ebf25e9a..7385d3bdcb 100644 --- a/gst/rtp/gstrtpj2kpay.c +++ b/gst/rtp/gstrtpj2kpay.c @@ -124,7 +124,7 @@ gst_rtp_j2k_pay_base_init (gpointer klass) gst_static_pad_template_get (&gst_rtp_j2k_pay_sink_template)); gst_element_class_set_details_simple (element_class, - "RTP JPEG 2000 payloader", "Codec/Payloader/Network", + "RTP JPEG 2000 payloader", "Codec/Payloader/Network/RTP", "Payload-encodes JPEG 2000 pictures into RTP packets (RFC 5371)", "Wim Taymans "); } diff --git a/gst/rtp/gstrtpjpegdepay.c b/gst/rtp/gstrtpjpegdepay.c index dbd2446903..e5c457f533 100644 --- a/gst/rtp/gstrtpjpegdepay.c +++ b/gst/rtp/gstrtpjpegdepay.c @@ -90,7 +90,7 @@ gst_rtp_jpeg_depay_base_init (gpointer klass) gst_static_pad_template_get (&gst_rtp_jpeg_depay_sink_template)); gst_element_class_set_details_simple (element_class, "RTP JPEG depayloader", - "Codec/Depayloader/Network", + "Codec/Depayloader/Network/RTP", "Extracts JPEG video from RTP packets (RFC 2435)", "Wim Taymans "); } diff --git a/gst/rtp/gstrtpjpegpay.c b/gst/rtp/gstrtpjpegpay.c index 12af2809b4..fc4096d5c2 100644 --- a/gst/rtp/gstrtpjpegpay.c +++ b/gst/rtp/gstrtpjpegpay.c @@ -248,7 +248,7 @@ gst_rtp_jpeg_pay_base_init (gpointer klass) gst_static_pad_template_get (&gst_rtp_jpeg_pay_sink_template)); gst_element_class_set_details_simple (element_class, "RTP JPEG payloader", - "Codec/Payloader/Network", + "Codec/Payloader/Network/RTP", "Payload-encodes JPEG pictures into RTP packets (RFC 2435)", "Axis Communications "); } diff --git a/gst/rtp/gstrtpmp1sdepay.c b/gst/rtp/gstrtpmp1sdepay.c index af4f03c657..4887c4cdea 100644 --- a/gst/rtp/gstrtpmp1sdepay.c +++ b/gst/rtp/gstrtpmp1sdepay.c @@ -81,7 +81,7 @@ gst_rtp_mp1s_depay_base_init (gpointer klass) gst_static_pad_template_get (&gst_rtp_mp1s_depay_sink_template)); gst_element_class_set_details_simple (element_class, - "RTP MPEG1 System Stream depayloader", "Codec/Depayloader/Network", + "RTP MPEG1 System Stream depayloader", "Codec/Depayloader/Network/RTP", "Extracts MPEG1 System Streams from RTP packets (RFC 3555)", "Wim Taymans "); } diff --git a/gst/rtp/gstrtpmp2tdepay.c b/gst/rtp/gstrtpmp2tdepay.c index 543de0a78e..fe6c5dee2a 100644 --- a/gst/rtp/gstrtpmp2tdepay.c +++ b/gst/rtp/gstrtpmp2tdepay.c @@ -92,7 +92,7 @@ gst_rtp_mp2t_depay_base_init (gpointer klass) gst_static_pad_template_get (&gst_rtp_mp2t_depay_sink_template)); gst_element_class_set_details_simple (element_class, - "RTP MPEG Transport Stream depayloader", "Codec/Depayloader/Network", + "RTP MPEG Transport Stream depayloader", "Codec/Depayloader/Network/RTP", "Extracts MPEG2 TS from RTP packets (RFC 2250)", "Wim Taymans , " "Thijs Vermeir "); diff --git a/gst/rtp/gstrtpmp2tpay.c b/gst/rtp/gstrtpmp2tpay.c index 64127772a5..3ddb5cb793 100644 --- a/gst/rtp/gstrtpmp2tpay.c +++ b/gst/rtp/gstrtpmp2tpay.c @@ -65,7 +65,7 @@ gst_rtp_mp2t_pay_base_init (gpointer klass) gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_mp2t_pay_src_template)); gst_element_class_set_details_simple (element_class, - "RTP MPEG2 Transport Stream payloader", "Codec/Payloader/Network", + "RTP MPEG2 Transport Stream payloader", "Codec/Payloader/Network/RTP", "Payload-encodes MPEG2 TS into RTP packets (RFC 2250)", "Wim Taymans "); } diff --git a/gst/rtp/gstrtpmp4adepay.c b/gst/rtp/gstrtpmp4adepay.c index 60d30f4d7c..3579d900b8 100644 --- a/gst/rtp/gstrtpmp4adepay.c +++ b/gst/rtp/gstrtpmp4adepay.c @@ -80,7 +80,7 @@ gst_rtp_mp4a_depay_base_init (gpointer klass) gst_static_pad_template_get (&gst_rtp_mp4a_depay_sink_template)); gst_element_class_set_details_simple (element_class, - "RTP MPEG4 audio depayloader", "Codec/Depayloader/Network", + "RTP MPEG4 audio depayloader", "Codec/Depayloader/Network/RTP", "Extracts MPEG4 audio from RTP packets (RFC 3016)", "Nokia Corporation (contact ), " "Wim Taymans "); diff --git a/gst/rtp/gstrtpmp4apay.c b/gst/rtp/gstrtpmp4apay.c index 1bb34adb4e..70fdbed7a5 100644 --- a/gst/rtp/gstrtpmp4apay.c +++ b/gst/rtp/gstrtpmp4apay.c @@ -77,7 +77,7 @@ GST_BOILERPLATE (GstRtpMP4APay, gst_rtp_mp4a_pay, GstBaseRTPPayload, gst_static_pad_template_get (&gst_rtp_mp4a_pay_sink_template)); gst_element_class_set_details_simple (element_class, - "RTP MPEG4 audio payloader", "Codec/Payloader/Network", + "RTP MPEG4 audio payloader", "Codec/Payloader/Network/RTP", "Payload MPEG4 audio as RTP packets (RFC 3016)", "Wim Taymans "); } diff --git a/gst/rtp/gstrtpmp4gdepay.c b/gst/rtp/gstrtpmp4gdepay.c index 2c468b7c78..67d5217ec6 100644 --- a/gst/rtp/gstrtpmp4gdepay.c +++ b/gst/rtp/gstrtpmp4gdepay.c @@ -151,7 +151,7 @@ gst_rtp_mp4g_depay_base_init (gpointer klass) gst_static_pad_template_get (&gst_rtp_mp4g_depay_sink_template)); gst_element_class_set_details_simple (element_class, - "RTP MPEG4 ES depayloader", "Codec/Depayloader/Network", + "RTP MPEG4 ES depayloader", "Codec/Depayloader/Network/RTP", "Extracts MPEG4 elementary streams from RTP packets (RFC 3640)", "Wim Taymans "); } diff --git a/gst/rtp/gstrtpmp4gpay.c b/gst/rtp/gstrtpmp4gpay.c index 51d8d25d01..764a8038d1 100644 --- a/gst/rtp/gstrtpmp4gpay.c +++ b/gst/rtp/gstrtpmp4gpay.c @@ -96,7 +96,7 @@ GST_BOILERPLATE (GstRtpMP4GPay, gst_rtp_mp4g_pay, GstBaseRTPPayload, gst_static_pad_template_get (&gst_rtp_mp4g_pay_sink_template)); gst_element_class_set_details_simple (element_class, "RTP MPEG4 ES payloader", - "Codec/Payloader/Network", + "Codec/Payloader/Network/RTP", "Payload MPEG4 elementary streams as RTP packets (RFC 3640)", "Wim Taymans "); } diff --git a/gst/rtp/gstrtpmp4vdepay.c b/gst/rtp/gstrtpmp4vdepay.c index 447198ad8f..ce18cecdd5 100644 --- a/gst/rtp/gstrtpmp4vdepay.c +++ b/gst/rtp/gstrtpmp4vdepay.c @@ -78,7 +78,7 @@ gst_rtp_mp4v_depay_base_init (gpointer klass) gst_static_pad_template_get (&gst_rtp_mp4v_depay_sink_template)); gst_element_class_set_details_simple (element_class, - "RTP MPEG4 video depayloader", "Codec/Depayloader/Network", + "RTP MPEG4 video depayloader", "Codec/Depayloader/Network/RTP", "Extracts MPEG4 video from RTP packets (RFC 3016)", "Wim Taymans "); } diff --git a/gst/rtp/gstrtpmp4vpay.c b/gst/rtp/gstrtpmp4vpay.c index 708494f766..811e1b4a58 100644 --- a/gst/rtp/gstrtpmp4vpay.c +++ b/gst/rtp/gstrtpmp4vpay.c @@ -93,7 +93,7 @@ GST_BOILERPLATE (GstRtpMP4VPay, gst_rtp_mp4v_pay, GstBaseRTPPayload, gst_static_pad_template_get (&gst_rtp_mp4v_pay_sink_template)); gst_element_class_set_details_simple (element_class, - "RTP MPEG4 Video payloader", "Codec/Payloader/Network", + "RTP MPEG4 Video payloader", "Codec/Payloader/Network/RTP", "Payload MPEG-4 video as RTP packets (RFC 3016)", "Wim Taymans "); } diff --git a/gst/rtp/gstrtpmpadepay.c b/gst/rtp/gstrtpmpadepay.c index f04d36bdf7..a76a7f39e6 100644 --- a/gst/rtp/gstrtpmpadepay.c +++ b/gst/rtp/gstrtpmpadepay.c @@ -69,7 +69,7 @@ gst_rtp_mpa_depay_base_init (gpointer klass) gst_static_pad_template_get (&gst_rtp_mpa_depay_sink_template)); gst_element_class_set_details_simple (element_class, - "RTP MPEG audio depayloader", "Codec/Depayloader/Network", + "RTP MPEG audio depayloader", "Codec/Depayloader/Network/RTP", "Extracts MPEG audio from RTP packets (RFC 2038)", "Wim Taymans "); } diff --git a/gst/rtp/gstrtpmpapay.c b/gst/rtp/gstrtpmpapay.c index 78e0b0d63e..9f17da11f9 100644 --- a/gst/rtp/gstrtpmpapay.c +++ b/gst/rtp/gstrtpmpapay.c @@ -73,7 +73,7 @@ GST_BOILERPLATE (GstRtpMPAPay, gst_rtp_mpa_pay, GstBaseRTPPayload, gst_static_pad_template_get (&gst_rtp_mpa_pay_sink_template)); gst_element_class_set_details_simple (element_class, - "RTP MPEG audio payloader", "Codec/Payloader/Network", + "RTP MPEG audio payloader", "Codec/Payloader/Network/RTP", "Payload MPEG audio as RTP packets (RFC 2038)", "Wim Taymans "); } diff --git a/gst/rtp/gstrtpmparobustdepay.c b/gst/rtp/gstrtpmparobustdepay.c index 61102b774b..0f3c04959c 100644 --- a/gst/rtp/gstrtpmparobustdepay.c +++ b/gst/rtp/gstrtpmparobustdepay.c @@ -91,7 +91,7 @@ gst_rtp_mpa_robust_depay_base_init (gpointer klass) gst_static_pad_template_get (&gst_rtp_mpa_robust_depay_sink_template)); gst_element_class_set_details_simple (element_class, - "RTP MPEG audio depayloader", "Codec/Depayloader/Network", + "RTP MPEG audio depayloader", "Codec/Depayloader/Network/RTP", "Extracts MPEG audio from RTP packets (RFC 5219)", "Mark Nauwelaerts "); } diff --git a/gst/rtp/gstrtpmpvdepay.c b/gst/rtp/gstrtpmpvdepay.c index e6528d5bb4..f9786048d4 100644 --- a/gst/rtp/gstrtpmpvdepay.c +++ b/gst/rtp/gstrtpmpvdepay.c @@ -72,7 +72,7 @@ gst_rtp_mpv_depay_base_init (gpointer klass) gst_static_pad_template_get (&gst_rtp_mpv_depay_sink_template)); gst_element_class_set_details_simple (element_class, - "RTP MPEG video depayloader", "Codec/Depayloader/Network", + "RTP MPEG video depayloader", "Codec/Depayloader/Network/RTP", "Extracts MPEG video from RTP packets (RFC 2250)", "Wim Taymans "); } diff --git a/gst/rtp/gstrtpmpvpay.c b/gst/rtp/gstrtpmpvpay.c index 5338c87df6..7cca09eccb 100644 --- a/gst/rtp/gstrtpmpvpay.c +++ b/gst/rtp/gstrtpmpvpay.c @@ -68,7 +68,7 @@ gst_rtp_mpv_pay_base_init (gpointer klass) gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_mpv_pay_src_template)); gst_element_class_set_details_simple (element_class, - "RTP MPEG2 ES video payloader", "Codec/Payloader/Network", + "RTP MPEG2 ES video payloader", "Codec/Payloader/Network/RTP", "Payload-encodes MPEG2 ES into RTP packets (RFC 2250)", "Thijs Vermeir "); } diff --git a/gst/rtp/gstrtppcmadepay.c b/gst/rtp/gstrtppcmadepay.c index 3b9695d7e0..35fc0a9891 100644 --- a/gst/rtp/gstrtppcmadepay.c +++ b/gst/rtp/gstrtppcmadepay.c @@ -78,7 +78,7 @@ gst_rtp_pcma_depay_base_init (gpointer klass) gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_pcma_depay_sink_template)); gst_element_class_set_details_simple (element_class, "RTP PCMA depayloader", - "Codec/Depayloader/Network", + "Codec/Depayloader/Network/RTP", "Extracts PCMA audio from RTP packets", "Edgard Lima , Zeeshan Ali "); } diff --git a/gst/rtp/gstrtppcmapay.c b/gst/rtp/gstrtppcmapay.c index b1531c61ba..97b31a9d31 100644 --- a/gst/rtp/gstrtppcmapay.c +++ b/gst/rtp/gstrtppcmapay.c @@ -66,7 +66,7 @@ gst_rtp_pcma_pay_base_init (gpointer klass) gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_pcma_pay_src_template)); gst_element_class_set_details_simple (element_class, "RTP PCMA payloader", - "Codec/Payloader/Network", + "Codec/Payloader/Network/RTP", "Payload-encodes PCMA audio into a RTP packet", "Edgard Lima "); } diff --git a/gst/rtp/gstrtppcmudepay.c b/gst/rtp/gstrtppcmudepay.c index a94e9991b1..fd1f1e5021 100644 --- a/gst/rtp/gstrtppcmudepay.c +++ b/gst/rtp/gstrtppcmudepay.c @@ -78,7 +78,7 @@ gst_rtp_pcmu_depay_base_init (gpointer klass) gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_pcmu_depay_sink_template)); gst_element_class_set_details_simple (element_class, "RTP PCMU depayloader", - "Codec/Depayloader/Network", + "Codec/Depayloader/Network/RTP", "Extracts PCMU audio from RTP packets", "Edgard Lima , Zeeshan Ali "); } diff --git a/gst/rtp/gstrtppcmupay.c b/gst/rtp/gstrtppcmupay.c index 0537b8433e..e7a0995539 100644 --- a/gst/rtp/gstrtppcmupay.c +++ b/gst/rtp/gstrtppcmupay.c @@ -66,7 +66,7 @@ gst_rtp_pcmu_pay_base_init (gpointer klass) gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_pcmu_pay_src_template)); gst_element_class_set_details_simple (element_class, "RTP PCMU payloader", - "Codec/Payloader/Network", + "Codec/Payloader/Network/RTP", "Payload-encodes PCMU audio into a RTP packet", "Edgard Lima "); } diff --git a/gst/rtp/gstrtpqcelpdepay.c b/gst/rtp/gstrtpqcelpdepay.c index 492fe9a24f..badeb9f244 100644 --- a/gst/rtp/gstrtpqcelpdepay.c +++ b/gst/rtp/gstrtpqcelpdepay.c @@ -91,7 +91,7 @@ gst_rtp_qcelp_depay_base_init (gpointer klass) gst_static_pad_template_get (&gst_rtp_qcelp_depay_sink_template)); gst_element_class_set_details_simple (element_class, "RTP QCELP depayloader", - "Codec/Depayloader/Network", + "Codec/Depayloader/Network/RTP", "Extracts QCELP (PureVoice) audio from RTP packets (RFC 2658)", "Wim Taymans "); } diff --git a/gst/rtp/gstrtpqdmdepay.c b/gst/rtp/gstrtpqdmdepay.c index 001b93a4f1..7e2e50b071 100644 --- a/gst/rtp/gstrtpqdmdepay.c +++ b/gst/rtp/gstrtpqdmdepay.c @@ -77,7 +77,7 @@ gst_rtp_qdm2_depay_base_init (gpointer klass) gst_element_class_set_details_simple (element_class, "RTP QDM2 depayloader", - "Codec/Depayloader/Network", + "Codec/Depayloader/Network/RTP", "Extracts QDM2 audio from RTP packets (no RFC)", "Edward Hervey "); } diff --git a/gst/rtp/gstrtpsirendepay.c b/gst/rtp/gstrtpsirendepay.c index 3c46ccb7cb..2746e74f82 100644 --- a/gst/rtp/gstrtpsirendepay.c +++ b/gst/rtp/gstrtpsirendepay.c @@ -64,7 +64,7 @@ gst_rtp_siren_depay_base_init (gpointer klass) gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_siren_depay_sink_template)); gst_element_class_set_details_simple (element_class, - "RTP Siren packet depayloader", "Codec/Depayloader/Network", + "RTP Siren packet depayloader", "Codec/Depayloader/Network/RTP", "Extracts Siren audio from RTP packets", "Philippe Kalaf "); } diff --git a/gst/rtp/gstrtpsirenpay.c b/gst/rtp/gstrtpsirenpay.c index 0ddf0af5e8..c7c9582af0 100644 --- a/gst/rtp/gstrtpsirenpay.c +++ b/gst/rtp/gstrtpsirenpay.c @@ -64,7 +64,7 @@ gst_rtp_siren_pay_base_init (gpointer klass) gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_siren_pay_src_template)); gst_element_class_set_details_simple (element_class, - "RTP Payloader for Siren Audio", "Codec/Payloader/Network", + "RTP Payloader for Siren Audio", "Codec/Payloader/Network/RTP", "Packetize Siren audio streams into RTP packets", "Youness Alaoui "); } diff --git a/gst/rtp/gstrtpspeexdepay.c b/gst/rtp/gstrtpspeexdepay.c index 4bb10aaf26..10dea1a703 100644 --- a/gst/rtp/gstrtpspeexdepay.c +++ b/gst/rtp/gstrtpspeexdepay.c @@ -76,7 +76,7 @@ gst_rtp_speex_depay_base_init (gpointer klass) gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_speex_depay_sink_template)); gst_element_class_set_details_simple (element_class, "RTP Speex depayloader", - "Codec/Depayloader/Network", + "Codec/Depayloader/Network/RTP", "Extracts Speex audio from RTP packets", "Edgard Lima "); } diff --git a/gst/rtp/gstrtpspeexpay.c b/gst/rtp/gstrtpspeexpay.c index 424070a9e0..a551fa9439 100644 --- a/gst/rtp/gstrtpspeexpay.c +++ b/gst/rtp/gstrtpspeexpay.c @@ -73,7 +73,7 @@ gst_rtp_speex_pay_base_init (gpointer klass) gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_speex_pay_src_template)); gst_element_class_set_details_simple (element_class, "RTP Speex payloader", - "Codec/Payloader/Network", + "Codec/Payloader/Network/RTP", "Payload-encodes Speex audio into a RTP packet", "Edgard Lima "); diff --git a/gst/rtp/gstrtpsv3vdepay.c b/gst/rtp/gstrtpsv3vdepay.c index f67f9853fa..a99559cfe1 100644 --- a/gst/rtp/gstrtpsv3vdepay.c +++ b/gst/rtp/gstrtpsv3vdepay.c @@ -72,7 +72,7 @@ gst_rtp_sv3v_depay_base_init (gpointer klass) gst_element_class_set_details_simple (element_class, "RTP SVQ3 depayloader", - "Codec/Depayloader/Network", + "Codec/Depayloader/Network/RTP", "Extracts SVQ3 video from RTP packets (no RFC)", "Wim Taymans "); } diff --git a/gst/rtp/gstrtptheoradepay.c b/gst/rtp/gstrtptheoradepay.c index fa6ef62450..dfe6de3d45 100644 --- a/gst/rtp/gstrtptheoradepay.c +++ b/gst/rtp/gstrtptheoradepay.c @@ -82,7 +82,7 @@ gst_rtp_theora_depay_base_init (gpointer klass) gst_static_pad_template_get (&gst_rtp_theora_depay_src_template)); gst_element_class_set_details_simple (element_class, "RTP Theora depayloader", - "Codec/Depayloader/Network", + "Codec/Depayloader/Network/RTP", "Extracts Theora video from RTP packets (draft-01 of RFC XXXX)", "Wim Taymans "); } diff --git a/gst/rtp/gstrtptheorapay.c b/gst/rtp/gstrtptheorapay.c index 7fd1182848..7782853508 100644 --- a/gst/rtp/gstrtptheorapay.c +++ b/gst/rtp/gstrtptheorapay.c @@ -104,7 +104,7 @@ gst_rtp_theora_pay_base_init (gpointer klass) gst_static_pad_template_get (&gst_rtp_theora_pay_sink_template)); gst_element_class_set_details_simple (element_class, "RTP Theora payloader", - "Codec/Payloader/Network", + "Codec/Payloader/Network/RTP", "Payload-encode Theora video into RTP packets (draft-01 RFC XXXX)", "Wim Taymans "); } diff --git a/gst/rtp/gstrtpvorbisdepay.c b/gst/rtp/gstrtpvorbisdepay.c index 5cf5ef8ebc..506558defe 100644 --- a/gst/rtp/gstrtpvorbisdepay.c +++ b/gst/rtp/gstrtpvorbisdepay.c @@ -82,7 +82,7 @@ gst_rtp_vorbis_depay_base_init (gpointer klass) gst_static_pad_template_get (&gst_rtp_vorbis_depay_src_template)); gst_element_class_set_details_simple (element_class, "RTP Vorbis depayloader", - "Codec/Depayloader/Network", + "Codec/Depayloader/Network/RTP", "Extracts Vorbis Audio from RTP packets (RFC 5215)", "Wim Taymans "); } diff --git a/gst/rtp/gstrtpvorbispay.c b/gst/rtp/gstrtpvorbispay.c index 76f3fe84da..a7841c382a 100644 --- a/gst/rtp/gstrtpvorbispay.c +++ b/gst/rtp/gstrtpvorbispay.c @@ -81,7 +81,7 @@ gst_rtp_vorbis_pay_base_init (gpointer klass) gst_static_pad_template_get (&gst_rtp_vorbis_pay_sink_template)); gst_element_class_set_details_simple (element_class, "RTP Vorbis depayloader", - "Codec/Payloader/Network", + "Codec/Payloader/Network/RTP", "Payload-encode Vorbis audio into RTP packets (RFC 5215)", "Wim Taymans "); } diff --git a/gst/rtp/gstrtpvrawdepay.c b/gst/rtp/gstrtpvrawdepay.c index fae1692401..8e1af75e72 100644 --- a/gst/rtp/gstrtpvrawdepay.c +++ b/gst/rtp/gstrtpvrawdepay.c @@ -72,7 +72,7 @@ gst_rtp_vraw_depay_base_init (gpointer klass) gst_static_pad_template_get (&gst_rtp_vraw_depay_sink_template)); gst_element_class_set_details_simple (element_class, - "RTP Raw Video depayloader", "Codec/Depayloader/Network", + "RTP Raw Video depayloader", "Codec/Depayloader/Network/RTP", "Extracts raw video from RTP packets (RFC 4175)", "Wim Taymans "); } diff --git a/gst/rtp/gstrtpvrawpay.c b/gst/rtp/gstrtpvrawpay.c index 911aed34b5..12b72e3eb5 100644 --- a/gst/rtp/gstrtpvrawpay.c +++ b/gst/rtp/gstrtpvrawpay.c @@ -122,7 +122,7 @@ GST_BOILERPLATE (GstRtpVRawPay, gst_rtp_vraw_pay, GstBaseRTPPayload, gst_static_pad_template_get (&gst_rtp_vraw_pay_sink_template)); gst_element_class_set_details_simple (element_class, - "RTP Raw Video payloader", "Codec/Payloader/Network", + "RTP Raw Video payloader", "Codec/Payloader/Network/RTP", "Payload raw video as RTP packets (RFC 4175)", "Wim Taymans "); } From dcbba0932dc579abd6aab4460fa1a416374eda1b Mon Sep 17 00:00:00 2001 From: Thiago Santos Date: Tue, 21 Dec 2010 13:37:40 -0300 Subject: [PATCH 054/351] jpegenc: sink pad's getcaps shouldn't use the src pad getcaps Instead of using get_allowed_caps on the srcpad, the sinkpad getcaps should use the getcaps of the srcpad's peer. This way the srcpad can keep using fixed_caps and sinkpad getcaps exposes all caps that can be negotiated https://bugzilla.gnome.org/show_bug.cgi?id=637686 --- ext/jpeg/gstjpegenc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/ext/jpeg/gstjpegenc.c b/ext/jpeg/gstjpegenc.c index e562867216..60e2e99bb3 100644 --- a/ext/jpeg/gstjpegenc.c +++ b/ext/jpeg/gstjpegenc.c @@ -345,7 +345,7 @@ gst_jpegenc_getcaps (GstPad * pad) /* we want to proxy properties like width, height and framerate from the other end of the element */ - othercaps = gst_pad_get_allowed_caps (jpegenc->srcpad); + othercaps = gst_pad_peer_get_caps_reffed (jpegenc->srcpad); if (othercaps == NULL || gst_caps_is_empty (othercaps) || gst_caps_is_any (othercaps)) { caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad)); From dde7af4b9d55463201d649427ad52b4642b6f945 Mon Sep 17 00:00:00 2001 From: Thiago Santos Date: Mon, 20 Dec 2010 14:49:02 -0300 Subject: [PATCH 055/351] jpegenc: Adds another test case Adds a test for jpegenc to check that is possible to negotiate and push buffers with different resolution one after another. https://bugzilla.gnome.org/show_bug.cgi?id=637686 --- tests/check/Makefile.am | 3 ++ tests/check/elements/jpegenc.c | 88 ++++++++++++++++++++++++++++++++++ 2 files changed, 91 insertions(+) diff --git a/tests/check/Makefile.am b/tests/check/Makefile.am index 059eedaefe..5079bc7274 100644 --- a/tests/check/Makefile.am +++ b/tests/check/Makefile.am @@ -202,6 +202,9 @@ elements_interleave_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR elements_imagefreeze_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(AM_CFLAGS) elements_imagefreeze_LDADD = $(GST_BASE_LIBS) $(LDADD) -lgstvideo-$(GST_MAJORMINOR) +elements_jpegenc_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(AM_CFLAGS) +elements_jpegenc_LDADD = $(GST_BASE_LIBS) $(LDADD) -lgstapp-0.10 + elements_level_LDADD = $(LDADD) $(LIBM) elements_matroskamux_LDADD = $(GST_BASE_LIBS) $(LDADD) $(LIBM) diff --git a/tests/check/elements/jpegenc.c b/tests/check/elements/jpegenc.c index 674617d6bc..2c992f77b6 100644 --- a/tests/check/elements/jpegenc.c +++ b/tests/check/elements/jpegenc.c @@ -23,11 +23,13 @@ #include #include +#include /* For ease of programming we use globals to keep refs for our floating * sink pads we create; otherwise we always have to do get_pad, * get_peer, and then remove references in every test function */ static GstPad *mysinkpad; +static GstPad *mysrcpad; #define JPEG_CAPS_STRING "image/jpeg" @@ -52,6 +54,12 @@ GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_ALWAYS, GST_STATIC_CAPS (JPEG_CAPS_RESTRICTIVE)); +static GstStaticPadTemplate any_srctemplate = GST_STATIC_PAD_TEMPLATE ("src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + GST_STATIC_CAPS_ANY); + + static GstElement * setup_jpegenc (GstStaticPadTemplate * sinktemplate) { @@ -60,6 +68,8 @@ setup_jpegenc (GstStaticPadTemplate * sinktemplate) GST_DEBUG ("setup_jpegenc"); jpegenc = gst_check_setup_element ("jpegenc"); mysinkpad = gst_check_setup_sink_pad (jpegenc, sinktemplate, NULL); + mysrcpad = gst_check_setup_src_pad (jpegenc, &any_srctemplate, NULL); + gst_pad_set_active (mysrcpad, TRUE); gst_pad_set_active (mysinkpad, TRUE); return jpegenc; @@ -71,11 +81,44 @@ cleanup_jpegenc (GstElement * jpegenc) GST_DEBUG ("cleanup_jpegenc"); gst_element_set_state (jpegenc, GST_STATE_NULL); + gst_pad_set_active (mysrcpad, FALSE); gst_pad_set_active (mysinkpad, FALSE); gst_check_teardown_sink_pad (jpegenc); + gst_check_teardown_src_pad (jpegenc); gst_check_teardown_element (jpegenc); } +static GstBuffer * +create_video_buffer (GstCaps * caps) +{ + GstElement *pipeline; + GstElement *cf; + GstElement *sink; + GstBuffer *buffer; + + pipeline = + gst_parse_launch + ("videotestsrc num-buffers=1 ! capsfilter name=cf ! appsink name=sink", + NULL); + g_assert (pipeline != NULL); + + cf = gst_bin_get_by_name (GST_BIN (pipeline), "cf"); + sink = gst_bin_get_by_name (GST_BIN (pipeline), "sink"); + + g_object_set (G_OBJECT (cf), "caps", caps, NULL); + + gst_element_set_state (pipeline, GST_STATE_PLAYING); + + buffer = gst_app_sink_pull_buffer (GST_APP_SINK (sink)); + + gst_element_set_state (pipeline, GST_STATE_NULL); + gst_object_unref (pipeline); + gst_object_unref (sink); + gst_object_unref (cf); + return buffer; +} + + GST_START_TEST (test_jpegenc_getcaps) { GstElement *jpegenc; @@ -129,6 +172,50 @@ GST_START_TEST (test_jpegenc_getcaps) GST_END_TEST; + +GST_START_TEST (test_jpegenc_different_caps) +{ + GstElement *jpegenc; + GstBuffer *buffer; + GstCaps *caps; + GstCaps *allowed_caps; + + /* now use a more restricted one and check the resulting caps */ + jpegenc = setup_jpegenc (&any_sinktemplate); + gst_element_set_state (jpegenc, GST_STATE_PLAYING); + + /* push first buffer with 800x600 resolution */ + caps = gst_caps_new_simple ("video/x-raw-yuv", "width", G_TYPE_INT, + 800, "height", G_TYPE_INT, 600, "framerate", + GST_TYPE_FRACTION, 1, 1, "format", GST_TYPE_FOURCC, + GST_MAKE_FOURCC ('I', '4', '2', '0'), NULL); + buffer = create_video_buffer (caps); + gst_caps_unref (caps); + fail_unless (gst_pad_push (mysrcpad, buffer) == GST_FLOW_OK); + + /* check the allowed caps to see if a second buffer with a different + * caps could be negotiated */ + allowed_caps = gst_pad_get_allowed_caps (mysrcpad); + + /* the caps we want to negotiate to */ + caps = gst_caps_new_simple ("video/x-raw-yuv", "width", G_TYPE_INT, + 640, "height", G_TYPE_INT, 480, "framerate", + GST_TYPE_FRACTION, 1, 1, "format", GST_TYPE_FOURCC, + GST_MAKE_FOURCC ('I', '4', '2', '0'), NULL); + fail_unless (gst_caps_can_intersect (allowed_caps, caps)); + + /* push second buffer with 640x480 resolution */ + buffer = create_video_buffer (caps); + gst_caps_unref (caps); + fail_unless (gst_pad_push (mysrcpad, buffer) == GST_FLOW_OK); + + gst_caps_unref (allowed_caps); + gst_element_set_state (jpegenc, GST_STATE_NULL); + cleanup_jpegenc (jpegenc); +} + +GST_END_TEST; + static Suite * jpegenc_suite (void) { @@ -137,6 +224,7 @@ jpegenc_suite (void) suite_add_tcase (s, tc_chain); tcase_add_test (tc_chain, test_jpegenc_getcaps); + tcase_add_test (tc_chain, test_jpegenc_different_caps); return s; } From e2f4fe8d3df0a32684486a091b6ecbbb77061bcb Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Tue, 21 Dec 2010 22:16:42 +0100 Subject: [PATCH 056/351] mpapay: add debug category --- gst/rtp/gstrtpmpapay.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/gst/rtp/gstrtpmpapay.c b/gst/rtp/gstrtpmpapay.c index 9f17da11f9..f685ef271d 100644 --- a/gst/rtp/gstrtpmpapay.c +++ b/gst/rtp/gstrtpmpapay.c @@ -27,6 +27,9 @@ #include "gstrtpmpapay.h" +GST_DEBUG_CATEGORY_STATIC (rtpmpapay_debug); +#define GST_CAT_DEFAULT (rtpmpapay_debug) + static GstStaticPadTemplate gst_rtp_mpa_pay_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, @@ -96,6 +99,9 @@ gst_rtp_mpa_pay_class_init (GstRtpMPAPayClass * klass) gstbasertppayload_class->set_caps = gst_rtp_mpa_pay_setcaps; gstbasertppayload_class->handle_event = gst_rtp_mpa_pay_handle_event; gstbasertppayload_class->handle_buffer = gst_rtp_mpa_pay_handle_buffer; + + GST_DEBUG_CATEGORY_INIT (rtpmpapay_debug, "rtpmpapay", 0, + "MPEG Audio RTP Depayloader"); } static void From 97993d31194b3f817509a91f835febbbca923b9c Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Tue, 21 Dec 2010 22:17:19 +0100 Subject: [PATCH 057/351] ac3depay: fix debug category description --- gst/rtp/gstrtpac3depay.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/gst/rtp/gstrtpac3depay.c b/gst/rtp/gstrtpac3depay.c index 80f91b3341..2ba4cefe77 100644 --- a/gst/rtp/gstrtpac3depay.c +++ b/gst/rtp/gstrtpac3depay.c @@ -82,7 +82,7 @@ gst_rtp_ac3_depay_class_init (GstRtpAC3DepayClass * klass) gstbasertpdepayload_class->process = gst_rtp_ac3_depay_process; GST_DEBUG_CATEGORY_INIT (rtpac3depay_debug, "rtpac3depay", 0, - "MPEG Audio RTP Depayloader"); + "AC3 Audio RTP Depayloader"); } static void From 6db12cb0036c7aeb49ec71f26a3ce9a6b11d48d9 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Tue, 21 Dec 2010 22:34:49 +0100 Subject: [PATCH 058/351] rtpac3pay: add AC3 payloader --- gst/rtp/Makefile.am | 2 + gst/rtp/gstrtp.c | 4 + gst/rtp/gstrtpac3pay.c | 452 +++++++++++++++++++++++++++++++++++++++++ gst/rtp/gstrtpac3pay.h | 64 ++++++ 4 files changed, 522 insertions(+) create mode 100644 gst/rtp/gstrtpac3pay.c create mode 100644 gst/rtp/gstrtpac3pay.h diff --git a/gst/rtp/Makefile.am b/gst/rtp/Makefile.am index 5a300b33c6..9d431084d3 100644 --- a/gst/rtp/Makefile.am +++ b/gst/rtp/Makefile.am @@ -6,6 +6,7 @@ libgstrtp_la_SOURCES = \ gstrtpchannels.c \ gstrtpdepay.c \ gstrtpac3depay.c \ + gstrtpac3pay.c \ gstrtpbvdepay.c \ gstrtpbvpay.c \ gstrtpceltdepay.c \ @@ -94,6 +95,7 @@ noinst_HEADERS = \ gstrtpL16depay.h \ gstrtpL16pay.h \ gstrtpac3depay.h \ + gstrtpac3pay.h \ gstrtpbvdepay.h \ gstrtpbvpay.h \ gstrtpceltpay.h \ diff --git a/gst/rtp/gstrtp.c b/gst/rtp/gstrtp.c index 339ff97470..e2fa2ed413 100644 --- a/gst/rtp/gstrtp.c +++ b/gst/rtp/gstrtp.c @@ -23,6 +23,7 @@ #include "gstrtpdepay.h" #include "gstrtpac3depay.h" +#include "gstrtpac3pay.h" #include "gstrtpbvdepay.h" #include "gstrtpbvpay.h" #include "gstrtpceltdepay.h" @@ -97,6 +98,9 @@ plugin_init (GstPlugin * plugin) if (!gst_rtp_ac3_depay_plugin_init (plugin)) return FALSE; + if (!gst_rtp_ac3_pay_plugin_init (plugin)) + return FALSE; + if (!gst_rtp_bv_depay_plugin_init (plugin)) return FALSE; diff --git a/gst/rtp/gstrtpac3pay.c b/gst/rtp/gstrtpac3pay.c new file mode 100644 index 0000000000..fddea38472 --- /dev/null +++ b/gst/rtp/gstrtpac3pay.c @@ -0,0 +1,452 @@ +/* GStreamer + * Copyright (C) <2010> Wim Taymans + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifdef HAVE_CONFIG_H +# include "config.h" +#endif + +#include + +#include + +#include "gstrtpac3pay.h" + +GST_DEBUG_CATEGORY_STATIC (rtpac3pay_debug); +#define GST_CAT_DEFAULT (rtpac3pay_debug) + +static GstStaticPadTemplate gst_rtp_ac3_pay_sink_template = + GST_STATIC_PAD_TEMPLATE ("sink", + GST_PAD_SINK, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/ac3; " "audio/x-ac3; ") + ); + +static GstStaticPadTemplate gst_rtp_ac3_pay_src_template = +GST_STATIC_PAD_TEMPLATE ("src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("application/x-rtp, " + "media = (string) \"audio\", " + "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " + "clock-rate = (int) { 32000, 44100, 48000 }, " + "encoding-name = (string) \"AC3\"") + ); + +static void gst_rtp_ac3_pay_finalize (GObject * object); + +static GstStateChangeReturn gst_rtp_ac3_pay_change_state (GstElement * element, + GstStateChange transition); + +static gboolean gst_rtp_ac3_pay_setcaps (GstBaseRTPPayload * payload, + GstCaps * caps); +static gboolean gst_rtp_ac3_pay_handle_event (GstPad * pad, GstEvent * event); +static GstFlowReturn gst_rtp_ac3_pay_flush (GstRtpAC3Pay * rtpac3pay); +static GstFlowReturn gst_rtp_ac3_pay_handle_buffer (GstBaseRTPPayload * payload, + GstBuffer * buffer); + +GST_BOILERPLATE (GstRtpAC3Pay, gst_rtp_ac3_pay, GstBaseRTPPayload, + GST_TYPE_BASE_RTP_PAYLOAD) + + static void gst_rtp_ac3_pay_base_init (gpointer klass) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (klass); + + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&gst_rtp_ac3_pay_src_template)); + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&gst_rtp_ac3_pay_sink_template)); + + gst_element_class_set_details_simple (element_class, + "RTP AC3 audio payloader", "Codec/Payloader/Network/RTP", + "Payload AC3 audio as RTP packets (RFC 4184)", + "Wim Taymans "); +} + +static void +gst_rtp_ac3_pay_class_init (GstRtpAC3PayClass * klass) +{ + GObjectClass *gobject_class; + GstElementClass *gstelement_class; + GstBaseRTPPayloadClass *gstbasertppayload_class; + + gobject_class = (GObjectClass *) klass; + gstelement_class = (GstElementClass *) klass; + gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass; + + gobject_class->finalize = gst_rtp_ac3_pay_finalize; + + gstelement_class->change_state = gst_rtp_ac3_pay_change_state; + + gstbasertppayload_class->set_caps = gst_rtp_ac3_pay_setcaps; + gstbasertppayload_class->handle_event = gst_rtp_ac3_pay_handle_event; + gstbasertppayload_class->handle_buffer = gst_rtp_ac3_pay_handle_buffer; + + GST_DEBUG_CATEGORY_INIT (rtpac3pay_debug, "rtpac3pay", 0, + "AC3 Audio RTP Depayloader"); +} + +static void +gst_rtp_ac3_pay_init (GstRtpAC3Pay * rtpac3pay, GstRtpAC3PayClass * klass) +{ + rtpac3pay->adapter = gst_adapter_new (); +} + +static void +gst_rtp_ac3_pay_finalize (GObject * object) +{ + GstRtpAC3Pay *rtpac3pay; + + rtpac3pay = GST_RTP_AC3_PAY (object); + + g_object_unref (rtpac3pay->adapter); + + G_OBJECT_CLASS (parent_class)->finalize (object); +} + +static void +gst_rtp_ac3_pay_reset (GstRtpAC3Pay * pay) +{ + pay->first_ts = -1; + pay->duration = 0; + gst_adapter_clear (pay->adapter); + GST_DEBUG_OBJECT (pay, "reset depayloader"); +} + +static gboolean +gst_rtp_ac3_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps) +{ + gboolean res; + gint rate; + GstStructure *structure; + + structure = gst_caps_get_structure (caps, 0); + + if (!gst_structure_get_int (structure, "rate", &rate)) + rate = 90000; /* default */ + + gst_basertppayload_set_options (payload, "audio", TRUE, "AC3", rate); + res = gst_basertppayload_set_outcaps (payload, NULL); + + return res; +} + +static gboolean +gst_rtp_ac3_pay_handle_event (GstPad * pad, GstEvent * event) +{ + GstRtpAC3Pay *rtpac3pay; + + rtpac3pay = GST_RTP_AC3_PAY (gst_pad_get_parent (pad)); + + switch (GST_EVENT_TYPE (event)) { + case GST_EVENT_EOS: + /* make sure we push the last packets in the adapter on EOS */ + gst_rtp_ac3_pay_flush (rtpac3pay); + break; + case GST_EVENT_FLUSH_STOP: + gst_rtp_ac3_pay_reset (rtpac3pay); + break; + default: + break; + } + + gst_object_unref (rtpac3pay); + + /* FALSE to let the parent handle the event as well */ + return FALSE; +} + +struct frmsize_s +{ + guint16 bit_rate; + guint16 frm_size[3]; +}; + +static const struct frmsize_s frmsizecod_tbl[] = { + {32, {64, 69, 96}}, + {32, {64, 70, 96}}, + {40, {80, 87, 120}}, + {40, {80, 88, 120}}, + {48, {96, 104, 144}}, + {48, {96, 105, 144}}, + {56, {112, 121, 168}}, + {56, {112, 122, 168}}, + {64, {128, 139, 192}}, + {64, {128, 140, 192}}, + {80, {160, 174, 240}}, + {80, {160, 175, 240}}, + {96, {192, 208, 288}}, + {96, {192, 209, 288}}, + {112, {224, 243, 336}}, + {112, {224, 244, 336}}, + {128, {256, 278, 384}}, + {128, {256, 279, 384}}, + {160, {320, 348, 480}}, + {160, {320, 349, 480}}, + {192, {384, 417, 576}}, + {192, {384, 418, 576}}, + {224, {448, 487, 672}}, + {224, {448, 488, 672}}, + {256, {512, 557, 768}}, + {256, {512, 558, 768}}, + {320, {640, 696, 960}}, + {320, {640, 697, 960}}, + {384, {768, 835, 1152}}, + {384, {768, 836, 1152}}, + {448, {896, 975, 1344}}, + {448, {896, 976, 1344}}, + {512, {1024, 1114, 1536}}, + {512, {1024, 1115, 1536}}, + {576, {1152, 1253, 1728}}, + {576, {1152, 1254, 1728}}, + {640, {1280, 1393, 1920}}, + {640, {1280, 1394, 1920}} +}; + +static GstFlowReturn +gst_rtp_ac3_pay_flush (GstRtpAC3Pay * rtpac3pay) +{ + guint avail, FT, NF, mtu; + GstBuffer *outbuf; + GstFlowReturn ret; + + /* the data available in the adapter is either smaller + * than the MTU or bigger. In the case it is smaller, the complete + * adapter contents can be put in one packet. In the case the + * adapter has more than one MTU, we need to split the AC3 data + * over multiple packets. */ + avail = gst_adapter_available (rtpac3pay->adapter); + + ret = GST_FLOW_OK; + + FT = 0; + /* number of frames */ + NF = rtpac3pay->NF; + + mtu = GST_BASE_RTP_PAYLOAD_MTU (rtpac3pay); + + GST_LOG_OBJECT (rtpac3pay, "flushing %u bytes", avail); + + while (avail > 0) { + guint towrite; + guint8 *payload; + guint payload_len; + guint packet_len; + + /* this will be the total length of the packet */ + packet_len = gst_rtp_buffer_calc_packet_len (2 + avail, 0, 0); + + /* fill one MTU or all available bytes */ + towrite = MIN (packet_len, mtu); + + /* this is the payload length */ + payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0); + + /* create buffer to hold the payload */ + outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0); + + if (FT == 0) { + /* check if it all fits */ + if (towrite < packet_len) { + guint maxlen; + + GST_LOG_OBJECT (rtpac3pay, "we need to fragment"); + /* check if we will be able to put at least 5/8th of the total + * frame in this first frame. */ + if ((avail * 5) / 8 >= (payload_len - 2)) + FT = 1; + else + FT = 2; + /* check how many fragments we will need */ + maxlen = gst_rtp_buffer_calc_payload_len (mtu - 2, 0, 0); + NF = (avail + maxlen - 1) / maxlen; + } + } else if (FT != 3) { + /* remaining fragment */ + FT = 3; + } + + /* + * 0 1 + * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 + * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + * | MBZ | FT| NF | + * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + * + * FT: 0: one or more complete frames + * 1: initial 5/8 fragment + * 2: initial fragment not 5/8 + * 3: other fragment + * NF: amount of frames if FT = 0, else number of fragments. + */ + GST_LOG_OBJECT (rtpac3pay, "FT %u, NF %u", FT, NF); + payload = gst_rtp_buffer_get_payload (outbuf); + payload[0] = (FT & 3); + payload[1] = NF; + payload_len -= 2; + + gst_adapter_copy (rtpac3pay->adapter, &payload[2], 0, payload_len); + gst_adapter_flush (rtpac3pay->adapter, payload_len); + + avail -= payload_len; + if (avail == 0) + gst_rtp_buffer_set_marker (outbuf, TRUE); + + GST_BUFFER_TIMESTAMP (outbuf) = rtpac3pay->first_ts; + GST_BUFFER_DURATION (outbuf) = rtpac3pay->duration; + + ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpac3pay), outbuf); + } + + return ret; +} + +static GstFlowReturn +gst_rtp_ac3_pay_handle_buffer (GstBaseRTPPayload * basepayload, + GstBuffer * buffer) +{ + GstRtpAC3Pay *rtpac3pay; + GstFlowReturn ret; + guint size, avail, left, NF; + guint8 *data, *p; + guint packet_len; + GstClockTime duration, timestamp; + + rtpac3pay = GST_RTP_AC3_PAY (basepayload); + + size = GST_BUFFER_SIZE (buffer); + data = GST_BUFFER_DATA (buffer); + duration = GST_BUFFER_DURATION (buffer); + timestamp = GST_BUFFER_TIMESTAMP (buffer); + + if (GST_BUFFER_IS_DISCONT (buffer)) { + GST_DEBUG_OBJECT (rtpac3pay, "DISCONT"); + gst_rtp_ac3_pay_reset (rtpac3pay); + } + + /* count the amount of incomming packets */ + NF = 0; + left = size; + p = data; + while (TRUE) { + guint bsid, fscod, frmsizecod, frame_size; + + if (left < 6) + break; + + if (p[0] != 0x0b || p[1] != 0x77) + break; + + bsid = p[5] >> 3; + if (bsid > 8) + break; + + frmsizecod = p[4] & 0x3f; + fscod = p[4] >> 6; + + GST_DEBUG_OBJECT (rtpac3pay, "fscod %u, %u", fscod, frmsizecod); + + if (fscod >= 3 || frmsizecod >= 38) + break; + + frame_size = frmsizecod_tbl[frmsizecod].frm_size[fscod] * 2; + if (frame_size > left) + break; + + NF++; + GST_DEBUG_OBJECT (rtpac3pay, "found frame %u of size %u", NF, frame_size); + + p += frame_size; + left -= frame_size; + } + if (NF == 0) + goto no_frames; + + avail = gst_adapter_available (rtpac3pay->adapter); + + /* get packet length of previous data and this new data, + * payload length includes a 4 byte header */ + packet_len = gst_rtp_buffer_calc_packet_len (2 + avail + size, 0, 0); + + /* if this buffer is going to overflow the packet, flush what we + * have. */ + if (gst_basertppayload_is_filled (basepayload, + packet_len, rtpac3pay->duration + duration)) { + ret = gst_rtp_ac3_pay_flush (rtpac3pay); + avail = 0; + } else { + ret = GST_FLOW_OK; + } + + if (avail == 0) { + GST_DEBUG_OBJECT (rtpac3pay, + "first packet, save timestamp %" GST_TIME_FORMAT, + GST_TIME_ARGS (timestamp)); + rtpac3pay->first_ts = timestamp; + rtpac3pay->duration = 0; + rtpac3pay->NF = 0; + } + + gst_adapter_push (rtpac3pay->adapter, buffer); + rtpac3pay->duration += duration; + rtpac3pay->NF += NF; + + return ret; + + /* ERRORS */ +no_frames: + { + GST_WARNING_OBJECT (rtpac3pay, "no valid AC3 frames found"); + return GST_FLOW_OK; + } +} + +static GstStateChangeReturn +gst_rtp_ac3_pay_change_state (GstElement * element, GstStateChange transition) +{ + GstRtpAC3Pay *rtpac3pay; + GstStateChangeReturn ret; + + rtpac3pay = GST_RTP_AC3_PAY (element); + + switch (transition) { + case GST_STATE_CHANGE_READY_TO_PAUSED: + gst_rtp_ac3_pay_reset (rtpac3pay); + break; + default: + break; + } + + ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); + + switch (transition) { + case GST_STATE_CHANGE_PAUSED_TO_READY: + gst_rtp_ac3_pay_reset (rtpac3pay); + break; + default: + break; + } + return ret; +} + +gboolean +gst_rtp_ac3_pay_plugin_init (GstPlugin * plugin) +{ + return gst_element_register (plugin, "rtpac3pay", + GST_RANK_SECONDARY, GST_TYPE_RTP_AC3_PAY); +} diff --git a/gst/rtp/gstrtpac3pay.h b/gst/rtp/gstrtpac3pay.h new file mode 100644 index 0000000000..2ca9d997d2 --- /dev/null +++ b/gst/rtp/gstrtpac3pay.h @@ -0,0 +1,64 @@ +/* GStreamer + * Copyright (C) <2010> Wim Taymans + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifndef __GST_RTP_AC3_PAY_H__ +#define __GST_RTP_AC3_PAY_H__ + +#include +#include +#include + +G_BEGIN_DECLS + +#define GST_TYPE_RTP_AC3_PAY \ + (gst_rtp_ac3_pay_get_type()) +#define GST_RTP_AC3_PAY(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_AC3_PAY,GstRtpAC3Pay)) +#define GST_RTP_AC3_PAY_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_AC3_PAY,GstRtpAC3PayClass)) +#define GST_IS_RTP_AC3_PAY(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_AC3_PAY)) +#define GST_IS_RTP_AC3_PAY_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_AC3_PAY)) + +typedef struct _GstRtpAC3Pay GstRtpAC3Pay; +typedef struct _GstRtpAC3PayClass GstRtpAC3PayClass; + +struct _GstRtpAC3Pay +{ + GstBaseRTPPayload payload; + + GstAdapter *adapter; + GstClockTime first_ts; + GstClockTime duration; + guint NF; +}; + +struct _GstRtpAC3PayClass +{ + GstBaseRTPPayloadClass parent_class; +}; + +GType gst_rtp_ac3_pay_get_type (void); + +gboolean gst_rtp_ac3_pay_plugin_init (GstPlugin * plugin); + +G_END_DECLS + +#endif /* __GST_RTP_AC3_PAY_H__ */ From 2b53cbe923419a87df9cccd8fbcb911e61b46c9e Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Wed, 22 Dec 2010 11:42:31 +0100 Subject: [PATCH 059/351] rtpsession: unlock before emitting signals --- gst/rtpmanager/gstrtpsession.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c index fd2663652e..751b7f8cce 100644 --- a/gst/rtpmanager/gstrtpsession.c +++ b/gst/rtpmanager/gstrtpsession.c @@ -1215,9 +1215,13 @@ gst_rtp_session_get_caps_for_pt (GstRtpSession * rtpsession, guint payload) g_value_init (&ret, GST_TYPE_CAPS); g_value_set_boxed (&ret, NULL); + GST_RTP_SESSION_UNLOCK (rtpsession); + g_signal_emitv (args, gst_rtp_session_signals[SIGNAL_REQUEST_PT_MAP], 0, &ret); + GST_RTP_SESSION_LOCK (rtpsession); + g_value_unset (&args[0]); g_value_unset (&args[1]); caps = (GstCaps *) g_value_dup_boxed (&ret); From 7759ad0db2a8b8d2517c9178a9f456b6a392f062 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Thu, 16 Dec 2010 11:44:44 +0000 Subject: [PATCH 060/351] docs: update rtspsrc docs, rtpbin is not in -bad any more --- gst/rtsp/gstrtspsrc.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c index ced8ae5759..a21a14ad2a 100644 --- a/gst/rtsp/gstrtspsrc.c +++ b/gst/rtsp/gstrtspsrc.c @@ -61,8 +61,7 @@ * rtspsrc will internally instantiate an RTP session manager element * that will handle the RTCP messages to and from the server, jitter removal, * packet reordering along with providing a clock for the pipeline. - * This feature is currently fully implemented with the gstrtpbin in the - * gst-plugins-bad module. + * This feature is implemented using the gstrtpbin element. * * rtspsrc acts like a live source and will therefore only generate data in the * PLAYING state. From 331bcfbdfc1a00149fe5e989cdb739387f1a2030 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Wed, 22 Dec 2010 14:14:08 +0000 Subject: [PATCH 061/351] matroskamux: fix subtitle pad template, we only handle kate for now --- gst/matroska/matroska-mux.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/gst/matroska/matroska-mux.c b/gst/matroska/matroska-mux.c index fea9c356fe..7da4e6fa92 100644 --- a/gst/matroska/matroska-mux.c +++ b/gst/matroska/matroska-mux.c @@ -203,7 +203,7 @@ static GstStaticPadTemplate subtitlesink_templ = GST_STATIC_PAD_TEMPLATE ("subtitle_%d", GST_PAD_SINK, GST_PAD_REQUEST, - GST_STATIC_CAPS_ANY); + GST_STATIC_CAPS ("subtitle/x-kate")); static GArray *used_uids; G_LOCK_DEFINE_STATIC (used_uids); From 1d74f568a3ae8ba24fd6909b838e8a9fb7d47a90 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Sebastian=20Dr=C3=B6ge?= Date: Wed, 22 Dec 2010 19:58:21 +0100 Subject: [PATCH 062/351] matroskademux: Don't use gst_pad_alloc_buffer() Using this in a demuxer will cause deadlocks if there's a pad with a pending pad-block downstream, no matter if there is a queue between the pad or not. Queues pass bufferalloc downstream from the same thread and only act as a thread boundary for events and buffers. --- gst/matroska/matroska-demux.c | 48 ++++++----------------------------- 1 file changed, 8 insertions(+), 40 deletions(-) diff --git a/gst/matroska/matroska-demux.c b/gst/matroska/matroska-demux.c index 7a0eb3183b..8332996d56 100644 --- a/gst/matroska/matroska-demux.c +++ b/gst/matroska/matroska-demux.c @@ -4157,17 +4157,10 @@ gst_matroska_demux_push_hdr_buf (GstMatroskaDemux * demux, GstMatroskaTrackContext * stream, guint8 * data, guint len) { GstFlowReturn ret, cret; - GstBuffer *header_buf = NULL; - - ret = gst_pad_alloc_buffer_and_set_caps (stream->pad, - GST_BUFFER_OFFSET_NONE, len, stream->caps, &header_buf); - - /* we combine but don't use the combined value to check if we have a buffer - * or not. The combined value is what we return. */ - cret = gst_matroska_demux_combine_flows (demux, stream, ret); - if (ret != GST_FLOW_OK) - goto no_buffer; + GstBuffer *header_buf; + header_buf = gst_buffer_new_and_alloc (len); + gst_buffer_set_caps (header_buf, stream->caps); memcpy (GST_BUFFER_DATA (header_buf), data, len); if (stream->set_discont) { @@ -4181,14 +4174,6 @@ gst_matroska_demux_push_hdr_buf (GstMatroskaDemux * demux, cret = gst_matroska_demux_combine_flows (demux, stream, ret); return cret; - - /* ERRORS */ -no_buffer: - { - GST_DEBUG_OBJECT (demux, "could not alloc buffer: %s, combined %s", - gst_flow_get_name (ret), gst_flow_get_name (cret)); - return cret; - } } static GstFlowReturn @@ -4423,18 +4408,9 @@ gst_matroska_demux_add_mpeg_seq_header (GstElement * element, /* Sequence start code, if not found prepend */ if (header != 0x000001b3) { GstBuffer *newbuf; - GstFlowReturn ret, cret; - ret = gst_pad_alloc_buffer_and_set_caps (stream->pad, - GST_BUFFER_OFFSET_NONE, GST_BUFFER_SIZE (*buf) + seq_header_len, - stream->caps, &newbuf); - cret = gst_matroska_demux_combine_flows (demux, stream, ret); - if (ret != GST_FLOW_OK) { - GST_WARNING_OBJECT (demux, "Reallocating buffer for sequence header " - "failed: %s, combined flow return: %s", gst_flow_get_name (ret), - gst_flow_get_name (cret)); - return cret; - } + newbuf = gst_buffer_new_and_alloc (GST_BUFFER_SIZE (*buf) + seq_header_len); + gst_buffer_set_caps (newbuf, stream->caps); GST_DEBUG_OBJECT (demux, "Prepending MPEG sequence header"); gst_buffer_copy_metadata (newbuf, *buf, GST_BUFFER_COPY_TIMESTAMPS | @@ -4459,7 +4435,6 @@ gst_matroska_demux_add_wvpk_header (GstElement * element, GstBuffer *newbuf = NULL; guint8 *data; guint newlen; - GstFlowReturn ret, cret = GST_FLOW_OK; Wavpack4Header wvh; wvh.ck_id[0] = 'w'; @@ -4486,15 +4461,8 @@ gst_matroska_demux_add_wvpk_header (GstElement * element, /* block_samples, flags and crc are already in the buffer */ newlen = GST_BUFFER_SIZE (*buf) + sizeof (Wavpack4Header) - 12; - ret = - gst_pad_alloc_buffer_and_set_caps (stream->pad, GST_BUFFER_OFFSET_NONE, - newlen, stream->caps, &newbuf); - cret = gst_matroska_demux_combine_flows (demux, stream, ret); - if (ret != GST_FLOW_OK) { - GST_DEBUG_OBJECT (demux, "pad_alloc failed %s, combined %s", - gst_flow_get_name (ret), gst_flow_get_name (cret)); - return cret; - } + newbuf = gst_buffer_new_and_alloc (newlen); + gst_buffer_set_caps (newbuf, stream->caps); data = GST_BUFFER_DATA (newbuf); data[0] = 'w'; @@ -4590,7 +4558,7 @@ gst_matroska_demux_add_wvpk_header (GstElement * element, audiocontext->wvpk_block_index += block_samples; } - return cret; + return GST_FLOW_OK; } static GstFlowReturn From d2a43eb8ff3bfe874ad6a1f758959f6d7959a028 Mon Sep 17 00:00:00 2001 From: Edward Hervey Date: Wed, 22 Dec 2010 21:27:11 +0100 Subject: [PATCH 063/351] ximagesrc: remote is a boolean (and not uint) property --- sys/ximage/gstximagesrc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sys/ximage/gstximagesrc.c b/sys/ximage/gstximagesrc.c index e6a15a1ebd..e705ccbe32 100644 --- a/sys/ximage/gstximagesrc.c +++ b/sys/ximage/gstximagesrc.c @@ -876,7 +876,7 @@ gst_ximage_src_get_property (GObject * object, guint prop_id, GValue * value, g_value_set_uint (value, src->endy); break; case PROP_REMOTE: - g_value_set_uint (value, src->remote); + g_value_set_boolean (value, src->remote); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); From 1230258e6fb8b7bf9e88e153ea14783dc7608a19 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Thu, 23 Dec 2010 11:10:55 +0100 Subject: [PATCH 064/351] docs: add some more gstrtpbin docs --- gst/rtpmanager/gstrtpbin.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/gst/rtpmanager/gstrtpbin.c b/gst/rtpmanager/gstrtpbin.c index 48e533ff78..5c61460049 100644 --- a/gst/rtpmanager/gstrtpbin.c +++ b/gst/rtpmanager/gstrtpbin.c @@ -58,6 +58,11 @@ * mapping. One can clear the cached values with the #GstRtpSession::clear-pt-map * signal. * + * Access to the internal statistics of gstrtpbin is provided with the + * get-internal-session property. This action signal gives access to the + * RTPSession object which further provides action signals to retrieve the + * internal source and other sources. + * * * Example pipelines * |[ From 10a5a795eabf2f97a4903c9d08082d9ecfcfca77 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Thu, 23 Dec 2010 12:58:05 +0100 Subject: [PATCH 065/351] rtpsession: also emit RTCP activity on SR Also emit RTCP activity signals when we receive an SR packet without RB blocks, such as from a sender that is not receiving anything. --- gst/rtpmanager/rtpsession.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c index 0a0ee88ad3..83d763a7bf 100644 --- a/gst/rtpmanager/rtpsession.c +++ b/gst/rtpmanager/rtpsession.c @@ -1647,10 +1647,9 @@ rtp_session_process_rb (RTPSession * sess, RTPSource * source, * the other sender to see if we are better or worse. */ rtp_source_process_rb (source, arrival->current_time, fractionlost, packetslost, exthighestseq, jitter, lsr, dlsr); - - on_ssrc_active (sess, source); } } + on_ssrc_active (sess, source); } /* A Sender report contains statistics about how the sender is doing. This From 17c45a886913f4475744674420d00601636aff12 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Thu, 23 Dec 2010 12:59:59 +0100 Subject: [PATCH 066/351] examples: add example RTP stats Add some more RTP examples for how to retrieve RTP stats in a receiver. --- tests/examples/rtp/client-PCMA.c | 46 ++++++++++++++++++++++++++++++++ 1 file changed, 46 insertions(+) diff --git a/tests/examples/rtp/client-PCMA.c b/tests/examples/rtp/client-PCMA.c index 0c895a2397..885b8264e2 100644 --- a/tests/examples/rtp/client-PCMA.c +++ b/tests/examples/rtp/client-PCMA.c @@ -55,6 +55,48 @@ * from another machine, change this address. */ #define DEST_HOST "127.0.0.1" +/* print the stats of a source */ +static void +print_source_stats (GObject * source) +{ + GstStructure *stats; + gchar *str; + + g_return_if_fail (source != NULL); + + /* get the source stats */ + g_object_get (source, "stats", &stats, NULL); + + /* simply dump the stats structure */ + str = gst_structure_to_string (stats); + g_print ("source stats: %s\n", str); + + gst_structure_free (stats); + g_free (str); +} + +/* will be called when gstrtpbin signals on-ssrc-active. It means that an RTCP + * packet was received from another source. */ +static void +on_ssrc_active_cb (GstElement * rtpbin, guint sessid, guint ssrc, + GstElement * depay) +{ + GObject *session, *isrc, *osrc; + + g_print ("got RTCP from session %u, SSRC %u\n", sessid, ssrc); + + /* get the right session */ + g_signal_emit_by_name (rtpbin, "get-internal-session", sessid, &session); + + /* get the internal source (the SSRC allocated to us, the receiver */ + g_object_get (session, "internal-source", &isrc, NULL); + print_source_stats (isrc); + + /* get the remote source that sent us RTCP */ + g_signal_emit_by_name (session, "get-source-by-ssrc", ssrc, &osrc); + print_source_stats (osrc); +} + /* will be called when rtpbin has validated a payload that we can depayload */ static void pad_added_cb (GstElement * rtpbin, GstPad * new_pad, GstElement * depay) @@ -174,6 +216,10 @@ main (int argc, char *argv[]) * user_data so that we can link to it. */ g_signal_connect (rtpbin, "pad-added", G_CALLBACK (pad_added_cb), audiodepay); + /* give some stats when we receive RTCP */ + g_signal_connect (rtpbin, "on-ssrc-active", G_CALLBACK (on_ssrc_active_cb), + audiodepay); + /* set the pipeline to playing */ g_print ("starting receiver pipeline\n"); gst_element_set_state (pipeline, GST_STATE_PLAYING); From 6035ee08c0de420a358e5ecedbc6b8102046ddf1 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Thu, 23 Dec 2010 13:00:49 +0100 Subject: [PATCH 067/351] rtpsource: include all stats and document Include all possible stats of a source in the stats structure because we might be interested in what happened in the past. Document the stats property and the fields. --- gst/rtpmanager/rtpsource.c | 162 +++++++++++++++++++++++-------------- 1 file changed, 101 insertions(+), 61 deletions(-) diff --git a/gst/rtpmanager/rtpsource.c b/gst/rtpmanager/rtpsource.c index e0432b76d5..ad6160fada 100644 --- a/gst/rtpmanager/rtpsource.c +++ b/gst/rtpmanager/rtpsource.c @@ -123,6 +123,62 @@ rtp_source_class_init (RTPSourceClass * klass) * The statistics of the source. This property returns a GstStructure with * name application/x-rtp-source-stats with the following fields: * + * "ssrc" G_TYPE_UINT The SSRC of this source + * "internal" G_TYPE_BOOLEAN If this source is the source of the session + * "validated" G_TYPE_BOOLEAN If the source is validated + * "received-bye" G_TYPE_BOOLEAN If we received a BYE from this source + * "is-csrc" G_TYPE_BOOLEAN If this source was found as CSRC + * "is-sender" G_TYPE_BOOLEAN If this source is a sender + * "seqnum-base" G_TYPE_INT first seqnum if known + * "clock-rate" G_TYPE_INT the clock rate of the media + * + * The following two fields are only present when known. + * + * "rtp-from" G_TYPE_STRING where we received the last RTP packet from + * "rtcp-from" G_TYPE_STRING where we received the last RTCP packet from + * + * The following fields make sense for internal sources and will only increase + * when "is-sender" is TRUE: + * + * "octets-sent" G_TYPE_UINT64 number of bytes we sent + * "packets-sent" G_TYPE_UINT64 number of packets we sent + * + * The following fields make sense for non-internal sources and will only + * increase when "is-sender" is TRUE. + * + * "octets-received" G_TYPE_UINT64 total number of bytes received + * "packets-received" G_TYPE_UINT64 total number of packets received + * + * Following fields are updated when "is-sender" is TRUE. + * + * "bitrate" G_TYPE_UINT64 bitrate in bits per second + * "jitter" G_TYPE_UINT estimated jitter + * "packets-lost" G_TYPE_INT estimated amount of packets lost + * + * The last SR report this source sent. This only updates when "is-sender" is + * TRUE. + * + * "have-sr" G_TYPE_BOOLEAN the source has sent SR + * "sr-ntptime" G_TYPE_UINT64 ntptime of SR + * "sr-rtptime" G_TYPE_UINT rtptime of SR + * "sr-octet-count" G_TYPE_UINT the number of bytes in the SR + * "sr-packet-count" G_TYPE_UINT the number of packets in the SR + * + * The last RB that this source sent. This is only updated when the source is + * receiving data. + * + * "have-rb" G_TYPE_BOOLEAN the source has sent RB + * "rb-fractionlost" G_TYPE_UINT lost fraction + * "rb-packetslost" G_TYPE_INT lost packets + * "rb-exthighestseq" G_TYPE_UINT highest received seqnum + * "rb-jitter" G_TYPE_UINT reception jitter + * "rb-lsr" G_TYPE_UINT last SR time + * "rb-dlsr" G_TYPE_UINT delay since last SR + * + * The round trip of this source. This is calculated from the last RB + * values and the recption time of the last RB packet. + * + * "rb-round-trip" G_TYPE_UINT the round trip time in nanoseconds */ g_object_class_install_property (gobject_class, PROP_STATS, g_param_spec_boxed ("stats", "Stats", @@ -202,6 +258,21 @@ rtp_source_create_stats (RTPSource * src) gboolean is_sender = src->is_sender; gboolean internal = src->internal; gchar address_str[GST_NETADDRESS_MAX_LEN]; + gboolean have_rb; + guint8 fractionlost = 0; + gint32 packetslost = 0; + guint32 exthighestseq = 0; + guint32 jitter = 0; + guint32 lsr = 0; + guint32 dlsr = 0; + guint32 round_trip = 0; + gboolean have_sr; + GstClockTime time = 0; + guint64 ntptime = 0; + guint32 rtptime = 0; + guint32 packet_count = 0; + guint32 octet_count = 0; + /* common data for all types of sources */ s = gst_structure_new ("application/x-rtp-source-stats", @@ -226,70 +297,39 @@ rtp_source_create_stats (RTPSource * src) gst_structure_set (s, "rtcp-from", G_TYPE_STRING, address_str, NULL); } - if (internal) { - /* our internal source */ + gst_structure_set (s, + "octets-sent", G_TYPE_UINT64, src->stats.octets_sent, + "packets-sent", G_TYPE_UINT64, src->stats.packets_sent, + "octets-received", G_TYPE_UINT64, src->stats.octets_received, + "packets-received", G_TYPE_UINT64, src->stats.packets_received, + "bitrate", G_TYPE_UINT64, src->bitrate, + "packets-lost", G_TYPE_INT, + (gint) rtp_stats_get_packets_lost (&src->stats), "jitter", G_TYPE_UINT, + (guint) (src->stats.jitter >> 4), NULL); - /* report accumulated send statistics, other sources will have a RB with - * info on reception. */ - gst_structure_set (s, - "octets-sent", G_TYPE_UINT64, src->stats.octets_sent, - "packets-sent", G_TYPE_UINT64, src->stats.packets_sent, NULL); + /* get the last SR. */ + have_sr = rtp_source_get_last_sr (src, &time, &ntptime, &rtptime, + &packet_count, &octet_count); + gst_structure_set (s, + "have-sr", G_TYPE_BOOLEAN, have_sr, + "sr-ntptime", G_TYPE_UINT64, ntptime, + "sr-rtptime", G_TYPE_UINT, (guint) rtptime, + "sr-octet-count", G_TYPE_UINT, (guint) octet_count, + "sr-packet-count", G_TYPE_UINT, (guint) packet_count, NULL); - if (is_sender) - gst_structure_set (s, "bitrate", G_TYPE_UINT64, src->bitrate, NULL); + /* get the last RB */ + have_rb = rtp_source_get_last_rb (src, &fractionlost, &packetslost, + &exthighestseq, &jitter, &lsr, &dlsr, &round_trip); - } else { - /* other sources */ - gboolean have_rb; - guint8 fractionlost = 0; - gint32 packetslost = 0; - guint32 exthighestseq = 0; - guint32 jitter = 0; - guint32 lsr = 0; - guint32 dlsr = 0; - guint32 round_trip = 0; - - gst_structure_set (s, - "octets-received", G_TYPE_UINT64, src->stats.octets_received, - "packets-received", G_TYPE_UINT64, src->stats.packets_received, - "bitrate", G_TYPE_UINT64, src->bitrate, - "packets-lost", G_TYPE_INT, - (gint) rtp_stats_get_packets_lost (&src->stats), "jitter", G_TYPE_UINT, - (guint) (src->stats.jitter >> 4), NULL); - - if (is_sender) { - gboolean have_sr; - GstClockTime time = 0; - guint64 ntptime = 0; - guint32 rtptime = 0; - guint32 packet_count = 0; - guint32 octet_count = 0; - - /* this source is sending to us, get the last SR. */ - have_sr = rtp_source_get_last_sr (src, &time, &ntptime, &rtptime, - &packet_count, &octet_count); - gst_structure_set (s, - "have-sr", G_TYPE_BOOLEAN, have_sr, - "sr-ntptime", G_TYPE_UINT64, ntptime, - "sr-rtptime", G_TYPE_UINT, (guint) rtptime, - "sr-octet-count", G_TYPE_UINT, (guint) octet_count, - "sr-packet-count", G_TYPE_UINT, (guint) packet_count, NULL); - } - /* we might be sending to this SSRC so we report about how it is - * receiving our data */ - have_rb = rtp_source_get_last_rb (src, &fractionlost, &packetslost, - &exthighestseq, &jitter, &lsr, &dlsr, &round_trip); - - gst_structure_set (s, - "have-rb", G_TYPE_BOOLEAN, have_rb, - "rb-fractionlost", G_TYPE_UINT, (guint) fractionlost, - "rb-packetslost", G_TYPE_INT, (gint) packetslost, - "rb-exthighestseq", G_TYPE_UINT, (guint) exthighestseq, - "rb-jitter", G_TYPE_UINT, (guint) jitter, - "rb-lsr", G_TYPE_UINT, (guint) lsr, - "rb-dlsr", G_TYPE_UINT, (guint) dlsr, - "rb-round-trip", G_TYPE_UINT, (guint) round_trip, NULL); - } + gst_structure_set (s, + "have-rb", G_TYPE_BOOLEAN, have_rb, + "rb-fractionlost", G_TYPE_UINT, (guint) fractionlost, + "rb-packetslost", G_TYPE_INT, (gint) packetslost, + "rb-exthighestseq", G_TYPE_UINT, (guint) exthighestseq, + "rb-jitter", G_TYPE_UINT, (guint) jitter, + "rb-lsr", G_TYPE_UINT, (guint) lsr, + "rb-dlsr", G_TYPE_UINT, (guint) dlsr, + "rb-round-trip", G_TYPE_UINT, (guint) round_trip, NULL); return s; } From 8fa5ddab9a7a9ed009bbfce9ce3de1d1867d249c Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Thu, 23 Dec 2010 13:52:57 +0100 Subject: [PATCH 068/351] rtpsession: remember last sent RB values. --- gst/rtpmanager/rtpsession.c | 9 +++++++++ gst/rtpmanager/rtpsource.h | 1 + 2 files changed, 10 insertions(+) diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c index 83d763a7bf..2698f3c7a9 100644 --- a/gst/rtpmanager/rtpsession.c +++ b/gst/rtpmanager/rtpsession.c @@ -2357,6 +2357,15 @@ session_report_blocks (const gchar * key, RTPSource * source, ReportData * data) rtp_source_get_new_rb (source, data->current_time, &fractionlost, &packetslost, &exthighestseq, &jitter, &lsr, &dlsr); + /* store last generated RR packet */ + source->last_rr.is_valid = TRUE; + source->last_rr.fractionlost = fractionlost; + source->last_rr.packetslost = packetslost; + source->last_rr.exthighestseq = exthighestseq; + source->last_rr.jitter = jitter; + source->last_rr.lsr = lsr; + source->last_rr.dlsr = dlsr; + /* packet is not yet filled, add report block for this source. */ gst_rtcp_packet_add_rb (packet, source->ssrc, fractionlost, packetslost, exthighestseq, jitter, lsr, dlsr); diff --git a/gst/rtpmanager/rtpsource.h b/gst/rtpmanager/rtpsource.h index 54a86ac47c..94c22b3f5a 100644 --- a/gst/rtpmanager/rtpsource.h +++ b/gst/rtpmanager/rtpsource.h @@ -167,6 +167,7 @@ struct _RTPSource { gpointer user_data; RTPSourceStats stats; + RTPReceiverReport last_rr; GList *conflicting_addresses; }; From 7caad21a57d196894991daddd9fcd5a1311a1f87 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Thu, 23 Dec 2010 13:55:31 +0100 Subject: [PATCH 069/351] rtpsource: include last send RB block Only report RB values for non-internal sources. Report not only the RB blocks we last received from but also the last RB block we sent to a source. --- gst/rtpmanager/rtpsource.c | 57 ++++++++++++++++++++++++++++---------- 1 file changed, 42 insertions(+), 15 deletions(-) diff --git a/gst/rtpmanager/rtpsource.c b/gst/rtpmanager/rtpsource.c index ad6160fada..3d5b6c1f57 100644 --- a/gst/rtpmanager/rtpsource.c +++ b/gst/rtpmanager/rtpsource.c @@ -164,8 +164,21 @@ rtp_source_class_init (RTPSourceClass * klass) * "sr-octet-count" G_TYPE_UINT the number of bytes in the SR * "sr-packet-count" G_TYPE_UINT the number of packets in the SR * - * The last RB that this source sent. This is only updated when the source is - * receiving data. + * The following fields are only present for non-internal sources and + * represent the content of the last RB packet that was sent to this source. + * These values are only updated when the source is sending. + * + * "sent-rb" G_TYPE_BOOLEAN we have sent an RB + * "sent-rb-fractionlost" G_TYPE_UINT calculated lost fraction + * "sent-rb-packetslost" G_TYPE_INT lost packets + * "sent-rb-exthighestseq" G_TYPE_UINT last seen seqnum + * "sent-rb-jitter" G_TYPE_UINT jitter + * "sent-rb-lsr" G_TYPE_UINT last SR time + * "sent-rb-dlsr" G_TYPE_UINT delay since last SR + * + * The following fields are only present for non-internal sources and + * represents the last RB that this source sent. This is only updated + * when the source is receiving data and sending RB blocks. * * "have-rb" G_TYPE_BOOLEAN the source has sent RB * "rb-fractionlost" G_TYPE_UINT lost fraction @@ -176,7 +189,8 @@ rtp_source_class_init (RTPSourceClass * klass) * "rb-dlsr" G_TYPE_UINT delay since last SR * * The round trip of this source. This is calculated from the last RB - * values and the recption time of the last RB packet. + * values and the recption time of the last RB packet. Only present for + * non-internal sources. * * "rb-round-trip" G_TYPE_UINT the round trip time in nanoseconds */ @@ -317,19 +331,32 @@ rtp_source_create_stats (RTPSource * src) "sr-octet-count", G_TYPE_UINT, (guint) octet_count, "sr-packet-count", G_TYPE_UINT, (guint) packet_count, NULL); - /* get the last RB */ - have_rb = rtp_source_get_last_rb (src, &fractionlost, &packetslost, - &exthighestseq, &jitter, &lsr, &dlsr, &round_trip); + if (!internal) { + /* get the last RB we sent */ + gst_structure_set (s, + "sent-rb", G_TYPE_BOOLEAN, have_rb, + "sent-rb-fractionlost", G_TYPE_UINT, (guint) src->last_rr.fractionlost, + "sent-rb-packetslost", G_TYPE_INT, (gint) src->last_rr.packetslost, + "sent-rb-exthighestseq", G_TYPE_UINT, + (guint) src->last_rr.exthighestseq, "sent-rb-jitter", G_TYPE_UINT, + (guint) src->last_rr.jitter, "sent-rb-lsr", G_TYPE_UINT, + (guint) src->last_rr.lsr, "sent-rb-dlsr", G_TYPE_UINT, + (guint) src->last_rr.dlsr, NULL); - gst_structure_set (s, - "have-rb", G_TYPE_BOOLEAN, have_rb, - "rb-fractionlost", G_TYPE_UINT, (guint) fractionlost, - "rb-packetslost", G_TYPE_INT, (gint) packetslost, - "rb-exthighestseq", G_TYPE_UINT, (guint) exthighestseq, - "rb-jitter", G_TYPE_UINT, (guint) jitter, - "rb-lsr", G_TYPE_UINT, (guint) lsr, - "rb-dlsr", G_TYPE_UINT, (guint) dlsr, - "rb-round-trip", G_TYPE_UINT, (guint) round_trip, NULL); + /* get the last RB */ + have_rb = rtp_source_get_last_rb (src, &fractionlost, &packetslost, + &exthighestseq, &jitter, &lsr, &dlsr, &round_trip); + + gst_structure_set (s, + "have-rb", G_TYPE_BOOLEAN, have_rb, + "rb-fractionlost", G_TYPE_UINT, (guint) fractionlost, + "rb-packetslost", G_TYPE_INT, (gint) packetslost, + "rb-exthighestseq", G_TYPE_UINT, (guint) exthighestseq, + "rb-jitter", G_TYPE_UINT, (guint) jitter, + "rb-lsr", G_TYPE_UINT, (guint) lsr, + "rb-dlsr", G_TYPE_UINT, (guint) dlsr, + "rb-round-trip", G_TYPE_UINT, (guint) round_trip, NULL); + } return s; } From 12bc7258b923effe77e4efc92295a0cc42c01f10 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Thu, 23 Dec 2010 15:24:29 +0100 Subject: [PATCH 070/351] rtspsrc: improve RTP session handling Store the RTP session in the stream so that we can more efficiently perform actions on the stream based on RTP signals. --- gst/rtsp/gstrtspsrc.c | 165 +++++++++++++++++++++--------------------- gst/rtsp/gstrtspsrc.h | 9 ++- 2 files changed, 90 insertions(+), 84 deletions(-) diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c index a21a14ad2a..5b2e84d70f 100644 --- a/gst/rtsp/gstrtspsrc.c +++ b/gst/rtsp/gstrtspsrc.c @@ -1126,6 +1126,10 @@ gst_rtspsrc_stream_free (GstRTSPSrc * src, GstRTSPStream * stream) gst_object_unref (stream->rtcppad); stream->rtcppad = NULL; } + if (stream->session) { + g_object_unref (stream->session); + stream->session = NULL; + } g_free (stream); } @@ -1143,14 +1147,14 @@ gst_rtspsrc_cleanup (GstRTSPSrc * src) } g_list_free (src->streams); src->streams = NULL; - if (src->session) { - if (src->session_sig_id) { - g_signal_handler_disconnect (src->session, src->session_sig_id); - src->session_sig_id = 0; + if (src->manager) { + if (src->manager_sig_id) { + g_signal_handler_disconnect (src->manager, src->manager_sig_id); + src->manager_sig_id = 0; } - gst_element_set_state (src->session, GST_STATE_NULL); - gst_bin_remove (GST_BIN_CAST (src), src->session); - src->session = NULL; + gst_element_set_state (src->manager, GST_STATE_NULL); + gst_bin_remove (GST_BIN_CAST (src), src->manager); + src->manager = NULL; } src->numstreams = 0; if (src->props) @@ -1672,8 +1676,8 @@ gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush) if (base_time != -1) gst_element_set_base_time (GST_ELEMENT_CAST (src), base_time); /* to manage jitterbuffer buffer mode */ - if (src->session) - gst_element_set_base_time (GST_ELEMENT_CAST (src->session), base_time); + if (src->manager) + gst_element_set_base_time (GST_ELEMENT_CAST (src->manager), base_time); } static GstRTSPResult @@ -2138,7 +2142,7 @@ was_ok: /* this callback is called when the session manager generated a new src pad with * payloaded RTP packets. We simply ghost the pad here. */ static void -new_session_pad (GstElement * session, GstPad * pad, GstRTSPSrc * src) +new_manager_pad (GstElement * manager, GstPad * pad, GstRTSPSrc * src) { gchar *name; GstPadTemplate *template; @@ -2147,7 +2151,7 @@ new_session_pad (GstElement * session, GstPad * pad, GstRTSPSrc * src) GstRTSPStream *stream; gboolean all_added; - GST_DEBUG_OBJECT (src, "got new session pad %" GST_PTR_FORMAT, pad); + GST_DEBUG_OBJECT (src, "got new manager pad %" GST_PTR_FORMAT, pad); GST_RTSP_STATE_LOCK (src); /* find stream */ @@ -2210,7 +2214,7 @@ unknown_stream: } static GstCaps * -request_pt_map (GstElement * sess, guint session, guint pt, GstRTSPSrc * src) +request_pt_map (GstElement * manager, guint session, guint pt, GstRTSPSrc * src) { GstRTSPStream *stream; GstCaps *caps; @@ -2238,17 +2242,12 @@ unknown_stream: } static void -gst_rtspsrc_do_stream_eos (GstRTSPSrc * src, guint session) +gst_rtspsrc_do_stream_eos (GstRTSPSrc * src, GstRTSPStream * stream) { - GstRTSPStream *stream; + guint session = stream->id; GST_DEBUG_OBJECT (src, "setting stream for session %u to EOS", session); - /* get stream for session */ - stream = find_stream (src, &session, (gpointer) find_stream_by_id); - if (!stream) - goto unknown_stream; - if (stream->eos) goto was_eos; @@ -2257,11 +2256,6 @@ gst_rtspsrc_do_stream_eos (GstRTSPSrc * src, guint session) return; /* ERRORS */ -unknown_stream: - { - GST_DEBUG_OBJECT (src, "unknown stream for session %u", session); - return; - } was_eos: { GST_DEBUG_OBJECT (src, "stream for session %u was already EOS", session); @@ -2270,30 +2264,41 @@ was_eos: } static void -on_bye_ssrc (GstElement * manager, guint session, guint32 ssrc, - GstRTSPSrc * src) +on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream) { - GST_DEBUG_OBJECT (src, "SSRC %08x in session %u received BYE", ssrc, session); + GstRTSPSrc *src = stream->parent; - gst_rtspsrc_do_stream_eos (src, session); + GST_DEBUG_OBJECT (src, "source in session %u received BYE", stream->id); + + gst_rtspsrc_do_stream_eos (src, stream); } static void -on_timeout (GstElement * manager, guint session, guint32 ssrc, GstRTSPSrc * src) +on_timeout (GObject * session, GObject * source, GstRTSPStream * stream) { - GST_DEBUG_OBJECT (src, "SSRC %08x in session %u timed out", ssrc, session); + GstRTSPSrc *src = stream->parent; - gst_rtspsrc_do_stream_eos (src, session); + GST_DEBUG_OBJECT (src, "source in session %u timed out", stream->id); + + gst_rtspsrc_do_stream_eos (src, stream); } static void -on_npt_stop (GstElement * manager, guint session, guint32 ssrc, - GstRTSPSrc * src) +on_npt_stop (GObject * session, GObject * source, GstRTSPStream * stream) { - GST_DEBUG_OBJECT (src, "SSRC %08x in session %u reached the NPT stop", ssrc, - session); + GstRTSPSrc *src = stream->parent; - gst_rtspsrc_do_stream_eos (src, session); + GST_DEBUG_OBJECT (src, "source in session %u reached NPT stop", stream->id); + + gst_rtspsrc_do_stream_eos (src, stream); +} + +static void +on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream) +{ + GstRTSPSrc *src = stream->parent; + + GST_DEBUG_OBJECT (src, "source in session %u is active", stream->id); } /* try to get and configure a manager */ @@ -2313,11 +2318,11 @@ gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream, GST_DEBUG_OBJECT (src, "using manager %s", manager); /* configure the manager */ - if (src->session == NULL) { + if (src->manager == NULL) { GObjectClass *klass; GstState target; - if (!(src->session = gst_element_factory_make (manager, NULL))) { + if (!(src->manager = gst_element_factory_make (manager, NULL))) { /* fallback */ if (gst_rtsp_transport_get_manager (transport->trans, &manager, 1) < 0) goto no_manager; @@ -2325,27 +2330,27 @@ gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream, if (!manager) goto use_no_manager; - if (!(src->session = gst_element_factory_make (manager, NULL))) + if (!(src->manager = gst_element_factory_make (manager, NULL))) goto manager_failed; } /* we manage this element */ - gst_bin_add (GST_BIN_CAST (src), src->session); + gst_bin_add (GST_BIN_CAST (src), src->manager); GST_OBJECT_LOCK (src); target = GST_STATE_TARGET (src); GST_OBJECT_UNLOCK (src); - ret = gst_element_set_state (src->session, target); + ret = gst_element_set_state (src->manager, target); if (ret == GST_STATE_CHANGE_FAILURE) - goto start_session_failure; + goto start_manager_failure; - g_object_set (src->session, "latency", src->latency, NULL); + g_object_set (src->manager, "latency", src->latency, NULL); - klass = G_OBJECT_GET_CLASS (G_OBJECT (src->session)); + klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager)); if (g_object_class_find_property (klass, "buffer-mode")) { if (src->buffer_mode != BUFFER_MODE_AUTO) { - g_object_set (src->session, "buffer-mode", src->buffer_mode, NULL); + g_object_set (src->manager, "buffer-mode", src->buffer_mode, NULL); } else { gboolean need_slave; GstStructure *s; @@ -2368,11 +2373,11 @@ gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream, if (GST_CLOCK_TIME_IS_VALID (src->segment.duration) && src->segment.duration && !need_slave) { GST_DEBUG_OBJECT (src, "selected buffer"); - g_object_set (src->session, "buffer-mode", BUFFER_MODE_BUFFER, + g_object_set (src->manager, "buffer-mode", BUFFER_MODE_BUFFER, NULL); } else { GST_DEBUG_OBJECT (src, "selected slave"); - g_object_set (src->session, "buffer-mode", BUFFER_MODE_SLAVE, NULL); + g_object_set (src->manager, "buffer-mode", BUFFER_MODE_SLAVE, NULL); } } } @@ -2380,46 +2385,35 @@ gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream, /* connect to signals if we did not already do so */ GST_DEBUG_OBJECT (src, "connect to signals on session manager, stream %p", stream); - src->session_sig_id = - g_signal_connect (src->session, "pad-added", - (GCallback) new_session_pad, src); - src->session_ptmap_id = - g_signal_connect (src->session, "request-pt-map", + src->manager_sig_id = + g_signal_connect (src->manager, "pad-added", + (GCallback) new_manager_pad, src); + src->manager_ptmap_id = + g_signal_connect (src->manager, "request-pt-map", (GCallback) request_pt_map, src); - g_signal_connect (src->session, "on-bye-ssrc", (GCallback) on_bye_ssrc, - src); - g_signal_connect (src->session, "on-bye-timeout", (GCallback) on_timeout, - src); - g_signal_connect (src->session, "on-timeout", (GCallback) on_timeout, - src); - /* FIXME: remove this once the rdtmanager is released */ - if (g_signal_lookup ("on-npt-stop", G_OBJECT_TYPE (src->session)) != 0) { - g_signal_connect (src->session, "on-npt-stop", (GCallback) on_npt_stop, - src); - } else { - GST_INFO_OBJECT (src, "skipping on-npt-stop handling, not implemented"); - } } /* we stream directly to the manager, get some pads. Each RTSP stream goes * into a separate RTP session. */ name = g_strdup_printf ("recv_rtp_sink_%d", stream->id); - stream->channelpad[0] = gst_element_get_request_pad (src->session, name); + stream->channelpad[0] = gst_element_get_request_pad (src->manager, name); g_free (name); name = g_strdup_printf ("recv_rtcp_sink_%d", stream->id); - stream->channelpad[1] = gst_element_get_request_pad (src->session, name); + stream->channelpad[1] = gst_element_get_request_pad (src->manager, name); g_free (name); - /* now configure the bandwidth in the session */ + /* now configure the bandwidth in the manager */ if (g_signal_lookup ("get-internal-session", - G_OBJECT_TYPE (src->session)) != 0) { + G_OBJECT_TYPE (src->manager)) != 0) { GObject *rtpsession; - g_signal_emit_by_name (src->session, "get-internal-session", stream->id, + g_signal_emit_by_name (src->manager, "get-internal-session", stream->id, &rtpsession); if (rtpsession) { GST_INFO_OBJECT (src, "configure bandwidth in session %p", rtpsession); + stream->session = rtpsession; + if (stream->as_bandwidth != -1) { GST_INFO_OBJECT (src, "setting AS: %f", (gdouble) (stream->as_bandwidth * 1000)); @@ -2436,7 +2430,16 @@ gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream, g_object_set (rtpsession, "rtcp-rs-bandwidth", stream->rs_bandwidth, NULL); } - g_object_unref (rtpsession); + g_signal_connect (rtpsession, "on-bye-ssrc", (GCallback) on_bye_ssrc, + stream); + g_signal_connect (rtpsession, "on-bye-timeout", (GCallback) on_timeout, + stream); + g_signal_connect (rtpsession, "on-timeout", (GCallback) on_timeout, + stream); + g_signal_connect (rtpsession, "on-npt-stop", (GCallback) on_npt_stop, + stream); + g_signal_connect (rtpsession, "on-ssrc-active", + (GCallback) on_ssrc_active, stream); } } } @@ -2455,9 +2458,9 @@ manager_failed: GST_DEBUG_OBJECT (src, "no session manager element %s found", manager); return FALSE; } -start_session_failure: +start_manager_failure: { - GST_DEBUG_OBJECT (src, "could not start session"); + GST_DEBUG_OBJECT (src, "could not start session manager"); return FALSE; } } @@ -2545,7 +2548,7 @@ gst_rtspsrc_stream_configure_tcp (GstRTSPSrc * src, GstRTSPStream * stream, gst_object_unref (template); } /* setup RTCP transport back to the server if we have to. */ - if (src->session && src->do_rtcp) { + if (src->manager && src->do_rtcp) { GstPad *pad; template = gst_static_pad_template_get (&anysinktemplate); @@ -2557,7 +2560,7 @@ gst_rtspsrc_stream_configure_tcp (GstRTSPSrc * src, GstRTSPStream * stream, /* get session RTCP pad */ name = g_strdup_printf ("send_rtcp_src_%d", stream->id); - pad = gst_element_get_request_pad (src->session, name); + pad = gst_element_get_request_pad (src->manager, name); g_free (name); /* and link */ @@ -2780,7 +2783,7 @@ gst_rtspsrc_stream_configure_udp_sinks (GstRTSPSrc * src, do_rtp = (rtp_port != -1); /* it's possible that the server does not want us to send RTCP in which case * the port is -1 */ - do_rtcp = (rtcp_port != -1 && src->session != NULL && src->do_rtcp); + do_rtcp = (rtcp_port != -1 && src->manager != NULL && src->do_rtcp); /* we need a destination when we have RTP or RTCP ports */ if (destination == NULL && (do_rtp || do_rtcp)) @@ -2889,7 +2892,7 @@ gst_rtspsrc_stream_configure_udp_sinks (GstRTSPSrc * src, /* get session RTCP pad */ name = g_strdup_printf ("send_rtcp_src_%d", stream->id); - pad = gst_element_get_request_pad (src->session, name); + pad = gst_element_get_request_pad (src->manager, name); g_free (name); /* and link */ @@ -3066,7 +3069,7 @@ gst_rtspsrc_activate_streams (GstRTSPSrc * src) if (stream->srcpad) { /* if we don't have a session manager, set the caps now. If we have a * session, we will get a notification of the pad and the caps. */ - if (!src->session) { + if (!src->manager) { GST_DEBUG_OBJECT (src, "setting pad caps for stream %p", stream); gst_pad_set_caps (stream->srcpad, stream->caps); } @@ -3134,9 +3137,9 @@ gst_rtspsrc_configure_caps (GstRTSPSrc * src, GstSegment * segment) } GST_DEBUG_OBJECT (src, "stream %p, caps %" GST_PTR_FORMAT, stream, caps); } - if (src->session) { + if (src->manager) { GST_DEBUG_OBJECT (src, "clear session"); - g_signal_emit_by_name (src->session, "clear-pt-map", NULL); + g_signal_emit_by_name (src->manager, "clear-pt-map", NULL); } } @@ -3592,7 +3595,7 @@ gst_rtspsrc_loop_interleaved (GstRTSPSrc * src) src->need_activate = FALSE; } - if (!src->session) { + if (!src->manager) { /* set stream caps on buffer when we don't have a session manager to do it * for us */ gst_buffer_set_caps (buf, stream->caps); diff --git a/gst/rtsp/gstrtspsrc.h b/gst/rtsp/gstrtspsrc.h index bdaff49b0d..00861b6da9 100644 --- a/gst/rtsp/gstrtspsrc.h +++ b/gst/rtsp/gstrtspsrc.h @@ -138,6 +138,9 @@ struct _GstRTSPStream { /* per stream connection */ GstRTSPConnInfo conninfo; + /* session */ + GObject *session; + /* bandwidth */ guint as_bandwidth; guint rs_bandwidth; @@ -233,9 +236,9 @@ struct _GstRTSPSrc { gboolean seekable; /* session management */ - GstElement *session; - gulong session_sig_id; - gulong session_ptmap_id; + GstElement *manager; + gulong manager_sig_id; + gulong manager_ptmap_id; GstRTSPConnInfo conninfo; From 2a49d34c3e79dc392be453985bfb26342e1bbf9b Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Thu, 23 Dec 2010 16:25:15 +0100 Subject: [PATCH 071/351] rtspsrc: on-npt-stop is a manager signal --- gst/rtsp/gstrtspsrc.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c index 5b2e84d70f..b8e6cc6e6d 100644 --- a/gst/rtsp/gstrtspsrc.c +++ b/gst/rtsp/gstrtspsrc.c @@ -2284,13 +2284,17 @@ on_timeout (GObject * session, GObject * source, GstRTSPStream * stream) } static void -on_npt_stop (GObject * session, GObject * source, GstRTSPStream * stream) +on_npt_stop (GstElement * rtpbin, guint session, guint ssrc, GstRTSPSrc * src) { - GstRTSPSrc *src = stream->parent; + GstRTSPStream *stream; - GST_DEBUG_OBJECT (src, "source in session %u reached NPT stop", stream->id); + GST_DEBUG_OBJECT (src, "source in session %u reached NPT stop", session); - gst_rtspsrc_do_stream_eos (src, stream); + /* get stream for session */ + stream = find_stream (src, &session, (gpointer) find_stream_by_id); + if (stream) { + gst_rtspsrc_do_stream_eos (src, stream); + } } static void @@ -2391,6 +2395,9 @@ gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream, src->manager_ptmap_id = g_signal_connect (src->manager, "request-pt-map", (GCallback) request_pt_map, src); + + g_signal_connect (src->manager, "on-npt-stop", (GCallback) on_npt_stop, + src); } /* we stream directly to the manager, get some pads. Each RTSP stream goes @@ -2436,8 +2443,6 @@ gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream, stream); g_signal_connect (rtpsession, "on-timeout", (GCallback) on_timeout, stream); - g_signal_connect (rtpsession, "on-npt-stop", (GCallback) on_npt_stop, - stream); g_signal_connect (rtpsession, "on-ssrc-active", (GCallback) on_ssrc_active, stream); } From 1dd71cc63f01e48610e85fd72d28e4f089183d0c Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Thu, 23 Dec 2010 16:26:07 +0100 Subject: [PATCH 072/351] mp4gdepay: flush state on flush-stop --- gst/rtp/gstrtpmp4gdepay.c | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+) diff --git a/gst/rtp/gstrtpmp4gdepay.c b/gst/rtp/gstrtpmp4gdepay.c index 67d5217ec6..02ac8d681a 100644 --- a/gst/rtp/gstrtpmp4gdepay.c +++ b/gst/rtp/gstrtpmp4gdepay.c @@ -135,6 +135,8 @@ static gboolean gst_rtp_mp4g_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps); static GstBuffer *gst_rtp_mp4g_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf); +static gboolean gst_rtp_mp4g_depay_handle_event (GstBaseRTPDepayload * filter, + GstEvent * event); static GstStateChangeReturn gst_rtp_mp4g_depay_change_state (GstElement * element, GstStateChange transition); @@ -173,6 +175,7 @@ gst_rtp_mp4g_depay_class_init (GstRtpMP4GDepayClass * klass) gstbasertpdepayload_class->process = gst_rtp_mp4g_depay_process; gstbasertpdepayload_class->set_caps = gst_rtp_mp4g_depay_setcaps; + gstbasertpdepayload_class->handle_event = gst_rtp_mp4g_depay_handle_event; GST_DEBUG_CATEGORY_INIT (rtpmp4gdepay_debug, "rtpmp4gdepay", 0, "MP4-generic RTP Depayloader"); @@ -711,6 +714,28 @@ short_payload: } } +static gboolean +gst_rtp_mp4g_depay_handle_event (GstBaseRTPDepayload * filter, GstEvent * event) +{ + gboolean ret; + GstRtpMP4GDepay *rtpmp4gdepay; + + rtpmp4gdepay = GST_RTP_MP4G_DEPAY (filter); + + switch (GST_EVENT_TYPE (event)) { + case GST_EVENT_FLUSH_STOP: + gst_rtp_mp4g_depay_reset (rtpmp4gdepay); + break; + default: + break; + } + + ret = + GST_BASE_RTP_DEPAYLOAD_CLASS (parent_class)->handle_event (filter, event); + + return ret; +} + static GstStateChangeReturn gst_rtp_mp4g_depay_change_state (GstElement * element, GstStateChange transition) From e13340ccb535178a35083350a76bef4045bf742c Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Thu, 23 Dec 2010 17:06:58 +0100 Subject: [PATCH 073/351] mp4gpay: reset state on flush-stop --- gst/rtp/gstrtpmp4gpay.c | 51 ++++++++++++++++++++++++++++++++++++++--- 1 file changed, 48 insertions(+), 3 deletions(-) diff --git a/gst/rtp/gstrtpmp4gpay.c b/gst/rtp/gstrtpmp4gpay.c index 764a8038d1..c095064062 100644 --- a/gst/rtp/gstrtpmp4gpay.c +++ b/gst/rtp/gstrtpmp4gpay.c @@ -82,6 +82,7 @@ static gboolean gst_rtp_mp4g_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps); static GstFlowReturn gst_rtp_mp4g_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buffer); +static gboolean gst_rtp_mp4g_pay_handle_event (GstPad * pad, GstEvent * event); GST_BOILERPLATE (GstRtpMP4GPay, gst_rtp_mp4g_pay, GstBaseRTPPayload, GST_TYPE_BASE_RTP_PAYLOAD) @@ -118,6 +119,7 @@ gst_rtp_mp4g_pay_class_init (GstRtpMP4GPayClass * klass) gstbasertppayload_class->set_caps = gst_rtp_mp4g_pay_setcaps; gstbasertppayload_class->handle_buffer = gst_rtp_mp4g_pay_handle_buffer; + gstbasertppayload_class->handle_event = gst_rtp_mp4g_pay_handle_event; GST_DEBUG_CATEGORY_INIT (rtpmp4gpay_debug, "rtpmp4gpay", 0, "MP4-generic RTP Payloader"); @@ -135,6 +137,13 @@ gst_rtp_mp4g_pay_reset (GstRtpMP4GPay * rtpmp4gpay) GST_DEBUG_OBJECT (rtpmp4gpay, "reset"); gst_adapter_clear (rtpmp4gpay->adapter); + rtpmp4gpay->offset = 0; +} + +static void +gst_rtp_mp4g_pay_cleanup (GstRtpMP4GPay * rtpmp4gpay) +{ + gst_rtp_mp4g_pay_reset (rtpmp4gpay); g_free (rtpmp4gpay->params); rtpmp4gpay->params = NULL; @@ -150,7 +159,6 @@ gst_rtp_mp4g_pay_reset (GstRtpMP4GPay * rtpmp4gpay) rtpmp4gpay->mode = NULL; rtpmp4gpay->frame_len = 0; - rtpmp4gpay->offset = 0; } static void @@ -160,7 +168,7 @@ gst_rtp_mp4g_pay_finalize (GObject * object) rtpmp4gpay = GST_RTP_MP4G_PAY (object); - gst_rtp_mp4g_pay_reset (rtpmp4gpay); + gst_rtp_mp4g_pay_cleanup (rtpmp4gpay); g_object_unref (rtpmp4gpay->adapter); rtpmp4gpay->adapter = NULL; @@ -561,6 +569,35 @@ gst_rtp_mp4g_pay_handle_buffer (GstBaseRTPPayload * basepayload, return gst_rtp_mp4g_pay_flush (rtpmp4gpay); } +static gboolean +gst_rtp_mp4g_pay_handle_event (GstPad * pad, GstEvent * event) +{ + GstRtpMP4GPay *rtpmp4gpay; + + rtpmp4gpay = GST_RTP_MP4G_PAY (gst_pad_get_parent (pad)); + + GST_DEBUG ("Got event: %s", GST_EVENT_TYPE_NAME (event)); + + switch (GST_EVENT_TYPE (event)) { + case GST_EVENT_NEWSEGMENT: + case GST_EVENT_EOS: + /* This flush call makes sure that the last buffer is always pushed + * to the base payloader */ + gst_rtp_mp4g_pay_flush (rtpmp4gpay); + break; + case GST_EVENT_FLUSH_STOP: + gst_rtp_mp4g_pay_reset (rtpmp4gpay); + break; + default: + break; + } + + g_object_unref (rtpmp4gpay); + + /* let parent handle event too */ + return FALSE; +} + static GstStateChangeReturn gst_rtp_mp4g_pay_change_state (GstElement * element, GstStateChange transition) { @@ -571,7 +608,7 @@ gst_rtp_mp4g_pay_change_state (GstElement * element, GstStateChange transition) switch (transition) { case GST_STATE_CHANGE_READY_TO_PAUSED: - gst_rtp_mp4g_pay_reset (rtpmp4gpay); + gst_rtp_mp4g_pay_cleanup (rtpmp4gpay); break; default: break; @@ -579,6 +616,14 @@ gst_rtp_mp4g_pay_change_state (GstElement * element, GstStateChange transition) ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); + switch (transition) { + case GST_STATE_CHANGE_PAUSED_TO_READY: + gst_rtp_mp4g_pay_cleanup (rtpmp4gpay); + break; + default: + break; + } + return ret; } From 7c9b91d2d811475f26541f56ba1f0f3bf9ea6ad2 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Sun, 12 Dec 2010 05:10:01 +0100 Subject: [PATCH 074/351] gstpay/depay: add generic gstreamer payloader Add the beginnings of a generic GStreamer buffers payloader. --- gst/rtp/Makefile.am | 4 + gst/rtp/gstrtp.c | 8 ++ gst/rtp/gstrtpgstdepay.c | 302 +++++++++++++++++++++++++++++++++++++++ gst/rtp/gstrtpgstdepay.h | 63 ++++++++ gst/rtp/gstrtpgstpay.c | 239 +++++++++++++++++++++++++++++++ gst/rtp/gstrtpgstpay.h | 58 ++++++++ 6 files changed, 674 insertions(+) create mode 100644 gst/rtp/gstrtpgstdepay.c create mode 100644 gst/rtp/gstrtpgstdepay.h create mode 100644 gst/rtp/gstrtpgstpay.c create mode 100644 gst/rtp/gstrtpgstpay.h diff --git a/gst/rtp/Makefile.am b/gst/rtp/Makefile.am index 9d431084d3..82ccd5fedb 100644 --- a/gst/rtp/Makefile.am +++ b/gst/rtp/Makefile.am @@ -13,6 +13,8 @@ libgstrtp_la_SOURCES = \ gstrtpceltpay.c \ gstrtpdvdepay.c \ gstrtpdvpay.c \ + gstrtpgstdepay.c \ + gstrtpgstpay.c \ gstrtpilbcdepay.c \ gstrtpilbcpay.c \ gstrtpmpadepay.c \ @@ -104,6 +106,8 @@ noinst_HEADERS = \ gstrtpdvpay.h \ gstrtpamrdepay.h \ gstrtpamrpay.h \ + gstrtpgstdepay.h \ + gstrtpgstpay.h \ gstrtpilbcdepay.h \ gstrtpilbcpay.h \ gstrtppcmadepay.h \ diff --git a/gst/rtp/gstrtp.c b/gst/rtp/gstrtp.c index e2fa2ed413..721cee3c84 100644 --- a/gst/rtp/gstrtp.c +++ b/gst/rtp/gstrtp.c @@ -30,6 +30,8 @@ #include "gstrtpceltpay.h" #include "gstrtpdvdepay.h" #include "gstrtpdvpay.h" +#include "gstrtpgstdepay.h" +#include "gstrtpgstpay.h" #include "gstrtpilbcdepay.h" #include "gstrtpilbcpay.h" #include "gstrtppcmupay.h" @@ -119,6 +121,12 @@ plugin_init (GstPlugin * plugin) if (!gst_rtp_dv_pay_plugin_init (plugin)) return FALSE; + if (!gst_rtp_gst_depay_plugin_init (plugin)) + return FALSE; + + if (!gst_rtp_gst_pay_plugin_init (plugin)) + return FALSE; + if (!gst_rtp_ilbc_pay_plugin_init (plugin)) return FALSE; diff --git a/gst/rtp/gstrtpgstdepay.c b/gst/rtp/gstrtpgstdepay.c new file mode 100644 index 0000000000..633971d31b --- /dev/null +++ b/gst/rtp/gstrtpgstdepay.c @@ -0,0 +1,302 @@ +/* GStreamer + * Copyright (C) <2010> Wim Taymans + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifdef HAVE_CONFIG_H +# include "config.h" +#endif + +#include + +#include +#include "gstrtpgstdepay.h" + +GST_DEBUG_CATEGORY_STATIC (rtpgstdepay_debug); +#define GST_CAT_DEFAULT (rtpgstdepay_debug) + +static GstStaticPadTemplate gst_rtp_gst_depay_src_template = +GST_STATIC_PAD_TEMPLATE ("src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + GST_STATIC_CAPS_ANY); + +static GstStaticPadTemplate gst_rtp_gst_depay_sink_template = +GST_STATIC_PAD_TEMPLATE ("sink", + GST_PAD_SINK, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("application/x-rtp, " + "media = (string) \"application\", " + "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " + "clock-rate = (int) 90000, " "encoding-name = (string) \"X-GST\"") + ); + +GST_BOILERPLATE (GstRtpGSTDepay, gst_rtp_gst_depay, GstBaseRTPDepayload, + GST_TYPE_BASE_RTP_DEPAYLOAD); + +static void gst_rtp_gst_depay_finalize (GObject * object); + +static gboolean gst_rtp_gst_depay_setcaps (GstBaseRTPDepayload * depayload, + GstCaps * caps); +static GstBuffer *gst_rtp_gst_depay_process (GstBaseRTPDepayload * depayload, + GstBuffer * buf); + +static void +gst_rtp_gst_depay_base_init (gpointer klass) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (klass); + + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&gst_rtp_gst_depay_src_template)); + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&gst_rtp_gst_depay_sink_template)); + + gst_element_class_set_details_simple (element_class, + "GStreamer depayloader", "Codec/Depayloader/Network", + "Extracts GStreamer buffers from RTP packets", + "Wim Taymans "); +} + +static void +gst_rtp_gst_depay_class_init (GstRtpGSTDepayClass * klass) +{ + GObjectClass *gobject_class; + GstBaseRTPDepayloadClass *gstbasertpdepayload_class; + + gobject_class = (GObjectClass *) klass; + gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass; + + gobject_class->finalize = gst_rtp_gst_depay_finalize; + + gstbasertpdepayload_class->set_caps = gst_rtp_gst_depay_setcaps; + gstbasertpdepayload_class->process = gst_rtp_gst_depay_process; + + GST_DEBUG_CATEGORY_INIT (rtpgstdepay_debug, "rtpgstdepay", 0, + "Gstreamer RTP Depayloader"); +} + +static void +gst_rtp_gst_depay_init (GstRtpGSTDepay * rtpgstdepay, + GstRtpGSTDepayClass * klass) +{ + rtpgstdepay->adapter = gst_adapter_new (); +} + +static void +gst_rtp_gst_depay_finalize (GObject * object) +{ + GstRtpGSTDepay *rtpgstdepay; + + rtpgstdepay = GST_RTP_GST_DEPAY (object); + + g_object_unref (rtpgstdepay->adapter); + + G_OBJECT_CLASS (parent_class)->finalize (object); +} + +static gboolean +gst_rtp_gst_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps) +{ + GstRtpGSTDepay *rtpgstdepay; + GstStructure *structure; + GstCaps *outcaps; + gint clock_rate; + gboolean res; + const gchar *capsenc; + gchar *capsstr; + + rtpgstdepay = GST_RTP_GST_DEPAY (depayload); + + structure = gst_caps_get_structure (caps, 0); + + if (!gst_structure_get_int (structure, "clock-rate", &clock_rate)) + clock_rate = 90000; + depayload->clock_rate = clock_rate; + + capsenc = gst_structure_get_string (structure, "caps"); + if (capsenc) { + gsize out_len; + + capsstr = (gchar *) g_base64_decode (capsenc, &out_len); + outcaps = gst_caps_from_string (capsstr); + g_free (capsstr); + + /* we have the SDP caps as output caps */ + rtpgstdepay->current_CV = 0; + } else { + outcaps = gst_caps_new_any (); + } + res = gst_pad_set_caps (depayload->srcpad, outcaps); + gst_caps_unref (outcaps); + + return res; +} + +static GstBuffer * +gst_rtp_gst_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf) +{ + GstRtpGSTDepay *rtpgstdepay; + GstBuffer *subbuf, *outbuf = NULL; + gint payload_len; + guint8 *payload; + guint16 frag_offset; + guint CV; + + rtpgstdepay = GST_RTP_GST_DEPAY (depayload); + + payload_len = gst_rtp_buffer_get_payload_len (buf); + + if (payload_len <= 8) + goto empty_packet; + + if (GST_BUFFER_IS_DISCONT (buf)) { + GST_WARNING_OBJECT (rtpgstdepay, "DISCONT, clear adapter"); + gst_adapter_clear (rtpgstdepay->adapter); + } + + payload = gst_rtp_buffer_get_payload (buf); + + /* strip off header + * + * 0 1 2 3 + * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 + * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + * |C| CV |D|X|Y|Z| MBZ | + * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + * | Frag_offset | + * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + */ + frag_offset = + (payload[4] << 24) | (payload[5] << 16) | (payload[6] << 8) | payload[7]; + + /* subbuffer skipping the 8 header bytes */ + subbuf = gst_rtp_buffer_get_payload_subbuffer (buf, 8, -1); + gst_adapter_push (rtpgstdepay->adapter, subbuf); + + if (gst_rtp_buffer_get_marker (buf)) { + guint avail; + GstCaps *outcaps; + + /* take the buffer */ + avail = gst_adapter_available (rtpgstdepay->adapter); + outbuf = gst_adapter_take_buffer (rtpgstdepay->adapter, avail); + + CV = (payload[0] >> 4) & 0x7; + + if (payload[0] & 0x80) { + guint b, csize, size, offset; + guint8 *data; + GstBuffer *subbuf; + + /* C bit, we have inline caps */ + data = GST_BUFFER_DATA (outbuf); + size = GST_BUFFER_SIZE (outbuf); + + /* start reading the length, we need this to skip to the data later */ + csize = offset = 0; + do { + if (offset >= size) + goto too_small; + b = data[offset++]; + csize = (csize << 7) | (b & 0x7f); + } while (b & 0x80); + + if (size < csize) + goto too_small; + + /* parse and store in cache */ + outcaps = gst_caps_from_string ((gchar *) & data[offset]); + if (rtpgstdepay->CV_cache[CV]) + gst_caps_unref (rtpgstdepay->CV_cache[CV]); + rtpgstdepay->CV_cache[CV] = outcaps; + + /* skip caps */ + offset += csize; + size -= csize; + + GST_DEBUG_OBJECT (rtpgstdepay, + "inline caps %u, length %u, %" GST_PTR_FORMAT, CV, csize, outcaps); + + /* create real data buffer when needed */ + if (size) + subbuf = gst_buffer_create_sub (outbuf, offset, size); + else + subbuf = NULL; + + gst_buffer_unref (outbuf); + outbuf = subbuf; + } + + /* see what caps we need */ + if (CV != rtpgstdepay->current_CV) { + /* we need to switch caps, check if we have the caps */ + if ((outcaps = rtpgstdepay->CV_cache[CV]) == NULL) + goto missing_caps; + + GST_DEBUG_OBJECT (rtpgstdepay, + "need caps switch from %u to %u, %" GST_PTR_FORMAT, + rtpgstdepay->current_CV, CV, outcaps); + + /* and set caps */ + if (gst_pad_set_caps (depayload->srcpad, outcaps)) + rtpgstdepay->current_CV = CV; + } + + if (outbuf) { + if (payload[0] & 0x8) + GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DELTA_UNIT); + if (payload[0] & 0x4) + GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_MEDIA1); + if (payload[0] & 0x2) + GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_MEDIA2); + if (payload[0] & 0x1) + GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_MEDIA3); + } + } + return outbuf; + + /* ERRORS */ +empty_packet: + { + GST_ELEMENT_WARNING (rtpgstdepay, STREAM, DECODE, + ("Empty Payload."), (NULL)); + return NULL; + } +too_small: + { + GST_ELEMENT_WARNING (rtpgstdepay, STREAM, DECODE, + ("Buffer too small."), (NULL)); + if (outbuf) + gst_buffer_unref (outbuf); + return NULL; + } +missing_caps: + { + GST_ELEMENT_WARNING (rtpgstdepay, STREAM, DECODE, + ("Missing caps %u.", CV), (NULL)); + if (outbuf) + gst_buffer_unref (outbuf); + return NULL; + } +} + +gboolean +gst_rtp_gst_depay_plugin_init (GstPlugin * plugin) +{ + return gst_element_register (plugin, "rtpgstdepay", + GST_RANK_MARGINAL, GST_TYPE_RTP_GST_DEPAY); +} diff --git a/gst/rtp/gstrtpgstdepay.h b/gst/rtp/gstrtpgstdepay.h new file mode 100644 index 0000000000..7a50856d32 --- /dev/null +++ b/gst/rtp/gstrtpgstdepay.h @@ -0,0 +1,63 @@ +/* GStreamer + * Copyright (C) <2010> Wim Taymans + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifndef __GST_RTP_GST_DEPAY_H__ +#define __GST_RTP_GST_DEPAY_H__ + +#include +#include +#include + +G_BEGIN_DECLS + +#define GST_TYPE_RTP_GST_DEPAY \ + (gst_rtp_gst_depay_get_type()) +#define GST_RTP_GST_DEPAY(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_GST_DEPAY,GstRtpGSTDepay)) +#define GST_RTP_GST_DEPAY_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_GST_DEPAY,GstRtpGSTDepayClass)) +#define GST_IS_RTP_GST_DEPAY(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_GST_DEPAY)) +#define GST_IS_RTP_GST_DEPAY_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_GST_DEPAY)) + +typedef struct _GstRtpGSTDepay GstRtpGSTDepay; +typedef struct _GstRtpGSTDepayClass GstRtpGSTDepayClass; + +struct _GstRtpGSTDepay +{ + GstBaseRTPDepayload depayload; + + GstAdapter *adapter; + guint current_CV; + GstCaps *CV_cache[8]; +}; + +struct _GstRtpGSTDepayClass +{ + GstBaseRTPDepayloadClass parent_class; +}; + +GType gst_rtp_gst_depay_get_type (void); + +gboolean gst_rtp_gst_depay_plugin_init (GstPlugin * plugin); + +G_END_DECLS + +#endif /* __GST_RTP_GST_DEPAY_H__ */ diff --git a/gst/rtp/gstrtpgstpay.c b/gst/rtp/gstrtpgstpay.c new file mode 100644 index 0000000000..5fa315f321 --- /dev/null +++ b/gst/rtp/gstrtpgstpay.c @@ -0,0 +1,239 @@ +/* GStreamer + * Copyright (C) <2010> Wim Taymans + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifdef HAVE_CONFIG_H +# include "config.h" +#endif + +#include + +#include + +#include "gstrtpgstpay.h" + +/* + * 0 1 2 3 + * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 + * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + * |C| CV |D|X|Y|Z| MBZ | + * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + * | Frag_offset | + * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + * + * C: caps inlined flag + * When C set, first part of payload contains caps definition. Caps definition + * starts with variable length length prefix and then a string of that length. + * the length is encoded in big endian 7 bit chunks, the top 1 bit of a byte + * is the continuation marker and the 7 next bits the data. A continuation + * marker of 1 means that the next byte contains more data. + * + * CV: caps version, 0 = caps from SDP, 1 - 7 inlined caps + * D: delta unit buffer + * X: media 1 flag + * Y: media 2 flag + * Z: media 3 flag + * + * + */ + +static GstStaticPadTemplate gst_rtp_gst_pay_sink_template = +GST_STATIC_PAD_TEMPLATE ("sink", + GST_PAD_SINK, + GST_PAD_ALWAYS, + GST_STATIC_CAPS_ANY); + +static GstStaticPadTemplate gst_rtp_gst_pay_src_template = +GST_STATIC_PAD_TEMPLATE ("src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("application/x-rtp, " + "media = (string) \"application\", " + "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " + "clock-rate = (int) 90000, " "encoding-name = (string) \"X-GST\"") + ); + +static void gst_rtp_gst_pay_finalize (GObject * object); + +static gboolean gst_rtp_gst_pay_setcaps (GstBaseRTPPayload * payload, + GstCaps * caps); +static GstFlowReturn gst_rtp_gst_pay_handle_buffer (GstBaseRTPPayload * payload, + GstBuffer * buffer); + +GST_BOILERPLATE (GstRtpGSTPay, gst_rtp_gst_pay, GstBaseRTPPayload, + GST_TYPE_BASE_RTP_PAYLOAD) + + static void gst_rtp_gst_pay_base_init (gpointer klass) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (klass); + + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&gst_rtp_gst_pay_src_template)); + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&gst_rtp_gst_pay_sink_template)); + + gst_element_class_set_details_simple (element_class, + "RTP GStreamer payloader", "Codec/Payloader/Network", + "Payload GStreamer buffers as RTP packets", + "Wim Taymans "); +} + +static void +gst_rtp_gst_pay_class_init (GstRtpGSTPayClass * klass) +{ + GObjectClass *gobject_class; + GstBaseRTPPayloadClass *gstbasertppayload_class; + + gobject_class = (GObjectClass *) klass; + gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass; + + gobject_class->finalize = gst_rtp_gst_pay_finalize; + + gstbasertppayload_class->set_caps = gst_rtp_gst_pay_setcaps; + gstbasertppayload_class->handle_buffer = gst_rtp_gst_pay_handle_buffer; +} + +static void +gst_rtp_gst_pay_init (GstRtpGSTPay * rtpgstpay, GstRtpGSTPayClass * klass) +{ +} + +static void +gst_rtp_gst_pay_finalize (GObject * object) +{ + GstRtpGSTPay *rtpgstpay; + + rtpgstpay = GST_RTP_GST_PAY (object); + + G_OBJECT_CLASS (parent_class)->finalize (object); +} + +static gboolean +gst_rtp_gst_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps) +{ + gboolean res; + gchar *capsstr, *capsenc; + + capsstr = gst_caps_to_string (caps); + capsenc = g_base64_encode ((guchar *) capsstr, strlen (capsstr)); + g_free (capsstr); + + gst_basertppayload_set_options (payload, "application", TRUE, "X-GST", 90000); + res = + gst_basertppayload_set_outcaps (payload, "caps", G_TYPE_STRING, capsenc, + NULL); + g_free (capsenc); + + return res; +} + +static GstFlowReturn +gst_rtp_gst_pay_handle_buffer (GstBaseRTPPayload * basepayload, + GstBuffer * buffer) +{ + GstRtpGSTPay *rtpgstpay; + guint8 *data; + guint size; + GstBuffer *outbuf; + GstFlowReturn ret; + GstClockTime timestamp; + guint16 frag_offset; + guint flags; + + rtpgstpay = GST_RTP_GST_PAY (basepayload); + + size = GST_BUFFER_SIZE (buffer); + data = GST_BUFFER_DATA (buffer); + timestamp = GST_BUFFER_TIMESTAMP (buffer); + + ret = GST_FLOW_OK; + + /* caps always from SDP for now */ + flags = 0; + if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DELTA_UNIT)) + flags |= (1 << 3); + if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_MEDIA1)) + flags |= (1 << 2); + if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_MEDIA2)) + flags |= (1 << 1); + if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_MEDIA3)) + flags |= (1 << 0); + + /* + * 0 1 2 3 + * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 + * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + * |C| CV |D|X|Y|Z| MBZ | + * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + * | Frag_offset | + * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + */ + frag_offset = 0; + + while (size > 0) { + guint towrite; + guint8 *payload; + guint payload_len; + guint packet_len; + + /* this will be the total lenght of the packet */ + packet_len = gst_rtp_buffer_calc_packet_len (8 + size, 0, 0); + + /* fill one MTU or all available bytes */ + towrite = MIN (packet_len, GST_BASE_RTP_PAYLOAD_MTU (rtpgstpay)); + + /* this is the payload length */ + payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0); + + /* create buffer to hold the payload */ + outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0); + payload = gst_rtp_buffer_get_payload (outbuf); + + payload[0] = flags; + payload[1] = payload[2] = payload[3] = 0; + payload[4] = frag_offset >> 24; + payload[5] = frag_offset >> 16; + payload[6] = frag_offset >> 8; + payload[7] = frag_offset & 0xff; + + payload += 8; + payload_len -= 8; + + memcpy (payload, data, payload_len); + + data += payload_len; + size -= payload_len; + frag_offset += payload_len; + + if (size == 0) + gst_rtp_buffer_set_marker (outbuf, TRUE); + + GST_BUFFER_TIMESTAMP (outbuf) = timestamp; + + ret = gst_basertppayload_push (basepayload, outbuf); + } + + return ret; +} + +gboolean +gst_rtp_gst_pay_plugin_init (GstPlugin * plugin) +{ + return gst_element_register (plugin, "rtpgstpay", + GST_RANK_NONE, GST_TYPE_RTP_GST_PAY); +} diff --git a/gst/rtp/gstrtpgstpay.h b/gst/rtp/gstrtpgstpay.h new file mode 100644 index 0000000000..c7a7c02744 --- /dev/null +++ b/gst/rtp/gstrtpgstpay.h @@ -0,0 +1,58 @@ +/* GStreamer + * Copyright (C) <2010> Wim Taymans + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifndef __GST_RTP_GST_PAY_H__ +#define __GST_RTP_GST_PAY_H__ + +#include +#include + +G_BEGIN_DECLS + +#define GST_TYPE_RTP_GST_PAY \ + (gst_rtp_gst_pay_get_type()) +#define GST_RTP_GST_PAY(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_GST_PAY,GstRtpGSTPay)) +#define GST_RTP_GST_PAY_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_GST_PAY,GstRtpGSTPayClass)) +#define GST_IS_RTP_GST_PAY(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_GST_PAY)) +#define GST_IS_RTP_GST_PAY_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_GST_PAY)) + +typedef struct _GstRtpGSTPay GstRtpGSTPay; +typedef struct _GstRtpGSTPayClass GstRtpGSTPayClass; + +struct _GstRtpGSTPay +{ + GstBaseRTPPayload payload; +}; + +struct _GstRtpGSTPayClass +{ + GstBaseRTPPayloadClass parent_class; +}; + +GType gst_rtp_gst_pay_get_type (void); + +gboolean gst_rtp_gst_pay_plugin_init (GstPlugin * plugin); + +G_END_DECLS + +#endif /* __GST_RTP_GST_PAY_H__ */ From 5fe6046c20a21900db77cf62e2c5dbb3a5a50274 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Sun, 12 Dec 2010 15:10:47 +0100 Subject: [PATCH 075/351] gstdepay: cleanup the cache --- gst/rtp/gstrtpgstdepay.c | 63 ++++++++++++++++++++++++++++++++++++++-- 1 file changed, 60 insertions(+), 3 deletions(-) diff --git a/gst/rtp/gstrtpgstdepay.c b/gst/rtp/gstrtpgstdepay.c index 633971d31b..4e536bafac 100644 --- a/gst/rtp/gstrtpgstdepay.c +++ b/gst/rtp/gstrtpgstdepay.c @@ -50,6 +50,10 @@ GST_BOILERPLATE (GstRtpGSTDepay, gst_rtp_gst_depay, GstBaseRTPDepayload, static void gst_rtp_gst_depay_finalize (GObject * object); +static GstStateChangeReturn gst_rtp_gst_depay_change_state (GstElement * + element, GstStateChange transition); + +static void gst_rtp_gst_depay_reset (GstRtpGSTDepay * rtpgstdepay); static gboolean gst_rtp_gst_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps); static GstBuffer *gst_rtp_gst_depay_process (GstBaseRTPDepayload * depayload, @@ -75,13 +79,17 @@ static void gst_rtp_gst_depay_class_init (GstRtpGSTDepayClass * klass) { GObjectClass *gobject_class; + GstElementClass *gstelement_class; GstBaseRTPDepayloadClass *gstbasertpdepayload_class; gobject_class = (GObjectClass *) klass; + gstelement_class = (GstElementClass *) klass; gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass; gobject_class->finalize = gst_rtp_gst_depay_finalize; + gstelement_class->change_state = gst_rtp_gst_depay_change_state; + gstbasertpdepayload_class->set_caps = gst_rtp_gst_depay_setcaps; gstbasertpdepayload_class->process = gst_rtp_gst_depay_process; @@ -103,11 +111,31 @@ gst_rtp_gst_depay_finalize (GObject * object) rtpgstdepay = GST_RTP_GST_DEPAY (object); + gst_rtp_gst_depay_reset (rtpgstdepay); g_object_unref (rtpgstdepay->adapter); G_OBJECT_CLASS (parent_class)->finalize (object); } +static void +store_cache (GstRtpGSTDepay * rtpgstdepay, guint CV, GstCaps * caps) +{ + if (rtpgstdepay->CV_cache[CV]) + gst_caps_unref (rtpgstdepay->CV_cache[CV]); + rtpgstdepay->CV_cache[CV] = caps; +} + +static void +gst_rtp_gst_depay_reset (GstRtpGSTDepay * rtpgstdepay) +{ + guint i; + + gst_adapter_clear (rtpgstdepay->adapter); + rtpgstdepay->current_CV = 0; + for (i = 0; i < 8; i++) + store_cache (rtpgstdepay, i, NULL); +} + static gboolean gst_rtp_gst_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps) { @@ -137,6 +165,8 @@ gst_rtp_gst_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps) /* we have the SDP caps as output caps */ rtpgstdepay->current_CV = 0; + gst_caps_ref (outcaps); + store_cache (rtpgstdepay, 0, outcaps); } else { outcaps = gst_caps_new_any (); } @@ -220,9 +250,7 @@ gst_rtp_gst_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf) /* parse and store in cache */ outcaps = gst_caps_from_string ((gchar *) & data[offset]); - if (rtpgstdepay->CV_cache[CV]) - gst_caps_unref (rtpgstdepay->CV_cache[CV]); - rtpgstdepay->CV_cache[CV] = outcaps; + store_cache (rtpgstdepay, CV, outcaps); /* skip caps */ offset += csize; @@ -294,6 +322,35 @@ missing_caps: } } +static GstStateChangeReturn +gst_rtp_gst_depay_change_state (GstElement * element, GstStateChange transition) +{ + GstRtpGSTDepay *rtpgstdepay; + GstStateChangeReturn ret; + + rtpgstdepay = GST_RTP_GST_DEPAY (element); + + switch (transition) { + case GST_STATE_CHANGE_READY_TO_PAUSED: + gst_rtp_gst_depay_reset (rtpgstdepay); + break; + default: + break; + } + + ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); + + switch (transition) { + case GST_STATE_CHANGE_PAUSED_TO_READY: + gst_rtp_gst_depay_reset (rtpgstdepay); + break; + default: + break; + } + return ret; +} + + gboolean gst_rtp_gst_depay_plugin_init (GstPlugin * plugin) { From ef0bc7558d2503551f2eb5f70943737d82a21cd1 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Wed, 22 Dec 2010 11:41:59 +0100 Subject: [PATCH 076/351] gstpay: fix klass, add RTP as a use case --- gst/rtp/gstrtpgstpay.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/gst/rtp/gstrtpgstpay.c b/gst/rtp/gstrtpgstpay.c index 5fa315f321..3e1aaad9dc 100644 --- a/gst/rtp/gstrtpgstpay.c +++ b/gst/rtp/gstrtpgstpay.c @@ -38,7 +38,7 @@ * * C: caps inlined flag * When C set, first part of payload contains caps definition. Caps definition - * starts with variable length length prefix and then a string of that length. + * starts with variable-length length prefix and then a string of that length. * the length is encoded in big endian 7 bit chunks, the top 1 bit of a byte * is the continuation marker and the 7 next bits the data. A continuation * marker of 1 means that the next byte contains more data. @@ -88,7 +88,7 @@ GST_BOILERPLATE (GstRtpGSTPay, gst_rtp_gst_pay, GstBaseRTPPayload, gst_static_pad_template_get (&gst_rtp_gst_pay_sink_template)); gst_element_class_set_details_simple (element_class, - "RTP GStreamer payloader", "Codec/Payloader/Network", + "RTP GStreamer payloader", "Codec/Payloader/Network/RTP", "Payload GStreamer buffers as RTP packets", "Wim Taymans "); } From 02cb1dd7b5df3d13ac81f4306cbd782e92e189a4 Mon Sep 17 00:00:00 2001 From: Rob Clark Date: Thu, 23 Dec 2010 16:03:00 -0600 Subject: [PATCH 077/351] matroska-demux: add stream-format and alignment properties for h264 --- gst/matroska/matroska-demux.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/gst/matroska/matroska-demux.c b/gst/matroska/matroska-demux.c index 8332996d56..47f0ed3255 100644 --- a/gst/matroska/matroska-demux.c +++ b/gst/matroska/matroska-demux.c @@ -6444,6 +6444,8 @@ gst_matroska_demux_video_caps (GstMatroskaTrackVideoContext * gst_caps_set_simple (caps, "codec_data", GST_TYPE_BUFFER, priv, NULL); gst_buffer_unref (priv); + gst_caps_set_simple (caps, "stream-format", G_TYPE_STRING, "avc", + "alignment", G_TYPE_STRING, "au", NULL); } *codec_name = g_strdup ("H264"); } else if ((!strcmp (codec_id, GST_MATROSKA_CODEC_ID_VIDEO_REALVIDEO1)) || From 60c1a44917efa32bccc29800df5d9a84c21ccd4b Mon Sep 17 00:00:00 2001 From: Rob Clark Date: Thu, 23 Dec 2010 16:42:29 -0600 Subject: [PATCH 078/351] v4l2: fix typo --- sys/v4l2/gstv4l2bufferpool.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sys/v4l2/gstv4l2bufferpool.c b/sys/v4l2/gstv4l2bufferpool.c index b8c61bac56..cf439095f6 100644 --- a/sys/v4l2/gstv4l2bufferpool.c +++ b/sys/v4l2/gstv4l2bufferpool.c @@ -263,7 +263,7 @@ gst_v4l2_buffer_pool_get_type (void) if (G_UNLIKELY (_gst_v4l2_buffer_pool_type == 0)) { static const GTypeInfo v4l2_buffer_pool_info = { - sizeof (GstBufferClass), + sizeof (GstMiniObjectClass), NULL, NULL, gst_v4l2_buffer_pool_class_init, From b5647685c4688836581cf3bde925051f0a378987 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Mon, 27 Dec 2010 13:11:59 +0100 Subject: [PATCH 079/351] rtpsource: use the right variable Use the right variable for specifying that we sent a receiver report. --- gst/rtpmanager/rtpsource.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/gst/rtpmanager/rtpsource.c b/gst/rtpmanager/rtpsource.c index 3d5b6c1f57..4a3af168a7 100644 --- a/gst/rtpmanager/rtpsource.c +++ b/gst/rtpmanager/rtpsource.c @@ -334,7 +334,7 @@ rtp_source_create_stats (RTPSource * src) if (!internal) { /* get the last RB we sent */ gst_structure_set (s, - "sent-rb", G_TYPE_BOOLEAN, have_rb, + "sent-rb", G_TYPE_BOOLEAN, src->last_rr.is_valid, "sent-rb-fractionlost", G_TYPE_UINT, (guint) src->last_rr.fractionlost, "sent-rb-packetslost", G_TYPE_INT, (gint) src->last_rr.packetslost, "sent-rb-exthighestseq", G_TYPE_UINT, From fafd0b7bc35f60d077fc3bc1287454778619d7ca Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Wed, 29 Dec 2010 14:40:05 +0000 Subject: [PATCH 080/351] rtpjpegdepay: fix framerate parsing for locales that use a comma as floating point atof() converts strings according to the current locale, but the framerate string will likely always use a dot as floating point separator, so use g_ascii_strtod() instead (but also canonicalise the string before, so we can handle both formats as input). --- gst/rtp/gstrtpjpegdepay.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/gst/rtp/gstrtpjpegdepay.c b/gst/rtp/gstrtpjpegdepay.c index e5c457f533..5ab88fb7bf 100644 --- a/gst/rtp/gstrtpjpegdepay.c +++ b/gst/rtp/gstrtpjpegdepay.c @@ -460,15 +460,23 @@ gst_rtp_jpeg_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps) if (media_attr) { GValue src = { 0 }; GValue dest = { 0 }; + gchar *s; + + /* canonicalise floating point string so we can handle framerate strings + * in the form "24.930" or "24,930" irrespective of the current locale */ + s = g_strdup (media_attr); + g_strdelimit (s, ",", '.'); /* convert the float to a fraction */ g_value_init (&src, G_TYPE_DOUBLE); - g_value_set_double (&src, atof (media_attr)); + g_value_set_double (&src, g_ascii_strtod (s, NULL)); g_value_init (&dest, GST_TYPE_FRACTION); g_value_transform (&src, &dest); rtpjpegdepay->frate_num = gst_value_get_fraction_numerator (&dest); rtpjpegdepay->frate_denom = gst_value_get_fraction_denominator (&dest); + + g_free (s); } return TRUE; From 96830324a577caae9e2d77e346b98d4a29cc4a76 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Wed, 29 Dec 2010 15:54:46 +0000 Subject: [PATCH 081/351] rtspsrc: serialise/deserialise floats without changing locale Use g_ascii_dtostr() and g_ascii_strtod() to serialise/deserialise floating point numbers, instead of ugly hacks that switch locale before and after calling libc functions (which is not a good idea in a multi-threaded application). --- gst/rtsp/gstrtspsrc.c | 64 +++++++++++++++---------------------------- 1 file changed, 22 insertions(+), 42 deletions(-) diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c index b8e6cc6e6d..7e003d3c87 100644 --- a/gst/rtsp/gstrtspsrc.c +++ b/gst/rtsp/gstrtspsrc.c @@ -260,7 +260,6 @@ static gboolean gst_rtspsrc_stream_push_event (GstRTSPSrc * src, GstRTSPStream * stream, GstEvent * event, gboolean source); static gboolean gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event, gboolean source); -static gchar *gst_rtspsrc_dup_printf (const gchar * format, ...); /* commands we send to out loop to notify it of events */ #define CMD_WAIT 0 @@ -4968,7 +4967,7 @@ gst_rtspsrc_setup_streams (GstRTSPSrc * src) /* if the user wants a non default RTP packet size we add the blocksize * parameter */ if (src->rtp_blocksize > 0) { - hval = gst_rtspsrc_dup_printf ("%d", src->rtp_blocksize); + hval = g_strdup_printf ("%d", src->rtp_blocksize); gst_rtsp_message_add_header (&request, GST_RTSP_HDR_BLOCKSIZE, hval); g_free (hval); } @@ -5744,51 +5743,34 @@ gst_rtspsrc_parse_rtpinfo (GstRTSPSrc * src, gchar * rtpinfo) return TRUE; } -#define USE_POSIX_LOCALE { \ - gchar *__old_locale = g_strdup (setlocale (LC_NUMERIC, NULL)); \ - setlocale (LC_NUMERIC, "POSIX"); - -#define RESTORE_LOCALE \ - setlocale (LC_NUMERIC, __old_locale); \ - g_free (__old_locale);} - -static gchar * -gst_rtspsrc_dup_printf (const gchar * format, ...) +static gdouble +gst_rtspsrc_get_float (const gchar * dstr) { - gchar *result; - va_list varargs; + gchar s[G_ASCII_DTOSTR_BUF_SIZE] = { 0, }; - USE_POSIX_LOCALE va_start (varargs, format); - - result = g_strdup_vprintf (format, varargs); - va_end (varargs); - RESTORE_LOCALE return result; -} - -static gint -gst_rtspsrc_get_float (const char *str, gfloat * val) -{ - gint result; - - USE_POSIX_LOCALE result = sscanf (str, "%f", val); - RESTORE_LOCALE return result; + /* canonicalise floating point string so we can handle float strings + * in the form "24.930" or "24,930" irrespective of the current locale */ + g_strlcpy (s, dstr, sizeof (s)); + g_strdelimit (s, ",", '.'); + return g_ascii_strtod (s, NULL); } static gchar * gen_range_header (GstRTSPSrc * src, GstSegment * segment) { - gchar *res; + gchar val_str[G_ASCII_DTOSTR_BUF_SIZE] = { 0, }; if (src->range && src->range->min.type == GST_RTSP_TIME_NOW) { - res = g_strdup_printf ("npt=now-"); + g_strlcpy (val_str, "now", sizeof (val_str)); } else { - if (segment->last_stop == 0) - res = g_strdup_printf ("npt=0-"); - else - res = gst_rtspsrc_dup_printf ("npt=%f-", + if (segment->last_stop == 0) { + g_strlcpy (val_str, "0", sizeof (val_str)); + } else { + g_ascii_dtostr (val_str, sizeof (val_str), ((gdouble) segment->last_stop) / GST_SECOND); + } } - return res; + return g_strdup_printf ("npt=%s-", val_str); } static void @@ -5813,7 +5795,6 @@ gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment) GstRTSPResult res; GList *walk; gchar *hval; - gfloat fval; gint hval_idx; gchar *control; @@ -5878,12 +5859,13 @@ gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment) } if (segment->rate != 1.0) { - hval = gst_rtspsrc_dup_printf ("%f", segment->rate); + gchar hval[G_ASCII_DTOSTR_BUF_SIZE]; + + g_ascii_dtostr (hval, sizeof (hval), segment->rate); if (src->skip) gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SCALE, hval); else gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, hval); - g_free (hval); } if (gst_rtspsrc_send (src, conn, &request, &response, NULL) < 0) @@ -5926,12 +5908,10 @@ gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment) * and should be put in the NEWSEGMENT rate field. */ if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SPEED, &hval, 0) == GST_RTSP_OK) { - if (gst_rtspsrc_get_float (hval, &fval) > 0) - segment->rate = fval; + segment->rate = gst_rtspsrc_get_float (hval); } else if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SCALE, &hval, 0) == GST_RTSP_OK) { - if (gst_rtspsrc_get_float (hval, &fval) > 0) - segment->rate = fval; + segment->rate = gst_rtspsrc_get_float (hval); } /* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp From 82ee35372bfbe1ba78071502b2f28a25696f6f95 Mon Sep 17 00:00:00 2001 From: Andy Wingo Date: Mon, 15 Nov 2010 15:58:28 +0100 Subject: [PATCH 082/351] add "client" property * ext/pulse/pulsesrc.c (gst_pulsesrc_class_init, gst_pulsesrc_init) (gst_pulsesrc_set_property, gst_pulsesrc_get_property) (gst_pulsesrc_open): Add a "client" property, as in pulsesink. Fixes #634914 --- ext/pulse/pulsesrc.c | 34 ++++++++++++++++++++++++++++++---- ext/pulse/pulsesrc.h | 3 +-- 2 files changed, 31 insertions(+), 6 deletions(-) diff --git a/ext/pulse/pulsesrc.c b/ext/pulse/pulsesrc.c index 0509bfd6e1..06fb2069c2 100644 --- a/ext/pulse/pulsesrc.c +++ b/ext/pulse/pulsesrc.c @@ -61,6 +61,7 @@ enum PROP_SERVER, PROP_DEVICE, PROP_DEVICE_NAME, + PROP_CLIENT, PROP_STREAM_PROPERTIES, PROP_LAST }; @@ -246,6 +247,20 @@ gst_pulsesrc_class_init (GstPulseSrcClass * klass) "Human-readable name of the sound device", DEFAULT_DEVICE_NAME, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); + /** + * GstPulseSink:client + * + * The PulseAudio client name to use. + * + * Since: 0.10.27 + */ + g_object_class_install_property (gobject_class, + PROP_CLIENT, + g_param_spec_string ("client", "Client", + "The PulseAudio client_name_to_use", gst_pulse_client_name (), + G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | + GST_PARAM_MUTABLE_READY)); + /** * GstPulseSrc:stream-properties * @@ -275,6 +290,7 @@ gst_pulsesrc_init (GstPulseSrc * pulsesrc, GstPulseSrcClass * klass) { pulsesrc->server = NULL; pulsesrc->device = NULL; + pulsesrc->client_name = gst_pulse_client_name (); pulsesrc->device_description = NULL; pulsesrc->context = NULL; @@ -340,6 +356,7 @@ gst_pulsesrc_finalize (GObject * object) g_free (pulsesrc->server); g_free (pulsesrc->device); + g_free (pulsesrc->client_name); if (pulsesrc->properties) gst_structure_free (pulsesrc->properties); @@ -463,6 +480,15 @@ gst_pulsesrc_set_property (GObject * object, g_free (pulsesrc->device); pulsesrc->device = g_value_dup_string (value); break; + case PROP_CLIENT: + g_free (pulsesrc->client_name); + if (!g_value_get_string (value)) { + GST_WARNING_OBJECT (pulsesrc, + "Empty PulseAudio client name not allowed. Resetting to default value"); + pulsesrc->client_name = gst_pulse_client_name (); + } else + pulsesrc->client_name = g_value_dup_string (value); + break; case PROP_STREAM_PROPERTIES: if (pulsesrc->properties) gst_structure_free (pulsesrc->properties); @@ -495,6 +521,9 @@ gst_pulsesrc_get_property (GObject * object, case PROP_DEVICE_NAME: g_value_take_string (value, gst_pulsesrc_device_description (pulsesrc)); break; + case PROP_CLIENT: + g_value_set_string (value, pulsesrc->client_name); + break; case PROP_STREAM_PROPERTIES: gst_value_set_structure (value, pulsesrc->properties); break; @@ -599,7 +628,6 @@ static gboolean gst_pulsesrc_open (GstAudioSrc * asrc) { GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc); - gchar *name = gst_pulse_client_name (); pa_threaded_mainloop_lock (pulsesrc->mainloop); @@ -610,7 +638,7 @@ gst_pulsesrc_open (GstAudioSrc * asrc) if (!(pulsesrc->context = pa_context_new (pa_threaded_mainloop_get_api (pulsesrc->mainloop), - name))) { + pulsesrc->client_name))) { GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Failed to create context"), (NULL)); goto unlock_and_fail; @@ -649,7 +677,6 @@ gst_pulsesrc_open (GstAudioSrc * asrc) pa_threaded_mainloop_unlock (pulsesrc->mainloop); - g_free (name); return TRUE; /* ERRORS */ @@ -659,7 +686,6 @@ unlock_and_fail: pa_threaded_mainloop_unlock (pulsesrc->mainloop); - g_free (name); return FALSE; } } diff --git a/ext/pulse/pulsesrc.h b/ext/pulse/pulsesrc.h index fb5006d6f6..6e6322b9c3 100644 --- a/ext/pulse/pulsesrc.h +++ b/ext/pulse/pulsesrc.h @@ -55,7 +55,7 @@ struct _GstPulseSrc { GstAudioSrc src; - gchar *server, *device; + gchar *server, *device, *client_name; pa_threaded_mainloop *mainloop; @@ -68,7 +68,6 @@ struct _GstPulseSrc size_t read_buffer_length; gchar *device_description; - GstPulseMixerCtrl *mixer; GstPulseProbe *probe; From 48a8b53bdde342b3536143a84175a21a12ea1aea Mon Sep 17 00:00:00 2001 From: Rob Clark Date: Sun, 4 Apr 2010 06:22:43 -0500 Subject: [PATCH 083/351] v4l2sink: Improve behavior for shared buffers. When the decoder is using pad_alloc(), v4l2sink would behave badly if the number of buffers ('queue-size' property) was not high enough to account for all the buffers needed by the decoder, and other elements (such as queues) between the decoder and v4l2sink. This patch slightly increases the default number of buffers, and changes v4l2sink to drop frames rather than return an error in case the number of buffers is not high enough. --- sys/v4l2/gstv4l2sink.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sys/v4l2/gstv4l2sink.c b/sys/v4l2/gstv4l2sink.c index 1fde82c292..9bef34fb04 100644 --- a/sys/v4l2/gstv4l2sink.c +++ b/sys/v4l2/gstv4l2sink.c @@ -56,7 +56,7 @@ GST_DEBUG_CATEGORY (v4l2sink_debug); #define GST_CAT_DEFAULT v4l2sink_debug -#define PROP_DEF_QUEUE_SIZE 8 +#define PROP_DEF_QUEUE_SIZE 12 #define DEFAULT_PROP_DEVICE "/dev/video1" enum @@ -670,7 +670,9 @@ gst_v4l2sink_show_frame (GstBaseSink * bsink, GstBuffer * buf) &newbuf); if (GST_FLOW_OK != ret) { - return ret; + GST_DEBUG_OBJECT (v4l2sink, + "dropping frame! Consider increasing 'queue-size' property!"); + return GST_FLOW_OK; } memcpy (GST_BUFFER_DATA (newbuf), From e24cfa72505327a247b2af9db39c98167f62ee7f Mon Sep 17 00:00:00 2001 From: Rob Clark Date: Sun, 4 Apr 2010 06:23:31 -0500 Subject: [PATCH 084/351] v4l2sink: don't render preroll buffers Most v4l2 drivers will get upset when you queue the same buffer twice in a row without first dequeueing it. Rendering of pre-roll buffers can be re-introduced later, but will require tracking the state of the buffer, and avoiding to re-QBUF if the buffer has already been passed to the driver. --- sys/v4l2/gstv4l2sink.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sys/v4l2/gstv4l2sink.c b/sys/v4l2/gstv4l2sink.c index 9bef34fb04..0dcce397ea 100644 --- a/sys/v4l2/gstv4l2sink.c +++ b/sys/v4l2/gstv4l2sink.c @@ -249,7 +249,6 @@ gst_v4l2sink_class_init (GstV4l2SinkClass * klass) basesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_v4l2sink_get_caps); basesink_class->set_caps = GST_DEBUG_FUNCPTR (gst_v4l2sink_set_caps); basesink_class->buffer_alloc = GST_DEBUG_FUNCPTR (gst_v4l2sink_buffer_alloc); - basesink_class->preroll = GST_DEBUG_FUNCPTR (gst_v4l2sink_show_frame); basesink_class->render = GST_DEBUG_FUNCPTR (gst_v4l2sink_show_frame); } From ecfbaf58a502ddfc0c605bba9e30f78d4609ea4d Mon Sep 17 00:00:00 2001 From: Rob Clark Date: Sun, 4 Apr 2010 06:24:41 -0500 Subject: [PATCH 085/351] v4l2: clear flags before reusing buffer from buffer pool note: this really only affects v4l2sink since gst_v4l2_buffer_pool_get() is only called once per buffer in the v4l2src case (in gst_v4l2src_buffer_pool_activate()) --- sys/v4l2/gstv4l2bufferpool.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sys/v4l2/gstv4l2bufferpool.c b/sys/v4l2/gstv4l2bufferpool.c index cf439095f6..a981c68db7 100644 --- a/sys/v4l2/gstv4l2bufferpool.c +++ b/sys/v4l2/gstv4l2bufferpool.c @@ -463,8 +463,10 @@ gst_v4l2_buffer_pool_get (GstV4l2BufferPool * pool) { GstV4l2Buffer *buf = g_async_queue_try_pop (pool->avail_buffers); - if (buf) + if (buf) { GST_BUFFER_SIZE (buf) = buf->vbuffer.length; + GST_BUFFER_FLAG_UNSET (buf, 0xffffffff); + } pool->running = TRUE; From 0cec72b97e3110d0009c35b6513afff92b194453 Mon Sep 17 00:00:00 2001 From: Rob Clark Date: Sun, 4 Apr 2010 06:26:50 -0500 Subject: [PATCH 086/351] v4l2sink: Add support for blocking dequeue. We'd prefer to throttle the decoder if we run out of buffers, to keep a bound on memory usage. Also, for OMAP4 it is a requirement of the decoder to not alternate between memory alloced by the display driver and malloc'd userspace memory. --- sys/v4l2/gstv4l2bufferpool.c | 14 +++++++++++--- sys/v4l2/gstv4l2bufferpool.h | 2 +- sys/v4l2/gstv4l2sink.c | 2 +- sys/v4l2/v4l2src_calls.c | 2 +- 4 files changed, 14 insertions(+), 6 deletions(-) diff --git a/sys/v4l2/gstv4l2bufferpool.c b/sys/v4l2/gstv4l2bufferpool.c index a981c68db7..2d9893cf77 100644 --- a/sys/v4l2/gstv4l2bufferpool.c +++ b/sys/v4l2/gstv4l2bufferpool.c @@ -454,14 +454,22 @@ gst_v4l2_buffer_pool_destroy (GstV4l2BufferPool * pool) /** * gst_v4l2_buffer_pool_get: - * @pool: the pool + * @pool: the "this" object + * @blocking: should this call suspend until there is a buffer available + * in the buffer pool? * * Get an available buffer in the pool */ GstV4l2Buffer * -gst_v4l2_buffer_pool_get (GstV4l2BufferPool * pool) +gst_v4l2_buffer_pool_get (GstV4l2BufferPool * pool, gboolean blocking) { - GstV4l2Buffer *buf = g_async_queue_try_pop (pool->avail_buffers); + GstV4l2Buffer *buf; + + if (blocking) { + buf = g_async_queue_pop (pool->avail_buffers); + } else { + buf = g_async_queue_try_pop (pool->avail_buffers); + } if (buf) { GST_BUFFER_SIZE (buf) = buf->vbuffer.length; diff --git a/sys/v4l2/gstv4l2bufferpool.h b/sys/v4l2/gstv4l2bufferpool.h index 70ab082f7b..caad9ac2fc 100644 --- a/sys/v4l2/gstv4l2bufferpool.h +++ b/sys/v4l2/gstv4l2bufferpool.h @@ -82,7 +82,7 @@ void gst_v4l2_buffer_pool_destroy (GstV4l2BufferPool * pool); GstV4l2BufferPool *gst_v4l2_buffer_pool_new (GstElement *v4l2elem, gint fd, gint num_buffers, GstCaps * caps, gboolean requeuebuf, enum v4l2_buf_type type); -GstV4l2Buffer *gst_v4l2_buffer_pool_get (GstV4l2BufferPool *pool); +GstV4l2Buffer *gst_v4l2_buffer_pool_get (GstV4l2BufferPool *pool, gboolean blocking); gboolean gst_v4l2_buffer_pool_qbuf (GstV4l2BufferPool *pool, GstV4l2Buffer *buf); GstV4l2Buffer *gst_v4l2_buffer_pool_dqbuf (GstV4l2BufferPool *pool); diff --git a/sys/v4l2/gstv4l2sink.c b/sys/v4l2/gstv4l2sink.c index 0dcce397ea..c673a86a4c 100644 --- a/sys/v4l2/gstv4l2sink.c +++ b/sys/v4l2/gstv4l2sink.c @@ -632,7 +632,7 @@ gst_v4l2sink_buffer_alloc (GstBaseSink * bsink, guint64 offset, guint size, } } - v4l2buf = gst_v4l2_buffer_pool_get (v4l2sink->pool); + v4l2buf = gst_v4l2_buffer_pool_get (v4l2sink->pool, TRUE); if (G_LIKELY (v4l2buf)) { GST_DEBUG_OBJECT (v4l2sink, "allocated buffer: %p", v4l2buf); diff --git a/sys/v4l2/v4l2src_calls.c b/sys/v4l2/v4l2src_calls.c index e33fa618a8..ecd54b843f 100644 --- a/sys/v4l2/v4l2src_calls.c +++ b/sys/v4l2/v4l2src_calls.c @@ -68,7 +68,7 @@ gst_v4l2src_buffer_pool_activate (GstV4l2BufferPool * pool, { GstV4l2Buffer *buf; - while ((buf = gst_v4l2_buffer_pool_get (pool)) != NULL) + while ((buf = gst_v4l2_buffer_pool_get (pool, FALSE)) != NULL) if (!gst_v4l2_buffer_pool_qbuf (pool, buf)) goto queue_failed; From 61db502a127fa723d112b189c90e21b0692c4d04 Mon Sep 17 00:00:00 2001 From: Rob Clark Date: Sun, 4 Apr 2010 06:28:51 -0500 Subject: [PATCH 087/351] v4l2sink: add "min-queued-bufs" property --- sys/v4l2/gstv4l2sink.c | 18 +++++++++++++++++- sys/v4l2/gstv4l2sink.h | 1 + 2 files changed, 18 insertions(+), 1 deletion(-) diff --git a/sys/v4l2/gstv4l2sink.c b/sys/v4l2/gstv4l2sink.c index c673a86a4c..46c66ad95c 100644 --- a/sys/v4l2/gstv4l2sink.c +++ b/sys/v4l2/gstv4l2sink.c @@ -57,6 +57,7 @@ GST_DEBUG_CATEGORY (v4l2sink_debug); #define GST_CAT_DEFAULT v4l2sink_debug #define PROP_DEF_QUEUE_SIZE 12 +#define PROP_DEF_MIN_QUEUED_BUFS 1 #define DEFAULT_PROP_DEVICE "/dev/video1" enum @@ -64,6 +65,7 @@ enum PROP_0, V4L2_STD_OBJECT_PROPS, PROP_QUEUE_SIZE, + PROP_MIN_QUEUED_BUFS, PROP_OVERLAY_TOP, PROP_OVERLAY_LEFT, PROP_OVERLAY_WIDTH, @@ -229,6 +231,12 @@ gst_v4l2sink_class_init (GstV4l2SinkClass * klass) "Number of buffers to be enqueud in the driver in streaming mode", GST_V4L2_MIN_BUFFERS, GST_V4L2_MAX_BUFFERS, PROP_DEF_QUEUE_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + g_object_class_install_property (gobject_class, PROP_MIN_QUEUED_BUFS, + g_param_spec_uint ("min-queued-bufs", "Minimum queued bufs", + "Minimum number of queued bufs; v4l2sink won't dqbuf if the driver " + "doesn't have more than this number (which normally you shouldn't change)", + 0, GST_V4L2_MAX_BUFFERS, PROP_DEF_MIN_QUEUED_BUFS, + G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_OVERLAY_TOP, g_param_spec_int ("overlay-top", "Overlay top", "The topmost (y) coordinate of the video overlay; top left corner of screen is 0,0", @@ -268,6 +276,7 @@ gst_v4l2sink_init (GstV4l2Sink * v4l2sink, GstV4l2SinkClass * klass) /* number of buffers requested */ v4l2sink->num_buffers = PROP_DEF_QUEUE_SIZE; + v4l2sink->min_queued_bufs = PROP_DEF_MIN_QUEUED_BUFS; v4l2sink->probed_caps = NULL; v4l2sink->current_caps = NULL; @@ -370,6 +379,9 @@ gst_v4l2sink_set_property (GObject * object, case PROP_QUEUE_SIZE: v4l2sink->num_buffers = g_value_get_uint (value); break; + case PROP_MIN_QUEUED_BUFS: + v4l2sink->min_queued_bufs = g_value_get_uint (value); + break; case PROP_OVERLAY_TOP: v4l2sink->overlay.top = g_value_get_int (value); v4l2sink->overlay_fields_set |= OVERLAY_TOP_SET; @@ -410,6 +422,9 @@ gst_v4l2sink_get_property (GObject * object, case PROP_QUEUE_SIZE: g_value_set_uint (value, v4l2sink->num_buffers); break; + case PROP_MIN_QUEUED_BUFS: + g_value_set_uint (value, v4l2sink->min_queued_bufs); + break; case PROP_OVERLAY_TOP: g_value_set_int (value, v4l2sink->overlay.top); break; @@ -701,7 +716,8 @@ gst_v4l2sink_show_frame (GstBaseSink * bsink, GstBuffer * buf) * just queued, then dequeue one immediately to make it available via * _buffer_alloc(): */ - if (gst_v4l2_buffer_pool_available_buffers (v4l2sink->pool) > 1) { + if (gst_v4l2_buffer_pool_available_buffers (v4l2sink->pool) > + v4l2sink->min_queued_bufs) { GstV4l2Buffer *v4l2buf = gst_v4l2_buffer_pool_dqbuf (v4l2sink->pool); /* note: if we get a buf, we don't want to use it directly (because diff --git a/sys/v4l2/gstv4l2sink.h b/sys/v4l2/gstv4l2sink.h index 71553cbae7..22d2559109 100644 --- a/sys/v4l2/gstv4l2sink.h +++ b/sys/v4l2/gstv4l2sink.h @@ -58,6 +58,7 @@ struct _GstV4l2Sink { GstCaps *current_caps; /* the current negotiated caps */ GstV4l2BufferPool *pool; guint32 num_buffers; + guint32 min_queued_bufs; /* * field to store requested overlay-top/left/width/height props: From 417b899c44c2eb4a05cb11cba70733ab0dfba442 Mon Sep 17 00:00:00 2001 From: Rob Clark Date: Sat, 25 Dec 2010 11:52:36 -0600 Subject: [PATCH 088/351] v4l2sink: fix for PAUSED->READY->PAUSED state transitions When v4l2sink goes to PAUSED->READY it only stops streaming, so the state should be set to STATE_PENDING_STREAMON in case the element transitions back to PLAYING. --- sys/v4l2/gstv4l2sink.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sys/v4l2/gstv4l2sink.c b/sys/v4l2/gstv4l2sink.c index 46c66ad95c..703c304807 100644 --- a/sys/v4l2/gstv4l2sink.c +++ b/sys/v4l2/gstv4l2sink.c @@ -472,7 +472,7 @@ gst_v4l2sink_change_state (GstElement * element, GstStateChange transition) if (!gst_v4l2_object_stop_streaming (v4l2sink->v4l2object)) { return GST_STATE_CHANGE_FAILURE; } - v4l2sink->state = STATE_OFF; + v4l2sink->state = STATE_PENDING_STREAMON; } break; case GST_STATE_CHANGE_READY_TO_NULL: @@ -482,6 +482,7 @@ gst_v4l2sink_change_state (GstElement * element, GstStateChange transition) /* close the device */ if (!gst_v4l2_object_stop (v4l2sink->v4l2object)) return GST_STATE_CHANGE_FAILURE; + v4l2sink->state = STATE_OFF; break; default: break; From 67cd90408f54a61e37e6168b77d104dc3af8e206 Mon Sep 17 00:00:00 2001 From: Rob Clark Date: Sun, 4 Apr 2010 06:37:16 -0500 Subject: [PATCH 089/351] v4l2: re-enable x-overlay support --- sys/v4l2/Makefile.am | 15 +++++++-------- sys/v4l2/gstv4l2object.c | 6 +++--- sys/v4l2/gstv4l2sink.c | 18 ++++++++++++------ sys/v4l2/gstv4l2src.c | 12 ++++++------ sys/v4l2/gstv4l2xoverlay.c | 5 ++++- 5 files changed, 32 insertions(+), 24 deletions(-) diff --git a/sys/v4l2/Makefile.am b/sys/v4l2/Makefile.am index 8d46d438d9..22cfec4dd3 100644 --- a/sys/v4l2/Makefile.am +++ b/sys/v4l2/Makefile.am @@ -1,13 +1,12 @@ plugin_LTLIBRARIES = libgstvideo4linux2.la -# overlay is still not supported in current implementation -# if USE_XVIDEO -#xv_source = gstv4l2xoverlay.c -#xv_libs = $(X_LIBS) $(XVIDEO_LIBS) -#else -#xv_source = -#xv_libs = -#endif +if USE_XVIDEO +xv_source = gstv4l2xoverlay.c +xv_libs = $(X_LIBS) $(XVIDEO_LIBS) +else +xv_source = +xv_libs = +endif libgstvideo4linux2_la_SOURCES = gstv4l2.c \ gstv4l2colorbalance.c \ diff --git a/sys/v4l2/gstv4l2object.c b/sys/v4l2/gstv4l2object.c index 4abdb107a3..aaa88d4d78 100644 --- a/sys/v4l2/gstv4l2object.c +++ b/sys/v4l2/gstv4l2object.c @@ -34,7 +34,7 @@ #include "v4l2_calls.h" #include "gstv4l2tuner.h" -#if 0 /* overlay is still not implemented #ifdef HAVE_XVIDEO */ +#ifdef HAVE_XVIDEO #include "gstv4l2xoverlay.h" #endif #include "gstv4l2colorbalance.h" @@ -695,7 +695,7 @@ gst_v4l2_object_start (GstV4l2Object * v4l2object) else return FALSE; -#if 0 /* overlay is still not implemented #ifdef HAVE_XVIDEO */ +#ifdef HAVE_XVIDEO gst_v4l2_xoverlay_start (v4l2object); #endif @@ -705,7 +705,7 @@ gst_v4l2_object_start (GstV4l2Object * v4l2object) gboolean gst_v4l2_object_stop (GstV4l2Object * v4l2object) { -#if 0 /* overlay is still not implemented #ifdef HAVE_XVIDEO */ +#ifdef HAVE_XVIDEO gst_v4l2_xoverlay_stop (v4l2object); #endif diff --git a/sys/v4l2/gstv4l2sink.c b/sys/v4l2/gstv4l2sink.c index 703c304807..cec4c80f66 100644 --- a/sys/v4l2/gstv4l2sink.c +++ b/sys/v4l2/gstv4l2sink.c @@ -43,7 +43,7 @@ #include "gstv4l2colorbalance.h" -#if 0 /* overlay is still not implemented #ifdef HAVE_XVIDEO */ +#ifdef HAVE_XVIDEO #include "gstv4l2xoverlay.h" #endif #include "gstv4l2vidorient.h" @@ -75,7 +75,7 @@ enum GST_IMPLEMENT_V4L2_PROBE_METHODS (GstV4l2SinkClass, gst_v4l2sink); GST_IMPLEMENT_V4L2_COLOR_BALANCE_METHODS (GstV4l2Sink, gst_v4l2sink); -#if 0 /* overlay is still not implemented #ifdef HAVE_XVIDEO */ +#ifdef HAVE_XVIDEO GST_IMPLEMENT_V4L2_XOVERLAY_METHODS (GstV4l2Sink, gst_v4l2sink); #endif GST_IMPLEMENT_V4L2_VIDORIENT_METHODS (GstV4l2Sink, gst_v4l2sink); @@ -85,7 +85,7 @@ gst_v4l2sink_iface_supported (GstImplementsInterface * iface, GType iface_type) { GstV4l2Object *v4l2object = GST_V4L2SINK (iface)->v4l2object; -#if 0 /* overlay is still not implemented #ifdef HAVE_XVIDEO */ +#ifdef HAVE_XVIDEO g_assert (iface_type == GST_TYPE_X_OVERLAY || iface_type == GST_TYPE_COLOR_BALANCE || iface_type == GST_TYPE_VIDEO_ORIENTATION); @@ -97,7 +97,7 @@ gst_v4l2sink_iface_supported (GstImplementsInterface * iface, GType iface_type) if (v4l2object->video_fd == -1) return FALSE; -#if 0 /* overlay is still not implemented #ifdef HAVE_XVIDEO */ +#ifdef HAVE_XVIDEO if (iface_type == GST_TYPE_X_OVERLAY && !GST_V4L2_IS_OVERLAY (v4l2object)) return FALSE; #endif @@ -122,7 +122,7 @@ gst_v4l2sink_init_interfaces (GType type) NULL, NULL, }; -#if 0 /* overlay is still not implemented #ifdef HAVE_XVIDEO */ +#ifdef HAVE_XVIDEO static const GInterfaceInfo v4l2_xoverlay_info = { (GInterfaceInitFunc) gst_v4l2sink_xoverlay_interface_init, NULL, @@ -147,7 +147,7 @@ gst_v4l2sink_init_interfaces (GType type) g_type_add_interface_static (type, GST_TYPE_IMPLEMENTS_INTERFACE, &v4l2iface_info); -#if 0 /* overlay is still not implemented #ifdef HAVE_XVIDEO */ +#ifdef HAVE_XVIDEO g_type_add_interface_static (type, GST_TYPE_X_OVERLAY, &v4l2_xoverlay_info); #endif g_type_add_interface_static (type, @@ -638,6 +638,12 @@ gst_v4l2sink_buffer_alloc (GstBaseSink * bsink, guint64 offset, guint size, V4L2_BUF_TYPE_VIDEO_OUTPUT))) { return GST_FLOW_ERROR; } +#ifdef HAVE_XVIDEO + if (GST_V4L2_IS_OVERLAY (v4l2sink->v4l2object)) { + gst_x_overlay_prepare_xwindow_id (GST_X_OVERLAY (v4l2sink)); + } +#endif + v4l2sink->state = STATE_PENDING_STREAMON; GST_INFO_OBJECT (v4l2sink, "outputting buffers via mmap()"); diff --git a/sys/v4l2/gstv4l2src.c b/sys/v4l2/gstv4l2src.c index 4724fe9675..3fd26519ee 100644 --- a/sys/v4l2/gstv4l2src.c +++ b/sys/v4l2/gstv4l2src.c @@ -51,7 +51,7 @@ #include "gstv4l2colorbalance.h" #include "gstv4l2tuner.h" -#if 0 /* overlay is still not implemented #ifdef HAVE_XVIDEO */ +#ifdef HAVE_XVIDEO #include "gstv4l2xoverlay.h" #endif #include "gstv4l2vidorient.h" @@ -79,7 +79,7 @@ enum GST_IMPLEMENT_V4L2_PROBE_METHODS (GstV4l2SrcClass, gst_v4l2src); GST_IMPLEMENT_V4L2_COLOR_BALANCE_METHODS (GstV4l2Src, gst_v4l2src); GST_IMPLEMENT_V4L2_TUNER_METHODS (GstV4l2Src, gst_v4l2src); -#if 0 /* overlay is still not implemented #ifdef HAVE_XVIDEO */ +#ifdef HAVE_XVIDEO GST_IMPLEMENT_V4L2_XOVERLAY_METHODS (GstV4l2Src, gst_v4l2src); #endif GST_IMPLEMENT_V4L2_VIDORIENT_METHODS (GstV4l2Src, gst_v4l2src); @@ -92,7 +92,7 @@ gst_v4l2src_iface_supported (GstImplementsInterface * iface, GType iface_type) { GstV4l2Object *v4l2object = GST_V4L2SRC (iface)->v4l2object; -#if 0 /* overlay is still not implemented #ifdef HAVE_XVIDEO */ +#ifdef HAVE_XVIDEO g_assert (iface_type == GST_TYPE_TUNER || iface_type == GST_TYPE_X_OVERLAY || iface_type == GST_TYPE_COLOR_BALANCE || @@ -106,7 +106,7 @@ gst_v4l2src_iface_supported (GstImplementsInterface * iface, GType iface_type) if (v4l2object->video_fd == -1) return FALSE; -#if 0 /* overlay is still not implemented #ifdef HAVE_XVIDEO */ +#ifdef HAVE_XVIDEO if (iface_type == GST_TYPE_X_OVERLAY && !GST_V4L2_IS_OVERLAY (v4l2object)) return FALSE; #endif @@ -142,7 +142,7 @@ gst_v4l2src_init_interfaces (GType type) NULL, NULL, }; -#if 0 /* overlay is still not implemented #ifdef HAVE_XVIDEO */ +#ifdef HAVE_XVIDEO static const GInterfaceInfo v4l2_xoverlay_info = { (GInterfaceInitFunc) gst_v4l2src_xoverlay_interface_init, NULL, @@ -169,7 +169,7 @@ gst_v4l2src_init_interfaces (GType type) g_type_add_interface_static (type, GST_TYPE_IMPLEMENTS_INTERFACE, &v4l2iface_info); g_type_add_interface_static (type, GST_TYPE_TUNER, &v4l2_tuner_info); -#if 0 /* overlay is still not implemented #ifdef HAVE_XVIDEO */ +#ifdef HAVE_XVIDEO g_type_add_interface_static (type, GST_TYPE_X_OVERLAY, &v4l2_xoverlay_info); #endif g_type_add_interface_static (type, diff --git a/sys/v4l2/gstv4l2xoverlay.c b/sys/v4l2/gstv4l2xoverlay.c index 4c13e5bfdd..1b1e63b67d 100644 --- a/sys/v4l2/gstv4l2xoverlay.c +++ b/sys/v4l2/gstv4l2xoverlay.c @@ -37,6 +37,8 @@ #include "gstv4l2object.h" #include "v4l2_calls.h" +#include "gst/gst-i18n-plugin.h" + struct _GstV4l2Xv { Display *dpy; @@ -101,7 +103,8 @@ gst_v4l2_xoverlay_open (GstV4l2Object * v4l2object) } min = s.st_rdev & 0xff; for (i = 0; i < anum; i++) { - if (!strcmp (ai[i].name, "video4linux2")) { + if (!strcmp (ai[i].name, "video4linux2") || + !strcmp (ai[i].name, "video4linux")) { if (first_id == 0) first_id = ai[i].base_id; From 34e448980b4de9afff83c674a4d46178fa999611 Mon Sep 17 00:00:00 2001 From: Rob Clark Date: Sun, 4 Apr 2010 06:39:08 -0500 Subject: [PATCH 090/351] v4l2sink: add properties to control crop --- sys/v4l2/gstv4l2sink.c | 134 +++++++++++++++++++++++++++++++++++------ sys/v4l2/gstv4l2sink.h | 8 +-- 2 files changed, 121 insertions(+), 21 deletions(-) diff --git a/sys/v4l2/gstv4l2sink.c b/sys/v4l2/gstv4l2sink.c index cec4c80f66..a7f07cfb1e 100644 --- a/sys/v4l2/gstv4l2sink.c +++ b/sys/v4l2/gstv4l2sink.c @@ -70,6 +70,10 @@ enum PROP_OVERLAY_LEFT, PROP_OVERLAY_WIDTH, PROP_OVERLAY_HEIGHT, + PROP_CROP_TOP, + PROP_CROP_LEFT, + PROP_CROP_WIDTH, + PROP_CROP_HEIGHT, }; @@ -254,6 +258,23 @@ gst_v4l2sink_class_init (GstV4l2SinkClass * klass) "The height of the video overlay; default is equal to negotiated image height", 0, G_MAXUINT, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + g_object_class_install_property (gobject_class, PROP_CROP_TOP, + g_param_spec_int ("crop-top", "Crop top", + "The topmost (y) coordinate of the video crop; top left corner of image is 0,0", + 0x80000000, 0x7fffffff, 0, G_PARAM_READWRITE)); + g_object_class_install_property (gobject_class, PROP_CROP_LEFT, + g_param_spec_int ("crop-left", "Crop left", + "The leftmost (x) coordinate of the video crop; top left corner of image is 0,0", + 0x80000000, 0x7fffffff, 0, G_PARAM_READWRITE)); + g_object_class_install_property (gobject_class, PROP_CROP_WIDTH, + g_param_spec_uint ("crop-width", "Crop width", + "The width of the video crop; default is equal to negotiated image width", + 0, 0xffffffff, 0, G_PARAM_READWRITE)); + g_object_class_install_property (gobject_class, PROP_CROP_HEIGHT, + g_param_spec_uint ("crop-height", "Crop height", + "The height of the video crop; default is equal to negotiated image height", + 0, 0xffffffff, 0, G_PARAM_READWRITE)); + basesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_v4l2sink_get_caps); basesink_class->set_caps = GST_DEBUG_FUNCPTR (gst_v4l2sink_set_caps); basesink_class->buffer_alloc = GST_DEBUG_FUNCPTR (gst_v4l2sink_buffer_alloc); @@ -282,6 +303,7 @@ gst_v4l2sink_init (GstV4l2Sink * v4l2sink, GstV4l2SinkClass * klass) v4l2sink->current_caps = NULL; v4l2sink->overlay_fields_set = 0; + v4l2sink->crop_fields_set = 0; v4l2sink->state = 0; } @@ -323,15 +345,15 @@ enum }; /* - * flags to indicate which overlay properties the user has set (and therefore - * which ones should override the defaults from the driver) + * flags to indicate which overlay/crop properties the user has set (and + * therefore which ones should override the defaults from the driver) */ enum { - OVERLAY_TOP_SET = 0x01, - OVERLAY_LEFT_SET = 0x02, - OVERLAY_WIDTH_SET = 0x04, - OVERLAY_HEIGHT_SET = 0x08 + RECT_TOP_SET = 0x01, + RECT_LEFT_SET = 0x02, + RECT_WIDTH_SET = 0x04, + RECT_HEIGHT_SET = 0x08 }; static void @@ -350,22 +372,66 @@ gst_v4l2sink_sync_overlay_fields (GstV4l2Sink * v4l2sink) g_return_if_fail (v4l2_ioctl (fd, VIDIOC_G_FMT, &format) >= 0); - if (v4l2sink->overlay_fields_set & OVERLAY_TOP_SET) + GST_DEBUG_OBJECT (v4l2sink, + "setting overlay: overlay_fields_set=0x%02x, top=%d, left=%d, width=%d, height=%d", + v4l2sink->overlay_fields_set, + v4l2sink->overlay.top, v4l2sink->overlay.left, + v4l2sink->overlay.width, v4l2sink->overlay.height); + + if (v4l2sink->overlay_fields_set & RECT_TOP_SET) format.fmt.win.w.top = v4l2sink->overlay.top; - if (v4l2sink->overlay_fields_set & OVERLAY_LEFT_SET) + if (v4l2sink->overlay_fields_set & RECT_LEFT_SET) format.fmt.win.w.left = v4l2sink->overlay.left; - if (v4l2sink->overlay_fields_set & OVERLAY_WIDTH_SET) + if (v4l2sink->overlay_fields_set & RECT_WIDTH_SET) format.fmt.win.w.width = v4l2sink->overlay.width; - if (v4l2sink->overlay_fields_set & OVERLAY_HEIGHT_SET) + if (v4l2sink->overlay_fields_set & RECT_HEIGHT_SET) format.fmt.win.w.height = v4l2sink->overlay.height; g_return_if_fail (v4l2_ioctl (fd, VIDIOC_S_FMT, &format) >= 0); - v4l2sink->overlay_fields_set = 0; + v4l2sink->overlay_fields_set = 0; v4l2sink->overlay = format.fmt.win.w; } } +static void +gst_v4l2sink_sync_crop_fields (GstV4l2Sink * v4l2sink) +{ + if (!v4l2sink->crop_fields_set) + return; + + if (GST_V4L2_IS_OPEN (v4l2sink->v4l2object)) { + + gint fd = v4l2sink->v4l2object->video_fd; + struct v4l2_crop crop; + + memset (&crop, 0x00, sizeof (struct v4l2_crop)); + crop.type = V4L2_BUF_TYPE_VIDEO_OUTPUT; + + g_return_if_fail (v4l2_ioctl (fd, VIDIOC_G_CROP, &crop) >= 0); + + GST_DEBUG_OBJECT (v4l2sink, + "setting crop: crop_fields_set=0x%02x, top=%d, left=%d, width=%d, height=%d", + v4l2sink->crop_fields_set, + v4l2sink->crop.top, v4l2sink->crop.left, + v4l2sink->crop.width, v4l2sink->crop.height); + + if (v4l2sink->crop_fields_set & RECT_TOP_SET) + crop.c.top = v4l2sink->crop.top; + if (v4l2sink->crop_fields_set & RECT_LEFT_SET) + crop.c.left = v4l2sink->crop.left; + if (v4l2sink->crop_fields_set & RECT_WIDTH_SET) + crop.c.width = v4l2sink->crop.width; + if (v4l2sink->crop_fields_set & RECT_HEIGHT_SET) + crop.c.height = v4l2sink->crop.height; + + g_return_if_fail (v4l2_ioctl (fd, VIDIOC_S_CROP, &crop) >= 0); + + v4l2sink->crop_fields_set = 0; + v4l2sink->crop = crop.c; + } +} + static void gst_v4l2sink_set_property (GObject * object, @@ -384,24 +450,44 @@ gst_v4l2sink_set_property (GObject * object, break; case PROP_OVERLAY_TOP: v4l2sink->overlay.top = g_value_get_int (value); - v4l2sink->overlay_fields_set |= OVERLAY_TOP_SET; + v4l2sink->overlay_fields_set |= RECT_TOP_SET; gst_v4l2sink_sync_overlay_fields (v4l2sink); break; case PROP_OVERLAY_LEFT: v4l2sink->overlay.left = g_value_get_int (value); - v4l2sink->overlay_fields_set |= OVERLAY_LEFT_SET; + v4l2sink->overlay_fields_set |= RECT_LEFT_SET; gst_v4l2sink_sync_overlay_fields (v4l2sink); break; case PROP_OVERLAY_WIDTH: v4l2sink->overlay.width = g_value_get_uint (value); - v4l2sink->overlay_fields_set |= OVERLAY_WIDTH_SET; + v4l2sink->overlay_fields_set |= RECT_WIDTH_SET; gst_v4l2sink_sync_overlay_fields (v4l2sink); break; case PROP_OVERLAY_HEIGHT: v4l2sink->overlay.height = g_value_get_uint (value); - v4l2sink->overlay_fields_set |= OVERLAY_HEIGHT_SET; + v4l2sink->overlay_fields_set |= RECT_HEIGHT_SET; gst_v4l2sink_sync_overlay_fields (v4l2sink); break; + case PROP_CROP_TOP: + v4l2sink->crop.top = g_value_get_int (value); + v4l2sink->crop_fields_set |= RECT_TOP_SET; + gst_v4l2sink_sync_crop_fields (v4l2sink); + break; + case PROP_CROP_LEFT: + v4l2sink->crop.left = g_value_get_int (value); + v4l2sink->crop_fields_set |= RECT_LEFT_SET; + gst_v4l2sink_sync_crop_fields (v4l2sink); + break; + case PROP_CROP_WIDTH: + v4l2sink->crop.width = g_value_get_uint (value); + v4l2sink->crop_fields_set |= RECT_WIDTH_SET; + gst_v4l2sink_sync_crop_fields (v4l2sink); + break; + case PROP_CROP_HEIGHT: + v4l2sink->crop.height = g_value_get_uint (value); + v4l2sink->crop_fields_set |= RECT_HEIGHT_SET; + gst_v4l2sink_sync_crop_fields (v4l2sink); + break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; @@ -437,6 +523,18 @@ gst_v4l2sink_get_property (GObject * object, case PROP_OVERLAY_HEIGHT: g_value_set_uint (value, v4l2sink->overlay.height); break; + case PROP_CROP_TOP: + g_value_set_int (value, v4l2sink->crop.top); + break; + case PROP_CROP_LEFT: + g_value_set_int (value, v4l2sink->crop.left); + break; + case PROP_CROP_WIDTH: + g_value_set_uint (value, v4l2sink->crop.width); + break; + case PROP_CROP_HEIGHT: + g_value_set_uint (value, v4l2sink->crop.height); + break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; @@ -605,8 +703,6 @@ gst_v4l2sink_set_caps (GstBaseSink * bsink, GstCaps * caps) return FALSE; } - gst_v4l2sink_sync_overlay_fields (v4l2sink); - v4l2sink->current_caps = gst_caps_ref (caps); return TRUE; @@ -638,6 +734,10 @@ gst_v4l2sink_buffer_alloc (GstBaseSink * bsink, guint64 offset, guint size, V4L2_BUF_TYPE_VIDEO_OUTPUT))) { return GST_FLOW_ERROR; } + + gst_v4l2sink_sync_overlay_fields (v4l2sink); + gst_v4l2sink_sync_crop_fields (v4l2sink); + #ifdef HAVE_XVIDEO if (GST_V4L2_IS_OVERLAY (v4l2sink->v4l2object)) { gst_x_overlay_prepare_xwindow_id (GST_X_OVERLAY (v4l2sink)); diff --git a/sys/v4l2/gstv4l2sink.h b/sys/v4l2/gstv4l2sink.h index 22d2559109..07a32bf3c2 100644 --- a/sys/v4l2/gstv4l2sink.h +++ b/sys/v4l2/gstv4l2sink.h @@ -61,16 +61,16 @@ struct _GstV4l2Sink { guint32 min_queued_bufs; /* - * field to store requested overlay-top/left/width/height props: + * field to store requested overlay and crop top/left/width/height props: * note, could maybe be combined with 'vwin' field in GstV4l2Object? */ - struct v4l2_rect overlay; + struct v4l2_rect overlay, crop; /* - * bitmask to track which 'overlay' fields user has requested by + * bitmask to track which overlay and crop fields user has requested by * setting properties: */ - guint8 overlay_fields_set; + guint8 overlay_fields_set, crop_fields_set; guint8 state; }; From c3721534c9ede49b9482b2065b591a05c495bb30 Mon Sep 17 00:00:00 2001 From: Rob Clark Date: Sun, 4 Apr 2010 06:39:52 -0500 Subject: [PATCH 091/351] v4l2: fix race condition The size of the buffer would be zero'd out in gst_v4l2_buffer_finalize() after the buffer is qbuf'd or pushed onto the queue of available buffers.. leaving a race condition where the thread waiting for the buffer could awake and set back a valid size before the finalizing thread zeros out the length. This would result that the newly allocated buffer has length of zero. --- sys/v4l2/gstv4l2bufferpool.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sys/v4l2/gstv4l2bufferpool.c b/sys/v4l2/gstv4l2bufferpool.c index 2d9893cf77..a8e7e79fe1 100644 --- a/sys/v4l2/gstv4l2bufferpool.c +++ b/sys/v4l2/gstv4l2bufferpool.c @@ -472,8 +472,10 @@ gst_v4l2_buffer_pool_get (GstV4l2BufferPool * pool, gboolean blocking) } if (buf) { + GST_V4L2_BUFFER_POOL_LOCK (pool); GST_BUFFER_SIZE (buf) = buf->vbuffer.length; GST_BUFFER_FLAG_UNSET (buf, 0xffffffff); + GST_V4L2_BUFFER_POOL_UNLOCK (pool); } pool->running = TRUE; @@ -555,8 +557,6 @@ gst_v4l2_buffer_pool_dqbuf (GstV4l2BufferPool * pool) GST_DEBUG_OBJECT (pool->v4l2elem, "num_live_buffers++: %d", pool->num_live_buffers); - GST_V4L2_BUFFER_POOL_UNLOCK (pool); - /* set top/bottom field first if v4l2_buffer has the information */ if (buffer.field == V4L2_FIELD_INTERLACED_TB) GST_BUFFER_FLAG_SET (pool_buffer, GST_VIDEO_BUFFER_TFF); @@ -566,6 +566,8 @@ gst_v4l2_buffer_pool_dqbuf (GstV4l2BufferPool * pool) /* this can change at every frame, esp. with jpeg */ GST_BUFFER_SIZE (pool_buffer) = buffer.bytesused; + GST_V4L2_BUFFER_POOL_UNLOCK (pool); + return pool_buffer; } From ff5494046afeb64d8cf7aaea1c58d07d7e480508 Mon Sep 17 00:00:00 2001 From: Rob Clark Date: Sun, 4 Apr 2010 06:41:28 -0500 Subject: [PATCH 092/351] v4l2sink: special handling for cases gst_buffer_make_metadata_writable() Special case check for sub-buffers: In certain cases, places like GstBaseTransform, which might check that the buffer is writable before copying metadata, timestamp, and such, will find that the buffer has more than one reference to it. In these cases, they will create a sub-buffer with an offset=0 and length equal to the original buffer size. This could happen in two scenarios: (1) a tee in the pipeline, and (2) because the refcnt is incremented in gst_mini_object_free() before the finalize function is called, and decremented after it returns.. but returning this buffer to the buffer pool in the finalize function, could wake up a thread blocked in _buffer_alloc() which could run and get a buffer w/ refcnt==2 before the thread originally unref'ing the buffer returns from finalize function and decrements the refcnt back to 1! This is related to issue #545501 --- sys/v4l2/gstv4l2sink.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) diff --git a/sys/v4l2/gstv4l2sink.c b/sys/v4l2/gstv4l2sink.c index a7f07cfb1e..9679b26db7 100644 --- a/sys/v4l2/gstv4l2sink.c +++ b/sys/v4l2/gstv4l2sink.c @@ -783,6 +783,29 @@ gst_v4l2sink_show_frame (GstBaseSink * bsink, GstBuffer * buf) if (!GST_IS_V4L2_BUFFER (buf)) { GstFlowReturn ret; + /* special case check for sub-buffers: In certain cases, places like + * GstBaseTransform, which might check that the buffer is writable + * before copying metadata, timestamp, and such, will find that the + * buffer has more than one reference to it. In these cases, they + * will create a sub-buffer with an offset=0 and length equal to the + * original buffer size. + * + * This could happen in two scenarios: (1) a tee in the pipeline, and + * (2) because the refcnt is incremented in gst_mini_object_free() + * before the finalize function is called, and decremented after it + * returns.. but returning this buffer to the buffer pool in the + * finalize function, could wake up a thread blocked in _buffer_alloc() + * which could run and get a buffer w/ refcnt==2 before the thread + * originally unref'ing the buffer returns from finalize function and + * decrements the refcnt back to 1! + */ + if (buf->parent && + (GST_BUFFER_DATA (buf) == GST_BUFFER_DATA (buf->parent)) && + (GST_BUFFER_SIZE (buf) == GST_BUFFER_SIZE (buf->parent))) { + GST_DEBUG_OBJECT (v4l2sink, "I have a sub-buffer!"); + return gst_v4l2sink_show_frame (bsink, buf->parent); + } + GST_DEBUG_OBJECT (v4l2sink, "slow-path.. I got a %s so I need to memcpy", g_type_name (G_OBJECT_TYPE (buf))); From bf885474891ce13331a082b46bdc5127ea871b04 Mon Sep 17 00:00:00 2001 From: Rob Clark Date: Thu, 10 Jun 2010 11:15:46 -0500 Subject: [PATCH 093/351] v4l2xoverlay: add support to create window If xoverlay is available, v4l2sink should create a window for the overlay to display in. The window automatically tries to make itself as large as possible. This works well on a small screen, but perhaps should first attempt to use the size of the video that is played (no scaling). --- sys/v4l2/gstv4l2sink.c | 4 +- sys/v4l2/gstv4l2xoverlay.c | 132 +++++++++++++++++++++++++++++++++---- sys/v4l2/gstv4l2xoverlay.h | 2 + 3 files changed, 124 insertions(+), 14 deletions(-) diff --git a/sys/v4l2/gstv4l2sink.c b/sys/v4l2/gstv4l2sink.c index 9679b26db7..3de98512ee 100644 --- a/sys/v4l2/gstv4l2sink.c +++ b/sys/v4l2/gstv4l2sink.c @@ -739,9 +739,7 @@ gst_v4l2sink_buffer_alloc (GstBaseSink * bsink, guint64 offset, guint size, gst_v4l2sink_sync_crop_fields (v4l2sink); #ifdef HAVE_XVIDEO - if (GST_V4L2_IS_OVERLAY (v4l2sink->v4l2object)) { - gst_x_overlay_prepare_xwindow_id (GST_X_OVERLAY (v4l2sink)); - } + gst_v4l2_xoverlay_prepare_xwindow_id (v4l2sink->v4l2object, TRUE); #endif v4l2sink->state = STATE_PENDING_STREAMON; diff --git a/sys/v4l2/gstv4l2xoverlay.c b/sys/v4l2/gstv4l2xoverlay.c index 1b1e63b67d..d11b8a35ae 100644 --- a/sys/v4l2/gstv4l2xoverlay.c +++ b/sys/v4l2/gstv4l2xoverlay.c @@ -42,7 +42,7 @@ struct _GstV4l2Xv { Display *dpy; - gint port, idle_id; + gint port, idle_id, event_id; GMutex *mutex; }; @@ -103,11 +103,15 @@ gst_v4l2_xoverlay_open (GstV4l2Object * v4l2object) } min = s.st_rdev & 0xff; for (i = 0; i < anum; i++) { + GST_DEBUG_OBJECT (v4l2object->element, "found adapter: %s", ai[i].name); if (!strcmp (ai[i].name, "video4linux2") || !strcmp (ai[i].name, "video4linux")) { if (first_id == 0) first_id = ai[i].base_id; + GST_DEBUG_OBJECT (v4l2object->element, + "first_id=%d, base_id=%lu, min=%d", first_id, ai[i].base_id, min); + /* hmm... */ if (first_id != 0 && ai[i].base_id == first_id + min) id = ai[i].base_id; @@ -127,6 +131,7 @@ gst_v4l2_xoverlay_open (GstV4l2Object * v4l2object) v4l2xv->port = id; v4l2xv->mutex = g_mutex_new (); v4l2xv->idle_id = 0; + v4l2xv->event_id = 0; v4l2object->xv = v4l2xv; if (v4l2object->xwindow_id) { @@ -150,6 +155,8 @@ gst_v4l2_xoverlay_close (GstV4l2Object * v4l2object) g_mutex_free (v4l2xv->mutex); if (v4l2xv->idle_id) g_source_remove (v4l2xv->idle_id); + if (v4l2xv->event_id) + g_source_remove (v4l2xv->event_id); g_free (v4l2xv); v4l2object->xv = NULL; } @@ -168,20 +175,29 @@ gst_v4l2_xoverlay_stop (GstV4l2Object * v4l2object) gst_v4l2_xoverlay_close (v4l2object); } +static void +update_geometry (GstV4l2Object * v4l2object) +{ + GstV4l2Xv *v4l2xv = v4l2object->xv; + XWindowAttributes attr; + XGetWindowAttributes (v4l2xv->dpy, v4l2object->xwindow_id, &attr); + XvPutVideo (v4l2xv->dpy, v4l2xv->port, v4l2object->xwindow_id, + DefaultGC (v4l2xv->dpy, DefaultScreen (v4l2xv->dpy)), + 0, 0, attr.width, attr.height, 0, 0, attr.width, attr.height); +} + static gboolean idle_refresh (gpointer data) { GstV4l2Object *v4l2object = GST_V4L2_OBJECT (data); GstV4l2Xv *v4l2xv = v4l2object->xv; - XWindowAttributes attr; + + GST_LOG_OBJECT (v4l2object->element, "idle refresh"); if (v4l2xv) { g_mutex_lock (v4l2xv->mutex); - XGetWindowAttributes (v4l2xv->dpy, v4l2object->xwindow_id, &attr); - XvPutVideo (v4l2xv->dpy, v4l2xv->port, v4l2object->xwindow_id, - DefaultGC (v4l2xv->dpy, DefaultScreen (v4l2xv->dpy)), - 0, 0, attr.width, attr.height, 0, 0, attr.width, attr.height); + update_geometry (v4l2object); v4l2xv->idle_id = 0; g_mutex_unlock (v4l2xv->mutex); @@ -191,11 +207,42 @@ idle_refresh (gpointer data) return FALSE; } + +static gboolean +event_refresh (gpointer data) +{ + GstV4l2Object *v4l2object = GST_V4L2_OBJECT (data); + GstV4l2Xv *v4l2xv = v4l2object->xv; + + GST_LOG_OBJECT (v4l2object->element, "event refresh"); + + if (v4l2xv) { + XEvent e; + + g_mutex_lock (v4l2xv->mutex); + + /* Handle ConfigureNotify */ + while (XCheckWindowEvent (v4l2xv->dpy, v4l2object->xwindow_id, + StructureNotifyMask, &e)) { + switch (e.type) { + case ConfigureNotify: + update_geometry (v4l2object); + break; + default: + break; + } + } + g_mutex_unlock (v4l2xv->mutex); + } + + /* repeat */ + return TRUE; +} + void gst_v4l2_xoverlay_set_window_handle (GstV4l2Object * v4l2object, guintptr id) { GstV4l2Xv *v4l2xv; - XWindowAttributes attr; XID xwindow_id = id; gboolean change = (v4l2object->xwindow_id != xwindow_id); @@ -238,13 +285,76 @@ gst_v4l2_xoverlay_set_window_handle (GstV4l2Object * v4l2object, guintptr id) XvSelectVideoNotify (v4l2xv->dpy, v4l2object->xwindow_id, 1); } - XGetWindowAttributes (v4l2xv->dpy, v4l2object->xwindow_id, &attr); - XvPutVideo (v4l2xv->dpy, v4l2xv->port, v4l2object->xwindow_id, - DefaultGC (v4l2xv->dpy, DefaultScreen (v4l2xv->dpy)), - 0, 0, attr.width, attr.height, 0, 0, attr.width, attr.height); + update_geometry (v4l2object); if (v4l2xv->idle_id) g_source_remove (v4l2xv->idle_id); v4l2xv->idle_id = g_idle_add (idle_refresh, v4l2object); g_mutex_unlock (v4l2xv->mutex); } + +/** + * gst_v4l2_xoverlay_prepare_xwindow_id: + * @param v4l2object + * @param required TRUE if display is required (ie. TRUE for v4l2sink, but + * FALSE for any other element with optional overlay capabilities) + */ +void +gst_v4l2_xoverlay_prepare_xwindow_id (GstV4l2Object * v4l2object, + gboolean required) +{ + if (!GST_V4L2_IS_OVERLAY (v4l2object)) + return; + + gst_x_overlay_prepare_xwindow_id (GST_X_OVERLAY (v4l2object->element)); + + if (required && !v4l2object->xwindow_id) { + GstV4l2Xv *v4l2xv; + Window win; + int width, height; + + if (!v4l2object->xv && GST_V4L2_IS_OPEN (v4l2object)) + gst_v4l2_xoverlay_open (v4l2object); + + v4l2xv = v4l2object->xv; + + /* if xoverlay is not supported, just bail */ + if (!v4l2xv) + return; + + /* xoverlay is supported, but we don't have a window.. so create one */ + GST_DEBUG_OBJECT (v4l2object->element, "creating window"); + + g_mutex_lock (v4l2xv->mutex); + + width = XDisplayWidth (v4l2xv->dpy, DefaultScreen (v4l2xv->dpy)); + height = XDisplayHeight (v4l2xv->dpy, DefaultScreen (v4l2xv->dpy)); + GST_DEBUG_OBJECT (v4l2object->element, "dpy=%p", v4l2xv->dpy); + + win = XCreateSimpleWindow (v4l2xv->dpy, + DefaultRootWindow (v4l2xv->dpy), + 0, 0, width, height, 0, 0, + XBlackPixel (v4l2xv->dpy, DefaultScreen (v4l2xv->dpy))); + + GST_DEBUG_OBJECT (v4l2object->element, "win=%lu", win); + + /* @todo add mouse events for all windows, and button events for self + * created windows, and hook up to navigation interface.. note that at + * least some of the events we want to handle regardless of whether it + * is a self created window or not.. such as mouse/button events in + * order to implement navigation interface? + */ + XSelectInput (v4l2xv->dpy, win, ExposureMask | StructureNotifyMask); + v4l2xv->event_id = g_timeout_add (45, event_refresh, v4l2object); + + XMapRaised (v4l2xv->dpy, win); + + XSync (v4l2xv->dpy, FALSE); + + g_mutex_unlock (v4l2xv->mutex); + + GST_DEBUG_OBJECT (v4l2object->element, "got window"); + + gst_v4l2_xoverlay_set_window_handle (v4l2object, win); + } +} diff --git a/sys/v4l2/gstv4l2xoverlay.h b/sys/v4l2/gstv4l2xoverlay.h index bd77e8a2d1..6af5c2218a 100644 --- a/sys/v4l2/gstv4l2xoverlay.h +++ b/sys/v4l2/gstv4l2xoverlay.h @@ -39,6 +39,8 @@ void gst_v4l2_xoverlay_stop (GstV4l2Object *v4l2object); void gst_v4l2_xoverlay_interface_init (GstXOverlayClass * klass); void gst_v4l2_xoverlay_set_window_handle (GstV4l2Object * v4l2object, guintptr id); +void gst_v4l2_xoverlay_prepare_xwindow_id (GstV4l2Object * v4l2object, + gboolean required); #define GST_IMPLEMENT_V4L2_XOVERLAY_METHODS(Type, interface_as_function) \ From bb07a39c6766bc925ebd15af9de8ea9bee923210 Mon Sep 17 00:00:00 2001 From: Rob Clark Date: Tue, 13 Jul 2010 10:03:51 -0500 Subject: [PATCH 094/351] v4l2: cleanup get/set input/output output devices should use get/set output, and in either case we should not print a warning message if the ioctl fails but the device does not claim to support the tuner interface --- sys/v4l2/gstv4l2sink.c | 2 +- sys/v4l2/v4l2_calls.c | 71 +++++++++++++++++++++++++++++++++++++++--- sys/v4l2/v4l2_calls.h | 2 -- 3 files changed, 68 insertions(+), 7 deletions(-) diff --git a/sys/v4l2/gstv4l2sink.c b/sys/v4l2/gstv4l2sink.c index 3de98512ee..421f639d84 100644 --- a/sys/v4l2/gstv4l2sink.c +++ b/sys/v4l2/gstv4l2sink.c @@ -286,7 +286,7 @@ gst_v4l2sink_init (GstV4l2Sink * v4l2sink, GstV4l2SinkClass * klass) { v4l2sink->v4l2object = gst_v4l2_object_new (GST_ELEMENT (v4l2sink), V4L2_BUF_TYPE_VIDEO_OUTPUT, DEFAULT_PROP_DEVICE, - gst_v4l2_get_input, gst_v4l2_set_input, NULL); + gst_v4l2_get_output, gst_v4l2_set_output, NULL); /* same default value for video output device as is used for * v4l2src/capture is no good.. so lets set a saner default diff --git a/sys/v4l2/v4l2_calls.c b/sys/v4l2/v4l2_calls.c index f2c2c081d1..b4feef3eb7 100644 --- a/sys/v4l2/v4l2_calls.c +++ b/sys/v4l2/v4l2_calls.c @@ -824,11 +824,14 @@ gst_v4l2_get_input (GstV4l2Object * v4l2object, gint * input) /* ERRORS */ input_failed: - { + if (v4l2object->vcap.capabilities & V4L2_CAP_TUNER) { + /* only give a warning message if driver actually claims to have tuner + * support + */ GST_ELEMENT_WARNING (v4l2object->element, RESOURCE, SETTINGS, (_("Failed to get current input on device '%s'. May be it is a radio device"), v4l2object->videodev), GST_ERROR_SYSTEM); - return FALSE; } + return FALSE; } gboolean @@ -846,10 +849,70 @@ gst_v4l2_set_input (GstV4l2Object * v4l2object, gint input) /* ERRORS */ input_failed: - { + if (v4l2object->vcap.capabilities & V4L2_CAP_TUNER) { + /* only give a warning message if driver actually claims to have tuner + * support + */ GST_ELEMENT_WARNING (v4l2object->element, RESOURCE, SETTINGS, (_("Failed to set input %d on device %s."), input, v4l2object->videodev), GST_ERROR_SYSTEM); - return FALSE; } + return FALSE; +} + +gboolean +gst_v4l2_get_output (GstV4l2Object * v4l2object, gint * output) +{ + gint n; + + GST_DEBUG_OBJECT (v4l2object->element, "trying to get output"); + + if (!GST_V4L2_IS_OPEN (v4l2object)) + return FALSE; + + if (v4l2_ioctl (v4l2object->video_fd, VIDIOC_G_OUTPUT, &n) < 0) + goto output_failed; + + *output = n; + + GST_DEBUG_OBJECT (v4l2object->element, "output: %d", n); + + return TRUE; + + /* ERRORS */ +output_failed: + if (v4l2object->vcap.capabilities & V4L2_CAP_TUNER) { + /* only give a warning message if driver actually claims to have tuner + * support + */ + GST_ELEMENT_WARNING (v4l2object->element, RESOURCE, SETTINGS, + (_("Failed to get current output on device '%s'. May be it is a radio device"), v4l2object->videodev), GST_ERROR_SYSTEM); + } + return FALSE; +} + +gboolean +gst_v4l2_set_output (GstV4l2Object * v4l2object, gint output) +{ + GST_DEBUG_OBJECT (v4l2object->element, "trying to set output to %d", output); + + if (!GST_V4L2_IS_OPEN (v4l2object)) + return FALSE; + + if (v4l2_ioctl (v4l2object->video_fd, VIDIOC_S_OUTPUT, &output) < 0) + goto output_failed; + + return TRUE; + + /* ERRORS */ +output_failed: + if (v4l2object->vcap.capabilities & V4L2_CAP_TUNER) { + /* only give a warning message if driver actually claims to have tuner + * support + */ + GST_ELEMENT_WARNING (v4l2object->element, RESOURCE, SETTINGS, + (_("Failed to set output %d on device %s."), + output, v4l2object->videodev), GST_ERROR_SYSTEM); + } + return FALSE; } diff --git a/sys/v4l2/v4l2_calls.h b/sys/v4l2/v4l2_calls.h index c5c1808a4e..a47872dd43 100644 --- a/sys/v4l2/v4l2_calls.h +++ b/sys/v4l2/v4l2_calls.h @@ -111,12 +111,10 @@ gboolean gst_v4l2_get_input (GstV4l2Object * v4l2object, gint * input); gboolean gst_v4l2_set_input (GstV4l2Object * v4l2object, gint input); -#if 0 /* output not handled by now */ gboolean gst_v4l2_get_output (GstV4l2Object *v4l2object, gint *output); gboolean gst_v4l2_set_output (GstV4l2Object *v4l2object, gint output); -#endif /* #if 0 - output not handled by now */ /* frequency control */ gboolean gst_v4l2_get_frequency (GstV4l2Object *v4l2object, From 9e1d419d07337e6db2cc3936472be205ce927e54 Mon Sep 17 00:00:00 2001 From: Rob Clark Date: Sun, 4 Apr 2010 06:43:41 -0500 Subject: [PATCH 095/351] v4l2: add norm property Based on a patch by Guennadi Liakhovetski. --- sys/v4l2/gstv4l2object.c | 81 +++++++++++++++++++++++++++++++++++----- sys/v4l2/gstv4l2object.h | 5 ++- sys/v4l2/gstv4l2sink.c | 16 +++++--- sys/v4l2/gstv4l2tuner.c | 23 +++++++++++- sys/v4l2/gstv4l2tuner.h | 5 +++ sys/v4l2/v4l2_calls.c | 3 ++ 6 files changed, 115 insertions(+), 18 deletions(-) diff --git a/sys/v4l2/gstv4l2object.c b/sys/v4l2/gstv4l2object.c index aaa88d4d78..e1cce2b877 100644 --- a/sys/v4l2/gstv4l2object.c +++ b/sys/v4l2/gstv4l2object.c @@ -49,7 +49,7 @@ GST_DEBUG_CATEGORY_EXTERN (v4l2_debug); #define DEFAULT_PROP_DEVICE_NAME NULL #define DEFAULT_PROP_DEVICE_FD -1 #define DEFAULT_PROP_FLAGS 0 -#define DEFAULT_PROP_NORM NULL +#define DEFAULT_PROP_TV_NORM 0 #define DEFAULT_PROP_CHANNEL NULL #define DEFAULT_PROP_FREQUENCY 0 @@ -309,6 +309,54 @@ gst_v4l2_device_get_type (void) return v4l2_device_type; } +#define GST_TYPE_V4L2_TV_NORM (gst_v4l2_tv_norm_get_type ()) +static GType +gst_v4l2_tv_norm_get_type (void) +{ + static GType v4l2_tv_norm = 0; + + if (!v4l2_tv_norm) { + static const GEnumValue tv_norms[] = { + {0, "none", "none"}, + {V4L2_STD_NTSC, "NTSC", "NTSC"}, + {V4L2_STD_NTSC_M, "NTSC-M", "NTSC-M"}, + {V4L2_STD_NTSC_M_JP, "NTSC-M-JP", "NTSC-M-JP"}, + {V4L2_STD_NTSC_M_KR, "NTSC-M-KR", "NTSC-M-KR"}, + {V4L2_STD_NTSC_443, "NTSC-443", "NTSC-443"}, + {V4L2_STD_PAL, "PAL", "PAL"}, + {V4L2_STD_PAL_BG, "PAL-BG", "PAL-BG"}, + {V4L2_STD_PAL_B, "PAL-B", "PAL-B"}, + {V4L2_STD_PAL_B1, "PAL-B1", "PAL-B1"}, + {V4L2_STD_PAL_G, "PAL-G", "PAL-G"}, + {V4L2_STD_PAL_H, "PAL-H", "PAL-H"}, + {V4L2_STD_PAL_I, "PAL-I", "PAL-I"}, + {V4L2_STD_PAL_DK, "PAL-DK", "PAL-DK"}, + {V4L2_STD_PAL_D, "PAL-D", "PAL-D"}, + {V4L2_STD_PAL_D1, "PAL-D1", "PAL-D1"}, + {V4L2_STD_PAL_K, "PAL-K", "PAL-K"}, + {V4L2_STD_PAL_M, "PAL-M", "PAL-M"}, + {V4L2_STD_PAL_N, "PAL-N", "PAL-N"}, + {V4L2_STD_PAL_Nc, "PAL-Nc", "PAL-Nc"}, + {V4L2_STD_PAL_60, "PAL-60", "PAL-60"}, + {V4L2_STD_SECAM, "SECAM", "SECAM"}, + {V4L2_STD_SECAM_B, "SECAM-B", "SECAM-B"}, + {V4L2_STD_SECAM_G, "SECAM-G", "SECAM-G"}, + {V4L2_STD_SECAM_H, "SECAM-H", "SECAM-H"}, + {V4L2_STD_SECAM_DK, "SECAM-DK", "SECAM-DK"}, + {V4L2_STD_SECAM_D, "SECAM-D", "SECAM-D"}, + {V4L2_STD_SECAM_K, "SECAM-K", "SECAM-K"}, + {V4L2_STD_SECAM_K1, "SECAM-K1", "SECAM-K1"}, + {V4L2_STD_SECAM_L, "SECAM-L", "SECAM-L"}, + {V4L2_STD_SECAM_LC, "SECAM-Lc", "SECAM-Lc"}, + {0, NULL, NULL} + }; + + v4l2_tv_norm = g_enum_register_static ("V4L2_TV_norms", tv_norms); + } + + return v4l2_tv_norm; +} + void gst_v4l2_object_install_properties_helper (GObjectClass * gobject_class, const char *default_device) @@ -377,6 +425,12 @@ gst_v4l2_object_install_properties_helper (GObjectClass * gobject_class, "Hue or color balance", G_MININT, G_MAXINT, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | GST_PARAM_CONTROLLABLE)); + + g_object_class_install_property (gobject_class, PROP_TV_NORM, + g_param_spec_enum ("norm", "TV norm", + "video standard", + GST_TYPE_V4L2_TV_NORM, DEFAULT_PROP_TV_NORM, + G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); } GstV4l2Object * @@ -433,9 +487,6 @@ gst_v4l2_object_destroy (GstV4l2Object * v4l2object) if (v4l2object->channel) g_free (v4l2object->channel); - if (v4l2object->norm) - g_free (v4l2object->norm); - if (v4l2object->formats) { gst_v4l2_object_clear_format_list (v4l2object); } @@ -520,6 +571,9 @@ gst_v4l2_object_set_property_helper (GstV4l2Object * v4l2object, v4l2object->norm = g_value_dup_string (value); } break; + case PROP_TV_NORM: + v4l2object->tv_norm = g_value_get_enum (value); + break; case PROP_CHANNEL: if (GST_V4L2_IS_OPEN (v4l2object)) { GstTuner *tuner = GST_TUNER (v4l2object->element); @@ -624,6 +678,9 @@ gst_v4l2_object_get_property_helper (GstV4l2Object * v4l2object, return TRUE; } break; + case PROP_TV_NORM: + g_value_set_enum (value, v4l2object->tv_norm); + break; default: return FALSE; break; @@ -643,16 +700,18 @@ gst_v4l2_set_defaults (GstV4l2Object * v4l2object) tuner = GST_TUNER (v4l2object->element); - if (v4l2object->norm) - norm = gst_tuner_find_norm_by_name (tuner, v4l2object->norm); + if (v4l2object->tv_norm) + norm = gst_v4l2_tuner_get_norm_by_std_id (v4l2object, v4l2object->tv_norm); + GST_DEBUG_OBJECT (v4l2object->element, "tv_norm=%llx, norm=%p", + v4l2object->tv_norm, norm); if (norm) { gst_tuner_set_norm (tuner, norm); } else { norm = GST_TUNER_NORM (gst_tuner_get_norm (GST_TUNER (v4l2object->element))); if (norm) { - g_free (v4l2object->norm); - v4l2object->norm = g_strdup (norm->label); + v4l2object->tv_norm = + gst_v4l2_tuner_get_std_id_by_norm (v4l2object, norm); gst_tuner_norm_changed (tuner, norm); } } @@ -1859,13 +1918,15 @@ default_frame_sizes: } /* Since we can't get framerate directly, try to use the current norm */ - if (v4l2object->norm && v4l2object->norms) { + if (v4l2object->tv_norm && v4l2object->norms) { GList *norms; GstTunerNorm *norm = NULL; + GstTunerNorm *current = + gst_v4l2_tuner_get_norm_by_std_id (v4l2object, v4l2object->tv_norm); for (norms = v4l2object->norms; norms != NULL; norms = norms->next) { norm = (GstTunerNorm *) norms->data; - if (!strcmp (norm->label, v4l2object->norm)) + if (!strcmp (norm->label, current->label)) break; } /* If it's possible, set framerate to that (discrete) value */ diff --git a/sys/v4l2/gstv4l2object.h b/sys/v4l2/gstv4l2object.h index 1b38672103..b21c27d766 100644 --- a/sys/v4l2/gstv4l2object.h +++ b/sys/v4l2/gstv4l2object.h @@ -108,7 +108,7 @@ struct _GstV4l2Object { GList *channels; /* properties */ - gchar *norm; + v4l2_std_id tv_norm; gchar *channel; gulong frequency; @@ -137,7 +137,8 @@ GType gst_v4l2_object_get_type (void); PROP_BRIGHTNESS, \ PROP_CONTRAST, \ PROP_SATURATION, \ - PROP_HUE + PROP_HUE, \ + PROP_TV_NORM /* create/destroy */ GstV4l2Object * gst_v4l2_object_new (GstElement * element, diff --git a/sys/v4l2/gstv4l2sink.c b/sys/v4l2/gstv4l2sink.c index 421f639d84..9fce498fd4 100644 --- a/sys/v4l2/gstv4l2sink.c +++ b/sys/v4l2/gstv4l2sink.c @@ -43,6 +43,7 @@ #include "gstv4l2colorbalance.h" +#include "gstv4l2tuner.h" #ifdef HAVE_XVIDEO #include "gstv4l2xoverlay.h" #endif @@ -79,6 +80,7 @@ enum GST_IMPLEMENT_V4L2_PROBE_METHODS (GstV4l2SinkClass, gst_v4l2sink); GST_IMPLEMENT_V4L2_COLOR_BALANCE_METHODS (GstV4l2Sink, gst_v4l2sink); +GST_IMPLEMENT_V4L2_TUNER_METHODS (GstV4l2Sink, gst_v4l2sink); #ifdef HAVE_XVIDEO GST_IMPLEMENT_V4L2_XOVERLAY_METHODS (GstV4l2Sink, gst_v4l2sink); #endif @@ -89,14 +91,12 @@ gst_v4l2sink_iface_supported (GstImplementsInterface * iface, GType iface_type) { GstV4l2Object *v4l2object = GST_V4L2SINK (iface)->v4l2object; + g_assert (iface_type == GST_TYPE_TUNER || #ifdef HAVE_XVIDEO - g_assert (iface_type == GST_TYPE_X_OVERLAY || + iface_type == GST_TYPE_X_OVERLAY || +#endif iface_type == GST_TYPE_COLOR_BALANCE || iface_type == GST_TYPE_VIDEO_ORIENTATION); -#else - g_assert (iface_type == GST_TYPE_COLOR_BALANCE || - iface_type == GST_TYPE_VIDEO_ORIENTATION); -#endif if (v4l2object->video_fd == -1) return FALSE; @@ -126,6 +126,11 @@ gst_v4l2sink_init_interfaces (GType type) NULL, NULL, }; + static const GInterfaceInfo v4l2_tuner_info = { + (GInterfaceInitFunc) gst_v4l2sink_tuner_interface_init, + NULL, + NULL, + }; #ifdef HAVE_XVIDEO static const GInterfaceInfo v4l2_xoverlay_info = { (GInterfaceInitFunc) gst_v4l2sink_xoverlay_interface_init, @@ -151,6 +156,7 @@ gst_v4l2sink_init_interfaces (GType type) g_type_add_interface_static (type, GST_TYPE_IMPLEMENTS_INTERFACE, &v4l2iface_info); + g_type_add_interface_static (type, GST_TYPE_TUNER, &v4l2_tuner_info); #ifdef HAVE_XVIDEO g_type_add_interface_static (type, GST_TYPE_X_OVERLAY, &v4l2_xoverlay_info); #endif diff --git a/sys/v4l2/gstv4l2tuner.c b/sys/v4l2/gstv4l2tuner.c index c5f84426bd..a805396cac 100644 --- a/sys/v4l2/gstv4l2tuner.c +++ b/sys/v4l2/gstv4l2tuner.c @@ -237,7 +237,6 @@ gst_v4l2_tuner_set_norm (GstV4l2Object * v4l2object, GstTunerNorm * norm) GstTunerNorm * gst_v4l2_tuner_get_norm (GstV4l2Object * v4l2object) { - GList *item; v4l2_std_id norm; /* assert that we're opened and that we're using a known item */ @@ -245,6 +244,14 @@ gst_v4l2_tuner_get_norm (GstV4l2Object * v4l2object) gst_v4l2_get_norm (v4l2object, &norm); + return gst_v4l2_tuner_get_norm_by_std_id (v4l2object, norm); +} + +GstTunerNorm * +gst_v4l2_tuner_get_norm_by_std_id (GstV4l2Object * v4l2object, v4l2_std_id norm) +{ + GList *item; + for (item = v4l2object->norms; item != NULL; item = item->next) { if (norm & GST_V4L2_TUNER_NORM (item->data)->index) return (GstTunerNorm *) item->data; @@ -253,6 +260,20 @@ gst_v4l2_tuner_get_norm (GstV4l2Object * v4l2object) return NULL; } +v4l2_std_id +gst_v4l2_tuner_get_std_id_by_norm (GstV4l2Object * v4l2object, + GstTunerNorm * norm) +{ + GList *item; + + for (item = v4l2object->norms; item != NULL; item = item->next) { + if (norm == GST_TUNER_NORM (item->data)) + return GST_V4L2_TUNER_NORM (item->data)->index; + } + + return 0; +} + void gst_v4l2_tuner_set_frequency_and_notify (GstV4l2Object * v4l2object, GstTunerChannel * channel, gulong frequency) diff --git a/sys/v4l2/gstv4l2tuner.h b/sys/v4l2/gstv4l2tuner.h index bf9ff94da5..699ca87e10 100644 --- a/sys/v4l2/gstv4l2tuner.h +++ b/sys/v4l2/gstv4l2tuner.h @@ -94,6 +94,11 @@ void gst_v4l2_tuner_set_norm_and_notify (GstV4l2Object * v4l2o GstTunerNorm* gst_v4l2_tuner_get_norm (GstV4l2Object * v4l2object); gboolean gst_v4l2_tuner_set_norm (GstV4l2Object * v4l2object, GstTunerNorm * norm); +GstTunerNorm* gst_v4l2_tuner_get_norm_by_std_id (GstV4l2Object * v4l2object, + v4l2_std_id norm); +v4l2_std_id gst_v4l2_tuner_get_std_id_by_norm (GstV4l2Object * v4l2object, + GstTunerNorm * norm); + /* frequency */ void gst_v4l2_tuner_set_frequency_and_notify (GstV4l2Object * v4l2object, GstTunerChannel * channel, diff --git a/sys/v4l2/v4l2_calls.c b/sys/v4l2/v4l2_calls.c index b4feef3eb7..74b2e0adf2 100644 --- a/sys/v4l2/v4l2_calls.c +++ b/sys/v4l2/v4l2_calls.c @@ -226,6 +226,9 @@ gst_v4l2_fill_lists (GstV4l2Object * v4l2object) standard.frameperiod.denominator, standard.frameperiod.numerator); v4l2norm->index = standard.id; + GST_DEBUG_OBJECT (v4l2object->element, "index=%08x, label=%s", + (unsigned int) v4l2norm->index, norm->label); + v4l2object->norms = g_list_prepend (v4l2object->norms, (gpointer) norm); } v4l2object->norms = g_list_reverse (v4l2object->norms); From 538d3a959b3f358477077b2c733c9027991c32aa Mon Sep 17 00:00:00 2001 From: Rob Clark Date: Sat, 2 Oct 2010 14:45:14 -0500 Subject: [PATCH 096/351] v4l2sink: add navigation support --- sys/v4l2/gstv4l2sink.c | 90 +++++++++++++++++++++- sys/v4l2/gstv4l2sink.h | 2 + sys/v4l2/gstv4l2xoverlay.c | 149 ++++++++++++++++++++++++++++++++++--- sys/v4l2/gstv4l2xoverlay.h | 4 + 4 files changed, 231 insertions(+), 14 deletions(-) diff --git a/sys/v4l2/gstv4l2sink.c b/sys/v4l2/gstv4l2sink.c index 9fce498fd4..8652a9fde5 100644 --- a/sys/v4l2/gstv4l2sink.c +++ b/sys/v4l2/gstv4l2sink.c @@ -33,6 +33,18 @@ * |[ * gst-launch videotestsrc ! v4l2sink device=/dev/video1 * ]| This pipeline displays a test pattern on /dev/video1 + * |[ + * gst-launch -v videotestsrc ! navigationtest ! v4l2sink + * ]| A pipeline to test navigation events. + * While moving the mouse pointer over the test signal you will see a black box + * following the mouse pointer. If you press the mouse button somewhere on the + * video and release it somewhere else a green box will appear where you pressed + * the button and a red one where you released it. (The navigationtest element + * is part of gst-plugins-good.) You can observe here that even if the images + * are scaled through hardware the pointer coordinates are converted back to the + * original video frame geometry so that the box can be drawn to the correct + * position. This also handles borders correctly, limiting coordinates to the + * image area * */ @@ -94,6 +106,7 @@ gst_v4l2sink_iface_supported (GstImplementsInterface * iface, GType iface_type) g_assert (iface_type == GST_TYPE_TUNER || #ifdef HAVE_XVIDEO iface_type == GST_TYPE_X_OVERLAY || + iface_type == GST_TYPE_NAVIGATION || #endif iface_type == GST_TYPE_COLOR_BALANCE || iface_type == GST_TYPE_VIDEO_ORIENTATION); @@ -102,8 +115,10 @@ gst_v4l2sink_iface_supported (GstImplementsInterface * iface, GType iface_type) return FALSE; #ifdef HAVE_XVIDEO - if (iface_type == GST_TYPE_X_OVERLAY && !GST_V4L2_IS_OVERLAY (v4l2object)) - return FALSE; + if (!GST_V4L2_IS_OVERLAY (v4l2object)) { + if (iface_type == GST_TYPE_X_OVERLAY || iface_type == GST_TYPE_NAVIGATION) + return FALSE; + } #endif return TRUE; @@ -118,6 +133,16 @@ gst_v4l2sink_interface_init (GstImplementsInterfaceClass * klass) klass->supported = gst_v4l2sink_iface_supported; } +#ifdef HAVE_XVIDEO +static void gst_v4l2sink_navigation_send_event (GstNavigation * navigation, + GstStructure * structure); +static void +gst_v4l2sink_navigation_init (GstNavigationInterface * iface) +{ + iface->send_event = gst_v4l2sink_navigation_send_event; +} +#endif + static void gst_v4l2sink_init_interfaces (GType type) { @@ -137,6 +162,11 @@ gst_v4l2sink_init_interfaces (GType type) NULL, NULL, }; + static const GInterfaceInfo v4l2_navigation_info = { + (GInterfaceInitFunc) gst_v4l2sink_navigation_init, + NULL, + NULL, + }; #endif static const GInterfaceInfo v4l2_colorbalance_info = { (GInterfaceInitFunc) gst_v4l2sink_color_balance_interface_init, @@ -159,6 +189,8 @@ gst_v4l2sink_init_interfaces (GType type) g_type_add_interface_static (type, GST_TYPE_TUNER, &v4l2_tuner_info); #ifdef HAVE_XVIDEO g_type_add_interface_static (type, GST_TYPE_X_OVERLAY, &v4l2_xoverlay_info); + g_type_add_interface_static (type, + GST_TYPE_NAVIGATION, &v4l2_navigation_info); #endif g_type_add_interface_static (type, GST_TYPE_COLOR_BALANCE, &v4l2_colorbalance_info); @@ -195,7 +227,6 @@ static GstFlowReturn gst_v4l2sink_buffer_alloc (GstBaseSink * bsink, static GstFlowReturn gst_v4l2sink_show_frame (GstBaseSink * bsink, GstBuffer * buf); - static void gst_v4l2sink_base_init (gpointer g_class) { @@ -709,6 +740,15 @@ gst_v4l2sink_set_caps (GstBaseSink * bsink, GstCaps * caps) return FALSE; } + v4l2sink->video_width = w; + v4l2sink->video_height = h; + + /* TODO: videosink width/height should be scaled according to + * pixel-aspect-ratio + */ + GST_VIDEO_SINK_WIDTH (v4l2sink) = w; + GST_VIDEO_SINK_HEIGHT (v4l2sink) = h; + v4l2sink->current_caps = gst_caps_ref (caps); return TRUE; @@ -866,3 +906,47 @@ gst_v4l2sink_show_frame (GstBaseSink * bsink, GstBuffer * buf) return GST_FLOW_OK; } + +#ifdef HAVE_XVIDEO +static void +gst_v4l2sink_navigation_send_event (GstNavigation * navigation, + GstStructure * structure) +{ + GstV4l2Sink *v4l2sink = GST_V4L2SINK (navigation); + GstV4l2Xv *xv = v4l2sink->v4l2object->xv; + GstPad *peer; + + if (!xv) + return; + + if ((peer = gst_pad_get_peer (GST_VIDEO_SINK_PAD (v4l2sink)))) { + GstVideoRectangle rect; + gdouble x, y, xscale = 1.0, yscale = 1.0; + + gst_v4l2_xoverlay_get_render_rect (v4l2sink->v4l2object, &rect); + + /* We calculate scaling using the original video frames geometry to + * include pixel aspect ratio scaling. + */ + xscale = (gdouble) v4l2sink->video_width / rect.w; + yscale = (gdouble) v4l2sink->video_height / rect.h; + + /* Converting pointer coordinates to the non scaled geometry */ + if (gst_structure_get_double (structure, "pointer_x", &x)) { + x = MIN (x, rect.x + rect.w); + x = MAX (x - rect.x, 0); + gst_structure_set (structure, "pointer_x", G_TYPE_DOUBLE, + (gdouble) x * xscale, NULL); + } + if (gst_structure_get_double (structure, "pointer_y", &y)) { + y = MIN (y, rect.y + rect.h); + y = MAX (y - rect.y, 0); + gst_structure_set (structure, "pointer_y", G_TYPE_DOUBLE, + (gdouble) y * yscale, NULL); + } + + gst_pad_send_event (peer, gst_event_new_navigation (structure)); + gst_object_unref (peer); + } +} +#endif diff --git a/sys/v4l2/gstv4l2sink.h b/sys/v4l2/gstv4l2sink.h index 07a32bf3c2..8fe82221ae 100644 --- a/sys/v4l2/gstv4l2sink.h +++ b/sys/v4l2/gstv4l2sink.h @@ -60,6 +60,8 @@ struct _GstV4l2Sink { guint32 num_buffers; guint32 min_queued_bufs; + gint video_width, video_height; /* original (unscaled) video w/h */ + /* * field to store requested overlay and crop top/left/width/height props: * note, could maybe be combined with 'vwin' field in GstV4l2Object? diff --git a/sys/v4l2/gstv4l2xoverlay.c b/sys/v4l2/gstv4l2xoverlay.c index d11b8a35ae..2ad5631249 100644 --- a/sys/v4l2/gstv4l2xoverlay.c +++ b/sys/v4l2/gstv4l2xoverlay.c @@ -33,6 +33,8 @@ #include #include +#include + #include "gstv4l2xoverlay.h" #include "gstv4l2object.h" #include "v4l2_calls.h" @@ -43,7 +45,7 @@ struct _GstV4l2Xv { Display *dpy; gint port, idle_id, event_id; - GMutex *mutex; + GMutex *mutex; /* to serialize calls to X11 */ }; GST_DEBUG_CATEGORY_STATIC (v4l2xv_debug); @@ -175,15 +177,53 @@ gst_v4l2_xoverlay_stop (GstV4l2Object * v4l2object) gst_v4l2_xoverlay_close (v4l2object); } +/* should be called with mutex held */ +static gboolean +get_render_rect (GstV4l2Object * v4l2object, GstVideoRectangle *rect) +{ + GstV4l2Xv *v4l2xv = v4l2object->xv; + if (v4l2xv && v4l2xv->dpy && v4l2object->xwindow_id) { + XWindowAttributes attr; + XGetWindowAttributes (v4l2xv->dpy, v4l2object->xwindow_id, &attr); + /* this is where we'd add support to maintain aspect ratio */ + rect->x = 0; + rect->y = 0; + rect->w = attr.width; + rect->h = attr.height; + return TRUE; + } else { + return FALSE; + } +} + +gboolean +gst_v4l2_xoverlay_get_render_rect (GstV4l2Object *v4l2object, + GstVideoRectangle *rect) +{ + GstV4l2Xv *v4l2xv = v4l2object->xv; + gboolean ret = FALSE; + if (v4l2xv) { + g_mutex_lock (v4l2xv->mutex); + ret = get_render_rect (v4l2object, rect); + g_mutex_unlock (v4l2xv->mutex); + } + return ret; +} + static void update_geometry (GstV4l2Object * v4l2object) { GstV4l2Xv *v4l2xv = v4l2object->xv; - XWindowAttributes attr; - XGetWindowAttributes (v4l2xv->dpy, v4l2object->xwindow_id, &attr); + GstVideoRectangle rect; + if (!get_render_rect (v4l2object, &rect)) + return; + /* note: we don't pass in valid video x/y/w/h.. currently the xserver + * doesn't need to know these, as they come from v4l2 by setting the + * crop.. + */ XvPutVideo (v4l2xv->dpy, v4l2xv->port, v4l2object->xwindow_id, DefaultGC (v4l2xv->dpy, DefaultScreen (v4l2xv->dpy)), - 0, 0, attr.width, attr.height, 0, 0, attr.width, attr.height); + 0, 0, rect.w, rect.h, rect.x, rect.y, rect.w, rect.h); } static gboolean @@ -221,6 +261,92 @@ event_refresh (gpointer data) g_mutex_lock (v4l2xv->mutex); + /* If the element supports navigation, collect the relavent input + * events and push them upstream as navigation events + */ + if (GST_IS_NAVIGATION (v4l2object->element)) { + guint pointer_x = 0, pointer_y = 0; + gboolean pointer_moved = FALSE; + + /* We get all pointer motion events, only the last position is + * interesting. + */ + while (XCheckWindowEvent (v4l2xv->dpy, v4l2object->xwindow_id, + PointerMotionMask, &e)) { + switch (e.type) { + case MotionNotify: + pointer_x = e.xmotion.x; + pointer_y = e.xmotion.y; + pointer_moved = TRUE; + break; + default: + break; + } + } + if (pointer_moved) { + GST_DEBUG_OBJECT (v4l2object->element, + "pointer moved over window at %d,%d", pointer_x, pointer_y); + g_mutex_unlock (v4l2xv->mutex); + gst_navigation_send_mouse_event (GST_NAVIGATION (v4l2object->element), + "mouse-move", 0, e.xbutton.x, e.xbutton.y); + g_mutex_lock (v4l2xv->mutex); + } + + /* We get all events on our window to throw them upstream + */ + while (XCheckWindowEvent (v4l2xv->dpy, v4l2object->xwindow_id, + KeyPressMask | KeyReleaseMask | + ButtonPressMask | ButtonReleaseMask, &e)) { + KeySym keysym; + const char *key_str = NULL; + + g_mutex_unlock (v4l2xv->mutex); + + switch (e.type) { + case ButtonPress: + GST_DEBUG_OBJECT (v4l2object->element, + "button %d pressed over window at %d,%d", + e.xbutton.button, e.xbutton.x, e.xbutton.y); + gst_navigation_send_mouse_event ( + GST_NAVIGATION (v4l2object->element), + "mouse-button-press", e.xbutton.button, + e.xbutton.x, e.xbutton.y); + break; + case ButtonRelease: + GST_DEBUG_OBJECT (v4l2object->element, + "button %d released over window at %d,%d", + e.xbutton.button, e.xbutton.x, e.xbutton.y); + gst_navigation_send_mouse_event ( + GST_NAVIGATION (v4l2object->element), + "mouse-button-release", e.xbutton.button, + e.xbutton.x, e.xbutton.y); + break; + case KeyPress: + case KeyRelease: + g_mutex_lock (v4l2xv->mutex); + keysym = XKeycodeToKeysym (v4l2xv->dpy, e.xkey.keycode, 0); + if (keysym != NoSymbol) { + key_str = XKeysymToString (keysym); + } else { + key_str = "unknown"; + } + g_mutex_unlock (v4l2xv->mutex); + GST_DEBUG_OBJECT (v4l2object->element, + "key %d pressed over window at %d,%d (%s)", + e.xkey.keycode, e.xkey.x, e.xkey.y, key_str); + gst_navigation_send_key_event ( + GST_NAVIGATION (v4l2object->element), + e.type == KeyPress ? "key-press" : "key-release", key_str); + break; + default: + GST_DEBUG_OBJECT (v4l2object->element, + "unhandled X event (%d)", e.type); + } + + g_mutex_lock (v4l2xv->mutex); + } + } + /* Handle ConfigureNotify */ while (XCheckWindowEvent (v4l2xv->dpy, v4l2object->xwindow_id, StructureNotifyMask, &e)) { @@ -312,6 +438,7 @@ gst_v4l2_xoverlay_prepare_xwindow_id (GstV4l2Object * v4l2object, GstV4l2Xv *v4l2xv; Window win; int width, height; + long event_mask; if (!v4l2object->xv && GST_V4L2_IS_OPEN (v4l2object)) gst_v4l2_xoverlay_open (v4l2object); @@ -338,13 +465,13 @@ gst_v4l2_xoverlay_prepare_xwindow_id (GstV4l2Object * v4l2object, GST_DEBUG_OBJECT (v4l2object->element, "win=%lu", win); - /* @todo add mouse events for all windows, and button events for self - * created windows, and hook up to navigation interface.. note that at - * least some of the events we want to handle regardless of whether it - * is a self created window or not.. such as mouse/button events in - * order to implement navigation interface? - */ - XSelectInput (v4l2xv->dpy, win, ExposureMask | StructureNotifyMask); + event_mask = ExposureMask | StructureNotifyMask; + if (GST_IS_NAVIGATION (v4l2object->element)) { + event_mask |= PointerMotionMask | + KeyPressMask | KeyReleaseMask | + ButtonPressMask | ButtonReleaseMask; + } + XSelectInput (v4l2xv->dpy, win, event_mask); v4l2xv->event_id = g_timeout_add (45, event_refresh, v4l2object); XMapRaised (v4l2xv->dpy, win); diff --git a/sys/v4l2/gstv4l2xoverlay.h b/sys/v4l2/gstv4l2xoverlay.h index 6af5c2218a..1a09306836 100644 --- a/sys/v4l2/gstv4l2xoverlay.h +++ b/sys/v4l2/gstv4l2xoverlay.h @@ -28,6 +28,8 @@ #include #include +#include +#include /* for GstVideoRectange */ #include "gstv4l2object.h" @@ -35,6 +37,8 @@ G_BEGIN_DECLS void gst_v4l2_xoverlay_start (GstV4l2Object *v4l2object); void gst_v4l2_xoverlay_stop (GstV4l2Object *v4l2object); +gboolean gst_v4l2_xoverlay_get_render_rect (GstV4l2Object *v4l2object, + GstVideoRectangle *rect); void gst_v4l2_xoverlay_interface_init (GstXOverlayClass * klass); void gst_v4l2_xoverlay_set_window_handle (GstV4l2Object * v4l2object, From 6c8268dbfd5c88fac28c882ef2e4598a6522e2d6 Mon Sep 17 00:00:00 2001 From: Rob Clark Date: Sat, 4 Dec 2010 13:43:11 -0600 Subject: [PATCH 097/351] v4l2: add interlaced support --- sys/v4l2/gstv4l2object.c | 50 +++++++++++++++++++++------------------- sys/v4l2/gstv4l2object.h | 4 ++-- sys/v4l2/gstv4l2sink.c | 7 +++--- sys/v4l2/gstv4l2src.c | 7 +++--- sys/v4l2/v4l2src_calls.c | 5 ++-- sys/v4l2/v4l2src_calls.h | 3 ++- 6 files changed, 41 insertions(+), 35 deletions(-) diff --git a/sys/v4l2/gstv4l2object.c b/sys/v4l2/gstv4l2object.c index e1cce2b877..f4512187c9 100644 --- a/sys/v4l2/gstv4l2object.c +++ b/sys/v4l2/gstv4l2object.c @@ -1361,8 +1361,8 @@ gst_v4l2_object_get_all_caps (void) */ gboolean gst_v4l2_object_get_caps_info (GstV4l2Object * v4l2object, GstCaps * caps, - struct v4l2_fmtdesc ** format, gint * w, gint * h, guint * fps_n, - guint * fps_d, guint * size) + struct v4l2_fmtdesc ** format, gint * w, gint * h, + gboolean * interlaced, guint * fps_n, guint * fps_d, guint * size) { GstStructure *structure; const GValue *framerate; @@ -1391,6 +1391,9 @@ gst_v4l2_object_get_caps_info (GstV4l2Object * v4l2object, GstCaps * caps, if (!gst_structure_get_int (structure, "height", h)) return FALSE; + if (!gst_structure_get_boolean (structure, "interlaced", interlaced)) + *interlaced = FALSE; + framerate = gst_structure_get_value (structure, "framerate"); if (!framerate) return FALSE; @@ -1987,15 +1990,15 @@ gst_v4l2_object_get_nearest_size (GstV4l2Object * v4l2object, fmt.fmt.pix.width = *width; fmt.fmt.pix.height = *height; fmt.fmt.pix.pixelformat = pixelformat; - fmt.fmt.pix.field = V4L2_FIELD_INTERLACED; + fmt.fmt.pix.field = V4L2_FIELD_NONE; r = v4l2_ioctl (fd, VIDIOC_TRY_FMT, &fmt); if (r < 0 && errno == EINVAL) { - /* try again with progressive video */ + /* try again with interlaced video */ fmt.fmt.pix.width = *width; fmt.fmt.pix.height = *height; fmt.fmt.pix.pixelformat = pixelformat; - fmt.fmt.pix.field = V4L2_FIELD_NONE; + fmt.fmt.pix.field = V4L2_FIELD_INTERLACED; r = v4l2_ioctl (fd, VIDIOC_TRY_FMT, &fmt); } @@ -2037,6 +2040,7 @@ gst_v4l2_object_get_nearest_size (GstV4l2Object * v4l2object, *height = fmt.fmt.pix.height; switch (fmt.fmt.pix.field) { + case V4L2_FIELD_ANY: case V4L2_FIELD_NONE: *interlaced = FALSE; break; @@ -2058,10 +2062,22 @@ gst_v4l2_object_get_nearest_size (GstV4l2Object * v4l2object, gboolean gst_v4l2_object_set_format (GstV4l2Object * v4l2object, guint32 pixelformat, - guint32 width, guint32 height) + guint32 width, guint32 height, gboolean interlaced) { gint fd = v4l2object->video_fd; struct v4l2_format format; + enum v4l2_field field; + + if (interlaced) { + GST_DEBUG_OBJECT (v4l2object->element, "interlaced video"); + /* ideally we would differentiate between types of interlaced video + * but there is not sufficient information in the caps.. + */ + field = V4L2_FIELD_SEQ_TB; + } else { + GST_DEBUG_OBJECT (v4l2object->element, "progressive video"); + field = V4L2_FIELD_NONE; + } GST_DEBUG_OBJECT (v4l2object->element, "Setting format to %dx%d, format " "%" GST_FOURCC_FORMAT, width, height, GST_FOURCC_ARGS (pixelformat)); @@ -2081,7 +2097,8 @@ gst_v4l2_object_set_format (GstV4l2Object * v4l2object, guint32 pixelformat, if (format.type == v4l2object->type && format.fmt.pix.width == width && format.fmt.pix.height == height && - format.fmt.pix.pixelformat == pixelformat) { + format.fmt.pix.pixelformat == pixelformat && + format.fmt.pix.field == field) { /* Nothing to do. We want to succeed immediately * here because setting the same format back * can still fail due to EBUSY. By short-circuiting @@ -2098,25 +2115,10 @@ gst_v4l2_object_set_format (GstV4l2Object * v4l2object, guint32 pixelformat, format.fmt.pix.width = width; format.fmt.pix.height = height; format.fmt.pix.pixelformat = pixelformat; - /* FIXME: request whole frames; need to use gstreamer interlace support - * (INTERLACED mode returns frames where the fields have already been - * combined, there are other modes for requesting fields individually) */ - format.fmt.pix.field = V4L2_FIELD_INTERLACED; + format.fmt.pix.field = field; if (v4l2_ioctl (fd, VIDIOC_S_FMT, &format) < 0) { - /* we might also get EBUSY here */ - if (errno != EINVAL) - goto set_fmt_failed; - - GST_DEBUG_OBJECT (v4l2object->element, "trying again..."); - - /* try again with progressive video */ - format.fmt.pix.width = width; - format.fmt.pix.height = height; - format.fmt.pix.pixelformat = pixelformat; - format.fmt.pix.field = V4L2_FIELD_NONE; - if (v4l2_ioctl (fd, VIDIOC_S_FMT, &format) < 0) - goto set_fmt_failed; + goto set_fmt_failed; } if (format.fmt.pix.width != width || format.fmt.pix.height != height) diff --git a/sys/v4l2/gstv4l2object.h b/sys/v4l2/gstv4l2object.h index b21c27d766..d83861a4d7 100644 --- a/sys/v4l2/gstv4l2object.h +++ b/sys/v4l2/gstv4l2object.h @@ -181,7 +181,7 @@ GstCaps* gst_v4l2_object_probe_caps_for_format (GstV4l2Object *v4l2object, gboolean gst_v4l2_object_get_caps_info (GstV4l2Object *v4l2object, GstCaps *caps, struct v4l2_fmtdesc **format, gint *w, gint *h, - guint *fps_n, guint *fps_d, guint *size); + gboolean * interlaced, guint *fps_n, guint *fps_d, guint *size); GSList* gst_v4l2_object_get_format_list (GstV4l2Object *v4l2object); @@ -190,7 +190,7 @@ GstCaps* gst_v4l2_object_get_all_caps (void); GstStructure* gst_v4l2_object_v4l2fourcc_to_structure (guint32 fourcc); -gboolean gst_v4l2_object_set_format (GstV4l2Object *v4l2object, guint32 pixelformat, guint32 width, guint32 height); +gboolean gst_v4l2_object_set_format (GstV4l2Object *v4l2object, guint32 pixelformat, guint32 width, guint32 height, gboolean interlaced); gboolean gst_v4l2_object_start_streaming (GstV4l2Object *v4l2object); gboolean gst_v4l2_object_stop_streaming (GstV4l2Object *v4l2object); diff --git a/sys/v4l2/gstv4l2sink.c b/sys/v4l2/gstv4l2sink.c index 8652a9fde5..6b3da4e1a8 100644 --- a/sys/v4l2/gstv4l2sink.c +++ b/sys/v4l2/gstv4l2sink.c @@ -689,6 +689,7 @@ gst_v4l2sink_set_caps (GstBaseSink * bsink, GstCaps * caps) { GstV4l2Sink *v4l2sink = GST_V4L2SINK (bsink); gint w = 0, h = 0; + gboolean interlaced; struct v4l2_fmtdesc *format; guint fps_n, fps_d; guint size; @@ -724,7 +725,7 @@ gst_v4l2sink_set_caps (GstBaseSink * bsink, GstCaps * caps) /* we want our own v4l2 type of fourcc codes */ if (!gst_v4l2_object_get_caps_info (v4l2sink->v4l2object, caps, - &format, &w, &h, &fps_n, &fps_d, &size)) { + &format, &w, &h, &interlaced, &fps_n, &fps_d, &size)) { GST_DEBUG_OBJECT (v4l2sink, "can't get capture format from caps %p", caps); return FALSE; } @@ -734,8 +735,8 @@ gst_v4l2sink_set_caps (GstBaseSink * bsink, GstCaps * caps) return FALSE; } - if (!gst_v4l2_object_set_format (v4l2sink->v4l2object, format->pixelformat, w, - h)) { + if (!gst_v4l2_object_set_format (v4l2sink->v4l2object, format->pixelformat, + w, h, interlaced)) { /* error already posted */ return FALSE; } diff --git a/sys/v4l2/gstv4l2src.c b/sys/v4l2/gstv4l2src.c index 3fd26519ee..ed3b2ac6ef 100644 --- a/sys/v4l2/gstv4l2src.c +++ b/sys/v4l2/gstv4l2src.c @@ -611,6 +611,7 @@ gst_v4l2src_set_caps (GstBaseSrc * src, GstCaps * caps) { GstV4l2Src *v4l2src; gint w = 0, h = 0; + gboolean interlaced; struct v4l2_fmtdesc *format; guint fps_n, fps_d; guint size; @@ -632,7 +633,7 @@ gst_v4l2src_set_caps (GstBaseSrc * src, GstCaps * caps) /* we want our own v4l2 type of fourcc codes */ if (!gst_v4l2_object_get_caps_info (v4l2src->v4l2object, caps, &format, &w, - &h, &fps_n, &fps_d, &size)) { + &h, &interlaced, &fps_n, &fps_d, &size)) { GST_INFO_OBJECT (v4l2src, "can't get capture format from caps %" GST_PTR_FORMAT, caps); return FALSE; @@ -641,8 +642,8 @@ gst_v4l2src_set_caps (GstBaseSrc * src, GstCaps * caps) GST_DEBUG_OBJECT (v4l2src, "trying to set_capture %dx%d at %d/%d fps, " "format %s", w, h, fps_n, fps_d, format->description); - if (!gst_v4l2src_set_capture (v4l2src, format->pixelformat, w, h, fps_n, - fps_d)) + if (!gst_v4l2src_set_capture (v4l2src, format->pixelformat, w, h, + interlaced, fps_n, fps_d)) /* error already posted */ return FALSE; diff --git a/sys/v4l2/v4l2src_calls.c b/sys/v4l2/v4l2src_calls.c index ecd54b843f..fb6c37456b 100644 --- a/sys/v4l2/v4l2src_calls.c +++ b/sys/v4l2/v4l2src_calls.c @@ -214,7 +214,8 @@ too_many_trials: ******************************************************/ gboolean gst_v4l2src_set_capture (GstV4l2Src * v4l2src, guint32 pixelformat, - guint32 width, guint32 height, guint fps_n, guint fps_d) + guint32 width, guint32 height, gboolean interlaced, + guint fps_n, guint fps_d) { gint fd = v4l2src->v4l2object->video_fd; struct v4l2_streamparm stream; @@ -223,7 +224,7 @@ gst_v4l2src_set_capture (GstV4l2Src * v4l2src, guint32 pixelformat, return TRUE; if (!gst_v4l2_object_set_format (v4l2src->v4l2object, pixelformat, width, - height)) { + height, interlaced)) { /* error already reported */ return FALSE; } diff --git a/sys/v4l2/v4l2src_calls.h b/sys/v4l2/v4l2src_calls.h index 1fc7411fae..709191870e 100644 --- a/sys/v4l2/v4l2src_calls.h +++ b/sys/v4l2/v4l2src_calls.h @@ -31,7 +31,8 @@ gboolean gst_v4l2src_get_capture (GstV4l2Src * v4l2src); gboolean gst_v4l2src_set_capture (GstV4l2Src * v4l2src, guint32 pixelformat, guint32 width, guint32 height, - guint32 fps_n, guint32 fps_d); + gboolean interlaced, + guint32 fps_n, guint32 fps_d); gboolean gst_v4l2src_capture_init (GstV4l2Src * v4l2src, GstCaps *caps); gboolean gst_v4l2src_capture_start (GstV4l2Src * v4l2src); From c8fc90fd37821ef0be2ea2075ac2bd22beefcda0 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Wed, 29 Dec 2010 23:09:04 +0000 Subject: [PATCH 098/351] matroska-demux: assume matroska if no doctype is specified https://bugzilla.gnome.org/show_bug.cgi?id=638019 --- gst/matroska/matroska-demux.c | 28 ++++++++++++++++------------ 1 file changed, 16 insertions(+), 12 deletions(-) diff --git a/gst/matroska/matroska-demux.c b/gst/matroska/matroska-demux.c index 47f0ed3255..96c9a1593f 100644 --- a/gst/matroska/matroska-demux.c +++ b/gst/matroska/matroska-demux.c @@ -3099,26 +3099,30 @@ gst_matroska_demux_parse_header (GstMatroskaDemux * demux, GstEbmlRead * ebml) } exit: - ret = GST_FLOW_ERROR; - if (doctype) { - if (g_str_equal (doctype, GST_MATROSKA_DOCTYPE_MATROSKA) || - g_str_equal (doctype, GST_MATROSKA_DOCTYPE_WEBM)) { - if (version <= 2) { + + if ((doctype != NULL && !strcmp (doctype, GST_MATROSKA_DOCTYPE_MATROSKA)) || + (doctype != NULL && !strcmp (doctype, GST_MATROSKA_DOCTYPE_WEBM)) || + (doctype == NULL)) { + if (version <= 2) { + if (doctype) { GST_INFO_OBJECT (demux, "Input is %s version %d", doctype, version); - ret = GST_FLOW_OK; } else { - GST_ELEMENT_ERROR (demux, STREAM, DEMUX, (NULL), - ("Demuxer version (2) is too old to read %s version %d", - doctype, version)); + GST_WARNING_OBJECT (demux, "Input is EBML without doctype, assuming " + "matroska (version %d)", version); } + ret = GST_FLOW_OK; } else { - GST_ELEMENT_ERROR (demux, STREAM, WRONG_TYPE, (NULL), - ("Input is not a matroska stream (doctype=%s)", doctype)); + GST_ELEMENT_ERROR (demux, STREAM, DEMUX, (NULL), + ("Demuxer version (2) is too old to read %s version %d", + GST_STR_NULL (doctype), version)); + ret = GST_FLOW_ERROR; } g_free (doctype); } else { GST_ELEMENT_ERROR (demux, STREAM, WRONG_TYPE, (NULL), - ("Input is not a matroska stream")); + ("Input is not a matroska stream (doctype=%s)", doctype)); + ret = GST_FLOW_ERROR; + g_free (doctype); } return ret; From 3a6682908724f25f0f604fe0c5b1a8609d2325b4 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Wed, 29 Dec 2010 23:38:18 +0000 Subject: [PATCH 099/351] matroska-demux: increase allowed max. block size for push mode from 10M to 15M It was an arbitrary limit from the start, meant as a basic sanity check, so may just as well increase it a little. Would be good to provide progress reporting while completing the block in any case.. https://bugzilla.gnome.org/show_bug.cgi?id=637060 --- gst/matroska/matroska-demux.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/gst/matroska/matroska-demux.c b/gst/matroska/matroska-demux.c index 96c9a1593f..29ba781ad0 100644 --- a/gst/matroska/matroska-demux.c +++ b/gst/matroska/matroska-demux.c @@ -5436,10 +5436,12 @@ gst_matroska_demux_parse_contents (GstMatroskaDemux * demux, GstEbmlRead * ebml) #define GST_FLOW_OVERFLOW GST_FLOW_CUSTOM_ERROR +#define MAX_BLOCK_SIZE (15 * 1024 * 1024) + static inline GstFlowReturn gst_matroska_demux_check_read_size (GstMatroskaDemux * demux, guint64 bytes) { - if (G_UNLIKELY (bytes > 10 * 1024 * 1024)) { + if (G_UNLIKELY (bytes > MAX_BLOCK_SIZE)) { /* only a few blocks are expected/allowed to be large, * and will be recursed into, whereas others will be read and must fit */ if (demux->streaming) { From 756869421c7d39ca6588307a3cb7c6a9b8adc985 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Thu, 30 Dec 2010 16:22:48 +0100 Subject: [PATCH 100/351] mpvpay: fix flushing and discont Fix flushing and disconts. Clean up in state changes. --- gst/rtp/gstrtpmpvpay.c | 82 ++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 80 insertions(+), 2 deletions(-) diff --git a/gst/rtp/gstrtpmpvpay.c b/gst/rtp/gstrtpmpvpay.c index 7cca09eccb..a96afbf0f1 100644 --- a/gst/rtp/gstrtpmpvpay.c +++ b/gst/rtp/gstrtpmpvpay.c @@ -48,12 +48,17 @@ GST_STATIC_PAD_TEMPLATE ("src", "clock-rate = (int) 90000, " "encoding-name = (string) \"MPV\"") ); +static GstStateChangeReturn gst_rtp_mpv_pay_change_state (GstElement * element, + GstStateChange transition); + +static void gst_rtp_mpv_pay_finalize (GObject * object); + +static GstFlowReturn gst_rtp_mpv_pay_flush (GstRTPMPVPay * rtpmpvpay); static gboolean gst_rtp_mpv_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps); static GstFlowReturn gst_rtp_mpv_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buffer); -static GstFlowReturn gst_rtp_mpv_pay_flush (GstRTPMPVPay * rtpmpvpay); -static void gst_rtp_mpv_pay_finalize (GObject * object); +static gboolean gst_rtp_mpv_pay_handle_event (GstPad * pad, GstEvent * event); GST_BOILERPLATE (GstRTPMPVPay, gst_rtp_mpv_pay, GstBaseRTPPayload, GST_TYPE_BASE_RTP_PAYLOAD); @@ -77,15 +82,20 @@ static void gst_rtp_mpv_pay_class_init (GstRTPMPVPayClass * klass) { GObjectClass *gobject_class; + GstElementClass *gstelement_class; GstBaseRTPPayloadClass *gstbasertppayload_class; gobject_class = (GObjectClass *) klass; + gstelement_class = (GstElementClass *) klass; gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass; gobject_class->finalize = gst_rtp_mpv_pay_finalize; + gstelement_class->change_state = gst_rtp_mpv_pay_change_state; + gstbasertppayload_class->set_caps = gst_rtp_mpv_pay_setcaps; gstbasertppayload_class->handle_buffer = gst_rtp_mpv_pay_handle_buffer; + gstbasertppayload_class->handle_event = gst_rtp_mpv_pay_handle_event; GST_DEBUG_CATEGORY_INIT (rtpmpvpay_debug, "rtpmpvpay", 0, "MPEG2 ES Video RTP Payloader"); @@ -113,6 +123,15 @@ gst_rtp_mpv_pay_finalize (GObject * object) G_OBJECT_CLASS (parent_class)->finalize (object); } +static void +gst_rtp_mpv_pay_reset (GstRTPMPVPay * pay) +{ + pay->first_ts = -1; + pay->duration = 0; + gst_adapter_clear (pay->adapter); + GST_DEBUG_OBJECT (pay, "reset depayloader"); +} + static gboolean gst_rtp_mpv_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps) { @@ -120,6 +139,31 @@ gst_rtp_mpv_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps) return gst_basertppayload_set_outcaps (payload, NULL); } +static gboolean +gst_rtp_mpv_pay_handle_event (GstPad * pad, GstEvent * event) +{ + GstRTPMPVPay *rtpmpvpay; + + rtpmpvpay = GST_RTP_MPV_PAY (gst_pad_get_parent (pad)); + + switch (GST_EVENT_TYPE (event)) { + case GST_EVENT_EOS: + /* make sure we push the last packets in the adapter on EOS */ + gst_rtp_mpv_pay_flush (rtpmpvpay); + break; + case GST_EVENT_FLUSH_STOP: + gst_rtp_mpv_pay_reset (rtpmpvpay); + break; + default: + break; + } + + gst_object_unref (rtpmpvpay); + + /* FALSE to let the parent handle the event as well */ + return FALSE; +} + static GstFlowReturn gst_rtp_mpv_pay_flush (GstRTPMPVPay * rtpmpvpay) { @@ -191,6 +235,11 @@ gst_rtp_mpv_pay_handle_buffer (GstBaseRTPPayload * basepayload, timestamp = GST_BUFFER_TIMESTAMP (buffer); duration = GST_BUFFER_DURATION (buffer); + if (GST_BUFFER_IS_DISCONT (buffer)) { + GST_DEBUG_OBJECT (rtpmpvpay, "DISCONT"); + gst_rtp_mpv_pay_reset (rtpmpvpay); + } + avail = gst_adapter_available (rtpmpvpay->adapter); if (duration == -1) @@ -224,6 +273,35 @@ gst_rtp_mpv_pay_handle_buffer (GstBaseRTPPayload * basepayload, return ret; } +static GstStateChangeReturn +gst_rtp_mpv_pay_change_state (GstElement * element, GstStateChange transition) +{ + GstRTPMPVPay *rtpmpvpay; + GstStateChangeReturn ret; + + rtpmpvpay = GST_RTP_MPV_PAY (element); + + switch (transition) { + case GST_STATE_CHANGE_READY_TO_PAUSED: + gst_rtp_mpv_pay_reset (rtpmpvpay); + break; + default: + break; + } + + ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); + + switch (transition) { + case GST_STATE_CHANGE_PAUSED_TO_READY: + gst_rtp_mpv_pay_reset (rtpmpvpay); + break; + default: + break; + } + return ret; +} + + gboolean gst_rtp_mpv_pay_plugin_init (GstPlugin * plugin) { From 9c2393490f07bfb094d6983d9a54a972eb975e67 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Thu, 30 Dec 2010 16:24:16 +0100 Subject: [PATCH 101/351] mp4adepay: fix timestamps on buffers --- gst/rtp/gstrtpmp4adepay.c | 19 +++++++++---------- 1 file changed, 9 insertions(+), 10 deletions(-) diff --git a/gst/rtp/gstrtpmp4adepay.c b/gst/rtp/gstrtpmp4adepay.c index 3579d900b8..836ff9f095 100644 --- a/gst/rtp/gstrtpmp4adepay.c +++ b/gst/rtp/gstrtpmp4adepay.c @@ -263,8 +263,11 @@ gst_rtp_mp4a_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf) guint i; guint8 *data; guint pos; + GstClockTime timestamp; + guint offset; avail = gst_adapter_available (rtpmp4adepay->adapter); + timestamp = gst_adapter_prev_timestamp (rtpmp4adepay->adapter, NULL); GST_LOG_OBJECT (rtpmp4adepay, "have marker and %u available", avail); @@ -272,18 +275,16 @@ gst_rtp_mp4a_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf) data = GST_BUFFER_DATA (outbuf); /* position in data we are at */ pos = 0; + /* timestamp offset */ + offset = 0; /* looping through the number of sub-frames in the audio payload */ for (i = 0; i <= rtpmp4adepay->numSubFrames; i++) { /* determine payload length and set buffer data pointer accordingly */ guint skip; guint data_len; - guint32 timestamp; - GstBuffer *tmp = NULL; - timestamp = gst_rtp_buffer_get_timestamp (buf); - /* each subframe starts with a variable length encoding */ data_len = 0; for (skip = 0; skip < avail; skip++) { @@ -314,13 +315,11 @@ gst_rtp_mp4a_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf) data += skip; avail -= skip; - gst_buffer_set_caps (tmp, GST_PAD_CAPS (depayload->srcpad)); + GST_BUFFER_TIMESTAMP (tmp) = timestamp; + gst_base_rtp_depayload_push (depayload, tmp); - /* only apply the timestamp for the first buffer. Based on gstrtpmp4gdepay.c */ - if (i == 0) - gst_base_rtp_depayload_push_ts (depayload, timestamp, tmp); - else - gst_base_rtp_depayload_push (depayload, tmp); + /* only apply the timestamp for the first buffer */ + timestamp = -1; } /* just a check that lengths match */ From d4167c4feecc7485d9e16c11978d4d8f58689f4b Mon Sep 17 00:00:00 2001 From: David Schleef Date: Thu, 30 Dec 2010 12:07:52 -0800 Subject: [PATCH 102/351] videobalance: use G_OS_WIN32 for windows check --- gst/videofilter/gstvideobalance.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/gst/videofilter/gstvideobalance.c b/gst/videofilter/gstvideobalance.c index cc27b0ba23..379330bf35 100644 --- a/gst/videofilter/gstvideobalance.c +++ b/gst/videofilter/gstvideobalance.c @@ -55,7 +55,7 @@ #define M_PI 3.14159265358979323846 #endif -#ifdef WIN32 +#ifdef G_OS_WIN32 #define rint(x) (floor((x)+0.5)) #endif From 7b8981766b184457f362341faaf70c260de2185e Mon Sep 17 00:00:00 2001 From: David Schleef Date: Thu, 30 Dec 2010 14:20:52 -0800 Subject: [PATCH 103/351] Change M_PI to G_PI --- gst/audiofx/audiochebband.c | 16 ++++++------- gst/audiofx/audiocheblimit.c | 8 +++---- gst/audiofx/audiokaraoke.c | 4 ++-- gst/audiofx/audiowsincband.c | 16 ++++++------- gst/audiofx/audiowsinclimit.c | 8 +++---- gst/effectv/gstop.c | 4 ++-- gst/equalizer/gstiirequalizer.c | 10 ++++----- gst/goom/convolve_fx.c | 2 +- gst/goom/ifs.c | 4 ++-- gst/goom/lines.c | 4 ++-- gst/goom/tentacle3d.c | 26 +++++++++++----------- tests/examples/audiofx/firfilter-example.c | 2 +- tests/examples/audiofx/iirfilter-example.c | 2 +- 13 files changed, 53 insertions(+), 53 deletions(-) diff --git a/gst/audiofx/audiochebband.c b/gst/audiofx/audiochebband.c index 037304c58e..89a950c23e 100644 --- a/gst/audiofx/audiochebband.c +++ b/gst/audiofx/audiochebband.c @@ -229,7 +229,7 @@ generate_biquad_coefficients (GstAudioChebBand * filter, /* Calculate pole location for lowpass at frequency 1 */ { - gdouble angle = (M_PI / 2.0) * (2.0 * p - 1) / np; + gdouble angle = (G_PI / 2.0) * (2.0 * p - 1) / np; rp = -sin (angle); ip = cos (angle); @@ -266,7 +266,7 @@ generate_biquad_coefficients (GstAudioChebBand * filter, /* Calculate zero location for frequency 1 on the * unit circle for type 2 */ if (type == 2) { - gdouble angle = M_PI / (np * 2.0) + ((p - 1) * M_PI) / (np); + gdouble angle = G_PI / (np * 2.0) + ((p - 1) * G_PI) / (np); gdouble mag2; iz = cos (angle); @@ -339,10 +339,10 @@ generate_biquad_coefficients (GstAudioChebBand * filter, gdouble a, b, d; gdouble alpha, beta; gdouble w0 = - 2.0 * M_PI * (filter->lower_frequency / + 2.0 * G_PI * (filter->lower_frequency / GST_AUDIO_FILTER (filter)->format.rate); gdouble w1 = - 2.0 * M_PI * (filter->upper_frequency / + 2.0 * G_PI * (filter->upper_frequency / GST_AUDIO_FILTER (filter)->format.rate); if (filter->mode == MODE_BAND_PASS) { @@ -498,10 +498,10 @@ generate_coefficients (GstAudioChebBand * filter) /* gain is H(wc), wc = center frequency */ gdouble w1 = - 2.0 * M_PI * (filter->lower_frequency / + 2.0 * G_PI * (filter->lower_frequency / GST_AUDIO_FILTER (filter)->format.rate); gdouble w2 = - 2.0 * M_PI * (filter->upper_frequency / + 2.0 * G_PI * (filter->upper_frequency / GST_AUDIO_FILTER (filter)->format.rate); gdouble w0 = (w2 + w1) / 2.0; gdouble zr = cos (w0), zi = sin (w0); @@ -530,10 +530,10 @@ generate_coefficients (GstAudioChebBand * filter) np + 1, 1.0, 0.0))); { gdouble w1 = - 2.0 * M_PI * (filter->lower_frequency / + 2.0 * G_PI * (filter->lower_frequency / GST_AUDIO_FILTER (filter)->format.rate); gdouble w2 = - 2.0 * M_PI * (filter->upper_frequency / + 2.0 * G_PI * (filter->upper_frequency / GST_AUDIO_FILTER (filter)->format.rate); gdouble w0 = (w2 + w1) / 2.0; gdouble zr, zi; diff --git a/gst/audiofx/audiocheblimit.c b/gst/audiofx/audiocheblimit.c index 02cbc14bf4..8d51fbbf66 100644 --- a/gst/audiofx/audiocheblimit.c +++ b/gst/audiofx/audiocheblimit.c @@ -222,7 +222,7 @@ generate_biquad_coefficients (GstAudioChebLimit * filter, /* Calculate pole location for lowpass at frequency 1 */ { - gdouble angle = (M_PI / 2.0) * (2.0 * p - 1) / np; + gdouble angle = (G_PI / 2.0) * (2.0 * p - 1) / np; rp = -sin (angle); ip = cos (angle); @@ -259,7 +259,7 @@ generate_biquad_coefficients (GstAudioChebLimit * filter, /* Calculate zero location for frequency 1 on the * unit circle for type 2 */ if (type == 2) { - gdouble angle = M_PI / (np * 2.0) + ((p - 1) * M_PI) / (np); + gdouble angle = G_PI / (np * 2.0) + ((p - 1) * G_PI) / (np); gdouble mag2; iz = cos (angle); @@ -324,7 +324,7 @@ generate_biquad_coefficients (GstAudioChebLimit * filter, { gdouble k, d; gdouble omega = - 2.0 * M_PI * (filter->cutoff / GST_AUDIO_FILTER (filter)->format.rate); + 2.0 * G_PI * (filter->cutoff / GST_AUDIO_FILTER (filter)->format.rate); if (filter->mode == MODE_LOW_PASS) k = sin ((1.0 - omega) / 2.0) / sin ((1.0 + omega) / 2.0); @@ -456,7 +456,7 @@ generate_coefficients (GstAudioChebLimit * filter) #ifndef GST_DISABLE_GST_DEBUG { gdouble wc = - 2.0 * M_PI * (filter->cutoff / + 2.0 * G_PI * (filter->cutoff / GST_AUDIO_FILTER (filter)->format.rate); gdouble zr = cos (wc), zi = sin (wc); diff --git a/gst/audiofx/audiokaraoke.c b/gst/audiofx/audiokaraoke.c index aeda8d3b82..c6fb93ee49 100644 --- a/gst/audiofx/audiokaraoke.c +++ b/gst/audiofx/audiokaraoke.c @@ -179,8 +179,8 @@ update_filter (GstAudioKaraoke * filter, gint rate) if (rate == 0) return; - C = exp (-2 * M_PI * filter->filter_width / rate); - B = -4 * C / (1 + C) * cos (2 * M_PI * filter->filter_band / rate); + C = exp (-2 * G_PI * filter->filter_width / rate); + B = -4 * C / (1 + C) * cos (2 * G_PI * filter->filter_band / rate); A = sqrt (1 - B * B / (4 * C)) * (1 - C); filter->A = A; diff --git a/gst/audiofx/audiowsincband.c b/gst/audiofx/audiowsincband.c index 4b76a16a36..c6e2a0d1ae 100644 --- a/gst/audiofx/audiowsincband.c +++ b/gst/audiofx/audiowsincband.c @@ -255,7 +255,7 @@ gst_audio_wsincband_build_kernel (GstAudioWSincBand * self) (self->mode == MODE_BAND_PASS) ? "band-pass" : "band-reject"); /* fill the lp kernel */ - w = 2 * M_PI * (self->lower_frequency / GST_AUDIO_FILTER (self)->format.rate); + w = 2 * G_PI * (self->lower_frequency / GST_AUDIO_FILTER (self)->format.rate); kernel_lp = g_new (gdouble, len); for (i = 0; i < len; ++i) { if (i == len / 2) @@ -265,11 +265,11 @@ gst_audio_wsincband_build_kernel (GstAudioWSincBand * self) / (i - len / 2); /* Windowing */ if (self->window == WINDOW_HAMMING) - kernel_lp[i] *= (0.54 - 0.46 * cos (2 * M_PI * i / len)); + kernel_lp[i] *= (0.54 - 0.46 * cos (2 * G_PI * i / len)); else kernel_lp[i] *= - (0.42 - 0.5 * cos (2 * M_PI * i / len) + - 0.08 * cos (4 * M_PI * i / len)); + (0.42 - 0.5 * cos (2 * G_PI * i / len) + + 0.08 * cos (4 * G_PI * i / len)); } /* normalize for unity gain at DC */ @@ -280,7 +280,7 @@ gst_audio_wsincband_build_kernel (GstAudioWSincBand * self) kernel_lp[i] /= sum; /* fill the hp kernel */ - w = 2 * M_PI * (self->upper_frequency / GST_AUDIO_FILTER (self)->format.rate); + w = 2 * G_PI * (self->upper_frequency / GST_AUDIO_FILTER (self)->format.rate); kernel_hp = g_new (gdouble, len); for (i = 0; i < len; ++i) { if (i == len / 2) @@ -290,11 +290,11 @@ gst_audio_wsincband_build_kernel (GstAudioWSincBand * self) / (i - len / 2); /* Windowing */ if (self->window == WINDOW_HAMMING) - kernel_hp[i] *= (0.54 - 0.46 * cos (2 * M_PI * i / len)); + kernel_hp[i] *= (0.54 - 0.46 * cos (2 * G_PI * i / len)); else kernel_hp[i] *= - (0.42 - 0.5 * cos (2 * M_PI * i / len) + - 0.08 * cos (4 * M_PI * i / len)); + (0.42 - 0.5 * cos (2 * G_PI * i / len) + + 0.08 * cos (4 * G_PI * i / len)); } /* normalize for unity gain at DC */ diff --git a/gst/audiofx/audiowsinclimit.c b/gst/audiofx/audiowsinclimit.c index 4b79e930be..046f1ca268 100644 --- a/gst/audiofx/audiowsinclimit.c +++ b/gst/audiofx/audiowsinclimit.c @@ -239,7 +239,7 @@ gst_audio_wsinclimit_build_kernel (GstAudioWSincLimit * self) (self->mode == MODE_LOW_PASS) ? "low-pass" : "high-pass"); /* fill the kernel */ - w = 2 * M_PI * (self->cutoff / GST_AUDIO_FILTER (self)->format.rate); + w = 2 * G_PI * (self->cutoff / GST_AUDIO_FILTER (self)->format.rate); kernel = g_new (gdouble, len); @@ -250,10 +250,10 @@ gst_audio_wsinclimit_build_kernel (GstAudioWSincLimit * self) kernel[i] = sin (w * (i - len / 2)) / (i - len / 2); /* windowing */ if (self->window == WINDOW_HAMMING) - kernel[i] *= (0.54 - 0.46 * cos (2 * M_PI * i / len)); + kernel[i] *= (0.54 - 0.46 * cos (2 * G_PI * i / len)); else - kernel[i] *= (0.42 - 0.5 * cos (2 * M_PI * i / len) + - 0.08 * cos (4 * M_PI * i / len)); + kernel[i] *= (0.42 - 0.5 * cos (2 * G_PI * i / len) + + 0.08 * cos (4 * G_PI * i / len)); } /* normalize for unity gain at DC */ diff --git a/gst/effectv/gstop.c b/gst/effectv/gstop.c index d9e35dfe52..de798dc89b 100644 --- a/gst/effectv/gstop.c +++ b/gst/effectv/gstop.c @@ -161,14 +161,14 @@ setOpmap (gint8 * opmap[4], gint width, gint height) #endif opmap[OP_SPIRAL1][i] = ((guint) - ((at / M_PI * 256) + (r * 4000))) & 255; + ((at / G_PI * 256) + (r * 4000))) & 255; j = r * 300 / 32; rr = r * 300 - j * 32; j *= 64; j += (rr > 28) ? (rr - 28) * 16 : 0; opmap[OP_SPIRAL2][i] = ((guint) - ((at / M_PI * 4096) + (r * 1600) - j)) & 255; + ((at / G_PI * 4096) + (r * 1600) - j)) & 255; opmap[OP_PARABOLA][i] = ((guint) (yy / (xx * xx * 0.3 + 0.1) * 400)) & 255; diff --git a/gst/equalizer/gstiirequalizer.c b/gst/equalizer/gstiirequalizer.c index b679d4d9d7..0b767489ea 100644 --- a/gst/equalizer/gstiirequalizer.c +++ b/gst/equalizer/gstiirequalizer.c @@ -441,11 +441,11 @@ calculate_omega (gdouble freq, gint rate) gdouble omega; if (freq / rate >= 0.5) - omega = M_PI; + omega = G_PI; else if (freq <= 0.0) omega = 0.0; else - omega = 2.0 * M_PI * (freq / rate); + omega = 2.0 * G_PI * (freq / rate); return omega; } @@ -456,10 +456,10 @@ calculate_bw (GstIirEqualizerBand * band, gint rate) gdouble bw = 0.0; if (band->width / rate >= 0.5) { - /* If bandwidth == 0.5 the calculation below fails as tan(M_PI/2) + /* If bandwidth == 0.5 the calculation below fails as tan(G_PI/2) * is undefined. So set the bandwidth to a slightly smaller value. */ - bw = M_PI - 0.00000001; + bw = G_PI - 0.00000001; } else if (band->width <= 0.0) { /* If bandwidth == 0 this band won't change anything so set * the coefficients accordingly. The coefficient calculation @@ -472,7 +472,7 @@ calculate_bw (GstIirEqualizerBand * band, gint rate) band->b1 = 0.0; band->b2 = 0.0; } else { - bw = 2.0 * M_PI * (band->width / rate); + bw = 2.0 * G_PI * (band->width / rate); } return bw; } diff --git a/gst/goom/convolve_fx.c b/gst/goom/convolve_fx.c index 349de4d28e..457b5eb2cd 100644 --- a/gst/goom/convolve_fx.c +++ b/gst/goom/convolve_fx.c @@ -76,7 +76,7 @@ compute_tables (VisualFX * _this, PluginInfo * info) data->h_height = info->screen.height; for (i = 0; i < NB_THETA; i++) { - radian = 2 * i * M_PI / NB_THETA; + radian = 2 * i * G_PI / NB_THETA; h = (0.2 + cos (radian) / 15.0 * sin (radian * 2.0 + 12.123)) * screen_coef; data->h_cos[i] = 0x10000 * (-h * cos (radian) * cos (radian)); data->h_sin[i] = 0x10000 * (h * sin (radian + 1.57) * sin (radian)); diff --git a/gst/goom/ifs.c b/gst/goom/ifs.c index 9c59d927fb..3684dcc516 100644 --- a/gst/goom/ifs.c +++ b/gst/goom/ifs.c @@ -177,8 +177,8 @@ Random_Simis (PluginInfo * goomInfo, FRACTAL * F, SIMI * Cur, int i) Cur->c_y = Gauss_Rand (goomInfo, 0.0, .8, 4.0); Cur->r = Gauss_Rand (goomInfo, F->r_mean, F->dr_mean, 3.0); Cur->r2 = Half_Gauss_Rand (goomInfo, 0.0, F->dr2_mean, 2.0); - Cur->A = Gauss_Rand (goomInfo, 0.0, 360.0, 4.0) * (M_PI / 180.0); - Cur->A2 = Gauss_Rand (goomInfo, 0.0, 360.0, 4.0) * (M_PI / 180.0); + Cur->A = Gauss_Rand (goomInfo, 0.0, 360.0, 4.0) * (G_PI / 180.0); + Cur->A2 = Gauss_Rand (goomInfo, 0.0, 360.0, 4.0) * (G_PI / 180.0); Cur++; } } diff --git a/gst/goom/lines.c b/gst/goom/lines.c index 849fbe6e8f..096240cda9 100644 --- a/gst/goom/lines.c +++ b/gst/goom/lines.c @@ -70,7 +70,7 @@ genline (int id, float param, GMUnitPointer * l, int rx, int ry) for (i = 0; i < 512; i++) { l[i].x = ((float) i * rx) / 512.0f; l[i].y = param; - l[i].angle = M_PI / 2.0f; + l[i].angle = G_PI / 2.0f; } return; case GML_VLINE: @@ -84,7 +84,7 @@ genline (int id, float param, GMUnitPointer * l, int rx, int ry) for (i = 0; i < 512; i++) { float cosa, sina; - l[i].angle = 2.0f * M_PI * (float) i / 512.0f; + l[i].angle = 2.0f * G_PI * (float) i / 512.0f; cosa = param * cos (l[i].angle); sina = param * sin (l[i].angle); l[i].x = ((float) rx / 2.0f) + cosa; diff --git a/gst/goom/tentacle3d.c b/gst/goom/tentacle3d.c index 83de43e6b7..4c0d1438d1 100644 --- a/gst/goom/tentacle3d.c +++ b/gst/goom/tentacle3d.c @@ -51,7 +51,7 @@ typedef struct _TENTACLE_FX_DATA /* statics from pretty_move */ float distt; float distt2; - float rot; /* entre 0 et 2 * M_PI */ + float rot; /* entre 0 et 2 * G_PI */ int happens; int rotation; int lock; @@ -85,7 +85,7 @@ tentacle_fx_init (VisualFX * _this, PluginInfo * info) data->distt = 10.0f; data->distt2 = 0.0f; - data->rot = 0.0f; /* entre 0 et 2 * M_PI */ + data->rot = 0.0f; /* entre 0 et 2 * G_PI */ data->happens = 0; data->rotation = 0; @@ -253,27 +253,27 @@ pretty_move (PluginInfo * goomInfo, float cycle, float *dist, float *dist2, *dist = fx_data->distt = (tmp + 3.0f * fx_data->distt) / 4.0f; if (!fx_data->happens) { - tmp = M_PI * sin (cycle) / 32 + 3 * M_PI / 2; + tmp = G_PI * sin (cycle) / 32 + 3 * G_PI / 2; } else { fx_data->rotation = goom_irand (goomInfo->gRandom, 500) ? fx_data->rotation : goom_irand (goomInfo->gRandom, 2); if (fx_data->rotation) - cycle *= 2.0f * M_PI; + cycle *= 2.0f * G_PI; else - cycle *= -1.0f * M_PI; - tmp = cycle - (M_PI * 2.0) * floor (cycle / (M_PI * 2.0)); + cycle *= -1.0f * G_PI; + tmp = cycle - (G_PI * 2.0) * floor (cycle / (G_PI * 2.0)); } - if (abs (tmp - fx_data->rot) > abs (tmp - (fx_data->rot + 2.0 * M_PI))) { - fx_data->rot = (tmp + 15.0f * (fx_data->rot + 2 * M_PI)) / 16.0f; - if (fx_data->rot > 2.0 * M_PI) - fx_data->rot -= 2.0 * M_PI; + if (abs (tmp - fx_data->rot) > abs (tmp - (fx_data->rot + 2.0 * G_PI))) { + fx_data->rot = (tmp + 15.0f * (fx_data->rot + 2 * G_PI)) / 16.0f; + if (fx_data->rot > 2.0 * G_PI) + fx_data->rot -= 2.0 * G_PI; *rotangle = fx_data->rot; - } else if (abs (tmp - fx_data->rot) > abs (tmp - (fx_data->rot - 2.0 * M_PI))) { - fx_data->rot = (tmp + 15.0f * (fx_data->rot - 2.0 * M_PI)) / 16.0f; + } else if (abs (tmp - fx_data->rot) > abs (tmp - (fx_data->rot - 2.0 * G_PI))) { + fx_data->rot = (tmp + 15.0f * (fx_data->rot - 2.0 * G_PI)) / 16.0f; if (fx_data->rot < 0.0f) - fx_data->rot += 2.0 * M_PI; + fx_data->rot += 2.0 * G_PI; *rotangle = fx_data->rot; } else *rotangle = fx_data->rot = (tmp + 15.0f * fx_data->rot) / 16.0f; diff --git a/tests/examples/audiofx/firfilter-example.c b/tests/examples/audiofx/firfilter-example.c index a0c31578d5..b344e74e63 100644 --- a/tests/examples/audiofx/firfilter-example.c +++ b/tests/examples/audiofx/firfilter-example.c @@ -90,7 +90,7 @@ on_rate_changed (GstElement * element, gint rate, gpointer user_data) * a better result than given from the rectangular window */ for (i = 0; i < 32; i++) - filter_kernel[i] *= (0.54 - 0.46 * cos (2 * M_PI * i / 32)); + filter_kernel[i] *= (0.54 - 0.46 * cos (2 * G_PI * i / 32)); va = g_value_array_new (1); diff --git a/tests/examples/audiofx/iirfilter-example.c b/tests/examples/audiofx/iirfilter-example.c index 8ccffb7433..7fac2ac92b 100644 --- a/tests/examples/audiofx/iirfilter-example.c +++ b/tests/examples/audiofx/iirfilter-example.c @@ -63,7 +63,7 @@ on_rate_changed (GstElement * element, gint rate, gpointer user_data) gdouble x; if (rate / 2.0 > CUTOFF) - x = exp (-2.0 * M_PI * (CUTOFF / rate)); + x = exp (-2.0 * G_PI * (CUTOFF / rate)); else x = 0.0; From 954a04e2b57d98c4fd5dd004470abfa603a077ef Mon Sep 17 00:00:00 2001 From: David Schleef Date: Thu, 30 Dec 2010 14:21:37 -0800 Subject: [PATCH 104/351] cairo: Don't use #ifdefs inside macros --- ext/cairo/gstcairorender.c | 61 ++++++++++++++++++++++---------------- 1 file changed, 36 insertions(+), 25 deletions(-) diff --git a/ext/cairo/gstcairorender.c b/ext/cairo/gstcairorender.c index 827ef3d26b..cf1b563aa8 100644 --- a/ext/cairo/gstcairorender.c +++ b/ext/cairo/gstcairorender.c @@ -264,52 +264,63 @@ gst_cairo_render_setcaps_sink (GstPad * pad, GstCaps * caps) return TRUE; } -static GstStaticPadTemplate t_src = GST_STATIC_PAD_TEMPLATE ("src", - GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ( + +#define SIZE_CAPS "width = (int) [ 1, MAX], height = (int) [ 1, MAX] " #if CAIRO_HAS_PDF_SURFACE - "application/pdf, " - "width = (int) [ 1, MAX], " "height = (int) [ 1, MAX] " +#define PDF_CAPS "application/pdf, " SIZE_CAPS +#else +#define PDF_CAPS #endif #if CAIRO_HAS_PDF_SURFACE && (CAIRO_HAS_PS_SURFACE || CAIRO_HAS_SVG_SURFACE || CAIRO_HAS_PNG_FUNCTIONS) - ";" +#define JOIN1 ";" +#else +#define JOIN1 #endif #if CAIRO_HAS_PS_SURFACE - "application/postscript, " - "width = (int) [ 1, MAX], " "height = (int) [ 1, MAX] " +#define PS_CAPS "application/postscript, " SIZE_CAPS +#else +#define PS_CAPS #endif #if (CAIRO_HAS_PDF_SURFACE || CAIRO_HAS_PS_SURFACE) && (CAIRO_HAS_SVG_SURFACE || CAIRO_HAS_PNG_FUNCTIONS) - ";" +#define JOIN2 ";" +#else +#define JOIN2 #endif #if CAIRO_HAS_SVG_SURFACE - "image/svg+xml, " - "width = (int) [ 1, MAX], " "height = (int) [ 1, MAX] " +#define SVG_CAPS "image/svg+xml, " SIZE_CAPS +#else +#define SVG_CAPS #endif #if (CAIRO_HAS_PDF_SURFACE || CAIRO_HAS_PS_SURFACE || CAIRO_HAS_SVG_SURFACE) && CAIRO_HAS_PNG_FUNCTIONS - ";" +#define JOIN3 ";" +#else +#define JOIN3 #endif #if CAIRO_HAS_PNG_FUNCTIONS - "image/png, " "width = (int) [ 1, MAX], " "height = (int) [ 1, MAX] " -#endif - )); -static GstStaticPadTemplate t_snk = GST_STATIC_PAD_TEMPLATE ("sink", - GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ( -#if G_BYTE_ORDER == G_LITTLE_ENDIAN - GST_VIDEO_CAPS_BGRx " ; " GST_VIDEO_CAPS_BGRA " ; " +#define PNG_CAPS "image/png, " SIZE_CAPS +#define PNG_CAPS2 "; image/png, " SIZE_CAPS #else - GST_VIDEO_CAPS_xRGB " ; " GST_VIDEO_CAPS_ARGB " ; " +#define PNG_CAPS +#define PNG_CAPS2 #endif + +#if G_BYTE_ORDER == G_LITTLE_ENDIAN +#define ARGB_CAPS GST_VIDEO_CAPS_BGRx " ; " GST_VIDEO_CAPS_BGRA " ; " +#else +#define ARGB_CAPS GST_VIDEO_CAPS_xRGB " ; " GST_VIDEO_CAPS_ARGB " ; " +#endif +static GstStaticPadTemplate t_src = GST_STATIC_PAD_TEMPLATE ("src", + GST_PAD_SRC, GST_PAD_ALWAYS, + GST_STATIC_CAPS (PDF_CAPS JOIN1 PS_CAPS JOIN2 SVG_CAPS JOIN3 PNG_CAPS)); +static GstStaticPadTemplate t_snk = GST_STATIC_PAD_TEMPLATE ("sink", + GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS (ARGB_CAPS GST_VIDEO_CAPS_YUV ("Y800") " ; " "video/x-raw-gray, " "bpp = 8, " "depth = 8, " "width = " GST_VIDEO_SIZE_RANGE ", " "height = " GST_VIDEO_SIZE_RANGE ", " "framerate = " GST_VIDEO_FPS_RANGE - " ; " -#if CAIRO_HAS_PNG_FUNCTIONS - "image/png, " - "width = " GST_VIDEO_SIZE_RANGE ", " "height = " GST_VIDEO_SIZE_RANGE -#endif - )); + PNG_CAPS2)); GST_BOILERPLATE (GstCairoRender, gst_cairo_render, GstElement, GST_TYPE_ELEMENT); From cdaf72b614f93f355ea4615dd0313d4a848b4f6a Mon Sep 17 00:00:00 2001 From: David Schleef Date: Thu, 30 Dec 2010 14:30:27 -0800 Subject: [PATCH 105/351] videobalance: Check for HAVE_RINT instead Also change M_PI to G_PI for giggles. --- gst/videofilter/gstvideobalance.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) diff --git a/gst/videofilter/gstvideobalance.c b/gst/videofilter/gstvideobalance.c index 379330bf35..136acaea33 100644 --- a/gst/videofilter/gstvideobalance.c +++ b/gst/videofilter/gstvideobalance.c @@ -51,11 +51,7 @@ #include #include -#ifndef M_PI -#define M_PI 3.14159265358979323846 -#endif - -#ifdef G_OS_WIN32 +#ifndef HAVE_RINT #define rint(x) (floor((x)+0.5)) #endif @@ -170,8 +166,8 @@ gst_video_balance_update_tables (GstVideoBalance * vb) vb->tabley[i] = rint (y); } - hue_cos = cos (M_PI * vb->hue); - hue_sin = sin (M_PI * vb->hue); + hue_cos = cos (G_PI * vb->hue); + hue_sin = sin (G_PI * vb->hue); /* U/V lookup tables are 2D, since we need both U/V for each table * separately. */ From cfa7225898948551b590653a10265d6c6344eaf6 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Fri, 31 Dec 2010 02:16:54 +0000 Subject: [PATCH 106/351] cairo, videofilter: use gst/math-compat.h header for rint --- ext/cairo/gsttimeoverlay.c | 7 ++----- gst/videofilter/gstvideobalance.c | 7 ++----- 2 files changed, 4 insertions(+), 10 deletions(-) diff --git a/ext/cairo/gsttimeoverlay.c b/ext/cairo/gsttimeoverlay.c index 18f18f8fe5..b731c37894 100644 --- a/ext/cairo/gsttimeoverlay.c +++ b/ext/cairo/gsttimeoverlay.c @@ -36,19 +36,16 @@ #include "config.h" #endif +#include + #include #include -#include #include #include -#ifndef HAVE_RINT -#define rint(x) ((double) floor((x)+(((x) < 0)? -0.5 : 0.5))) -#endif - static GstStaticPadTemplate gst_cairo_time_overlay_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, diff --git a/gst/videofilter/gstvideobalance.c b/gst/videofilter/gstvideobalance.c index 136acaea33..968e2596c2 100644 --- a/gst/videofilter/gstvideobalance.c +++ b/gst/videofilter/gstvideobalance.c @@ -44,17 +44,14 @@ #include "config.h" #endif +#include + #include "gstvideobalance.h" #include -#include #include #include -#ifndef HAVE_RINT -#define rint(x) (floor((x)+0.5)) -#endif - GST_DEBUG_CATEGORY_STATIC (videobalance_debug); #define GST_CAT_DEFAULT videobalance_debug From 5ed3701a2d0c7f22473d6d4ae551b58adb8049b3 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Fri, 31 Dec 2010 13:57:05 +0100 Subject: [PATCH 107/351] mp4adepay: improve timestamps on outgoing packets Improve parsing of the samplerate. Parse the framelen so that we can calculate timestamps. When interpollate the incomming timestamp on outgoing buffers when there are multiple subframes. fixes #625825 --- gst/rtp/gstrtpmp4adepay.c | 97 +++++++++++++++++++++++++++++++-------- gst/rtp/gstrtpmp4adepay.h | 1 + 2 files changed, 80 insertions(+), 18 deletions(-) diff --git a/gst/rtp/gstrtpmp4adepay.c b/gst/rtp/gstrtpmp4adepay.c index 836ff9f095..394fc0d288 100644 --- a/gst/rtp/gstrtpmp4adepay.c +++ b/gst/rtp/gstrtpmp4adepay.c @@ -21,6 +21,7 @@ # include "config.h" #endif +#include #include #include @@ -128,6 +129,10 @@ gst_rtp_mp4a_depay_finalize (GObject * object) G_OBJECT_CLASS (parent_class)->finalize (object); } +static const guint aac_sample_rates[] = { 96000, 88200, 64000, 48000, + 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000 +}; + static gboolean gst_rtp_mp4a_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps) { @@ -165,10 +170,9 @@ gst_rtp_mp4a_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps) guint8 *data; guint size; gint i; - guint sr_idx; - static const guint aac_sample_rates[] = { 96000, 88200, 64000, 48000, - 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000 - }; + guint32 rate; + guint8 obj_type, sr_idx, channels; + GstBitReader br; buffer = gst_value_get_buffer (&v); gst_buffer_ref (buffer); @@ -210,22 +214,68 @@ gst_rtp_mp4a_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps) for (i = 0; i < size; i++) { data[i] = ((data[i + 1] & 1) << 7) | ((data[i + 2] & 0xfe) >> 1); } - - /* grab and set sampling rate */ - sr_idx = ((data[0] & 0x07) << 1) | ((data[1] & 0x80) >> 7); - if (sr_idx < G_N_ELEMENTS (aac_sample_rates)) { - gst_caps_set_simple (srccaps, - "rate", G_TYPE_INT, (gint) aac_sample_rates[sr_idx], NULL); - GST_DEBUG_OBJECT (depayload, "sampling rate from stream-config %u", - aac_sample_rates[sr_idx]); - } else { - GST_WARNING_OBJECT (depayload, "Invalid sample rate index %u", sr_idx); - } - /* ignore remaining bit, we're only interested in full bytes */ GST_BUFFER_SIZE (buffer) = size; + gst_bit_reader_init (&br, data, size); + + /* any object type is fine, we need to copy it to the profile-level-id field. */ + if (!gst_bit_reader_get_bits_uint8 (&br, &obj_type, 5)) + goto bad_config; + if (obj_type == 0) { + GST_WARNING_OBJECT (depayload, "invalid object type 0"); + goto bad_config; + } + + if (!gst_bit_reader_get_bits_uint8 (&br, &sr_idx, 4)) + goto bad_config; + if (sr_idx > 12 && sr_idx != 15) { + GST_WARNING_OBJECT (depayload, "invalid sample rate index %d", sr_idx); + goto bad_config; + } + GST_LOG_OBJECT (rtpmp4adepay, "sample rate index %u", sr_idx); + + if (!gst_bit_reader_get_bits_uint8 (&br, &channels, 4)) + goto bad_config; + if (channels > 7) { + GST_WARNING_OBJECT (depayload, "invalid channels %u", (guint) channels); + goto bad_config; + } + + /* rtp rate depends on sampling rate of the audio */ + if (sr_idx == 15) { + /* index of 15 means we get the rate in the next 24 bits */ + if (!gst_bit_reader_get_bits_uint32 (&br, &rate, 24)) + goto bad_config; + } else { + /* else use the rate from the table */ + rate = aac_sample_rates[sr_idx]; + } + + rtpmp4adepay->frame_len = 1024; + + switch (obj_type) { + case 1: + case 2: + case 3: + case 4: + case 6: + case 7: + { + guint8 frameLenFlag = 0; + + if (gst_bit_reader_get_bits_uint8 (&br, &frameLenFlag, 1)) + if (frameLenFlag) + rtpmp4adepay->frame_len = 960; + break; + } + default: + break; + } + gst_caps_set_simple (srccaps, + "channels", G_TYPE_INT, (gint) channels, + "rate", G_TYPE_INT, (gint) rate, "codec_data", GST_TYPE_BUFFER, buffer, NULL); gst_buffer_unref (buffer); } else { @@ -254,6 +304,7 @@ gst_rtp_mp4a_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf) outbuf = gst_rtp_buffer_get_payload_buffer (buf); + gst_buffer_copy_metadata (outbuf, buf, GST_BUFFER_COPY_TIMESTAMPS); gst_adapter_push (rtpmp4adepay->adapter, outbuf); /* RTP marker bit indicates the last packet of the AudioMuxElement => create @@ -315,11 +366,19 @@ gst_rtp_mp4a_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf) data += skip; avail -= skip; + if (offset > 0 && timestamp != -1 && depayload->clock_rate != 0) { + timestamp += + gst_util_uint64_scale_int (offset, GST_SECOND, + depayload->clock_rate); + } + GST_BUFFER_TIMESTAMP (tmp) = timestamp; gst_base_rtp_depayload_push (depayload, tmp); - /* only apply the timestamp for the first buffer */ - timestamp = -1; + /* calculate offsets for next buffers */ + if (rtpmp4adepay->frame_len) { + offset += rtpmp4adepay->frame_len; + } } /* just a check that lengths match */ @@ -355,6 +414,8 @@ gst_rtp_mp4a_depay_change_state (GstElement * element, switch (transition) { case GST_STATE_CHANGE_READY_TO_PAUSED: gst_adapter_clear (rtpmp4adepay->adapter); + rtpmp4adepay->frame_len = 0; + rtpmp4adepay->numSubFrames = 0; break; default: break; diff --git a/gst/rtp/gstrtpmp4adepay.h b/gst/rtp/gstrtpmp4adepay.h index 16c20b7f7a..33622fcaed 100644 --- a/gst/rtp/gstrtpmp4adepay.h +++ b/gst/rtp/gstrtpmp4adepay.h @@ -44,6 +44,7 @@ struct _GstRtpMP4ADepay GstBaseRTPDepayload depayload; GstAdapter *adapter; guint8 numSubFrames; + guint frame_len; }; struct _GstRtpMP4ADepayClass From 88e90f36b5a6a05d829ff09d678fe4838deb1daf Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Sat, 1 Jan 2011 23:26:33 +0000 Subject: [PATCH 108/351] tools: remove unused left-over directory These are all in -base/tools. --- tools/.gitignore | 4 - tools/Makefile.am | 39 --------- tools/README.filterstamp | 24 ------ tools/filterstamp.sh | 57 ------------- tools/gst-launch-ext-m.m | 169 -------------------------------------- tools/gst-launch-ext.1.in | 45 ---------- tools/gst-visualise-m.m | 79 ------------------ tools/gst-visualise.1.in | 35 -------- 8 files changed, 452 deletions(-) delete mode 100644 tools/.gitignore delete mode 100644 tools/Makefile.am delete mode 100644 tools/README.filterstamp delete mode 100755 tools/filterstamp.sh delete mode 100644 tools/gst-launch-ext-m.m delete mode 100644 tools/gst-launch-ext.1.in delete mode 100644 tools/gst-visualise-m.m delete mode 100644 tools/gst-visualise.1.in diff --git a/tools/.gitignore b/tools/.gitignore deleted file mode 100644 index e3970bba92..0000000000 --- a/tools/.gitignore +++ /dev/null @@ -1,4 +0,0 @@ -gst-launch-ext-?.? -gst-visualise-?.? -gst-launch-ext-?.?.1 -gst-visualise-?.?.1 diff --git a/tools/Makefile.am b/tools/Makefile.am deleted file mode 100644 index 1b8ab850ad..0000000000 --- a/tools/Makefile.am +++ /dev/null @@ -1,39 +0,0 @@ -bin_SCRIPTS = \ - gst-launch-ext-@GST_MAJORMINOR@ \ - gst-visualise-@GST_MAJORMINOR@ - -man_MANS = \ - gst-launch-ext-@GST_MAJORMINOR@.1 \ - gst-visualise-@GST_MAJORMINOR@.1 - -CLEANFILES = $(man_MANS) $(bin_SCRIPTS) - -EXTRA_DIST = \ - gst-launch-ext-m.m gst-visualise-m.m \ - gst-launch-ext.1.in gst-visualise.1.in - -# generate versioned scripts from templates -%-@GST_MAJORMINOR@: %-m.m - sed -e s,\@GST_MAJORMINOR\@,@GST_MAJORMINOR@,g $< > $@ - chmod +x $@ - -# generate man pages -%-@GST_MAJORMINOR@.1: %.1.in - sed \ - -e s,gst-complete,gst-complete-@GST_MAJORMINOR@,g \ - -e s,gst-compprep,gst-compprep-@GST_MAJORMINOR@,g \ - -e s,gst-feedback,gst-feedback-@GST_MAJORMINOR@,g \ - -e s,gst-inspect,gst-inspect-@GST_MAJORMINOR@,g \ - -e s,gst-launch,gst-launch-@GST_MAJORMINOR@,g \ - -e s,gst-launch-ext,gst-launch-ext-@GST_MAJORMINOR@,g \ - -e s,gst-md5sum,gst-md5sum-@GST_MAJORMINOR@,g \ - -e s,gst-register,gst-register-@GST_MAJORMINOR@,g \ - -e s,gst-typefind,gst-typefind-@GST_MAJORMINOR@,g \ - -e s,gst-visualise,gst-visualise-@GST_MAJORMINOR@,g \ - -e s,gst-xmllaunch,gst-xmllaunch-@GST_MAJORMINOR@,g \ - $< >$@ - -all: all-am chmod - -chmod: $(bin_SCRIPTS) - chmod +x $^ diff --git a/tools/README.filterstamp b/tools/README.filterstamp deleted file mode 100644 index a7b3d21e6f..0000000000 --- a/tools/README.filterstamp +++ /dev/null @@ -1,24 +0,0 @@ -filterstamp.sh is a script to copy a filter and change filenames and all -occurrences of the old name to the new name. - -This is used for writing new audio filters. The best one to copy for now is -passthrough; it works on raw/audio int or float data. - -If your new filter is called StereoPan, for example, then do this : - -cd gst -../tools/filterstamp.sh Passthrough StereoPan -cd stereopan -make - -(Please note the upper- and lower-case !) - -You should also add a line to configure.ac to make sure the Makefiles are built -correctly. Just search for "passthrough" and add corresponding "stereopan" (or -whatever your plugin is) lines. - -Register it, then try it out. It shouldn't do anything ! - -Now edit the filter.func in the new directory; this file contains the body -of the main processing loop. - diff --git a/tools/filterstamp.sh b/tools/filterstamp.sh deleted file mode 100755 index 56b3d8d9ab..0000000000 --- a/tools/filterstamp.sh +++ /dev/null @@ -1,57 +0,0 @@ -#!/bin/bash - -# copies from gstreamer filter boilerplate -# to new filter -# changing file names and function references - -# thomas@apestaart.org - -# modified 23 aug 2001 apwingo@eos.ncsu.edu: -# conform better to gtk naming conventions (GstFoo->gst_foo in functions, etc) - -if [ "$1" = "" ] -then - echo "please specify the filter to copy FROM (e.g. Passthrough)" - exit -fi - -if [ "$2" = "" ] -then - echo "please specify the filter to copy TO (e.g. NewFilter)" - exit -fi - -FROM=$1 -TO=$2 -FROM_LC=`echo $FROM | tr [A-Z] [a-z]` -TO_LC=`echo $TO | tr [A-Z] [a-z]` -FROM_UC=`echo $FROM | tr [a-z] [A-Z]` -TO_UC=`echo $TO | tr [a-z] [A-Z]` -FROM_LC_UNDERSCORE=`echo $FROM | perl -n -p -e 's/([a-z])([A-Z])/$1_$2/g; tr/A-Z/a-z/'` -TO_LC_UNDERSCORE=`echo $TO | perl -n -p -e 's/([a-z])([A-Z])/$1_$2/g; tr/A-Z/a-z/'` - -echo "Copying filter boilerplate $FROM to new filter $TO..." - -if [ ! -d $FROM_LC ] -then - echo "Filter directory $FROM_LC does not exist !" - exit -fi - -if [ -d $TO_LC ] -then - echo "Filter directory $TO_LC already exists !" - exit -fi - -cp -r $FROM_LC $TO_LC - -cd $TO_LC - -for a in *$FROM_LC*; do mv $a `echo $a | sed s/$FROM_LC/$TO_LC/g`; done - -perl -i -p -e "s/$FROM/$TO/g" * -perl -i -p -e "s/${FROM_LC_UNDERSCORE}_/${TO_LC_UNDERSCORE}_/g" * -perl -i -p -e "s/$FROM_LC/$TO_LC/g" * -perl -i -p -e "s/$FROM_UC/$TO_UC/g" * - diff --git a/tools/gst-launch-ext-m.m b/tools/gst-launch-ext-m.m deleted file mode 100644 index 915f2dfa86..0000000000 --- a/tools/gst-launch-ext-m.m +++ /dev/null @@ -1,169 +0,0 @@ -#!/usr/bin/perl -w -use strict; - -# launch a gst-launch pipeline for the supplied media file -# use the extension to determine the gst-launch pipeline -# make use of default output sinks - -my (%pipes, %cfg); - -sub extension -{ - my $path = shift; - my $ext; - - # get only the bit after the last period. We don't deal with - # .tar.gz extensions do we ? - if ($path =~ /\./) - { - $ext = $path; - $ext =~ s/^.*\.//; - } - else { $ext = ""; } - - return $ext; -} - -sub read_config -{ - my $command = shift; - - my $config_file = $ENV{HOME}."/.gst"; - if (-e $config_file) - { - open CONFIG, $config_file; - while () - { - chomp; - s/#.*//; - s/\s+$//; - next unless length; - my ($var, $value) = split (/\s*=\s*/, $_, 2); - $cfg{$var} = $value; - } - if (!($cfg{AUDIOSINK})) - { - print "Please add an AUDIOSINK to $config_file !\n"; - } - if (!($cfg{VIDEOSINK})) - { - print "Please add a VIDEOSINK to $config_file !\n"; - } - } - else - { - print "No configuration file $config_file found. You might want to create one.\n"; - print "This is not an error, just a friendly reminder... Check the man page.\n\n"; - } - if (!defined $cfg{AUDIOSINK}) { $cfg{AUDIOSINK} = "osssink"; } - if (!defined $cfg{VIDEOSINK}) { $cfg{VIDEOSINK} = "ffmpegcolorspace ! xvimagesink"; } - if (!defined $cfg{CVS_PATH}) { $cfg{CVS_PATH} = $ENV{HOME}."/gst/cvs"; } - - if ($command =~ /(.+)\/gst-launch-ext-@GST_MAJORMINOR@$/) - { $cfg{COMMAND_PATH} = "$1"; } - else - { $cfg{COMMAND_PATH} = ""; } -} - -sub playfile($$) -{ - my ($file, $ext) = @_; - my $command; - my $pipe; - my $path = "\$PATH:".$cfg{CVS_PATH}."/gstreamer/tools"; - - if ($cfg{COMMAND_PATH} ne "") { - $path = $cfg{COMMAND_PATH}.":$path"; - } - - $ext = lc $ext; - - if ($cfg{VISUALIZER} && ($pipe = $pipes{"vis." . $ext})) - { - $command = "gst-launch-@GST_MAJORMINOR@ filesrc location=\"$file\" ! $pipe"; - print "Running command-line\n$command\n\n"; - system ("PATH=$path $command"); - } - elsif ($pipe = $pipes{$ext}) - { - $command = "gst-launch-@GST_MAJORMINOR@ filesrc location=\"$file\" ! $pipe"; - print "Running command-line\n$command\n\n"; - system ("PATH=$path $command"); - } - else - { - print "No suitable pipe found for extension $ext.\n"; - } -} - -### main - -read_config ($0); - -%pipes = ( - "ac3", "a52dec ! $cfg{AUDIOSINK}", - "au", "auparse ! $cfg{AUDIOSINK}", - "avi", "decodebin name=d { d. ! queue ! $cfg{VIDEOSINK} } { d. ! queue ! audioconvert ! audioscale ! $cfg{AUDIOSINK} }", - "asf", "decodebin name=d ! { d. ! queue ! $cfg{VIDEOSINK} } { d. ! queue ! audioconvert ! audioscale ! $cfg{AUDIOSINK} }", - "flac", "flacdec ! $cfg{AUDIOSINK}", - "fli", "flxdec ! colorspace ! $cfg{VIDEOSINK}", - "m1v", "mpeg2dec ! $cfg{VIDEOSINK}", - "m2v", "mpeg2dec ! $cfg{VIDEOSINK}", - "m4a", "qtdemux .audio_00 ! { queue ! faad ! $cfg{AUDIOSINK} }", - "aac", "faad ! $cfg{AUDIOSINK}", - "mod", "modplug ! $cfg{AUDIOSINK}", - "mpc", "musepackdec ! $cfg{AUDIOSINK}", - "mp2", "mad ! $cfg{AUDIOSINK}", - "mp3", "mad ! $cfg{AUDIOSINK}", - "mp4", "decodebin name=d { d. ! queue ! ffmpegcolorspace ! videoscale ! $cfg{VIDEOSINK} } { d. ! queue ! audioconvert ! audioscale ! $cfg{AUDIOSINK} }", - "mpeg", "mpegdemux name=d { d.audio_00 ! queue ! mad ! audioconvert ! audioscale ! $cfg{AUDIOSINK} } { d.video_00 ! queue ! mpeg2dec ! $cfg{VIDEOSINK} }", - "mpg", "mpegdemux name=demux { demux.video_00 ! queue ! mpeg2dec ! ffmpegcolorspace ! $cfg{VIDEOSINK} } { demux.audio_00 ! queue ! mad ! $cfg{AUDIOSINK} }", - "ogg", "oggdemux ! vorbisdec ! audioconvert ! $cfg{AUDIOSINK}", - "sid", "siddec ! $cfg{AUDIOSINK}", - "swf", "swfdec name=swfdec ! { queue ! colorspace ! $cfg{VIDEOSINK} } { swfdec. ! queue ! $cfg{AUDIOSINK} }", - "vob", "mpegdemux name=d { d.video_00 ! queue ! mpeg2dec ! $cfg{VIDEOSINK} } { d.audio_00 ! queue ! a52dec ! audioconvert ! audioscale ! $cfg{AUDIOSINK} }", - "wav", "wavparse ! $cfg{AUDIOSINK}", - "wm", "asfdemux name=demux ! { queue ! spider ! $cfg{VIDEOSINK} } { demux. ! queue ! spider ! $cfg{AUDIOSINK} }", -### a wma file can use wmav1 or wmav2 codec so we must use spider to decode it - "wma", "asfdemux name=demux ! spider ! $cfg{AUDIOSINK}", - "wmv", "asfdemux name=demux ! { queue ! spider ! $cfg{VIDEOSINK} } { demux. ! queue ! spider ! $cfg{AUDIOSINK} }", - "mkv", "matroskademux name=demux ! { queue ! spider ! $cfg{VIDEOSINK} } { demux. ! queue ! spider ! $cfg{AUDIOSINK} }", - "mka", "matroskademux ! spider ! $cfg{AUDIOSINK}", - "mov", "decodebin name=d { d. ! queue ! ffmpegcolorspace ! videoscale ! $cfg{VIDEOSINK} } { d. ! queue ! audioconvert ! audioscale ! $cfg{AUDIOSINK} }", -); - -if ($cfg{VISUALIZER}) { - %pipes = ( - %pipes, - "vis.mp3", "mad ! tee name=tee silent=true ! queue leaky=1 ! { $cfg{VISUALIZER} ! ffmpegcolorspace ! $cfg{VIDEOSINK} } tee. ! $cfg{AUDIOSINK}", - "vis.ogg", "vorbisdec ! tee name=tee silent=true ! queue leaky=1 ! { $cfg{VISUALIZER} ! ffmpegcolorspace ! $cfg{VIDEOSINK} } tee. ! $cfg{AUDIOSINK}", - "vis.wav", "wavparse ! tee name=tee silent=true ! queue leaky=1 ! { $cfg{VISUALIZER} ! ffmpegcolorspace ! $cfg{VIDEOSINK} } tee. ! $cfg{AUDIOSINK}", - ); -} - -if ($#ARGV == -1) { - print STDERR "Usage: gst-launch-ext-@GST_MAJORMINOR@ filename[s]\n"; - exit 1; -} - -my $file; -while ($file = shift @ARGV) { - my $ext = extension ($file); - if (!$ext) { - print "file $file doesn't have an extension !\n"; - exit; - } - if ($ext eq 'm3u') - { - open (PLAYLIST, '<', $file); - my $file2; - while ($file2 = ) { - chomp $file2; - my $ext2 = extension ($file2); - playfile($file2, $ext2); - } - close PLAYLIST; - } else { - playfile($file, $ext); - } -} diff --git a/tools/gst-launch-ext.1.in b/tools/gst-launch-ext.1.in deleted file mode 100644 index fd2ebf1be3..0000000000 --- a/tools/gst-launch-ext.1.in +++ /dev/null @@ -1,45 +0,0 @@ -.TH "GStreamer" "1" "February 2002" "" "" -.SH "NAME" -gst\-launch\-ext \- Run a predefined GStreamer pipeline -.SH "SYNOPSIS" -\fBgst\-launch\-ext\fR \fIfilename [filename...]\fR -.SH "DESCRIPTION" -.LP -\fIgst\-launch\-ext\fP is a tool that is used to run a basic predefined -\fIGStreamer\fP pipeline. This application is only used as a quick test to -ensure proper working of codecs and GStreamer. It doesn't handle more advanced -features like synchronisation. - -All supported formats in GStreamer should be playable by simply typing: - - gst\-launch\-ext filename - -It will also print out the pipeline it uses, so you can customize it using -cut and paste. - -.SH "CONFIGURATION" -.LP -\fIgst\-launch\-ext\fP can be configured by creating a .gst file in your -home directory. This is a perl-style configuration file and can override -the defaults for audio and video output sinks. - -Here is an example .gst file that implements the same defaults as hard-coded -in the script : - -AUDIOSINK = osssink -VIDEOSINK = ffmpegcolorspace ! xvimagesink - -You can change osssink to esdsink or alsasink (if you have -the plug-in), and you can change xvimagesink to ximagesink, aasink -or sdlvideosink. - -Other plug-ins might be used as well if GStreamer has them. - -.SH "SEE ALSO" -.BR gst\-guilaunch (1), -.BR gst\-complete (1), -.BR gst\-register (1), -.BR gst\-inspect (1), -.BR gst\-launch (1), -.SH "AUTHOR" -The GStreamer team at http://gstreamer.net/ diff --git a/tools/gst-visualise-m.m b/tools/gst-visualise-m.m deleted file mode 100644 index 36d4497413..0000000000 --- a/tools/gst-visualise-m.m +++ /dev/null @@ -1,79 +0,0 @@ -#!/usr/bin/perl -w - -# launch a gst-launch pipeline to display a visualisation of the -# input audio. -# make use of default input srcs. -# visualisation plugin is specified on command line. - -### packages - -use File::Basename; - - -my (%pipes, %cfg); - -sub read_config -{ - my $config_file = `echo -n ~`."/.gst"; - if (-e $config_file) - { - open CONFIG, $config_file; - while () - { - chomp; - s/#.*//; - s/\s+$//; - next unless length; - my ($var, $value) = split (/\s*=\s*/, $_, 2); - $cfg{$var} = $value; - } - if (!($cfg{AUDIOSRC})) - { - print "Please add an AUDIOSRC to $config_file !\n"; - } - if (!($cfg{VIDEOSINK})) - { - print "Please add a VIDEOSINK to $config_file !\n"; - } - } - else - { - print "No configuration file $config_file found. You might want to create one.\n"; - } - if (!defined $cfg{AUDIOSRC}) { $cfg{AUDIOSRC} = "osssrc"; } - if (!defined $cfg{VIDEOSINK}) { $cfg{VIDEOSINK} = "xvimagesink"; } - if (!defined $cfg{CVS_PATH}) { $cfg{CVS_PATH} = `echo -n ~`."/gst/cvs"; } -} - -sub visualise(@) -{ - my $vis = $cfg{VISUALIZER}; - $vis = shift() if ($#_ != -1); - $vis = "goom" unless $vis; - - my $pipe; - $pipe = $vis unless $pipe = $pipes{$vis}; - - $command = "gst-launch-@GST_MAJORMINOR@ $cfg{AUDIOSRC} ! $pipe ! { queue ! ffmpegcolorspace ! $cfg{VIDEOSINK} }"; - print "Running $command\n"; - system ("PATH=\$PATH:".$cfg{CVS_PATH}."/gstreamer/tools $command"); -} - -### main - -read_config (); - -%pipes = ( - "goom", "goom", - "chart", "audioconvert ! chart", - "synaesthesia", "synaesthesia", - "monoscope", "audioconvert ! monoscope" -); - -if ($#ARGV > 0) { - print STDERR "Usage: gst-visualise [visualiser]\n"; - exit 1; -} - -visualise(@ARGV); - diff --git a/tools/gst-visualise.1.in b/tools/gst-visualise.1.in deleted file mode 100644 index f61446ece9..0000000000 --- a/tools/gst-visualise.1.in +++ /dev/null @@ -1,35 +0,0 @@ -.TH "GStreamer" "1" "February 2002" "" "" -.SH "NAME" -gst\-visualise \- Run a GStreamer pipeline to display an audio visualisation -.SH "SYNOPSIS" -\fBgst\-visualise\fR \fI[visualiser]\fR -.SH "DESCRIPTION" -.LP -\fIgst\-visualise\fP is a tool that is used to run a basic \fIGStreamer\fP pipeline, to display a graphical visualisation of an audio stream. - -By default, the audio stream is read from ESD (the Enlightened Sound Daemon), -but this can be changed by setting the AUDIOSRC parameter in ~/.gst. For -example, you might set "AUDIOSRC=osssrc" to display a visualisation of the -sound input to your soundcard. - -You can select a visualiser by providing a parameter naming the visualiser. -For example: - - gst\-visualise synaesthesia - -will use the synaesthesia plugin. If no visualiser is named, the VISUALIZER -property in ~/.gst will be used. If this is not specified either, the goom -visualiser will be used. - -The videosink to use to display the visualisation will be read from the -VIDEOSINK parameter in ~/.gst, defaulting to sdlvideosink. - -.SH "SEE ALSO" -.BR gst\-launch\-ext (1), -.BR gst\-guilaunch (1), -.BR gst\-complete (1), -.BR gst\-register (1), -.BR gst\-inspect (1), -.BR gst\-launch (1), -.SH "AUTHOR" -The GStreamer team at http://gstreamer.net/ From 0be6bd355a8900cc35a1cb625edda97830f4e98a Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Sun, 2 Jan 2011 01:24:21 +0000 Subject: [PATCH 109/351] Revert "v4l2: add norm property" This reverts commit 9e1d419d07337e6db2cc3936472be205ce927e54. Reverting this since it adds unreviewed and bad API to v4l2src (property of type enum, with seemingly random and unsorted values). --- sys/v4l2/gstv4l2object.c | 81 +++++----------------------------------- sys/v4l2/gstv4l2object.h | 5 +-- sys/v4l2/gstv4l2sink.c | 17 +++------ sys/v4l2/gstv4l2tuner.c | 23 +----------- sys/v4l2/gstv4l2tuner.h | 5 --- sys/v4l2/v4l2_calls.c | 3 -- 6 files changed, 18 insertions(+), 116 deletions(-) diff --git a/sys/v4l2/gstv4l2object.c b/sys/v4l2/gstv4l2object.c index f4512187c9..21ec2bc622 100644 --- a/sys/v4l2/gstv4l2object.c +++ b/sys/v4l2/gstv4l2object.c @@ -49,7 +49,7 @@ GST_DEBUG_CATEGORY_EXTERN (v4l2_debug); #define DEFAULT_PROP_DEVICE_NAME NULL #define DEFAULT_PROP_DEVICE_FD -1 #define DEFAULT_PROP_FLAGS 0 -#define DEFAULT_PROP_TV_NORM 0 +#define DEFAULT_PROP_NORM NULL #define DEFAULT_PROP_CHANNEL NULL #define DEFAULT_PROP_FREQUENCY 0 @@ -309,54 +309,6 @@ gst_v4l2_device_get_type (void) return v4l2_device_type; } -#define GST_TYPE_V4L2_TV_NORM (gst_v4l2_tv_norm_get_type ()) -static GType -gst_v4l2_tv_norm_get_type (void) -{ - static GType v4l2_tv_norm = 0; - - if (!v4l2_tv_norm) { - static const GEnumValue tv_norms[] = { - {0, "none", "none"}, - {V4L2_STD_NTSC, "NTSC", "NTSC"}, - {V4L2_STD_NTSC_M, "NTSC-M", "NTSC-M"}, - {V4L2_STD_NTSC_M_JP, "NTSC-M-JP", "NTSC-M-JP"}, - {V4L2_STD_NTSC_M_KR, "NTSC-M-KR", "NTSC-M-KR"}, - {V4L2_STD_NTSC_443, "NTSC-443", "NTSC-443"}, - {V4L2_STD_PAL, "PAL", "PAL"}, - {V4L2_STD_PAL_BG, "PAL-BG", "PAL-BG"}, - {V4L2_STD_PAL_B, "PAL-B", "PAL-B"}, - {V4L2_STD_PAL_B1, "PAL-B1", "PAL-B1"}, - {V4L2_STD_PAL_G, "PAL-G", "PAL-G"}, - {V4L2_STD_PAL_H, "PAL-H", "PAL-H"}, - {V4L2_STD_PAL_I, "PAL-I", "PAL-I"}, - {V4L2_STD_PAL_DK, "PAL-DK", "PAL-DK"}, - {V4L2_STD_PAL_D, "PAL-D", "PAL-D"}, - {V4L2_STD_PAL_D1, "PAL-D1", "PAL-D1"}, - {V4L2_STD_PAL_K, "PAL-K", "PAL-K"}, - {V4L2_STD_PAL_M, "PAL-M", "PAL-M"}, - {V4L2_STD_PAL_N, "PAL-N", "PAL-N"}, - {V4L2_STD_PAL_Nc, "PAL-Nc", "PAL-Nc"}, - {V4L2_STD_PAL_60, "PAL-60", "PAL-60"}, - {V4L2_STD_SECAM, "SECAM", "SECAM"}, - {V4L2_STD_SECAM_B, "SECAM-B", "SECAM-B"}, - {V4L2_STD_SECAM_G, "SECAM-G", "SECAM-G"}, - {V4L2_STD_SECAM_H, "SECAM-H", "SECAM-H"}, - {V4L2_STD_SECAM_DK, "SECAM-DK", "SECAM-DK"}, - {V4L2_STD_SECAM_D, "SECAM-D", "SECAM-D"}, - {V4L2_STD_SECAM_K, "SECAM-K", "SECAM-K"}, - {V4L2_STD_SECAM_K1, "SECAM-K1", "SECAM-K1"}, - {V4L2_STD_SECAM_L, "SECAM-L", "SECAM-L"}, - {V4L2_STD_SECAM_LC, "SECAM-Lc", "SECAM-Lc"}, - {0, NULL, NULL} - }; - - v4l2_tv_norm = g_enum_register_static ("V4L2_TV_norms", tv_norms); - } - - return v4l2_tv_norm; -} - void gst_v4l2_object_install_properties_helper (GObjectClass * gobject_class, const char *default_device) @@ -425,12 +377,6 @@ gst_v4l2_object_install_properties_helper (GObjectClass * gobject_class, "Hue or color balance", G_MININT, G_MAXINT, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | GST_PARAM_CONTROLLABLE)); - - g_object_class_install_property (gobject_class, PROP_TV_NORM, - g_param_spec_enum ("norm", "TV norm", - "video standard", - GST_TYPE_V4L2_TV_NORM, DEFAULT_PROP_TV_NORM, - G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); } GstV4l2Object * @@ -487,6 +433,9 @@ gst_v4l2_object_destroy (GstV4l2Object * v4l2object) if (v4l2object->channel) g_free (v4l2object->channel); + if (v4l2object->norm) + g_free (v4l2object->norm); + if (v4l2object->formats) { gst_v4l2_object_clear_format_list (v4l2object); } @@ -571,9 +520,6 @@ gst_v4l2_object_set_property_helper (GstV4l2Object * v4l2object, v4l2object->norm = g_value_dup_string (value); } break; - case PROP_TV_NORM: - v4l2object->tv_norm = g_value_get_enum (value); - break; case PROP_CHANNEL: if (GST_V4L2_IS_OPEN (v4l2object)) { GstTuner *tuner = GST_TUNER (v4l2object->element); @@ -678,9 +624,6 @@ gst_v4l2_object_get_property_helper (GstV4l2Object * v4l2object, return TRUE; } break; - case PROP_TV_NORM: - g_value_set_enum (value, v4l2object->tv_norm); - break; default: return FALSE; break; @@ -700,18 +643,16 @@ gst_v4l2_set_defaults (GstV4l2Object * v4l2object) tuner = GST_TUNER (v4l2object->element); - if (v4l2object->tv_norm) - norm = gst_v4l2_tuner_get_norm_by_std_id (v4l2object, v4l2object->tv_norm); - GST_DEBUG_OBJECT (v4l2object->element, "tv_norm=%llx, norm=%p", - v4l2object->tv_norm, norm); + if (v4l2object->norm) + norm = gst_tuner_find_norm_by_name (tuner, v4l2object->norm); if (norm) { gst_tuner_set_norm (tuner, norm); } else { norm = GST_TUNER_NORM (gst_tuner_get_norm (GST_TUNER (v4l2object->element))); if (norm) { - v4l2object->tv_norm = - gst_v4l2_tuner_get_std_id_by_norm (v4l2object, norm); + g_free (v4l2object->norm); + v4l2object->norm = g_strdup (norm->label); gst_tuner_norm_changed (tuner, norm); } } @@ -1921,15 +1862,13 @@ default_frame_sizes: } /* Since we can't get framerate directly, try to use the current norm */ - if (v4l2object->tv_norm && v4l2object->norms) { + if (v4l2object->norm && v4l2object->norms) { GList *norms; GstTunerNorm *norm = NULL; - GstTunerNorm *current = - gst_v4l2_tuner_get_norm_by_std_id (v4l2object, v4l2object->tv_norm); for (norms = v4l2object->norms; norms != NULL; norms = norms->next) { norm = (GstTunerNorm *) norms->data; - if (!strcmp (norm->label, current->label)) + if (!strcmp (norm->label, v4l2object->norm)) break; } /* If it's possible, set framerate to that (discrete) value */ diff --git a/sys/v4l2/gstv4l2object.h b/sys/v4l2/gstv4l2object.h index d83861a4d7..a0dd41ce11 100644 --- a/sys/v4l2/gstv4l2object.h +++ b/sys/v4l2/gstv4l2object.h @@ -108,7 +108,7 @@ struct _GstV4l2Object { GList *channels; /* properties */ - v4l2_std_id tv_norm; + gchar *norm; gchar *channel; gulong frequency; @@ -137,8 +137,7 @@ GType gst_v4l2_object_get_type (void); PROP_BRIGHTNESS, \ PROP_CONTRAST, \ PROP_SATURATION, \ - PROP_HUE, \ - PROP_TV_NORM + PROP_HUE /* create/destroy */ GstV4l2Object * gst_v4l2_object_new (GstElement * element, diff --git a/sys/v4l2/gstv4l2sink.c b/sys/v4l2/gstv4l2sink.c index 6b3da4e1a8..d938ba3e34 100644 --- a/sys/v4l2/gstv4l2sink.c +++ b/sys/v4l2/gstv4l2sink.c @@ -55,7 +55,6 @@ #include "gstv4l2colorbalance.h" -#include "gstv4l2tuner.h" #ifdef HAVE_XVIDEO #include "gstv4l2xoverlay.h" #endif @@ -92,7 +91,6 @@ enum GST_IMPLEMENT_V4L2_PROBE_METHODS (GstV4l2SinkClass, gst_v4l2sink); GST_IMPLEMENT_V4L2_COLOR_BALANCE_METHODS (GstV4l2Sink, gst_v4l2sink); -GST_IMPLEMENT_V4L2_TUNER_METHODS (GstV4l2Sink, gst_v4l2sink); #ifdef HAVE_XVIDEO GST_IMPLEMENT_V4L2_XOVERLAY_METHODS (GstV4l2Sink, gst_v4l2sink); #endif @@ -103,13 +101,14 @@ gst_v4l2sink_iface_supported (GstImplementsInterface * iface, GType iface_type) { GstV4l2Object *v4l2object = GST_V4L2SINK (iface)->v4l2object; - g_assert (iface_type == GST_TYPE_TUNER || #ifdef HAVE_XVIDEO - iface_type == GST_TYPE_X_OVERLAY || - iface_type == GST_TYPE_NAVIGATION || -#endif + g_assert (iface_type == GST_TYPE_X_OVERLAY || iface_type == GST_TYPE_COLOR_BALANCE || iface_type == GST_TYPE_VIDEO_ORIENTATION); +#else + g_assert (iface_type == GST_TYPE_COLOR_BALANCE || + iface_type == GST_TYPE_VIDEO_ORIENTATION); +#endif if (v4l2object->video_fd == -1) return FALSE; @@ -151,11 +150,6 @@ gst_v4l2sink_init_interfaces (GType type) NULL, NULL, }; - static const GInterfaceInfo v4l2_tuner_info = { - (GInterfaceInitFunc) gst_v4l2sink_tuner_interface_init, - NULL, - NULL, - }; #ifdef HAVE_XVIDEO static const GInterfaceInfo v4l2_xoverlay_info = { (GInterfaceInitFunc) gst_v4l2sink_xoverlay_interface_init, @@ -186,7 +180,6 @@ gst_v4l2sink_init_interfaces (GType type) g_type_add_interface_static (type, GST_TYPE_IMPLEMENTS_INTERFACE, &v4l2iface_info); - g_type_add_interface_static (type, GST_TYPE_TUNER, &v4l2_tuner_info); #ifdef HAVE_XVIDEO g_type_add_interface_static (type, GST_TYPE_X_OVERLAY, &v4l2_xoverlay_info); g_type_add_interface_static (type, diff --git a/sys/v4l2/gstv4l2tuner.c b/sys/v4l2/gstv4l2tuner.c index a805396cac..c5f84426bd 100644 --- a/sys/v4l2/gstv4l2tuner.c +++ b/sys/v4l2/gstv4l2tuner.c @@ -237,6 +237,7 @@ gst_v4l2_tuner_set_norm (GstV4l2Object * v4l2object, GstTunerNorm * norm) GstTunerNorm * gst_v4l2_tuner_get_norm (GstV4l2Object * v4l2object) { + GList *item; v4l2_std_id norm; /* assert that we're opened and that we're using a known item */ @@ -244,14 +245,6 @@ gst_v4l2_tuner_get_norm (GstV4l2Object * v4l2object) gst_v4l2_get_norm (v4l2object, &norm); - return gst_v4l2_tuner_get_norm_by_std_id (v4l2object, norm); -} - -GstTunerNorm * -gst_v4l2_tuner_get_norm_by_std_id (GstV4l2Object * v4l2object, v4l2_std_id norm) -{ - GList *item; - for (item = v4l2object->norms; item != NULL; item = item->next) { if (norm & GST_V4L2_TUNER_NORM (item->data)->index) return (GstTunerNorm *) item->data; @@ -260,20 +253,6 @@ gst_v4l2_tuner_get_norm_by_std_id (GstV4l2Object * v4l2object, v4l2_std_id norm) return NULL; } -v4l2_std_id -gst_v4l2_tuner_get_std_id_by_norm (GstV4l2Object * v4l2object, - GstTunerNorm * norm) -{ - GList *item; - - for (item = v4l2object->norms; item != NULL; item = item->next) { - if (norm == GST_TUNER_NORM (item->data)) - return GST_V4L2_TUNER_NORM (item->data)->index; - } - - return 0; -} - void gst_v4l2_tuner_set_frequency_and_notify (GstV4l2Object * v4l2object, GstTunerChannel * channel, gulong frequency) diff --git a/sys/v4l2/gstv4l2tuner.h b/sys/v4l2/gstv4l2tuner.h index 699ca87e10..bf9ff94da5 100644 --- a/sys/v4l2/gstv4l2tuner.h +++ b/sys/v4l2/gstv4l2tuner.h @@ -94,11 +94,6 @@ void gst_v4l2_tuner_set_norm_and_notify (GstV4l2Object * v4l2o GstTunerNorm* gst_v4l2_tuner_get_norm (GstV4l2Object * v4l2object); gboolean gst_v4l2_tuner_set_norm (GstV4l2Object * v4l2object, GstTunerNorm * norm); -GstTunerNorm* gst_v4l2_tuner_get_norm_by_std_id (GstV4l2Object * v4l2object, - v4l2_std_id norm); -v4l2_std_id gst_v4l2_tuner_get_std_id_by_norm (GstV4l2Object * v4l2object, - GstTunerNorm * norm); - /* frequency */ void gst_v4l2_tuner_set_frequency_and_notify (GstV4l2Object * v4l2object, GstTunerChannel * channel, diff --git a/sys/v4l2/v4l2_calls.c b/sys/v4l2/v4l2_calls.c index 74b2e0adf2..b4feef3eb7 100644 --- a/sys/v4l2/v4l2_calls.c +++ b/sys/v4l2/v4l2_calls.c @@ -226,9 +226,6 @@ gst_v4l2_fill_lists (GstV4l2Object * v4l2object) standard.frameperiod.denominator, standard.frameperiod.numerator); v4l2norm->index = standard.id; - GST_DEBUG_OBJECT (v4l2object->element, "index=%08x, label=%s", - (unsigned int) v4l2norm->index, norm->label); - v4l2object->norms = g_list_prepend (v4l2object->norms, (gpointer) norm); } v4l2object->norms = g_list_reverse (v4l2object->norms); From b0eb10cc533a3f88a5ed98feff4b88d4eae69cea Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Sun, 2 Jan 2011 11:37:14 +0000 Subject: [PATCH 110/351] v4l2: mark v4l2sink as experimental and build only if --enable-experimental is passed It's not really of 'good' quality yet, but there's a lot of code shared with v4l2src, so not so easy to move it elswhere. https://bugzilla.gnome.org/show_bug.cgi?id=612244 --- sys/v4l2/Makefile.am | 5 ++++- sys/v4l2/gstv4l2.c | 4 ++++ sys/v4l2/gstv4l2bufferpool.c | 4 ++++ sys/v4l2/v4l2_calls.c | 7 +++++++ 4 files changed, 19 insertions(+), 1 deletion(-) diff --git a/sys/v4l2/Makefile.am b/sys/v4l2/Makefile.am index 22cfec4dd3..ab996a15a8 100644 --- a/sys/v4l2/Makefile.am +++ b/sys/v4l2/Makefile.am @@ -13,13 +13,16 @@ libgstvideo4linux2_la_SOURCES = gstv4l2.c \ gstv4l2object.c \ gstv4l2bufferpool.c \ gstv4l2src.c \ - gstv4l2sink.c \ gstv4l2tuner.c \ gstv4l2vidorient.c \ v4l2_calls.c \ v4l2src_calls.c \ $(xv_source) +if BUILD_EXPERIMENTAL +libgstvideo4linux2_la_SOURCES += gstv4l2sink.c +endif + libgstvideo4linux2_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) \ $(GST_BASE_CFLAGS) \ $(GST_CONTROLLER_CFLAGS) \ diff --git a/sys/v4l2/gstv4l2.c b/sys/v4l2/gstv4l2.c index 2599f77286..4a7056fd52 100644 --- a/sys/v4l2/gstv4l2.c +++ b/sys/v4l2/gstv4l2.c @@ -32,7 +32,9 @@ #include "gstv4l2object.h" #include "gstv4l2src.h" +#ifdef HAVE_EXPERIMENTAL #include "gstv4l2sink.h" +#endif /* #include "gstv4l2jpegsrc.h" */ /* #include "gstv4l2mjpegsrc.h" */ /* #include "gstv4l2mjpegsink.h" */ @@ -52,8 +54,10 @@ plugin_init (GstPlugin * plugin) if (!gst_element_register (plugin, "v4l2src", GST_RANK_PRIMARY, GST_TYPE_V4L2SRC) || +#ifdef HAVE_EXPERIMENTAL !gst_element_register (plugin, "v4l2sink", GST_RANK_NONE, GST_TYPE_V4L2SINK) || +#endif /* !gst_element_register (plugin, "v4l2jpegsrc", */ /* GST_RANK_NONE, GST_TYPE_V4L2JPEGSRC) || */ /* !gst_element_register (plugin, "v4l2mjpegsrc", */ diff --git a/sys/v4l2/gstv4l2bufferpool.c b/sys/v4l2/gstv4l2bufferpool.c index a8e7e79fe1..505689a66e 100644 --- a/sys/v4l2/gstv4l2bufferpool.c +++ b/sys/v4l2/gstv4l2bufferpool.c @@ -34,7 +34,9 @@ #include #include "gstv4l2src.h" +#ifdef HAVE_EXPERIMENTAL #include "gstv4l2sink.h" +#endif #include "v4l2_calls.h" #include "gst/gst-i18n-plugin.h" @@ -290,8 +292,10 @@ get_v4l2_object (GstElement * v4l2elem) GstV4l2Object *v4l2object = NULL; if (GST_IS_V4L2SRC (v4l2elem)) { v4l2object = (GST_V4L2SRC (v4l2elem))->v4l2object; +#ifdef HAVE_EXPERIMENTAL } else if (GST_IS_V4L2SINK (v4l2elem)) { v4l2object = (GST_V4L2SINK (v4l2elem))->v4l2object; +#endif } else { GST_ERROR_OBJECT (v4l2elem, "unknown v4l2 element"); } diff --git a/sys/v4l2/v4l2_calls.c b/sys/v4l2/v4l2_calls.c index b4feef3eb7..e9d306999f 100644 --- a/sys/v4l2/v4l2_calls.c +++ b/sys/v4l2/v4l2_calls.c @@ -46,7 +46,10 @@ #include "gstv4l2colorbalance.h" #include "gstv4l2src.h" + +#ifdef HAVE_EXPERIMENTAL #include "gstv4l2sink.h" +#endif #include "gst/gst-i18n-plugin.h" @@ -462,9 +465,11 @@ gst_v4l2_open (GstV4l2Object * v4l2object) !(v4l2object->vcap.capabilities & V4L2_CAP_VIDEO_CAPTURE)) goto not_capture; +#ifdef HAVE_EXPERIMENTAL if (GST_IS_V4L2SINK (v4l2object->element) && !(v4l2object->vcap.capabilities & V4L2_CAP_VIDEO_OUTPUT)) goto not_output; +#endif /* create enumerations, posts errors. */ if (!gst_v4l2_fill_lists (v4l2object)) @@ -510,6 +515,7 @@ not_capture: ("Capabilities: 0x%x", v4l2object->vcap.capabilities)); goto error; } +#ifdef HAVE_EXPERIMENTAL not_output: { GST_ELEMENT_ERROR (v4l2object->element, RESOURCE, NOT_FOUND, @@ -518,6 +524,7 @@ not_output: ("Capabilities: 0x%x", v4l2object->vcap.capabilities)); goto error; } +#endif error: { if (GST_V4L2_IS_OPEN (v4l2object)) { From 90ee1e400a48dee19b746e0e60b5f7efcaa515b7 Mon Sep 17 00:00:00 2001 From: Andy Wingo Date: Mon, 18 Mar 2002 04:41:35 +0000 Subject: [PATCH 111/351] s/gst_element_install_std_props/gst_element_class_install_std_props/ -- it just makes more sense that way added jack ... Original commit message from CVS: * s/gst_element_install_std_props/gst_element_class_install_std_props/ -- it just makes more sense that way * added jack element, doesn't quite work right yet but i didn't want to lose the work -- it does build, register, and attempt to run though * imposed some restrictions on the naming of request pads to better allow for reverse parsing * added '%s' to reverse parsing * added new bin flag to indicate that it is self-iterating, and some lame code in gst-launch to test it out * fixen on launch-gui * added pkg-config stuff for the editor's libs --- ext/jack/Makefile.am | 12 + ext/jack/README | 4 + ext/jack/gstjack.c | 593 +++++++++++++++++++++++++++++++++++++++++++ ext/jack/gstjack.h | 115 +++++++++ 4 files changed, 724 insertions(+) create mode 100644 ext/jack/Makefile.am create mode 100644 ext/jack/README create mode 100644 ext/jack/gstjack.c create mode 100644 ext/jack/gstjack.h diff --git a/ext/jack/Makefile.am b/ext/jack/Makefile.am new file mode 100644 index 0000000000..97f3f6f0c9 --- /dev/null +++ b/ext/jack/Makefile.am @@ -0,0 +1,12 @@ +plugindir = $(libdir)/gst + +plugin_LTLIBRARIES = libgstjack.la + +libgstjack_la_SOURCES = gstjack.c +libgstjack_la_CFLAGS = $(GST_CFLAGS) $(JACK_CFLAGS) +libgstjack_la_LIBADD = $(GST_LIBS) $(JACK_LIBS) +libgstjack_la_LDFLAGS = @GST_PLUGIN_LDFLAGS@ + +noinst_HEADERS = gstjack.h + +EXTRA_DIST = README diff --git a/ext/jack/README b/ext/jack/README new file mode 100644 index 0000000000..2bb97bebad --- /dev/null +++ b/ext/jack/README @@ -0,0 +1,4 @@ +to be written, la dee da + +jackit.sf.net + diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c new file mode 100644 index 0000000000..82da2aeecd --- /dev/null +++ b/ext/jack/gstjack.c @@ -0,0 +1,593 @@ +/* + Copyright (C) 2002 Andy Wingo + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU General Public + License as published by the Free Software Foundation; either + version 2 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + General Public License for more details. + + You should have received a copy of the GNU General Public + License along with this library; if not, write to the Free + Software Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. +*/ + +#include +#include "gstjack.h" +#include + + +/* TODO: + + this element is still nonfunctional + + - work out the src side (caps setting, etc) + + future core TODO: + - make a jack clock provider + - add GST_ELEMENT_FIXED_DATA_RATE, GST_ELEMENT_QOS, + GST_ELEMENT_CHANGES_DATA_RATE element flags, and make the scheduler + sensitive to them +*/ + + +static GstElementDetails gst_jack_sink_details = { + "Jack Sink", + "Sink/Audio", + "Output to a Jack processing network: see README for more info", + VERSION, + "Andy Wingo ", + "(C) 2002 " +}; + +static GstElementDetails gst_jack_src_details = { + "Jack Src", + "Source/Audio", + "Input from a Jack processing network: see README for more info", + VERSION, + "Andy Wingo ", + "(C) 2002", +}; + + +static GstElement *parent_class = NULL; + +static void gst_jack_init(GstJack *this); +static void gst_jack_class_init(GstJackClass *klass); + +static GstPadTemplate *gst_jack_src_request_pad_factory(); +static GstPadTemplate *gst_jack_sink_request_pad_factory(); + +static GstPad* gst_jack_request_new_pad (GstElement *element, GstPadTemplate *templ, const + gchar *name); + +static void gst_jack_set_property(GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec); +static void gst_jack_get_property(GObject *object, guint prop_id, GValue *value, GParamSpec *pspec); +static GstElementStateReturn gst_jack_change_state(GstElement *element); +static GstPadConnectReturn gst_jack_connect (GstPad *pad, GstCaps *caps); + +static void gst_jack_loop (GstElement *element); + +/* jack callbacks */ +static int process (nframes_t nframes, void *arg); +static int buffer_size (nframes_t nframes, void *arg); +static int sample_rate (nframes_t nframes, void *arg); +static void shutdown (void *arg); + + +enum { + ARG_0, + ARG_PORT_NAME_PREFIX, +}; + + +GType +gst_jack_get_type (void) +{ + static GType jack_type = 0; + + if (!jack_type) { + static const GTypeInfo jack_info = { + sizeof(GstJackClass), + NULL, + NULL, + NULL, + NULL, + NULL, + sizeof(GstJack), + 0, + NULL, + }; + jack_type = g_type_register_static (GST_TYPE_ELEMENT, "GstJack", &jack_info, 0); + } + return jack_type; +} + +GType +gst_jack_sink_get_type (void) +{ + static GType jack_type = 0; + + if (!jack_type) { + static const GTypeInfo jack_info = { + sizeof(GstJackClass), + NULL, + NULL, + (GClassInitFunc)gst_jack_class_init, + NULL, + NULL, + sizeof(GstJack), + 0, + (GInstanceInitFunc)gst_jack_init, + }; + jack_type = g_type_register_static (GST_TYPE_JACK, "GstJackSink", &jack_info, 0); + } + return jack_type; +} + +GType +gst_jack_src_get_type (void) +{ + static GType jack_type = 0; + + if (!jack_type) { + static const GTypeInfo jack_info = { + sizeof(GstJackClass), + NULL, + NULL, + (GClassInitFunc)gst_jack_class_init, + NULL, + NULL, + sizeof(GstJack), + 0, + (GInstanceInitFunc)gst_jack_init, + }; + jack_type = g_type_register_static (GST_TYPE_JACK, "GstJackSrc", &jack_info, 0); + } + return jack_type; +} + +static GstPadTemplate* +gst_jack_src_request_pad_factory(void) +{ + static GstPadTemplate *template = NULL; + + if (!template) + template = gst_padtemplate_new("%s", GST_PAD_SRC, GST_PAD_REQUEST, + gst_caps_new("src", "audio/raw", + GST_AUDIO_FLOAT_MONO_PAD_TEMPLATE_PROPS), + NULL); + + return template; +} + +static GstPadTemplate* +gst_jack_sink_request_pad_factory(void) +{ + static GstPadTemplate *template = NULL; + + if (!template) + template = gst_padtemplate_new("%s", GST_PAD_SINK, GST_PAD_REQUEST, + gst_caps_new("sink", "audio/raw", + GST_AUDIO_FLOAT_MONO_PAD_TEMPLATE_PROPS), + NULL); + + return template; +} + +static void +gst_jack_class_init(GstJackClass *klass) +{ + GObjectClass *object_class; + GstElementClass *element_class; + + object_class = (GObjectClass *)klass; + element_class = (GstElementClass *)klass; + + if (parent_class == NULL) + parent_class = g_type_class_ref(GST_TYPE_ELEMENT); + + object_class->get_property = gst_jack_get_property; + object_class->set_property = gst_jack_set_property; + + g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_PORT_NAME_PREFIX, + g_param_spec_string("port-name-prefix","Port name prefix", + "String to prepend to jack port names", + "gst-", + G_PARAM_READWRITE | G_PARAM_CONSTRUCT)); + + element_class->change_state = gst_jack_change_state; + + element_class->request_new_pad = gst_jack_request_new_pad; +} + +static void +gst_jack_init(GstJack *this) +{ + if (G_OBJECT_TYPE (this) == GST_TYPE_JACK_SRC) { + this->direction = GST_PAD_SRC; + } else if (G_OBJECT_TYPE (this) == GST_TYPE_JACK_SINK) { + this->direction = GST_PAD_SINK; + } else { + g_assert_not_reached (); + } + + gst_element_set_loop_function(GST_ELEMENT(this), gst_jack_loop); +} + +static GstPad* +gst_jack_request_new_pad (GstElement *element, GstPadTemplate *templ, const gchar *name) +{ + GstJack *this; + gchar *newname; + GList *l, **pad_list; + GstJackPad *pad; + + g_return_val_if_fail ((this = GST_JACK (element)), NULL); + + if (!this->client) + pad_list = &this->pads; + else if (this->direction == GST_PAD_SRC) + pad_list = &this->client->src_pads; + else + pad_list = &this->client->src_pads; + + if (name) { + l = *pad_list; + while (l) { + if (strcmp (GST_JACK_PAD(l)->name, name) == 0) { + g_warning("requested port name %s already in use.", name); + return NULL; + } + l = l->next; + } + newname = g_strdup (name); + } else { + newname = g_strdup ("alsa_pcm:out_1"); + } + + pad = g_new0(GstJackPad, 1); + pad->name = g_strdup_printf ("%s%d", this->port_name_prefix, this->client->default_new_port_number++); + pad->peer_name = newname; + pad->pad = gst_pad_new_from_template (templ, newname); + gst_element_add_pad (GST_ELEMENT (this), pad->pad); + gst_pad_set_connect_function (pad->pad, gst_jack_connect); + + if (this->client) + *pad_list = g_list_append (*pad_list, pad); +// this->pads = g_list_append (this->pads, pad); + + return pad->pad; +} + +static void +gst_jack_set_property(GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec) +{ + GstJack *this; + + this = (GstJack *)object; + switch (prop_id) { + case ARG_PORT_NAME_PREFIX: + if (this->port_name_prefix) + g_free (this->port_name_prefix); + this->port_name_prefix = g_strdup (g_value_get_string (value)); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + return; + } +} + +static void +gst_jack_get_property(GObject *object, guint prop_id, GValue *value, GParamSpec *pspec) +{ + GstJack *this; + + this = (GstJack *)object; + + switch (prop_id) { + case ARG_PORT_NAME_PREFIX: + g_value_set_string (value, this->port_name_prefix); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static GstElementStateReturn +gst_jack_change_state (GstElement *element) +{ + GstJack *this; + GstJackClient *client; + GstBin *bin; + GList *l = NULL, **pads; + enum JackPortFlags flags; + GstJackPad *pad; + + g_return_val_if_fail (element != NULL, FALSE); + this = GST_JACK (element); + client = this->client; + + switch (GST_STATE_PENDING (element)) { + case GST_STATE_NULL: + if (client) { + jack_client_close (client->client); + } + + break; + + case GST_STATE_READY: + if (!this->client) { + if (!(bin = gst_element_get_managing_bin (element))) { + g_warning ("jack element %s cannot be brought to READY state without a managing bin", + GST_OBJECT_NAME (GST_OBJECT (element))); + return GST_STATE_FAILURE; + } + + client = g_object_get_data (G_OBJECT (bin), "gst-jack-client"); + + if (!client) { + client = g_new0 (GstJackClient, 1); + if (!(client->client = jack_client_new ("gst-jack"))) { + g_warning ("jack server not running?"); + return GST_STATE_FAILURE; + } + + jack_set_process_callback (client->client, process, client); + jack_set_buffer_size_callback (client->client, buffer_size, client); + jack_set_sample_rate_callback (client->client, sample_rate, client); + jack_on_shutdown (client->client, shutdown, client); + + client->refcount = 1; + g_object_set_data (G_OBJECT (bin), "gst-jack-client", client); + client->manager = bin; + GST_FLAG_SET (GST_OBJECT (bin), GST_BIN_SELF_ITERATING); + } else { + client->refcount++; + } + this->client = client; + + /* fixme: verify that all names are unique */ + l = this->pads; + pads = (this->direction == GST_PAD_SRC) ? &client->src_pads : &client->sink_pads; + while (l) { + pad = GST_JACK_PAD (l); + *pads = g_list_append (*pads, pad); + l = g_list_next (l); + } + } + + /* fixme: there are a *lot* of problems here */ + if (GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_OPEN)) { + l = this->pads; + while (l) { + jack_port_unregister (client->client, GST_JACK_PAD (l)->port); + l = g_list_next (l); + } + GST_FLAG_UNSET (GST_OBJECT (this), GST_JACK_OPEN); + } + break; + + case GST_STATE_PAUSED: + g_assert (client); + + if (!GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_OPEN)) { + l = this->pads; + while (l) { + pad = GST_JACK_PAD (l); + + if (this->direction == GST_PAD_SINK) + flags = JackPortIsOutput; + else + flags = JackPortIsInput; + + pad->port = jack_port_register (client->client, pad->name, JACK_DEFAULT_AUDIO_TYPE, flags, 0); + if (jack_connect (client->client, pad->peer_name, jack_port_name (pad->port))) { + g_warning ("could not connect %s and %s", pad->peer_name, jack_port_name (pad->port)); + return GST_STATE_FAILURE; + } + + l = g_list_next (l); + } + GST_FLAG_SET (GST_OBJECT (this), GST_JACK_OPEN); + } + + if (GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_ACTIVE)) { + jack_deactivate (client->client); + GST_FLAG_UNSET (GST_OBJECT (this), GST_JACK_ACTIVE); + } + break; + case GST_STATE_PLAYING: + if (!GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_ACTIVE)) { + jack_activate (client->client); + GST_FLAG_SET (GST_OBJECT (this), GST_JACK_ACTIVE); + } + break; + } + + if (GST_ELEMENT_CLASS (parent_class)->change_state) + return GST_ELEMENT_CLASS (parent_class)->change_state (element); + + return GST_STATE_SUCCESS; +} + +static GstPadConnectReturn +gst_jack_connect (GstPad *pad, GstCaps *caps) +{ + GstJack *this; + + this = GST_JACK (gst_pad_get_parent (pad)); + g_return_val_if_fail (this != NULL, GST_PAD_CONNECT_REFUSED); + g_return_val_if_fail (GST_IS_JACK (this), GST_PAD_CONNECT_REFUSED); + + if (GST_CAPS_IS_FIXED (caps)) { + if (this->client && gst_caps_get_int (caps, "rate") != this->client->rate) + return GST_PAD_CONNECT_REFUSED; + + return GST_PAD_CONNECT_OK; + } + + return GST_PAD_CONNECT_DELAYED; +} + +static void +gst_jack_loop (GstElement *element) +{ + GstJack *this; + GList *pads; + gint len; + gchar *peeked; + gint avail; + GstEvent *event; + GstJackPad *pad; + GstBuffer *buffer; + + this = GST_JACK (element); + + g_return_if_fail(this != NULL); + len = this->client->nframes * sizeof (sample_t); + + do { + pads = this->pads; + while (pads) { + pad = GST_JACK_PAD (pads); + + if (this->direction == GST_PAD_SINK) { + if (pad->bs) + pad->bs = gst_bytestream_new (pad->pad); + + if (!(peeked = gst_bytestream_peek_bytes (pad->bs, len))) { + gst_bytestream_get_status(pad->bs, &avail, &event); + if (event) { + g_warning("got an event on jacksink"); + if (GST_EVENT_TYPE (event) == GST_EVENT_EOS) { + /* really, we should just cut this pad out of the graph. let + * me know when this is needed ;) + * also, for sample accuracy etc, we should play avail + * bytes, but hey. */ + gst_element_set_eos(GST_ELEMENT(this)); + gst_event_free(event); + } + } else { + /* the element at the top of the chain did not emit an eos + * event. this is a Bug(tm) */ + g_assert_not_reached(); + } + } + + memcpy (pad->data, peeked, len); + gst_bytestream_flush (pad->bs, len); + } else { + buffer = gst_buffer_new (); + GST_BUFFER_DATA (buffer) = pad->data; + GST_BUFFER_SIZE (buffer) = len; + GST_BUFFER_MAXSIZE (buffer) = len; + + gst_pad_push (pad->pad, buffer); + } + pads = g_list_next (pads); + } + + gst_element_yield (element); + } while (TRUE); +} + +/* jack callbacks */ + +static int +process (nframes_t nframes, void *arg) +{ + GstJackClient *client = (GstJackClient*) arg; + GstJackPad *pad; + GList *l; + + g_assert (client); + + l = client->src_pads; + while (l) { + pad = GST_JACK_PAD (l); + pad->data = jack_port_get_buffer (pad->port, nframes); + l = g_list_next (l); + } + + l = client->sink_pads; + while (l) { + pad = GST_JACK_PAD (l); + pad->data = jack_port_get_buffer (pad->port, nframes); + l = g_list_next (l); + } + + client->nframes = nframes; + + if (!gst_bin_iterate (client->manager)) { + g_warning ("bin failed to iterate"); + return -1; + } + + /* that's all folks */ + + return 0; +} + +static int +buffer_size (nframes_t nframes, void *arg) +{ + printf ("the maximum buffer size is now %lu\n", nframes); + return 0; +} + +static int +sample_rate (nframes_t nframes, void *arg) +{ + GstJackClient *client = (GstJackClient*) arg; + printf ("the sample rate is now %lu/sec\n", nframes); + client->rate = nframes; + return 0; +} + +static void +shutdown (void *arg) +{ + GstJackClient *client = (GstJackClient*) arg; + printf ("shutdown\n"); +// gst_element_set_state (GST_ELEMENT (client->manager), GST_STATE_READY); +} + +static gboolean +plugin_init (GModule *module, GstPlugin *plugin) +{ + GstElementFactory *factory; + + if (!gst_library_load ("gstbytestream")) { + gst_info("jack: could not load support library: 'gstbytestream'\n"); + return FALSE; + } + + factory = gst_elementfactory_new ("jacksrc", GST_TYPE_JACK_SRC, &gst_jack_src_details); + g_return_val_if_fail (factory != NULL, FALSE); + gst_elementfactory_add_padtemplate (factory, gst_jack_src_request_pad_factory()); + gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory)); + + factory = gst_elementfactory_new ("jacksink", GST_TYPE_JACK_SINK, &gst_jack_sink_details); + g_return_val_if_fail (factory != NULL, FALSE); + gst_elementfactory_add_padtemplate (factory, gst_jack_sink_request_pad_factory()); + gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory)); + + gst_plugin_set_longname(plugin, "JACK plugin library"); + + return TRUE; +} + +GstPluginDesc plugin_desc = { + GST_VERSION_MAJOR, + GST_VERSION_MINOR, + "jack", + plugin_init +}; + + diff --git a/ext/jack/gstjack.h b/ext/jack/gstjack.h new file mode 100644 index 0000000000..da6940f2fd --- /dev/null +++ b/ext/jack/gstjack.h @@ -0,0 +1,115 @@ +/* + Copyright (C) 2002 Andy Wingo + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU Library General Public + License as published by the Free Software Foundation; either + version 2 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + Library General Public License for more details. + + You should have received a copy of the GNU Library General Public + License along with this library; if not, write to the Free + Software Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. +*/ + +#ifndef __GST_JACK_H__ +#define __GST_JACK_H__ + +#include +#include +#include + + +#define GST_JACK(obj) G_TYPE_CHECK_INSTANCE_CAST(obj, GST_TYPE_JACK, GstJack) +#define GST_JACK_CLASS(klass) G_TYPE_CHECK_CLASS_CAST(klass, GST_TYPE_JACK, GstJackClass) +#define GST_IS_JACK(obj) G_TYPE_CHECK_INSTANCE_TYPE(obj, GST_TYPE_JACK) +#define GST_IS_JACK_CLASS(klass) G_TYPE_CHECK_CLASS_TYPE(klass, GST_TYPE_JACK) +#define GST_TYPE_JACK gst_jack_get_type() + +#define GST_JACK_SINK(obj) G_TYPE_CHECK_INSTANCE_CAST(obj, GST_TYPE_JACK_SINK, GstJack) +#define GST_JACK_SINK_CLASS(klass) G_TYPE_CHECK_CLASS_CAST(klass, GST_TYPE_JACK_SINK, GstJackClass) +#define GST_IS_JACK_SINK(obj) G_TYPE_CHECK_INSTANCE_TYPE(obj, GST_TYPE_JACK_SINK) +#define GST_IS_JACK_SINK_CLASS(klass) G_TYPE_CHECK_CLASS_TYPE(klass, GST_TYPE_JACK_SINK) +#define GST_TYPE_JACK_SINK gst_jack_sink_get_type() + +#define GST_JACK_SRC(obj) G_TYPE_CHECK_INSTANCE_CAST(obj, GST_TYPE_JACK_SRC, GstJack) +#define GST_JACK_SRC_CLASS(klass) G_TYPE_CHECK_CLASS_CAST(klass, GST_TYPE_JACK_SRC, GstJackClass) +#define GST_IS_JACK_SRC(obj) G_TYPE_CHECK_INSTANCE_TYPE(obj, GST_TYPE_JACK_SRC) +#define GST_IS_JACK_SRC_CLASS(klass) G_TYPE_CHECK_CLASS_TYPE(klass, GST_TYPE_JACK_SRC) +#define GST_TYPE_JACK_SRC gst_jack_src_get_type() + +#define GST_JACK_PAD(l) ((GstJackPad*)l->data) /* l is a GList */ + + +typedef struct _GstJack GstJack; +typedef struct _GstJackClass GstJackClass; +typedef GstJack GstJackSink; +typedef GstJackClass GstJackSinkClass; +typedef GstJack GstJackSrc; +typedef GstJackClass GstJackSrcClass; + + +enum { + GST_JACK_OPEN = GST_ELEMENT_FLAG_LAST, + GST_JACK_ACTIVE, + GST_JACK_FLAG_LAST = GST_ELEMENT_FLAG_LAST + 3, +}; + + +typedef struct { + GstPad *pad; + void *data; + const gchar *name; + const gchar *peer_name; + GstByteStream *bs; + jack_port_t *port; +} GstJackPad; + +typedef struct { + GstBin *manager; + + jack_client_t *client; + gint default_new_port_number; + + /* lists of GstJackPads */ + GList *sink_pads; + GList *src_pads; + + gchar *client_name; + + guint rate; + nframes_t nframes; + + guint refcount; +} GstJackClient; + +struct _GstJack { + GstElement element; + + /* list of GstJackPads */ + GList *pads; + + /* for convenience */ + GstPadDirection direction; + + gchar *port_name_prefix; + + /* there is exactly one client per managing bin */ + GstJackClient *client; +}; + +struct _GstJackClass { + GstElementClass parent_class; +}; + + +GType gst_jack_get_type (void); +GType gst_jack_sink_get_type (void); +GType gst_jack_src_get_type (void); + + +#endif /* __GST_JACK_H__ */ From 3b8cad1d4958694d8c14cb8858f3eb011eb5545b Mon Sep 17 00:00:00 2001 From: Andy Wingo Date: Tue, 19 Mar 2002 01:39:43 +0000 Subject: [PATCH 112/351] s/@GST_PLUGIN_LDFLAGS@/$(GST_PLUGIN_LDFLAGS)/ @-substitued variables variables are defined as make variables automagi... Original commit message from CVS: s/@GST_PLUGIN_LDFLAGS@/$(GST_PLUGIN_LDFLAGS)/ @-substitued variables variables are defined as make variables automagically, and this gives the user the freedom to say make GST_PLUGIN_LDFLAGS=-myflag --- ext/jack/Makefile.am | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/ext/jack/Makefile.am b/ext/jack/Makefile.am index 97f3f6f0c9..fc5c110d7e 100644 --- a/ext/jack/Makefile.am +++ b/ext/jack/Makefile.am @@ -5,7 +5,7 @@ plugin_LTLIBRARIES = libgstjack.la libgstjack_la_SOURCES = gstjack.c libgstjack_la_CFLAGS = $(GST_CFLAGS) $(JACK_CFLAGS) libgstjack_la_LIBADD = $(GST_LIBS) $(JACK_LIBS) -libgstjack_la_LDFLAGS = @GST_PLUGIN_LDFLAGS@ +libgstjack_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) noinst_HEADERS = gstjack.h From fe8058a641138f874760ae51e37053abdaa42750 Mon Sep 17 00:00:00 2001 From: Andy Wingo Date: Tue, 19 Mar 2002 04:10:06 +0000 Subject: [PATCH 113/351] removal of //-style comments don't link plugins to core libs -- the versioning is done internally to the plugins with... Original commit message from CVS: * removal of //-style comments * don't link plugins to core libs -- the versioning is done internally to the plugins with the plugin_info struct, and symbol resolution is lazy, so we can always know if a plugin can be loaded by the plugin_info data. in theory. --- ext/jack/Makefile.am | 2 +- ext/jack/gstjack.c | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/ext/jack/Makefile.am b/ext/jack/Makefile.am index fc5c110d7e..7c1b5c2376 100644 --- a/ext/jack/Makefile.am +++ b/ext/jack/Makefile.am @@ -4,7 +4,7 @@ plugin_LTLIBRARIES = libgstjack.la libgstjack_la_SOURCES = gstjack.c libgstjack_la_CFLAGS = $(GST_CFLAGS) $(JACK_CFLAGS) -libgstjack_la_LIBADD = $(GST_LIBS) $(JACK_LIBS) +libgstjack_la_LIBADD = $(JACK_LIBS) libgstjack_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) noinst_HEADERS = gstjack.h diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 82da2aeecd..816c503691 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -259,7 +259,7 @@ gst_jack_request_new_pad (GstElement *element, GstPadTemplate *templ, const gcha if (this->client) *pad_list = g_list_append (*pad_list, pad); -// this->pads = g_list_append (this->pads, pad); +/* this->pads = g_list_append (this->pads, pad); */ return pad->pad; } @@ -555,7 +555,7 @@ shutdown (void *arg) { GstJackClient *client = (GstJackClient*) arg; printf ("shutdown\n"); -// gst_element_set_state (GST_ELEMENT (client->manager), GST_STATE_READY); +/* gst_element_set_state (GST_ELEMENT (client->manager), GST_STATE_READY); */ } static gboolean From 2f2a02b8a9ff6b9fb9d63da2d659c95c1bbbe94e Mon Sep 17 00:00:00 2001 From: Andy Wingo Date: Wed, 20 Mar 2002 21:45:04 +0000 Subject: [PATCH 114/351] s/Gnome-Streamer/GStreamer/ Original commit message from CVS: s/Gnome-Streamer/GStreamer/ --- ext/jack/gstjack.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 816c503691..13a29b10a9 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -251,7 +251,7 @@ gst_jack_request_new_pad (GstElement *element, GstPadTemplate *templ, const gcha } pad = g_new0(GstJackPad, 1); - pad->name = g_strdup_printf ("%s%d", this->port_name_prefix, this->client->default_new_port_number++); + pad->name = g_strdup_printf ("%s%d", this->port_name_prefix, 1); /* fixme :) */ pad->peer_name = newname; pad->pad = gst_pad_new_from_template (templ, newname); gst_element_add_pad (GST_ELEMENT (this), pad->pad); @@ -261,6 +261,7 @@ gst_jack_request_new_pad (GstElement *element, GstPadTemplate *templ, const gcha *pad_list = g_list_append (*pad_list, pad); /* this->pads = g_list_append (this->pads, pad); */ + g_print ("returning from request_new_pad, pad %s created, to connect to %s\n", pad->name, pad->peer_name); return pad->pad; } From b3ce1be5546823b7d2c3cee5aa19ad5bcbdfab1f Mon Sep 17 00:00:00 2001 From: Andy Wingo Date: Sat, 30 Mar 2002 19:31:13 +0000 Subject: [PATCH 115/351] add notify back to filesrc, it's needed for MVC applications remove notify printouts from gst-launch cleanup in gst-p... Original commit message from CVS: * add notify back to filesrc, it's needed for MVC applications * remove notify printouts from gst-launch * cleanup in gst-plugins configure.ac * some jack updates * remove SELF_ITERATING flag in favor of SEF_SCHEDULABLE (not a clear name, but it's what we have for the moment) * improve parsing of request pad names, no more sscanf * fixes to the fastscheduler Makefile.am --- ext/jack/gstjack.c | 27 ++++++++++++++++++++++----- 1 file changed, 22 insertions(+), 5 deletions(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 13a29b10a9..82bb4ce5df 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -251,15 +251,14 @@ gst_jack_request_new_pad (GstElement *element, GstPadTemplate *templ, const gcha } pad = g_new0(GstJackPad, 1); + pad->name = g_strdup_printf ("%s%d", this->port_name_prefix, 1); /* fixme :) */ pad->peer_name = newname; pad->pad = gst_pad_new_from_template (templ, newname); gst_element_add_pad (GST_ELEMENT (this), pad->pad); gst_pad_set_connect_function (pad->pad, gst_jack_connect); - if (this->client) - *pad_list = g_list_append (*pad_list, pad); -/* this->pads = g_list_append (this->pads, pad); */ + this->pads = g_list_append (this->pads, pad); g_print ("returning from request_new_pad, pad %s created, to connect to %s\n", pad->name, pad->peer_name); return pad->pad; @@ -316,13 +315,16 @@ gst_jack_change_state (GstElement *element) switch (GST_STATE_PENDING (element)) { case GST_STATE_NULL: + g_message ("jack: NULL state"); if (client) { + g_message ("jack: closing client"); jack_client_close (client->client); } break; case GST_STATE_READY: + g_message ("jack: READY"); if (!this->client) { if (!(bin = gst_element_get_managing_bin (element))) { g_warning ("jack element %s cannot be brought to READY state without a managing bin", @@ -333,6 +335,7 @@ gst_jack_change_state (GstElement *element) client = g_object_get_data (G_OBJECT (bin), "gst-jack-client"); if (!client) { + g_message ("jack: making new client"); client = g_new0 (GstJackClient, 1); if (!(client->client = jack_client_new ("gst-jack"))) { g_warning ("jack server not running?"); @@ -347,8 +350,9 @@ gst_jack_change_state (GstElement *element) client->refcount = 1; g_object_set_data (G_OBJECT (bin), "gst-jack-client", client); client->manager = bin; - GST_FLAG_SET (GST_OBJECT (bin), GST_BIN_SELF_ITERATING); + GST_FLAG_SET (GST_OBJECT (bin), GST_BIN_SELF_SCHEDULABLE); } else { + g_message ("jack: refcounting existing client"); client->refcount++; } this->client = client; @@ -358,6 +362,7 @@ gst_jack_change_state (GstElement *element) pads = (this->direction == GST_PAD_SRC) ? &client->src_pads : &client->sink_pads; while (l) { pad = GST_JACK_PAD (l); + g_message ("jack: appending pad %s:%s to list", pad->name, pad->peer_name); *pads = g_list_append (*pads, pad); l = g_list_next (l); } @@ -367,6 +372,7 @@ gst_jack_change_state (GstElement *element) if (GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_OPEN)) { l = this->pads; while (l) { + g_message ("jack: unregistering pad %s:%s", GST_JACK_PAD (l)->name, GST_JACK_PAD (l)->peer_name); jack_port_unregister (client->client, GST_JACK_PAD (l)->port); l = g_list_next (l); } @@ -375,6 +381,7 @@ gst_jack_change_state (GstElement *element) break; case GST_STATE_PAUSED: + g_message ("jack: PAUSED"); g_assert (client); if (!GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_OPEN)) { @@ -387,30 +394,38 @@ gst_jack_change_state (GstElement *element) else flags = JackPortIsInput; + g_message ("jack: registering pad %s:%s", pad->name, pad->peer_name); pad->port = jack_port_register (client->client, pad->name, JACK_DEFAULT_AUDIO_TYPE, flags, 0); - if (jack_connect (client->client, pad->peer_name, jack_port_name (pad->port))) { + g_message ("connecting gst jack port %s to jack port %s", jack_port_name (pad->port), pad->peer_name); + if (jack_connect (client->client, jack_port_name (pad->port), pad->peer_name)) { g_warning ("could not connect %s and %s", pad->peer_name, jack_port_name (pad->port)); return GST_STATE_FAILURE; } l = g_list_next (l); } + g_message ("jack: setting OPEN flag"); GST_FLAG_SET (GST_OBJECT (this), GST_JACK_OPEN); } if (GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_ACTIVE)) { + g_message ("jack: deactivating client"); jack_deactivate (client->client); GST_FLAG_UNSET (GST_OBJECT (this), GST_JACK_ACTIVE); } break; case GST_STATE_PLAYING: + g_message ("jack: PLAYING"); if (!GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_ACTIVE)) { + g_message ("jack: activating client"); jack_activate (client->client); GST_FLAG_SET (GST_OBJECT (this), GST_JACK_ACTIVE); } break; } + g_message ("jack: state change finished"); + if (GST_ELEMENT_CLASS (parent_class)->change_state) return GST_ELEMENT_CLASS (parent_class)->change_state (element); @@ -509,6 +524,8 @@ process (nframes_t nframes, void *arg) g_assert (client); + g_message ("jack: process()"); + l = client->src_pads; while (l) { pad = GST_JACK_PAD (l); From 2d8f33b3e4f2932386a0decf5df299cbea9843ac Mon Sep 17 00:00:00 2001 From: Andy Wingo Date: Sat, 30 Mar 2002 21:07:51 +0000 Subject: [PATCH 116/351] alphabetization fixen a jack caps fix Original commit message from CVS: * alphabetization fixen * a jack caps fix --- ext/jack/gstjack.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 82bb4ce5df..a61cdb9a7d 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -436,13 +436,15 @@ static GstPadConnectReturn gst_jack_connect (GstPad *pad, GstCaps *caps) { GstJack *this; + gint rate; this = GST_JACK (gst_pad_get_parent (pad)); g_return_val_if_fail (this != NULL, GST_PAD_CONNECT_REFUSED); g_return_val_if_fail (GST_IS_JACK (this), GST_PAD_CONNECT_REFUSED); if (GST_CAPS_IS_FIXED (caps)) { - if (this->client && gst_caps_get_int (caps, "rate") != this->client->rate) + gst_caps_get_int (caps, "rate", &rate); + if (this->client && rate != this->client->rate) return GST_PAD_CONNECT_REFUSED; return GST_PAD_CONNECT_OK; From a9161e18825d4a07f4fdcd6e5c7607571b1aae51 Mon Sep 17 00:00:00 2001 From: Andy Wingo Date: Thu, 11 Apr 2002 20:42:26 +0000 Subject: [PATCH 117/351] GstPadTemplate <-> gst_pad_template <-> GST_PAD_TEMPLATE same with *factory and typefind. Original commit message from CVS: GstPadTemplate <-> gst_pad_template <-> GST_PAD_TEMPLATE same with *factory and typefind. also, some -Werror fixes. --- ext/jack/gstjack.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index a61cdb9a7d..0567f47390 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -157,7 +157,7 @@ gst_jack_src_request_pad_factory(void) static GstPadTemplate *template = NULL; if (!template) - template = gst_padtemplate_new("%s", GST_PAD_SRC, GST_PAD_REQUEST, + template = gst_pad_template_new("%s", GST_PAD_SRC, GST_PAD_REQUEST, gst_caps_new("src", "audio/raw", GST_AUDIO_FLOAT_MONO_PAD_TEMPLATE_PROPS), NULL); @@ -171,7 +171,7 @@ gst_jack_sink_request_pad_factory(void) static GstPadTemplate *template = NULL; if (!template) - template = gst_padtemplate_new("%s", GST_PAD_SINK, GST_PAD_REQUEST, + template = gst_pad_template_new("%s", GST_PAD_SINK, GST_PAD_REQUEST, gst_caps_new("sink", "audio/raw", GST_AUDIO_FLOAT_MONO_PAD_TEMPLATE_PROPS), NULL); @@ -574,7 +574,7 @@ static void shutdown (void *arg) { GstJackClient *client = (GstJackClient*) arg; - printf ("shutdown\n"); + printf ("shutdown %p\n", client); /* gst_element_set_state (GST_ELEMENT (client->manager), GST_STATE_READY); */ } @@ -588,14 +588,14 @@ plugin_init (GModule *module, GstPlugin *plugin) return FALSE; } - factory = gst_elementfactory_new ("jacksrc", GST_TYPE_JACK_SRC, &gst_jack_src_details); + factory = gst_element_factory_new ("jacksrc", GST_TYPE_JACK_SRC, &gst_jack_src_details); g_return_val_if_fail (factory != NULL, FALSE); - gst_elementfactory_add_padtemplate (factory, gst_jack_src_request_pad_factory()); + gst_element_factory_add_pad_template (factory, gst_jack_src_request_pad_factory()); gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory)); - factory = gst_elementfactory_new ("jacksink", GST_TYPE_JACK_SINK, &gst_jack_sink_details); + factory = gst_element_factory_new ("jacksink", GST_TYPE_JACK_SINK, &gst_jack_sink_details); g_return_val_if_fail (factory != NULL, FALSE); - gst_elementfactory_add_padtemplate (factory, gst_jack_sink_request_pad_factory()); + gst_element_factory_add_pad_template (factory, gst_jack_sink_request_pad_factory()); gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory)); gst_plugin_set_longname(plugin, "JACK plugin library"); From 6ef5499ab50b92ab831e6608878decd048e618ba Mon Sep 17 00:00:00 2001 From: Andy Wingo Date: Tue, 16 Apr 2002 17:14:05 +0000 Subject: [PATCH 118/351] Finally we're on to a proper jack setup, with a specialized bin and elements that can only go in a jack bin. I had to... Original commit message from CVS: Finally we're on to a proper jack setup, with a specialized bin and elements that can only go in a jack bin. I had to fix the parser first to do this, but to run it, the syntax is like so: gst-launch jackbin.( filesrc ! mad ! jacksink ) But of course it's not fully functional yet. Sigh. --- ext/jack/Makefile.am | 2 +- ext/jack/gstjack.c | 192 +++++--------------------------- ext/jack/gstjack.h | 54 +++++---- ext/jack/gstjackbin.c | 250 ++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 311 insertions(+), 187 deletions(-) create mode 100644 ext/jack/gstjackbin.c diff --git a/ext/jack/Makefile.am b/ext/jack/Makefile.am index 7c1b5c2376..981c5a9f77 100644 --- a/ext/jack/Makefile.am +++ b/ext/jack/Makefile.am @@ -2,7 +2,7 @@ plugindir = $(libdir)/gst plugin_LTLIBRARIES = libgstjack.la -libgstjack_la_SOURCES = gstjack.c +libgstjack_la_SOURCES = gstjack.c gstjackbin.c libgstjack_la_CFLAGS = $(GST_CFLAGS) $(JACK_CFLAGS) libgstjack_la_LIBADD = $(JACK_LIBS) libgstjack_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 0567f47390..fc29775068 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -35,6 +35,15 @@ */ +static GstElementDetails gst_jack_bin_details = { + "Jack Bin", + "Bin/Audio", + "Jack processing bin: see README for more info", + VERSION, + "Andy Wingo ", + "(C) 2002 " +}; + static GstElementDetails gst_jack_sink_details = { "Jack Sink", "Sink/Audio", @@ -54,7 +63,7 @@ static GstElementDetails gst_jack_src_details = { }; -static GstElement *parent_class = NULL; +static GstElementClass *parent_class = NULL; static void gst_jack_init(GstJack *this); static void gst_jack_class_init(GstJackClass *klass); @@ -72,12 +81,6 @@ static GstPadConnectReturn gst_jack_connect (GstPad *pad, GstCaps *caps); static void gst_jack_loop (GstElement *element); -/* jack callbacks */ -static int process (nframes_t nframes, void *arg); -static int buffer_size (nframes_t nframes, void *arg); -static int sample_rate (nframes_t nframes, void *arg); -static void shutdown (void *arg); - enum { ARG_0, @@ -229,12 +232,12 @@ gst_jack_request_new_pad (GstElement *element, GstPadTemplate *templ, const gcha g_return_val_if_fail ((this = GST_JACK (element)), NULL); - if (!this->client) + if (!this->bin) pad_list = &this->pads; else if (this->direction == GST_PAD_SRC) - pad_list = &this->client->src_pads; + pad_list = &this->bin->src_pads; else - pad_list = &this->client->src_pads; + pad_list = &this->bin->src_pads; if (name) { l = *pad_list; @@ -303,63 +306,31 @@ static GstElementStateReturn gst_jack_change_state (GstElement *element) { GstJack *this; - GstJackClient *client; - GstBin *bin; GList *l = NULL, **pads; - enum JackPortFlags flags; GstJackPad *pad; g_return_val_if_fail (element != NULL, FALSE); this = GST_JACK (element); - client = this->client; switch (GST_STATE_PENDING (element)) { case GST_STATE_NULL: - g_message ("jack: NULL state"); - if (client) { - g_message ("jack: closing client"); - jack_client_close (client->client); - } - + g_message ("jack client %s: NULL", GST_OBJECT_NAME (GST_OBJECT (this))); break; case GST_STATE_READY: - g_message ("jack: READY"); - if (!this->client) { - if (!(bin = gst_element_get_managing_bin (element))) { - g_warning ("jack element %s cannot be brought to READY state without a managing bin", + g_message ("jack client %s: READY", GST_OBJECT_NAME (GST_OBJECT (this))); + if (!this->bin) { + if (!(this->bin = (GstJackBin*)gst_element_get_managing_bin (element)) + || !GST_IS_JACK_BIN (this->bin)) { + this->bin = NULL; + g_warning ("jack element %s needs to be contained in a jack bin.", GST_OBJECT_NAME (GST_OBJECT (element))); return GST_STATE_FAILURE; } - client = g_object_get_data (G_OBJECT (bin), "gst-jack-client"); - - if (!client) { - g_message ("jack: making new client"); - client = g_new0 (GstJackClient, 1); - if (!(client->client = jack_client_new ("gst-jack"))) { - g_warning ("jack server not running?"); - return GST_STATE_FAILURE; - } - - jack_set_process_callback (client->client, process, client); - jack_set_buffer_size_callback (client->client, buffer_size, client); - jack_set_sample_rate_callback (client->client, sample_rate, client); - jack_on_shutdown (client->client, shutdown, client); - - client->refcount = 1; - g_object_set_data (G_OBJECT (bin), "gst-jack-client", client); - client->manager = bin; - GST_FLAG_SET (GST_OBJECT (bin), GST_BIN_SELF_SCHEDULABLE); - } else { - g_message ("jack: refcounting existing client"); - client->refcount++; - } - this->client = client; - /* fixme: verify that all names are unique */ l = this->pads; - pads = (this->direction == GST_PAD_SRC) ? &client->src_pads : &client->sink_pads; + pads = (this->direction == GST_PAD_SRC) ? &this->bin->src_pads : &this->bin->sink_pads; while (l) { pad = GST_JACK_PAD (l); g_message ("jack: appending pad %s:%s to list", pad->name, pad->peer_name); @@ -367,60 +338,13 @@ gst_jack_change_state (GstElement *element) l = g_list_next (l); } } - - /* fixme: there are a *lot* of problems here */ - if (GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_OPEN)) { - l = this->pads; - while (l) { - g_message ("jack: unregistering pad %s:%s", GST_JACK_PAD (l)->name, GST_JACK_PAD (l)->peer_name); - jack_port_unregister (client->client, GST_JACK_PAD (l)->port); - l = g_list_next (l); - } - GST_FLAG_UNSET (GST_OBJECT (this), GST_JACK_OPEN); - } break; case GST_STATE_PAUSED: - g_message ("jack: PAUSED"); - g_assert (client); - - if (!GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_OPEN)) { - l = this->pads; - while (l) { - pad = GST_JACK_PAD (l); - - if (this->direction == GST_PAD_SINK) - flags = JackPortIsOutput; - else - flags = JackPortIsInput; - - g_message ("jack: registering pad %s:%s", pad->name, pad->peer_name); - pad->port = jack_port_register (client->client, pad->name, JACK_DEFAULT_AUDIO_TYPE, flags, 0); - g_message ("connecting gst jack port %s to jack port %s", jack_port_name (pad->port), pad->peer_name); - if (jack_connect (client->client, jack_port_name (pad->port), pad->peer_name)) { - g_warning ("could not connect %s and %s", pad->peer_name, jack_port_name (pad->port)); - return GST_STATE_FAILURE; - } - - l = g_list_next (l); - } - g_message ("jack: setting OPEN flag"); - GST_FLAG_SET (GST_OBJECT (this), GST_JACK_OPEN); - } - - if (GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_ACTIVE)) { - g_message ("jack: deactivating client"); - jack_deactivate (client->client); - GST_FLAG_UNSET (GST_OBJECT (this), GST_JACK_ACTIVE); - } + g_message ("jack client %s: PAUSED", GST_OBJECT_NAME (GST_OBJECT (this))); break; case GST_STATE_PLAYING: - g_message ("jack: PLAYING"); - if (!GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_ACTIVE)) { - g_message ("jack: activating client"); - jack_activate (client->client); - GST_FLAG_SET (GST_OBJECT (this), GST_JACK_ACTIVE); - } + g_message ("jack client %s: PLAYING", GST_OBJECT_NAME (GST_OBJECT (this))); break; } @@ -444,7 +368,7 @@ gst_jack_connect (GstPad *pad, GstCaps *caps) if (GST_CAPS_IS_FIXED (caps)) { gst_caps_get_int (caps, "rate", &rate); - if (this->client && rate != this->client->rate) + if (this->bin && rate != this->bin->rate) return GST_PAD_CONNECT_REFUSED; return GST_PAD_CONNECT_OK; @@ -468,7 +392,7 @@ gst_jack_loop (GstElement *element) this = GST_JACK (element); g_return_if_fail(this != NULL); - len = this->client->nframes * sizeof (sample_t); + len = this->bin->nframes * sizeof (sample_t); do { pads = this->pads; @@ -515,69 +439,6 @@ gst_jack_loop (GstElement *element) } while (TRUE); } -/* jack callbacks */ - -static int -process (nframes_t nframes, void *arg) -{ - GstJackClient *client = (GstJackClient*) arg; - GstJackPad *pad; - GList *l; - - g_assert (client); - - g_message ("jack: process()"); - - l = client->src_pads; - while (l) { - pad = GST_JACK_PAD (l); - pad->data = jack_port_get_buffer (pad->port, nframes); - l = g_list_next (l); - } - - l = client->sink_pads; - while (l) { - pad = GST_JACK_PAD (l); - pad->data = jack_port_get_buffer (pad->port, nframes); - l = g_list_next (l); - } - - client->nframes = nframes; - - if (!gst_bin_iterate (client->manager)) { - g_warning ("bin failed to iterate"); - return -1; - } - - /* that's all folks */ - - return 0; -} - -static int -buffer_size (nframes_t nframes, void *arg) -{ - printf ("the maximum buffer size is now %lu\n", nframes); - return 0; -} - -static int -sample_rate (nframes_t nframes, void *arg) -{ - GstJackClient *client = (GstJackClient*) arg; - printf ("the sample rate is now %lu/sec\n", nframes); - client->rate = nframes; - return 0; -} - -static void -shutdown (void *arg) -{ - GstJackClient *client = (GstJackClient*) arg; - printf ("shutdown %p\n", client); -/* gst_element_set_state (GST_ELEMENT (client->manager), GST_STATE_READY); */ -} - static gboolean plugin_init (GModule *module, GstPlugin *plugin) { @@ -588,6 +449,9 @@ plugin_init (GModule *module, GstPlugin *plugin) return FALSE; } + factory = gst_element_factory_new ("jackbin", GST_TYPE_JACK_BIN, &gst_jack_bin_details); + g_return_val_if_fail (factory != NULL, FALSE); + factory = gst_element_factory_new ("jacksrc", GST_TYPE_JACK_SRC, &gst_jack_src_details); g_return_val_if_fail (factory != NULL, FALSE); gst_element_factory_add_pad_template (factory, gst_jack_src_request_pad_factory()); diff --git a/ext/jack/gstjack.h b/ext/jack/gstjack.h index da6940f2fd..38763cd6e3 100644 --- a/ext/jack/gstjack.h +++ b/ext/jack/gstjack.h @@ -42,11 +42,19 @@ #define GST_IS_JACK_SRC_CLASS(klass) G_TYPE_CHECK_CLASS_TYPE(klass, GST_TYPE_JACK_SRC) #define GST_TYPE_JACK_SRC gst_jack_src_get_type() +#define GST_JACK_BIN(obj) G_TYPE_CHECK_INSTANCE_CAST(obj, GST_TYPE_JACK_BIN, GstJackBin) +#define GST_JACK_BIN_CLASS(klass) G_TYPE_CHECK_CLASS_CAST(klass, GST_TYPE_JACK_BIN, GstJackClass) +#define GST_IS_JACK_BIN(obj) G_TYPE_CHECK_INSTANCE_TYPE(obj, GST_TYPE_JACK_BIN) +#define GST_IS_JACK_BIN_CLASS(klass) G_TYPE_CHECK_CLASS_TYPE(klass, GST_TYPE_JACK_BIN) +#define GST_TYPE_JACK_BIN gst_jack_bin_get_type() + #define GST_JACK_PAD(l) ((GstJackPad*)l->data) /* l is a GList */ typedef struct _GstJack GstJack; typedef struct _GstJackClass GstJackClass; +typedef struct _GstJackBin GstJackBin; +typedef struct _GstJackBinClass GstJackBinClass; typedef GstJack GstJackSink; typedef GstJackClass GstJackSinkClass; typedef GstJack GstJackSrc; @@ -54,9 +62,9 @@ typedef GstJackClass GstJackSrcClass; enum { - GST_JACK_OPEN = GST_ELEMENT_FLAG_LAST, + GST_JACK_OPEN = GST_BIN_FLAG_LAST, GST_JACK_ACTIVE, - GST_JACK_FLAG_LAST = GST_ELEMENT_FLAG_LAST + 3, + GST_JACK_FLAG_LAST = GST_BIN_FLAG_LAST + 3, }; @@ -69,24 +77,6 @@ typedef struct { jack_port_t *port; } GstJackPad; -typedef struct { - GstBin *manager; - - jack_client_t *client; - gint default_new_port_number; - - /* lists of GstJackPads */ - GList *sink_pads; - GList *src_pads; - - gchar *client_name; - - guint rate; - nframes_t nframes; - - guint refcount; -} GstJackClient; - struct _GstJack { GstElement element; @@ -98,16 +88,36 @@ struct _GstJack { gchar *port_name_prefix; - /* there is exactly one client per managing bin */ - GstJackClient *client; + GstJackBin *bin; }; struct _GstJackClass { GstElementClass parent_class; }; +struct _GstJackBin { + GstBin bin; + + jack_client_t *client; + gint default_new_port_number; + + /* lists of GstJackPads */ + GList *sink_pads; + GList *src_pads; + + gchar *client_name; + + guint rate; + nframes_t nframes; +}; + +struct _GstJackBinClass { + GstBinClass parent_class; +}; + GType gst_jack_get_type (void); +GType gst_jack_bin_get_type (void); GType gst_jack_sink_get_type (void); GType gst_jack_src_get_type (void); diff --git a/ext/jack/gstjackbin.c b/ext/jack/gstjackbin.c new file mode 100644 index 0000000000..8a4cc7634a --- /dev/null +++ b/ext/jack/gstjackbin.c @@ -0,0 +1,250 @@ +/* + Copyright (C) 2002 Andy Wingo + + This library is free software; you can redistribute it and/or + modify it under the terms of the GNU General Public + License as published by the Free Software Foundation; either + version 2 of the License, or (at your option) any later version. + + This library is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + General Public License for more details. + + You should have received a copy of the GNU General Public + License along with this library; if not, write to the Free + Software Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. +*/ + +#include +#include "gstjack.h" + + +static GstBinClass *parent_class = NULL; + +static void gst_jack_bin_init(GstJack *this); +static void gst_jack_bin_class_init(GstJackClass *klass); + +static GstElementStateReturn gst_jack_bin_change_state(GstElement *element); + +/* jack callbacks */ +static int process (nframes_t nframes, void *arg); +static int buffer_size (nframes_t nframes, void *arg); +static int sample_rate (nframes_t nframes, void *arg); +static void shutdown (void *arg); + + +GType +gst_jack_bin_get_type (void) +{ + static GType jack_bin_type = 0; + + if (!jack_bin_type) { + static const GTypeInfo jack_bin_info = { + sizeof(GstJackBinClass), + NULL, + NULL, + (GClassInitFunc)gst_jack_bin_class_init, + NULL, + NULL, + sizeof(GstJackBin), + 0, + (GInstanceInitFunc)gst_jack_bin_init, + }; + jack_bin_type = g_type_register_static (GST_TYPE_BIN, "GstJackBin", &jack_bin_info, 0); + } + return jack_bin_type; +} + +static void +gst_jack_bin_class_init(GstJackClass *klass) +{ + GObjectClass *object_class; + GstElementClass *element_class; + + object_class = (GObjectClass *)klass; + element_class = (GstElementClass *)klass; + + parent_class = g_type_class_ref(GST_TYPE_BIN); + + element_class->change_state = gst_jack_bin_change_state; +} + +static void +gst_jack_bin_init(GstJack *this) +{ +} + +static GstElementStateReturn +gst_jack_bin_change_state (GstElement *element) +{ + GstJackBin *this; + GList *l = NULL; + GstJackPad *pad; + + g_return_val_if_fail (element != NULL, FALSE); + this = GST_JACK_BIN (element); + + switch (GST_STATE_PENDING (element)) { + case GST_STATE_NULL: + g_message ("jack: NULL state"); + if (this->client) { + g_message ("jack: closing client"); + jack_client_close (this->client); + } + + break; + + case GST_STATE_READY: + g_message ("jack: READY"); + if (!this->client) { + if (!(this->client = jack_client_new ("gst-jack"))) { + g_warning ("jack server not running?"); + return GST_STATE_FAILURE; + } + + jack_set_process_callback (this->client, process, this); + jack_set_buffer_size_callback (this->client, buffer_size, this); + jack_set_sample_rate_callback (this->client, sample_rate, this); + jack_on_shutdown (this->client, shutdown, this); + } + + /* fixme: there are a *lot* of problems here */ + if (GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_OPEN)) { + l = this->src_pads; + while (l) { + g_message ("jack: unregistering pad %s:%s", GST_JACK_PAD (l)->name, GST_JACK_PAD (l)->peer_name); + jack_port_unregister (this->client, GST_JACK_PAD (l)->port); + l = g_list_next (l); + } + l = this->sink_pads; + while (l) { + g_message ("jack: unregistering pad %s:%s", GST_JACK_PAD (l)->name, GST_JACK_PAD (l)->peer_name); + jack_port_unregister (this->client, GST_JACK_PAD (l)->port); + l = g_list_next (l); + } + GST_FLAG_UNSET (GST_OBJECT (this), GST_JACK_OPEN); + } + break; + + case GST_STATE_PAUSED: + g_message ("jack: PAUSED"); + + if (!GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_OPEN)) { + l = this->src_pads; + while (l) { + pad = GST_JACK_PAD (l); + g_message ("jack: registering pad %s:%s", pad->name, pad->peer_name); + pad->port = jack_port_register (this->client, pad->name, JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0); + g_message ("connecting gst jack port %s to jack port %s", jack_port_name (pad->port), pad->peer_name); + if (jack_connect (this->client, jack_port_name (pad->port), pad->peer_name)) { + g_warning ("could not connect %s and %s", pad->peer_name, jack_port_name (pad->port)); + return GST_STATE_FAILURE; + } + l = g_list_next (l); + } + l = this->sink_pads; + while (l) { + pad = GST_JACK_PAD (l); + g_message ("jack: registering pad %s:%s", pad->name, pad->peer_name); + pad->port = jack_port_register (this->client, pad->name, JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0); + g_message ("connecting gst jack port %s to jack port %s", jack_port_name (pad->port), pad->peer_name); + if (jack_connect (this->client, jack_port_name (pad->port), pad->peer_name)) { + g_warning ("could not connect %s and %s", pad->peer_name, jack_port_name (pad->port)); + return GST_STATE_FAILURE; + } + l = g_list_next (l); + } + g_message ("jack: setting OPEN flag"); + GST_FLAG_SET (GST_OBJECT (this), GST_JACK_OPEN); + } + + if (GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_ACTIVE)) { + g_message ("jack: deactivating client"); + jack_deactivate (this->client); + GST_FLAG_UNSET (GST_OBJECT (this), GST_JACK_ACTIVE); + } + break; + case GST_STATE_PLAYING: + g_message ("jack: PLAYING"); + if (!GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_ACTIVE)) { + g_message ("jack: activating client"); + jack_activate (this->client); + GST_FLAG_SET (GST_OBJECT (this), GST_JACK_ACTIVE); + } + break; + } + + g_message ("jack: state change finished"); + + if (GST_ELEMENT_CLASS (parent_class)->change_state) + return GST_ELEMENT_CLASS (parent_class)->change_state (element); + + return GST_STATE_SUCCESS; +} + +/* jack callbacks */ + +/* keep in mind that these run in another thread, mm-kay? */ + +static int +process (nframes_t nframes, void *arg) +{ + GstJackBin *bin = (GstJackBin*) arg; + GstJackPad *pad; + GList *l; + + g_assert (bin); + + g_message ("jack: process()"); + + l = bin->src_pads; + while (l) { + pad = GST_JACK_PAD (l); + pad->data = jack_port_get_buffer (pad->port, nframes); + l = g_list_next (l); + } + + l = bin->sink_pads; + while (l) { + pad = GST_JACK_PAD (l); + pad->data = jack_port_get_buffer (pad->port, nframes); + l = g_list_next (l); + } + + bin->nframes = nframes; + + if (!gst_bin_iterate (GST_BIN_CAST (bin))) { + g_warning ("bin failed to iterate"); + return -1; + } + + /* that's all folks */ + + return 0; +} + +static int +buffer_size (nframes_t nframes, void *arg) +{ + printf ("the maximum buffer size is now %lu\n", nframes); + return 0; +} + +static int +sample_rate (nframes_t nframes, void *arg) +{ + GstJackBin *bin = (GstJackBin*) arg; + printf ("the sample rate is now %lu/sec\n", nframes); + bin->rate = nframes; + return 0; +} + +static void +shutdown (void *arg) +{ +/* GstJackClient *client = (GstJackClient*) arg; */ + printf ("shutdown %p\n", arg); +/* gst_element_set_state (GST_ELEMENT (client->manager), GST_STATE_READY); */ +} + From 1613f0e51424f2a54bfa8d515e821c32165c8f04 Mon Sep 17 00:00:00 2001 From: Andy Wingo Date: Sat, 20 Apr 2002 21:42:51 +0000 Subject: [PATCH 119/351] a hack to work around intltool's brokenness a current check for mpeg2dec details->klass reorganizations an element br... Original commit message from CVS: * a hack to work around intltool's brokenness * a current check for mpeg2dec * details->klass reorganizations * an element browser that uses details->klass * separated cdxa parse out from the avi directory --- ext/jack/gstjack.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index fc29775068..87ce638c8b 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -37,7 +37,7 @@ static GstElementDetails gst_jack_bin_details = { "Jack Bin", - "Bin/Audio", + "Generic/Bin", "Jack processing bin: see README for more info", VERSION, "Andy Wingo ", @@ -451,6 +451,7 @@ plugin_init (GModule *module, GstPlugin *plugin) factory = gst_element_factory_new ("jackbin", GST_TYPE_JACK_BIN, &gst_jack_bin_details); g_return_val_if_fail (factory != NULL, FALSE); + gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory)); factory = gst_element_factory_new ("jacksrc", GST_TYPE_JACK_SRC, &gst_jack_src_details); g_return_val_if_fail (factory != NULL, FALSE); From fdd942826a34d0aa04940d5f83c752601ebfe4cc Mon Sep 17 00:00:00 2001 From: Andy Wingo Date: Sat, 4 May 2002 18:57:44 +0000 Subject: [PATCH 120/351] some jack fixes, alsa touchups, and add rtp by default to the build if there are any problems building rtp, we're mov... Original commit message from CVS: some jack fixes, alsa touchups, and add rtp by default to the build if there are any problems building rtp, we're moving it back to experimental ;) --- ext/jack/gstjack.c | 1 + ext/jack/gstjack.h | 5 +++++ ext/jack/gstjackbin.c | 47 ++++++++++++++++++++++++++++--------------- 3 files changed, 37 insertions(+), 16 deletions(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 87ce638c8b..1258e23351 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -1,3 +1,4 @@ +/* -*- Mode: C; c-basic-offset: 4 -*- */ /* Copyright (C) 2002 Andy Wingo diff --git a/ext/jack/gstjack.h b/ext/jack/gstjack.h index 38763cd6e3..9d07f86949 100644 --- a/ext/jack/gstjack.h +++ b/ext/jack/gstjack.h @@ -1,3 +1,4 @@ +/* -*- Mode: C; c-basic-offset: 4 -*- */ /* Copyright (C) 2002 Andy Wingo @@ -109,6 +110,10 @@ struct _GstJackBin { guint rate; nframes_t nframes; + + /* the scheduler needs to be setup from within the jack client thread; this + variable is to keep track of whether or not we have been set up yet */ + gboolean sched_setup; }; struct _GstJackBinClass { diff --git a/ext/jack/gstjackbin.c b/ext/jack/gstjackbin.c index 8a4cc7634a..dbaac6650f 100644 --- a/ext/jack/gstjackbin.c +++ b/ext/jack/gstjackbin.c @@ -1,3 +1,4 @@ +/* -*- Mode: C; c-basic-offset: 4 -*- */ /* Copyright (C) 2002 Andy Wingo @@ -37,23 +38,23 @@ static void shutdown (void *arg); GType gst_jack_bin_get_type (void) { - static GType jack_bin_type = 0; + static GType jack_bin_type = 0; - if (!jack_bin_type) { - static const GTypeInfo jack_bin_info = { - sizeof(GstJackBinClass), - NULL, - NULL, - (GClassInitFunc)gst_jack_bin_class_init, - NULL, - NULL, - sizeof(GstJackBin), - 0, - (GInstanceInitFunc)gst_jack_bin_init, - }; - jack_bin_type = g_type_register_static (GST_TYPE_BIN, "GstJackBin", &jack_bin_info, 0); - } - return jack_bin_type; + if (!jack_bin_type) { + static const GTypeInfo jack_bin_info = { + sizeof(GstJackBinClass), + NULL, + NULL, + (GClassInitFunc)gst_jack_bin_class_init, + NULL, + NULL, + sizeof(GstJackBin), + 0, + (GInstanceInitFunc)gst_jack_bin_init, + }; + jack_bin_type = g_type_register_static (GST_TYPE_BIN, "GstJackBin", &jack_bin_info, 0); + } + return jack_bin_type; } static void @@ -73,6 +74,14 @@ gst_jack_bin_class_init(GstJackClass *klass) static void gst_jack_bin_init(GstJack *this) { + GST_DEBUG (GST_CAT_THREAD, "initializing jack bin"); + + /* jack bins are managing bins and iterate themselves */ + GST_FLAG_SET (this, GST_BIN_FLAG_MANAGER); + GST_FLAG_SET (this, GST_BIN_SELF_SCHEDULABLE); + + /* make a new scheduler and associate it with the bin */ + gst_scheduler_factory_make (NULL, GST_ELEMENT (this)); } static GstElementStateReturn @@ -237,6 +246,12 @@ sample_rate (nframes_t nframes, void *arg) GstJackBin *bin = (GstJackBin*) arg; printf ("the sample rate is now %lu/sec\n", nframes); bin->rate = nframes; + + if (!bin->sched_setup) { + gst_scheduler_setup (GST_ELEMENT_SCHED (bin)); + bin->sched_setup = TRUE; + } + return 0; } From 35a88934f5b1b39e8a8dd3ef6c6ca1408f2a9574 Mon Sep 17 00:00:00 2001 From: Andy Wingo Date: Sat, 4 May 2002 20:53:35 +0000 Subject: [PATCH 121/351] set caps once we know the sample rate of the system Original commit message from CVS: set caps once we know the sample rate of the system --- ext/jack/gstjack.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 1258e23351..359ab3c1a2 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -309,6 +309,7 @@ gst_jack_change_state (GstElement *element) GstJack *this; GList *l = NULL, **pads; GstJackPad *pad; + GstCaps *caps; g_return_val_if_fail (element != NULL, FALSE); this = GST_JACK (element); @@ -316,10 +317,12 @@ gst_jack_change_state (GstElement *element) switch (GST_STATE_PENDING (element)) { case GST_STATE_NULL: g_message ("jack client %s: NULL", GST_OBJECT_NAME (GST_OBJECT (this))); + break; case GST_STATE_READY: g_message ("jack client %s: READY", GST_OBJECT_NAME (GST_OBJECT (this))); + if (!this->bin) { if (!(this->bin = (GstJackBin*)gst_element_get_managing_bin (element)) || !GST_IS_JACK_BIN (this->bin)) { @@ -343,6 +346,20 @@ gst_jack_change_state (GstElement *element) case GST_STATE_PAUSED: g_message ("jack client %s: PAUSED", GST_OBJECT_NAME (GST_OBJECT (this))); + + if (GST_STATE (element) == GST_STATE_READY) { + /* we're in READY->PAUSED */ + l = this->pads; + while (l) { + pad = GST_JACK_PAD (l); + caps = gst_pad_get_caps (pad->pad); + gst_caps_set (caps, "rate", GST_PROPS_INT_TYPE, (gint) this->bin->rate, NULL); + caps->fixed = TRUE; /* we know this to be true */ + if (!gst_pad_try_set_caps (pad->pad, caps)) + return GST_STATE_FAILURE; + l = g_list_next (l); + } + } break; case GST_STATE_PLAYING: g_message ("jack client %s: PLAYING", GST_OBJECT_NAME (GST_OBJECT (this))); From d9d5b97653f4842d2e6883ed9ed93460d103515c Mon Sep 17 00:00:00 2001 From: Andy Wingo Date: Sat, 4 May 2002 21:38:56 +0000 Subject: [PATCH 122/351] a commit so that jack will build without errors on Uraeus's system ;) Original commit message from CVS: a commit so that jack will build without errors on Uraeus's system ;) --- ext/jack/gstjackbin.c | 26 ++++++++++++++------------ 1 file changed, 14 insertions(+), 12 deletions(-) diff --git a/ext/jack/gstjackbin.c b/ext/jack/gstjackbin.c index dbaac6650f..0419ef7a32 100644 --- a/ext/jack/gstjackbin.c +++ b/ext/jack/gstjackbin.c @@ -18,13 +18,14 @@ */ #include +#include #include "gstjack.h" static GstBinClass *parent_class = NULL; -static void gst_jack_bin_init(GstJack *this); -static void gst_jack_bin_class_init(GstJackClass *klass); +static void gst_jack_bin_init(GstJackBin *this); +static void gst_jack_bin_class_init(GstJackBinClass *klass); static GstElementStateReturn gst_jack_bin_change_state(GstElement *element); @@ -58,7 +59,7 @@ gst_jack_bin_get_type (void) } static void -gst_jack_bin_class_init(GstJackClass *klass) +gst_jack_bin_class_init(GstJackBinClass *klass) { GObjectClass *object_class; GstElementClass *element_class; @@ -72,7 +73,7 @@ gst_jack_bin_class_init(GstJackClass *klass) } static void -gst_jack_bin_init(GstJack *this) +gst_jack_bin_init(GstJackBin *this) { GST_DEBUG (GST_CAT_THREAD, "initializing jack bin"); @@ -82,6 +83,8 @@ gst_jack_bin_init(GstJack *this) /* make a new scheduler and associate it with the bin */ gst_scheduler_factory_make (NULL, GST_ELEMENT (this)); + + this->sched_setup = FALSE; } static GstElementStateReturn @@ -158,10 +161,10 @@ gst_jack_bin_change_state (GstElement *element) g_message ("jack: registering pad %s:%s", pad->name, pad->peer_name); pad->port = jack_port_register (this->client, pad->name, JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0); g_message ("connecting gst jack port %s to jack port %s", jack_port_name (pad->port), pad->peer_name); - if (jack_connect (this->client, jack_port_name (pad->port), pad->peer_name)) { +/* if (jack_connect (this->client, jack_port_name (pad->port), pad->peer_name)) { g_warning ("could not connect %s and %s", pad->peer_name, jack_port_name (pad->port)); return GST_STATE_FAILURE; - } + } */ l = g_list_next (l); } g_message ("jack: setting OPEN flag"); @@ -207,6 +210,11 @@ process (nframes_t nframes, void *arg) g_message ("jack: process()"); + if (!bin->sched_setup) { + gst_scheduler_setup (GST_ELEMENT_SCHED (bin)); + bin->sched_setup = TRUE; + } + l = bin->src_pads; while (l) { pad = GST_JACK_PAD (l); @@ -246,12 +254,6 @@ sample_rate (nframes_t nframes, void *arg) GstJackBin *bin = (GstJackBin*) arg; printf ("the sample rate is now %lu/sec\n", nframes); bin->rate = nframes; - - if (!bin->sched_setup) { - gst_scheduler_setup (GST_ELEMENT_SCHED (bin)); - bin->sched_setup = TRUE; - } - return 0; } From e166e85ec1795a8b95738a736095d4a828ad311a Mon Sep 17 00:00:00 2001 From: Andy Wingo Date: Sun, 5 May 2002 01:08:05 +0000 Subject: [PATCH 123/351] better initialization. it doesn't work over here, though. Original commit message from CVS: better initialization. it doesn't work over here, though. --- ext/jack/gstjack.c | 2 +- ext/jack/gstjack.h | 2 + ext/jack/gstjackbin.c | 95 +++++++++++++++++++++++++++++++++---------- 3 files changed, 76 insertions(+), 23 deletions(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 359ab3c1a2..65544f1f86 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -418,7 +418,7 @@ gst_jack_loop (GstElement *element) pad = GST_JACK_PAD (pads); if (this->direction == GST_PAD_SINK) { - if (pad->bs) + if (!pad->bs) pad->bs = gst_bytestream_new (pad->pad); if (!(peeked = gst_bytestream_peek_bytes (pad->bs, len))) { diff --git a/ext/jack/gstjack.h b/ext/jack/gstjack.h index 9d07f86949..4801557eb1 100644 --- a/ext/jack/gstjack.h +++ b/ext/jack/gstjack.h @@ -114,6 +114,8 @@ struct _GstJackBin { /* the scheduler needs to be setup from within the jack client thread; this variable is to keep track of whether or not we have been set up yet */ gboolean sched_setup; + GCond *cond; + GMutex *lock; }; struct _GstJackBinClass { diff --git a/ext/jack/gstjackbin.c b/ext/jack/gstjackbin.c index 0419ef7a32..45ab94fd61 100644 --- a/ext/jack/gstjackbin.c +++ b/ext/jack/gstjackbin.c @@ -84,6 +84,9 @@ gst_jack_bin_init(GstJackBin *this) /* make a new scheduler and associate it with the bin */ gst_scheduler_factory_make (NULL, GST_ELEMENT (this)); + this->cond = g_cond_new (); + this->lock = g_mutex_new (); + g_mutex_lock (this->lock); this->sched_setup = FALSE; } @@ -104,7 +107,9 @@ gst_jack_bin_change_state (GstElement *element) g_message ("jack: closing client"); jack_client_close (this->client); } - + + if (GST_ELEMENT_CLASS (parent_class)->change_state) + return GST_ELEMENT_CLASS (parent_class)->change_state (element); break; case GST_STATE_READY: @@ -136,7 +141,16 @@ gst_jack_bin_change_state (GstElement *element) l = g_list_next (l); } GST_FLAG_UNSET (GST_OBJECT (this), GST_JACK_OPEN); + + if (GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_ACTIVE)) { + g_message ("jack: deactivating client"); + jack_deactivate (this->client); + GST_FLAG_UNSET (GST_OBJECT (this), GST_JACK_ACTIVE); + } } + + if (GST_ELEMENT_CLASS (parent_class)->change_state) + return GST_ELEMENT_CLASS (parent_class)->change_state (element); break; case GST_STATE_PAUSED: @@ -148,6 +162,27 @@ gst_jack_bin_change_state (GstElement *element) pad = GST_JACK_PAD (l); g_message ("jack: registering pad %s:%s", pad->name, pad->peer_name); pad->port = jack_port_register (this->client, pad->name, JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0); + l = g_list_next (l); + } + l = this->sink_pads; + while (l) { + pad = GST_JACK_PAD (l); + g_message ("jack: registering pad %s:%s", pad->name, pad->peer_name); + pad->port = jack_port_register (this->client, pad->name, JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0); + l = g_list_next (l); + } + + if (!GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_ACTIVE)) { + g_message ("jack: activating client"); + jack_activate (this->client); + GST_FLAG_SET (GST_OBJECT (this), GST_JACK_ACTIVE); + } + + g_cond_wait (this->cond, this->lock); + + l = this->src_pads; + while (l) { + pad = GST_JACK_PAD (l); g_message ("connecting gst jack port %s to jack port %s", jack_port_name (pad->port), pad->peer_name); if (jack_connect (this->client, jack_port_name (pad->port), pad->peer_name)) { g_warning ("could not connect %s and %s", pad->peer_name, jack_port_name (pad->port)); @@ -158,40 +193,30 @@ gst_jack_bin_change_state (GstElement *element) l = this->sink_pads; while (l) { pad = GST_JACK_PAD (l); - g_message ("jack: registering pad %s:%s", pad->name, pad->peer_name); - pad->port = jack_port_register (this->client, pad->name, JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0); g_message ("connecting gst jack port %s to jack port %s", jack_port_name (pad->port), pad->peer_name); -/* if (jack_connect (this->client, jack_port_name (pad->port), pad->peer_name)) { + if (jack_connect (this->client, jack_port_name (pad->port), pad->peer_name)) { g_warning ("could not connect %s and %s", pad->peer_name, jack_port_name (pad->port)); return GST_STATE_FAILURE; - } */ + } l = g_list_next (l); } + g_message ("jack: setting OPEN flag"); GST_FLAG_SET (GST_OBJECT (this), GST_JACK_OPEN); + } else { + g_cond_wait (this->cond, this->lock); } - if (GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_ACTIVE)) { - g_message ("jack: deactivating client"); - jack_deactivate (this->client); - GST_FLAG_UNSET (GST_OBJECT (this), GST_JACK_ACTIVE); - } break; case GST_STATE_PLAYING: g_message ("jack: PLAYING"); - if (!GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_ACTIVE)) { - g_message ("jack: activating client"); - jack_activate (this->client); - GST_FLAG_SET (GST_OBJECT (this), GST_JACK_ACTIVE); - } + + g_cond_wait (this->cond, this->lock); break; } g_message ("jack: state change finished"); - if (GST_ELEMENT_CLASS (parent_class)->change_state) - return GST_ELEMENT_CLASS (parent_class)->change_state (element); - return GST_STATE_SUCCESS; } @@ -210,11 +235,37 @@ process (nframes_t nframes, void *arg) g_message ("jack: process()"); - if (!bin->sched_setup) { - gst_scheduler_setup (GST_ELEMENT_SCHED (bin)); - bin->sched_setup = TRUE; + if (GST_STATE_PENDING (bin) != GST_STATE_VOID_PENDING) { + g_message ("jackbin: doing state change from %s to %s", + gst_element_state_get_name (GST_STATE (bin)), + gst_element_state_get_name (GST_STATE_PENDING (bin))); + + /* FIXME: this breaks ultra-low latency... */ + g_mutex_lock (bin->lock); + + switch (GST_STATE_TRANSITION (bin)) { + case GST_STATE_READY_TO_PAUSED: + if (!bin->sched_setup) { + gst_scheduler_setup (GST_ELEMENT_SCHED (bin)); + bin->sched_setup = TRUE; + } + break; + } + + /* do the chaining up from within the jack thread, so that child + elements are initialized from within the proper thread */ + if (GST_ELEMENT_CLASS (parent_class)->change_state) + GST_ELEMENT_CLASS (parent_class)->change_state (GST_ELEMENT_CAST (bin)); + + g_cond_signal (bin->cond); + g_mutex_unlock (bin->lock); } - + + if (GST_STATE (bin) != GST_STATE_PLAYING) { + g_message ("jackbin: bin is not PLAYING yet, returning"); + return 0; + } + l = bin->src_pads; while (l) { pad = GST_JACK_PAD (l); From b2ef7f94110931cf099733273b8e31b680d8951c Mon Sep 17 00:00:00 2001 From: Andy Wingo Date: Sun, 5 May 2002 19:39:17 +0000 Subject: [PATCH 124/351] add some includes Original commit message from CVS: add some includes --- ext/jack/gstjack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 65544f1f86..02696901b9 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -18,10 +18,10 @@ */ #include +#include #include "gstjack.h" #include - /* TODO: this element is still nonfunctional From f11d85cec43da3a49bbb6524521c8dc087f76711 Mon Sep 17 00:00:00 2001 From: Andy Wingo Date: Mon, 13 May 2002 18:08:33 +0000 Subject: [PATCH 125/351] update to new jack api Original commit message from CVS: update to new jack api --- ext/jack/gstjack.c | 2 +- ext/jack/gstjack.h | 2 +- ext/jack/gstjackbin.c | 12 ++++++------ 3 files changed, 8 insertions(+), 8 deletions(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 02696901b9..ab0e4a3c43 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -410,7 +410,7 @@ gst_jack_loop (GstElement *element) this = GST_JACK (element); g_return_if_fail(this != NULL); - len = this->bin->nframes * sizeof (sample_t); + len = this->bin->nframes * sizeof (jack_default_audio_sample_t); do { pads = this->pads; diff --git a/ext/jack/gstjack.h b/ext/jack/gstjack.h index 4801557eb1..d13a1dab7d 100644 --- a/ext/jack/gstjack.h +++ b/ext/jack/gstjack.h @@ -109,7 +109,7 @@ struct _GstJackBin { gchar *client_name; guint rate; - nframes_t nframes; + jack_nframes_t nframes; /* the scheduler needs to be setup from within the jack client thread; this variable is to keep track of whether or not we have been set up yet */ diff --git a/ext/jack/gstjackbin.c b/ext/jack/gstjackbin.c index 45ab94fd61..49ecca70a9 100644 --- a/ext/jack/gstjackbin.c +++ b/ext/jack/gstjackbin.c @@ -30,9 +30,9 @@ static void gst_jack_bin_class_init(GstJackBinClass *klass); static GstElementStateReturn gst_jack_bin_change_state(GstElement *element); /* jack callbacks */ -static int process (nframes_t nframes, void *arg); -static int buffer_size (nframes_t nframes, void *arg); -static int sample_rate (nframes_t nframes, void *arg); +static int process (jack_nframes_t nframes, void *arg); +static int buffer_size (jack_nframes_t nframes, void *arg); +static int sample_rate (jack_nframes_t nframes, void *arg); static void shutdown (void *arg); @@ -225,7 +225,7 @@ gst_jack_bin_change_state (GstElement *element) /* keep in mind that these run in another thread, mm-kay? */ static int -process (nframes_t nframes, void *arg) +process (jack_nframes_t nframes, void *arg) { GstJackBin *bin = (GstJackBin*) arg; GstJackPad *pad; @@ -293,14 +293,14 @@ process (nframes_t nframes, void *arg) } static int -buffer_size (nframes_t nframes, void *arg) +buffer_size (jack_nframes_t nframes, void *arg) { printf ("the maximum buffer size is now %lu\n", nframes); return 0; } static int -sample_rate (nframes_t nframes, void *arg) +sample_rate (jack_nframes_t nframes, void *arg) { GstJackBin *bin = (GstJackBin*) arg; printf ("the sample rate is now %lu/sec\n", nframes); From 5789a18e0aa53114716f3ff7b535a719c6e9122e Mon Sep 17 00:00:00 2001 From: Steve Baker Date: Wed, 15 May 2002 19:08:49 +0000 Subject: [PATCH 126/351] use new bytestream api Original commit message from CVS: use new bytestream api --- ext/jack/gstjack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index ab0e4a3c43..4cf4df5877 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -421,7 +421,7 @@ gst_jack_loop (GstElement *element) if (!pad->bs) pad->bs = gst_bytestream_new (pad->pad); - if (!(peeked = gst_bytestream_peek_bytes (pad->bs, len))) { + if (gst_bytestream_peek_bytes (pad->bs, (guint8**)&peeked, len) < len) { gst_bytestream_get_status(pad->bs, &avail, &event); if (event) { g_warning("got an event on jacksink"); From af472ca88be4af647adf4133824505f01e9a66f5 Mon Sep 17 00:00:00 2001 From: Andy Wingo Date: Wed, 12 Jun 2002 03:32:02 +0000 Subject: [PATCH 127/351] working jack elements (fixed a problem in upstream jack) random other fixen... Original commit message from CVS: * working jack elements (fixed a problem in upstream jack) * random other fixen... --- ext/jack/gstjack.c | 18 ++++++++++++------ ext/jack/gstjackbin.c | 13 +++++++------ 2 files changed, 19 insertions(+), 12 deletions(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 4cf4df5877..9311ff5298 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -400,7 +400,7 @@ gst_jack_loop (GstElement *element) { GstJack *this; GList *pads; - gint len; + gint len, peeked_len; gchar *peeked; gint avail; GstEvent *event; @@ -421,7 +421,9 @@ gst_jack_loop (GstElement *element) if (!pad->bs) pad->bs = gst_bytestream_new (pad->pad); - if (gst_bytestream_peek_bytes (pad->bs, (guint8**)&peeked, len) < len) { + read: + peeked_len = gst_bytestream_peek_bytes (pad->bs, (guint8**)&peeked, len); + if (peeked_len < len) { gst_bytestream_get_status(pad->bs, &avail, &event); if (event) { g_warning("got an event on jacksink"); @@ -430,9 +432,12 @@ gst_jack_loop (GstElement *element) * me know when this is needed ;) * also, for sample accuracy etc, we should play avail * bytes, but hey. */ - gst_element_set_eos(GST_ELEMENT(this)); - gst_event_free(event); + gst_element_set_eos (element); + gst_event_free (event); + gst_element_yield (element); /* shouldn't return */ + return; } + goto read; } else { /* the element at the top of the chain did not emit an eos * event. this is a Bug(tm) */ @@ -440,8 +445,9 @@ gst_jack_loop (GstElement *element) } } - memcpy (pad->data, peeked, len); - gst_bytestream_flush (pad->bs, len); + + memcpy (pad->data, peeked, peeked_len); + gst_bytestream_flush (pad->bs, peeked_len); } else { buffer = gst_buffer_new (); GST_BUFFER_DATA (buffer) = pad->data; diff --git a/ext/jack/gstjackbin.c b/ext/jack/gstjackbin.c index 49ecca70a9..5aee29248f 100644 --- a/ext/jack/gstjackbin.c +++ b/ext/jack/gstjackbin.c @@ -102,7 +102,7 @@ gst_jack_bin_change_state (GstElement *element) switch (GST_STATE_PENDING (element)) { case GST_STATE_NULL: - g_message ("jack: NULL state"); + g_message ("jackbin: NULL state"); if (this->client) { g_message ("jack: closing client"); jack_client_close (this->client); @@ -113,7 +113,7 @@ gst_jack_bin_change_state (GstElement *element) break; case GST_STATE_READY: - g_message ("jack: READY"); + g_message ("jackbin: READY"); if (!this->client) { if (!(this->client = jack_client_new ("gst-jack"))) { g_warning ("jack server not running?"); @@ -154,24 +154,25 @@ gst_jack_bin_change_state (GstElement *element) break; case GST_STATE_PAUSED: - g_message ("jack: PAUSED"); + g_message ("jackbin: PAUSED"); if (!GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_OPEN)) { l = this->src_pads; while (l) { pad = GST_JACK_PAD (l); g_message ("jack: registering pad %s:%s", pad->name, pad->peer_name); - pad->port = jack_port_register (this->client, pad->name, JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0); + pad->port = jack_port_register (this->client, pad->name, JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput|JackPortIsTerminal, 0); l = g_list_next (l); } l = this->sink_pads; while (l) { pad = GST_JACK_PAD (l); g_message ("jack: registering pad %s:%s", pad->name, pad->peer_name); - pad->port = jack_port_register (this->client, pad->name, JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0); + pad->port = jack_port_register (this->client, pad->name, JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput|JackPortIsTerminal, 0); l = g_list_next (l); } + /* must activate before connecting */ if (!GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_ACTIVE)) { g_message ("jack: activating client"); jack_activate (this->client); @@ -201,7 +202,7 @@ gst_jack_bin_change_state (GstElement *element) l = g_list_next (l); } - g_message ("jack: setting OPEN flag"); + g_message ("jackbin: setting OPEN flag"); GST_FLAG_SET (GST_OBJECT (this), GST_JACK_OPEN); } else { g_cond_wait (this->cond, this->lock); From c2880b1c6d93d061a8895e620ecea3406e65ed6d Mon Sep 17 00:00:00 2001 From: Andy Wingo Date: Tue, 2 Jul 2002 23:35:07 +0000 Subject: [PATCH 128/351] make jack work in all its full duplex glory Original commit message from CVS: make jack work in all its full duplex glory --- ext/jack/gstjack.c | 26 ++++++++++++++++++++++---- ext/jack/gstjackbin.c | 8 ++++---- 2 files changed, 26 insertions(+), 8 deletions(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 9311ff5298..c86f8566f3 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -64,6 +64,7 @@ static GstElementDetails gst_jack_src_details = { }; +static GHashTable *port_name_counts = NULL; static GstElementClass *parent_class = NULL; static void gst_jack_init(GstJack *this); @@ -188,6 +189,7 @@ gst_jack_class_init(GstJackClass *klass) { GObjectClass *object_class; GstElementClass *element_class; + gchar *prefix; object_class = (GObjectClass *)klass; element_class = (GstElementClass *)klass; @@ -198,10 +200,15 @@ gst_jack_class_init(GstJackClass *klass) object_class->get_property = gst_jack_get_property; object_class->set_property = gst_jack_set_property; + if (GST_IS_JACK_SINK_CLASS (klass)) + prefix = "gst-out-"; + else + prefix = "gst-in-"; + g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_PORT_NAME_PREFIX, g_param_spec_string("port-name-prefix","Port name prefix", "String to prepend to jack port names", - "gst-", + prefix, G_PARAM_READWRITE | G_PARAM_CONSTRUCT)); element_class->change_state = gst_jack_change_state; @@ -230,6 +237,7 @@ gst_jack_request_new_pad (GstElement *element, GstPadTemplate *templ, const gcha gchar *newname; GList *l, **pad_list; GstJackPad *pad; + gint count; g_return_val_if_fail ((this = GST_JACK (element)), NULL); @@ -238,7 +246,7 @@ gst_jack_request_new_pad (GstElement *element, GstPadTemplate *templ, const gcha else if (this->direction == GST_PAD_SRC) pad_list = &this->bin->src_pads; else - pad_list = &this->bin->src_pads; + pad_list = &this->bin->sink_pads; if (name) { l = *pad_list; @@ -251,12 +259,22 @@ gst_jack_request_new_pad (GstElement *element, GstPadTemplate *templ, const gcha } newname = g_strdup (name); } else { - newname = g_strdup ("alsa_pcm:out_1"); + if (this->direction == GST_PAD_SINK) + newname = g_strdup ("alsa_pcm:out_1"); + else + newname = g_strdup ("alsa_pcm:in_1"); } pad = g_new0(GstJackPad, 1); - pad->name = g_strdup_printf ("%s%d", this->port_name_prefix, 1); /* fixme :) */ + if (!port_name_counts) + port_name_counts = g_hash_table_new (g_str_hash, g_str_equal); + + count = GPOINTER_TO_INT (g_hash_table_lookup (port_name_counts, this->port_name_prefix)); + g_hash_table_insert (port_name_counts, g_strdup (this->port_name_prefix), GINT_TO_POINTER (count+1)); + + pad->name = g_strdup_printf ("%s%d", this->port_name_prefix, count); + pad->peer_name = newname; pad->pad = gst_pad_new_from_template (templ, newname); gst_element_add_pad (GST_ELEMENT (this), pad->pad); diff --git a/ext/jack/gstjackbin.c b/ext/jack/gstjackbin.c index 5aee29248f..64aab499de 100644 --- a/ext/jack/gstjackbin.c +++ b/ext/jack/gstjackbin.c @@ -160,14 +160,14 @@ gst_jack_bin_change_state (GstElement *element) l = this->src_pads; while (l) { pad = GST_JACK_PAD (l); - g_message ("jack: registering pad %s:%s", pad->name, pad->peer_name); + g_message ("jack: registering output port %s (peer %s)", pad->name, pad->peer_name); pad->port = jack_port_register (this->client, pad->name, JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput|JackPortIsTerminal, 0); l = g_list_next (l); } l = this->sink_pads; while (l) { pad = GST_JACK_PAD (l); - g_message ("jack: registering pad %s:%s", pad->name, pad->peer_name); + g_message ("jack: registering input port %s (peer %s)", pad->name, pad->peer_name); pad->port = jack_port_register (this->client, pad->name, JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput|JackPortIsTerminal, 0); l = g_list_next (l); } @@ -186,7 +186,7 @@ gst_jack_bin_change_state (GstElement *element) pad = GST_JACK_PAD (l); g_message ("connecting gst jack port %s to jack port %s", jack_port_name (pad->port), pad->peer_name); if (jack_connect (this->client, jack_port_name (pad->port), pad->peer_name)) { - g_warning ("could not connect %s and %s", pad->peer_name, jack_port_name (pad->port)); + g_warning ("jack: could not connect %s and %s", pad->peer_name, jack_port_name (pad->port)); return GST_STATE_FAILURE; } l = g_list_next (l); @@ -196,7 +196,7 @@ gst_jack_bin_change_state (GstElement *element) pad = GST_JACK_PAD (l); g_message ("connecting gst jack port %s to jack port %s", jack_port_name (pad->port), pad->peer_name); if (jack_connect (this->client, jack_port_name (pad->port), pad->peer_name)) { - g_warning ("could not connect %s and %s", pad->peer_name, jack_port_name (pad->port)); + g_warning ("jack: could not connect %s and %s", pad->peer_name, jack_port_name (pad->port)); return GST_STATE_FAILURE; } l = g_list_next (l); From 925babbf2a144316394ad51fa315c82d4d86170c Mon Sep 17 00:00:00 2001 From: Andy Wingo Date: Tue, 9 Jul 2002 17:39:17 +0000 Subject: [PATCH 129/351] compile fixen, and prepare to move MAINTAINER_MODE to as-version.m4 Original commit message from CVS: compile fixen, and prepare to move MAINTAINER_MODE to as-version.m4 --- ext/jack/gstjack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index c86f8566f3..3a9aef796c 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -451,7 +451,7 @@ gst_jack_loop (GstElement *element) * also, for sample accuracy etc, we should play avail * bytes, but hey. */ gst_element_set_eos (element); - gst_event_free (event); + gst_event_unref (event); gst_element_yield (element); /* shouldn't return */ return; } From 31295406d4ddbf79826534d001b3d0bf23a9baac Mon Sep 17 00:00:00 2001 From: Andy Wingo Date: Fri, 23 Aug 2002 04:04:11 +0000 Subject: [PATCH 130/351] fix jack input port connection Original commit message from CVS: fix jack input port connection --- ext/jack/gstjack.c | 1 + ext/jack/gstjackbin.c | 8 ++++---- 2 files changed, 5 insertions(+), 4 deletions(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 3a9aef796c..14e50d5e10 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -471,6 +471,7 @@ gst_jack_loop (GstElement *element) GST_BUFFER_DATA (buffer) = pad->data; GST_BUFFER_SIZE (buffer) = len; GST_BUFFER_MAXSIZE (buffer) = len; + GST_BUFFER_FLAG_SET(buffer, GST_BUFFER_DONTFREE); gst_pad_push (pad->pad, buffer); } diff --git a/ext/jack/gstjackbin.c b/ext/jack/gstjackbin.c index 64aab499de..52222b65a7 100644 --- a/ext/jack/gstjackbin.c +++ b/ext/jack/gstjackbin.c @@ -160,14 +160,14 @@ gst_jack_bin_change_state (GstElement *element) l = this->src_pads; while (l) { pad = GST_JACK_PAD (l); - g_message ("jack: registering output port %s (peer %s)", pad->name, pad->peer_name); + g_message ("jack: registering input port %s (peer %s)", pad->name, pad->peer_name); pad->port = jack_port_register (this->client, pad->name, JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput|JackPortIsTerminal, 0); l = g_list_next (l); } l = this->sink_pads; while (l) { pad = GST_JACK_PAD (l); - g_message ("jack: registering input port %s (peer %s)", pad->name, pad->peer_name); + g_message ("jack: registering output port %s (peer %s)", pad->name, pad->peer_name); pad->port = jack_port_register (this->client, pad->name, JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput|JackPortIsTerminal, 0); l = g_list_next (l); } @@ -184,8 +184,8 @@ gst_jack_bin_change_state (GstElement *element) l = this->src_pads; while (l) { pad = GST_JACK_PAD (l); - g_message ("connecting gst jack port %s to jack port %s", jack_port_name (pad->port), pad->peer_name); - if (jack_connect (this->client, jack_port_name (pad->port), pad->peer_name)) { + g_message ("connecting jack port %s to gst jack port %s", pad->peer_name, jack_port_name (pad->port)); + if (jack_connect (this->client, pad->peer_name, jack_port_name (pad->port))) { g_warning ("jack: could not connect %s and %s", pad->peer_name, jack_port_name (pad->port)); return GST_STATE_FAILURE; } From bec3f5c03dcb7343eebbfc809bc26851f88a0256 Mon Sep 17 00:00:00 2001 From: Thomas Vander Stichele Date: Mon, 9 Sep 2002 23:27:38 +0000 Subject: [PATCH 131/351] removing warnings as approved by wim Original commit message from CVS: removing warnings as approved by wim --- ext/jack/gstjack.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 14e50d5e10..18d4ce4379 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -487,10 +487,8 @@ plugin_init (GModule *module, GstPlugin *plugin) { GstElementFactory *factory; - if (!gst_library_load ("gstbytestream")) { - gst_info("jack: could not load support library: 'gstbytestream'\n"); + if (!gst_library_load ("gstbytestream")) return FALSE; - } factory = gst_element_factory_new ("jackbin", GST_TYPE_JACK_BIN, &gst_jack_bin_details); g_return_val_if_fail (factory != NULL, FALSE); From 3d25650da7df36e2e1fe9fe685f2fb1ff4717b78 Mon Sep 17 00:00:00 2001 From: "Ronald S. Bultje" Date: Tue, 10 Sep 2002 09:31:40 +0000 Subject: [PATCH 132/351] This updates all plugins to the new API for gst_pad_try_set_caps Original commit message from CVS: This updates all plugins to the new API for gst_pad_try_set_caps --- ext/jack/gstjack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 18d4ce4379..179645f4b0 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -373,7 +373,7 @@ gst_jack_change_state (GstElement *element) caps = gst_pad_get_caps (pad->pad); gst_caps_set (caps, "rate", GST_PROPS_INT_TYPE, (gint) this->bin->rate, NULL); caps->fixed = TRUE; /* we know this to be true */ - if (!gst_pad_try_set_caps (pad->pad, caps)) + if (gst_pad_try_set_caps (pad->pad, caps) <= 0) return GST_STATE_FAILURE; l = g_list_next (l); } From c7692e1fbe87ace727902270d86d623240de04e0 Mon Sep 17 00:00:00 2001 From: Christian Schaller Date: Wed, 18 Sep 2002 19:02:52 +0000 Subject: [PATCH 133/351] plugins part of license field patch Original commit message from CVS: plugins part of license field patch --- ext/jack/gstjack.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 179645f4b0..33901da740 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -35,10 +35,11 @@ sensitive to them */ - +/* elementfactory information */ static GstElementDetails gst_jack_bin_details = { "Jack Bin", "Generic/Bin", + "LGPL", "Jack processing bin: see README for more info", VERSION, "Andy Wingo ", From cc1c7106c1380e2ccc11bda4ffe53c5d3ee91dd8 Mon Sep 17 00:00:00 2001 From: Andy Wingo Date: Sun, 29 Sep 2002 18:12:18 +0000 Subject: [PATCH 134/351] licenses again Original commit message from CVS: licenses again --- ext/jack/gstjack.c | 4 +++- ext/jack/gstjackbin.c | 2 +- 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 33901da740..4a19f3fc33 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -39,7 +39,7 @@ static GstElementDetails gst_jack_bin_details = { "Jack Bin", "Generic/Bin", - "LGPL", + "GPL", "Jack processing bin: see README for more info", VERSION, "Andy Wingo ", @@ -49,6 +49,7 @@ static GstElementDetails gst_jack_bin_details = { static GstElementDetails gst_jack_sink_details = { "Jack Sink", "Sink/Audio", + "GPL", "Output to a Jack processing network: see README for more info", VERSION, "Andy Wingo ", @@ -58,6 +59,7 @@ static GstElementDetails gst_jack_sink_details = { static GstElementDetails gst_jack_src_details = { "Jack Src", "Source/Audio", + "GPL", "Input from a Jack processing network: see README for more info", VERSION, "Andy Wingo ", diff --git a/ext/jack/gstjackbin.c b/ext/jack/gstjackbin.c index 52222b65a7..1ba90507e7 100644 --- a/ext/jack/gstjackbin.c +++ b/ext/jack/gstjackbin.c @@ -106,6 +106,7 @@ gst_jack_bin_change_state (GstElement *element) if (this->client) { g_message ("jack: closing client"); jack_client_close (this->client); + this->client = NULL; } if (GST_ELEMENT_CLASS (parent_class)->change_state) @@ -126,7 +127,6 @@ gst_jack_bin_change_state (GstElement *element) jack_on_shutdown (this->client, shutdown, this); } - /* fixme: there are a *lot* of problems here */ if (GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_OPEN)) { l = this->src_pads; while (l) { From e2c03f4e9ca9f336355df95cac1df113400deff3 Mon Sep 17 00:00:00 2001 From: Thomas Vander Stichele Date: Sun, 8 Dec 2002 14:50:10 +0000 Subject: [PATCH 135/351] parallel install fixes Original commit message from CVS: parallel install fixes --- ext/jack/Makefile.am | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/ext/jack/Makefile.am b/ext/jack/Makefile.am index 981c5a9f77..3d05433206 100644 --- a/ext/jack/Makefile.am +++ b/ext/jack/Makefile.am @@ -1,4 +1,4 @@ -plugindir = $(libdir)/gst +plugindir = $(libdir)/gstreamer-@GST_MAJORMINOR@ plugin_LTLIBRARIES = libgstjack.la From 79806375c4de3a12ad00742a766c28a7d5972ccd Mon Sep 17 00:00:00 2001 From: Thomas Vander Stichele Date: Fri, 10 Jan 2003 10:22:25 +0000 Subject: [PATCH 136/351] another batch of connect->link fixes please let me know about issues and please refrain of making them yourself, so t... Original commit message from CVS: another batch of connect->link fixes please let me know about issues and please refrain of making them yourself, so that I don't spend double the time resolving conflicts --- ext/jack/gstjack.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 4a19f3fc33..25c57eb561 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -281,7 +281,7 @@ gst_jack_request_new_pad (GstElement *element, GstPadTemplate *templ, const gcha pad->peer_name = newname; pad->pad = gst_pad_new_from_template (templ, newname); gst_element_add_pad (GST_ELEMENT (this), pad->pad); - gst_pad_set_connect_function (pad->pad, gst_jack_connect); + gst_pad_set_link_function (pad->pad, gst_jack_connect); this->pads = g_list_append (this->pads, pad); @@ -402,18 +402,18 @@ gst_jack_connect (GstPad *pad, GstCaps *caps) gint rate; this = GST_JACK (gst_pad_get_parent (pad)); - g_return_val_if_fail (this != NULL, GST_PAD_CONNECT_REFUSED); - g_return_val_if_fail (GST_IS_JACK (this), GST_PAD_CONNECT_REFUSED); + g_return_val_if_fail (this != NULL, GST_PAD_LINK_REFUSED); + g_return_val_if_fail (GST_IS_JACK (this), GST_PAD_LINK_REFUSED); if (GST_CAPS_IS_FIXED (caps)) { gst_caps_get_int (caps, "rate", &rate); if (this->bin && rate != this->bin->rate) - return GST_PAD_CONNECT_REFUSED; + return GST_PAD_LINK_REFUSED; - return GST_PAD_CONNECT_OK; + return GST_PAD_LINK_OK; } - return GST_PAD_CONNECT_DELAYED; + return GST_PAD_LINK_DELAYED; } static void From 5781a1a912becbb3ee0c86c772998c0e3e31c5a5 Mon Sep 17 00:00:00 2001 From: Thomas Vander Stichele Date: Fri, 10 Jan 2003 13:38:32 +0000 Subject: [PATCH 137/351] PadConnect -> PadLink Original commit message from CVS: PadConnect -> PadLink --- ext/jack/gstjack.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 25c57eb561..11e80c081b 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -82,7 +82,7 @@ static GstPad* gst_jack_request_new_pad (GstElement *element, GstPadTemplate *te static void gst_jack_set_property(GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec); static void gst_jack_get_property(GObject *object, guint prop_id, GValue *value, GParamSpec *pspec); static GstElementStateReturn gst_jack_change_state(GstElement *element); -static GstPadConnectReturn gst_jack_connect (GstPad *pad, GstCaps *caps); +static GstPadLinkReturn gst_jack_connect (GstPad *pad, GstCaps *caps); static void gst_jack_loop (GstElement *element); @@ -395,7 +395,7 @@ gst_jack_change_state (GstElement *element) return GST_STATE_SUCCESS; } -static GstPadConnectReturn +static GstPadLinkReturn gst_jack_connect (GstPad *pad, GstCaps *caps) { GstJack *this; From a75afe6604c44a722f72be303a25475a5e581149 Mon Sep 17 00:00:00 2001 From: Jan Schmidt Date: Wed, 5 Feb 2003 20:38:41 +0000 Subject: [PATCH 138/351] Changed caps->fixed to use FLAG_SET Original commit message from CVS: Changed caps->fixed to use FLAG_SET --- ext/jack/gstjack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 11e80c081b..66548fe030 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -375,7 +375,7 @@ gst_jack_change_state (GstElement *element) pad = GST_JACK_PAD (l); caps = gst_pad_get_caps (pad->pad); gst_caps_set (caps, "rate", GST_PROPS_INT_TYPE, (gint) this->bin->rate, NULL); - caps->fixed = TRUE; /* we know this to be true */ + GST_CAPS_FLAG_SET (caps, GST_CAPS_FIXED); if (gst_pad_try_set_caps (pad->pad, caps) <= 0) return GST_STATE_FAILURE; l = g_list_next (l); From c357ceba44ed82625a48f9697cdcb995b0d0058b Mon Sep 17 00:00:00 2001 From: Andy Wingo Date: Tue, 4 Mar 2003 15:34:20 +0000 Subject: [PATCH 139/351] update for the latest jack cvs and non-cothreaded gst scheduler Original commit message from CVS: update for the latest jack cvs and non-cothreaded gst scheduler --- ext/jack/gstjack.c | 21 ++++---- ext/jack/gstjack.h | 8 +-- ext/jack/gstjackbin.c | 119 +++++++++++++++++++----------------------- 3 files changed, 66 insertions(+), 82 deletions(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 66548fe030..ea84afbf01 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -263,9 +263,9 @@ gst_jack_request_new_pad (GstElement *element, GstPadTemplate *templ, const gcha newname = g_strdup (name); } else { if (this->direction == GST_PAD_SINK) - newname = g_strdup ("alsa_pcm:out_1"); + newname = g_strdup ("alsa_pcm:playback_1"); else - newname = g_strdup ("alsa_pcm:in_1"); + newname = g_strdup ("alsa_pcm:capture_1"); } pad = g_new0(GstJackPad, 1); @@ -337,12 +337,12 @@ gst_jack_change_state (GstElement *element) switch (GST_STATE_PENDING (element)) { case GST_STATE_NULL: - g_message ("jack client %s: NULL", GST_OBJECT_NAME (GST_OBJECT (this))); + JACK_DEBUG ("%s: NULL", GST_OBJECT_NAME (GST_OBJECT (this))); break; case GST_STATE_READY: - g_message ("jack client %s: READY", GST_OBJECT_NAME (GST_OBJECT (this))); + JACK_DEBUG ("%s: READY", GST_OBJECT_NAME (GST_OBJECT (this))); if (!this->bin) { if (!(this->bin = (GstJackBin*)gst_element_get_managing_bin (element)) @@ -358,7 +358,7 @@ gst_jack_change_state (GstElement *element) pads = (this->direction == GST_PAD_SRC) ? &this->bin->src_pads : &this->bin->sink_pads; while (l) { pad = GST_JACK_PAD (l); - g_message ("jack: appending pad %s:%s to list", pad->name, pad->peer_name); + JACK_DEBUG ("%s: appending pad %s:%s to list", GST_OBJECT_NAME (this), pad->name, pad->peer_name); *pads = g_list_append (*pads, pad); l = g_list_next (l); } @@ -366,7 +366,7 @@ gst_jack_change_state (GstElement *element) break; case GST_STATE_PAUSED: - g_message ("jack client %s: PAUSED", GST_OBJECT_NAME (GST_OBJECT (this))); + JACK_DEBUG ("%s: PAUSED", GST_OBJECT_NAME (GST_OBJECT (this))); if (GST_STATE (element) == GST_STATE_READY) { /* we're in READY->PAUSED */ @@ -383,11 +383,11 @@ gst_jack_change_state (GstElement *element) } break; case GST_STATE_PLAYING: - g_message ("jack client %s: PLAYING", GST_OBJECT_NAME (GST_OBJECT (this))); + JACK_DEBUG ("%s: PLAYING", GST_OBJECT_NAME (GST_OBJECT (this))); break; } - g_message ("jack: state change finished"); + JACK_DEBUG ("%s: state change finished", GST_OBJECT_NAME (this)); if (GST_ELEMENT_CLASS (parent_class)->change_state) return GST_ELEMENT_CLASS (parent_class)->change_state (element); @@ -455,7 +455,7 @@ gst_jack_loop (GstElement *element) * bytes, but hey. */ gst_element_set_eos (element); gst_event_unref (event); - gst_element_yield (element); /* shouldn't return */ +// gst_element_yield (element); /* shouldn't return */ return; } goto read; @@ -481,7 +481,8 @@ gst_jack_loop (GstElement *element) pads = g_list_next (pads); } - gst_element_yield (element); + return; +// gst_element_yield (element); } while (TRUE); } diff --git a/ext/jack/gstjack.h b/ext/jack/gstjack.h index d13a1dab7d..3bf3f2d4ff 100644 --- a/ext/jack/gstjack.h +++ b/ext/jack/gstjack.h @@ -24,6 +24,8 @@ #include #include +//#define JACK_DEBUG(str, a...) g_message (str, ##a) +#define JACK_DEBUG(str, a...) #define GST_JACK(obj) G_TYPE_CHECK_INSTANCE_CAST(obj, GST_TYPE_JACK, GstJack) #define GST_JACK_CLASS(klass) G_TYPE_CHECK_CLASS_CAST(klass, GST_TYPE_JACK, GstJackClass) @@ -110,12 +112,6 @@ struct _GstJackBin { guint rate; jack_nframes_t nframes; - - /* the scheduler needs to be setup from within the jack client thread; this - variable is to keep track of whether or not we have been set up yet */ - gboolean sched_setup; - GCond *cond; - GMutex *lock; }; struct _GstJackBinClass { diff --git a/ext/jack/gstjackbin.c b/ext/jack/gstjackbin.c index 1ba90507e7..76c7e07527 100644 --- a/ext/jack/gstjackbin.c +++ b/ext/jack/gstjackbin.c @@ -19,6 +19,7 @@ #include #include +#include #include "gstjack.h" @@ -31,17 +32,21 @@ static GstElementStateReturn gst_jack_bin_change_state(GstElement *element); /* jack callbacks */ static int process (jack_nframes_t nframes, void *arg); -static int buffer_size (jack_nframes_t nframes, void *arg); static int sample_rate (jack_nframes_t nframes, void *arg); static void shutdown (void *arg); +static void sighup_handler (int sig); +static GstJackBin *_jackbin = NULL; +static gboolean watchdog_please_set_the_jackbin_to_ready = FALSE; + +/* fixme: we need a watchdog thread to see if we have received a SIGHUP, and if + * so set the state of the bin to READY. */ GType gst_jack_bin_get_type (void) { static GType jack_bin_type = 0; - - if (!jack_bin_type) { +if (!jack_bin_type) { static const GTypeInfo jack_bin_info = { sizeof(GstJackBinClass), NULL, @@ -83,11 +88,6 @@ gst_jack_bin_init(GstJackBin *this) /* make a new scheduler and associate it with the bin */ gst_scheduler_factory_make (NULL, GST_ELEMENT (this)); - - this->cond = g_cond_new (); - this->lock = g_mutex_new (); - g_mutex_lock (this->lock); - this->sched_setup = FALSE; } static GstElementStateReturn @@ -102,27 +102,36 @@ gst_jack_bin_change_state (GstElement *element) switch (GST_STATE_PENDING (element)) { case GST_STATE_NULL: - g_message ("jackbin: NULL state"); + JACK_DEBUG ("jackbin: NULL state"); if (this->client) { - g_message ("jack: closing client"); + JACK_DEBUG ("jackbin: closing client"); jack_client_close (this->client); this->client = NULL; } + if (_jackbin) + signal (SIGHUP, SIG_DFL); + _jackbin = NULL; + if (GST_ELEMENT_CLASS (parent_class)->change_state) return GST_ELEMENT_CLASS (parent_class)->change_state (element); break; case GST_STATE_READY: - g_message ("jackbin: READY"); + JACK_DEBUG ("jackbin: READY"); + + _jackbin = this; + signal (SIGHUP, sighup_handler); + if (!this->client) { if (!(this->client = jack_client_new ("gst-jack"))) { g_warning ("jack server not running?"); return GST_STATE_FAILURE; } + gst_scheduler_setup (GST_ELEMENT_SCHED (this)); + jack_set_process_callback (this->client, process, this); - jack_set_buffer_size_callback (this->client, buffer_size, this); jack_set_sample_rate_callback (this->client, sample_rate, this); jack_on_shutdown (this->client, shutdown, this); } @@ -130,20 +139,20 @@ gst_jack_bin_change_state (GstElement *element) if (GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_OPEN)) { l = this->src_pads; while (l) { - g_message ("jack: unregistering pad %s:%s", GST_JACK_PAD (l)->name, GST_JACK_PAD (l)->peer_name); + JACK_DEBUG ("jackbin: unregistering pad %s:%s", GST_JACK_PAD (l)->name, GST_JACK_PAD (l)->peer_name); jack_port_unregister (this->client, GST_JACK_PAD (l)->port); l = g_list_next (l); } l = this->sink_pads; while (l) { - g_message ("jack: unregistering pad %s:%s", GST_JACK_PAD (l)->name, GST_JACK_PAD (l)->peer_name); + JACK_DEBUG ("jackbin: unregistering pad %s:%s", GST_JACK_PAD (l)->name, GST_JACK_PAD (l)->peer_name); jack_port_unregister (this->client, GST_JACK_PAD (l)->port); l = g_list_next (l); } GST_FLAG_UNSET (GST_OBJECT (this), GST_JACK_OPEN); if (GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_ACTIVE)) { - g_message ("jack: deactivating client"); + JACK_DEBUG ("jackbin: deactivating client"); jack_deactivate (this->client); GST_FLAG_UNSET (GST_OBJECT (this), GST_JACK_ACTIVE); } @@ -154,39 +163,37 @@ gst_jack_bin_change_state (GstElement *element) break; case GST_STATE_PAUSED: - g_message ("jackbin: PAUSED"); + JACK_DEBUG ("jackbin: PAUSED"); if (!GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_OPEN)) { l = this->src_pads; while (l) { pad = GST_JACK_PAD (l); - g_message ("jack: registering input port %s (peer %s)", pad->name, pad->peer_name); + JACK_DEBUG ("jackbin: registering input port %s (peer %s)", pad->name, pad->peer_name); pad->port = jack_port_register (this->client, pad->name, JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput|JackPortIsTerminal, 0); l = g_list_next (l); } l = this->sink_pads; while (l) { pad = GST_JACK_PAD (l); - g_message ("jack: registering output port %s (peer %s)", pad->name, pad->peer_name); + JACK_DEBUG ("jackbin: registering output port %s (peer %s)", pad->name, pad->peer_name); pad->port = jack_port_register (this->client, pad->name, JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput|JackPortIsTerminal, 0); l = g_list_next (l); } /* must activate before connecting */ if (!GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_ACTIVE)) { - g_message ("jack: activating client"); + JACK_DEBUG ("jackbin: activating client"); jack_activate (this->client); GST_FLAG_SET (GST_OBJECT (this), GST_JACK_ACTIVE); } - g_cond_wait (this->cond, this->lock); - l = this->src_pads; while (l) { pad = GST_JACK_PAD (l); - g_message ("connecting jack port %s to gst jack port %s", pad->peer_name, jack_port_name (pad->port)); + JACK_DEBUG ("connecting jack port %s to gst jack port %s", pad->peer_name, jack_port_name (pad->port)); if (jack_connect (this->client, pad->peer_name, jack_port_name (pad->port))) { - g_warning ("jack: could not connect %s and %s", pad->peer_name, jack_port_name (pad->port)); + g_warning ("jackbin: could not connect %s and %s", pad->peer_name, jack_port_name (pad->port)); return GST_STATE_FAILURE; } l = g_list_next (l); @@ -194,29 +201,34 @@ gst_jack_bin_change_state (GstElement *element) l = this->sink_pads; while (l) { pad = GST_JACK_PAD (l); - g_message ("connecting gst jack port %s to jack port %s", jack_port_name (pad->port), pad->peer_name); + JACK_DEBUG ("connecting gst jack port %s to jack port %s", jack_port_name (pad->port), pad->peer_name); if (jack_connect (this->client, jack_port_name (pad->port), pad->peer_name)) { - g_warning ("jack: could not connect %s and %s", pad->peer_name, jack_port_name (pad->port)); + g_warning ("jackbin: could not connect %s and %s", pad->peer_name, jack_port_name (pad->port)); return GST_STATE_FAILURE; } l = g_list_next (l); } - g_message ("jackbin: setting OPEN flag"); + JACK_DEBUG ("jackbin: setting OPEN flag"); GST_FLAG_SET (GST_OBJECT (this), GST_JACK_OPEN); + + if (GST_ELEMENT_CLASS (parent_class)->change_state) + return GST_ELEMENT_CLASS (parent_class)->change_state (element); } else { - g_cond_wait (this->cond, this->lock); + if (GST_ELEMENT_CLASS (parent_class)->change_state) + return GST_ELEMENT_CLASS (parent_class)->change_state (element); } break; case GST_STATE_PLAYING: - g_message ("jack: PLAYING"); + JACK_DEBUG ("jackbin: PLAYING"); - g_cond_wait (this->cond, this->lock); + if (GST_ELEMENT_CLASS (parent_class)->change_state) + return GST_ELEMENT_CLASS (parent_class)->change_state (element); break; } - g_message ("jack: state change finished"); + JACK_DEBUG ("jackbin: state change finished"); return GST_STATE_SUCCESS; } @@ -234,37 +246,13 @@ process (jack_nframes_t nframes, void *arg) g_assert (bin); - g_message ("jack: process()"); - - if (GST_STATE_PENDING (bin) != GST_STATE_VOID_PENDING) { - g_message ("jackbin: doing state change from %s to %s", - gst_element_state_get_name (GST_STATE (bin)), - gst_element_state_get_name (GST_STATE_PENDING (bin))); - - /* FIXME: this breaks ultra-low latency... */ - g_mutex_lock (bin->lock); - - switch (GST_STATE_TRANSITION (bin)) { - case GST_STATE_READY_TO_PAUSED: - if (!bin->sched_setup) { - gst_scheduler_setup (GST_ELEMENT_SCHED (bin)); - bin->sched_setup = TRUE; - } - break; - } - - /* do the chaining up from within the jack thread, so that child - elements are initialized from within the proper thread */ - if (GST_ELEMENT_CLASS (parent_class)->change_state) - GST_ELEMENT_CLASS (parent_class)->change_state (GST_ELEMENT_CAST (bin)); - - g_cond_signal (bin->cond); - g_mutex_unlock (bin->lock); - } + JACK_DEBUG ("jackbin: process()"); if (GST_STATE (bin) != GST_STATE_PLAYING) { - g_message ("jackbin: bin is not PLAYING yet, returning"); + JACK_DEBUG ("jackbin: bin is not PLAYING yet, returning"); return 0; + } else { + JACK_DEBUG ("jackbin: we are PLAYING, let's process()"); } l = bin->src_pads; @@ -283,6 +271,7 @@ process (jack_nframes_t nframes, void *arg) bin->nframes = nframes; + JACK_DEBUG ("jackbin: iterating to process %ld frames of audio...", nframes); if (!gst_bin_iterate (GST_BIN_CAST (bin))) { g_warning ("bin failed to iterate"); return -1; @@ -293,18 +282,11 @@ process (jack_nframes_t nframes, void *arg) return 0; } -static int -buffer_size (jack_nframes_t nframes, void *arg) -{ - printf ("the maximum buffer size is now %lu\n", nframes); - return 0; -} - static int sample_rate (jack_nframes_t nframes, void *arg) { GstJackBin *bin = (GstJackBin*) arg; - printf ("the sample rate is now %lu/sec\n", nframes); + JACK_DEBUG ("the sample rate is now %lu/sec\n", nframes); bin->rate = nframes; return 0; } @@ -317,3 +299,8 @@ shutdown (void *arg) /* gst_element_set_state (GST_ELEMENT (client->manager), GST_STATE_READY); */ } +static void sighup_handler (int sig) +{ + g_message ("got sighup, setting state to READY"); + watchdog_please_set_the_jackbin_to_ready = TRUE; +} From c48b1d665358b40cce3ff2a91ccad04b6f361cf6 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Fri, 13 Jun 2003 21:21:17 +0000 Subject: [PATCH 140/351] Removed ugly caps fixed flag hack, will be done automatically in core soon Original commit message from CVS: Removed ugly caps fixed flag hack, will be done automatically in core soon --- ext/jack/gstjack.c | 1 - 1 file changed, 1 deletion(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index ea84afbf01..8a3357dd31 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -375,7 +375,6 @@ gst_jack_change_state (GstElement *element) pad = GST_JACK_PAD (l); caps = gst_pad_get_caps (pad->pad); gst_caps_set (caps, "rate", GST_PROPS_INT_TYPE, (gint) this->bin->rate, NULL); - GST_CAPS_FLAG_SET (caps, GST_CAPS_FIXED); if (gst_pad_try_set_caps (pad->pad, caps) <= 0) return GST_STATE_FAILURE; l = g_list_next (l); From c28be5c9674c15ab297af81e0d323e892935b491 Mon Sep 17 00:00:00 2001 From: Benjamin Otte Date: Sun, 29 Jun 2003 19:46:13 +0000 Subject: [PATCH 141/351] compatibility fix for new GST_DEBUG stuff. Original commit message from CVS: compatibility fix for new GST_DEBUG stuff. Includes fixes for missing includes for config.h and unistd.h I only ensured for plugins I can build that they work, so if some of them are still broken, you gotta fix them yourselves unfortunately. --- ext/jack/gstjack.c | 3 +++ ext/jack/gstjackbin.c | 2 +- 2 files changed, 4 insertions(+), 1 deletion(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 8a3357dd31..68d0775363 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -17,6 +17,9 @@ Software Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. */ +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif #include #include #include "gstjack.h" diff --git a/ext/jack/gstjackbin.c b/ext/jack/gstjackbin.c index 76c7e07527..959f58673a 100644 --- a/ext/jack/gstjackbin.c +++ b/ext/jack/gstjackbin.c @@ -80,7 +80,7 @@ gst_jack_bin_class_init(GstJackBinClass *klass) static void gst_jack_bin_init(GstJackBin *this) { - GST_DEBUG (GST_CAT_THREAD, "initializing jack bin"); + GST_DEBUG ("initializing jack bin"); /* jack bins are managing bins and iterate themselves */ GST_FLAG_SET (this, GST_BIN_FLAG_MANAGER); From 9a04390288b4d3edd304ab4c92e297289f157d77 Mon Sep 17 00:00:00 2001 From: David Schleef Date: Tue, 1 Jul 2003 02:27:06 +0000 Subject: [PATCH 142/351] fix type punning Original commit message from CVS: fix type punning --- ext/jack/gstjack.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 68d0775363..8db6c56b08 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -424,7 +424,7 @@ gst_jack_loop (GstElement *element) GstJack *this; GList *pads; gint len, peeked_len; - gchar *peeked; + guint8 *peeked; gint avail; GstEvent *event; GstJackPad *pad; @@ -445,7 +445,7 @@ gst_jack_loop (GstElement *element) pad->bs = gst_bytestream_new (pad->pad); read: - peeked_len = gst_bytestream_peek_bytes (pad->bs, (guint8**)&peeked, len); + peeked_len = gst_bytestream_peek_bytes (pad->bs, &peeked, len); if (peeked_len < len) { gst_bytestream_get_status(pad->bs, &avail, &event); if (event) { From 8bdd27fd624f0532c8a29e45b5baebf847f07abe Mon Sep 17 00:00:00 2001 From: "Ronald S. Bultje" Date: Sun, 6 Jul 2003 20:49:52 +0000 Subject: [PATCH 143/351] New mimetypes gone into effect today - this commit changes all old mimetypes over to the new mimetypes spec as descri... Original commit message from CVS: New mimetypes gone into effect today - this commit changes all old mimetypes over to the new mimetypes spec as described in the previous commit's document. Note: some plugins will break, some pipelines will break, expect HEAD to be broken or at least not 100% working for a few days, but don't forget to report bugs --- ext/jack/gstjack.c | 26 ++++++++++++++++---------- 1 file changed, 16 insertions(+), 10 deletions(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 8db6c56b08..700cf0aead 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -167,11 +167,14 @@ gst_jack_src_request_pad_factory(void) { static GstPadTemplate *template = NULL; - if (!template) - template = gst_pad_template_new("%s", GST_PAD_SRC, GST_PAD_REQUEST, - gst_caps_new("src", "audio/raw", - GST_AUDIO_FLOAT_MONO_PAD_TEMPLATE_PROPS), - NULL); + if (!template) { + GstCaps *caps; + caps = gst_caps_new("src", + "audio/x-raw-float", + GST_AUDIO_FLOAT_MONO_PAD_TEMPLATE_PROPS); + template = gst_pad_template_new("%s", GST_PAD_SRC, + GST_PAD_REQUEST, caps, NULL); + } return template; } @@ -181,11 +184,14 @@ gst_jack_sink_request_pad_factory(void) { static GstPadTemplate *template = NULL; - if (!template) - template = gst_pad_template_new("%s", GST_PAD_SINK, GST_PAD_REQUEST, - gst_caps_new("sink", "audio/raw", - GST_AUDIO_FLOAT_MONO_PAD_TEMPLATE_PROPS), - NULL); + if (!template) { + GstCaps *caps; + caps = gst_caps_new ("sink", + "audio/x-raw-float", + GST_AUDIO_FLOAT_MONO_PAD_TEMPLATE_PROPS); + template = gst_pad_template_new("%s", GST_PAD_SINK, + GST_PAD_REQUEST, caps, NULL); + } return template; } From 25dd7cc50c91096f2bbd5ee6099e2a8069182743 Mon Sep 17 00:00:00 2001 From: Leif Johnson Date: Sat, 19 Jul 2003 23:25:25 +0000 Subject: [PATCH 144/351] + changes for new float caps without slope/intercept + some category changes for plugins Original commit message from CVS: + changes for new float caps without slope/intercept + some category changes for plugins --- ext/jack/gstjack.c | 8 ++++---- ext/jack/gstjack.h | 8 ++++---- 2 files changed, 8 insertions(+), 8 deletions(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 700cf0aead..6e057f6cd4 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -26,7 +26,7 @@ #include /* TODO: - + this element is still nonfunctional - work out the src side (caps setting, etc) @@ -39,7 +39,7 @@ */ /* elementfactory information */ -static GstElementDetails gst_jack_bin_details = { +static GstElementDetails gst_jack_bin_details = { "Jack Bin", "Generic/Bin", "GPL", @@ -171,7 +171,7 @@ gst_jack_src_request_pad_factory(void) GstCaps *caps; caps = gst_caps_new("src", "audio/x-raw-float", - GST_AUDIO_FLOAT_MONO_PAD_TEMPLATE_PROPS); + GST_AUDIO_FLOAT_STANDARD_PAD_TEMPLATE_PROPS); template = gst_pad_template_new("%s", GST_PAD_SRC, GST_PAD_REQUEST, caps, NULL); } @@ -188,7 +188,7 @@ gst_jack_sink_request_pad_factory(void) GstCaps *caps; caps = gst_caps_new ("sink", "audio/x-raw-float", - GST_AUDIO_FLOAT_MONO_PAD_TEMPLATE_PROPS); + GST_AUDIO_FLOAT_STANDARD_PAD_TEMPLATE_PROPS); template = gst_pad_template_new("%s", GST_PAD_SINK, GST_PAD_REQUEST, caps, NULL); } diff --git a/ext/jack/gstjack.h b/ext/jack/gstjack.h index 3bf3f2d4ff..216c305292 100644 --- a/ext/jack/gstjack.h +++ b/ext/jack/gstjack.h @@ -1,7 +1,7 @@ /* -*- Mode: C; c-basic-offset: 4 -*- */ /* Copyright (C) 2002 Andy Wingo - + This library is free software; you can redistribute it and/or modify it under the terms of the GNU Library General Public License as published by the Free Software Foundation; either @@ -85,12 +85,12 @@ struct _GstJack { /* list of GstJackPads */ GList *pads; - + /* for convenience */ GstPadDirection direction; - + gchar *port_name_prefix; - + GstJackBin *bin; }; From b87729bd6969f21f7c29a6a54c8521bb254a1b3a Mon Sep 17 00:00:00 2001 From: David Schleef Date: Sun, 10 Aug 2003 00:01:58 +0000 Subject: [PATCH 145/351] Remove redundant plugindir definition Original commit message from CVS: Remove redundant plugindir definition --- ext/jack/Makefile.am | 1 - 1 file changed, 1 deletion(-) diff --git a/ext/jack/Makefile.am b/ext/jack/Makefile.am index 3d05433206..a97a2ad7a9 100644 --- a/ext/jack/Makefile.am +++ b/ext/jack/Makefile.am @@ -1,4 +1,3 @@ -plugindir = $(libdir)/gstreamer-@GST_MAJORMINOR@ plugin_LTLIBRARIES = libgstjack.la From b5802bdbb6814d493202531c9af5efad9ecf016d Mon Sep 17 00:00:00 2001 From: Andy Wingo Date: Tue, 30 Sep 2003 12:56:27 +0000 Subject: [PATCH 146/351] conform to the buffer-frames props entry -- much nicer now... Original commit message from CVS: conform to the buffer-frames props entry -- much nicer now... --- ext/jack/gstjack.c | 313 +++++++++++++++++++----------------------- ext/jack/gstjack.h | 3 +- ext/jack/gstjackbin.c | 14 +- 3 files changed, 160 insertions(+), 170 deletions(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 6e057f6cd4..4835813a29 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -1,6 +1,6 @@ /* -*- Mode: C; c-basic-offset: 4 -*- */ /* - Copyright (C) 2002 Andy Wingo + Copyright (C) 2002, 2003 Andy Wingo This library is free software; you can redistribute it and/or modify it under the terms of the GNU General Public @@ -27,8 +27,6 @@ /* TODO: - this element is still nonfunctional - - work out the src side (caps setting, etc) future core TODO: @@ -43,51 +41,51 @@ static GstElementDetails gst_jack_bin_details = { "Jack Bin", "Generic/Bin", "GPL", - "Jack processing bin: see README for more info", + "Jack processing bin", VERSION, "Andy Wingo ", - "(C) 2002 " + "(C) 2002, 2003" }; static GstElementDetails gst_jack_sink_details = { "Jack Sink", "Sink/Audio", "GPL", - "Output to a Jack processing network: see README for more info", + "Output to a Jack processing network", VERSION, "Andy Wingo ", - "(C) 2002 " + "(C) 2002, 2003" }; static GstElementDetails gst_jack_src_details = { "Jack Src", "Source/Audio", "GPL", - "Input from a Jack processing network: see README for more info", + "Input from a Jack processing network", VERSION, "Andy Wingo ", - "(C) 2002", + "(C) 2002, 2003", }; static GHashTable *port_name_counts = NULL; static GstElementClass *parent_class = NULL; -static void gst_jack_init(GstJack *this); -static void gst_jack_class_init(GstJackClass *klass); +static void gst_jack_init(GstJack *this); +static void gst_jack_class_init(GstJackClass *klass); +static void gst_jack_set_property (GObject *object, guint prop_id, + const GValue *value, GParamSpec *pspec); +static void gst_jack_get_property (GObject *object, guint prop_id, + GValue *value, GParamSpec *pspec); -static GstPadTemplate *gst_jack_src_request_pad_factory(); -static GstPadTemplate *gst_jack_sink_request_pad_factory(); +static GstPadTemplate* gst_jack_src_request_pad_factory(); +static GstPadTemplate* gst_jack_sink_request_pad_factory(); +static GstPad* gst_jack_request_new_pad (GstElement *element, GstPadTemplate *templ, + const gchar *name); +static GstElementStateReturn gst_jack_change_state (GstElement *element); +static GstPadLinkReturn gst_jack_link (GstPad *pad, GstCaps *caps); -static GstPad* gst_jack_request_new_pad (GstElement *element, GstPadTemplate *templ, const - gchar *name); - -static void gst_jack_set_property(GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec); -static void gst_jack_get_property(GObject *object, guint prop_id, GValue *value, GParamSpec *pspec); -static GstElementStateReturn gst_jack_change_state(GstElement *element); -static GstPadLinkReturn gst_jack_connect (GstPad *pad, GstCaps *caps); - -static void gst_jack_loop (GstElement *element); +static void gst_jack_loop (GstElement *element); enum { @@ -162,45 +160,12 @@ gst_jack_src_get_type (void) return jack_type; } -static GstPadTemplate* -gst_jack_src_request_pad_factory(void) -{ - static GstPadTemplate *template = NULL; - - if (!template) { - GstCaps *caps; - caps = gst_caps_new("src", - "audio/x-raw-float", - GST_AUDIO_FLOAT_STANDARD_PAD_TEMPLATE_PROPS); - template = gst_pad_template_new("%s", GST_PAD_SRC, - GST_PAD_REQUEST, caps, NULL); - } - - return template; -} - -static GstPadTemplate* -gst_jack_sink_request_pad_factory(void) -{ - static GstPadTemplate *template = NULL; - - if (!template) { - GstCaps *caps; - caps = gst_caps_new ("sink", - "audio/x-raw-float", - GST_AUDIO_FLOAT_STANDARD_PAD_TEMPLATE_PROPS); - template = gst_pad_template_new("%s", GST_PAD_SINK, - GST_PAD_REQUEST, caps, NULL); - } - - return template; -} - static void gst_jack_class_init(GstJackClass *klass) { GObjectClass *object_class; GstElementClass *element_class; + GParamSpec *pspec; gchar *prefix; object_class = (GObjectClass *)klass; @@ -217,11 +182,10 @@ gst_jack_class_init(GstJackClass *klass) else prefix = "gst-in-"; - g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_PORT_NAME_PREFIX, - g_param_spec_string("port-name-prefix","Port name prefix", - "String to prepend to jack port names", - prefix, - G_PARAM_READWRITE | G_PARAM_CONSTRUCT)); + pspec = g_param_spec_string ("port-name-prefix", "Port name prefix", + "String to prepend to jack port names", + prefix, G_PARAM_READWRITE | G_PARAM_CONSTRUCT); + g_object_class_install_property (object_class, ARG_PORT_NAME_PREFIX, pspec); element_class->change_state = gst_jack_change_state; @@ -231,15 +195,80 @@ gst_jack_class_init(GstJackClass *klass) static void gst_jack_init(GstJack *this) { - if (G_OBJECT_TYPE (this) == GST_TYPE_JACK_SRC) { + if (G_OBJECT_TYPE (this) == GST_TYPE_JACK_SRC) this->direction = GST_PAD_SRC; - } else if (G_OBJECT_TYPE (this) == GST_TYPE_JACK_SINK) { + else if (G_OBJECT_TYPE (this) == GST_TYPE_JACK_SINK) this->direction = GST_PAD_SINK; - } else { + else g_assert_not_reached (); + + gst_element_set_loop_function (GST_ELEMENT (this), gst_jack_loop); +} + +static void +gst_jack_set_property(GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec) +{ + GstJack *this = (GstJack*)object; + + switch (prop_id) { + case ARG_PORT_NAME_PREFIX: + if (this->port_name_prefix) + g_free (this->port_name_prefix); + this->port_name_prefix = g_strdup (g_value_get_string (value)); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + return; + } +} + +static void +gst_jack_get_property(GObject *object, guint prop_id, GValue *value, GParamSpec *pspec) +{ + GstJack *this = (GstJack*)object; + + switch (prop_id) { + case ARG_PORT_NAME_PREFIX: + g_value_set_string (value, this->port_name_prefix); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static GstPadTemplate* +gst_jack_src_request_pad_factory (void) +{ + static GstPadTemplate *template = NULL; + + if (!template) { + GstCaps *caps; + caps = gst_caps_new ("src", + "audio/x-raw-float", + GST_AUDIO_FLOAT_STANDARD_PAD_TEMPLATE_PROPS); + template = gst_pad_template_new ("%s", GST_PAD_SRC, + GST_PAD_REQUEST, caps, NULL); } - gst_element_set_loop_function(GST_ELEMENT(this), gst_jack_loop); + return template; +} + +static GstPadTemplate* +gst_jack_sink_request_pad_factory (void) +{ + static GstPadTemplate *template = NULL; + + if (!template) { + GstCaps *caps; + caps = gst_caps_new ("sink", + "audio/x-raw-float", + GST_AUDIO_FLOAT_STANDARD_PAD_TEMPLATE_PROPS); + template = gst_pad_template_new ("%s", GST_PAD_SINK, + GST_PAD_REQUEST, caps, NULL); + } + + return template; } static GstPad* @@ -277,7 +306,7 @@ gst_jack_request_new_pad (GstElement *element, GstPadTemplate *templ, const gcha newname = g_strdup ("alsa_pcm:capture_1"); } - pad = g_new0(GstJackPad, 1); + pad = g_new0 (GstJackPad, 1); if (!port_name_counts) port_name_counts = g_hash_table_new (g_str_hash, g_str_equal); @@ -290,7 +319,7 @@ gst_jack_request_new_pad (GstElement *element, GstPadTemplate *templ, const gcha pad->peer_name = newname; pad->pad = gst_pad_new_from_template (templ, newname); gst_element_add_pad (GST_ELEMENT (this), pad->pad); - gst_pad_set_link_function (pad->pad, gst_jack_connect); + gst_pad_set_link_function (pad->pad, gst_jack_link); this->pads = g_list_append (this->pads, pad); @@ -298,41 +327,6 @@ gst_jack_request_new_pad (GstElement *element, GstPadTemplate *templ, const gcha return pad->pad; } -static void -gst_jack_set_property(GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec) -{ - GstJack *this; - - this = (GstJack *)object; - switch (prop_id) { - case ARG_PORT_NAME_PREFIX: - if (this->port_name_prefix) - g_free (this->port_name_prefix); - this->port_name_prefix = g_strdup (g_value_get_string (value)); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - return; - } -} - -static void -gst_jack_get_property(GObject *object, guint prop_id, GValue *value, GParamSpec *pspec) -{ - GstJack *this; - - this = (GstJack *)object; - - switch (prop_id) { - case ARG_PORT_NAME_PREFIX: - g_value_set_string (value, this->port_name_prefix); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - static GstElementStateReturn gst_jack_change_state (GstElement *element) { @@ -358,7 +352,7 @@ gst_jack_change_state (GstElement *element) || !GST_IS_JACK_BIN (this->bin)) { this->bin = NULL; g_warning ("jack element %s needs to be contained in a jack bin.", - GST_OBJECT_NAME (GST_OBJECT (element))); + GST_OBJECT_NAME (element)); return GST_STATE_FAILURE; } @@ -383,7 +377,10 @@ gst_jack_change_state (GstElement *element) while (l) { pad = GST_JACK_PAD (l); caps = gst_pad_get_caps (pad->pad); - gst_caps_set (caps, "rate", GST_PROPS_INT_TYPE, (gint) this->bin->rate, NULL); + gst_caps_set (caps, "rate", GST_PROPS_INT_TYPE, + (gint)this->bin->rate, NULL); + gst_caps_set (caps, "buffer-frames", GST_PROPS_INT_TYPE, + (gint)this->bin->nframes, NULL); if (gst_pad_try_set_caps (pad->pad, caps) <= 0) return GST_STATE_FAILURE; l = g_list_next (l); @@ -404,18 +401,18 @@ gst_jack_change_state (GstElement *element) } static GstPadLinkReturn -gst_jack_connect (GstPad *pad, GstCaps *caps) +gst_jack_link (GstPad *pad, GstCaps *caps) { GstJack *this; - gint rate; + gint rate, buffer_frames; - this = GST_JACK (gst_pad_get_parent (pad)); - g_return_val_if_fail (this != NULL, GST_PAD_LINK_REFUSED); - g_return_val_if_fail (GST_IS_JACK (this), GST_PAD_LINK_REFUSED); + this = GST_JACK (GST_OBJECT_PARENT (pad)); if (GST_CAPS_IS_FIXED (caps)) { gst_caps_get_int (caps, "rate", &rate); - if (this->bin && rate != this->bin->rate) + gst_caps_get_int (caps, "buffer-frames", &buffer_frames); + if (this->bin && (rate != this->bin->rate || + buffer_frames != this->bin->nframes)) return GST_PAD_LINK_REFUSED; return GST_PAD_LINK_OK; @@ -429,69 +426,54 @@ gst_jack_loop (GstElement *element) { GstJack *this; GList *pads; - gint len, peeked_len; - guint8 *peeked; - gint avail; - GstEvent *event; + gint len; GstJackPad *pad; GstBuffer *buffer; this = GST_JACK (element); - g_return_if_fail(this != NULL); - len = this->bin->nframes * sizeof (jack_default_audio_sample_t); + len = this->bin->nframes * sizeof (sample_t); - do { - pads = this->pads; - while (pads) { - pad = GST_JACK_PAD (pads); + pads = this->pads; + while (pads) { + pad = GST_JACK_PAD (pads); - if (this->direction == GST_PAD_SINK) { - if (!pad->bs) - pad->bs = gst_bytestream_new (pad->pad); + if (this->direction == GST_PAD_SINK) { + buffer = gst_pad_pull (pad->pad); - read: - peeked_len = gst_bytestream_peek_bytes (pad->bs, &peeked, len); - if (peeked_len < len) { - gst_bytestream_get_status(pad->bs, &avail, &event); - if (event) { - g_warning("got an event on jacksink"); - if (GST_EVENT_TYPE (event) == GST_EVENT_EOS) { - /* really, we should just cut this pad out of the graph. let - * me know when this is needed ;) - * also, for sample accuracy etc, we should play avail - * bytes, but hey. */ - gst_element_set_eos (element); - gst_event_unref (event); -// gst_element_yield (element); /* shouldn't return */ - return; - } - goto read; - } else { - /* the element at the top of the chain did not emit an eos - * event. this is a Bug(tm) */ - g_assert_not_reached(); - } + if (GST_IS_EVENT (buffer)) { + GstEvent *event = GST_EVENT (buffer); + switch (GST_EVENT_TYPE (buffer)) { + case GST_EVENT_EOS: + gst_element_set_eos (element); + gst_event_unref (event); + return; + default: + gst_pad_event_default (pad->pad, event); + return; } - - - memcpy (pad->data, peeked, peeked_len); - gst_bytestream_flush (pad->bs, peeked_len); - } else { - buffer = gst_buffer_new (); - GST_BUFFER_DATA (buffer) = pad->data; - GST_BUFFER_SIZE (buffer) = len; - GST_BUFFER_MAXSIZE (buffer) = len; - GST_BUFFER_FLAG_SET(buffer, GST_BUFFER_DONTFREE); - - gst_pad_push (pad->pad, buffer); } - pads = g_list_next (pads); + + /* if the other plugins only give out buffer-frames or less (as + they should), if the length of the GstBuffer is different + from nframes then the buffer is short and we will get EOS + next */ + memcpy (pad->data, GST_BUFFER_DATA (buffer), + GST_BUFFER_SIZE (buffer)); + if (len != GST_BUFFER_SIZE (buffer)) + memset (pad->data + GST_BUFFER_SIZE (buffer), 0, + len - GST_BUFFER_SIZE (buffer)); + + gst_buffer_unref (buffer); + } else { + buffer = gst_buffer_new (); + gst_buffer_set_data (buffer, pad->data, len); + GST_BUFFER_FLAG_SET(buffer, GST_BUFFER_DONTFREE); + + gst_pad_push (pad->pad, buffer); } - - return; -// gst_element_yield (element); - } while (TRUE); + pads = g_list_next (pads); + } } static gboolean @@ -499,9 +481,6 @@ plugin_init (GModule *module, GstPlugin *plugin) { GstElementFactory *factory; - if (!gst_library_load ("gstbytestream")) - return FALSE; - factory = gst_element_factory_new ("jackbin", GST_TYPE_JACK_BIN, &gst_jack_bin_details); g_return_val_if_fail (factory != NULL, FALSE); gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory)); @@ -516,7 +495,7 @@ plugin_init (GModule *module, GstPlugin *plugin) gst_element_factory_add_pad_template (factory, gst_jack_sink_request_pad_factory()); gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory)); - gst_plugin_set_longname(plugin, "JACK plugin library"); + gst_plugin_set_longname (plugin, "JACK plugin library"); return TRUE; } @@ -527,5 +506,3 @@ GstPluginDesc plugin_desc = { "jack", plugin_init }; - - diff --git a/ext/jack/gstjack.h b/ext/jack/gstjack.h index 216c305292..87d7d61151 100644 --- a/ext/jack/gstjack.h +++ b/ext/jack/gstjack.h @@ -71,12 +71,13 @@ enum { }; +typedef jack_default_audio_sample_t sample_t; + typedef struct { GstPad *pad; void *data; const gchar *name; const gchar *peer_name; - GstByteStream *bs; jack_port_t *port; } GstJackPad; diff --git a/ext/jack/gstjackbin.c b/ext/jack/gstjackbin.c index 959f58673a..01713a70cb 100644 --- a/ext/jack/gstjackbin.c +++ b/ext/jack/gstjackbin.c @@ -1,6 +1,6 @@ /* -*- Mode: C; c-basic-offset: 4 -*- */ /* - Copyright (C) 2002 Andy Wingo + Copyright (C) 2002, 2003 Andy Wingo This library is free software; you can redistribute it and/or modify it under the terms of the GNU General Public @@ -33,6 +33,7 @@ static GstElementStateReturn gst_jack_bin_change_state(GstElement *element); /* jack callbacks */ static int process (jack_nframes_t nframes, void *arg); static int sample_rate (jack_nframes_t nframes, void *arg); +static int buffer_size (jack_nframes_t nframes, void *arg); static void shutdown (void *arg); static void sighup_handler (int sig); @@ -133,6 +134,8 @@ gst_jack_bin_change_state (GstElement *element) jack_set_process_callback (this->client, process, this); jack_set_sample_rate_callback (this->client, sample_rate, this); + jack_set_buffer_size_callback (this->client, buffer_size, this); + this->nframes = jack_get_buffer_size (this->client); jack_on_shutdown (this->client, shutdown, this); } @@ -291,6 +294,15 @@ sample_rate (jack_nframes_t nframes, void *arg) return 0; } +static int +buffer_size (jack_nframes_t nframes, void *arg) +{ + GstJackBin *bin = (GstJackBin*) arg; + JACK_DEBUG ("the buffer size is now %lu\n", nframes); + bin->nframes = nframes; + return 0; +} + static void shutdown (void *arg) { From fc1b8b479fb0b6ed02303dfe49fa606997eaf13e Mon Sep 17 00:00:00 2001 From: "Ronald S. Bultje" Date: Wed, 1 Oct 2003 13:14:50 +0000 Subject: [PATCH 147/351] New typefind system: bytestream is now part of the core all plugins have been modified to use this new typefind syste... Original commit message from CVS: New typefind system: * bytestream is now part of the core * all plugins have been modified to use this new typefind system * asf typefinding added * mpeg video stream typefiding removed because it's broken * duplicate typefind entries removed * extra id3 typefinding added, because we've seen 4 types of files (riff/wav, flac, vorbis, mp3) with id3 headers and each of these needs to work. Instead, I've added an id3 element and let it redo typefiding after the id3 header. this needs a hack because spider only typefinds once. We can remove this hack once spider supports multiple typefinds. * with all this, mp3 typefinding is semi-rewritten * id3 typefinding in flac/vorbis is removed, it's no longer needed * fixed spider and gst-typefind to use this, too. * Other general cleanups --- ext/jack/gstjack.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/ext/jack/gstjack.h b/ext/jack/gstjack.h index 87d7d61151..80fd2b63a1 100644 --- a/ext/jack/gstjack.h +++ b/ext/jack/gstjack.h @@ -22,7 +22,7 @@ #include #include -#include +#include //#define JACK_DEBUG(str, a...) g_message (str, ##a) #define JACK_DEBUG(str, a...) From 45aedd595ec04aa8818f47fecf3bc031ceac4297 Mon Sep 17 00:00:00 2001 From: Andy Wingo Date: Wed, 8 Oct 2003 16:08:19 +0000 Subject: [PATCH 148/351] /GstBuffer/GstData/ in the API where you can pass events. Fix the plugins to deal with that. Fixes #113488. Original commit message from CVS: /GstBuffer/GstData/ in the API where you can pass events. Fix the plugins to deal with that. Fixes #113488. --- ext/jack/gstjack.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 4835813a29..98ec91d27e 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -439,7 +439,7 @@ gst_jack_loop (GstElement *element) pad = GST_JACK_PAD (pads); if (this->direction == GST_PAD_SINK) { - buffer = gst_pad_pull (pad->pad); + buffer = GST_BUFFER (gst_pad_pull (pad->pad)); if (GST_IS_EVENT (buffer)) { GstEvent *event = GST_EVENT (buffer); @@ -470,7 +470,7 @@ gst_jack_loop (GstElement *element) gst_buffer_set_data (buffer, pad->data, len); GST_BUFFER_FLAG_SET(buffer, GST_BUFFER_DONTFREE); - gst_pad_push (pad->pad, buffer); + gst_pad_push (pad->pad, GST_DATA (buffer)); } pads = g_list_next (pads); } From bbc133ec8f8829d8f66593264cf78c32ef5ff1e1 Mon Sep 17 00:00:00 2001 From: Benjamin Otte Date: Tue, 28 Oct 2003 20:52:41 +0000 Subject: [PATCH 149/351] merge TYPEFIND branch. Major changes: Original commit message from CVS: merge TYPEFIND branch. Major changes: - totally reworked type(find) system - all typefind functions are in gst/typefind now - more typefind functions then before - some plugins might fail to compile now because I don't have them installed and they a) require bytestream or b) haven't had their typefind fixed. Please fix those plugins and put the typefind functions into gst/typefind if they don't have dependencies --- ext/jack/gstjack.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/ext/jack/gstjack.h b/ext/jack/gstjack.h index 80fd2b63a1..2bb13858e4 100644 --- a/ext/jack/gstjack.h +++ b/ext/jack/gstjack.h @@ -22,7 +22,7 @@ #include #include -#include +#include //#define JACK_DEBUG(str, a...) g_message (str, ##a) #define JACK_DEBUG(str, a...) From bc5b62d93880bff660e0cc06ca50ce4ce8d7e829 Mon Sep 17 00:00:00 2001 From: David Schleef Date: Wed, 29 Oct 2003 03:15:55 +0000 Subject: [PATCH 150/351] change gst/bytestream.h to gst/bytestream/bytestream.h Original commit message from CVS: change gst/bytestream.h to gst/bytestream/bytestream.h --- ext/jack/gstjack.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/ext/jack/gstjack.h b/ext/jack/gstjack.h index 2bb13858e4..87d7d61151 100644 --- a/ext/jack/gstjack.h +++ b/ext/jack/gstjack.h @@ -22,7 +22,7 @@ #include #include -#include +#include //#define JACK_DEBUG(str, a...) g_message (str, ##a) #define JACK_DEBUG(str, a...) From 04c63f1f5cf7511784733362dfed80c777c9c20b Mon Sep 17 00:00:00 2001 From: Iain Holmes Date: Sat, 1 Nov 2003 23:43:13 +0000 Subject: [PATCH 151/351] Jack fixed too Original commit message from CVS: Jack fixed too --- ext/jack/gstjack.c | 154 +++++++++++++++++++++++++-------------------- 1 file changed, 85 insertions(+), 69 deletions(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 98ec91d27e..3aff6d7cdb 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -40,37 +40,31 @@ static GstElementDetails gst_jack_bin_details = { "Jack Bin", "Generic/Bin", - "GPL", "Jack processing bin", - VERSION, "Andy Wingo ", - "(C) 2002, 2003" }; static GstElementDetails gst_jack_sink_details = { "Jack Sink", "Sink/Audio", - "GPL", "Output to a Jack processing network", - VERSION, "Andy Wingo ", - "(C) 2002, 2003" }; static GstElementDetails gst_jack_src_details = { "Jack Src", "Source/Audio", - "GPL", "Input from a Jack processing network", - VERSION, "Andy Wingo ", - "(C) 2002, 2003", }; static GHashTable *port_name_counts = NULL; static GstElementClass *parent_class = NULL; +static void gst_jack_base_init (gpointer g_class); +static void gst_jack_src_base_init (gpointer g_class); +static void gst_jack_sink_base_init (gpointer g_class); static void gst_jack_init(GstJack *this); static void gst_jack_class_init(GstJackClass *klass); static void gst_jack_set_property (GObject *object, guint prop_id, @@ -102,7 +96,7 @@ gst_jack_get_type (void) if (!jack_type) { static const GTypeInfo jack_info = { sizeof(GstJackClass), - NULL, + gst_jack_base_init, NULL, NULL, NULL, @@ -119,45 +113,71 @@ gst_jack_get_type (void) GType gst_jack_sink_get_type (void) { - static GType jack_type = 0; - - if (!jack_type) { - static const GTypeInfo jack_info = { - sizeof(GstJackClass), - NULL, - NULL, - (GClassInitFunc)gst_jack_class_init, - NULL, - NULL, - sizeof(GstJack), - 0, - (GInstanceInitFunc)gst_jack_init, - }; - jack_type = g_type_register_static (GST_TYPE_JACK, "GstJackSink", &jack_info, 0); - } - return jack_type; + static GType jack_type = 0; + + if (!jack_type) { + static const GTypeInfo jack_info = { + sizeof(GstJackClass), + gst_jack_sink_base_init, + NULL, + (GClassInitFunc)gst_jack_class_init, + NULL, + NULL, + sizeof(GstJack), + 0, + (GInstanceInitFunc)gst_jack_init, + }; + jack_type = g_type_register_static (GST_TYPE_JACK, "GstJackSink", &jack_info, 0); + } + return jack_type; } GType gst_jack_src_get_type (void) { - static GType jack_type = 0; - - if (!jack_type) { - static const GTypeInfo jack_info = { - sizeof(GstJackClass), - NULL, - NULL, - (GClassInitFunc)gst_jack_class_init, - NULL, - NULL, - sizeof(GstJack), - 0, - (GInstanceInitFunc)gst_jack_init, - }; - jack_type = g_type_register_static (GST_TYPE_JACK, "GstJackSrc", &jack_info, 0); - } - return jack_type; + static GType jack_type = 0; + + if (!jack_type) { + static const GTypeInfo jack_info = { + sizeof(GstJackClass), + gst_jack_src_base_init, + NULL, + (GClassInitFunc)gst_jack_class_init, + NULL, + NULL, + sizeof(GstJack), + 0, + (GInstanceInitFunc)gst_jack_init, + }; + jack_type = g_type_register_static (GST_TYPE_JACK, "GstJackSrc", &jack_info, 0); + } + return jack_type; +} + +static void +gst_jack_base_init (gpointer g_class) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); + + gst_element_class_set_details (element_class, &gst_jack_bin_details); +} + +static void +gst_jack_src_base_init (gpointer g_class) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); + + gst_element_class_add_pad_template (element_class, gst_jack_src_request_pad_factory ()); + gst_element_class_set_details (element_class, &gst_jack_src_details); +} + +static void +gst_jack_sink_base_init (gpointer g_class) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); + + gst_element_class_add_pad_template (element_class, gst_jack_sink_request_pad_factory ()); + gst_element_class_set_details (element_class, &gst_jack_sink_details); } static void @@ -477,32 +497,28 @@ gst_jack_loop (GstElement *element) } static gboolean -plugin_init (GModule *module, GstPlugin *plugin) +plugin_init (GstPlugin *plugin) { - GstElementFactory *factory; + if (!gst_element_register (plugin, "jackbin", GST_RANK_NONE, GST_TYPE_JACK_BIN)) + return FALSE; + + if (!gst_element_register (plugin, "jacksrc", GST_RANK_NONE, GST_TYPE_JACK_SRC)) + return FALSE; + + if (!gst_element_register (plugin, "jacksink", GST_RANK_NONE, GST_TYPE_JACK_SINK)) + return FALSE; - factory = gst_element_factory_new ("jackbin", GST_TYPE_JACK_BIN, &gst_jack_bin_details); - g_return_val_if_fail (factory != NULL, FALSE); - gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory)); - - factory = gst_element_factory_new ("jacksrc", GST_TYPE_JACK_SRC, &gst_jack_src_details); - g_return_val_if_fail (factory != NULL, FALSE); - gst_element_factory_add_pad_template (factory, gst_jack_src_request_pad_factory()); - gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory)); - - factory = gst_element_factory_new ("jacksink", GST_TYPE_JACK_SINK, &gst_jack_sink_details); - g_return_val_if_fail (factory != NULL, FALSE); - gst_element_factory_add_pad_template (factory, gst_jack_sink_request_pad_factory()); - gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory)); - - gst_plugin_set_longname (plugin, "JACK plugin library"); - - return TRUE; + return TRUE; } -GstPluginDesc plugin_desc = { - GST_VERSION_MAJOR, - GST_VERSION_MINOR, - "jack", - plugin_init -}; +GST_PLUGIN_DEFINE ( + GST_VERSION_MAJOR, + GST_VERSION_MINOR, + "jack", + "Jack Plugin Library", + plugin_init, + VERSION, + "GPL", + GST_COPYRIGHT, + GST_PACKAGE, + GST_ORIGIN) From 56906e2c2e366f0b7692f91c0cfbb17cd44c588f Mon Sep 17 00:00:00 2001 From: "Ronald S. Bultje" Date: Fri, 7 Nov 2003 12:47:02 +0000 Subject: [PATCH 152/351] Remove all config.h includes from header files, add it to each source file and remove duplicate config.h includes fro... Original commit message from CVS: Remove all config.h includes from header files, add it to each source file and remove duplicate config.h includes from several source files --- ext/jack/gstjackbin.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/ext/jack/gstjackbin.c b/ext/jack/gstjackbin.c index 01713a70cb..cb1839c41c 100644 --- a/ext/jack/gstjackbin.c +++ b/ext/jack/gstjackbin.c @@ -17,6 +17,10 @@ Software Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. */ +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + #include #include #include From d8b76118bd30d6274d066ace328471728f1e2747 Mon Sep 17 00:00:00 2001 From: Andy Wingo Date: Thu, 4 Dec 2003 10:37:38 +0000 Subject: [PATCH 153/351] remove copyright field from plugins Original commit message from CVS: remove copyright field from plugins --- ext/jack/gstjack.c | 1 - 1 file changed, 1 deletion(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 3aff6d7cdb..85d57658c6 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -519,6 +519,5 @@ GST_PLUGIN_DEFINE ( plugin_init, VERSION, "GPL", - GST_COPYRIGHT, GST_PACKAGE, GST_ORIGIN) From 93cb77d0f0bf8410db9b8bc87c9b8515cf1c9d75 Mon Sep 17 00:00:00 2001 From: Benjamin Otte Date: Sat, 13 Dec 2003 16:59:51 +0000 Subject: [PATCH 154/351] removed GST_*_CAST. Disabling of type checking is done in glib. Original commit message from CVS: removed GST_*_CAST. Disabling of type checking is done in glib. --- ext/jack/gstjackbin.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/ext/jack/gstjackbin.c b/ext/jack/gstjackbin.c index cb1839c41c..b0cea9bc32 100644 --- a/ext/jack/gstjackbin.c +++ b/ext/jack/gstjackbin.c @@ -279,7 +279,7 @@ process (jack_nframes_t nframes, void *arg) bin->nframes = nframes; JACK_DEBUG ("jackbin: iterating to process %ld frames of audio...", nframes); - if (!gst_bin_iterate (GST_BIN_CAST (bin))) { + if (!gst_bin_iterate (GST_BIN (bin))) { g_warning ("bin failed to iterate"); return -1; } From 9d2631e7d6d73f6dd9d6a200694b7baffd786ccd Mon Sep 17 00:00:00 2001 From: David Schleef Date: Mon, 22 Dec 2003 01:47:09 +0000 Subject: [PATCH 155/351] Merge CAPS branch Original commit message from CVS: Merge CAPS branch --- ext/jack/gstjack.c | 42 ++++++++++++++++++------------------------ 1 file changed, 18 insertions(+), 24 deletions(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 85d57658c6..4a987ae17b 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -77,7 +77,7 @@ static GstPadTemplate* gst_jack_sink_request_pad_factory(); static GstPad* gst_jack_request_new_pad (GstElement *element, GstPadTemplate *templ, const gchar *name); static GstElementStateReturn gst_jack_change_state (GstElement *element); -static GstPadLinkReturn gst_jack_link (GstPad *pad, GstCaps *caps); +static GstPadLinkReturn gst_jack_link (GstPad *pad, const GstCaps *caps); static void gst_jack_loop (GstElement *element); @@ -264,11 +264,9 @@ gst_jack_src_request_pad_factory (void) if (!template) { GstCaps *caps; - caps = gst_caps_new ("src", - "audio/x-raw-float", - GST_AUDIO_FLOAT_STANDARD_PAD_TEMPLATE_PROPS); + caps = gst_caps_from_string (GST_AUDIO_FLOAT_STANDARD_PAD_TEMPLATE_CAPS); template = gst_pad_template_new ("%s", GST_PAD_SRC, - GST_PAD_REQUEST, caps, NULL); + GST_PAD_REQUEST, caps); } return template; @@ -281,11 +279,9 @@ gst_jack_sink_request_pad_factory (void) if (!template) { GstCaps *caps; - caps = gst_caps_new ("sink", - "audio/x-raw-float", - GST_AUDIO_FLOAT_STANDARD_PAD_TEMPLATE_PROPS); + caps = gst_caps_from_string (GST_AUDIO_FLOAT_STANDARD_PAD_TEMPLATE_CAPS); template = gst_pad_template_new ("%s", GST_PAD_SINK, - GST_PAD_REQUEST, caps, NULL); + GST_PAD_REQUEST, caps); } return template; @@ -397,10 +393,10 @@ gst_jack_change_state (GstElement *element) while (l) { pad = GST_JACK_PAD (l); caps = gst_pad_get_caps (pad->pad); - gst_caps_set (caps, "rate", GST_PROPS_INT_TYPE, - (gint)this->bin->rate, NULL); - gst_caps_set (caps, "buffer-frames", GST_PROPS_INT_TYPE, - (gint)this->bin->nframes, NULL); + gst_caps_set_simple (caps, + "rate", G_TYPE_INT, (int)this->bin->rate, + "buffer-frames", G_TYPE_INT, (gint)this->bin->nframes, + NULL); if (gst_pad_try_set_caps (pad->pad, caps) <= 0) return GST_STATE_FAILURE; l = g_list_next (l); @@ -421,24 +417,22 @@ gst_jack_change_state (GstElement *element) } static GstPadLinkReturn -gst_jack_link (GstPad *pad, GstCaps *caps) +gst_jack_link (GstPad *pad, const GstCaps *caps) { GstJack *this; gint rate, buffer_frames; + GstStructure *structure; this = GST_JACK (GST_OBJECT_PARENT (pad)); - if (GST_CAPS_IS_FIXED (caps)) { - gst_caps_get_int (caps, "rate", &rate); - gst_caps_get_int (caps, "buffer-frames", &buffer_frames); - if (this->bin && (rate != this->bin->rate || - buffer_frames != this->bin->nframes)) - return GST_PAD_LINK_REFUSED; - - return GST_PAD_LINK_OK; - } + structure = gst_caps_get_structure (caps, 0); + gst_structure_get_int (structure, "rate", &rate); + gst_structure_get_int (structure, "buffer-frames", &buffer_frames); + if (this->bin && (rate != this->bin->rate || + buffer_frames != this->bin->nframes)) + return GST_PAD_LINK_REFUSED; - return GST_PAD_LINK_DELAYED; + return GST_PAD_LINK_OK; } static void From d952e72d42bf32bb4f7ef87cf2e6bed003eb86ee Mon Sep 17 00:00:00 2001 From: David Schleef Date: Mon, 12 Jan 2004 03:40:18 +0000 Subject: [PATCH 156/351] Remove all usage of gst_pad_get_caps(), and replace it with gst_pad_get_allowed_caps() or gst_pad_get_negotiated_cap(). Original commit message from CVS: Remove all usage of gst_pad_get_caps(), and replace it with gst_pad_get_allowed_caps() or gst_pad_get_negotiated_cap(). --- ext/jack/gstjack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 4a987ae17b..a43c6707ff 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -392,7 +392,7 @@ gst_jack_change_state (GstElement *element) l = this->pads; while (l) { pad = GST_JACK_PAD (l); - caps = gst_pad_get_caps (pad->pad); + caps = gst_caps_copy (gst_pad_get_negotiated_caps (pad->pad)); gst_caps_set_simple (caps, "rate", G_TYPE_INT, (int)this->bin->rate, "buffer-frames", G_TYPE_INT, (gint)this->bin->nframes, From c2bcfc8446aecdde3628bec0e3ffe34340cc1f58 Mon Sep 17 00:00:00 2001 From: Thomas Vander Stichele Date: Sun, 14 Mar 2004 22:34:33 +0000 Subject: [PATCH 157/351] gst-indent Original commit message from CVS: gst-indent --- ext/jack/gstjack.c | 632 +++++++++++++++++++++--------------------- ext/jack/gstjack.h | 72 ++--- ext/jack/gstjackbin.c | 448 ++++++++++++++++-------------- 3 files changed, 595 insertions(+), 557 deletions(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index a43c6707ff..41f078bc44 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -38,118 +38,122 @@ /* elementfactory information */ static GstElementDetails gst_jack_bin_details = { - "Jack Bin", - "Generic/Bin", - "Jack processing bin", - "Andy Wingo ", + "Jack Bin", + "Generic/Bin", + "Jack processing bin", + "Andy Wingo ", }; -static GstElementDetails gst_jack_sink_details = { - "Jack Sink", - "Sink/Audio", - "Output to a Jack processing network", - "Andy Wingo ", +static GstElementDetails gst_jack_sink_details = { + "Jack Sink", + "Sink/Audio", + "Output to a Jack processing network", + "Andy Wingo ", }; -static GstElementDetails gst_jack_src_details = { - "Jack Src", - "Source/Audio", - "Input from a Jack processing network", - "Andy Wingo ", +static GstElementDetails gst_jack_src_details = { + "Jack Src", + "Source/Audio", + "Input from a Jack processing network", + "Andy Wingo ", }; static GHashTable *port_name_counts = NULL; static GstElementClass *parent_class = NULL; -static void gst_jack_base_init (gpointer g_class); -static void gst_jack_src_base_init (gpointer g_class); -static void gst_jack_sink_base_init (gpointer g_class); -static void gst_jack_init(GstJack *this); -static void gst_jack_class_init(GstJackClass *klass); -static void gst_jack_set_property (GObject *object, guint prop_id, - const GValue *value, GParamSpec *pspec); -static void gst_jack_get_property (GObject *object, guint prop_id, - GValue *value, GParamSpec *pspec); +static void gst_jack_base_init (gpointer g_class); +static void gst_jack_src_base_init (gpointer g_class); +static void gst_jack_sink_base_init (gpointer g_class); +static void gst_jack_init (GstJack * this); +static void gst_jack_class_init (GstJackClass * klass); +static void gst_jack_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec); +static void gst_jack_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec); -static GstPadTemplate* gst_jack_src_request_pad_factory(); -static GstPadTemplate* gst_jack_sink_request_pad_factory(); -static GstPad* gst_jack_request_new_pad (GstElement *element, GstPadTemplate *templ, - const gchar *name); -static GstElementStateReturn gst_jack_change_state (GstElement *element); -static GstPadLinkReturn gst_jack_link (GstPad *pad, const GstCaps *caps); +static GstPadTemplate *gst_jack_src_request_pad_factory (); +static GstPadTemplate *gst_jack_sink_request_pad_factory (); +static GstPad *gst_jack_request_new_pad (GstElement * element, + GstPadTemplate * templ, const gchar * name); +static GstElementStateReturn gst_jack_change_state (GstElement * element); +static GstPadLinkReturn gst_jack_link (GstPad * pad, const GstCaps * caps); -static void gst_jack_loop (GstElement *element); +static void gst_jack_loop (GstElement * element); -enum { - ARG_0, - ARG_PORT_NAME_PREFIX, +enum +{ + ARG_0, + ARG_PORT_NAME_PREFIX, }; GType -gst_jack_get_type (void) +gst_jack_get_type (void) { static GType jack_type = 0; if (!jack_type) { static const GTypeInfo jack_info = { - sizeof(GstJackClass), + sizeof (GstJackClass), gst_jack_base_init, NULL, NULL, NULL, NULL, - sizeof(GstJack), + sizeof (GstJack), 0, NULL, }; - jack_type = g_type_register_static (GST_TYPE_ELEMENT, "GstJack", &jack_info, 0); + jack_type = + g_type_register_static (GST_TYPE_ELEMENT, "GstJack", &jack_info, 0); } return jack_type; } GType -gst_jack_sink_get_type (void) +gst_jack_sink_get_type (void) { static GType jack_type = 0; - + if (!jack_type) { static const GTypeInfo jack_info = { - sizeof(GstJackClass), + sizeof (GstJackClass), gst_jack_sink_base_init, NULL, - (GClassInitFunc)gst_jack_class_init, + (GClassInitFunc) gst_jack_class_init, NULL, NULL, - sizeof(GstJack), + sizeof (GstJack), 0, - (GInstanceInitFunc)gst_jack_init, + (GInstanceInitFunc) gst_jack_init, }; - jack_type = g_type_register_static (GST_TYPE_JACK, "GstJackSink", &jack_info, 0); + jack_type = + g_type_register_static (GST_TYPE_JACK, "GstJackSink", &jack_info, 0); } return jack_type; } GType -gst_jack_src_get_type (void) +gst_jack_src_get_type (void) { static GType jack_type = 0; - + if (!jack_type) { static const GTypeInfo jack_info = { - sizeof(GstJackClass), + sizeof (GstJackClass), gst_jack_src_base_init, NULL, - (GClassInitFunc)gst_jack_class_init, + (GClassInitFunc) gst_jack_class_init, NULL, NULL, - sizeof(GstJack), + sizeof (GstJack), 0, - (GInstanceInitFunc)gst_jack_init, + (GInstanceInitFunc) gst_jack_init, }; - jack_type = g_type_register_static (GST_TYPE_JACK, "GstJackSrc", &jack_info, 0); + jack_type = + g_type_register_static (GST_TYPE_JACK, "GstJackSrc", &jack_info, 0); } return jack_type; } @@ -167,7 +171,8 @@ gst_jack_src_base_init (gpointer g_class) { GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); - gst_element_class_add_pad_template (element_class, gst_jack_src_request_pad_factory ()); + gst_element_class_add_pad_template (element_class, + gst_jack_src_request_pad_factory ()); gst_element_class_set_details (element_class, &gst_jack_src_details); } @@ -176,342 +181,349 @@ gst_jack_sink_base_init (gpointer g_class) { GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); - gst_element_class_add_pad_template (element_class, gst_jack_sink_request_pad_factory ()); + gst_element_class_add_pad_template (element_class, + gst_jack_sink_request_pad_factory ()); gst_element_class_set_details (element_class, &gst_jack_sink_details); } static void -gst_jack_class_init(GstJackClass *klass) +gst_jack_class_init (GstJackClass * klass) { - GObjectClass *object_class; - GstElementClass *element_class; - GParamSpec *pspec; - gchar *prefix; - - object_class = (GObjectClass *)klass; - element_class = (GstElementClass *)klass; - - if (parent_class == NULL) - parent_class = g_type_class_ref(GST_TYPE_ELEMENT); - - object_class->get_property = gst_jack_get_property; - object_class->set_property = gst_jack_set_property; - - if (GST_IS_JACK_SINK_CLASS (klass)) - prefix = "gst-out-"; - else - prefix = "gst-in-"; - - pspec = g_param_spec_string ("port-name-prefix", "Port name prefix", - "String to prepend to jack port names", - prefix, G_PARAM_READWRITE | G_PARAM_CONSTRUCT); - g_object_class_install_property (object_class, ARG_PORT_NAME_PREFIX, pspec); + GObjectClass *object_class; + GstElementClass *element_class; + GParamSpec *pspec; + gchar *prefix; - element_class->change_state = gst_jack_change_state; - - element_class->request_new_pad = gst_jack_request_new_pad; + object_class = (GObjectClass *) klass; + element_class = (GstElementClass *) klass; + + if (parent_class == NULL) + parent_class = g_type_class_ref (GST_TYPE_ELEMENT); + + object_class->get_property = gst_jack_get_property; + object_class->set_property = gst_jack_set_property; + + if (GST_IS_JACK_SINK_CLASS (klass)) + prefix = "gst-out-"; + else + prefix = "gst-in-"; + + pspec = g_param_spec_string ("port-name-prefix", "Port name prefix", + "String to prepend to jack port names", + prefix, G_PARAM_READWRITE | G_PARAM_CONSTRUCT); + g_object_class_install_property (object_class, ARG_PORT_NAME_PREFIX, pspec); + + element_class->change_state = gst_jack_change_state; + + element_class->request_new_pad = gst_jack_request_new_pad; } static void -gst_jack_init(GstJack *this) +gst_jack_init (GstJack * this) { - if (G_OBJECT_TYPE (this) == GST_TYPE_JACK_SRC) - this->direction = GST_PAD_SRC; - else if (G_OBJECT_TYPE (this) == GST_TYPE_JACK_SINK) - this->direction = GST_PAD_SINK; - else - g_assert_not_reached (); - - gst_element_set_loop_function (GST_ELEMENT (this), gst_jack_loop); + if (G_OBJECT_TYPE (this) == GST_TYPE_JACK_SRC) + this->direction = GST_PAD_SRC; + else if (G_OBJECT_TYPE (this) == GST_TYPE_JACK_SINK) + this->direction = GST_PAD_SINK; + else + g_assert_not_reached (); + + gst_element_set_loop_function (GST_ELEMENT (this), gst_jack_loop); } static void -gst_jack_set_property(GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec) +gst_jack_set_property (GObject * object, guint prop_id, const GValue * value, + GParamSpec * pspec) { - GstJack *this = (GstJack*)object; + GstJack *this = (GstJack *) object; - switch (prop_id) { + switch (prop_id) { case ARG_PORT_NAME_PREFIX: - if (this->port_name_prefix) - g_free (this->port_name_prefix); - this->port_name_prefix = g_strdup (g_value_get_string (value)); - break; + if (this->port_name_prefix) + g_free (this->port_name_prefix); + this->port_name_prefix = g_strdup (g_value_get_string (value)); + break; default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - return; - } + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + return; + } } static void -gst_jack_get_property(GObject *object, guint prop_id, GValue *value, GParamSpec *pspec) +gst_jack_get_property (GObject * object, guint prop_id, GValue * value, + GParamSpec * pspec) { - GstJack *this = (GstJack*)object; + GstJack *this = (GstJack *) object; - switch (prop_id) { + switch (prop_id) { case ARG_PORT_NAME_PREFIX: - g_value_set_string (value, this->port_name_prefix); - break; + g_value_set_string (value, this->port_name_prefix); + break; default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } } -static GstPadTemplate* +static GstPadTemplate * gst_jack_src_request_pad_factory (void) { - static GstPadTemplate *template = NULL; - - if (!template) { - GstCaps *caps; - caps = gst_caps_from_string (GST_AUDIO_FLOAT_STANDARD_PAD_TEMPLATE_CAPS); - template = gst_pad_template_new ("%s", GST_PAD_SRC, - GST_PAD_REQUEST, caps); - } - - return template; + static GstPadTemplate *template = NULL; + + if (!template) { + GstCaps *caps; + + caps = gst_caps_from_string (GST_AUDIO_FLOAT_STANDARD_PAD_TEMPLATE_CAPS); + template = gst_pad_template_new ("%s", GST_PAD_SRC, GST_PAD_REQUEST, caps); + } + + return template; } -static GstPadTemplate* +static GstPadTemplate * gst_jack_sink_request_pad_factory (void) { - static GstPadTemplate *template = NULL; - - if (!template) { - GstCaps *caps; - caps = gst_caps_from_string (GST_AUDIO_FLOAT_STANDARD_PAD_TEMPLATE_CAPS); - template = gst_pad_template_new ("%s", GST_PAD_SINK, - GST_PAD_REQUEST, caps); - } - - return template; + static GstPadTemplate *template = NULL; + + if (!template) { + GstCaps *caps; + + caps = gst_caps_from_string (GST_AUDIO_FLOAT_STANDARD_PAD_TEMPLATE_CAPS); + template = gst_pad_template_new ("%s", GST_PAD_SINK, GST_PAD_REQUEST, caps); + } + + return template; } -static GstPad* -gst_jack_request_new_pad (GstElement *element, GstPadTemplate *templ, const gchar *name) +static GstPad * +gst_jack_request_new_pad (GstElement * element, GstPadTemplate * templ, + const gchar * name) { - GstJack *this; - gchar *newname; - GList *l, **pad_list; - GstJackPad *pad; - gint count; - - g_return_val_if_fail ((this = GST_JACK (element)), NULL); - - if (!this->bin) - pad_list = &this->pads; - else if (this->direction == GST_PAD_SRC) - pad_list = &this->bin->src_pads; - else - pad_list = &this->bin->sink_pads; - - if (name) { - l = *pad_list; - while (l) { - if (strcmp (GST_JACK_PAD(l)->name, name) == 0) { - g_warning("requested port name %s already in use.", name); - return NULL; - } - l = l->next; - } - newname = g_strdup (name); - } else { - if (this->direction == GST_PAD_SINK) - newname = g_strdup ("alsa_pcm:playback_1"); - else - newname = g_strdup ("alsa_pcm:capture_1"); + GstJack *this; + gchar *newname; + GList *l, **pad_list; + GstJackPad *pad; + gint count; + + g_return_val_if_fail ((this = GST_JACK (element)), NULL); + + if (!this->bin) + pad_list = &this->pads; + else if (this->direction == GST_PAD_SRC) + pad_list = &this->bin->src_pads; + else + pad_list = &this->bin->sink_pads; + + if (name) { + l = *pad_list; + while (l) { + if (strcmp (GST_JACK_PAD (l)->name, name) == 0) { + g_warning ("requested port name %s already in use.", name); + return NULL; + } + l = l->next; } - - pad = g_new0 (GstJackPad, 1); - - if (!port_name_counts) - port_name_counts = g_hash_table_new (g_str_hash, g_str_equal); + newname = g_strdup (name); + } else { + if (this->direction == GST_PAD_SINK) + newname = g_strdup ("alsa_pcm:playback_1"); + else + newname = g_strdup ("alsa_pcm:capture_1"); + } - count = GPOINTER_TO_INT (g_hash_table_lookup (port_name_counts, this->port_name_prefix)); - g_hash_table_insert (port_name_counts, g_strdup (this->port_name_prefix), GINT_TO_POINTER (count+1)); + pad = g_new0 (GstJackPad, 1); - pad->name = g_strdup_printf ("%s%d", this->port_name_prefix, count); + if (!port_name_counts) + port_name_counts = g_hash_table_new (g_str_hash, g_str_equal); - pad->peer_name = newname; - pad->pad = gst_pad_new_from_template (templ, newname); - gst_element_add_pad (GST_ELEMENT (this), pad->pad); - gst_pad_set_link_function (pad->pad, gst_jack_link); - - this->pads = g_list_append (this->pads, pad); - - g_print ("returning from request_new_pad, pad %s created, to connect to %s\n", pad->name, pad->peer_name); - return pad->pad; + count = + GPOINTER_TO_INT (g_hash_table_lookup (port_name_counts, + this->port_name_prefix)); + g_hash_table_insert (port_name_counts, g_strdup (this->port_name_prefix), + GINT_TO_POINTER (count + 1)); + + pad->name = g_strdup_printf ("%s%d", this->port_name_prefix, count); + + pad->peer_name = newname; + pad->pad = gst_pad_new_from_template (templ, newname); + gst_element_add_pad (GST_ELEMENT (this), pad->pad); + gst_pad_set_link_function (pad->pad, gst_jack_link); + + this->pads = g_list_append (this->pads, pad); + + g_print ("returning from request_new_pad, pad %s created, to connect to %s\n", + pad->name, pad->peer_name); + return pad->pad; } static GstElementStateReturn -gst_jack_change_state (GstElement *element) +gst_jack_change_state (GstElement * element) { - GstJack *this; - GList *l = NULL, **pads; - GstJackPad *pad; - GstCaps *caps; - - g_return_val_if_fail (element != NULL, FALSE); - this = GST_JACK (element); - - switch (GST_STATE_PENDING (element)) { + GstJack *this; + GList *l = NULL, **pads; + GstJackPad *pad; + GstCaps *caps; + + g_return_val_if_fail (element != NULL, FALSE); + this = GST_JACK (element); + + switch (GST_STATE_PENDING (element)) { case GST_STATE_NULL: - JACK_DEBUG ("%s: NULL", GST_OBJECT_NAME (GST_OBJECT (this))); + JACK_DEBUG ("%s: NULL", GST_OBJECT_NAME (GST_OBJECT (this))); + + break; - break; - case GST_STATE_READY: - JACK_DEBUG ("%s: READY", GST_OBJECT_NAME (GST_OBJECT (this))); + JACK_DEBUG ("%s: READY", GST_OBJECT_NAME (GST_OBJECT (this))); - if (!this->bin) { - if (!(this->bin = (GstJackBin*)gst_element_get_managing_bin (element)) - || !GST_IS_JACK_BIN (this->bin)) { - this->bin = NULL; - g_warning ("jack element %s needs to be contained in a jack bin.", - GST_OBJECT_NAME (element)); - return GST_STATE_FAILURE; - } + if (!this->bin) { + if (!(this->bin = (GstJackBin *) gst_element_get_managing_bin (element)) + || !GST_IS_JACK_BIN (this->bin)) { + this->bin = NULL; + g_warning ("jack element %s needs to be contained in a jack bin.", + GST_OBJECT_NAME (element)); + return GST_STATE_FAILURE; + } + + /* fixme: verify that all names are unique */ + l = this->pads; + pads = + (this->direction == + GST_PAD_SRC) ? &this->bin->src_pads : &this->bin->sink_pads; + while (l) { + pad = GST_JACK_PAD (l); + JACK_DEBUG ("%s: appending pad %s:%s to list", GST_OBJECT_NAME (this), + pad->name, pad->peer_name); + *pads = g_list_append (*pads, pad); + l = g_list_next (l); + } + } + break; - /* fixme: verify that all names are unique */ - l = this->pads; - pads = (this->direction == GST_PAD_SRC) ? &this->bin->src_pads : &this->bin->sink_pads; - while (l) { - pad = GST_JACK_PAD (l); - JACK_DEBUG ("%s: appending pad %s:%s to list", GST_OBJECT_NAME (this), pad->name, pad->peer_name); - *pads = g_list_append (*pads, pad); - l = g_list_next (l); - } - } - break; - case GST_STATE_PAUSED: - JACK_DEBUG ("%s: PAUSED", GST_OBJECT_NAME (GST_OBJECT (this))); + JACK_DEBUG ("%s: PAUSED", GST_OBJECT_NAME (GST_OBJECT (this))); - if (GST_STATE (element) == GST_STATE_READY) { - /* we're in READY->PAUSED */ - l = this->pads; - while (l) { - pad = GST_JACK_PAD (l); - caps = gst_caps_copy (gst_pad_get_negotiated_caps (pad->pad)); - gst_caps_set_simple (caps, - "rate", G_TYPE_INT, (int)this->bin->rate, - "buffer-frames", G_TYPE_INT, (gint)this->bin->nframes, - NULL); - if (gst_pad_try_set_caps (pad->pad, caps) <= 0) - return GST_STATE_FAILURE; - l = g_list_next (l); - } - } - break; + if (GST_STATE (element) == GST_STATE_READY) { + /* we're in READY->PAUSED */ + l = this->pads; + while (l) { + pad = GST_JACK_PAD (l); + caps = gst_caps_copy (gst_pad_get_negotiated_caps (pad->pad)); + gst_caps_set_simple (caps, + "rate", G_TYPE_INT, (int) this->bin->rate, + "buffer-frames", G_TYPE_INT, (gint) this->bin->nframes, NULL); + if (gst_pad_try_set_caps (pad->pad, caps) <= 0) + return GST_STATE_FAILURE; + l = g_list_next (l); + } + } + break; case GST_STATE_PLAYING: - JACK_DEBUG ("%s: PLAYING", GST_OBJECT_NAME (GST_OBJECT (this))); - break; - } - - JACK_DEBUG ("%s: state change finished", GST_OBJECT_NAME (this)); - - if (GST_ELEMENT_CLASS (parent_class)->change_state) - return GST_ELEMENT_CLASS (parent_class)->change_state (element); + JACK_DEBUG ("%s: PLAYING", GST_OBJECT_NAME (GST_OBJECT (this))); + break; + } - return GST_STATE_SUCCESS; + JACK_DEBUG ("%s: state change finished", GST_OBJECT_NAME (this)); + + if (GST_ELEMENT_CLASS (parent_class)->change_state) + return GST_ELEMENT_CLASS (parent_class)->change_state (element); + + return GST_STATE_SUCCESS; } static GstPadLinkReturn -gst_jack_link (GstPad *pad, const GstCaps *caps) +gst_jack_link (GstPad * pad, const GstCaps * caps) { GstJack *this; gint rate, buffer_frames; GstStructure *structure; - + this = GST_JACK (GST_OBJECT_PARENT (pad)); - + structure = gst_caps_get_structure (caps, 0); - gst_structure_get_int (structure, "rate", &rate); - gst_structure_get_int (structure, "buffer-frames", &buffer_frames); + gst_structure_get_int (structure, "rate", &rate); + gst_structure_get_int (structure, "buffer-frames", &buffer_frames); if (this->bin && (rate != this->bin->rate || - buffer_frames != this->bin->nframes)) - return GST_PAD_LINK_REFUSED; - + buffer_frames != this->bin->nframes)) + return GST_PAD_LINK_REFUSED; + return GST_PAD_LINK_OK; } static void -gst_jack_loop (GstElement *element) +gst_jack_loop (GstElement * element) { - GstJack *this; - GList *pads; - gint len; - GstJackPad *pad; - GstBuffer *buffer; - - this = GST_JACK (element); - - len = this->bin->nframes * sizeof (sample_t); - - pads = this->pads; - while (pads) { - pad = GST_JACK_PAD (pads); - - if (this->direction == GST_PAD_SINK) { - buffer = GST_BUFFER (gst_pad_pull (pad->pad)); - - if (GST_IS_EVENT (buffer)) { - GstEvent *event = GST_EVENT (buffer); - switch (GST_EVENT_TYPE (buffer)) { - case GST_EVENT_EOS: - gst_element_set_eos (element); - gst_event_unref (event); - return; - default: - gst_pad_event_default (pad->pad, event); - return; - } - } - - /* if the other plugins only give out buffer-frames or less (as - they should), if the length of the GstBuffer is different - from nframes then the buffer is short and we will get EOS - next */ - memcpy (pad->data, GST_BUFFER_DATA (buffer), - GST_BUFFER_SIZE (buffer)); - if (len != GST_BUFFER_SIZE (buffer)) - memset (pad->data + GST_BUFFER_SIZE (buffer), 0, - len - GST_BUFFER_SIZE (buffer)); + GstJack *this; + GList *pads; + gint len; + GstJackPad *pad; + GstBuffer *buffer; - gst_buffer_unref (buffer); - } else { - buffer = gst_buffer_new (); - gst_buffer_set_data (buffer, pad->data, len); - GST_BUFFER_FLAG_SET(buffer, GST_BUFFER_DONTFREE); - - gst_pad_push (pad->pad, GST_DATA (buffer)); - } - pads = g_list_next (pads); + this = GST_JACK (element); + + len = this->bin->nframes * sizeof (sample_t); + + pads = this->pads; + while (pads) { + pad = GST_JACK_PAD (pads); + + if (this->direction == GST_PAD_SINK) { + buffer = GST_BUFFER (gst_pad_pull (pad->pad)); + + if (GST_IS_EVENT (buffer)) { + GstEvent *event = GST_EVENT (buffer); + + switch (GST_EVENT_TYPE (buffer)) { + case GST_EVENT_EOS: + gst_element_set_eos (element); + gst_event_unref (event); + return; + default: + gst_pad_event_default (pad->pad, event); + return; + } + } + + /* if the other plugins only give out buffer-frames or less (as + they should), if the length of the GstBuffer is different + from nframes then the buffer is short and we will get EOS + next */ + memcpy (pad->data, GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer)); + if (len != GST_BUFFER_SIZE (buffer)) + memset (pad->data + GST_BUFFER_SIZE (buffer), 0, + len - GST_BUFFER_SIZE (buffer)); + + gst_buffer_unref (buffer); + } else { + buffer = gst_buffer_new (); + gst_buffer_set_data (buffer, pad->data, len); + GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_DONTFREE); + + gst_pad_push (pad->pad, GST_DATA (buffer)); } + pads = g_list_next (pads); + } } static gboolean -plugin_init (GstPlugin *plugin) +plugin_init (GstPlugin * plugin) { - if (!gst_element_register (plugin, "jackbin", GST_RANK_NONE, GST_TYPE_JACK_BIN)) + if (!gst_element_register (plugin, "jackbin", GST_RANK_NONE, + GST_TYPE_JACK_BIN)) return FALSE; - if (!gst_element_register (plugin, "jacksrc", GST_RANK_NONE, GST_TYPE_JACK_SRC)) + if (!gst_element_register (plugin, "jacksrc", GST_RANK_NONE, + GST_TYPE_JACK_SRC)) return FALSE; - if (!gst_element_register (plugin, "jacksink", GST_RANK_NONE, GST_TYPE_JACK_SINK)) + if (!gst_element_register (plugin, "jacksink", GST_RANK_NONE, + GST_TYPE_JACK_SINK)) return FALSE; - + return TRUE; } -GST_PLUGIN_DEFINE ( - GST_VERSION_MAJOR, - GST_VERSION_MINOR, - "jack", - "Jack Plugin Library", - plugin_init, - VERSION, - "GPL", - GST_PACKAGE, - GST_ORIGIN) +GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, + GST_VERSION_MINOR, + "jack", + "Jack Plugin Library", plugin_init, VERSION, "GPL", GST_PACKAGE, GST_ORIGIN) diff --git a/ext/jack/gstjack.h b/ext/jack/gstjack.h index 87d7d61151..f5a5416fba 100644 --- a/ext/jack/gstjack.h +++ b/ext/jack/gstjack.h @@ -51,7 +51,7 @@ #define GST_IS_JACK_BIN_CLASS(klass) G_TYPE_CHECK_CLASS_TYPE(klass, GST_TYPE_JACK_BIN) #define GST_TYPE_JACK_BIN gst_jack_bin_get_type() -#define GST_JACK_PAD(l) ((GstJackPad*)l->data) /* l is a GList */ +#define GST_JACK_PAD(l) ((GstJackPad*)l->data) /* l is a GList */ typedef struct _GstJack GstJack; @@ -64,59 +64,65 @@ typedef GstJack GstJackSrc; typedef GstJackClass GstJackSrcClass; -enum { - GST_JACK_OPEN = GST_BIN_FLAG_LAST, - GST_JACK_ACTIVE, - GST_JACK_FLAG_LAST = GST_BIN_FLAG_LAST + 3, +enum +{ + GST_JACK_OPEN = GST_BIN_FLAG_LAST, + GST_JACK_ACTIVE, + GST_JACK_FLAG_LAST = GST_BIN_FLAG_LAST + 3, }; typedef jack_default_audio_sample_t sample_t; -typedef struct { - GstPad *pad; - void *data; - const gchar *name; - const gchar *peer_name; - jack_port_t *port; +typedef struct +{ + GstPad *pad; + void *data; + const gchar *name; + const gchar *peer_name; + jack_port_t *port; } GstJackPad; -struct _GstJack { - GstElement element; +struct _GstJack +{ + GstElement element; - /* list of GstJackPads */ - GList *pads; + /* list of GstJackPads */ + GList *pads; - /* for convenience */ - GstPadDirection direction; + /* for convenience */ + GstPadDirection direction; - gchar *port_name_prefix; + gchar *port_name_prefix; - GstJackBin *bin; + GstJackBin *bin; }; -struct _GstJackClass { - GstElementClass parent_class; +struct _GstJackClass +{ + GstElementClass parent_class; }; -struct _GstJackBin { - GstBin bin; +struct _GstJackBin +{ + GstBin bin; - jack_client_t *client; - gint default_new_port_number; + jack_client_t *client; + gint default_new_port_number; - /* lists of GstJackPads */ - GList *sink_pads; - GList *src_pads; + /* lists of GstJackPads */ + GList *sink_pads; + GList *src_pads; - gchar *client_name; + gchar *client_name; - guint rate; - jack_nframes_t nframes; + guint rate; + jack_nframes_t nframes; }; -struct _GstJackBinClass { - GstBinClass parent_class; +struct _GstJackBinClass +{ + GstBinClass parent_class; }; diff --git a/ext/jack/gstjackbin.c b/ext/jack/gstjackbin.c index b0cea9bc32..9be6e91563 100644 --- a/ext/jack/gstjackbin.c +++ b/ext/jack/gstjackbin.c @@ -29,10 +29,10 @@ static GstBinClass *parent_class = NULL; -static void gst_jack_bin_init(GstJackBin *this); -static void gst_jack_bin_class_init(GstJackBinClass *klass); +static void gst_jack_bin_init (GstJackBin * this); +static void gst_jack_bin_class_init (GstJackBinClass * klass); -static GstElementStateReturn gst_jack_bin_change_state(GstElement *element); +static GstElementStateReturn gst_jack_bin_change_state (GstElement * element); /* jack callbacks */ static int process (jack_nframes_t nframes, void *arg); @@ -48,196 +48,213 @@ static gboolean watchdog_please_set_the_jackbin_to_ready = FALSE; * so set the state of the bin to READY. */ GType -gst_jack_bin_get_type (void) +gst_jack_bin_get_type (void) { - static GType jack_bin_type = 0; -if (!jack_bin_type) { - static const GTypeInfo jack_bin_info = { - sizeof(GstJackBinClass), - NULL, - NULL, - (GClassInitFunc)gst_jack_bin_class_init, - NULL, - NULL, - sizeof(GstJackBin), - 0, - (GInstanceInitFunc)gst_jack_bin_init, - }; - jack_bin_type = g_type_register_static (GST_TYPE_BIN, "GstJackBin", &jack_bin_info, 0); - } - return jack_bin_type; + static GType jack_bin_type = 0; + + if (!jack_bin_type) { + static const GTypeInfo jack_bin_info = { + sizeof (GstJackBinClass), + NULL, + NULL, + (GClassInitFunc) gst_jack_bin_class_init, + NULL, + NULL, + sizeof (GstJackBin), + 0, + (GInstanceInitFunc) gst_jack_bin_init, + }; + jack_bin_type = + g_type_register_static (GST_TYPE_BIN, "GstJackBin", &jack_bin_info, 0); + } + return jack_bin_type; } static void -gst_jack_bin_class_init(GstJackBinClass *klass) +gst_jack_bin_class_init (GstJackBinClass * klass) { - GObjectClass *object_class; - GstElementClass *element_class; - - object_class = (GObjectClass *)klass; - element_class = (GstElementClass *)klass; - - parent_class = g_type_class_ref(GST_TYPE_BIN); + GObjectClass *object_class; + GstElementClass *element_class; - element_class->change_state = gst_jack_bin_change_state; + object_class = (GObjectClass *) klass; + element_class = (GstElementClass *) klass; + + parent_class = g_type_class_ref (GST_TYPE_BIN); + + element_class->change_state = gst_jack_bin_change_state; } static void -gst_jack_bin_init(GstJackBin *this) +gst_jack_bin_init (GstJackBin * this) { - GST_DEBUG ("initializing jack bin"); - - /* jack bins are managing bins and iterate themselves */ - GST_FLAG_SET (this, GST_BIN_FLAG_MANAGER); - GST_FLAG_SET (this, GST_BIN_SELF_SCHEDULABLE); - - /* make a new scheduler and associate it with the bin */ - gst_scheduler_factory_make (NULL, GST_ELEMENT (this)); + GST_DEBUG ("initializing jack bin"); + + /* jack bins are managing bins and iterate themselves */ + GST_FLAG_SET (this, GST_BIN_FLAG_MANAGER); + GST_FLAG_SET (this, GST_BIN_SELF_SCHEDULABLE); + + /* make a new scheduler and associate it with the bin */ + gst_scheduler_factory_make (NULL, GST_ELEMENT (this)); } static GstElementStateReturn -gst_jack_bin_change_state (GstElement *element) +gst_jack_bin_change_state (GstElement * element) { - GstJackBin *this; - GList *l = NULL; - GstJackPad *pad; - - g_return_val_if_fail (element != NULL, FALSE); - this = GST_JACK_BIN (element); - - switch (GST_STATE_PENDING (element)) { + GstJackBin *this; + GList *l = NULL; + GstJackPad *pad; + + g_return_val_if_fail (element != NULL, FALSE); + this = GST_JACK_BIN (element); + + switch (GST_STATE_PENDING (element)) { case GST_STATE_NULL: - JACK_DEBUG ("jackbin: NULL state"); - if (this->client) { - JACK_DEBUG ("jackbin: closing client"); - jack_client_close (this->client); - this->client = NULL; - } - - if (_jackbin) - signal (SIGHUP, SIG_DFL); - _jackbin = NULL; + JACK_DEBUG ("jackbin: NULL state"); + if (this->client) { + JACK_DEBUG ("jackbin: closing client"); + jack_client_close (this->client); + this->client = NULL; + } + + if (_jackbin) + signal (SIGHUP, SIG_DFL); + _jackbin = NULL; + + if (GST_ELEMENT_CLASS (parent_class)->change_state) + return GST_ELEMENT_CLASS (parent_class)->change_state (element); + break; - if (GST_ELEMENT_CLASS (parent_class)->change_state) - return GST_ELEMENT_CLASS (parent_class)->change_state (element); - break; - case GST_STATE_READY: - JACK_DEBUG ("jackbin: READY"); + JACK_DEBUG ("jackbin: READY"); - _jackbin = this; - signal (SIGHUP, sighup_handler); + _jackbin = this; + signal (SIGHUP, sighup_handler); - if (!this->client) { - if (!(this->client = jack_client_new ("gst-jack"))) { - g_warning ("jack server not running?"); - return GST_STATE_FAILURE; - } - - gst_scheduler_setup (GST_ELEMENT_SCHED (this)); + if (!this->client) { + if (!(this->client = jack_client_new ("gst-jack"))) { + g_warning ("jack server not running?"); + return GST_STATE_FAILURE; + } - jack_set_process_callback (this->client, process, this); - jack_set_sample_rate_callback (this->client, sample_rate, this); - jack_set_buffer_size_callback (this->client, buffer_size, this); - this->nframes = jack_get_buffer_size (this->client); - jack_on_shutdown (this->client, shutdown, this); - } - - if (GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_OPEN)) { - l = this->src_pads; - while (l) { - JACK_DEBUG ("jackbin: unregistering pad %s:%s", GST_JACK_PAD (l)->name, GST_JACK_PAD (l)->peer_name); - jack_port_unregister (this->client, GST_JACK_PAD (l)->port); - l = g_list_next (l); - } - l = this->sink_pads; - while (l) { - JACK_DEBUG ("jackbin: unregistering pad %s:%s", GST_JACK_PAD (l)->name, GST_JACK_PAD (l)->peer_name); - jack_port_unregister (this->client, GST_JACK_PAD (l)->port); - l = g_list_next (l); - } - GST_FLAG_UNSET (GST_OBJECT (this), GST_JACK_OPEN); + gst_scheduler_setup (GST_ELEMENT_SCHED (this)); + + jack_set_process_callback (this->client, process, this); + jack_set_sample_rate_callback (this->client, sample_rate, this); + jack_set_buffer_size_callback (this->client, buffer_size, this); + this->nframes = jack_get_buffer_size (this->client); + jack_on_shutdown (this->client, shutdown, this); + } + + if (GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_OPEN)) { + l = this->src_pads; + while (l) { + JACK_DEBUG ("jackbin: unregistering pad %s:%s", + GST_JACK_PAD (l)->name, GST_JACK_PAD (l)->peer_name); + jack_port_unregister (this->client, GST_JACK_PAD (l)->port); + l = g_list_next (l); + } + l = this->sink_pads; + while (l) { + JACK_DEBUG ("jackbin: unregistering pad %s:%s", + GST_JACK_PAD (l)->name, GST_JACK_PAD (l)->peer_name); + jack_port_unregister (this->client, GST_JACK_PAD (l)->port); + l = g_list_next (l); + } + GST_FLAG_UNSET (GST_OBJECT (this), GST_JACK_OPEN); + + if (GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_ACTIVE)) { + JACK_DEBUG ("jackbin: deactivating client"); + jack_deactivate (this->client); + GST_FLAG_UNSET (GST_OBJECT (this), GST_JACK_ACTIVE); + } + } + + if (GST_ELEMENT_CLASS (parent_class)->change_state) + return GST_ELEMENT_CLASS (parent_class)->change_state (element); + break; - if (GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_ACTIVE)) { - JACK_DEBUG ("jackbin: deactivating client"); - jack_deactivate (this->client); - GST_FLAG_UNSET (GST_OBJECT (this), GST_JACK_ACTIVE); - } - } - - if (GST_ELEMENT_CLASS (parent_class)->change_state) - return GST_ELEMENT_CLASS (parent_class)->change_state (element); - break; - case GST_STATE_PAUSED: - JACK_DEBUG ("jackbin: PAUSED"); - - if (!GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_OPEN)) { - l = this->src_pads; - while (l) { - pad = GST_JACK_PAD (l); - JACK_DEBUG ("jackbin: registering input port %s (peer %s)", pad->name, pad->peer_name); - pad->port = jack_port_register (this->client, pad->name, JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput|JackPortIsTerminal, 0); - l = g_list_next (l); - } - l = this->sink_pads; - while (l) { - pad = GST_JACK_PAD (l); - JACK_DEBUG ("jackbin: registering output port %s (peer %s)", pad->name, pad->peer_name); - pad->port = jack_port_register (this->client, pad->name, JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput|JackPortIsTerminal, 0); - l = g_list_next (l); - } + JACK_DEBUG ("jackbin: PAUSED"); - /* must activate before connecting */ - if (!GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_ACTIVE)) { - JACK_DEBUG ("jackbin: activating client"); - jack_activate (this->client); - GST_FLAG_SET (GST_OBJECT (this), GST_JACK_ACTIVE); - } + if (!GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_OPEN)) { + l = this->src_pads; + while (l) { + pad = GST_JACK_PAD (l); + JACK_DEBUG ("jackbin: registering input port %s (peer %s)", pad->name, + pad->peer_name); + pad->port = + jack_port_register (this->client, pad->name, + JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput | JackPortIsTerminal, 0); + l = g_list_next (l); + } + l = this->sink_pads; + while (l) { + pad = GST_JACK_PAD (l); + JACK_DEBUG ("jackbin: registering output port %s (peer %s)", + pad->name, pad->peer_name); + pad->port = + jack_port_register (this->client, pad->name, + JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput | JackPortIsTerminal, + 0); + l = g_list_next (l); + } - l = this->src_pads; - while (l) { - pad = GST_JACK_PAD (l); - JACK_DEBUG ("connecting jack port %s to gst jack port %s", pad->peer_name, jack_port_name (pad->port)); - if (jack_connect (this->client, pad->peer_name, jack_port_name (pad->port))) { - g_warning ("jackbin: could not connect %s and %s", pad->peer_name, jack_port_name (pad->port)); - return GST_STATE_FAILURE; - } - l = g_list_next (l); - } - l = this->sink_pads; - while (l) { - pad = GST_JACK_PAD (l); - JACK_DEBUG ("connecting gst jack port %s to jack port %s", jack_port_name (pad->port), pad->peer_name); - if (jack_connect (this->client, jack_port_name (pad->port), pad->peer_name)) { - g_warning ("jackbin: could not connect %s and %s", pad->peer_name, jack_port_name (pad->port)); - return GST_STATE_FAILURE; - } - l = g_list_next (l); - } + /* must activate before connecting */ + if (!GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_ACTIVE)) { + JACK_DEBUG ("jackbin: activating client"); + jack_activate (this->client); + GST_FLAG_SET (GST_OBJECT (this), GST_JACK_ACTIVE); + } - JACK_DEBUG ("jackbin: setting OPEN flag"); - GST_FLAG_SET (GST_OBJECT (this), GST_JACK_OPEN); + l = this->src_pads; + while (l) { + pad = GST_JACK_PAD (l); + JACK_DEBUG ("connecting jack port %s to gst jack port %s", + pad->peer_name, jack_port_name (pad->port)); + if (jack_connect (this->client, pad->peer_name, + jack_port_name (pad->port))) { + g_warning ("jackbin: could not connect %s and %s", pad->peer_name, + jack_port_name (pad->port)); + return GST_STATE_FAILURE; + } + l = g_list_next (l); + } + l = this->sink_pads; + while (l) { + pad = GST_JACK_PAD (l); + JACK_DEBUG ("connecting gst jack port %s to jack port %s", + jack_port_name (pad->port), pad->peer_name); + if (jack_connect (this->client, jack_port_name (pad->port), + pad->peer_name)) { + g_warning ("jackbin: could not connect %s and %s", pad->peer_name, + jack_port_name (pad->port)); + return GST_STATE_FAILURE; + } + l = g_list_next (l); + } - if (GST_ELEMENT_CLASS (parent_class)->change_state) - return GST_ELEMENT_CLASS (parent_class)->change_state (element); - } else { - if (GST_ELEMENT_CLASS (parent_class)->change_state) - return GST_ELEMENT_CLASS (parent_class)->change_state (element); - } + JACK_DEBUG ("jackbin: setting OPEN flag"); + GST_FLAG_SET (GST_OBJECT (this), GST_JACK_OPEN); - break; + if (GST_ELEMENT_CLASS (parent_class)->change_state) + return GST_ELEMENT_CLASS (parent_class)->change_state (element); + } else { + if (GST_ELEMENT_CLASS (parent_class)->change_state) + return GST_ELEMENT_CLASS (parent_class)->change_state (element); + } + + break; case GST_STATE_PLAYING: - JACK_DEBUG ("jackbin: PLAYING"); + JACK_DEBUG ("jackbin: PLAYING"); - if (GST_ELEMENT_CLASS (parent_class)->change_state) - return GST_ELEMENT_CLASS (parent_class)->change_state (element); - break; - } - - JACK_DEBUG ("jackbin: state change finished"); - - return GST_STATE_SUCCESS; + if (GST_ELEMENT_CLASS (parent_class)->change_state) + return GST_ELEMENT_CLASS (parent_class)->change_state (element); + break; + } + + JACK_DEBUG ("jackbin: state change finished"); + + return GST_STATE_SUCCESS; } /* jack callbacks */ @@ -247,76 +264,79 @@ gst_jack_bin_change_state (GstElement *element) static int process (jack_nframes_t nframes, void *arg) { - GstJackBin *bin = (GstJackBin*) arg; - GstJackPad *pad; - GList *l; - - g_assert (bin); - - JACK_DEBUG ("jackbin: process()"); + GstJackBin *bin = (GstJackBin *) arg; + GstJackPad *pad; + GList *l; - if (GST_STATE (bin) != GST_STATE_PLAYING) { - JACK_DEBUG ("jackbin: bin is not PLAYING yet, returning"); - return 0; - } else { - JACK_DEBUG ("jackbin: we are PLAYING, let's process()"); - } + g_assert (bin); - l = bin->src_pads; - while (l) { - pad = GST_JACK_PAD (l); - pad->data = jack_port_get_buffer (pad->port, nframes); - l = g_list_next (l); - } - - l = bin->sink_pads; - while (l) { - pad = GST_JACK_PAD (l); - pad->data = jack_port_get_buffer (pad->port, nframes); - l = g_list_next (l); - } - - bin->nframes = nframes; - - JACK_DEBUG ("jackbin: iterating to process %ld frames of audio...", nframes); - if (!gst_bin_iterate (GST_BIN (bin))) { - g_warning ("bin failed to iterate"); - return -1; - } - - /* that's all folks */ - - return 0; + JACK_DEBUG ("jackbin: process()"); + + if (GST_STATE (bin) != GST_STATE_PLAYING) { + JACK_DEBUG ("jackbin: bin is not PLAYING yet, returning"); + return 0; + } else { + JACK_DEBUG ("jackbin: we are PLAYING, let's process()"); + } + + l = bin->src_pads; + while (l) { + pad = GST_JACK_PAD (l); + pad->data = jack_port_get_buffer (pad->port, nframes); + l = g_list_next (l); + } + + l = bin->sink_pads; + while (l) { + pad = GST_JACK_PAD (l); + pad->data = jack_port_get_buffer (pad->port, nframes); + l = g_list_next (l); + } + + bin->nframes = nframes; + + JACK_DEBUG ("jackbin: iterating to process %ld frames of audio...", nframes); + if (!gst_bin_iterate (GST_BIN (bin))) { + g_warning ("bin failed to iterate"); + return -1; + } + + /* that's all folks */ + + return 0; } static int sample_rate (jack_nframes_t nframes, void *arg) { - GstJackBin *bin = (GstJackBin*) arg; - JACK_DEBUG ("the sample rate is now %lu/sec\n", nframes); - bin->rate = nframes; - return 0; + GstJackBin *bin = (GstJackBin *) arg; + + JACK_DEBUG ("the sample rate is now %lu/sec\n", nframes); + bin->rate = nframes; + return 0; } static int buffer_size (jack_nframes_t nframes, void *arg) { - GstJackBin *bin = (GstJackBin*) arg; - JACK_DEBUG ("the buffer size is now %lu\n", nframes); - bin->nframes = nframes; - return 0; + GstJackBin *bin = (GstJackBin *) arg; + + JACK_DEBUG ("the buffer size is now %lu\n", nframes); + bin->nframes = nframes; + return 0; } static void shutdown (void *arg) { /* GstJackClient *client = (GstJackClient*) arg; */ - printf ("shutdown %p\n", arg); + printf ("shutdown %p\n", arg); /* gst_element_set_state (GST_ELEMENT (client->manager), GST_STATE_READY); */ } -static void sighup_handler (int sig) +static void +sighup_handler (int sig) { - g_message ("got sighup, setting state to READY"); - watchdog_please_set_the_jackbin_to_ready = TRUE; + g_message ("got sighup, setting state to READY"); + watchdog_please_set_the_jackbin_to_ready = TRUE; } From 44f714a5fd82aa290f63d1adaca77afd43a087af Mon Sep 17 00:00:00 2001 From: Johan Dahlin Date: Mon, 15 Mar 2004 16:32:54 +0000 Subject: [PATCH 158/351] *.h: Revert indenting Original commit message from CVS: * *.h: Revert indenting --- ext/jack/gstjack.h | 72 +++++++++++++++++++++------------------------- 1 file changed, 33 insertions(+), 39 deletions(-) diff --git a/ext/jack/gstjack.h b/ext/jack/gstjack.h index f5a5416fba..87d7d61151 100644 --- a/ext/jack/gstjack.h +++ b/ext/jack/gstjack.h @@ -51,7 +51,7 @@ #define GST_IS_JACK_BIN_CLASS(klass) G_TYPE_CHECK_CLASS_TYPE(klass, GST_TYPE_JACK_BIN) #define GST_TYPE_JACK_BIN gst_jack_bin_get_type() -#define GST_JACK_PAD(l) ((GstJackPad*)l->data) /* l is a GList */ +#define GST_JACK_PAD(l) ((GstJackPad*)l->data) /* l is a GList */ typedef struct _GstJack GstJack; @@ -64,65 +64,59 @@ typedef GstJack GstJackSrc; typedef GstJackClass GstJackSrcClass; -enum -{ - GST_JACK_OPEN = GST_BIN_FLAG_LAST, - GST_JACK_ACTIVE, - GST_JACK_FLAG_LAST = GST_BIN_FLAG_LAST + 3, +enum { + GST_JACK_OPEN = GST_BIN_FLAG_LAST, + GST_JACK_ACTIVE, + GST_JACK_FLAG_LAST = GST_BIN_FLAG_LAST + 3, }; typedef jack_default_audio_sample_t sample_t; -typedef struct -{ - GstPad *pad; - void *data; - const gchar *name; - const gchar *peer_name; - jack_port_t *port; +typedef struct { + GstPad *pad; + void *data; + const gchar *name; + const gchar *peer_name; + jack_port_t *port; } GstJackPad; -struct _GstJack -{ - GstElement element; +struct _GstJack { + GstElement element; - /* list of GstJackPads */ - GList *pads; + /* list of GstJackPads */ + GList *pads; - /* for convenience */ - GstPadDirection direction; + /* for convenience */ + GstPadDirection direction; - gchar *port_name_prefix; + gchar *port_name_prefix; - GstJackBin *bin; + GstJackBin *bin; }; -struct _GstJackClass -{ - GstElementClass parent_class; +struct _GstJackClass { + GstElementClass parent_class; }; -struct _GstJackBin -{ - GstBin bin; +struct _GstJackBin { + GstBin bin; - jack_client_t *client; - gint default_new_port_number; + jack_client_t *client; + gint default_new_port_number; - /* lists of GstJackPads */ - GList *sink_pads; - GList *src_pads; + /* lists of GstJackPads */ + GList *sink_pads; + GList *src_pads; - gchar *client_name; + gchar *client_name; - guint rate; - jack_nframes_t nframes; + guint rate; + jack_nframes_t nframes; }; -struct _GstJackBinClass -{ - GstBinClass parent_class; +struct _GstJackBinClass { + GstBinClass parent_class; }; From f88ab240e5c127353df2b5a5dc8081f07fea105a Mon Sep 17 00:00:00 2001 From: Thomas Vander Stichele Date: Mon, 15 Mar 2004 19:32:27 +0000 Subject: [PATCH 159/351] don't mix tabs and spaces Original commit message from CVS: don't mix tabs and spaces --- ext/jack/gstjack.c | 111 ++++++++++++------------ ext/jack/gstjackbin.c | 195 +++++++++++++++++++++--------------------- 2 files changed, 155 insertions(+), 151 deletions(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 41f078bc44..643896a8a4 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -106,8 +106,9 @@ gst_jack_get_type (void) 0, NULL, }; + jack_type = - g_type_register_static (GST_TYPE_ELEMENT, "GstJack", &jack_info, 0); + g_type_register_static (GST_TYPE_ELEMENT, "GstJack", &jack_info, 0); } return jack_type; } @@ -129,8 +130,9 @@ gst_jack_sink_get_type (void) 0, (GInstanceInitFunc) gst_jack_init, }; + jack_type = - g_type_register_static (GST_TYPE_JACK, "GstJackSink", &jack_info, 0); + g_type_register_static (GST_TYPE_JACK, "GstJackSink", &jack_info, 0); } return jack_type; } @@ -152,8 +154,9 @@ gst_jack_src_get_type (void) 0, (GInstanceInitFunc) gst_jack_init, }; + jack_type = - g_type_register_static (GST_TYPE_JACK, "GstJackSrc", &jack_info, 0); + g_type_register_static (GST_TYPE_JACK, "GstJackSrc", &jack_info, 0); } return jack_type; } @@ -240,7 +243,7 @@ gst_jack_set_property (GObject * object, guint prop_id, const GValue * value, switch (prop_id) { case ARG_PORT_NAME_PREFIX: if (this->port_name_prefix) - g_free (this->port_name_prefix); + g_free (this->port_name_prefix); this->port_name_prefix = g_strdup (g_value_get_string (value)); break; default: @@ -318,8 +321,8 @@ gst_jack_request_new_pad (GstElement * element, GstPadTemplate * templ, l = *pad_list; while (l) { if (strcmp (GST_JACK_PAD (l)->name, name) == 0) { - g_warning ("requested port name %s already in use.", name); - return NULL; + g_warning ("requested port name %s already in use.", name); + return NULL; } l = l->next; } @@ -338,7 +341,7 @@ gst_jack_request_new_pad (GstElement * element, GstPadTemplate * templ, count = GPOINTER_TO_INT (g_hash_table_lookup (port_name_counts, - this->port_name_prefix)); + this->port_name_prefix)); g_hash_table_insert (port_name_counts, g_strdup (this->port_name_prefix), GINT_TO_POINTER (count + 1)); @@ -377,26 +380,26 @@ gst_jack_change_state (GstElement * element) JACK_DEBUG ("%s: READY", GST_OBJECT_NAME (GST_OBJECT (this))); if (!this->bin) { - if (!(this->bin = (GstJackBin *) gst_element_get_managing_bin (element)) - || !GST_IS_JACK_BIN (this->bin)) { - this->bin = NULL; - g_warning ("jack element %s needs to be contained in a jack bin.", - GST_OBJECT_NAME (element)); - return GST_STATE_FAILURE; - } + if (!(this->bin = (GstJackBin *) gst_element_get_managing_bin (element)) + || !GST_IS_JACK_BIN (this->bin)) { + this->bin = NULL; + g_warning ("jack element %s needs to be contained in a jack bin.", + GST_OBJECT_NAME (element)); + return GST_STATE_FAILURE; + } - /* fixme: verify that all names are unique */ - l = this->pads; - pads = - (this->direction == - GST_PAD_SRC) ? &this->bin->src_pads : &this->bin->sink_pads; - while (l) { - pad = GST_JACK_PAD (l); - JACK_DEBUG ("%s: appending pad %s:%s to list", GST_OBJECT_NAME (this), - pad->name, pad->peer_name); - *pads = g_list_append (*pads, pad); - l = g_list_next (l); - } + /* fixme: verify that all names are unique */ + l = this->pads; + pads = + (this->direction == + GST_PAD_SRC) ? &this->bin->src_pads : &this->bin->sink_pads; + while (l) { + pad = GST_JACK_PAD (l); + JACK_DEBUG ("%s: appending pad %s:%s to list", GST_OBJECT_NAME (this), + pad->name, pad->peer_name); + *pads = g_list_append (*pads, pad); + l = g_list_next (l); + } } break; @@ -404,18 +407,18 @@ gst_jack_change_state (GstElement * element) JACK_DEBUG ("%s: PAUSED", GST_OBJECT_NAME (GST_OBJECT (this))); if (GST_STATE (element) == GST_STATE_READY) { - /* we're in READY->PAUSED */ - l = this->pads; - while (l) { - pad = GST_JACK_PAD (l); - caps = gst_caps_copy (gst_pad_get_negotiated_caps (pad->pad)); - gst_caps_set_simple (caps, - "rate", G_TYPE_INT, (int) this->bin->rate, - "buffer-frames", G_TYPE_INT, (gint) this->bin->nframes, NULL); - if (gst_pad_try_set_caps (pad->pad, caps) <= 0) - return GST_STATE_FAILURE; - l = g_list_next (l); - } + /* we're in READY->PAUSED */ + l = this->pads; + while (l) { + pad = GST_JACK_PAD (l); + caps = gst_caps_copy (gst_pad_get_negotiated_caps (pad->pad)); + gst_caps_set_simple (caps, + "rate", G_TYPE_INT, (int) this->bin->rate, + "buffer-frames", G_TYPE_INT, (gint) this->bin->nframes, NULL); + if (gst_pad_try_set_caps (pad->pad, caps) <= 0) + return GST_STATE_FAILURE; + l = g_list_next (l); + } } break; case GST_STATE_PLAYING: @@ -444,7 +447,7 @@ gst_jack_link (GstPad * pad, const GstCaps * caps) gst_structure_get_int (structure, "rate", &rate); gst_structure_get_int (structure, "buffer-frames", &buffer_frames); if (this->bin && (rate != this->bin->rate || - buffer_frames != this->bin->nframes)) + buffer_frames != this->bin->nframes)) return GST_PAD_LINK_REFUSED; return GST_PAD_LINK_OK; @@ -471,17 +474,17 @@ gst_jack_loop (GstElement * element) buffer = GST_BUFFER (gst_pad_pull (pad->pad)); if (GST_IS_EVENT (buffer)) { - GstEvent *event = GST_EVENT (buffer); + GstEvent *event = GST_EVENT (buffer); - switch (GST_EVENT_TYPE (buffer)) { - case GST_EVENT_EOS: - gst_element_set_eos (element); - gst_event_unref (event); - return; - default: - gst_pad_event_default (pad->pad, event); - return; - } + switch (GST_EVENT_TYPE (buffer)) { + case GST_EVENT_EOS: + gst_element_set_eos (element); + gst_event_unref (event); + return; + default: + gst_pad_event_default (pad->pad, event); + return; + } } /* if the other plugins only give out buffer-frames or less (as @@ -490,8 +493,8 @@ gst_jack_loop (GstElement * element) next */ memcpy (pad->data, GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer)); if (len != GST_BUFFER_SIZE (buffer)) - memset (pad->data + GST_BUFFER_SIZE (buffer), 0, - len - GST_BUFFER_SIZE (buffer)); + memset (pad->data + GST_BUFFER_SIZE (buffer), 0, + len - GST_BUFFER_SIZE (buffer)); gst_buffer_unref (buffer); } else { @@ -509,15 +512,15 @@ static gboolean plugin_init (GstPlugin * plugin) { if (!gst_element_register (plugin, "jackbin", GST_RANK_NONE, - GST_TYPE_JACK_BIN)) + GST_TYPE_JACK_BIN)) return FALSE; if (!gst_element_register (plugin, "jacksrc", GST_RANK_NONE, - GST_TYPE_JACK_SRC)) + GST_TYPE_JACK_SRC)) return FALSE; if (!gst_element_register (plugin, "jacksink", GST_RANK_NONE, - GST_TYPE_JACK_SINK)) + GST_TYPE_JACK_SINK)) return FALSE; return TRUE; diff --git a/ext/jack/gstjackbin.c b/ext/jack/gstjackbin.c index 9be6e91563..c2c4957015 100644 --- a/ext/jack/gstjackbin.c +++ b/ext/jack/gstjackbin.c @@ -64,8 +64,9 @@ gst_jack_bin_get_type (void) 0, (GInstanceInitFunc) gst_jack_bin_init, }; + jack_bin_type = - g_type_register_static (GST_TYPE_BIN, "GstJackBin", &jack_bin_info, 0); + g_type_register_static (GST_TYPE_BIN, "GstJackBin", &jack_bin_info, 0); } return jack_bin_type; } @@ -111,17 +112,17 @@ gst_jack_bin_change_state (GstElement * element) case GST_STATE_NULL: JACK_DEBUG ("jackbin: NULL state"); if (this->client) { - JACK_DEBUG ("jackbin: closing client"); - jack_client_close (this->client); - this->client = NULL; + JACK_DEBUG ("jackbin: closing client"); + jack_client_close (this->client); + this->client = NULL; } if (_jackbin) - signal (SIGHUP, SIG_DFL); + signal (SIGHUP, SIG_DFL); _jackbin = NULL; if (GST_ELEMENT_CLASS (parent_class)->change_state) - return GST_ELEMENT_CLASS (parent_class)->change_state (element); + return GST_ELEMENT_CLASS (parent_class)->change_state (element); break; case GST_STATE_READY: @@ -131,116 +132,116 @@ gst_jack_bin_change_state (GstElement * element) signal (SIGHUP, sighup_handler); if (!this->client) { - if (!(this->client = jack_client_new ("gst-jack"))) { - g_warning ("jack server not running?"); - return GST_STATE_FAILURE; - } + if (!(this->client = jack_client_new ("gst-jack"))) { + g_warning ("jack server not running?"); + return GST_STATE_FAILURE; + } - gst_scheduler_setup (GST_ELEMENT_SCHED (this)); + gst_scheduler_setup (GST_ELEMENT_SCHED (this)); - jack_set_process_callback (this->client, process, this); - jack_set_sample_rate_callback (this->client, sample_rate, this); - jack_set_buffer_size_callback (this->client, buffer_size, this); - this->nframes = jack_get_buffer_size (this->client); - jack_on_shutdown (this->client, shutdown, this); + jack_set_process_callback (this->client, process, this); + jack_set_sample_rate_callback (this->client, sample_rate, this); + jack_set_buffer_size_callback (this->client, buffer_size, this); + this->nframes = jack_get_buffer_size (this->client); + jack_on_shutdown (this->client, shutdown, this); } if (GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_OPEN)) { - l = this->src_pads; - while (l) { - JACK_DEBUG ("jackbin: unregistering pad %s:%s", - GST_JACK_PAD (l)->name, GST_JACK_PAD (l)->peer_name); - jack_port_unregister (this->client, GST_JACK_PAD (l)->port); - l = g_list_next (l); - } - l = this->sink_pads; - while (l) { - JACK_DEBUG ("jackbin: unregistering pad %s:%s", - GST_JACK_PAD (l)->name, GST_JACK_PAD (l)->peer_name); - jack_port_unregister (this->client, GST_JACK_PAD (l)->port); - l = g_list_next (l); - } - GST_FLAG_UNSET (GST_OBJECT (this), GST_JACK_OPEN); + l = this->src_pads; + while (l) { + JACK_DEBUG ("jackbin: unregistering pad %s:%s", + GST_JACK_PAD (l)->name, GST_JACK_PAD (l)->peer_name); + jack_port_unregister (this->client, GST_JACK_PAD (l)->port); + l = g_list_next (l); + } + l = this->sink_pads; + while (l) { + JACK_DEBUG ("jackbin: unregistering pad %s:%s", + GST_JACK_PAD (l)->name, GST_JACK_PAD (l)->peer_name); + jack_port_unregister (this->client, GST_JACK_PAD (l)->port); + l = g_list_next (l); + } + GST_FLAG_UNSET (GST_OBJECT (this), GST_JACK_OPEN); - if (GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_ACTIVE)) { - JACK_DEBUG ("jackbin: deactivating client"); - jack_deactivate (this->client); - GST_FLAG_UNSET (GST_OBJECT (this), GST_JACK_ACTIVE); - } + if (GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_ACTIVE)) { + JACK_DEBUG ("jackbin: deactivating client"); + jack_deactivate (this->client); + GST_FLAG_UNSET (GST_OBJECT (this), GST_JACK_ACTIVE); + } } if (GST_ELEMENT_CLASS (parent_class)->change_state) - return GST_ELEMENT_CLASS (parent_class)->change_state (element); + return GST_ELEMENT_CLASS (parent_class)->change_state (element); break; case GST_STATE_PAUSED: JACK_DEBUG ("jackbin: PAUSED"); if (!GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_OPEN)) { - l = this->src_pads; - while (l) { - pad = GST_JACK_PAD (l); - JACK_DEBUG ("jackbin: registering input port %s (peer %s)", pad->name, - pad->peer_name); - pad->port = - jack_port_register (this->client, pad->name, - JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput | JackPortIsTerminal, 0); - l = g_list_next (l); - } - l = this->sink_pads; - while (l) { - pad = GST_JACK_PAD (l); - JACK_DEBUG ("jackbin: registering output port %s (peer %s)", - pad->name, pad->peer_name); - pad->port = - jack_port_register (this->client, pad->name, - JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput | JackPortIsTerminal, - 0); - l = g_list_next (l); - } + l = this->src_pads; + while (l) { + pad = GST_JACK_PAD (l); + JACK_DEBUG ("jackbin: registering input port %s (peer %s)", pad->name, + pad->peer_name); + pad->port = + jack_port_register (this->client, pad->name, + JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput | JackPortIsTerminal, 0); + l = g_list_next (l); + } + l = this->sink_pads; + while (l) { + pad = GST_JACK_PAD (l); + JACK_DEBUG ("jackbin: registering output port %s (peer %s)", + pad->name, pad->peer_name); + pad->port = + jack_port_register (this->client, pad->name, + JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput | JackPortIsTerminal, + 0); + l = g_list_next (l); + } - /* must activate before connecting */ - if (!GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_ACTIVE)) { - JACK_DEBUG ("jackbin: activating client"); - jack_activate (this->client); - GST_FLAG_SET (GST_OBJECT (this), GST_JACK_ACTIVE); - } + /* must activate before connecting */ + if (!GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_ACTIVE)) { + JACK_DEBUG ("jackbin: activating client"); + jack_activate (this->client); + GST_FLAG_SET (GST_OBJECT (this), GST_JACK_ACTIVE); + } - l = this->src_pads; - while (l) { - pad = GST_JACK_PAD (l); - JACK_DEBUG ("connecting jack port %s to gst jack port %s", - pad->peer_name, jack_port_name (pad->port)); - if (jack_connect (this->client, pad->peer_name, - jack_port_name (pad->port))) { - g_warning ("jackbin: could not connect %s and %s", pad->peer_name, - jack_port_name (pad->port)); - return GST_STATE_FAILURE; - } - l = g_list_next (l); - } - l = this->sink_pads; - while (l) { - pad = GST_JACK_PAD (l); - JACK_DEBUG ("connecting gst jack port %s to jack port %s", - jack_port_name (pad->port), pad->peer_name); - if (jack_connect (this->client, jack_port_name (pad->port), - pad->peer_name)) { - g_warning ("jackbin: could not connect %s and %s", pad->peer_name, - jack_port_name (pad->port)); - return GST_STATE_FAILURE; - } - l = g_list_next (l); - } + l = this->src_pads; + while (l) { + pad = GST_JACK_PAD (l); + JACK_DEBUG ("connecting jack port %s to gst jack port %s", + pad->peer_name, jack_port_name (pad->port)); + if (jack_connect (this->client, pad->peer_name, + jack_port_name (pad->port))) { + g_warning ("jackbin: could not connect %s and %s", pad->peer_name, + jack_port_name (pad->port)); + return GST_STATE_FAILURE; + } + l = g_list_next (l); + } + l = this->sink_pads; + while (l) { + pad = GST_JACK_PAD (l); + JACK_DEBUG ("connecting gst jack port %s to jack port %s", + jack_port_name (pad->port), pad->peer_name); + if (jack_connect (this->client, jack_port_name (pad->port), + pad->peer_name)) { + g_warning ("jackbin: could not connect %s and %s", pad->peer_name, + jack_port_name (pad->port)); + return GST_STATE_FAILURE; + } + l = g_list_next (l); + } - JACK_DEBUG ("jackbin: setting OPEN flag"); - GST_FLAG_SET (GST_OBJECT (this), GST_JACK_OPEN); + JACK_DEBUG ("jackbin: setting OPEN flag"); + GST_FLAG_SET (GST_OBJECT (this), GST_JACK_OPEN); - if (GST_ELEMENT_CLASS (parent_class)->change_state) - return GST_ELEMENT_CLASS (parent_class)->change_state (element); + if (GST_ELEMENT_CLASS (parent_class)->change_state) + return GST_ELEMENT_CLASS (parent_class)->change_state (element); } else { - if (GST_ELEMENT_CLASS (parent_class)->change_state) - return GST_ELEMENT_CLASS (parent_class)->change_state (element); + if (GST_ELEMENT_CLASS (parent_class)->change_state) + return GST_ELEMENT_CLASS (parent_class)->change_state (element); } break; @@ -248,7 +249,7 @@ gst_jack_bin_change_state (GstElement * element) JACK_DEBUG ("jackbin: PLAYING"); if (GST_ELEMENT_CLASS (parent_class)->change_state) - return GST_ELEMENT_CLASS (parent_class)->change_state (element); + return GST_ELEMENT_CLASS (parent_class)->change_state (element); break; } From da195d321181a05c683b8e6d60c72bad2b71efae Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?St=C3=A9phane=20Loeuillet?= Date: Fri, 21 May 2004 23:28:57 +0000 Subject: [PATCH 160/351] second batch : remove ',' at end of enums as they could confuse older gcc, foreign compilers (forte) and gtk-doc (in ... Original commit message from CVS: second batch : remove ',' at end of enums as they could confuse older gcc, foreign compilers (forte) and gtk-doc (in gst-plugins/ext/ this time) --- ext/jack/gstjack.c | 2 +- ext/jack/gstjack.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 643896a8a4..8b1296eac9 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -85,7 +85,7 @@ static void gst_jack_loop (GstElement * element); enum { ARG_0, - ARG_PORT_NAME_PREFIX, + ARG_PORT_NAME_PREFIX }; diff --git a/ext/jack/gstjack.h b/ext/jack/gstjack.h index 87d7d61151..e0b255a60e 100644 --- a/ext/jack/gstjack.h +++ b/ext/jack/gstjack.h @@ -67,7 +67,7 @@ typedef GstJackClass GstJackSrcClass; enum { GST_JACK_OPEN = GST_BIN_FLAG_LAST, GST_JACK_ACTIVE, - GST_JACK_FLAG_LAST = GST_BIN_FLAG_LAST + 3, + GST_JACK_FLAG_LAST = GST_BIN_FLAG_LAST + 3 }; From 567ac42d7e20a847621cc0bb0bba7564d5bb4306 Mon Sep 17 00:00:00 2001 From: Benjamin Otte Date: Tue, 3 Aug 2004 14:28:12 +0000 Subject: [PATCH 161/351] fixes for G_DISABLE_ASSERT and friends Original commit message from CVS: * examples/dynparams/filter.c: (ui_control_create): * examples/gstplay/player.c: (print_tag): * ext/alsa/gstalsa.c: (gst_alsa_request_new_pad): * ext/gdk_pixbuf/gstgdkanimation.c: (gst_gdk_animation_iter_may_advance): * ext/jack/gstjack.c: (gst_jack_request_new_pad): * ext/mad/gstid3tag.c: (gst_mad_id3_to_tag_list), (tag_list_to_id3_tag_foreach), (gst_id3_tag_handle_event): * ext/vorbis/oggvorbisenc.c: (gst_oggvorbisenc_get_tag_value): * ext/vorbis/vorbisenc.c: (gst_vorbisenc_get_tag_value): * ext/xine/xineaudiodec.c: (gst_xine_audio_dec_chain): * gst-libs/gst/media-info/media-info-test.c: (print_tag): * gst/sine/demo-dparams.c: (main): * gst/tags/gstvorbistag.c: (gst_tag_to_vorbis_comments): * testsuite/alsa/formats.c: (create_pipeline): * testsuite/alsa/sinesrc.c: (sinesrc_force_caps), (sinesrc_get): fixes for G_DISABLE_ASSERT and friends * gst/typefind/gsttypefindfunctions.c: (aac_type_find), (mp3_type_frame_length_from_header), (mp3_type_find), (plugin_init): require mp3 typefinding to have at least MIN_HEADERS valid headers add typefinding for AAC adts files --- ext/jack/gstjack.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 8b1296eac9..58678f8715 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -308,7 +308,8 @@ gst_jack_request_new_pad (GstElement * element, GstPadTemplate * templ, GstJackPad *pad; gint count; - g_return_val_if_fail ((this = GST_JACK (element)), NULL); + g_return_val_if_fail (GST_IS_JACK (element), NULL); + this = GST_JACK (element); if (!this->bin) pad_list = &this->pads; From 473a76d22120e711f2b38a15c4df4b4b1e60da1e Mon Sep 17 00:00:00 2001 From: Jan Schmidt Date: Mon, 5 Sep 2005 17:20:29 +0000 Subject: [PATCH 162/351] Fix up all the state change functions. Original commit message from CVS: Fix up all the state change functions. --- ext/jack/gstjack.c | 15 ++++++++------- ext/jack/gstjackbin.c | 30 ++++++++++++++++++------------ 2 files changed, 26 insertions(+), 19 deletions(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 58678f8715..4ea178de06 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -76,7 +76,8 @@ static GstPadTemplate *gst_jack_src_request_pad_factory (); static GstPadTemplate *gst_jack_sink_request_pad_factory (); static GstPad *gst_jack_request_new_pad (GstElement * element, GstPadTemplate * templ, const gchar * name); -static GstElementStateReturn gst_jack_change_state (GstElement * element); +static GstStateChangeReturn gst_jack_change_state (GstElement * element, + GstStateChange transition); static GstPadLinkReturn gst_jack_link (GstPad * pad, const GstCaps * caps); static void gst_jack_loop (GstElement * element); @@ -360,8 +361,8 @@ gst_jack_request_new_pad (GstElement * element, GstPadTemplate * templ, return pad->pad; } -static GstElementStateReturn -gst_jack_change_state (GstElement * element) +static GstStateChangeReturn +gst_jack_change_state (GstElement * element, GstStateChange transition) { GstJack *this; GList *l = NULL, **pads; @@ -386,7 +387,7 @@ gst_jack_change_state (GstElement * element) this->bin = NULL; g_warning ("jack element %s needs to be contained in a jack bin.", GST_OBJECT_NAME (element)); - return GST_STATE_FAILURE; + return GST_STATE_CHANGE_FAILURE; } /* fixme: verify that all names are unique */ @@ -417,7 +418,7 @@ gst_jack_change_state (GstElement * element) "rate", G_TYPE_INT, (int) this->bin->rate, "buffer-frames", G_TYPE_INT, (gint) this->bin->nframes, NULL); if (gst_pad_try_set_caps (pad->pad, caps) <= 0) - return GST_STATE_FAILURE; + return GST_STATE_CHANGE_FAILURE; l = g_list_next (l); } } @@ -430,9 +431,9 @@ gst_jack_change_state (GstElement * element) JACK_DEBUG ("%s: state change finished", GST_OBJECT_NAME (this)); if (GST_ELEMENT_CLASS (parent_class)->change_state) - return GST_ELEMENT_CLASS (parent_class)->change_state (element); + return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); - return GST_STATE_SUCCESS; + return GST_STATE_CHANGE_SUCCESS; } static GstPadLinkReturn diff --git a/ext/jack/gstjackbin.c b/ext/jack/gstjackbin.c index c2c4957015..85a31b1b89 100644 --- a/ext/jack/gstjackbin.c +++ b/ext/jack/gstjackbin.c @@ -32,7 +32,8 @@ static GstBinClass *parent_class = NULL; static void gst_jack_bin_init (GstJackBin * this); static void gst_jack_bin_class_init (GstJackBinClass * klass); -static GstElementStateReturn gst_jack_bin_change_state (GstElement * element); +static GstStateChangeReturn gst_jack_bin_change_state (GstElement * element, + GstStateChange transition); /* jack callbacks */ static int process (jack_nframes_t nframes, void *arg); @@ -98,8 +99,8 @@ gst_jack_bin_init (GstJackBin * this) gst_scheduler_factory_make (NULL, GST_ELEMENT (this)); } -static GstElementStateReturn -gst_jack_bin_change_state (GstElement * element) +static GstStateChangeReturn +gst_jack_bin_change_state (GstElement * element, GstStateChange transition) { GstJackBin *this; GList *l = NULL; @@ -122,7 +123,8 @@ gst_jack_bin_change_state (GstElement * element) _jackbin = NULL; if (GST_ELEMENT_CLASS (parent_class)->change_state) - return GST_ELEMENT_CLASS (parent_class)->change_state (element); + return GST_ELEMENT_CLASS (parent_class)->change_state (element, + transition); break; case GST_STATE_READY: @@ -134,7 +136,7 @@ gst_jack_bin_change_state (GstElement * element) if (!this->client) { if (!(this->client = jack_client_new ("gst-jack"))) { g_warning ("jack server not running?"); - return GST_STATE_FAILURE; + return GST_STATE_CHANGE_FAILURE; } gst_scheduler_setup (GST_ELEMENT_SCHED (this)); @@ -171,7 +173,8 @@ gst_jack_bin_change_state (GstElement * element) } if (GST_ELEMENT_CLASS (parent_class)->change_state) - return GST_ELEMENT_CLASS (parent_class)->change_state (element); + return GST_ELEMENT_CLASS (parent_class)->change_state (element, + transition); break; case GST_STATE_PAUSED: @@ -216,7 +219,7 @@ gst_jack_bin_change_state (GstElement * element) jack_port_name (pad->port))) { g_warning ("jackbin: could not connect %s and %s", pad->peer_name, jack_port_name (pad->port)); - return GST_STATE_FAILURE; + return GST_STATE_CHANGE_FAILURE; } l = g_list_next (l); } @@ -229,7 +232,7 @@ gst_jack_bin_change_state (GstElement * element) pad->peer_name)) { g_warning ("jackbin: could not connect %s and %s", pad->peer_name, jack_port_name (pad->port)); - return GST_STATE_FAILURE; + return GST_STATE_CHANGE_FAILURE; } l = g_list_next (l); } @@ -238,10 +241,12 @@ gst_jack_bin_change_state (GstElement * element) GST_FLAG_SET (GST_OBJECT (this), GST_JACK_OPEN); if (GST_ELEMENT_CLASS (parent_class)->change_state) - return GST_ELEMENT_CLASS (parent_class)->change_state (element); + return GST_ELEMENT_CLASS (parent_class)->change_state (element, + transition); } else { if (GST_ELEMENT_CLASS (parent_class)->change_state) - return GST_ELEMENT_CLASS (parent_class)->change_state (element); + return GST_ELEMENT_CLASS (parent_class)->change_state (element, + transition); } break; @@ -249,13 +254,14 @@ gst_jack_bin_change_state (GstElement * element) JACK_DEBUG ("jackbin: PLAYING"); if (GST_ELEMENT_CLASS (parent_class)->change_state) - return GST_ELEMENT_CLASS (parent_class)->change_state (element); + return GST_ELEMENT_CLASS (parent_class)->change_state (element, + transition); break; } JACK_DEBUG ("jackbin: state change finished"); - return GST_STATE_SUCCESS; + return GST_STATE_CHANGE_SUCCESS; } /* jack callbacks */ From c038b32b44f63335116a67a195d1baac79ed2221 Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Wed, 12 Oct 2005 14:29:55 +0000 Subject: [PATCH 163/351] renamed GST_FLAGS macros to GST_OBJECT_FLAGS moved bitshift from macro to enum definition Original commit message from CVS: * examples/indexing/indexmpeg.c: (main): * ext/artsd/gstartsdsink.c: (gst_artsdsink_open_audio), (gst_artsdsink_close_audio), (gst_artsdsink_change_state): * ext/artsd/gstartsdsink.h: * ext/audiofile/gstafparse.c: (gst_afparse_open_file), (gst_afparse_close_file): * ext/audiofile/gstafparse.h: * ext/audiofile/gstafsink.c: (gst_afsink_open_file), (gst_afsink_close_file), (gst_afsink_chain), (gst_afsink_change_state): * ext/audiofile/gstafsink.h: * ext/audiofile/gstafsrc.c: (gst_afsrc_open_file), (gst_afsrc_close_file), (gst_afsrc_change_state): * ext/audiofile/gstafsrc.h: * ext/cdaudio/gstcdaudio.c: (gst_cdaudio_init): * ext/directfb/directfbvideosink.c: (gst_directfbvideosink_init): * ext/dts/gstdtsdec.c: (gst_dtsdec_init): * ext/jack/gstjack.h: * ext/jack/gstjackbin.c: (gst_jack_bin_init), (gst_jack_bin_change_state): * ext/musepack/gstmusepackdec.c: (gst_musepackdec_init): * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_init): * ext/nas/nassink.c: (gst_nassink_open_audio), (gst_nassink_close_audio), (gst_nassink_change_state): * ext/nas/nassink.h: * ext/polyp/polypsink.c: (gst_polypsink_init): * ext/sdl/sdlvideosink.c: (gst_sdlvideosink_change_state): * ext/sdl/sdlvideosink.h: * ext/smoothwave/gstsmoothwave.c: (gst_smoothwave_init): * ext/sndfile/gstsf.c: (gst_sf_set_property), (gst_sf_change_state), (gst_sf_release_request_pad), (gst_sf_open_file), (gst_sf_close_file), (gst_sf_loop): * ext/sndfile/gstsf.h: * ext/swfdec/gstswfdec.c: (gst_swfdec_init): * ext/tarkin/gsttarkindec.c: (gst_tarkindec_init): * gst/apetag/apedemux.c: (gst_ape_demux_init): * gst/cdxaparse/gstcdxaparse.c: (gst_cdxaparse_init): * gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_init): * gst/festival/gstfestival.c: (gst_festival_change_state): * gst/festival/gstfestival.h: * gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_init): * gst/multifilesink/gstmultifilesink.c: (gst_multifilesink_init), (gst_multifilesink_set_location), (gst_multifilesink_open_file), (gst_multifilesink_close_file), (gst_multifilesink_next_file), (gst_multifilesink_pad_query), (gst_multifilesink_handle_event), (gst_multifilesink_chain), (gst_multifilesink_change_state): * gst/multifilesink/gstmultifilesink.h: * gst/videodrop/gstvideodrop.c: (gst_videodrop_init): * sys/cdrom/gstcdplayer.c: (cdplayer_init): * sys/dxr3/dxr3audiosink.c: (dxr3audiosink_init), (dxr3audiosink_open), (dxr3audiosink_close), (dxr3audiosink_chain_pcm), (dxr3audiosink_chain_ac3), (dxr3audiosink_change_state): * sys/dxr3/dxr3audiosink.h: * sys/dxr3/dxr3spusink.c: (dxr3spusink_init), (dxr3spusink_open), (dxr3spusink_close), (dxr3spusink_chain), (dxr3spusink_change_state): * sys/dxr3/dxr3spusink.h: * sys/dxr3/dxr3videosink.c: (dxr3videosink_init), (dxr3videosink_open), (dxr3videosink_close), (dxr3videosink_write_data), (dxr3videosink_change_state): * sys/dxr3/dxr3videosink.h: * sys/glsink/glimagesink.c: (gst_glimagesink_init): * sys/qcam/gstqcamsrc.c: (gst_qcamsrc_change_state), (gst_qcamsrc_open), (gst_qcamsrc_close): * sys/qcam/gstqcamsrc.h: * sys/v4l2/gstv4l2src.c: (gst_v4l2src_init): * sys/vcd/vcdsrc.c: (gst_vcdsrc_set_property), (gst_vcdsrc_get), (gst_vcdsrc_open_file), (gst_vcdsrc_close_file), (gst_vcdsrc_change_state), (gst_vcdsrc_recalculate): * sys/vcd/vcdsrc.h: renamed GST_FLAGS macros to GST_OBJECT_FLAGS moved bitshift from macro to enum definition --- ext/jack/gstjack.h | 6 +++--- ext/jack/gstjackbin.c | 20 ++++++++++---------- 2 files changed, 13 insertions(+), 13 deletions(-) diff --git a/ext/jack/gstjack.h b/ext/jack/gstjack.h index e0b255a60e..738168ca7e 100644 --- a/ext/jack/gstjack.h +++ b/ext/jack/gstjack.h @@ -65,9 +65,9 @@ typedef GstJackClass GstJackSrcClass; enum { - GST_JACK_OPEN = GST_BIN_FLAG_LAST, - GST_JACK_ACTIVE, - GST_JACK_FLAG_LAST = GST_BIN_FLAG_LAST + 3 + GST_JACK_OPEN = (GST_BIN_FLAG_LAST << 0), + GST_JACK_ACTIVE = (GST_BIN_FLAG_LAST << 1), + GST_JACK_FLAG_LAST = (GST_BIN_FLAG_LAST << 3) }; diff --git a/ext/jack/gstjackbin.c b/ext/jack/gstjackbin.c index 85a31b1b89..1d79b72e6f 100644 --- a/ext/jack/gstjackbin.c +++ b/ext/jack/gstjackbin.c @@ -92,8 +92,8 @@ gst_jack_bin_init (GstJackBin * this) GST_DEBUG ("initializing jack bin"); /* jack bins are managing bins and iterate themselves */ - GST_FLAG_SET (this, GST_BIN_FLAG_MANAGER); - GST_FLAG_SET (this, GST_BIN_SELF_SCHEDULABLE); + GST_OBJECT_FLAG_SET (this, GST_BIN_FLAG_MANAGER); + GST_OBJECT_FLAG_SET (this, GST_BIN_SELF_SCHEDULABLE); /* make a new scheduler and associate it with the bin */ gst_scheduler_factory_make (NULL, GST_ELEMENT (this)); @@ -148,7 +148,7 @@ gst_jack_bin_change_state (GstElement * element, GstStateChange transition) jack_on_shutdown (this->client, shutdown, this); } - if (GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_OPEN)) { + if (GST_OBJECT_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_OPEN)) { l = this->src_pads; while (l) { JACK_DEBUG ("jackbin: unregistering pad %s:%s", @@ -163,12 +163,12 @@ gst_jack_bin_change_state (GstElement * element, GstStateChange transition) jack_port_unregister (this->client, GST_JACK_PAD (l)->port); l = g_list_next (l); } - GST_FLAG_UNSET (GST_OBJECT (this), GST_JACK_OPEN); + GST_OBJECT_FLAG_UNSET (GST_OBJECT (this), GST_JACK_OPEN); - if (GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_ACTIVE)) { + if (GST_OBJECT_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_ACTIVE)) { JACK_DEBUG ("jackbin: deactivating client"); jack_deactivate (this->client); - GST_FLAG_UNSET (GST_OBJECT (this), GST_JACK_ACTIVE); + GST_OBJECT_FLAG_UNSET (GST_OBJECT (this), GST_JACK_ACTIVE); } } @@ -180,7 +180,7 @@ gst_jack_bin_change_state (GstElement * element, GstStateChange transition) case GST_STATE_PAUSED: JACK_DEBUG ("jackbin: PAUSED"); - if (!GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_OPEN)) { + if (!GST_OBJECT_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_OPEN)) { l = this->src_pads; while (l) { pad = GST_JACK_PAD (l); @@ -204,10 +204,10 @@ gst_jack_bin_change_state (GstElement * element, GstStateChange transition) } /* must activate before connecting */ - if (!GST_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_ACTIVE)) { + if (!GST_OBJECT_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_ACTIVE)) { JACK_DEBUG ("jackbin: activating client"); jack_activate (this->client); - GST_FLAG_SET (GST_OBJECT (this), GST_JACK_ACTIVE); + GST_OBJECT_FLAG_SET (GST_OBJECT (this), GST_JACK_ACTIVE); } l = this->src_pads; @@ -238,7 +238,7 @@ gst_jack_bin_change_state (GstElement * element, GstStateChange transition) } JACK_DEBUG ("jackbin: setting OPEN flag"); - GST_FLAG_SET (GST_OBJECT (this), GST_JACK_OPEN); + GST_OBJECT_FLAG_SET (GST_OBJECT (this), GST_JACK_OPEN); if (GST_ELEMENT_CLASS (parent_class)->change_state) return GST_ELEMENT_CLASS (parent_class)->change_state (element, From cef77bfc9c843334f593ec3c2905837c06e21330 Mon Sep 17 00:00:00 2001 From: Thomas Vander Stichele Date: Sat, 1 Apr 2006 10:09:11 +0000 Subject: [PATCH 164/351] rework build; add translations for v4l2 Original commit message from CVS: rework build; add translations for v4l2 --- ext/jack/gstjack.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 4ea178de06..1afab02a6d 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -531,4 +531,5 @@ plugin_init (GstPlugin * plugin) GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, "jack", - "Jack Plugin Library", plugin_init, VERSION, "GPL", GST_PACKAGE, GST_ORIGIN) + "Jack Plugin Library", plugin_init, VERSION, "GPL", GST_PACKAGE_NAME, + GST_PACKAGE_ORIGIN) From 8fa70e36227fb40b43863323baacd3bb13e67f25 Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Sat, 8 Apr 2006 21:48:01 +0000 Subject: [PATCH 165/351] Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent) Original commit message from CVS: * ext/amrwb/gstamrwbdec.c: (gst_amrwbdec_class_init): * ext/amrwb/gstamrwbenc.c: (gst_amrwbenc_class_init): * ext/amrwb/gstamrwbparse.c: (gst_amrwbparse_class_init): * ext/arts/gst_arts.c: (gst_arts_class_init): * ext/artsd/gstartsdsink.c: (gst_artsdsink_class_init): * ext/audiofile/gstafsink.c: (gst_afsink_class_init): * ext/audiofile/gstafsrc.c: (gst_afsrc_class_init): * ext/audioresample/gstaudioresample.c: * ext/cdaudio/gstcdaudio.c: (gst_cdaudio_class_init): * ext/directfb/dfbvideosink.c: (gst_dfbvideosink_class_init): * ext/divx/gstdivxdec.c: (gst_divxdec_class_init): * ext/hermes/gsthermescolorspace.c: (gst_hermes_colorspace_class_init): * ext/ivorbis/vorbisfile.c: (gst_ivorbisfile_class_init): * ext/jack/gstjack.c: (gst_jack_class_init): * ext/jack/gstjackbin.c: (gst_jack_bin_class_init): * ext/lcs/gstcolorspace.c: (gst_colorspace_class_init): * ext/libfame/gstlibfame.c: (gst_fameenc_class_init): * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_class_init): * ext/nas/nassink.c: (gst_nassink_class_init): * ext/shout/gstshout.c: (gst_icecastsend_class_init): * ext/snapshot/gstsnapshot.c: (gst_snapshot_class_init): * ext/sndfile/gstsf.c: (gst_sf_class_init): * ext/swfdec/gstswfdec.c: (gst_swfdecbuffer_class_init), (gst_swfdec_class_init): * ext/tarkin/gsttarkindec.c: (gst_tarkindec_class_init): * ext/tarkin/gsttarkinenc.c: (gst_tarkinenc_class_init): * gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_class_init): * gst/chart/gstchart.c: (gst_chart_class_init): * gst/colorspace/gstcolorspace.c: (gst_colorspace_class_init): * gst/deinterlace/gstdeinterlace.c: (gst_deinterlace_class_init): * gst/festival/gstfestival.c: (gst_festival_class_init): * gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init): * gst/filter/gstiir.c: (gst_iir_class_init): * gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init): * gst/librfb/gstrfbsrc.c: (gst_rfbsrc_class_init): * gst/mixmatrix/mixmatrix.c: (gst_mixmatrix_class_init): * gst/mpeg1sys/gstmpeg1systemencode.c: (gst_system_encode_class_init): * gst/mpeg1videoparse/gstmp1videoparse.c: (gst_mp1videoparse_class_init): * gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_class_init): * gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_class_init): * gst/overlay/gstoverlay.c: (gst_overlay_class_init): * gst/passthrough/gstpassthrough.c: (passthrough_class_init): * gst/playondemand/gstplayondemand.c: (play_on_demand_class_init): * gst/rtjpeg/gstrtjpegdec.c: (gst_rtjpegdec_class_init): * gst/rtjpeg/gstrtjpegenc.c: (gst_rtjpegenc_class_init): * gst/smooth/gstsmooth.c: (gst_smooth_class_init): * gst/smoothwave/gstsmoothwave.c: (gst_smoothwave_class_init): * gst/spectrum/gstspectrum.c: (gst_spectrum_class_init): * gst/stereo/gststereo.c: (gst_stereo_class_init): * gst/switch/gstswitch.c: (gst_switch_class_init): * gst/tta/gstttadec.c: (gst_tta_dec_class_init): * gst/tta/gstttaparse.c: (gst_tta_parse_class_init): * gst/vbidec/gstvbidec.c: (gst_vbidec_class_init): * gst/videocrop/gstvideocrop.c: (gst_video_crop_class_init): * gst/virtualdub/gstxsharpen.c: (gst_xsharpen_class_init): * gst/y4m/gsty4mencode.c: (gst_y4mencode_class_init): * sys/cdrom/gstcdplayer.c: (cdplayer_class_init): * sys/directsound/gstdirectsoundsink.c: (gst_directsoundsink_class_init): * sys/dxr3/dxr3audiosink.c: (dxr3audiosink_class_init): * sys/dxr3/dxr3spusink.c: (dxr3spusink_class_init): * sys/dxr3/dxr3videosink.c: (dxr3videosink_class_init): * sys/qcam/gstqcamsrc.c: (gst_qcamsrc_class_init): * sys/v4l2/gstv4l2colorbalance.c: (gst_v4l2_color_balance_channel_class_init): * sys/v4l2/gstv4l2tuner.c: (gst_v4l2_tuner_channel_class_init), (gst_v4l2_tuner_norm_class_init): * sys/ximagesrc/ximagesrc.c: (gst_ximagesrc_class_init): Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent) --- ext/jack/gstjack.c | 2 +- ext/jack/gstjackbin.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 1afab02a6d..2f6affcc25 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -202,7 +202,7 @@ gst_jack_class_init (GstJackClass * klass) element_class = (GstElementClass *) klass; if (parent_class == NULL) - parent_class = g_type_class_ref (GST_TYPE_ELEMENT); + parent_class = g_type_class_peek_parent (klass); object_class->get_property = gst_jack_get_property; object_class->set_property = gst_jack_set_property; diff --git a/ext/jack/gstjackbin.c b/ext/jack/gstjackbin.c index 1d79b72e6f..7b219c7068 100644 --- a/ext/jack/gstjackbin.c +++ b/ext/jack/gstjackbin.c @@ -81,7 +81,7 @@ gst_jack_bin_class_init (GstJackBinClass * klass) object_class = (GObjectClass *) klass; element_class = (GstElementClass *) klass; - parent_class = g_type_class_ref (GST_TYPE_BIN); + parent_class = g_type_class_peek_parent (klass); element_class->change_state = gst_jack_bin_change_state; } From 32aaf1e95df9d65caa510f23682d683fb194155c Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Thu, 30 Nov 2006 11:49:36 +0000 Subject: [PATCH 166/351] Added fully functional jackaudiosink. Original commit message from CVS: * configure.ac: * ext/Makefile.am: * ext/jack/Makefile.am: * ext/jack/gstjack.c: (plugin_init): * ext/jack/gstjack.h: * ext/jack/gstjackaudiosink.c: (gst_jack_ring_buffer_get_type), (gst_jack_ring_buffer_class_init), (jack_process_cb), (jack_sample_rate_cb), (jack_buffer_size_cb), (jack_shutdown_cb), (gst_jack_ring_buffer_init), (gst_jack_ring_buffer_dispose), (gst_jack_ring_buffer_finalize), (gst_jack_ring_buffer_open_device), (gst_jack_ring_buffer_close_device), (gst_jack_ring_buffer_acquire), (gst_jack_ring_buffer_release), (gst_jack_ring_buffer_start), (gst_jack_ring_buffer_pause), (gst_jack_ring_buffer_stop), (gst_jack_ring_buffer_delay), (gst_jack_connect_get_type), (gst_jack_audio_sink_base_init), (gst_jack_audio_sink_class_init), (gst_jack_audio_sink_init), (gst_jack_audio_sink_set_property), (gst_jack_audio_sink_get_property), (gst_jack_audio_sink_getcaps), (gst_jack_audio_sink_create_ringbuffer): * ext/jack/gstjackaudiosink.h: Added fully functional jackaudiosink. --- ext/jack/Makefile.am | 8 +- ext/jack/gstjack.c | 542 +---------------------- ext/jack/gstjack.h | 129 ------ ext/jack/gstjackaudiosink.c | 846 ++++++++++++++++++++++++++++++++++++ ext/jack/gstjackaudiosink.h | 80 ++++ 5 files changed, 954 insertions(+), 651 deletions(-) delete mode 100644 ext/jack/gstjack.h create mode 100644 ext/jack/gstjackaudiosink.c create mode 100644 ext/jack/gstjackaudiosink.h diff --git a/ext/jack/Makefile.am b/ext/jack/Makefile.am index a97a2ad7a9..0a237dd6c9 100644 --- a/ext/jack/Makefile.am +++ b/ext/jack/Makefile.am @@ -1,11 +1,11 @@ plugin_LTLIBRARIES = libgstjack.la -libgstjack_la_SOURCES = gstjack.c gstjackbin.c -libgstjack_la_CFLAGS = $(GST_CFLAGS) $(JACK_CFLAGS) -libgstjack_la_LIBADD = $(JACK_LIBS) +libgstjack_la_SOURCES = gstjack.c gstjackaudiosink.c +libgstjack_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) $(JACK_CFLAGS) +libgstjack_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR) $(JACK_LIBS) libgstjack_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) -noinst_HEADERS = gstjack.h +noinst_HEADERS = gstjackaudiosink.h EXTRA_DIST = README diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 2f6affcc25..52b4b2455e 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -1,528 +1,33 @@ -/* -*- Mode: C; c-basic-offset: 4 -*- */ -/* - Copyright (C) 2002, 2003 Andy Wingo - - This library is free software; you can redistribute it and/or - modify it under the terms of the GNU General Public - License as published by the Free Software Foundation; either - version 2 of the License, or (at your option) any later version. - - This library is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the implied warranty of - MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - General Public License for more details. - - You should have received a copy of the GNU General Public - License along with this library; if not, write to the Free - Software Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. -*/ +/* GStreamer Jack plugins + * Copyright (C) 2006 Wim Taymans + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ #ifdef HAVE_CONFIG_H #include "config.h" #endif -#include -#include -#include "gstjack.h" -#include -/* TODO: - - - work out the src side (caps setting, etc) - - future core TODO: - - make a jack clock provider - - add GST_ELEMENT_FIXED_DATA_RATE, GST_ELEMENT_QOS, - GST_ELEMENT_CHANGES_DATA_RATE element flags, and make the scheduler - sensitive to them -*/ - -/* elementfactory information */ -static GstElementDetails gst_jack_bin_details = { - "Jack Bin", - "Generic/Bin", - "Jack processing bin", - "Andy Wingo ", -}; - -static GstElementDetails gst_jack_sink_details = { - "Jack Sink", - "Sink/Audio", - "Output to a Jack processing network", - "Andy Wingo ", -}; - -static GstElementDetails gst_jack_src_details = { - "Jack Src", - "Source/Audio", - "Input from a Jack processing network", - "Andy Wingo ", -}; - - -static GHashTable *port_name_counts = NULL; -static GstElementClass *parent_class = NULL; - -static void gst_jack_base_init (gpointer g_class); -static void gst_jack_src_base_init (gpointer g_class); -static void gst_jack_sink_base_init (gpointer g_class); -static void gst_jack_init (GstJack * this); -static void gst_jack_class_init (GstJackClass * klass); -static void gst_jack_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec); -static void gst_jack_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec); - -static GstPadTemplate *gst_jack_src_request_pad_factory (); -static GstPadTemplate *gst_jack_sink_request_pad_factory (); -static GstPad *gst_jack_request_new_pad (GstElement * element, - GstPadTemplate * templ, const gchar * name); -static GstStateChangeReturn gst_jack_change_state (GstElement * element, - GstStateChange transition); -static GstPadLinkReturn gst_jack_link (GstPad * pad, const GstCaps * caps); - -static void gst_jack_loop (GstElement * element); - - -enum -{ - ARG_0, - ARG_PORT_NAME_PREFIX -}; - - -GType -gst_jack_get_type (void) -{ - static GType jack_type = 0; - - if (!jack_type) { - static const GTypeInfo jack_info = { - sizeof (GstJackClass), - gst_jack_base_init, - NULL, - NULL, - NULL, - NULL, - sizeof (GstJack), - 0, - NULL, - }; - - jack_type = - g_type_register_static (GST_TYPE_ELEMENT, "GstJack", &jack_info, 0); - } - return jack_type; -} - -GType -gst_jack_sink_get_type (void) -{ - static GType jack_type = 0; - - if (!jack_type) { - static const GTypeInfo jack_info = { - sizeof (GstJackClass), - gst_jack_sink_base_init, - NULL, - (GClassInitFunc) gst_jack_class_init, - NULL, - NULL, - sizeof (GstJack), - 0, - (GInstanceInitFunc) gst_jack_init, - }; - - jack_type = - g_type_register_static (GST_TYPE_JACK, "GstJackSink", &jack_info, 0); - } - return jack_type; -} - -GType -gst_jack_src_get_type (void) -{ - static GType jack_type = 0; - - if (!jack_type) { - static const GTypeInfo jack_info = { - sizeof (GstJackClass), - gst_jack_src_base_init, - NULL, - (GClassInitFunc) gst_jack_class_init, - NULL, - NULL, - sizeof (GstJack), - 0, - (GInstanceInitFunc) gst_jack_init, - }; - - jack_type = - g_type_register_static (GST_TYPE_JACK, "GstJackSrc", &jack_info, 0); - } - return jack_type; -} - -static void -gst_jack_base_init (gpointer g_class) -{ - GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); - - gst_element_class_set_details (element_class, &gst_jack_bin_details); -} - -static void -gst_jack_src_base_init (gpointer g_class) -{ - GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); - - gst_element_class_add_pad_template (element_class, - gst_jack_src_request_pad_factory ()); - gst_element_class_set_details (element_class, &gst_jack_src_details); -} - -static void -gst_jack_sink_base_init (gpointer g_class) -{ - GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); - - gst_element_class_add_pad_template (element_class, - gst_jack_sink_request_pad_factory ()); - gst_element_class_set_details (element_class, &gst_jack_sink_details); -} - -static void -gst_jack_class_init (GstJackClass * klass) -{ - GObjectClass *object_class; - GstElementClass *element_class; - GParamSpec *pspec; - gchar *prefix; - - object_class = (GObjectClass *) klass; - element_class = (GstElementClass *) klass; - - if (parent_class == NULL) - parent_class = g_type_class_peek_parent (klass); - - object_class->get_property = gst_jack_get_property; - object_class->set_property = gst_jack_set_property; - - if (GST_IS_JACK_SINK_CLASS (klass)) - prefix = "gst-out-"; - else - prefix = "gst-in-"; - - pspec = g_param_spec_string ("port-name-prefix", "Port name prefix", - "String to prepend to jack port names", - prefix, G_PARAM_READWRITE | G_PARAM_CONSTRUCT); - g_object_class_install_property (object_class, ARG_PORT_NAME_PREFIX, pspec); - - element_class->change_state = gst_jack_change_state; - - element_class->request_new_pad = gst_jack_request_new_pad; -} - -static void -gst_jack_init (GstJack * this) -{ - if (G_OBJECT_TYPE (this) == GST_TYPE_JACK_SRC) - this->direction = GST_PAD_SRC; - else if (G_OBJECT_TYPE (this) == GST_TYPE_JACK_SINK) - this->direction = GST_PAD_SINK; - else - g_assert_not_reached (); - - gst_element_set_loop_function (GST_ELEMENT (this), gst_jack_loop); -} - -static void -gst_jack_set_property (GObject * object, guint prop_id, const GValue * value, - GParamSpec * pspec) -{ - GstJack *this = (GstJack *) object; - - switch (prop_id) { - case ARG_PORT_NAME_PREFIX: - if (this->port_name_prefix) - g_free (this->port_name_prefix); - this->port_name_prefix = g_strdup (g_value_get_string (value)); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - return; - } -} - -static void -gst_jack_get_property (GObject * object, guint prop_id, GValue * value, - GParamSpec * pspec) -{ - GstJack *this = (GstJack *) object; - - switch (prop_id) { - case ARG_PORT_NAME_PREFIX: - g_value_set_string (value, this->port_name_prefix); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static GstPadTemplate * -gst_jack_src_request_pad_factory (void) -{ - static GstPadTemplate *template = NULL; - - if (!template) { - GstCaps *caps; - - caps = gst_caps_from_string (GST_AUDIO_FLOAT_STANDARD_PAD_TEMPLATE_CAPS); - template = gst_pad_template_new ("%s", GST_PAD_SRC, GST_PAD_REQUEST, caps); - } - - return template; -} - -static GstPadTemplate * -gst_jack_sink_request_pad_factory (void) -{ - static GstPadTemplate *template = NULL; - - if (!template) { - GstCaps *caps; - - caps = gst_caps_from_string (GST_AUDIO_FLOAT_STANDARD_PAD_TEMPLATE_CAPS); - template = gst_pad_template_new ("%s", GST_PAD_SINK, GST_PAD_REQUEST, caps); - } - - return template; -} - -static GstPad * -gst_jack_request_new_pad (GstElement * element, GstPadTemplate * templ, - const gchar * name) -{ - GstJack *this; - gchar *newname; - GList *l, **pad_list; - GstJackPad *pad; - gint count; - - g_return_val_if_fail (GST_IS_JACK (element), NULL); - this = GST_JACK (element); - - if (!this->bin) - pad_list = &this->pads; - else if (this->direction == GST_PAD_SRC) - pad_list = &this->bin->src_pads; - else - pad_list = &this->bin->sink_pads; - - if (name) { - l = *pad_list; - while (l) { - if (strcmp (GST_JACK_PAD (l)->name, name) == 0) { - g_warning ("requested port name %s already in use.", name); - return NULL; - } - l = l->next; - } - newname = g_strdup (name); - } else { - if (this->direction == GST_PAD_SINK) - newname = g_strdup ("alsa_pcm:playback_1"); - else - newname = g_strdup ("alsa_pcm:capture_1"); - } - - pad = g_new0 (GstJackPad, 1); - - if (!port_name_counts) - port_name_counts = g_hash_table_new (g_str_hash, g_str_equal); - - count = - GPOINTER_TO_INT (g_hash_table_lookup (port_name_counts, - this->port_name_prefix)); - g_hash_table_insert (port_name_counts, g_strdup (this->port_name_prefix), - GINT_TO_POINTER (count + 1)); - - pad->name = g_strdup_printf ("%s%d", this->port_name_prefix, count); - - pad->peer_name = newname; - pad->pad = gst_pad_new_from_template (templ, newname); - gst_element_add_pad (GST_ELEMENT (this), pad->pad); - gst_pad_set_link_function (pad->pad, gst_jack_link); - - this->pads = g_list_append (this->pads, pad); - - g_print ("returning from request_new_pad, pad %s created, to connect to %s\n", - pad->name, pad->peer_name); - return pad->pad; -} - -static GstStateChangeReturn -gst_jack_change_state (GstElement * element, GstStateChange transition) -{ - GstJack *this; - GList *l = NULL, **pads; - GstJackPad *pad; - GstCaps *caps; - - g_return_val_if_fail (element != NULL, FALSE); - this = GST_JACK (element); - - switch (GST_STATE_PENDING (element)) { - case GST_STATE_NULL: - JACK_DEBUG ("%s: NULL", GST_OBJECT_NAME (GST_OBJECT (this))); - - break; - - case GST_STATE_READY: - JACK_DEBUG ("%s: READY", GST_OBJECT_NAME (GST_OBJECT (this))); - - if (!this->bin) { - if (!(this->bin = (GstJackBin *) gst_element_get_managing_bin (element)) - || !GST_IS_JACK_BIN (this->bin)) { - this->bin = NULL; - g_warning ("jack element %s needs to be contained in a jack bin.", - GST_OBJECT_NAME (element)); - return GST_STATE_CHANGE_FAILURE; - } - - /* fixme: verify that all names are unique */ - l = this->pads; - pads = - (this->direction == - GST_PAD_SRC) ? &this->bin->src_pads : &this->bin->sink_pads; - while (l) { - pad = GST_JACK_PAD (l); - JACK_DEBUG ("%s: appending pad %s:%s to list", GST_OBJECT_NAME (this), - pad->name, pad->peer_name); - *pads = g_list_append (*pads, pad); - l = g_list_next (l); - } - } - break; - - case GST_STATE_PAUSED: - JACK_DEBUG ("%s: PAUSED", GST_OBJECT_NAME (GST_OBJECT (this))); - - if (GST_STATE (element) == GST_STATE_READY) { - /* we're in READY->PAUSED */ - l = this->pads; - while (l) { - pad = GST_JACK_PAD (l); - caps = gst_caps_copy (gst_pad_get_negotiated_caps (pad->pad)); - gst_caps_set_simple (caps, - "rate", G_TYPE_INT, (int) this->bin->rate, - "buffer-frames", G_TYPE_INT, (gint) this->bin->nframes, NULL); - if (gst_pad_try_set_caps (pad->pad, caps) <= 0) - return GST_STATE_CHANGE_FAILURE; - l = g_list_next (l); - } - } - break; - case GST_STATE_PLAYING: - JACK_DEBUG ("%s: PLAYING", GST_OBJECT_NAME (GST_OBJECT (this))); - break; - } - - JACK_DEBUG ("%s: state change finished", GST_OBJECT_NAME (this)); - - if (GST_ELEMENT_CLASS (parent_class)->change_state) - return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); - - return GST_STATE_CHANGE_SUCCESS; -} - -static GstPadLinkReturn -gst_jack_link (GstPad * pad, const GstCaps * caps) -{ - GstJack *this; - gint rate, buffer_frames; - GstStructure *structure; - - this = GST_JACK (GST_OBJECT_PARENT (pad)); - - structure = gst_caps_get_structure (caps, 0); - gst_structure_get_int (structure, "rate", &rate); - gst_structure_get_int (structure, "buffer-frames", &buffer_frames); - if (this->bin && (rate != this->bin->rate || - buffer_frames != this->bin->nframes)) - return GST_PAD_LINK_REFUSED; - - return GST_PAD_LINK_OK; -} - -static void -gst_jack_loop (GstElement * element) -{ - GstJack *this; - GList *pads; - gint len; - GstJackPad *pad; - GstBuffer *buffer; - - this = GST_JACK (element); - - len = this->bin->nframes * sizeof (sample_t); - - pads = this->pads; - while (pads) { - pad = GST_JACK_PAD (pads); - - if (this->direction == GST_PAD_SINK) { - buffer = GST_BUFFER (gst_pad_pull (pad->pad)); - - if (GST_IS_EVENT (buffer)) { - GstEvent *event = GST_EVENT (buffer); - - switch (GST_EVENT_TYPE (buffer)) { - case GST_EVENT_EOS: - gst_element_set_eos (element); - gst_event_unref (event); - return; - default: - gst_pad_event_default (pad->pad, event); - return; - } - } - - /* if the other plugins only give out buffer-frames or less (as - they should), if the length of the GstBuffer is different - from nframes then the buffer is short and we will get EOS - next */ - memcpy (pad->data, GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer)); - if (len != GST_BUFFER_SIZE (buffer)) - memset (pad->data + GST_BUFFER_SIZE (buffer), 0, - len - GST_BUFFER_SIZE (buffer)); - - gst_buffer_unref (buffer); - } else { - buffer = gst_buffer_new (); - gst_buffer_set_data (buffer, pad->data, len); - GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_DONTFREE); - - gst_pad_push (pad->pad, GST_DATA (buffer)); - } - pads = g_list_next (pads); - } -} +#include "gstjackaudiosink.h" static gboolean plugin_init (GstPlugin * plugin) { - if (!gst_element_register (plugin, "jackbin", GST_RANK_NONE, - GST_TYPE_JACK_BIN)) - return FALSE; - - if (!gst_element_register (plugin, "jacksrc", GST_RANK_NONE, - GST_TYPE_JACK_SRC)) - return FALSE; - - if (!gst_element_register (plugin, "jacksink", GST_RANK_NONE, - GST_TYPE_JACK_SINK)) + if (!gst_element_register (plugin, "jackaudiosink", GST_RANK_PRIMARY, + GST_TYPE_JACK_AUDIO_SINK)) return FALSE; return TRUE; @@ -531,5 +36,6 @@ plugin_init (GstPlugin * plugin) GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, "jack", - "Jack Plugin Library", plugin_init, VERSION, "GPL", GST_PACKAGE_NAME, - GST_PACKAGE_ORIGIN) + "Jack elements", + plugin_init, + VERSION, "LGPL", "Gstreamer", "http://gstreamer.freedesktop.org") diff --git a/ext/jack/gstjack.h b/ext/jack/gstjack.h deleted file mode 100644 index 738168ca7e..0000000000 --- a/ext/jack/gstjack.h +++ /dev/null @@ -1,129 +0,0 @@ -/* -*- Mode: C; c-basic-offset: 4 -*- */ -/* - Copyright (C) 2002 Andy Wingo - - This library is free software; you can redistribute it and/or - modify it under the terms of the GNU Library General Public - License as published by the Free Software Foundation; either - version 2 of the License, or (at your option) any later version. - - This library is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the implied warranty of - MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - Library General Public License for more details. - - You should have received a copy of the GNU Library General Public - License along with this library; if not, write to the Free - Software Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. -*/ - -#ifndef __GST_JACK_H__ -#define __GST_JACK_H__ - -#include -#include -#include - -//#define JACK_DEBUG(str, a...) g_message (str, ##a) -#define JACK_DEBUG(str, a...) - -#define GST_JACK(obj) G_TYPE_CHECK_INSTANCE_CAST(obj, GST_TYPE_JACK, GstJack) -#define GST_JACK_CLASS(klass) G_TYPE_CHECK_CLASS_CAST(klass, GST_TYPE_JACK, GstJackClass) -#define GST_IS_JACK(obj) G_TYPE_CHECK_INSTANCE_TYPE(obj, GST_TYPE_JACK) -#define GST_IS_JACK_CLASS(klass) G_TYPE_CHECK_CLASS_TYPE(klass, GST_TYPE_JACK) -#define GST_TYPE_JACK gst_jack_get_type() - -#define GST_JACK_SINK(obj) G_TYPE_CHECK_INSTANCE_CAST(obj, GST_TYPE_JACK_SINK, GstJack) -#define GST_JACK_SINK_CLASS(klass) G_TYPE_CHECK_CLASS_CAST(klass, GST_TYPE_JACK_SINK, GstJackClass) -#define GST_IS_JACK_SINK(obj) G_TYPE_CHECK_INSTANCE_TYPE(obj, GST_TYPE_JACK_SINK) -#define GST_IS_JACK_SINK_CLASS(klass) G_TYPE_CHECK_CLASS_TYPE(klass, GST_TYPE_JACK_SINK) -#define GST_TYPE_JACK_SINK gst_jack_sink_get_type() - -#define GST_JACK_SRC(obj) G_TYPE_CHECK_INSTANCE_CAST(obj, GST_TYPE_JACK_SRC, GstJack) -#define GST_JACK_SRC_CLASS(klass) G_TYPE_CHECK_CLASS_CAST(klass, GST_TYPE_JACK_SRC, GstJackClass) -#define GST_IS_JACK_SRC(obj) G_TYPE_CHECK_INSTANCE_TYPE(obj, GST_TYPE_JACK_SRC) -#define GST_IS_JACK_SRC_CLASS(klass) G_TYPE_CHECK_CLASS_TYPE(klass, GST_TYPE_JACK_SRC) -#define GST_TYPE_JACK_SRC gst_jack_src_get_type() - -#define GST_JACK_BIN(obj) G_TYPE_CHECK_INSTANCE_CAST(obj, GST_TYPE_JACK_BIN, GstJackBin) -#define GST_JACK_BIN_CLASS(klass) G_TYPE_CHECK_CLASS_CAST(klass, GST_TYPE_JACK_BIN, GstJackClass) -#define GST_IS_JACK_BIN(obj) G_TYPE_CHECK_INSTANCE_TYPE(obj, GST_TYPE_JACK_BIN) -#define GST_IS_JACK_BIN_CLASS(klass) G_TYPE_CHECK_CLASS_TYPE(klass, GST_TYPE_JACK_BIN) -#define GST_TYPE_JACK_BIN gst_jack_bin_get_type() - -#define GST_JACK_PAD(l) ((GstJackPad*)l->data) /* l is a GList */ - - -typedef struct _GstJack GstJack; -typedef struct _GstJackClass GstJackClass; -typedef struct _GstJackBin GstJackBin; -typedef struct _GstJackBinClass GstJackBinClass; -typedef GstJack GstJackSink; -typedef GstJackClass GstJackSinkClass; -typedef GstJack GstJackSrc; -typedef GstJackClass GstJackSrcClass; - - -enum { - GST_JACK_OPEN = (GST_BIN_FLAG_LAST << 0), - GST_JACK_ACTIVE = (GST_BIN_FLAG_LAST << 1), - GST_JACK_FLAG_LAST = (GST_BIN_FLAG_LAST << 3) -}; - - -typedef jack_default_audio_sample_t sample_t; - -typedef struct { - GstPad *pad; - void *data; - const gchar *name; - const gchar *peer_name; - jack_port_t *port; -} GstJackPad; - -struct _GstJack { - GstElement element; - - /* list of GstJackPads */ - GList *pads; - - /* for convenience */ - GstPadDirection direction; - - gchar *port_name_prefix; - - GstJackBin *bin; -}; - -struct _GstJackClass { - GstElementClass parent_class; -}; - -struct _GstJackBin { - GstBin bin; - - jack_client_t *client; - gint default_new_port_number; - - /* lists of GstJackPads */ - GList *sink_pads; - GList *src_pads; - - gchar *client_name; - - guint rate; - jack_nframes_t nframes; -}; - -struct _GstJackBinClass { - GstBinClass parent_class; -}; - - -GType gst_jack_get_type (void); -GType gst_jack_bin_get_type (void); -GType gst_jack_sink_get_type (void); -GType gst_jack_src_get_type (void); - - -#endif /* __GST_JACK_H__ */ diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c new file mode 100644 index 0000000000..5a0cadf8be --- /dev/null +++ b/ext/jack/gstjackaudiosink.c @@ -0,0 +1,846 @@ +/* GStreamer + * Copyright (C) 2006 Wim Taymans + * + * gstjackaudiosink.c: jack audio sink implementation + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +/** + * SECTION:gstjacksink + * @short_description: JACK audio sink + * @see_also: #GstBaseAudioSink, #GstRingBuffer + * + * A Sink that outputs data to Jack ports. + * + * It will create N Jack ports named out_ where is starting from 1. + * Each port corresponds to a gstreamer channel. + * + * The samplerate as exposed on the caps is always the same as the samplerate of + * the jack server. + * + * When the ::connect property is set to auto, this element will try to connect + * each output port to a random physical jack input pin. In this mode, the sink + * will expose the number of physical channels on its pad caps. + * + * When the ::connect property is set to none, the element will accept any + * number of input channels and will create (but not connect) an output port for + * each channel. + * + * The element will generate an error when the Jack server is shut down when it + * was PAUSED or PLAYING. This element does not support dynamic rate and buffer + * size changes at runtime. + * + * Last reviewed on 2006-11-30 (0.10.4) + */ +#include + +#include "gstjackaudiosink.h" + +GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_sink_debug); +#define GST_CAT_DEFAULT gst_jack_audio_sink_debug + +typedef jack_default_audio_sample_t sample_t; + +#define GST_TYPE_JACK_RING_BUFFER \ + (gst_jack_ring_buffer_get_type()) +#define GST_JACK_RING_BUFFER(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_JACK_RING_BUFFER,GstJackRingBuffer)) +#define GST_JACK_RING_BUFFER_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_JACK_RING_BUFFER,GstJackRingBufferClass)) +#define GST_JACK_RING_BUFFER_GET_CLASS(obj) \ + (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_JACK_RING_BUFFER, GstJackRingBufferClass)) +#define GST_JACK_RING_BUFFER_CAST(obj) \ + ((GstJackRingBuffer *)obj) +#define GST_IS_JACK_RING_BUFFER(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_JACK_RING_BUFFER)) +#define GST_IS_JACK_RING_BUFFER_CLASS(klass)\ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_JACK_RING_BUFFER)) + +typedef struct _GstJackRingBuffer GstJackRingBuffer; +typedef struct _GstJackRingBufferClass GstJackRingBufferClass; + +struct _GstJackRingBuffer +{ + GstRingBuffer object; + + gint sample_rate; + gint buffer_size; + gint channels; + + jack_port_t **outport; +}; + +struct _GstJackRingBufferClass +{ + GstRingBufferClass parent_class; +}; + +static void gst_jack_ring_buffer_class_init (GstJackRingBufferClass * klass); +static void gst_jack_ring_buffer_init (GstJackRingBuffer * ringbuffer, + GstJackRingBufferClass * klass); +static void gst_jack_ring_buffer_dispose (GObject * object); +static void gst_jack_ring_buffer_finalize (GObject * object); + +static GstRingBufferClass *ring_parent_class = NULL; + +static gboolean gst_jack_ring_buffer_open_device (GstRingBuffer * buf); +static gboolean gst_jack_ring_buffer_close_device (GstRingBuffer * buf); +static gboolean gst_jack_ring_buffer_acquire (GstRingBuffer * buf, + GstRingBufferSpec * spec); +static gboolean gst_jack_ring_buffer_release (GstRingBuffer * buf); +static gboolean gst_jack_ring_buffer_start (GstRingBuffer * buf); +static gboolean gst_jack_ring_buffer_pause (GstRingBuffer * buf); +static gboolean gst_jack_ring_buffer_stop (GstRingBuffer * buf); +static guint gst_jack_ring_buffer_delay (GstRingBuffer * buf); + +/* ringbuffer abstract base class */ +static GType +gst_jack_ring_buffer_get_type (void) +{ + static GType ringbuffer_type = 0; + + if (!ringbuffer_type) { + static const GTypeInfo ringbuffer_info = { + sizeof (GstJackRingBufferClass), + NULL, + NULL, + (GClassInitFunc) gst_jack_ring_buffer_class_init, + NULL, + NULL, + sizeof (GstJackRingBuffer), + 0, + (GInstanceInitFunc) gst_jack_ring_buffer_init, + NULL + }; + + ringbuffer_type = + g_type_register_static (GST_TYPE_RING_BUFFER, + "GstJackAudioSinkRingBuffer", &ringbuffer_info, 0); + } + return ringbuffer_type; +} + +static void +gst_jack_ring_buffer_class_init (GstJackRingBufferClass * klass) +{ + GObjectClass *gobject_class; + GstObjectClass *gstobject_class; + GstRingBufferClass *gstringbuffer_class; + + gobject_class = (GObjectClass *) klass; + gstobject_class = (GstObjectClass *) klass; + gstringbuffer_class = (GstRingBufferClass *) klass; + + ring_parent_class = g_type_class_peek_parent (klass); + + gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_dispose); + gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_finalize); + + gstringbuffer_class->open_device = + GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_open_device); + gstringbuffer_class->close_device = + GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_close_device); + gstringbuffer_class->acquire = + GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_acquire); + gstringbuffer_class->release = + GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_release); + gstringbuffer_class->start = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_start); + gstringbuffer_class->pause = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_pause); + gstringbuffer_class->resume = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_start); + gstringbuffer_class->stop = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_stop); + + gstringbuffer_class->delay = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_delay); +} + +/* this is the callback of jack. This should RT-safe. + */ +static int +jack_process_cb (jack_nframes_t nframes, void *arg) +{ + GstJackAudioSink *sink; + GstRingBuffer *buf; + GstJackRingBuffer *abuf; + gint readseg, len; + guint8 *readptr; + gint i, j, flen, channels; + sample_t **buffers, *data; + + buf = GST_RING_BUFFER_CAST (arg); + abuf = GST_JACK_RING_BUFFER_CAST (arg); + sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf)); + + channels = buf->spec.channels; + + /* alloc pointers to samples */ + buffers = g_alloca (sizeof (sample_t *) * channels); + + /* get target buffers */ + for (i = 0; i < channels; i++) { + buffers[i] = (sample_t *) jack_port_get_buffer (abuf->outport[i], nframes); + } + + if (gst_ring_buffer_prepare_read (buf, &readseg, &readptr, &len)) { + flen = len / channels; + + if (nframes * sizeof (sample_t) != flen) + goto wrong_size; + + /* copy samples */ + GST_DEBUG ("copy %d frames: %p, %d bytes, %d channels", nframes, readptr, + flen, channels); + data = (sample_t *) readptr; + + /* copy and interleave into target buffers */ + for (i = 0; i < nframes; i++) { + for (j = 0; j < channels; j++) { + buffers[j][i] = *data++; + } + } + + /* clear written samples */ + gst_ring_buffer_clear (buf, readseg); + + /* we wrote one segment */ + gst_ring_buffer_advance (buf, 1); + } else { + /* write silence to all buffers */ + for (i = 0; i < channels; i++) { + memset (buffers[i], 0, nframes * sizeof (sample_t)); + } + } + return 0; + + /* ERRORS */ +wrong_size: + { + GST_ERROR_OBJECT (sink, "nbytes (%d) != flen (%d)", + nframes * sizeof (sample_t), flen); + return 1; + } +} + +/* we error out */ +static int +jack_sample_rate_cb (jack_nframes_t nframes, void *arg) +{ + GstJackAudioSink *sink; + GstJackRingBuffer *abuf; + + abuf = GST_JACK_RING_BUFFER_CAST (arg); + sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (arg)); + + if (abuf->sample_rate != -1 && abuf->sample_rate != nframes) + goto not_supported; + + return 0; + + /* ERRORS */ +not_supported: + { + GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, + (NULL), ("Jack changed the sample rate, which is not supported")); + return 1; + } +} + +/* we error out */ +static int +jack_buffer_size_cb (jack_nframes_t nframes, void *arg) +{ + GstJackAudioSink *sink; + GstJackRingBuffer *abuf; + + abuf = GST_JACK_RING_BUFFER_CAST (arg); + sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (arg)); + + if (abuf->buffer_size != -1 && abuf->buffer_size != nframes) + goto not_supported; + + return 0; + + /* ERRORS */ +not_supported: + { + GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, + (NULL), ("Jack changed the buffer size, which is not supported")); + return 1; + } +} + +static void +jack_shutdown_cb (void *arg) +{ + GstJackAudioSink *sink; + + sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (arg)); + + GST_DEBUG_OBJECT (sink, "shutdown"); + + GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND, + (NULL), ("Jack server shutdown")); +} + +static void +gst_jack_ring_buffer_init (GstJackRingBuffer * buf, + GstJackRingBufferClass * g_class) +{ + buf->channels = -1; + buf->buffer_size = -1; + buf->sample_rate = -1; +} + +static void +gst_jack_ring_buffer_dispose (GObject * object) +{ + G_OBJECT_CLASS (ring_parent_class)->dispose (object); +} + +static void +gst_jack_ring_buffer_finalize (GObject * object) +{ + GstJackRingBuffer *ringbuffer; + + ringbuffer = GST_JACK_RING_BUFFER_CAST (object); + + G_OBJECT_CLASS (ring_parent_class)->finalize (object); +} + +/* the _open_device method should make a connection with the server + */ +static gboolean +gst_jack_ring_buffer_open_device (GstRingBuffer * buf) +{ + GstJackAudioSink *sink; + jack_options_t options; + jack_status_t status = 0; + + sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf)); + + GST_DEBUG_OBJECT (sink, "open"); + + /* never start a server */ + options = JackNoStartServer; + /* if we have a servername, use it */ + if (sink->server != NULL) + options |= JackServerName; + /* open the client */ + sink->client = jack_client_open ("GStreamer", options, &status, sink->server); + if (sink->client == NULL) + goto could_not_open; + + /* set our callbacks */ + jack_set_process_callback (sink->client, jack_process_cb, buf); + /* these callbacks cause us to error */ + jack_set_buffer_size_callback (sink->client, jack_buffer_size_cb, buf); + jack_set_sample_rate_callback (sink->client, jack_sample_rate_cb, buf); + jack_on_shutdown (sink->client, jack_shutdown_cb, buf); + + GST_DEBUG_OBJECT (sink, "opened"); + + return TRUE; + + /* ERRORS */ +could_not_open: + { + if (status & JackServerFailed) { + GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND, + (NULL), ("Cannot connect to the Jack server (status %d)", status)); + } else { + GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE, + (NULL), ("Jack client open error (status %d)", status)); + } + return FALSE; + } +} + +/* close the connection with the server + */ +static gboolean +gst_jack_ring_buffer_close_device (GstRingBuffer * buf) +{ + GstJackAudioSink *sink; + gint res; + + sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf)); + + GST_DEBUG_OBJECT (sink, "close"); + + if ((res = jack_client_close (sink->client))) { + /* just a warning, we assume the client is gone. */ + GST_ELEMENT_WARNING (sink, RESOURCE, CLOSE, + (NULL), ("Jack client close error (%d)", res)); + } + sink->client = NULL; + + return TRUE; +} + +/* allocate a buffer and setup resources to process the audio samples of + * the format as specified in @spec. + * + * We allocate N jack ports for each channel. If we are asked to automatically + * make a connection with physical ports, we connect as many ports as there are + * physical ports, leaving leftover ports unconnected. + * + * It is assumed that samplerate and number of channels are acceptable since our + * getcaps method will always provide correct values. If unacceptable caps are + * received for some reason, we fail here. + */ +static gboolean +gst_jack_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec) +{ + GstJackAudioSink *sink; + GstJackRingBuffer *abuf; + const char **ports; + gint sample_rate, buffer_size; + gint i, channels, res; + + sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf)); + abuf = GST_JACK_RING_BUFFER_CAST (buf); + + GST_DEBUG_OBJECT (sink, "acquire"); + + /* sample rate must be that of the server */ + sample_rate = jack_get_sample_rate (sink->client); + if (sample_rate != spec->rate) + goto wrong_samplerate; + + channels = spec->channels; + + /* alloc enough output ports */ + abuf->outport = g_new (jack_port_t *, channels); + + /* create an output port for each channel */ + for (i = 0; i < channels; i++) { + gchar *name; + + /* port names start from 1 */ + name = g_strdup_printf ("out_%d", i + 1); + abuf->outport[i] = jack_port_register (sink->client, name, + JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0); + if (abuf->outport[i] == NULL) + goto out_of_ports; + + g_free (name); + } + + buffer_size = jack_get_buffer_size (sink->client); + + /* the segment size in bytes, this is large enough to hold a buffer of 32bit floats + * for all channels */ + spec->segsize = buffer_size * sizeof (gfloat) * channels; + spec->latency_time = gst_util_uint64_scale (spec->segsize, + (GST_SECOND / GST_USECOND), spec->rate * spec->bytes_per_sample); + /* segtotal based on buffer-time latency */ + spec->segtotal = spec->buffer_time / spec->latency_time; + + GST_DEBUG_OBJECT (sink, "segsize %d, segtotal %d", spec->segsize, + spec->segtotal); + + /* allocate the ringbuffer memory now */ + buf->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize); + memset (GST_BUFFER_DATA (buf->data), 0, GST_BUFFER_SIZE (buf->data)); + + if ((res = jack_activate (sink->client))) + goto could_not_activate; + + /* if we need to automatically connect the ports, do so now. We must do this + * after activating the client. */ + if (sink->connect == GST_JACK_CONNECT_AUTO) { + /* find all the physical input ports. A physical input port is a port + * associated with a hardware device. Someone needs connect to a physical + * port in order to hear something. */ + ports = jack_get_ports (sink->client, NULL, NULL, + JackPortIsPhysical | JackPortIsInput); + if (ports == NULL) { + /* no ports? fine then we don't do anything except for posting a warning + * message. */ + GST_ELEMENT_WARNING (sink, RESOURCE, NOT_FOUND, (NULL), + ("No physical input ports found, leaving ports unconnected")); + goto done; + } + + for (i = 0; i < channels; i++) { + /* stop when all input ports are exhausted */ + if (ports[i] == NULL) { + /* post a warning that we could not connect all ports */ + GST_ELEMENT_WARNING (sink, RESOURCE, NOT_FOUND, (NULL), + ("No more physical ports, leaving some ports unconnected")); + break; + } + /* connect the port to a physical port */ + if ((res = jack_connect (sink->client, jack_port_name (abuf->outport[i]), + ports[i]))) + goto cannot_connect; + } + free (ports); + } +done: + + abuf->sample_rate = sample_rate; + abuf->buffer_size = buffer_size; + abuf->channels = spec->channels; + + return TRUE; + + /* ERRORS */ +wrong_samplerate: + { + GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL), + ("Wrong samplerate, server is running at %d and we received %d", + sample_rate, spec->rate)); + return FALSE; + } +out_of_ports: + { + GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL), + ("Cannot allocate more Jack ports")); + return FALSE; + } +could_not_activate: + { + GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL), + ("Could not activate client (%d)", res)); + return FALSE; + } +cannot_connect: + { + GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL), + ("Could not connect output ports to physical ports (%d)", res)); + free (ports); + return FALSE; + } +} + +/* function is called with LOCK */ +static gboolean +gst_jack_ring_buffer_release (GstRingBuffer * buf) +{ + GstJackAudioSink *sink; + GstJackRingBuffer *abuf; + gint i, res; + + abuf = GST_JACK_RING_BUFFER_CAST (buf); + sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf)); + + GST_DEBUG_OBJECT (sink, "release"); + + if ((res = jack_deactivate (sink->client))) { + /* we only warn, this means the server is probably shut down and the client + * is gone anyway. */ + GST_ELEMENT_WARNING (sink, RESOURCE, CLOSE, (NULL), + ("Could not deactivate Jack client (%d)", res)); + } + + /* remove all ports */ + for (i = 0; i < abuf->channels; i++) { + GST_LOG_OBJECT (sink, "unregister port %d", i); + if ((res = jack_port_unregister (sink->client, abuf->outport[i]))) { + GST_DEBUG_OBJECT (sink, "unregister of port failed (%d)", res); + } + abuf->outport[i] = NULL; + } + g_free (abuf->outport); + abuf->outport = NULL; + abuf->channels = -1; + abuf->buffer_size = -1; + abuf->sample_rate = -1; + + /* free the buffer */ + gst_buffer_unref (buf->data); + buf->data = NULL; + + return TRUE; +} + +static gboolean +gst_jack_ring_buffer_start (GstRingBuffer * buf) +{ + GstJackAudioSink *sink; + + sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf)); + + GST_DEBUG_OBJECT (sink, "start"); + + return TRUE; +} + +static gboolean +gst_jack_ring_buffer_pause (GstRingBuffer * buf) +{ + GstJackAudioSink *sink; + + sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf)); + + GST_DEBUG_OBJECT (sink, "pause"); + + return TRUE; +} + +static gboolean +gst_jack_ring_buffer_stop (GstRingBuffer * buf) +{ + GstJackAudioSink *sink; + + sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf)); + + GST_DEBUG_OBJECT (sink, "stop"); + + return TRUE; +} + +static guint +gst_jack_ring_buffer_delay (GstRingBuffer * buf) +{ + GstJackAudioSink *sink; + guint res = 0; + + sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf)); + + GST_DEBUG_OBJECT (sink, "delay %u", res); + + return res; +} + +/* elementfactory information */ +static const GstElementDetails gst_jack_audio_sink_details = +GST_ELEMENT_DETAILS ("Audio Sink (Jack)", + "Sink/Audio", + "Output to Jack", + "Wim Taymans "); + +static GstStaticPadTemplate jackaudiosink_sink_factory = +GST_STATIC_PAD_TEMPLATE ("sink", + GST_PAD_SINK, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/x-raw-float, " + "endianness = (int) { " G_STRINGIFY (G_BYTE_ORDER) " }, " + "width = (int) 32, " + "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]") + ); + +/* AudioSink signals and args */ +enum +{ + /* FILL ME */ + SIGNAL_LAST +}; + +#define DEFAULT_PROP_CONNECT GST_JACK_CONNECT_AUTO +#define DEFAULT_PROP_SERVER NULL + +enum +{ + PROP_0, + PROP_CONNECT, + PROP_SERVER, + PROP_LAST +}; + +#define GST_TYPE_JACK_CONNECT (gst_jack_connect_get_type()) +static GType +gst_jack_connect_get_type (void) +{ + static GType jack_connect_type = 0; + static const GEnumValue jack_connect[] = { + {GST_JACK_CONNECT_NONE, + "Don't automatically connect ports to physical ports", "none"}, + {GST_JACK_CONNECT_AUTO, + "Automatically connect ports to physical ports", "auto"}, + {0, NULL, NULL}, + }; + + if (!jack_connect_type) { + jack_connect_type = g_enum_register_static ("GstJackConnect", jack_connect); + } + return jack_connect_type; +} + +#define _do_init(bla) \ + GST_DEBUG_CATEGORY_INIT (gst_jack_audio_sink_debug, "jacksink", 0, "jacksink element"); + +GST_BOILERPLATE_FULL (GstJackAudioSink, gst_jack_audio_sink, GstBaseAudioSink, + GST_TYPE_BASE_AUDIO_SINK, _do_init); + +static void gst_jack_audio_sink_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec); +static void gst_jack_audio_sink_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec); + +static GstCaps *gst_jack_audio_sink_getcaps (GstBaseSink * bsink); +static GstRingBuffer *gst_jack_audio_sink_create_ringbuffer (GstBaseAudioSink * + sink); + +static void +gst_jack_audio_sink_base_init (gpointer g_class) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); + + gst_element_class_set_details (element_class, &gst_jack_audio_sink_details); + + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&jackaudiosink_sink_factory)); +} + +static void +gst_jack_audio_sink_class_init (GstJackAudioSinkClass * klass) +{ + GObjectClass *gobject_class; + GstElementClass *gstelement_class; + GstBaseSinkClass *gstbasesink_class; + GstBaseAudioSinkClass *gstbaseaudiosink_class; + + gobject_class = (GObjectClass *) klass; + gstelement_class = (GstElementClass *) klass; + gstbasesink_class = (GstBaseSinkClass *) klass; + gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass; + + gobject_class->get_property = + GST_DEBUG_FUNCPTR (gst_jack_audio_sink_get_property); + gobject_class->set_property = + GST_DEBUG_FUNCPTR (gst_jack_audio_sink_set_property); + + g_object_class_install_property (gobject_class, PROP_CONNECT, + g_param_spec_enum ("connect", "Connect", + "Specify how the output ports will be connected", + GST_TYPE_JACK_CONNECT, DEFAULT_PROP_CONNECT, G_PARAM_READWRITE)); + + g_object_class_install_property (gobject_class, PROP_SERVER, + g_param_spec_string ("server", "Server", + "The Jack server to connect to (NULL = default)", + DEFAULT_PROP_SERVER, G_PARAM_READWRITE)); + + gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_jack_audio_sink_getcaps); + + gstbaseaudiosink_class->create_ringbuffer = + GST_DEBUG_FUNCPTR (gst_jack_audio_sink_create_ringbuffer); +} + +static void +gst_jack_audio_sink_init (GstJackAudioSink * sink, + GstJackAudioSinkClass * g_class) +{ + sink->connect = DEFAULT_PROP_CONNECT; + sink->server = g_strdup (DEFAULT_PROP_SERVER); +} + +static void +gst_jack_audio_sink_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec) +{ + GstJackAudioSink *sink; + + sink = GST_JACK_AUDIO_SINK (object); + + switch (prop_id) { + case PROP_CONNECT: + sink->connect = g_value_get_enum (value); + break; + case PROP_SERVER: + g_free (sink->server); + sink->server = g_value_dup_string (value); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static void +gst_jack_audio_sink_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec) +{ + GstJackAudioSink *sink; + + sink = GST_JACK_AUDIO_SINK (object); + + switch (prop_id) { + case PROP_CONNECT: + g_value_set_enum (value, sink->connect); + break; + case PROP_SERVER: + g_value_set_string (value, sink->server); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static GstCaps * +gst_jack_audio_sink_getcaps (GstBaseSink * bsink) +{ + GstJackAudioSink *sink = GST_JACK_AUDIO_SINK (bsink); + const char **ports; + gint min, max; + gint rate; + + if (sink->client == NULL) + goto no_client; + + if (sink->connect == GST_JACK_CONNECT_AUTO) { + /* get a port count, this is the number of channels we can automatically + * connect. */ + ports = jack_get_ports (sink->client, NULL, NULL, + JackPortIsPhysical | JackPortIsInput); + max = 0; + if (ports != NULL) { + for (; ports[max]; max++); + free (ports); + } else + max = 0; + } else { + /* we allow any number of pads, somoething else is going to connect the + * pads. */ + max = G_MAXINT; + } + min = MIN (1, max); + + rate = jack_get_sample_rate (sink->client); + + GST_DEBUG_OBJECT (sink, "got %d-%d ports, samplerate: %d", min, max, rate); + + if (!sink->caps) { + sink->caps = gst_caps_new_simple ("audio/x-raw-float", + "endianness", G_TYPE_INT, G_BYTE_ORDER, + "width", G_TYPE_INT, 32, + "rate", G_TYPE_INT, rate, + "channels", GST_TYPE_INT_RANGE, min, max, NULL); + } + GST_INFO_OBJECT (sink, "returning caps %" GST_PTR_FORMAT, sink->caps); + + return gst_caps_ref (sink->caps); + + /* ERRORS */ +no_client: + { + GST_DEBUG_OBJECT (sink, "device not open, using template caps"); + /* base class will get template caps for us when we return NULL */ + return NULL; + } +} + +static GstRingBuffer * +gst_jack_audio_sink_create_ringbuffer (GstBaseAudioSink * sink) +{ + GstRingBuffer *buffer; + + buffer = g_object_new (GST_TYPE_JACK_RING_BUFFER, NULL); + GST_DEBUG_OBJECT (sink, "created ringbuffer @%p", buffer); + + return buffer; +} diff --git a/ext/jack/gstjackaudiosink.h b/ext/jack/gstjackaudiosink.h new file mode 100644 index 0000000000..218e9de0c4 --- /dev/null +++ b/ext/jack/gstjackaudiosink.h @@ -0,0 +1,80 @@ +/* GStreamer + * Copyright (C) 2006 Wim Taymans + * + * gstjacksink.h: + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifndef __GST_JACK_AUDIO_SINK_H__ +#define __GST_JACK_AUDIO_SINK_H__ + +#include + +#include +#include + +G_BEGIN_DECLS + +#define GST_TYPE_JACK_AUDIO_SINK (gst_jack_audio_sink_get_type()) +#define GST_JACK_AUDIO_SINK(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_JACK_AUDIO_SINK,GstJackAudioSink)) +#define GST_JACK_AUDIO_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_JACK_AUDIO_SINK,GstJackAudioSinkClass)) +#define GST_JACK_AUDIO_SINK_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_JACK_AUDIO_SINK,GstJackAudioSinkClass)) +#define GST_IS_JACK_AUDIO_SINK(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_JACK_AUDIO_SINK)) +#define GST_IS_JACK_AUDIO_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_JACK_AUDIO_SINK)) + +typedef struct _GstJackAudioSink GstJackAudioSink; +typedef struct _GstJackAudioSinkClass GstJackAudioSinkClass; + +typedef enum { + GST_JACK_CONNECT_NONE, + GST_JACK_CONNECT_AUTO +} GstJackConnect; + +/** + * GstJackAudioSink: + * + * Opaque #GstJackAudioSink. + */ +struct _GstJackAudioSink { + GstBaseAudioSink element; + + /* cached caps */ + GstCaps *caps; + + /* properties */ + GstJackConnect connect; + gchar *server; + + /* our client */ + jack_client_t *client; + + /*< private >*/ + gpointer _gst_reserved[GST_PADDING]; +}; + +struct _GstJackAudioSinkClass { + GstBaseAudioSinkClass parent_class; + + /*< private >*/ + gpointer _gst_reserved[GST_PADDING]; +}; + +GType gst_jack_audio_sink_get_type (void); + +G_END_DECLS + +#endif /* __GST_JACK_AUDIO_SINK_H__ */ From e2546f8275307662723114359907ba32b22a783d Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Thu, 30 Nov 2006 11:59:04 +0000 Subject: [PATCH 167/351] ext/Makefile.am: Fix build. Original commit message from CVS: * ext/Makefile.am: Fix build. * ext/jack/gstjackaudiosink.c: (jack_process_cb), (jack_sample_rate_cb), (jack_buffer_size_cb), (jack_shutdown_cb), (gst_jack_ring_buffer_acquire): Small cleanups. --- ext/jack/gstjackaudiosink.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c index 5a0cadf8be..c2c5e65671 100644 --- a/ext/jack/gstjackaudiosink.c +++ b/ext/jack/gstjackaudiosink.c @@ -196,28 +196,30 @@ jack_process_cb (jack_nframes_t nframes, void *arg) if (gst_ring_buffer_prepare_read (buf, &readseg, &readptr, &len)) { flen = len / channels; + /* the number of samples must be exactly the segment size */ if (nframes * sizeof (sample_t) != flen) goto wrong_size; - /* copy samples */ GST_DEBUG ("copy %d frames: %p, %d bytes, %d channels", nframes, readptr, flen, channels); data = (sample_t *) readptr; - /* copy and interleave into target buffers */ + /* the samples in the ringbuffer have the channels interleaved, we need to + * deinterleave into the jack target buffers */ for (i = 0; i < nframes; i++) { for (j = 0; j < channels; j++) { buffers[j][i] = *data++; } } - /* clear written samples */ + /* clear written samples in the ringbuffer */ gst_ring_buffer_clear (buf, readseg); /* we wrote one segment */ gst_ring_buffer_advance (buf, 1); } else { - /* write silence to all buffers */ + /* We are not allowed to read from the ringbuffer, write silence to all + * jack output buffers */ for (i = 0; i < channels; i++) { memset (buffers[i], 0, nframes * sizeof (sample_t)); } From 8beff1f58c8c3eadf385e1f9b00f0d37237303be Mon Sep 17 00:00:00 2001 From: Jan Schmidt Date: Wed, 6 Dec 2006 16:57:17 +0000 Subject: [PATCH 168/351] ext/jack/.cvsignore: Ignore old files as requested by the build slave. Original commit message from CVS: * ext/jack/.cvsignore: Ignore old files as requested by the build slave. --- ext/jack/.gitignore | 1 + 1 file changed, 1 insertion(+) create mode 100644 ext/jack/.gitignore diff --git a/ext/jack/.gitignore b/ext/jack/.gitignore new file mode 100644 index 0000000000..916f265c7a --- /dev/null +++ b/ext/jack/.gitignore @@ -0,0 +1 @@ +*.loT From b0bfe6fcdde7da3947500105c1c2e381fad0ec8d Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Fri, 12 Jan 2007 10:25:40 +0000 Subject: [PATCH 169/351] ext/jack/gstjackaudiosink.*: Improve docs. Original commit message from CVS: * ext/jack/gstjackaudiosink.c: (jack_sample_rate_cb), (jack_buffer_size_cb), (jack_shutdown_cb), (gst_jack_ring_buffer_acquire): * ext/jack/gstjackaudiosink.h: Improve docs. --- ext/jack/gstjackaudiosink.c | 36 ++++++++++++++++++++++++++---------- ext/jack/gstjackaudiosink.h | 11 +++++++++++ 2 files changed, 37 insertions(+), 10 deletions(-) diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c index c2c5e65671..b57e51cf09 100644 --- a/ext/jack/gstjackaudiosink.c +++ b/ext/jack/gstjackaudiosink.c @@ -20,29 +20,45 @@ */ /** - * SECTION:gstjacksink + * SECTION:element-jackaudiosink * @short_description: JACK audio sink * @see_also: #GstBaseAudioSink, #GstRingBuffer * + * + * * A Sink that outputs data to Jack ports. - * - * It will create N Jack ports named out_ where is starting from 1. + * + * + * It will create N Jack ports named out_<num> where <num> is starting from 1. * Each port corresponds to a gstreamer channel. - * + * + * * The samplerate as exposed on the caps is always the same as the samplerate of * the jack server. - * + * + * * When the ::connect property is set to auto, this element will try to connect * each output port to a random physical jack input pin. In this mode, the sink * will expose the number of physical channels on its pad caps. - * + * + * * When the ::connect property is set to none, the element will accept any * number of input channels and will create (but not connect) an output port for * each channel. - * + * + * * The element will generate an error when the Jack server is shut down when it * was PAUSED or PLAYING. This element does not support dynamic rate and buffer * size changes at runtime. + * + * Example launch line + * + * + * gst-launch audiotestsrc ! jackaudiosink + * + * Play a sine wave to using jack. + * + * * * Last reviewed on 2006-11-30 (0.10.4) */ @@ -394,9 +410,9 @@ gst_jack_ring_buffer_close_device (GstRingBuffer * buf) /* allocate a buffer and setup resources to process the audio samples of * the format as specified in @spec. * - * We allocate N jack ports for each channel. If we are asked to automatically - * make a connection with physical ports, we connect as many ports as there are - * physical ports, leaving leftover ports unconnected. + * We allocate N jack ports, one for each channel. If we are asked to + * automatically make a connection with physical ports, we connect as many + * ports as there are physical ports, leaving leftover ports unconnected. * * It is assumed that samplerate and number of channels are acceptable since our * getcaps method will always provide correct values. If unacceptable caps are diff --git a/ext/jack/gstjackaudiosink.h b/ext/jack/gstjackaudiosink.h index 218e9de0c4..245521b2a1 100644 --- a/ext/jack/gstjackaudiosink.h +++ b/ext/jack/gstjackaudiosink.h @@ -39,6 +39,17 @@ G_BEGIN_DECLS typedef struct _GstJackAudioSink GstJackAudioSink; typedef struct _GstJackAudioSinkClass GstJackAudioSinkClass; +/** + * GstJackConnect: + * @GST_JACK_CONNECT_NONE: Don't automatically connect to physical ports. + * In this mode, the element will accept any number of input channels and will + * create (but not connect) an output port for each channel. + * @GST_JACK_CONNECT_AUTO: In this mode, the element will try to connect each + * output port to a random physical jack input pin. The sink will + * expose the number of physical channels on its pad caps. + * + * Specify how the output ports will be connected. + */ typedef enum { GST_JACK_CONNECT_NONE, GST_JACK_CONNECT_AUTO From 60bcffa5ef21831380190a125bdc1a15c0927996 Mon Sep 17 00:00:00 2001 From: Paul Davis Date: Thu, 8 Mar 2007 15:24:52 +0000 Subject: [PATCH 170/351] ext/jack/: Make an object to manage client connections to the jack server which we will use in the future to run sele... Original commit message from CVS: Includes patch by: Paul Davis * ext/jack/Makefile.am: * ext/jack/gstjackaudioclient.c: (gst_jack_audio_client_init), (jack_process_cb), (jack_sample_rate_cb), (jack_buffer_size_cb), (jack_shutdown_cb), (connection_find), (gst_jack_audio_make_connection), (gst_jack_audio_get_connection), (gst_jack_audio_unref_connection), (gst_jack_audio_connection_add_client), (gst_jack_audio_connection_remove_client), (gst_jack_audio_client_new), (gst_jack_audio_client_free), (gst_jack_audio_client_get_client), (gst_jack_audio_client_set_active): * ext/jack/gstjackaudioclient.h: Make an object to manage client connections to the jack server which we will use in the future to run selected jack elements with the same jack connection. Make some stuff a bit more threadsafe. Activate the jack client ASAP. * ext/jack/gstjackaudiosink.c: (gst_jack_audio_sink_allocate_channels), (gst_jack_audio_sink_free_channels), (jack_process_cb), (gst_jack_ring_buffer_open_device), (gst_jack_ring_buffer_close_device), (gst_jack_ring_buffer_acquire), (gst_jack_ring_buffer_release), (gst_jack_audio_sink_class_init), (gst_jack_audio_sink_init), (gst_jack_audio_sink_getcaps): * ext/jack/gstjackaudiosink.h: Use new client object to manage connections. Don't remove and recreate all ports, try to reuse them. --- ext/jack/Makefile.am | 4 +- ext/jack/gstjackaudioclient.c | 484 ++++++++++++++++++++++++++++++++++ ext/jack/gstjackaudioclient.h | 58 ++++ ext/jack/gstjackaudiosink.c | 143 +++++----- ext/jack/gstjackaudiosink.h | 13 +- 5 files changed, 634 insertions(+), 68 deletions(-) create mode 100644 ext/jack/gstjackaudioclient.c create mode 100644 ext/jack/gstjackaudioclient.h diff --git a/ext/jack/Makefile.am b/ext/jack/Makefile.am index 0a237dd6c9..17efdfa94c 100644 --- a/ext/jack/Makefile.am +++ b/ext/jack/Makefile.am @@ -1,11 +1,11 @@ plugin_LTLIBRARIES = libgstjack.la -libgstjack_la_SOURCES = gstjack.c gstjackaudiosink.c +libgstjack_la_SOURCES = gstjack.c gstjackaudiosink.c gstjackaudioclient.c libgstjack_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) $(JACK_CFLAGS) libgstjack_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR) $(JACK_LIBS) libgstjack_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) -noinst_HEADERS = gstjackaudiosink.h +noinst_HEADERS = gstjackaudiosink.h gstjackaudioclient.h EXTRA_DIST = README diff --git a/ext/jack/gstjackaudioclient.c b/ext/jack/gstjackaudioclient.c new file mode 100644 index 0000000000..c81125939d --- /dev/null +++ b/ext/jack/gstjackaudioclient.c @@ -0,0 +1,484 @@ +/* GStreamer + * Copyright (C) 2006 Wim Taymans + * + * gstjackaudioclient.c: jack audio client implementation + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#include + +#include "gstjackaudioclient.h" + +GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_client_debug); +#define GST_CAT_DEFAULT gst_jack_audio_client_debug + +void +gst_jack_audio_client_init (void) +{ + GST_DEBUG_CATEGORY_INIT (gst_jack_audio_client_debug, "jackclient", 0, + "jackclient helpers"); +} + +/* a list of global connections indexed by id and server. */ +G_LOCK_DEFINE_STATIC (connections_lock); +static GList *connections; + +/* the connection to a server */ +typedef struct +{ + gint refcount; + GMutex *lock; + + /* id/server pair and the connection */ + gchar *id; + gchar *server; + jack_client_t *client; + + /* lists of GstJackAudioClients */ + gint n_clients; + GList *src_clients; + GList *sink_clients; +} GstJackAudioConnection; + +/* an object sharing a jack_client_t connection. */ +struct _GstJackAudioClient +{ + GstJackAudioConnection *conn; + + GstJackClientType type; + gboolean active; + + void (*shutdown) (void *arg); + JackProcessCallback process; + JackBufferSizeCallback buffer_size; + JackSampleRateCallback sample_rate; + gpointer user_data; +}; + +typedef jack_default_audio_sample_t sample_t; + +typedef struct +{ + jack_nframes_t nframes; + gpointer user_data; +} JackCB; + +static int +jack_process_cb (jack_nframes_t nframes, void *arg) +{ + GstJackAudioConnection *conn = (GstJackAudioConnection *) arg; + GList *walk; + int res = 0; + + g_mutex_lock (conn->lock); + /* call sources first, then sinks. Sources will either push data into the + * ringbuffer of the sinks, which will then pull the data out of it, or + * sinks will pull the data from the sources. */ + for (walk = conn->src_clients; walk; walk = g_list_next (walk)) { + GstJackAudioClient *client = (GstJackAudioClient *) walk->data; + + /* only call active clients */ + if (client->active && client->process) + res = client->process (nframes, client->user_data); + } + for (walk = conn->sink_clients; walk; walk = g_list_next (walk)) { + GstJackAudioClient *client = (GstJackAudioClient *) walk->data; + + /* only call active clients */ + if (client->active && client->process) + res = client->process (nframes, client->user_data); + } + g_mutex_unlock (conn->lock); + + return res; +} + +/* we error out */ +static int +jack_sample_rate_cb (jack_nframes_t nframes, void *arg) +{ + return 0; +} + +/* we error out */ +static int +jack_buffer_size_cb (jack_nframes_t nframes, void *arg) +{ + return 0; +} + +static void +jack_shutdown_cb (void *arg) +{ + GstJackAudioConnection *conn = (GstJackAudioConnection *) arg; + GList *walk; + + g_mutex_lock (conn->lock); + for (walk = conn->src_clients; walk; walk = g_list_next (walk)) { + GstJackAudioClient *client = (GstJackAudioClient *) walk->data; + + if (client->shutdown) + client->shutdown (client->user_data); + } + for (walk = conn->sink_clients; walk; walk = g_list_next (walk)) { + GstJackAudioClient *client = (GstJackAudioClient *) walk->data; + + if (client->shutdown) + client->shutdown (client->user_data); + } + g_mutex_unlock (conn->lock); +} + +typedef struct +{ + const gchar *id; + const gchar *server; +} FindData; + +static gint +connection_find (GstJackAudioConnection * conn, FindData * data) +{ + /* id's must match */ + if (strcmp (conn->id, data->id)) + return 1; + + /* both the same or NULL */ + if (conn->server == data->server) + return 0; + + /* we cannot compare NULL */ + if (conn->server == NULL || data->server == NULL) + return 1; + + if (strcmp (conn->server, data->server)) + return 1; + + return 0; +} + +/* make a connection with @id and @server. Returns NULL on failure with the + * status set. */ +static GstJackAudioConnection * +gst_jack_audio_make_connection (const gchar * id, const gchar * server, + jack_status_t * status) +{ + GstJackAudioConnection *conn; + jack_options_t options; + jack_client_t *jclient; + gint res; + + *status = 0; + + GST_DEBUG ("new client %s, connecting to server %s", id, + GST_STR_NULL (server)); + + /* never start a server */ + options = JackNoStartServer; + /* if we have a servername, use it */ + if (server != NULL) + options |= JackServerName; + /* open the client */ + jclient = jack_client_open (id, options, status, server); + if (jclient == NULL) + goto could_not_open; + + /* now create object */ + conn = g_new (GstJackAudioConnection, 1); + conn->refcount = 1; + conn->lock = g_mutex_new (); + conn->id = g_strdup (id); + conn->server = g_strdup (server); + conn->client = jclient; + conn->n_clients = 0; + conn->src_clients = NULL; + conn->sink_clients = NULL; + + /* set our callbacks */ + jack_set_process_callback (jclient, jack_process_cb, conn); + /* these callbacks cause us to error */ + jack_set_buffer_size_callback (jclient, jack_buffer_size_cb, conn); + jack_set_sample_rate_callback (jclient, jack_sample_rate_cb, conn); + jack_on_shutdown (jclient, jack_shutdown_cb, conn); + + /* all callbacks are set, activate the client */ + if ((res = jack_activate (jclient))) + goto could_not_activate; + + GST_DEBUG ("opened connection %p", conn); + + return conn; + + /* ERRORS */ +could_not_open: + { + GST_DEBUG ("failed to open jack client, %d", *status); + return NULL; + } +could_not_activate: + { + GST_ERROR ("Could not activate client (%d)", res); + *status = JackFailure; + g_mutex_free (conn->lock); + g_free (conn->id); + g_free (conn->server); + g_free (conn); + return NULL; + } +} + +static GstJackAudioConnection * +gst_jack_audio_get_connection (const gchar * id, const gchar * server, + jack_status_t * status) +{ + GstJackAudioConnection *conn; + GList *found; + FindData data; + + GST_DEBUG ("getting connection for id %s, server %s", id, + GST_STR_NULL (server)); + + data.id = id; + data.server = server; + + G_LOCK (connections_lock); + found = + g_list_find_custom (connections, &data, (GCompareFunc) connection_find); + if (found != NULL) { + /* we found it, increase refcount and return it */ + conn = (GstJackAudioConnection *) found->data; + conn->refcount++; + + GST_DEBUG ("found connection %p", conn); + } else { + /* make new connection */ + conn = gst_jack_audio_make_connection (id, server, status); + if (conn != NULL) { + GST_DEBUG ("created connection %p", conn); + /* add to list on success */ + connections = g_list_prepend (connections, conn); + } else { + GST_WARNING ("could not create connection"); + } + } + G_UNLOCK (connections_lock); + + return conn; +} + +static void +gst_jack_audio_unref_connection (GstJackAudioConnection * conn) +{ + gint res; + + GST_DEBUG ("unref connection %p", conn); + + G_LOCK (connections_lock); + conn->refcount--; + if (conn->refcount == 0) { + GST_DEBUG ("closing connection %p", conn); + /* remove from list */ + connections = g_list_remove (connections, conn); + + /* grab lock to be sure that we are not in one of the callbacks */ + g_mutex_lock (conn->lock); + if ((res = jack_deactivate (conn->client))) { + /* we only warn, this means the server is probably shut down and the client + * is gone anyway. */ + GST_WARNING ("Could not deactivate Jack client (%d)", res); + } + /* close connection */ + if ((res = jack_client_close (conn->client))) { + /* we assume the client is gone. */ + GST_WARNING ("close failed (%d)", res); + } + g_mutex_unlock (conn->lock); + + /* free resources */ + g_mutex_free (conn->lock); + g_free (conn->id); + g_free (conn->server); + g_free (conn); + } + G_UNLOCK (connections_lock); +} + +static void +gst_jack_audio_connection_add_client (GstJackAudioConnection * conn, + GstJackAudioClient * client) +{ + g_mutex_lock (conn->lock); + switch (client->type) { + case GST_JACK_CLIENT_SOURCE: + conn->src_clients = g_list_append (conn->src_clients, client); + conn->n_clients++; + break; + case GST_JACK_CLIENT_SINK: + conn->sink_clients = g_list_append (conn->sink_clients, client); + conn->n_clients++; + break; + default: + g_warning ("trying to add unknown client type"); + break; + } + g_mutex_unlock (conn->lock); +} + +static void +gst_jack_audio_connection_remove_client (GstJackAudioConnection * conn, + GstJackAudioClient * client) +{ + g_mutex_lock (conn->lock); + switch (client->type) { + case GST_JACK_CLIENT_SOURCE: + conn->src_clients = g_list_remove (conn->src_clients, client); + conn->n_clients--; + break; + case GST_JACK_CLIENT_SINK: + conn->sink_clients = g_list_remove (conn->sink_clients, client); + conn->n_clients--; + break; + default: + g_warning ("trying to remove unknown client type"); + break; + } + g_mutex_unlock (conn->lock); +} + +/** + * gst_jack_audio_client_get: + * @id: the client id + * @server: the server to connect to or NULL for the default server + * @type: the client type + * @shutdown: a callback when the jack server shuts down + * @process: a callback when samples are available + * @buffer_size: a callback when the buffer_size changes + * @sample_rate: a callback when the sample_rate changes + * @user_data: user data passed to the callbacks + * @status: pointer to hold the jack status code in case of errors + * + * Get the jack client connection for @id and @server. Connections to the same + * @id and @server will receive the same physical Jack client connection and + * will therefore be scheduled in the same process callback. + * + * Returns: a #GstJackAudioClient. + */ +GstJackAudioClient * +gst_jack_audio_client_new (const gchar * id, const gchar * server, + GstJackClientType type, void (*shutdown) (void *arg), + JackProcessCallback process, JackBufferSizeCallback buffer_size, + JackSampleRateCallback sample_rate, gpointer user_data, + jack_status_t * status) +{ + GstJackAudioClient *client; + GstJackAudioConnection *conn; + + g_return_val_if_fail (id != NULL, NULL); + g_return_val_if_fail (status != NULL, NULL); + + /* first get a connection for the id/server pair */ + conn = gst_jack_audio_get_connection (id, server, status); + if (conn == NULL) + goto no_connection; + + /* make new client using the connection */ + client = g_new (GstJackAudioClient, 1); + client->active = FALSE; + client->conn = conn; + client->type = type; + client->shutdown = shutdown; + client->process = process; + client->buffer_size = buffer_size; + client->sample_rate = sample_rate; + client->user_data = user_data; + + /* add the client to the connection */ + gst_jack_audio_connection_add_client (conn, client); + + return client; + + /* ERRORS */ +no_connection: + { + GST_DEBUG ("Could not get server connection (%d)", *status); + return NULL; + } +} + +/** + * gst_jack_audio_client_free: + * @client: a #GstJackAudioClient + * + * Free the resources used by @client. + */ +void +gst_jack_audio_client_free (GstJackAudioClient * client) +{ + GstJackAudioConnection *conn; + + g_return_if_fail (client != NULL); + + conn = client->conn; + + /* remove from connection first so that it's not scheduled anymore after this + * call */ + gst_jack_audio_connection_remove_client (conn, client); + gst_jack_audio_unref_connection (conn); + + g_free (client); +} + +/** + * gst_jack_audio_client_get_client: + * @client: a #GstJackAudioClient + * + * Get the jack audio client for @client. This function is used to perform + * operations on the jack server from this client. + * + * Returns: The jack audio client. + */ +jack_client_t * +gst_jack_audio_client_get_client (GstJackAudioClient * client) +{ + g_return_val_if_fail (client != NULL, NULL); + + /* no lock needed, the connection and the client does not change + * once the client is created. */ + return client->conn->client; +} + +/** + * gst_jack_audio_client_set_active: + * @client: a #GstJackAudioClient + * @active: new mode for the client + * + * Activate or deactive @client. When a client is activated it will receive + * callbacks when data should be processed. + * + * Returns: 0 if all ok. + */ +gint +gst_jack_audio_client_set_active (GstJackAudioClient * client, gboolean active) +{ + g_return_val_if_fail (client != NULL, -1); + + /* make sure that we are not dispatching the client */ + g_mutex_lock (client->conn->lock); + client->active = active; + g_mutex_unlock (client->conn->lock); + + return 0; +} diff --git a/ext/jack/gstjackaudioclient.h b/ext/jack/gstjackaudioclient.h new file mode 100644 index 0000000000..9092e7ded5 --- /dev/null +++ b/ext/jack/gstjackaudioclient.h @@ -0,0 +1,58 @@ +/* GStreamer + * Copyright (C) 2006 Wim Taymans + * + * gstjackaudioclient.h: + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifndef __GST_JACK_AUDIO_CLIENT_H__ +#define __GST_JACK_AUDIO_CLIENT_H__ + +#include + +#include + +G_BEGIN_DECLS + +typedef enum +{ + GST_JACK_CLIENT_SOURCE, + GST_JACK_CLIENT_SINK +} GstJackClientType; + +typedef struct _GstJackAudioClient GstJackAudioClient; + +void gst_jack_audio_client_init (void); + + +GstJackAudioClient * gst_jack_audio_client_new (const gchar *id, const gchar *server, + GstJackClientType type, + void (*shutdown) (void *arg), + JackProcessCallback process, + JackBufferSizeCallback buffer_size, + JackSampleRateCallback sample_rate, + gpointer user_data, + jack_status_t *status); +void gst_jack_audio_client_free (GstJackAudioClient *client); + +jack_client_t * gst_jack_audio_client_get_client (GstJackAudioClient *client); + +gboolean gst_jack_audio_client_set_active (GstJackAudioClient *client, gboolean active); + +G_END_DECLS + +#endif /* __GST_JACK_AUDIO_CLIENT_H__ */ diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c index b57e51cf09..6f0ea57118 100644 --- a/ext/jack/gstjackaudiosink.c +++ b/ext/jack/gstjackaudiosink.c @@ -96,8 +96,6 @@ struct _GstJackRingBuffer gint sample_rate; gint buffer_size; gint channels; - - jack_port_t **outport; }; struct _GstJackRingBufferClass @@ -123,6 +121,60 @@ static gboolean gst_jack_ring_buffer_pause (GstRingBuffer * buf); static gboolean gst_jack_ring_buffer_stop (GstRingBuffer * buf); static guint gst_jack_ring_buffer_delay (GstRingBuffer * buf); +static gboolean +gst_jack_audio_sink_allocate_channels (GstJackAudioSink * sink, gint channels) +{ + jack_client_t *client; + + client = gst_jack_audio_client_get_client (sink->client); + + /* remove ports we don't need */ + while (sink->port_count > channels) { + jack_port_unregister (client, sink->ports[--sink->port_count]); + } + + /* alloc enough output ports */ + sink->ports = g_realloc (sink->ports, sizeof (jack_port_t *) * channels); + + /* create an output port for each channel */ + while (sink->port_count < channels) { + gchar *name; + + /* port names start from 1 */ + name = g_strdup_printf ("out_%d", sink->port_count + 1); + sink->ports[sink->port_count] = + jack_port_register (client, name, + JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0); + if (sink->ports[sink->port_count] == NULL) + return FALSE; + + sink->port_count++; + + g_free (name); + } + return TRUE; +} + +static void +gst_jack_audio_sink_free_channels (GstJackAudioSink * sink) +{ + gint res, i = 0; + jack_client_t *client; + + client = gst_jack_audio_client_get_client (sink->client); + + /* get rid of all ports */ + while (sink->port_count) { + GST_LOG_OBJECT (sink, "unregister port %d", i); + if ((res = jack_port_unregister (client, sink->ports[i++]))) { + GST_DEBUG_OBJECT (sink, "unregister of port failed (%d)", res); + } + sink->port_count--; + } + g_free (sink->ports); + sink->ports = NULL; +} + /* ringbuffer abstract base class */ static GType gst_jack_ring_buffer_get_type (void) @@ -206,7 +258,7 @@ jack_process_cb (jack_nframes_t nframes, void *arg) /* get target buffers */ for (i = 0; i < channels; i++) { - buffers[i] = (sample_t *) jack_port_get_buffer (abuf->outport[i], nframes); + buffers[i] = (sample_t *) jack_port_get_buffer (sink->ports[i], nframes); } if (gst_ring_buffer_prepare_read (buf, &readseg, &readptr, &len)) { @@ -343,30 +395,19 @@ static gboolean gst_jack_ring_buffer_open_device (GstRingBuffer * buf) { GstJackAudioSink *sink; - jack_options_t options; jack_status_t status = 0; sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf)); GST_DEBUG_OBJECT (sink, "open"); - /* never start a server */ - options = JackNoStartServer; - /* if we have a servername, use it */ - if (sink->server != NULL) - options |= JackServerName; - /* open the client */ - sink->client = jack_client_open ("GStreamer", options, &status, sink->server); + sink->client = gst_jack_audio_client_new ("GStreamer", sink->server, + GST_JACK_CLIENT_SINK, + jack_shutdown_cb, + jack_process_cb, jack_buffer_size_cb, jack_sample_rate_cb, buf, &status); if (sink->client == NULL) goto could_not_open; - /* set our callbacks */ - jack_set_process_callback (sink->client, jack_process_cb, buf); - /* these callbacks cause us to error */ - jack_set_buffer_size_callback (sink->client, jack_buffer_size_cb, buf); - jack_set_sample_rate_callback (sink->client, jack_sample_rate_cb, buf); - jack_on_shutdown (sink->client, jack_shutdown_cb, buf); - GST_DEBUG_OBJECT (sink, "opened"); return TRUE; @@ -391,17 +432,13 @@ static gboolean gst_jack_ring_buffer_close_device (GstRingBuffer * buf) { GstJackAudioSink *sink; - gint res; sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf)); GST_DEBUG_OBJECT (sink, "close"); - if ((res = jack_client_close (sink->client))) { - /* just a warning, we assume the client is gone. */ - GST_ELEMENT_WARNING (sink, RESOURCE, CLOSE, - (NULL), ("Jack client close error (%d)", res)); - } + gst_jack_audio_sink_free_channels (sink); + gst_jack_audio_client_free (sink->client); sink->client = NULL; return TRUE; @@ -426,37 +463,26 @@ gst_jack_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec) const char **ports; gint sample_rate, buffer_size; gint i, channels, res; + jack_client_t *client; sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf)); abuf = GST_JACK_RING_BUFFER_CAST (buf); GST_DEBUG_OBJECT (sink, "acquire"); + client = gst_jack_audio_client_get_client (sink->client); + /* sample rate must be that of the server */ - sample_rate = jack_get_sample_rate (sink->client); + sample_rate = jack_get_sample_rate (client); if (sample_rate != spec->rate) goto wrong_samplerate; channels = spec->channels; - /* alloc enough output ports */ - abuf->outport = g_new (jack_port_t *, channels); + if (!gst_jack_audio_sink_allocate_channels (sink, channels)) + goto out_of_ports; - /* create an output port for each channel */ - for (i = 0; i < channels; i++) { - gchar *name; - - /* port names start from 1 */ - name = g_strdup_printf ("out_%d", i + 1); - abuf->outport[i] = jack_port_register (sink->client, name, - JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0); - if (abuf->outport[i] == NULL) - goto out_of_ports; - - g_free (name); - } - - buffer_size = jack_get_buffer_size (sink->client); + buffer_size = jack_get_buffer_size (client); /* the segment size in bytes, this is large enough to hold a buffer of 32bit floats * for all channels */ @@ -473,7 +499,7 @@ gst_jack_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec) buf->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize); memset (GST_BUFFER_DATA (buf->data), 0, GST_BUFFER_SIZE (buf->data)); - if ((res = jack_activate (sink->client))) + if ((res = gst_jack_audio_client_set_active (sink->client, TRUE))) goto could_not_activate; /* if we need to automatically connect the ports, do so now. We must do this @@ -482,7 +508,7 @@ gst_jack_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec) /* find all the physical input ports. A physical input port is a port * associated with a hardware device. Someone needs connect to a physical * port in order to hear something. */ - ports = jack_get_ports (sink->client, NULL, NULL, + ports = jack_get_ports (client, NULL, NULL, JackPortIsPhysical | JackPortIsInput); if (ports == NULL) { /* no ports? fine then we don't do anything except for posting a warning @@ -501,7 +527,7 @@ gst_jack_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec) break; } /* connect the port to a physical port */ - if ((res = jack_connect (sink->client, jack_port_name (abuf->outport[i]), + if ((res = jack_connect (client, jack_port_name (sink->ports[i]), ports[i]))) goto cannot_connect; } @@ -550,30 +576,20 @@ gst_jack_ring_buffer_release (GstRingBuffer * buf) { GstJackAudioSink *sink; GstJackRingBuffer *abuf; - gint i, res; + gint res; abuf = GST_JACK_RING_BUFFER_CAST (buf); sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf)); GST_DEBUG_OBJECT (sink, "release"); - if ((res = jack_deactivate (sink->client))) { + if ((res = gst_jack_audio_client_set_active (sink->client, FALSE))) { /* we only warn, this means the server is probably shut down and the client * is gone anyway. */ GST_ELEMENT_WARNING (sink, RESOURCE, CLOSE, (NULL), ("Could not deactivate Jack client (%d)", res)); } - /* remove all ports */ - for (i = 0; i < abuf->channels; i++) { - GST_LOG_OBJECT (sink, "unregister port %d", i); - if ((res = jack_port_unregister (sink->client, abuf->outport[i]))) { - GST_DEBUG_OBJECT (sink, "unregister of port failed (%d)", res); - } - abuf->outport[i] = NULL; - } - g_free (abuf->outport); - abuf->outport = NULL; abuf->channels = -1; abuf->buffer_size = -1; abuf->sample_rate = -1; @@ -746,6 +762,8 @@ gst_jack_audio_sink_class_init (GstJackAudioSinkClass * klass) gstbaseaudiosink_class->create_ringbuffer = GST_DEBUG_FUNCPTR (gst_jack_audio_sink_create_ringbuffer); + + gst_jack_audio_client_init (); } static void @@ -754,6 +772,8 @@ gst_jack_audio_sink_init (GstJackAudioSink * sink, { sink->connect = DEFAULT_PROP_CONNECT; sink->server = g_strdup (DEFAULT_PROP_SERVER); + sink->ports = NULL; + sink->port_count = 0; } static void @@ -806,14 +826,17 @@ gst_jack_audio_sink_getcaps (GstBaseSink * bsink) const char **ports; gint min, max; gint rate; + jack_client_t *client; if (sink->client == NULL) goto no_client; + client = gst_jack_audio_client_get_client (sink->client); + if (sink->connect == GST_JACK_CONNECT_AUTO) { /* get a port count, this is the number of channels we can automatically * connect. */ - ports = jack_get_ports (sink->client, NULL, NULL, + ports = jack_get_ports (client, NULL, NULL, JackPortIsPhysical | JackPortIsInput); max = 0; if (ports != NULL) { @@ -822,13 +845,13 @@ gst_jack_audio_sink_getcaps (GstBaseSink * bsink) } else max = 0; } else { - /* we allow any number of pads, somoething else is going to connect the + /* we allow any number of pads, something else is going to connect the * pads. */ max = G_MAXINT; } min = MIN (1, max); - rate = jack_get_sample_rate (sink->client); + rate = jack_get_sample_rate (client); GST_DEBUG_OBJECT (sink, "got %d-%d ports, samplerate: %d", min, max, rate); diff --git a/ext/jack/gstjackaudiosink.h b/ext/jack/gstjackaudiosink.h index 245521b2a1..12c82a834f 100644 --- a/ext/jack/gstjackaudiosink.h +++ b/ext/jack/gstjackaudiosink.h @@ -27,6 +27,8 @@ #include #include +#include "gstjackaudioclient.h" + G_BEGIN_DECLS #define GST_TYPE_JACK_AUDIO_SINK (gst_jack_audio_sink_get_type()) @@ -63,6 +65,7 @@ typedef enum { struct _GstJackAudioSink { GstBaseAudioSink element; + /*< private >*/ /* cached caps */ GstCaps *caps; @@ -71,17 +74,15 @@ struct _GstJackAudioSink { gchar *server; /* our client */ - jack_client_t *client; + GstJackAudioClient *client; - /*< private >*/ - gpointer _gst_reserved[GST_PADDING]; + /* our ports */ + jack_port_t **ports; + int port_count; }; struct _GstJackAudioSinkClass { GstBaseAudioSinkClass parent_class; - - /*< private >*/ - gpointer _gst_reserved[GST_PADDING]; }; GType gst_jack_audio_sink_get_type (void); From a70ffeed7cc169e9c1586847b07b2bc5f566cf4c Mon Sep 17 00:00:00 2001 From: Paul Davis Date: Sun, 18 Mar 2007 17:57:48 +0000 Subject: [PATCH 171/351] ext/jack/gstjackaudioclient.c: Don't need to take the connection lock, it will not be used and could cause deadlocks. Original commit message from CVS: Based on patch by: Paul Davis * ext/jack/gstjackaudioclient.c: (gst_jack_audio_unref_connection): Don't need to take the connection lock, it will not be used and could cause deadlocks. --- ext/jack/gstjackaudioclient.c | 26 +++++++++++++++++++------- 1 file changed, 19 insertions(+), 7 deletions(-) diff --git a/ext/jack/gstjackaudioclient.c b/ext/jack/gstjackaudioclient.c index c81125939d..9777cd9781 100644 --- a/ext/jack/gstjackaudioclient.c +++ b/ext/jack/gstjackaudioclient.c @@ -283,18 +283,32 @@ static void gst_jack_audio_unref_connection (GstJackAudioConnection * conn) { gint res; + gboolean zero; - GST_DEBUG ("unref connection %p", conn); + GST_DEBUG ("unref connection %p refcnt %d", conn, conn->refcount); G_LOCK (connections_lock); conn->refcount--; - if (conn->refcount == 0) { + if ((zero = (conn->refcount == 0))) { GST_DEBUG ("closing connection %p", conn); - /* remove from list */ + /* remove from list, we can release the mutex after removing the connection + * from the list because after that, nobody can access the connection anymore. */ connections = g_list_remove (connections, conn); + } + G_UNLOCK (connections_lock); - /* grab lock to be sure that we are not in one of the callbacks */ - g_mutex_lock (conn->lock); + /* if we are zero, close and cleanup the connection */ + if (zero) { + /* don't use conn->lock here. two reasons: + * + * 1) its not necessary: jack_deactivate() will not return until the JACK thread + * associated with this connection is cleaned up by a thread join, hence + * no more callbacks can occur or be in progress. + * + * 2) it would deadlock anyway, because jack_deactivate() will sleep + * waiting for the JACK thread, and can thus cause deadlock in + * jack_process_cb() + */ if ((res = jack_deactivate (conn->client))) { /* we only warn, this means the server is probably shut down and the client * is gone anyway. */ @@ -305,7 +319,6 @@ gst_jack_audio_unref_connection (GstJackAudioConnection * conn) /* we assume the client is gone. */ GST_WARNING ("close failed (%d)", res); } - g_mutex_unlock (conn->lock); /* free resources */ g_mutex_free (conn->lock); @@ -313,7 +326,6 @@ gst_jack_audio_unref_connection (GstJackAudioConnection * conn) g_free (conn->server); g_free (conn); } - G_UNLOCK (connections_lock); } static void From 5b2d68977423ce0f5610e93e37752f0c2ce71fff Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Wed, 4 Apr 2007 07:36:28 +0000 Subject: [PATCH 172/351] ext/jack/gstjackaudiosink.c: Try t better name clients. properly handle return codes when re- establishing links. Original commit message from CVS: * ext/jack/gstjackaudiosink.c: (gst_jack_ring_buffer_open_device), (gst_jack_ring_buffer_acquire): Try t better name clients. properly handle return codes when re- establishing links. --- ext/jack/gstjackaudiosink.c | 22 +++++++++++++++------- 1 file changed, 15 insertions(+), 7 deletions(-) diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c index 6f0ea57118..d3efe8d657 100644 --- a/ext/jack/gstjackaudiosink.c +++ b/ext/jack/gstjackaudiosink.c @@ -26,7 +26,7 @@ * * * - * A Sink that outputs data to Jack ports. + * A Sink that outputs data to Jack ports. * * * It will create N Jack ports named out_<num> where <num> is starting from 1. @@ -39,7 +39,7 @@ * * When the ::connect property is set to auto, this element will try to connect * each output port to a random physical jack input pin. In this mode, the sink - * will expose the number of physical channels on its pad caps. + * will expose the number of physical channels on its pad caps. * * * When the ::connect property is set to none, the element will accept any @@ -396,12 +396,17 @@ gst_jack_ring_buffer_open_device (GstRingBuffer * buf) { GstJackAudioSink *sink; jack_status_t status = 0; + const gchar *name; sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf)); GST_DEBUG_OBJECT (sink, "open"); - sink->client = gst_jack_audio_client_new ("GStreamer", sink->server, + name = g_get_application_name (); + if (!name) + name = "GStreamer"; + + sink->client = gst_jack_audio_client_new (name, sink->server, GST_JACK_CLIENT_SINK, jack_shutdown_cb, jack_process_cb, jack_buffer_size_cb, jack_sample_rate_cb, buf, &status); @@ -526,9 +531,11 @@ gst_jack_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec) ("No more physical ports, leaving some ports unconnected")); break; } + GST_DEBUG_OBJECT (sink, "try connecting to %s", + jack_port_name (sink->ports[i])); /* connect the port to a physical port */ - if ((res = jack_connect (client, jack_port_name (sink->ports[i]), - ports[i]))) + res = jack_connect (client, jack_port_name (sink->ports[i]), ports[i]); + if (res != 0 && res != EEXIST) goto cannot_connect; } free (ports); @@ -558,13 +565,14 @@ out_of_ports: could_not_activate: { GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL), - ("Could not activate client (%d)", res)); + ("Could not activate client (%d:%s)", res, g_strerror (res))); return FALSE; } cannot_connect: { GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL), - ("Could not connect output ports to physical ports (%d)", res)); + ("Could not connect output ports to physical ports (%d:%s)", + res, g_strerror (res))); free (ports); return FALSE; } From be5f69be58fa5bce9376969933d8c3073aa4a0e0 Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Wed, 18 Jul 2007 07:42:47 +0000 Subject: [PATCH 173/351] ext/jack/gstjackaudiosink.c: Add stdlib include here too. Original commit message from CVS: * ext/jack/gstjackaudiosink.c: (gst_jack_ring_buffer_open_device), (gst_jack_ring_buffer_acquire): Add stdlib include here too. --- ext/jack/gstjackaudiosink.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c index d3efe8d657..42dda79a6f 100644 --- a/ext/jack/gstjackaudiosink.c +++ b/ext/jack/gstjackaudiosink.c @@ -62,6 +62,8 @@ * * Last reviewed on 2006-11-30 (0.10.4) */ + +#include #include #include "gstjackaudiosink.h" From 459484bc4d02ce17a7aeb34c3c3b161817fde905 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Sebastian=20Dr=C3=B6ge?= Date: Thu, 13 Mar 2008 14:25:20 +0000 Subject: [PATCH 174/351] Use GST_LICENSE, GST_PACKAGE_NAME and GST_PACKAGE_ORIGIN instead of hardcoding values where possible. Fixes bug #522212. Original commit message from CVS: * ext/alsaspdif/alsaspdifsink.c: * ext/gsm/gstgsm.c: * ext/jack/gstjack.c: * ext/libmms/gstmms.c: * ext/neon/gstneonhttpsrc.c: * ext/shout/gstshout.c: * ext/timidity/gsttimidity.c: * ext/timidity/gstwildmidi.c: * gst/nuvdemux/gstnuvdemux.c: * gst/tta/gsttta.c: Use GST_LICENSE, GST_PACKAGE_NAME and GST_PACKAGE_ORIGIN instead of hardcoding values where possible. Fixes bug #522212. --- ext/jack/gstjack.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 52b4b2455e..72f501d0f6 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -37,5 +37,4 @@ GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, "jack", "Jack elements", - plugin_init, - VERSION, "LGPL", "Gstreamer", "http://gstreamer.freedesktop.org") + plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN) From d418db9aedb945b58f567e71b9251be3cca074cb Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Sun, 6 Apr 2008 20:18:16 +0000 Subject: [PATCH 175/351] ext/jack/gstjackaudiosink.c: Work around missing bits of thread-safety on older GLibs some more to avoid assertions w... Original commit message from CVS: * ext/jack/gstjackaudiosink.c: (gst_jack_audio_sink_class_init): Work around missing bits of thread-safety on older GLibs some more to avoid assertions when starting up multiple playbin objects concurrently (see #512382). --- ext/jack/gstjackaudiosink.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c index 42dda79a6f..5223e4d00a 100644 --- a/ext/jack/gstjackaudiosink.c +++ b/ext/jack/gstjackaudiosink.c @@ -773,6 +773,10 @@ gst_jack_audio_sink_class_init (GstJackAudioSinkClass * klass) gstbaseaudiosink_class->create_ringbuffer = GST_DEBUG_FUNCPTR (gst_jack_audio_sink_create_ringbuffer); + /* ref class from a thread-safe context to work around missing bit of + * thread-safety in GObject */ + g_type_class_ref (GST_TYPE_JACK_RING_BUFFER); + gst_jack_audio_client_init (); } From 102b2e3fc453ecc48bc32e9edb1a125a820c7adb Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Mon, 26 May 2008 17:52:21 +0000 Subject: [PATCH 176/351] ext/jack/gstjackaudiosink.c: Include the element name in the port name to avoid duplicate port names. Original commit message from CVS: * ext/jack/gstjackaudiosink.c: (gst_jack_audio_sink_allocate_channels): Include the element name in the port name to avoid duplicate port names. --- ext/jack/gstjackaudiosink.c | 13 ++++++++----- 1 file changed, 8 insertions(+), 5 deletions(-) diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c index 5223e4d00a..27b0979292 100644 --- a/ext/jack/gstjackaudiosink.c +++ b/ext/jack/gstjackaudiosink.c @@ -29,7 +29,8 @@ * A Sink that outputs data to Jack ports. * * - * It will create N Jack ports named out_<num> where <num> is starting from 1. + * It will create N Jack ports named out_<name>_<num> where + * <name> is the element name and <num> is starting from 1. * Each port corresponds to a gstreamer channel. * * @@ -142,11 +143,13 @@ gst_jack_audio_sink_allocate_channels (GstJackAudioSink * sink, gint channels) while (sink->port_count < channels) { gchar *name; - /* port names start from 1 */ - name = g_strdup_printf ("out_%d", sink->port_count + 1); + /* port names start from 1 and are local to the element */ + name = + g_strdup_printf ("out_%s_%d", GST_ELEMENT_NAME (sink), + sink->port_count + 1); sink->ports[sink->port_count] = - jack_port_register (client, name, - JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0); + jack_port_register (client, name, JACK_DEFAULT_AUDIO_TYPE, + JackPortIsOutput, 0); if (sink->ports[sink->port_count] == NULL) return FALSE; From 8909b106f0ee5b5c9829a9fc0756951f899a61a4 Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Thu, 12 Jun 2008 14:49:18 +0000 Subject: [PATCH 177/351] Do not use short_description in section docs for elements. We extract them from element details and there will be war... Original commit message from CVS: * ext/dc1394/gstdc1394.c: * ext/ivorbis/vorbisdec.c: * ext/jack/gstjackaudiosink.c: * ext/metadata/gstmetadatademux.c: * ext/mythtv/gstmythtvsrc.c: * ext/theora/theoradec.c: * gst-libs/gst/app/gstappsink.c: * gst/bayer/gstbayer2rgb.c: * gst/deinterlace/gstdeinterlace.c: * gst/rawparse/gstaudioparse.c: * gst/rawparse/gstvideoparse.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpclient.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/selector/gstinputselector.c: * gst/selector/gstoutputselector.c: * gst/videosignal/gstvideoanalyse.c: * gst/videosignal/gstvideodetect.c: * gst/videosignal/gstvideomark.c: * sys/oss4/oss4-mixer.c: * sys/oss4/oss4-sink.c: * sys/oss4/oss4-source.c: Do not use short_description in section docs for elements. We extract them from element details and there will be warnings if they differ. Also fixing up the ChangeLog order. --- ext/jack/gstjackaudiosink.c | 1 - 1 file changed, 1 deletion(-) diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c index 27b0979292..5710eda612 100644 --- a/ext/jack/gstjackaudiosink.c +++ b/ext/jack/gstjackaudiosink.c @@ -21,7 +21,6 @@ /** * SECTION:element-jackaudiosink - * @short_description: JACK audio sink * @see_also: #GstBaseAudioSink, #GstRingBuffer * * From d32bf24142a45455f35ccff672ea4cea21de507f Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Fri, 13 Jun 2008 11:59:23 +0000 Subject: [PATCH 178/351] docs/plugins/: docs/plugins/inspect/plugin-mythtv.xml Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * docs/plugins/gst-plugins-bad-plugins.args: * docs/plugins/gst-plugins-bad-plugins.hierarchy: * docs/plugins/gst-plugins-bad-plugins.interfaces: * docs/plugins/gst-plugins-bad-plugins.prerequisites: * docs/plugins/gst-plugins-bad-plugins.signals: * docs/plugins/inspect/plugin-alsaspdif.xml: * docs/plugins/inspect/plugin-amrwb.xml: * docs/plugins/inspect/plugin-app.xml: * docs/plugins/inspect/plugin-bayer.xml: * docs/plugins/inspect/plugin-bz2.xml: * docs/plugins/inspect/plugin-cdaudio.xml: * docs/plugins/inspect/plugin-cdxaparse.xml: * docs/plugins/inspect/plugin-dtsdec.xml: * docs/plugins/inspect/plugin-dvb.xml: * docs/plugins/inspect/plugin-dvdspu.xml: * docs/plugins/inspect/plugin-faac.xml: * docs/plugins/inspect/plugin-faad.xml: * docs/plugins/inspect/plugin-fbdevsink.xml: * docs/plugins/inspect/plugin-festival.xml: * docs/plugins/inspect/plugin-filter.xml: * docs/plugins/inspect/plugin-flvdemux.xml: * docs/plugins/inspect/plugin-freeze.xml: * docs/plugins/inspect/plugin-gsm.xml: * docs/plugins/inspect/plugin-gstinterlace.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-h264parse.xml: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-ladspa.xml: * docs/plugins/inspect/plugin-metadata.xml: * docs/plugins/inspect/plugin-mms.xml: * docs/plugins/inspect/plugin-modplug.xml: * docs/plugins/inspect/plugin-mpeg2enc.xml: * docs/plugins/inspect/plugin-mpeg4videoparse.xml: * docs/plugins/inspect/plugin-mpegtsparse.xml: * docs/plugins/inspect/plugin-mpegvideoparse.xml: * docs/plugins/inspect/plugin-musepack.xml: * docs/plugins/inspect/plugin-musicbrainz.xml: * docs/plugins/inspect/plugin-mve.xml: * docs/plugins/inspect/plugin-mythtv.xml * docs/plugins/inspect/plugin-nas.xml: * docs/plugins/inspect/plugin-neon.xml: * docs/plugins/inspect/plugin-nsfdec.xml: * docs/plugins/inspect/plugin-nuvdemux.xml: * docs/plugins/inspect/plugin-oss4.xml * docs/plugins/inspect/plugin-rawparse.xml: * docs/plugins/inspect/plugin-real.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-rfbsrc.xml: * docs/plugins/inspect/plugin-sdl.xml: * docs/plugins/inspect/plugin-sdp.xml: * docs/plugins/inspect/plugin-selector.xml: * docs/plugins/inspect/plugin-sndfile.xml: * docs/plugins/inspect/plugin-soundtouch.xml: * docs/plugins/inspect/plugin-spcdec.xml: * docs/plugins/inspect/plugin-speed.xml: * docs/plugins/inspect/plugin-speexresample.xml: * docs/plugins/inspect/plugin-stereo.xml: * docs/plugins/inspect/plugin-subenc.xml * docs/plugins/inspect/plugin-timidity.xml: * docs/plugins/inspect/plugin-tta.xml: * docs/plugins/inspect/plugin-vcdsrc.xml: * docs/plugins/inspect/plugin-videosignal.xml: * docs/plugins/inspect/plugin-vmnc.xml: * docs/plugins/inspect/plugin-wildmidi.xml: * docs/plugins/inspect/plugin-x264.xml: * docs/plugins/inspect/plugin-xvid.xml: * docs/plugins/inspect/plugin-y4menc.xml: * ext/amrwb/gstamrwbdec.c: * ext/amrwb/gstamrwbenc.c: * ext/amrwb/gstamrwbparse.c: * ext/dc1394/gstdc1394.c: * ext/directfb/dfbvideosink.c: * ext/ivorbis/vorbisdec.c: * ext/jack/gstjackaudiosink.c: * ext/mpeg2enc/gstmpeg2enc.cc: * ext/mplex/gstmplex.cc: * ext/musicbrainz/gsttrm.c: * ext/mythtv/gstmythtvsrc.c: * ext/theora/theoradec.c: * ext/timidity/gsttimidity.c: * ext/timidity/gstwildmidi.c: * gst-libs/gst/app/gstappsink.c: * gst/deinterlace/gstdeinterlace.c: * gst/dvdspu/gstdvdspu.c: * gst/festival/gstfestival.c: * gst/freeze/gstfreeze.c: * gst/interleave/deinterleave.c: * gst/interleave/interleave.c: * gst/modplug/gstmodplug.cc: * gst/nuvdemux/gstnuvdemux.c: Add missing elements to docs. Fix doc-markup: use convinience syntax for examples (produces valid docbook), add several refsec2 when we have several titles. Fix some types. --- ext/jack/gstjackaudiosink.c | 40 +++++++++++++++---------------------- 1 file changed, 16 insertions(+), 24 deletions(-) diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c index 5710eda612..05571b2cbe 100644 --- a/ext/jack/gstjackaudiosink.c +++ b/ext/jack/gstjackaudiosink.c @@ -23,41 +23,33 @@ * SECTION:element-jackaudiosink * @see_also: #GstBaseAudioSink, #GstRingBuffer * - * - * * A Sink that outputs data to Jack ports. - * - * + * * It will create N Jack ports named out_<name>_<num> where * <name> is the element name and <num> is starting from 1. * Each port corresponds to a gstreamer channel. - * - * + * * The samplerate as exposed on the caps is always the same as the samplerate of * the jack server. - * - * - * When the ::connect property is set to auto, this element will try to connect - * each output port to a random physical jack input pin. In this mode, the sink - * will expose the number of physical channels on its pad caps. - * - * - * When the ::connect property is set to none, the element will accept any - * number of input channels and will create (but not connect) an output port for - * each channel. - * - * + * + * When the #GstJackAudioSink:connect property is set to auto, this element + * will try to connect each output port to a random physical jack input pin. In + * this mode, the sink will expose the number of physical channels on its pad + * caps. + * + * When the #GstJackAudioSink:connect property is set to none, the element will + * accept any number of input channels and will create (but not connect) an + * output port for each channel. + * * The element will generate an error when the Jack server is shut down when it * was PAUSED or PLAYING. This element does not support dynamic rate and buffer * size changes at runtime. - * + * + * * Example launch line - * - * + * |[ * gst-launch audiotestsrc ! jackaudiosink - * - * Play a sine wave to using jack. - * + * ]| Play a sine wave to using jack. * * * Last reviewed on 2006-11-30 (0.10.4) From b34bd1425c10b43d33a16570a7542f8068874bd8 Mon Sep 17 00:00:00 2001 From: Tristan Matthews Date: Thu, 7 Aug 2008 08:47:40 +0000 Subject: [PATCH 179/351] ext/jack/: Add a jackaudiosrc. Refactor sink slightly for better code reuse. Original commit message from CVS: patch by: Tristan Matthews * ext/jack/Makefile.am: * ext/jack/gstjack.c: * ext/jack/gstjackaudioclient.c: * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosink.h: * ext/jack/gstjackaudiosrc.c: * ext/jack/gstjackaudiosrc.h: * ext/jack/gstjackringbuffer.h: Add a jackaudiosrc. Refactor sink slightly for better code reuse. Fixes #545197. --- ext/jack/Makefile.am | 4 +- ext/jack/gstjack.c | 23 + ext/jack/gstjackaudioclient.c | 3 + ext/jack/gstjackaudiosink.c | 72 +-- ext/jack/gstjackaudiosink.h | 17 +- ext/jack/gstjackaudiosrc.c | 840 ++++++++++++++++++++++++++++++++++ ext/jack/gstjackaudiosrc.h | 94 ++++ ext/jack/gstjackringbuffer.h | 90 ++++ 8 files changed, 1054 insertions(+), 89 deletions(-) create mode 100644 ext/jack/gstjackaudiosrc.c create mode 100644 ext/jack/gstjackaudiosrc.h create mode 100644 ext/jack/gstjackringbuffer.h diff --git a/ext/jack/Makefile.am b/ext/jack/Makefile.am index 17efdfa94c..abcc39a597 100644 --- a/ext/jack/Makefile.am +++ b/ext/jack/Makefile.am @@ -1,11 +1,11 @@ plugin_LTLIBRARIES = libgstjack.la -libgstjack_la_SOURCES = gstjack.c gstjackaudiosink.c gstjackaudioclient.c +libgstjack_la_SOURCES = gstjack.c gstjackaudiosrc.c gstjackaudiosink.c gstjackaudioclient.c libgstjack_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) $(JACK_CFLAGS) libgstjack_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR) $(JACK_LIBS) libgstjack_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) -noinst_HEADERS = gstjackaudiosink.h gstjackaudioclient.h +noinst_HEADERS = gstjackaudiosrc.h gstjackaudiosink.h gstjackaudioclient.h gstjack.h gstjackringbuffer.h EXTRA_DIST = README diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 72f501d0f6..96afd06e63 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -21,11 +21,34 @@ #include "config.h" #endif +#include "gstjackaudiosrc.h" #include "gstjackaudiosink.h" +GType +gst_jack_connect_get_type (void) +{ + static GType jack_connect_type = 0; + static const GEnumValue jack_connect[] = { + {GST_JACK_CONNECT_NONE, + "Don't automatically connect ports to physical ports", "none"}, + {GST_JACK_CONNECT_AUTO, + "Automatically connect ports to physical ports", "auto"}, + {0, NULL, NULL}, + }; + + if (!jack_connect_type) { + jack_connect_type = g_enum_register_static ("GstJackConnect", jack_connect); + } + return jack_connect_type; +} + + static gboolean plugin_init (GstPlugin * plugin) { + if (!gst_element_register (plugin, "jackaudiosrc", GST_RANK_PRIMARY, + GST_TYPE_JACK_AUDIO_SRC)) + return FALSE; if (!gst_element_register (plugin, "jackaudiosink", GST_RANK_PRIMARY, GST_TYPE_JACK_AUDIO_SINK)) return FALSE; diff --git a/ext/jack/gstjackaudioclient.c b/ext/jack/gstjackaudioclient.c index 9777cd9781..1aa1baf8a7 100644 --- a/ext/jack/gstjackaudioclient.c +++ b/ext/jack/gstjackaudioclient.c @@ -127,6 +127,9 @@ jack_shutdown_cb (void *arg) GstJackAudioConnection *conn = (GstJackAudioConnection *) arg; GList *walk; + GST_DEBUG ("disconnect client %s from server %s", conn->id, + GST_STR_NULL (conn->server)); + g_mutex_lock (conn->lock); for (walk = conn->src_clients; walk; walk = g_list_next (walk)) { GstJackAudioClient *client = (GstJackAudioClient *) walk->data; diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c index 05571b2cbe..ec257deb17 100644 --- a/ext/jack/gstjackaudiosink.c +++ b/ext/jack/gstjackaudiosink.c @@ -59,62 +59,11 @@ #include #include "gstjackaudiosink.h" +#include "gstjackringbuffer.h" GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_sink_debug); #define GST_CAT_DEFAULT gst_jack_audio_sink_debug -typedef jack_default_audio_sample_t sample_t; - -#define GST_TYPE_JACK_RING_BUFFER \ - (gst_jack_ring_buffer_get_type()) -#define GST_JACK_RING_BUFFER(obj) \ - (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_JACK_RING_BUFFER,GstJackRingBuffer)) -#define GST_JACK_RING_BUFFER_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_JACK_RING_BUFFER,GstJackRingBufferClass)) -#define GST_JACK_RING_BUFFER_GET_CLASS(obj) \ - (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_JACK_RING_BUFFER, GstJackRingBufferClass)) -#define GST_JACK_RING_BUFFER_CAST(obj) \ - ((GstJackRingBuffer *)obj) -#define GST_IS_JACK_RING_BUFFER(obj) \ - (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_JACK_RING_BUFFER)) -#define GST_IS_JACK_RING_BUFFER_CLASS(klass)\ - (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_JACK_RING_BUFFER)) - -typedef struct _GstJackRingBuffer GstJackRingBuffer; -typedef struct _GstJackRingBufferClass GstJackRingBufferClass; - -struct _GstJackRingBuffer -{ - GstRingBuffer object; - - gint sample_rate; - gint buffer_size; - gint channels; -}; - -struct _GstJackRingBufferClass -{ - GstRingBufferClass parent_class; -}; - -static void gst_jack_ring_buffer_class_init (GstJackRingBufferClass * klass); -static void gst_jack_ring_buffer_init (GstJackRingBuffer * ringbuffer, - GstJackRingBufferClass * klass); -static void gst_jack_ring_buffer_dispose (GObject * object); -static void gst_jack_ring_buffer_finalize (GObject * object); - -static GstRingBufferClass *ring_parent_class = NULL; - -static gboolean gst_jack_ring_buffer_open_device (GstRingBuffer * buf); -static gboolean gst_jack_ring_buffer_close_device (GstRingBuffer * buf); -static gboolean gst_jack_ring_buffer_acquire (GstRingBuffer * buf, - GstRingBufferSpec * spec); -static gboolean gst_jack_ring_buffer_release (GstRingBuffer * buf); -static gboolean gst_jack_ring_buffer_start (GstRingBuffer * buf); -static gboolean gst_jack_ring_buffer_pause (GstRingBuffer * buf); -static gboolean gst_jack_ring_buffer_stop (GstRingBuffer * buf); -static guint gst_jack_ring_buffer_delay (GstRingBuffer * buf); - static gboolean gst_jack_audio_sink_allocate_channels (GstJackAudioSink * sink, gint channels) { @@ -689,25 +638,6 @@ enum PROP_LAST }; -#define GST_TYPE_JACK_CONNECT (gst_jack_connect_get_type()) -static GType -gst_jack_connect_get_type (void) -{ - static GType jack_connect_type = 0; - static const GEnumValue jack_connect[] = { - {GST_JACK_CONNECT_NONE, - "Don't automatically connect ports to physical ports", "none"}, - {GST_JACK_CONNECT_AUTO, - "Automatically connect ports to physical ports", "auto"}, - {0, NULL, NULL}, - }; - - if (!jack_connect_type) { - jack_connect_type = g_enum_register_static ("GstJackConnect", jack_connect); - } - return jack_connect_type; -} - #define _do_init(bla) \ GST_DEBUG_CATEGORY_INIT (gst_jack_audio_sink_debug, "jacksink", 0, "jacksink element"); diff --git a/ext/jack/gstjackaudiosink.h b/ext/jack/gstjackaudiosink.h index 12c82a834f..b4a770332f 100644 --- a/ext/jack/gstjackaudiosink.h +++ b/ext/jack/gstjackaudiosink.h @@ -27,6 +27,7 @@ #include #include +#include "gstjack.h" #include "gstjackaudioclient.h" G_BEGIN_DECLS @@ -41,22 +42,6 @@ G_BEGIN_DECLS typedef struct _GstJackAudioSink GstJackAudioSink; typedef struct _GstJackAudioSinkClass GstJackAudioSinkClass; -/** - * GstJackConnect: - * @GST_JACK_CONNECT_NONE: Don't automatically connect to physical ports. - * In this mode, the element will accept any number of input channels and will - * create (but not connect) an output port for each channel. - * @GST_JACK_CONNECT_AUTO: In this mode, the element will try to connect each - * output port to a random physical jack input pin. The sink will - * expose the number of physical channels on its pad caps. - * - * Specify how the output ports will be connected. - */ -typedef enum { - GST_JACK_CONNECT_NONE, - GST_JACK_CONNECT_AUTO -} GstJackConnect; - /** * GstJackAudioSink: * diff --git a/ext/jack/gstjackaudiosrc.c b/ext/jack/gstjackaudiosrc.c new file mode 100644 index 0000000000..d41b62ffd0 --- /dev/null +++ b/ext/jack/gstjackaudiosrc.c @@ -0,0 +1,840 @@ +/* GStreamer + * Copyright (C) 2008 Tristan Matthews + * + * Permission is hereby granted, free of charge, to any person obtaining a + * copy of this software and associated documentation files (the "Software"), + * to deal in the Software without restriction, including without limitation + * the rights to use, copy, modify, merge, publish, distribute, sublicense, + * and/or sell copies of the Software, and to permit persons to whom the + * Software is furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE + * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING + * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER + * DEALINGS IN THE SOFTWARE. + * + * Alternatively, the contents of this file may be used under the + * GNU Lesser General Public License Version 2.1 (the "LGPL"), in + * which case the following provisions apply instead of the ones + * mentioned above: + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +/** + * SECTION:element-jackaudiosrc + * @see_also: #GstBaseAudioSrc, #GstRingBuffer + * + * A Src that inputs data from Jack ports. + * + * It will create N Jack ports named in_<name>_<num> where + * <name> is the element name and <num> is starting from 1. + * Each port corresponds to a gstreamer channel. + * + * The samplerate as exposed on the caps is always the same as the samplerate of + * the jack server. + * + * When the #GstJackAudioSrc:connect property is set to auto, this element + * will try to connect each input port to a random physical jack output pin. + * + * When the #GstJackAudioSrc:connect property is set to none, the element will + * accept any number of output channels and will create (but not connect) an + * input port for each channel. + * + * The element will generate an error when the Jack server is shut down when it + * was PAUSED or PLAYING. This element does not support dynamic rate and buffer + * size changes at runtime. + * + * + * Example launch line + * |[ + * gst-launch jackaudiosrc connect=0 ! jackaudiosink connect=0 + * ]| Get audio input into gstreamer from jack. + * + * + * Last reviewed on 2008-07-22 (0.10.4) + */ + +#include +#include +#include + +#include "gstjackaudiosrc.h" +#include "gstjackringbuffer.h" + +GST_DEBUG_CATEGORY_STATIC (gst_jackaudiosrc_debug); +#define GST_CAT_DEFAULT gst_jackaudiosrc_debug + +static gboolean +gst_jack_audio_src_allocate_channels (GstJackAudioSrc * src, gint channels) +{ + jack_client_t *client; + + client = gst_jack_audio_client_get_client (src->client); + + /* remove ports we don't need */ + while (src->port_count > channels) + jack_port_unregister (client, src->ports[--src->port_count]); + + /* alloc enough input ports */ + src->ports = g_realloc (src->ports, sizeof (jack_port_t *) * channels); + + /* create an input port for each channel */ + while (src->port_count < channels) { + gchar *name; + + /* port names start from 1 and are local to the element */ + name = + g_strdup_printf ("in_%s_%d", GST_ELEMENT_NAME (src), + src->port_count + 1); + src->ports[src->port_count] = + jack_port_register (client, name, JACK_DEFAULT_AUDIO_TYPE, + JackPortIsInput, 0); + if (src->ports[src->port_count] == NULL) + return FALSE; + + src->port_count++; + + g_free (name); + } + return TRUE; +} + +static void +gst_jack_audio_src_free_channels (GstJackAudioSrc * src) +{ + gint res, i = 0; + jack_client_t *client; + + client = gst_jack_audio_client_get_client (src->client); + + /* get rid of all ports */ + while (src->port_count) { + GST_LOG_OBJECT (src, "unregister port %d", i); + if ((res = jack_port_unregister (client, src->ports[i++]))) + GST_DEBUG_OBJECT (src, "unregister of port failed (%d)", res); + + src->port_count--; + } + g_free (src->ports); + src->ports = NULL; +} + +/* ringbuffer abstract base class */ +static GType +gst_jack_ring_buffer_get_type () +{ + static GType ringbuffer_type = 0; + + if (!ringbuffer_type) { + static const GTypeInfo ringbuffer_info = { sizeof (GstJackRingBufferClass), + NULL, + NULL, + (GClassInitFunc) gst_jack_ring_buffer_class_init, + NULL, + NULL, + sizeof (GstJackRingBuffer), + 0, + (GInstanceInitFunc) gst_jack_ring_buffer_init, + NULL + }; + + ringbuffer_type = + g_type_register_static (GST_TYPE_RING_BUFFER, + "GstJackAudioSrcRingBuffer", &ringbuffer_info, 0); + } + return ringbuffer_type; +} + +static void +gst_jack_ring_buffer_class_init (GstJackRingBufferClass * klass) +{ + GObjectClass *gobject_class; + GstObjectClass *gstobject_class; + GstRingBufferClass *gstringbuffer_class; + + gobject_class = (GObjectClass *) klass; + gstobject_class = (GstObjectClass *) klass; + gstringbuffer_class = (GstRingBufferClass *) klass; + + ring_parent_class = g_type_class_peek_parent (klass); + + gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_dispose); + gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_finalize); + + gstringbuffer_class->open_device = + GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_open_device); + gstringbuffer_class->close_device = + GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_close_device); + gstringbuffer_class->acquire = + GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_acquire); + gstringbuffer_class->release = + GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_release); + gstringbuffer_class->start = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_start); + gstringbuffer_class->pause = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_pause); + gstringbuffer_class->resume = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_start); + gstringbuffer_class->stop = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_stop); + + gstringbuffer_class->delay = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_delay); +} + +/* this is the callback of jack. This should be RT-safe. + * Writes samples from the jack input port's buffer to the gst ring buffer. + */ +static int +jack_process_cb (jack_nframes_t nframes, void *arg) +{ + GstJackAudioSrc *src; + GstRingBuffer *buf; + GstJackRingBuffer *abuf; + gint len, givenLen; + guint8 *writeptr, *dataStart; + gint writeseg; + gint channels, i, j; + sample_t **buffers, *data; + + buf = GST_RING_BUFFER_CAST (arg); + abuf = GST_JACK_RING_BUFFER_CAST (arg); + src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf)); + + channels = buf->spec.channels; + len = sizeof (sample_t) * nframes * channels; + + /* alloc pointers to samples */ + buffers = g_alloca (sizeof (sample_t *) * channels); + data = g_alloca (len); + + /* get input buffers */ + for (i = 0; i < channels; i++) + buffers[i] = (sample_t *) jack_port_get_buffer (src->ports[i], nframes); + + //writeptr = data; + dataStart = (guint8 *) data; + + /* the samples in the jack input buffers have to be interleaved into the + * ringbuffer + */ + + for (i = 0; i < nframes; ++i) + for (j = 0; j < channels; ++j) + *data++ = buffers[j][i]; + + if (gst_ring_buffer_prepare_read (buf, &writeseg, &writeptr, &givenLen)) { + memcpy (writeptr, (char *) dataStart, givenLen); + + GST_DEBUG ("copy %d frames: %p, %d bytes, %d channels", nframes, writeptr, + len / channels, channels); + + /* clear written samples in the ringbuffer */ + // gst_ring_buffer_clear(buf, 0); + + /* we wrote one segment */ + gst_ring_buffer_advance (buf, 1); + } + return 0; +} + +/* we error out */ +static int +jack_sample_rate_cb (jack_nframes_t nframes, void *arg) +{ + GstJackAudioSrc *src; + GstJackRingBuffer *abuf; + + abuf = GST_JACK_RING_BUFFER_CAST (arg); + src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (arg)); + + if (abuf->sample_rate != -1 && abuf->sample_rate != nframes) + goto not_supported; + + return 0; + + /* ERRORS */ +not_supported: + { + GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, + (NULL), ("Jack changed the sample rate, which is not supported")); + return 1; + } +} + +/* we error out */ +static int +jack_buffer_size_cb (jack_nframes_t nframes, void *arg) +{ + GstJackAudioSrc *src; + GstJackRingBuffer *abuf; + + abuf = GST_JACK_RING_BUFFER_CAST (arg); + src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (arg)); + + if (abuf->buffer_size != -1 && abuf->buffer_size != nframes) + goto not_supported; + + return 0; + + /* ERRORS */ +not_supported: + { + GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, + (NULL), ("Jack changed the buffer size, which is not supported")); + return 1; + } +} + +static void +jack_shutdown_cb (void *arg) +{ + GstJackAudioSrc *src; + + src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (arg)); + + GST_DEBUG_OBJECT (src, "shutdown"); + + GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, + (NULL), ("Jack server shutdown")); +} + +static void +gst_jack_ring_buffer_init (GstJackRingBuffer * buf, + GstJackRingBufferClass * g_class) +{ + buf->channels = -1; + buf->buffer_size = -1; + buf->sample_rate = -1; +} + +static void +gst_jack_ring_buffer_dispose (GObject * object) +{ + G_OBJECT_CLASS (ring_parent_class)->dispose (object); +} + +static void +gst_jack_ring_buffer_finalize (GObject * object) +{ + GstJackRingBuffer *ringbuffer; + ringbuffer = GST_JACK_RING_BUFFER_CAST (object); + G_OBJECT_CLASS (ring_parent_class)->finalize (object); +} + +/* the _open_device method should make a connection with the server +*/ +static gboolean +gst_jack_ring_buffer_open_device (GstRingBuffer * buf) +{ + GstJackAudioSrc *src; + jack_status_t status = 0; + const gchar *name; + + src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf)); + + GST_DEBUG_OBJECT (src, "open"); + + name = g_get_application_name (); + if (!name) + name = "GStreamer"; + + src->client = gst_jack_audio_client_new (name, src->server, + GST_JACK_CLIENT_SOURCE, + jack_shutdown_cb, + jack_process_cb, jack_buffer_size_cb, jack_sample_rate_cb, buf, &status); + if (src->client == NULL) + goto could_not_open; + + GST_DEBUG_OBJECT (src, "opened"); + + return TRUE; + + /* ERRORS */ +could_not_open: + { + if (status & JackServerFailed) { + GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, + (NULL), ("Cannot connect to the Jack server (status %d)", status)); + } else { + GST_ELEMENT_ERROR (src, RESOURCE, OPEN_WRITE, + (NULL), ("Jack client open error (status %d)", status)); + } + return FALSE; + } +} + +/* close the connection with the server +*/ +static gboolean +gst_jack_ring_buffer_close_device (GstRingBuffer * buf) +{ + GstJackAudioSrc *src; + + src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf)); + + GST_DEBUG_OBJECT (src, "close"); + + gst_jack_audio_src_free_channels (src); + gst_jack_audio_client_free (src->client); + src->client = NULL; + + return TRUE; +} + + +/* allocate a buffer and setup resources to process the audio samples of + * the format as specified in @spec. + * + * We allocate N jack ports, one for each channel. If we are asked to + * automatically make a connection with physical ports, we connect as many + * ports as there are physical ports, leaving leftover ports unconnected. + * + * It is assumed that samplerate and number of channels are acceptable since our + * getcaps method will always provide correct values. If unacceptable caps are + * received for some reason, we fail here. + */ +static gboolean +gst_jack_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec) +{ + GstJackAudioSrc *src; + GstJackRingBuffer *abuf; + const char **ports; + gint sample_rate, buffer_size; + gint i, channels, res; + jack_client_t *client; + + src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf)); + abuf = GST_JACK_RING_BUFFER_CAST (buf); + + GST_DEBUG_OBJECT (src, "acquire"); + + client = gst_jack_audio_client_get_client (src->client); + + /* sample rate must be that of the server */ + sample_rate = jack_get_sample_rate (client); + if (sample_rate != spec->rate) + goto wrong_samplerate; + + channels = spec->channels; + + if (!gst_jack_audio_src_allocate_channels (src, channels)) + goto out_of_ports; + + buffer_size = jack_get_buffer_size (client); + + /* the segment size in bytes, this is large enough to hold a buffer of 32bit floats + * for all channels */ + spec->segsize = buffer_size * sizeof (gfloat) * channels; + spec->latency_time = gst_util_uint64_scale (spec->segsize, + (GST_SECOND / GST_USECOND), spec->rate * spec->bytes_per_sample); + /* segtotal based on buffer-time latency */ + spec->segtotal = spec->buffer_time / spec->latency_time; + + GST_DEBUG_OBJECT (src, "segsize %d, segtotal %d", spec->segsize, + spec->segtotal); + + /* allocate the ringbuffer memory now */ + buf->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize); + memset (GST_BUFFER_DATA (buf->data), 0, GST_BUFFER_SIZE (buf->data)); + + if ((res = gst_jack_audio_client_set_active (src->client, TRUE))) + goto could_not_activate; + + /* if we need to automatically connect the ports, do so now. We must do this + * after activating the client. */ + if (src->connect == GST_JACK_CONNECT_AUTO) { + /* find all the physical output ports. A physical output port is a port + * associated with a hardware device. Someone needs connect to a physical + * port in order to capture something. */ + ports = + jack_get_ports (client, NULL, NULL, + JackPortIsPhysical | JackPortIsOutput); + if (ports == NULL) { + /* no ports? fine then we don't do anything except for posting a warning + * message. */ + GST_ELEMENT_WARNING (src, RESOURCE, NOT_FOUND, (NULL), + ("No physical output ports found, leaving ports unconnected")); + goto done; + } + + for (i = 0; i < channels; i++) { + /* stop when all output ports are exhausted */ + if (ports[i] == NULL) { + /* post a warning that we could not connect all ports */ + GST_ELEMENT_WARNING (src, RESOURCE, NOT_FOUND, (NULL), + ("No more physical ports, leaving some ports unconnected")); + break; + } + GST_DEBUG_OBJECT (src, "try connecting to %s", + jack_port_name (src->ports[i])); + /* connect the physical port to a port */ + + res = jack_connect (client, ports[i], jack_port_name (src->ports[i])); + g_print ("connecting to %s\n", jack_port_name (src->ports[i])); + if (res != 0 && res != EEXIST) + goto cannot_connect; + } + free (ports); + } +done: + + abuf->sample_rate = sample_rate; + abuf->buffer_size = buffer_size; + abuf->channels = spec->channels; + + return TRUE; + + /* ERRORS */ +wrong_samplerate: + { + GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL), + ("Wrong samplerate, server is running at %d and we received %d", + sample_rate, spec->rate)); + return FALSE; + } +out_of_ports: + { + GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL), + ("Cannot allocate more Jack ports")); + return FALSE; + } +could_not_activate: + { + GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL), + ("Could not activate client (%d:%s)", res, g_strerror (res))); + return FALSE; + } +cannot_connect: + { + GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL), + ("Could not connect input ports to physical ports (%d:%s)", + res, g_strerror (res))); + free (ports); + return FALSE; + } +} + +/* function is called with LOCK */ +static gboolean +gst_jack_ring_buffer_release (GstRingBuffer * buf) +{ + GstJackAudioSrc *src; + GstJackRingBuffer *abuf; + gint res; + + abuf = GST_JACK_RING_BUFFER_CAST (buf); + src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf)); + + GST_DEBUG_OBJECT (src, "release"); + + if ((res = gst_jack_audio_client_set_active (src->client, FALSE))) { + /* we only warn, this means the server is probably shut down and the client + * is gone anyway. */ + GST_ELEMENT_WARNING (src, RESOURCE, CLOSE, (NULL), + ("Could not deactivate Jack client (%d)", res)); + } + + abuf->channels = -1; + abuf->buffer_size = -1; + abuf->sample_rate = -1; + + /* free the buffer */ + gst_buffer_unref (buf->data); + buf->data = NULL; + + return TRUE; +} + +static gboolean +gst_jack_ring_buffer_start (GstRingBuffer * buf) +{ + GstJackAudioSrc *src; + + src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf)); + + GST_DEBUG_OBJECT (src, "start"); + + return TRUE; +} + +static gboolean +gst_jack_ring_buffer_pause (GstRingBuffer * buf) +{ + GstJackAudioSrc *src; + + src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf)); + + GST_DEBUG_OBJECT (src, "pause"); + + return TRUE; +} + +static gboolean +gst_jack_ring_buffer_stop (GstRingBuffer * buf) +{ + GstJackAudioSrc *src; + + src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf)); + + GST_DEBUG_OBJECT (src, "stop"); + + return TRUE; +} + +static guint +gst_jack_ring_buffer_delay (GstRingBuffer * buf) +{ + GstJackAudioSrc *src; + guint res = 0; + + src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf)); + + GST_DEBUG_OBJECT (src, "delay %u", res); + + return res; +} + +/* Audiosrc signals and args */ +enum +{ + /* FILL ME */ + LAST_SIGNAL +}; + +#define DEFAULT_PROP_CONNECT GST_JACK_CONNECT_AUTO +#define DEFAULT_PROP_SERVER NULL + +enum +{ + PROP_0, + PROP_CONNECT, + PROP_SERVER, + PROP_LAST +}; + + +/* the capabilities of the inputs and outputs. + * + * describe the real formats here. + */ + +static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/x-raw-float, " + "endianness = (int) { " G_STRINGIFY (G_BYTE_ORDER) " }, " + "width = (int) 32, " + "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]") + ); + +#define _do_init(bla) \ + GST_DEBUG_CATEGORY_INIT(gst_jackaudiosrc_debug, "jacksrc", 0, "jacksrc element"); + +GST_BOILERPLATE_FULL (GstJackAudioSrc, gst_jackaudiosrc, GstBaseAudioSrc, + GST_TYPE_BASE_AUDIO_SRC, _do_init); + +static void gst_jackaudiosrc_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec); +static void gst_jackaudiosrc_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec); + +static GstCaps *gst_jackaudiosrc_getcaps (GstBaseSrc * bsrc); +static GstRingBuffer *gst_jackaudiosrc_create_ringbuffer (GstBaseAudioSrc * + src); + +/* GObject vmethod implementations */ + +static void +gst_jackaudiosrc_base_init (gpointer gclass) +{ + static GstElementDetails gst_jackaudiosrc_details = { + "Audio Source (Jack)", + "Source/Audio", + "Input from Jack", + "Tristan Matthews " + }; + GstElementClass *element_class = GST_ELEMENT_CLASS (gclass); + + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&src_factory)); + gst_element_class_set_details (element_class, &gst_jackaudiosrc_details); +} + +/* initialize the jackaudiosrc's class */ +static void +gst_jackaudiosrc_class_init (GstJackAudioSrcClass * klass) +{ + GObjectClass *gobject_class; + GstElementClass *gstelement_class; + GstBaseSrcClass *gstbasesrc_class; + GstBaseAudioSrcClass *gstbaseaudiosrc_class; + + gobject_class = (GObjectClass *) klass; + gstelement_class = (GstElementClass *) klass; + + gstbasesrc_class = (GstBaseSrcClass *) klass; + gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass; + + gobject_class->set_property = + GST_DEBUG_FUNCPTR (gst_jackaudiosrc_set_property); + gobject_class->get_property = + GST_DEBUG_FUNCPTR (gst_jackaudiosrc_get_property); + + g_object_class_install_property (gobject_class, PROP_CONNECT, + g_param_spec_enum ("connect", "Connect", + "Specify how the input ports will be connected", + GST_TYPE_JACK_CONNECT, DEFAULT_PROP_CONNECT, G_PARAM_READWRITE)); + + g_object_class_install_property (gobject_class, PROP_SERVER, + g_param_spec_string ("server", "Server", + "The Jack server to connect to (NULL = default)", + DEFAULT_PROP_SERVER, G_PARAM_READWRITE)); + + gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_jackaudiosrc_getcaps); + gstbaseaudiosrc_class->create_ringbuffer = + GST_DEBUG_FUNCPTR (gst_jackaudiosrc_create_ringbuffer); + + /* ref class from a thread-safe context to work around missing bit of + * thread-safety in GObject */ + g_type_class_ref (GST_TYPE_JACK_RING_BUFFER); + + gst_jack_audio_client_init (); +} + +/* initialize the new element + * instantiate pads and add them to element + * set pad calback functions + * initialize instance structure + */ +static void +gst_jackaudiosrc_init (GstJackAudioSrc * src, GstJackAudioSrcClass * gclass) +{ + //gst_base_src_set_live(GST_BASE_SRC (src), TRUE); + src->connect = DEFAULT_PROP_CONNECT; + src->server = g_strdup (DEFAULT_PROP_SERVER); + src->ports = NULL; + src->port_count = 0; +} + +static void +gst_jackaudiosrc_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec) +{ + GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object); + + switch (prop_id) { + case PROP_CONNECT: + src->connect = g_value_get_enum (value); + break; + case PROP_SERVER: + g_free (src->server); + src->server = g_value_dup_string (value); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static void +gst_jackaudiosrc_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec) +{ + GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object); + + switch (prop_id) { + case PROP_CONNECT: + g_value_set_enum (value, src->connect); + break; + case PROP_SERVER: + g_value_set_string (value, src->server); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static GstCaps * +gst_jackaudiosrc_getcaps (GstBaseSrc * bsrc) +{ + GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (bsrc); + const char **ports; + gint min, max; + gint rate; + jack_client_t *client; + + if (src->client == NULL) + goto no_client; + + client = gst_jack_audio_client_get_client (src->client); + + if (src->connect == GST_JACK_CONNECT_AUTO) { + /* get a port count, this is the number of channels we can automatically + * connect. */ + ports = jack_get_ports (client, NULL, NULL, + JackPortIsPhysical | JackPortIsOutput); + max = 0; + if (ports != NULL) { + for (; ports[max]; max++); + + free (ports); + } else + max = 0; + } else { + /* we allow any number of pads, something else is going to connect the + * pads. */ + max = G_MAXINT; + } + min = MIN (1, max); + + rate = jack_get_sample_rate (client); + + GST_DEBUG_OBJECT (src, "got %d-%d ports, samplerate: %d", min, max, rate); + + if (!src->caps) { + src->caps = gst_caps_new_simple ("audio/x-raw-float", + "endianness", G_TYPE_INT, G_BYTE_ORDER, + "width", G_TYPE_INT, 32, + "rate", G_TYPE_INT, rate, + "channels", GST_TYPE_INT_RANGE, min, max, NULL); + } + GST_INFO_OBJECT (src, "returning caps %" GST_PTR_FORMAT, src->caps); + + return gst_caps_ref (src->caps); + + /* ERRORS */ +no_client: + { + GST_DEBUG_OBJECT (src, "device not open, using template caps"); + /* base class will get template caps for us when we return NULL */ + return NULL; + } +} + +static GstRingBuffer * +gst_jackaudiosrc_create_ringbuffer (GstBaseAudioSrc * src) +{ + GstRingBuffer *buffer; + + buffer = g_object_new (GST_TYPE_JACK_RING_BUFFER, NULL); + GST_DEBUG_OBJECT (src, "created ringbuffer @%p", buffer); + + return buffer; +} diff --git a/ext/jack/gstjackaudiosrc.h b/ext/jack/gstjackaudiosrc.h new file mode 100644 index 0000000000..69230977d4 --- /dev/null +++ b/ext/jack/gstjackaudiosrc.h @@ -0,0 +1,94 @@ +/* GStreamer + * Copyright (C) 2008 Tristan Matthews + * + * Permission is hereby granted, free of charge, to any person obtaining a + * copy of this software and associated documentation files (the "Software"), + * to deal in the Software without restriction, including without limitation + * the rights to use, copy, modify, merge, publish, distribute, sublicense, + * and/or sell copies of the Software, and to permit persons to whom the + * Software is furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE + * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING + * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER + * DEALINGS IN THE SOFTWARE. + * + * Alternatively, the contents of this file may be used under the + * GNU Lesser General Public License Version 2.1 (the "LGPL"), in + * which case the following provisions apply instead of the ones + * mentioned above: + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifndef __GST_JACK_AUDIO_SRC_H__ +#define __GST_JACK_AUDIO_SRC_H__ + +#include + +#include +#include + +#include "gstjackaudioclient.h" +#include "gstjack.h" + +G_BEGIN_DECLS + +#define GST_TYPE_JACK_AUDIO_SRC (gst_jackaudiosrc_get_type()) +#define GST_JACK_AUDIO_SRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_JACK_AUDIO_SRC,GstJackAudioSrc)) +#define GST_JACK_AUDIO_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_JACK_AUDIO_SRC,GstJackAudioSrcClass)) +#define GST_IS_JACK_AUDIO_SRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_JACK_AUDIO_SRC)) +#define GST_IS_JACK_AUDIO_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_JACK_AUDIO_SRC)) + +typedef struct _GstJackAudioSrc GstJackAudioSrc; +typedef struct _GstJackAudioSrcClass GstJackAudioSrcClass; + +struct _GstJackAudioSrc +{ + GstBaseAudioSrc src; + + /*< private >*/ + /* cached caps */ + GstCaps *caps; + + /* properties */ + GstJackConnect connect; + gchar *server; + + /* our client */ + GstJackAudioClient *client; + + /* our ports */ + jack_port_t **ports; + int port_count; +}; + +struct _GstJackAudioSrcClass +{ + GstBaseAudioSrcClass parent_class; +}; + +GType gst_jackaudiosrc_get_type (void); + +G_END_DECLS + +#endif /* __GST_JACK_AUDIO_SRC_H__ */ diff --git a/ext/jack/gstjackringbuffer.h b/ext/jack/gstjackringbuffer.h new file mode 100644 index 0000000000..dfe21b1a9c --- /dev/null +++ b/ext/jack/gstjackringbuffer.h @@ -0,0 +1,90 @@ +/* + * GStreamer + * Copyright (C) 2006 Wim Taymans + * Copyright (C) 2008 Tristan Matthews + * + * Permission is hereby granted, free of charge, to any person obtaining a + * copy of this software and associated documentation files (the "Software"), + * to deal in the Software without restriction, including without limitation + * the rights to use, copy, modify, merge, publish, distribute, sublicense, + * and/or sell copies of the Software, and to permit persons to whom the + * Software is furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE + * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING + * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER + * DEALINGS IN THE SOFTWARE. + * + * Alternatively, the contents of this file may be used under the + * GNU Lesser General Public License Version 2.1 (the "LGPL"), in + * which case the following provisions apply instead of the ones + * mentioned above: + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifndef __GST_JACK_RING_BUFFER_H__ +#define __GST_JACK_RING_BUFFER_H__ + +#define GST_TYPE_JACK_RING_BUFFER (gst_jack_ring_buffer_get_type()) +#define GST_JACK_RING_BUFFER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_JACK_RING_BUFFER,GstJackRingBuffer)) +#define GST_JACK_RING_BUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_JACK_RING_BUFFER,GstJackRingBufferClass)) +#define GST_JACK_RING_BUFFER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_JACK_RING_BUFFER,GstJackRingBufferClass)) +#define GST_JACK_RING_BUFFER_CAST(obj) ((GstJackRingBuffer *)obj) +#define GST_IS_JACK_RING_BUFFER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_JACK_RING_BUFFER)) +#define GST_IS_JACK_RING_BUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_JACK_RING_BUFFER)) + +typedef struct _GstJackRingBuffer GstJackRingBuffer; +typedef struct _GstJackRingBufferClass GstJackRingBufferClass; + +struct _GstJackRingBuffer +{ + GstRingBuffer object; + + gint sample_rate; + gint buffer_size; + gint channels; +}; + +struct _GstJackRingBufferClass +{ + GstRingBufferClass parent_class; +}; + +static void gst_jack_ring_buffer_class_init(GstJackRingBufferClass * klass); +static void gst_jack_ring_buffer_init(GstJackRingBuffer * ringbuffer, + GstJackRingBufferClass * klass); +static void gst_jack_ring_buffer_dispose(GObject * object); +static void gst_jack_ring_buffer_finalize(GObject * object); + +static GstRingBufferClass *ring_parent_class = NULL; + +static gboolean gst_jack_ring_buffer_open_device(GstRingBuffer * buf); +static gboolean gst_jack_ring_buffer_close_device(GstRingBuffer * buf); +static gboolean gst_jack_ring_buffer_acquire(GstRingBuffer * buf,GstRingBufferSpec * spec); +static gboolean gst_jack_ring_buffer_release(GstRingBuffer * buf); +static gboolean gst_jack_ring_buffer_start(GstRingBuffer * buf); +static gboolean gst_jack_ring_buffer_pause(GstRingBuffer * buf); +static gboolean gst_jack_ring_buffer_stop(GstRingBuffer * buf); +static guint gst_jack_ring_buffer_delay(GstRingBuffer * buf); + +#endif From 37f51bb59dc64667c539ce0834f50d7d1b25edff Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Thu, 7 Aug 2008 09:09:44 +0000 Subject: [PATCH 180/351] docs/plugins/: docs/plugins/inspect/plugin-jack.xml Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * docs/plugins/gst-plugins-bad-plugins.args: * docs/plugins/gst-plugins-bad-plugins.hierarchy: * docs/plugins/gst-plugins-bad-plugins.interfaces: * docs/plugins/gst-plugins-bad-plugins.prerequisites: * docs/plugins/inspect/plugin-jack.xml Add new element to docs. * ext/jack/gstjack.h Add missing file. * ext/jack/gstjackaudiosrc.c: * ext/jack/gstjackaudiosrc.h: Rename jackaudiosrc to jack_audio_src. --- ext/jack/gstjack.h | 48 ++++++++++++++++++++++++++++++++++++++ ext/jack/gstjackaudiosrc.c | 12 +++++----- ext/jack/gstjackaudiosrc.h | 5 ++-- 3 files changed, 57 insertions(+), 8 deletions(-) create mode 100644 ext/jack/gstjack.h diff --git a/ext/jack/gstjack.h b/ext/jack/gstjack.h new file mode 100644 index 0000000000..585b72ea8d --- /dev/null +++ b/ext/jack/gstjack.h @@ -0,0 +1,48 @@ +/* GStreamer + * Copyright (C) 2006 Wim Taymans + * + * gstjack.h: + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifndef _GST_JACK_H_ +#define _GST_JACK_H_ + + +/** + * GstJackConnect: + * @GST_JACK_CONNECT_NONE: Don't automatically connect to physical ports. + * In this mode, the element will accept any number of input channels and will + * create (but not connect) an output port for each channel. + * @GST_JACK_CONNECT_AUTO: In this mode, the element will try to connect each + * output port to a random physical jack input pin. The sink will + * expose the number of physical channels on its pad caps. + * + * Specify how the output ports will be connected. + */ + +typedef enum { + GST_JACK_CONNECT_NONE, + GST_JACK_CONNECT_AUTO +} GstJackConnect; + +typedef jack_default_audio_sample_t sample_t; + +#define GST_TYPE_JACK_CONNECT (gst_jack_connect_get_type()) +GType gst_jack_connect_get_type(); + +#endif // _GST_JACK_H_ diff --git a/ext/jack/gstjackaudiosrc.c b/ext/jack/gstjackaudiosrc.c index d41b62ffd0..7be5ee3f20 100644 --- a/ext/jack/gstjackaudiosrc.c +++ b/ext/jack/gstjackaudiosrc.c @@ -81,7 +81,7 @@ #include "gstjackaudiosrc.h" #include "gstjackringbuffer.h" -GST_DEBUG_CATEGORY_STATIC (gst_jackaudiosrc_debug); +GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_src_debug); #define GST_CAT_DEFAULT gst_jackaudiosrc_debug static gboolean @@ -644,7 +644,7 @@ static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src", ); #define _do_init(bla) \ - GST_DEBUG_CATEGORY_INIT(gst_jackaudiosrc_debug, "jacksrc", 0, "jacksrc element"); + GST_DEBUG_CATEGORY_INIT(gst_jack_audio_src_debug, "jacksrc", 0, "jacksrc element"); GST_BOILERPLATE_FULL (GstJackAudioSrc, gst_jackaudiosrc, GstBaseAudioSrc, GST_TYPE_BASE_AUDIO_SRC, _do_init); @@ -692,9 +692,9 @@ gst_jackaudiosrc_class_init (GstJackAudioSrcClass * klass) gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass; gobject_class->set_property = - GST_DEBUG_FUNCPTR (gst_jackaudiosrc_set_property); + GST_DEBUG_FUNCPTR (gst_jack_audio_src_set_property); gobject_class->get_property = - GST_DEBUG_FUNCPTR (gst_jackaudiosrc_get_property); + GST_DEBUG_FUNCPTR (gst_jack_audio_src_get_property); g_object_class_install_property (gobject_class, PROP_CONNECT, g_param_spec_enum ("connect", "Connect", @@ -706,9 +706,9 @@ gst_jackaudiosrc_class_init (GstJackAudioSrcClass * klass) "The Jack server to connect to (NULL = default)", DEFAULT_PROP_SERVER, G_PARAM_READWRITE)); - gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_jackaudiosrc_getcaps); + gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_jack_audio_src_getcaps); gstbaseaudiosrc_class->create_ringbuffer = - GST_DEBUG_FUNCPTR (gst_jackaudiosrc_create_ringbuffer); + GST_DEBUG_FUNCPTR (gst_jack_audio_src_create_ringbuffer); /* ref class from a thread-safe context to work around missing bit of * thread-safety in GObject */ diff --git a/ext/jack/gstjackaudiosrc.h b/ext/jack/gstjackaudiosrc.h index 69230977d4..b8d451aebb 100644 --- a/ext/jack/gstjackaudiosrc.h +++ b/ext/jack/gstjackaudiosrc.h @@ -53,9 +53,10 @@ G_BEGIN_DECLS -#define GST_TYPE_JACK_AUDIO_SRC (gst_jackaudiosrc_get_type()) +#define GST_TYPE_JACK_AUDIO_SRC (gst_jack_audio_src_get_type()) #define GST_JACK_AUDIO_SRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_JACK_AUDIO_SRC,GstJackAudioSrc)) #define GST_JACK_AUDIO_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_JACK_AUDIO_SRC,GstJackAudioSrcClass)) +#define GST_JACK_AUDIO_SRC_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_JACK_AUDIO_SRC,GstJackAudioSrcClass)) #define GST_IS_JACK_AUDIO_SRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_JACK_AUDIO_SRC)) #define GST_IS_JACK_AUDIO_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_JACK_AUDIO_SRC)) @@ -87,7 +88,7 @@ struct _GstJackAudioSrcClass GstBaseAudioSrcClass parent_class; }; -GType gst_jackaudiosrc_get_type (void); +GType gst_jack_audio_src_get_type (void); G_END_DECLS From e0d25d42db70a4f87080ed8b993b23a37547fec7 Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Thu, 7 Aug 2008 13:15:21 +0000 Subject: [PATCH 181/351] ext/jack/gstjackaudiosrc.c: Try committing this once again. Now properly renamed. Original commit message from CVS: * ext/jack/gstjackaudiosrc.c: Try committing this once again. Now properly renamed. --- ext/jack/gstjackaudiosrc.c | 34 +++++++++++++++++----------------- 1 file changed, 17 insertions(+), 17 deletions(-) diff --git a/ext/jack/gstjackaudiosrc.c b/ext/jack/gstjackaudiosrc.c index 7be5ee3f20..6462d239f2 100644 --- a/ext/jack/gstjackaudiosrc.c +++ b/ext/jack/gstjackaudiosrc.c @@ -41,7 +41,7 @@ */ /** - * SECTION:element-jackaudiosrc + * SECTION:element-jack_audio_src * @see_also: #GstBaseAudioSrc, #GstRingBuffer * * A Src that inputs data from Jack ports. @@ -82,7 +82,7 @@ #include "gstjackringbuffer.h" GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_src_debug); -#define GST_CAT_DEFAULT gst_jackaudiosrc_debug +#define GST_CAT_DEFAULT gst_jack_audio_src_debug static gboolean gst_jack_audio_src_allocate_channels (GstJackAudioSrc * src, gint channels) @@ -646,24 +646,24 @@ static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src", #define _do_init(bla) \ GST_DEBUG_CATEGORY_INIT(gst_jack_audio_src_debug, "jacksrc", 0, "jacksrc element"); -GST_BOILERPLATE_FULL (GstJackAudioSrc, gst_jackaudiosrc, GstBaseAudioSrc, +GST_BOILERPLATE_FULL (GstJackAudioSrc, gst_jack_audio_src, GstBaseAudioSrc, GST_TYPE_BASE_AUDIO_SRC, _do_init); -static void gst_jackaudiosrc_set_property (GObject * object, guint prop_id, +static void gst_jack_audio_src_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); -static void gst_jackaudiosrc_get_property (GObject * object, guint prop_id, +static void gst_jack_audio_src_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); -static GstCaps *gst_jackaudiosrc_getcaps (GstBaseSrc * bsrc); -static GstRingBuffer *gst_jackaudiosrc_create_ringbuffer (GstBaseAudioSrc * +static GstCaps *gst_jack_audio_src_getcaps (GstBaseSrc * bsrc); +static GstRingBuffer *gst_jack_audio_src_create_ringbuffer (GstBaseAudioSrc * src); /* GObject vmethod implementations */ static void -gst_jackaudiosrc_base_init (gpointer gclass) +gst_jack_audio_src_base_init (gpointer gclass) { - static GstElementDetails gst_jackaudiosrc_details = { + static GstElementDetails gst_jack_audio_src_details = { "Audio Source (Jack)", "Source/Audio", "Input from Jack", @@ -673,12 +673,12 @@ gst_jackaudiosrc_base_init (gpointer gclass) gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&src_factory)); - gst_element_class_set_details (element_class, &gst_jackaudiosrc_details); + gst_element_class_set_details (element_class, &gst_jack_audio_src_details); } -/* initialize the jackaudiosrc's class */ +/* initialize the jack_audio_src's class */ static void -gst_jackaudiosrc_class_init (GstJackAudioSrcClass * klass) +gst_jack_audio_src_class_init (GstJackAudioSrcClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; @@ -723,7 +723,7 @@ gst_jackaudiosrc_class_init (GstJackAudioSrcClass * klass) * initialize instance structure */ static void -gst_jackaudiosrc_init (GstJackAudioSrc * src, GstJackAudioSrcClass * gclass) +gst_jack_audio_src_init (GstJackAudioSrc * src, GstJackAudioSrcClass * gclass) { //gst_base_src_set_live(GST_BASE_SRC (src), TRUE); src->connect = DEFAULT_PROP_CONNECT; @@ -733,7 +733,7 @@ gst_jackaudiosrc_init (GstJackAudioSrc * src, GstJackAudioSrcClass * gclass) } static void -gst_jackaudiosrc_set_property (GObject * object, guint prop_id, +gst_jack_audio_src_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object); @@ -753,7 +753,7 @@ gst_jackaudiosrc_set_property (GObject * object, guint prop_id, } static void -gst_jackaudiosrc_get_property (GObject * object, guint prop_id, +gst_jack_audio_src_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object); @@ -772,7 +772,7 @@ gst_jackaudiosrc_get_property (GObject * object, guint prop_id, } static GstCaps * -gst_jackaudiosrc_getcaps (GstBaseSrc * bsrc) +gst_jack_audio_src_getcaps (GstBaseSrc * bsrc) { GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (bsrc); const char **ports; @@ -829,7 +829,7 @@ no_client: } static GstRingBuffer * -gst_jackaudiosrc_create_ringbuffer (GstBaseAudioSrc * src) +gst_jack_audio_src_create_ringbuffer (GstBaseAudioSrc * src) { GstRingBuffer *buffer; From 051dba6c01de5c7cf9443eb6a3fc06a220e9ccdf Mon Sep 17 00:00:00 2001 From: Jan Schmidt Date: Wed, 17 Sep 2008 13:59:21 +0000 Subject: [PATCH 182/351] Fix compiler warnings on OS/X Original commit message from CVS: * ext/jack/gstjackaudiosink.c: (jack_process_cb): * gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew): Fix compiler warnings on OS/X --- ext/jack/gstjackaudiosink.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c index ec257deb17..265ec68c70 100644 --- a/ext/jack/gstjackaudiosink.c +++ b/ext/jack/gstjackaudiosink.c @@ -243,7 +243,7 @@ jack_process_cb (jack_nframes_t nframes, void *arg) wrong_size: { GST_ERROR_OBJECT (sink, "nbytes (%d) != flen (%d)", - nframes * sizeof (sample_t), flen); + (gint) (nframes * sizeof (sample_t)), flen); return 1; } } From 7ba1bcb6aadd78ab2dfb1fd866ad891a1425d2d4 Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Tue, 4 Nov 2008 12:42:30 +0000 Subject: [PATCH 183/351] Don't install static libs for plugins. Fixes #550851 for -bad. Original commit message from CVS: * ext/alsaspdif/Makefile.am: * ext/amrwb/Makefile.am: * ext/apexsink/Makefile.am: * ext/arts/Makefile.am: * ext/artsd/Makefile.am: * ext/audiofile/Makefile.am: * ext/audioresample/Makefile.am: * ext/bz2/Makefile.am: * ext/cdaudio/Makefile.am: * ext/celt/Makefile.am: * ext/dc1394/Makefile.am: * ext/dirac/Makefile.am: * ext/directfb/Makefile.am: * ext/divx/Makefile.am: * ext/dts/Makefile.am: * ext/faac/Makefile.am: * ext/faad/Makefile.am: * ext/gsm/Makefile.am: * ext/hermes/Makefile.am: * ext/ivorbis/Makefile.am: * ext/jack/Makefile.am: * ext/jp2k/Makefile.am: * ext/ladspa/Makefile.am: * ext/lcs/Makefile.am: * ext/libfame/Makefile.am: * ext/libmms/Makefile.am: * ext/metadata/Makefile.am: * ext/mpeg2enc/Makefile.am: * ext/mplex/Makefile.am: * ext/musepack/Makefile.am: * ext/musicbrainz/Makefile.am: * ext/mythtv/Makefile.am: * ext/nas/Makefile.am: * ext/neon/Makefile.am: * ext/ofa/Makefile.am: * ext/polyp/Makefile.am: * ext/resindvd/Makefile.am: * ext/sdl/Makefile.am: * ext/shout/Makefile.am: * ext/snapshot/Makefile.am: * ext/sndfile/Makefile.am: * ext/soundtouch/Makefile.am: * ext/spc/Makefile.am: * ext/swfdec/Makefile.am: * ext/tarkin/Makefile.am: * ext/theora/Makefile.am: * ext/timidity/Makefile.am: * ext/twolame/Makefile.am: * ext/x264/Makefile.am: * ext/xine/Makefile.am: * ext/xvid/Makefile.am: * gst-libs/gst/app/Makefile.am: * gst-libs/gst/dshow/Makefile.am: * gst/aiffparse/Makefile.am: * gst/app/Makefile.am: * gst/audiobuffer/Makefile.am: * gst/bayer/Makefile.am: * gst/cdxaparse/Makefile.am: * gst/chart/Makefile.am: * gst/colorspace/Makefile.am: * gst/dccp/Makefile.am: * gst/deinterlace/Makefile.am: * gst/deinterlace2/Makefile.am: * gst/dvdspu/Makefile.am: * gst/festival/Makefile.am: * gst/filter/Makefile.am: * gst/flacparse/Makefile.am: * gst/flv/Makefile.am: * gst/games/Makefile.am: * gst/h264parse/Makefile.am: * gst/librfb/Makefile.am: * gst/mixmatrix/Makefile.am: * gst/modplug/Makefile.am: * gst/mpeg1sys/Makefile.am: * gst/mpeg4videoparse/Makefile.am: * gst/mpegdemux/Makefile.am: * gst/mpegtsmux/Makefile.am: * gst/mpegvideoparse/Makefile.am: * gst/mve/Makefile.am: * gst/nsf/Makefile.am: * gst/nuvdemux/Makefile.am: * gst/overlay/Makefile.am: * gst/passthrough/Makefile.am: * gst/pcapparse/Makefile.am: * gst/playondemand/Makefile.am: * gst/rawparse/Makefile.am: * gst/real/Makefile.am: * gst/rtjpeg/Makefile.am: * gst/rtpmanager/Makefile.am: * gst/scaletempo/Makefile.am: * gst/sdp/Makefile.am: * gst/selector/Makefile.am: * gst/smooth/Makefile.am: * gst/smoothwave/Makefile.am: * gst/speed/Makefile.am: * gst/speexresample/Makefile.am: * gst/stereo/Makefile.am: * gst/subenc/Makefile.am: * gst/tta/Makefile.am: * gst/vbidec/Makefile.am: * gst/videodrop/Makefile.am: * gst/videosignal/Makefile.am: * gst/virtualdub/Makefile.am: * gst/vmnc/Makefile.am: * gst/y4m/Makefile.am: * sys/acmenc/Makefile.am: * sys/cdrom/Makefile.am: * sys/dshowdecwrapper/Makefile.am: * sys/dshowsrcwrapper/Makefile.am: * sys/dvb/Makefile.am: * sys/dxr3/Makefile.am: * sys/fbdev/Makefile.am: * sys/oss4/Makefile.am: * sys/qcam/Makefile.am: * sys/qtwrapper/Makefile.am: * sys/vcd/Makefile.am: * sys/wininet/Makefile.am: * win32/common/config.h: Don't install static libs for plugins. Fixes #550851 for -bad. --- ext/jack/Makefile.am | 1 + 1 file changed, 1 insertion(+) diff --git a/ext/jack/Makefile.am b/ext/jack/Makefile.am index abcc39a597..5e9c8af61d 100644 --- a/ext/jack/Makefile.am +++ b/ext/jack/Makefile.am @@ -5,6 +5,7 @@ libgstjack_la_SOURCES = gstjack.c gstjackaudiosrc.c gstjackaudiosink.c gstjackau libgstjack_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) $(JACK_CFLAGS) libgstjack_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR) $(JACK_LIBS) libgstjack_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) +libgstjack_la_LIBTOOLFLAGS = --tag=disable-static noinst_HEADERS = gstjackaudiosrc.h gstjackaudiosink.h gstjackaudioclient.h gstjack.h gstjackringbuffer.h From fb03eb6bacb1e224ab4f89198ef6be17600229ad Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Sun, 23 Nov 2008 17:50:08 +0000 Subject: [PATCH 184/351] ext/jack/: Query port latencies for sink/src delays. Original commit message from CVS: * ext/jack/gstjackaudiosink.c: * ext/jack/gstjackaudiosrc.c: Query port latencies for sink/src delays. * ext/jack/gstjackbin.c: No printf please. --- ext/jack/gstjackaudiosink.c | 10 +++++++++- ext/jack/gstjackaudiosrc.c | 10 +++++++++- ext/jack/gstjackbin.c | 2 +- 3 files changed, 19 insertions(+), 3 deletions(-) diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c index 265ec68c70..228a229768 100644 --- a/ext/jack/gstjackaudiosink.c +++ b/ext/jack/gstjackaudiosink.c @@ -594,9 +594,17 @@ static guint gst_jack_ring_buffer_delay (GstRingBuffer * buf) { GstJackAudioSink *sink; - guint res = 0; + guint i, res = 0, latency; + jack_client_t *client; sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf)); + client = gst_jack_audio_client_get_client (sink->client); + + for (i = 0; i < sink->port_count; i++) { + latency = jack_port_get_total_latency (client, sink->ports[i]); + if (latency > res) + res = latency; + } GST_DEBUG_OBJECT (sink, "delay %u", res); diff --git a/ext/jack/gstjackaudiosrc.c b/ext/jack/gstjackaudiosrc.c index 6462d239f2..682a684afe 100644 --- a/ext/jack/gstjackaudiosrc.c +++ b/ext/jack/gstjackaudiosrc.c @@ -601,9 +601,17 @@ static guint gst_jack_ring_buffer_delay (GstRingBuffer * buf) { GstJackAudioSrc *src; - guint res = 0; + guint i, res = 0, latency; + jack_client_t *client; src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf)); + client = gst_jack_audio_client_get_client (src->client); + + for (i = 0; i < src->port_count; i++) { + latency = jack_port_get_total_latency (client, src->ports[i]); + if (latency > res) + res = latency; + } GST_DEBUG_OBJECT (src, "delay %u", res); diff --git a/ext/jack/gstjackbin.c b/ext/jack/gstjackbin.c index 7b219c7068..08fed371e0 100644 --- a/ext/jack/gstjackbin.c +++ b/ext/jack/gstjackbin.c @@ -337,7 +337,7 @@ static void shutdown (void *arg) { /* GstJackClient *client = (GstJackClient*) arg; */ - printf ("shutdown %p\n", arg); + JACK_DEBUG ("shutdown %p\n", arg); /* gst_element_set_state (GST_ELEMENT (client->manager), GST_STATE_READY); */ } From 7dde7eca2896e3049747150ed8f869f51fe5e43b Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Mon, 16 Mar 2009 11:21:02 +0100 Subject: [PATCH 185/351] jack: Add new connection mode Add a new connection mode to jacksrc and jacksink. In this new auto-force connection mode jack will create as many ports as requested/needed in the pipeline and will then connect as many physical ports as possible, possibly leaving some ports unconnected. Also get rid of some leftover g_print. Fixes #575284. --- ext/jack/gstjack.c | 3 +++ ext/jack/gstjack.h | 6 +++++- ext/jack/gstjackaudiosink.c | 3 ++- ext/jack/gstjackaudiosrc.c | 7 +++---- 4 files changed, 13 insertions(+), 6 deletions(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 96afd06e63..083025016d 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -33,6 +33,9 @@ gst_jack_connect_get_type (void) "Don't automatically connect ports to physical ports", "none"}, {GST_JACK_CONNECT_AUTO, "Automatically connect ports to physical ports", "auto"}, + {GST_JACK_CONNECT_AUTO_FORCED, + "Automatically connect ports to as many physical ports as possible", + "auto-forced"}, {0, NULL, NULL}, }; diff --git a/ext/jack/gstjack.h b/ext/jack/gstjack.h index 585b72ea8d..fc26b71384 100644 --- a/ext/jack/gstjack.h +++ b/ext/jack/gstjack.h @@ -31,13 +31,17 @@ * @GST_JACK_CONNECT_AUTO: In this mode, the element will try to connect each * output port to a random physical jack input pin. The sink will * expose the number of physical channels on its pad caps. + * @GST_JACK_CONNECT_AUTO_FORCED: In this mode, the element will try to connect each + * output port to a random physical jack input pin. The element will accept any number + * of input channels. * * Specify how the output ports will be connected. */ typedef enum { GST_JACK_CONNECT_NONE, - GST_JACK_CONNECT_AUTO + GST_JACK_CONNECT_AUTO, + GST_JACK_CONNECT_AUTO_FORCED } GstJackConnect; typedef jack_default_audio_sample_t sample_t; diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c index 228a229768..01383cbbf7 100644 --- a/ext/jack/gstjackaudiosink.c +++ b/ext/jack/gstjackaudiosink.c @@ -454,7 +454,8 @@ gst_jack_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec) /* if we need to automatically connect the ports, do so now. We must do this * after activating the client. */ - if (sink->connect == GST_JACK_CONNECT_AUTO) { + if (sink->connect == GST_JACK_CONNECT_AUTO + || sink->connect == GST_JACK_CONNECT_AUTO_FORCED) { /* find all the physical input ports. A physical input port is a port * associated with a hardware device. Someone needs connect to a physical * port in order to hear something. */ diff --git a/ext/jack/gstjackaudiosrc.c b/ext/jack/gstjackaudiosrc.c index 682a684afe..ca4bf73ff9 100644 --- a/ext/jack/gstjackaudiosrc.c +++ b/ext/jack/gstjackaudiosrc.c @@ -233,7 +233,6 @@ jack_process_cb (jack_nframes_t nframes, void *arg) /* the samples in the jack input buffers have to be interleaved into the * ringbuffer */ - for (i = 0; i < nframes; ++i) for (j = 0; j < channels; ++j) *data++ = buffers[j][i]; @@ -458,7 +457,8 @@ gst_jack_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec) /* if we need to automatically connect the ports, do so now. We must do this * after activating the client. */ - if (src->connect == GST_JACK_CONNECT_AUTO) { + if (src->connect == GST_JACK_CONNECT_AUTO + || src->connect == GST_JACK_CONNECT_AUTO_FORCED) { /* find all the physical output ports. A physical output port is a port * associated with a hardware device. Someone needs connect to a physical * port in order to capture something. */ @@ -483,10 +483,9 @@ gst_jack_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec) } GST_DEBUG_OBJECT (src, "try connecting to %s", jack_port_name (src->ports[i])); - /* connect the physical port to a port */ + /* connect the physical port to a port */ res = jack_connect (client, ports[i], jack_port_name (src->ports[i])); - g_print ("connecting to %s\n", jack_port_name (src->ports[i])); if (res != 0 && res != EEXIST) goto cannot_connect; } From 8a002a11219a9bf4a12f5c6450134fdb1e0ff0f4 Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Mon, 11 May 2009 16:12:54 +0300 Subject: [PATCH 186/351] jack: when stopping playback, do one more cycle to flush the port. Fixes #582167 The gst_jack_audio_client_set_active() flags the port as deactivating and uses a GCond to wait until the jack_process_cb() has run once more and cleared the flag. This way the client zero's the buffer. This happens if one manyally go to PAUSED and then to READY, while leting the mainloop run inbetween. --- ext/jack/gstjackaudioclient.c | 32 +++++++++++++++++++++++++++++--- ext/jack/gstjackaudiosink.c | 7 ++++--- 2 files changed, 33 insertions(+), 6 deletions(-) diff --git a/ext/jack/gstjackaudioclient.c b/ext/jack/gstjackaudioclient.c index 1aa1baf8a7..60e413819d 100644 --- a/ext/jack/gstjackaudioclient.c +++ b/ext/jack/gstjackaudioclient.c @@ -42,6 +42,7 @@ typedef struct { gint refcount; GMutex *lock; + GCond *flush_cond; /* id/server pair and the connection */ gchar *id; @@ -61,6 +62,7 @@ struct _GstJackAudioClient GstJackClientType type; gboolean active; + gboolean deactivate; void (*shutdown) (void *arg); JackProcessCallback process; @@ -92,15 +94,25 @@ jack_process_cb (jack_nframes_t nframes, void *arg) GstJackAudioClient *client = (GstJackAudioClient *) walk->data; /* only call active clients */ - if (client->active && client->process) + if ((client->active || client->deactivate) && client->process) { res = client->process (nframes, client->user_data); + if (client->deactivate) { + client->deactivate = FALSE; + g_cond_signal (conn->flush_cond); + } + } } for (walk = conn->sink_clients; walk; walk = g_list_next (walk)) { GstJackAudioClient *client = (GstJackAudioClient *) walk->data; /* only call active clients */ - if (client->active && client->process) + if ((client->active || client->deactivate) && client->process) { res = client->process (nframes, client->user_data); + if (client->deactivate) { + client->deactivate = FALSE; + g_cond_signal (conn->flush_cond); + } + } } g_mutex_unlock (conn->lock); @@ -203,6 +215,7 @@ gst_jack_audio_make_connection (const gchar * id, const gchar * server, conn = g_new (GstJackAudioConnection, 1); conn->refcount = 1; conn->lock = g_mutex_new (); + conn->flush_cond = g_cond_new (); conn->id = g_strdup (id); conn->server = g_strdup (server); conn->client = jclient; @@ -325,6 +338,7 @@ gst_jack_audio_unref_connection (GstJackAudioConnection * conn) /* free resources */ g_mutex_free (conn->lock); + g_cond_free (conn->flush_cond); g_free (conn->id); g_free (conn->server); g_free (conn); @@ -409,9 +423,11 @@ gst_jack_audio_client_new (const gchar * id, const gchar * server, if (conn == NULL) goto no_connection; + GST_INFO ("new client %s", id); + /* make new client using the connection */ client = g_new (GstJackAudioClient, 1); - client->active = FALSE; + client->active = client->deactivate = FALSE; client->conn = conn; client->type = type; client->shutdown = shutdown; @@ -446,6 +462,8 @@ gst_jack_audio_client_free (GstJackAudioClient * client) g_return_if_fail (client != NULL); + GST_INFO ("free client"); + conn = client->conn; /* remove from connection first so that it's not scheduled anymore after this @@ -492,6 +510,14 @@ gst_jack_audio_client_set_active (GstJackAudioClient * client, gboolean active) /* make sure that we are not dispatching the client */ g_mutex_lock (client->conn->lock); + if (client->active && !active) { + /* we need to process once more to flush the port */ + client->deactivate = TRUE; + + /* need to wait for process_cb run once more */ + while (client->deactivate) + g_cond_wait (client->conn->flush_cond, client->conn->lock); + } client->active = active; g_mutex_unlock (client->conn->lock); diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c index 01383cbbf7..58a699d0c6 100644 --- a/ext/jack/gstjackaudiosink.c +++ b/ext/jack/gstjackaudiosink.c @@ -213,8 +213,8 @@ jack_process_cb (jack_nframes_t nframes, void *arg) if (nframes * sizeof (sample_t) != flen) goto wrong_size; - GST_DEBUG ("copy %d frames: %p, %d bytes, %d channels", nframes, readptr, - flen, channels); + GST_DEBUG_OBJECT (sink, "copy %d frames: %p, %d bytes, %d channels", + nframes, readptr, flen, channels); data = (sample_t *) readptr; /* the samples in the ringbuffer have the channels interleaved, we need to @@ -231,6 +231,7 @@ jack_process_cb (jack_nframes_t nframes, void *arg) /* we wrote one segment */ gst_ring_buffer_advance (buf, 1); } else { + GST_DEBUG_OBJECT (sink, "write %d frames silence", nframes); /* We are not allowed to read from the ringbuffer, write silence to all * jack output buffers */ for (i = 0; i < channels; i++) { @@ -607,7 +608,7 @@ gst_jack_ring_buffer_delay (GstRingBuffer * buf) res = latency; } - GST_DEBUG_OBJECT (sink, "delay %u", res); + GST_LOG_OBJECT (sink, "delay %u", res); return res; } From 45fb781f9be8bcc6dc702436ea460e254e9aa433 Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Mon, 12 Oct 2009 00:51:27 +0300 Subject: [PATCH 187/351] jack: don't crash in ringbuffer with SIGFPE on small buffer-times Jack overrides user-specified latency-time with the one it gets from jack itself. It also needs to adjust buffer-time somewhat to avoid segtotal being 0 --- ext/jack/gstjackaudiosink.c | 18 ++++++++++++++---- 1 file changed, 14 insertions(+), 4 deletions(-) diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c index 58a699d0c6..4c48ab982e 100644 --- a/ext/jack/gstjackaudiosink.c +++ b/ext/jack/gstjackaudiosink.c @@ -440,11 +440,21 @@ gst_jack_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec) spec->segsize = buffer_size * sizeof (gfloat) * channels; spec->latency_time = gst_util_uint64_scale (spec->segsize, (GST_SECOND / GST_USECOND), spec->rate * spec->bytes_per_sample); - /* segtotal based on buffer-time latency */ - spec->segtotal = spec->buffer_time / spec->latency_time; + if (spec->latency_time < spec->buffer_time) { + /* segtotal based on buffer-time latency */ + spec->segtotal = spec->buffer_time / spec->latency_time; + } else { + spec->segtotal = 2; + spec->buffer_time = spec->latency_time * spec->segtotal; + } - GST_DEBUG_OBJECT (sink, "segsize %d, segtotal %d", spec->segsize, - spec->segtotal); + GST_DEBUG_OBJECT (sink, "buffer time: %" G_GINT64_FORMAT " usec", + spec->buffer_time); + GST_DEBUG_OBJECT (sink, "latency time: %" G_GINT64_FORMAT " usec", + spec->latency_time); + + GST_DEBUG_OBJECT (sink, "buffer_size %d, segsize %d, segtotal %d", + buffer_size, spec->segsize, spec->segtotal); /* allocate the ringbuffer memory now */ buf->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize); From f20c73684b3a980c9d4f425e9cc1a1e9e6cc99d6 Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Mon, 12 Oct 2009 09:06:37 +0300 Subject: [PATCH 188/351] jack: ensure segtotal is at least 2 Not only adjust buffer-time and avoid segtotal=0, but instead ensure segtotal is atleast 2. Do same change on jacksrc. We could also check the latency and buffer time configured by the client and adjust buffer-time so that we get to the same number of segments. --- ext/jack/gstjackaudiosink.c | 8 +++----- ext/jack/gstjackaudiosrc.c | 12 ++++++++++-- 2 files changed, 13 insertions(+), 7 deletions(-) diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c index 4c48ab982e..0e75f16d25 100644 --- a/ext/jack/gstjackaudiosink.c +++ b/ext/jack/gstjackaudiosink.c @@ -440,10 +440,9 @@ gst_jack_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec) spec->segsize = buffer_size * sizeof (gfloat) * channels; spec->latency_time = gst_util_uint64_scale (spec->segsize, (GST_SECOND / GST_USECOND), spec->rate * spec->bytes_per_sample); - if (spec->latency_time < spec->buffer_time) { - /* segtotal based on buffer-time latency */ - spec->segtotal = spec->buffer_time / spec->latency_time; - } else { + /* segtotal based on buffer-time latency */ + spec->segtotal = spec->buffer_time / spec->latency_time; + if (spec->segtotal < 2) { spec->segtotal = 2; spec->buffer_time = spec->latency_time * spec->segtotal; } @@ -452,7 +451,6 @@ gst_jack_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec) spec->buffer_time); GST_DEBUG_OBJECT (sink, "latency time: %" G_GINT64_FORMAT " usec", spec->latency_time); - GST_DEBUG_OBJECT (sink, "buffer_size %d, segsize %d, segtotal %d", buffer_size, spec->segsize, spec->segtotal); diff --git a/ext/jack/gstjackaudiosrc.c b/ext/jack/gstjackaudiosrc.c index ca4bf73ff9..211e1d2e6d 100644 --- a/ext/jack/gstjackaudiosrc.c +++ b/ext/jack/gstjackaudiosrc.c @@ -444,9 +444,17 @@ gst_jack_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec) (GST_SECOND / GST_USECOND), spec->rate * spec->bytes_per_sample); /* segtotal based on buffer-time latency */ spec->segtotal = spec->buffer_time / spec->latency_time; + if (spec->segtotal < 2) { + spec->segtotal = 2; + spec->buffer_time = spec->latency_time * spec->segtotal; + } - GST_DEBUG_OBJECT (src, "segsize %d, segtotal %d", spec->segsize, - spec->segtotal); + GST_DEBUG_OBJECT (src, "buffer time: %" G_GINT64_FORMAT " usec", + spec->buffer_time); + GST_DEBUG_OBJECT (src, "latency time: %" G_GINT64_FORMAT " usec", + spec->latency_time); + GST_DEBUG_OBJECT (src, "buffer_size %d, segsize %d, segtotal %d", + buffer_size, spec->segsize, spec->segtotal); /* allocate the ringbuffer memory now */ buf->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize); From 4ccda2632782713d357105d214382b2741f69530 Mon Sep 17 00:00:00 2001 From: Benjamin Otte Date: Thu, 18 Mar 2010 17:30:26 +0100 Subject: [PATCH 189/351] gst_element_class_set_details => gst_element_class_set_details_simple --- ext/jack/gstjackaudiosink.c | 10 ++-------- ext/jack/gstjackaudiosrc.c | 11 ++++------- 2 files changed, 6 insertions(+), 15 deletions(-) diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c index 0e75f16d25..40ee5a0fea 100644 --- a/ext/jack/gstjackaudiosink.c +++ b/ext/jack/gstjackaudiosink.c @@ -621,13 +621,6 @@ gst_jack_ring_buffer_delay (GstRingBuffer * buf) return res; } -/* elementfactory information */ -static const GstElementDetails gst_jack_audio_sink_details = -GST_ELEMENT_DETAILS ("Audio Sink (Jack)", - "Sink/Audio", - "Output to Jack", - "Wim Taymans "); - static GstStaticPadTemplate jackaudiosink_sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, @@ -676,7 +669,8 @@ gst_jack_audio_sink_base_init (gpointer g_class) { GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); - gst_element_class_set_details (element_class, &gst_jack_audio_sink_details); + gst_element_class_set_details_simple (element_class, "Audio Sink (Jack)", + "Sink/Audio", "Output to Jack", "Wim Taymans "); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&jackaudiosink_sink_factory)); diff --git a/ext/jack/gstjackaudiosrc.c b/ext/jack/gstjackaudiosrc.c index 211e1d2e6d..da97404fa8 100644 --- a/ext/jack/gstjackaudiosrc.c +++ b/ext/jack/gstjackaudiosrc.c @@ -678,17 +678,14 @@ static GstRingBuffer *gst_jack_audio_src_create_ringbuffer (GstBaseAudioSrc * static void gst_jack_audio_src_base_init (gpointer gclass) { - static GstElementDetails gst_jack_audio_src_details = { - "Audio Source (Jack)", - "Source/Audio", - "Input from Jack", - "Tristan Matthews " - }; + GstElementClass *element_class = GST_ELEMENT_CLASS (gclass); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&src_factory)); - gst_element_class_set_details (element_class, &gst_jack_audio_src_details); + gst_element_class_set_details_simple (element_class, "Audio Source (Jack)", + "Source/Audio", + "Input from Jack", "Tristan Matthews "); } /* initialize the jack_audio_src's class */ From 71366f80d9e730cbc494680086b9311f87b04196 Mon Sep 17 00:00:00 2001 From: Benjamin Otte Date: Sun, 21 Mar 2010 21:39:18 +0100 Subject: [PATCH 190/351] Add -Wmissing-declarations -Wmissing-prototypes to configure flags And fix all warnings --- ext/jack/gstjack.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/ext/jack/gstjack.h b/ext/jack/gstjack.h index fc26b71384..4565e067d5 100644 --- a/ext/jack/gstjack.h +++ b/ext/jack/gstjack.h @@ -47,6 +47,6 @@ typedef enum { typedef jack_default_audio_sample_t sample_t; #define GST_TYPE_JACK_CONNECT (gst_jack_connect_get_type()) -GType gst_jack_connect_get_type(); +GType gst_jack_connect_get_type(void); #endif // _GST_JACK_H_ From de91f84dfb0ae3d79e67f0430011c34051d1e785 Mon Sep 17 00:00:00 2001 From: Benjamin Otte Date: Mon, 22 Mar 2010 16:56:03 +0100 Subject: [PATCH 191/351] Add -Wold-style-definition and fix the warnings --- ext/jack/gstjackaudiosrc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/ext/jack/gstjackaudiosrc.c b/ext/jack/gstjackaudiosrc.c index da97404fa8..4669bbc5d1 100644 --- a/ext/jack/gstjackaudiosrc.c +++ b/ext/jack/gstjackaudiosrc.c @@ -141,7 +141,7 @@ gst_jack_audio_src_free_channels (GstJackAudioSrc * src) /* ringbuffer abstract base class */ static GType -gst_jack_ring_buffer_get_type () +gst_jack_ring_buffer_get_type (void) { static GType ringbuffer_type = 0; From 0a22a422682a91ea8d83ddc819a2c0bd6625daea Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Wed, 24 Mar 2010 15:59:53 +0200 Subject: [PATCH 192/351] jack: fix element name in section doc blob --- ext/jack/gstjackaudiosrc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/ext/jack/gstjackaudiosrc.c b/ext/jack/gstjackaudiosrc.c index 4669bbc5d1..4f1dd4f30e 100644 --- a/ext/jack/gstjackaudiosrc.c +++ b/ext/jack/gstjackaudiosrc.c @@ -41,7 +41,7 @@ */ /** - * SECTION:element-jack_audio_src + * SECTION:element-jackaudiosrc * @see_also: #GstBaseAudioSrc, #GstRingBuffer * * A Src that inputs data from Jack ports. From 084e8269c29450820a44ab2b5258635223a58346 Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Tue, 27 Apr 2010 10:34:24 +0300 Subject: [PATCH 193/351] jack: don't use GST_DEBUG_FUNCPTR for gobject vmethods --- ext/jack/gstjackaudiosink.c | 10 ++++------ ext/jack/gstjackaudiosrc.c | 10 ++++------ 2 files changed, 8 insertions(+), 12 deletions(-) diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c index 40ee5a0fea..04c9d06c15 100644 --- a/ext/jack/gstjackaudiosink.c +++ b/ext/jack/gstjackaudiosink.c @@ -160,8 +160,8 @@ gst_jack_ring_buffer_class_init (GstJackRingBufferClass * klass) ring_parent_class = g_type_class_peek_parent (klass); - gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_dispose); - gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_finalize); + gobject_class->dispose = gst_jack_ring_buffer_dispose; + gobject_class->finalize = gst_jack_ring_buffer_finalize; gstringbuffer_class->open_device = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_open_device); @@ -689,10 +689,8 @@ gst_jack_audio_sink_class_init (GstJackAudioSinkClass * klass) gstbasesink_class = (GstBaseSinkClass *) klass; gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass; - gobject_class->get_property = - GST_DEBUG_FUNCPTR (gst_jack_audio_sink_get_property); - gobject_class->set_property = - GST_DEBUG_FUNCPTR (gst_jack_audio_sink_set_property); + gobject_class->get_property = gst_jack_audio_sink_get_property; + gobject_class->set_property = gst_jack_audio_sink_set_property; g_object_class_install_property (gobject_class, PROP_CONNECT, g_param_spec_enum ("connect", "Connect", diff --git a/ext/jack/gstjackaudiosrc.c b/ext/jack/gstjackaudiosrc.c index 4f1dd4f30e..75204473d2 100644 --- a/ext/jack/gstjackaudiosrc.c +++ b/ext/jack/gstjackaudiosrc.c @@ -178,8 +178,8 @@ gst_jack_ring_buffer_class_init (GstJackRingBufferClass * klass) ring_parent_class = g_type_class_peek_parent (klass); - gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_dispose); - gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_finalize); + gobject_class->dispose = gst_jack_ring_buffer_dispose; + gobject_class->finalize = gst_jack_ring_buffer_finalize; gstringbuffer_class->open_device = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_open_device); @@ -703,10 +703,8 @@ gst_jack_audio_src_class_init (GstJackAudioSrcClass * klass) gstbasesrc_class = (GstBaseSrcClass *) klass; gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass; - gobject_class->set_property = - GST_DEBUG_FUNCPTR (gst_jack_audio_src_set_property); - gobject_class->get_property = - GST_DEBUG_FUNCPTR (gst_jack_audio_src_get_property); + gobject_class->set_property = gst_jack_audio_src_set_property; + gobject_class->get_property = gst_jack_audio_src_get_property; g_object_class_install_property (gobject_class, PROP_CONNECT, g_param_spec_enum ("connect", "Connect", From 2b7ca1413a244fab1e73dd65403ce449aa2f6cf7 Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Tue, 27 Apr 2010 10:59:00 +0300 Subject: [PATCH 194/351] jack: don't leak caps Add dispose methods to clear caps. --- ext/jack/gstjackaudiosink.c | 11 +++++++++++ ext/jack/gstjackaudiosrc.c | 11 +++++++++++ 2 files changed, 22 insertions(+) diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c index 04c9d06c15..3b9df01959 100644 --- a/ext/jack/gstjackaudiosink.c +++ b/ext/jack/gstjackaudiosink.c @@ -655,6 +655,7 @@ enum GST_BOILERPLATE_FULL (GstJackAudioSink, gst_jack_audio_sink, GstBaseAudioSink, GST_TYPE_BASE_AUDIO_SINK, _do_init); +static void gst_jack_audio_sink_dispose (GObject * object); static void gst_jack_audio_sink_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_jack_audio_sink_get_property (GObject * object, guint prop_id, @@ -689,6 +690,7 @@ gst_jack_audio_sink_class_init (GstJackAudioSinkClass * klass) gstbasesink_class = (GstBaseSinkClass *) klass; gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass; + gobject_class->dispose = gst_jack_audio_sink_dispose; gobject_class->get_property = gst_jack_audio_sink_get_property; gobject_class->set_property = gst_jack_audio_sink_set_property; @@ -724,6 +726,15 @@ gst_jack_audio_sink_init (GstJackAudioSink * sink, sink->port_count = 0; } +static void +gst_jack_audio_sink_dispose (GObject * object) +{ + GstJackAudioSink *sink = GST_JACK_AUDIO_SINK (object); + + gst_caps_replace (&sink->caps, NULL); + G_OBJECT_CLASS (parent_class)->dispose (object); +} + static void gst_jack_audio_sink_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) diff --git a/ext/jack/gstjackaudiosrc.c b/ext/jack/gstjackaudiosrc.c index 75204473d2..ee4ae8ddc3 100644 --- a/ext/jack/gstjackaudiosrc.c +++ b/ext/jack/gstjackaudiosrc.c @@ -664,6 +664,7 @@ static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src", GST_BOILERPLATE_FULL (GstJackAudioSrc, gst_jack_audio_src, GstBaseAudioSrc, GST_TYPE_BASE_AUDIO_SRC, _do_init); +static void gst_jack_audio_src_dispose (GObject * object); static void gst_jack_audio_src_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_jack_audio_src_get_property (GObject * object, guint prop_id, @@ -703,6 +704,7 @@ gst_jack_audio_src_class_init (GstJackAudioSrcClass * klass) gstbasesrc_class = (GstBaseSrcClass *) klass; gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass; + gobject_class->dispose = gst_jack_audio_src_dispose; gobject_class->set_property = gst_jack_audio_src_set_property; gobject_class->get_property = gst_jack_audio_src_get_property; @@ -742,6 +744,15 @@ gst_jack_audio_src_init (GstJackAudioSrc * src, GstJackAudioSrcClass * gclass) src->port_count = 0; } +static void +gst_jack_audio_src_dispose (GObject * object) +{ + GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object); + + gst_caps_replace (&src->caps, NULL); + G_OBJECT_CLASS (parent_class)->dispose (object); +} + static void gst_jack_audio_src_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) From 3153aa995f2fe2555f9b7e73ddfbce4005f3c131 Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Tue, 27 Apr 2010 11:21:16 +0300 Subject: [PATCH 195/351] jack: remove empty dispose and finalize methods --- ext/jack/gstjackaudiosink.c | 19 ------------------- ext/jack/gstjackaudiosrc.c | 17 ----------------- ext/jack/gstjackringbuffer.h | 2 -- 3 files changed, 38 deletions(-) diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c index 3b9df01959..ed11ca2d38 100644 --- a/ext/jack/gstjackaudiosink.c +++ b/ext/jack/gstjackaudiosink.c @@ -160,9 +160,6 @@ gst_jack_ring_buffer_class_init (GstJackRingBufferClass * klass) ring_parent_class = g_type_class_peek_parent (klass); - gobject_class->dispose = gst_jack_ring_buffer_dispose; - gobject_class->finalize = gst_jack_ring_buffer_finalize; - gstringbuffer_class->open_device = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_open_device); gstringbuffer_class->close_device = @@ -319,22 +316,6 @@ gst_jack_ring_buffer_init (GstJackRingBuffer * buf, buf->sample_rate = -1; } -static void -gst_jack_ring_buffer_dispose (GObject * object) -{ - G_OBJECT_CLASS (ring_parent_class)->dispose (object); -} - -static void -gst_jack_ring_buffer_finalize (GObject * object) -{ - GstJackRingBuffer *ringbuffer; - - ringbuffer = GST_JACK_RING_BUFFER_CAST (object); - - G_OBJECT_CLASS (ring_parent_class)->finalize (object); -} - /* the _open_device method should make a connection with the server */ static gboolean diff --git a/ext/jack/gstjackaudiosrc.c b/ext/jack/gstjackaudiosrc.c index ee4ae8ddc3..f817c2bc1f 100644 --- a/ext/jack/gstjackaudiosrc.c +++ b/ext/jack/gstjackaudiosrc.c @@ -178,9 +178,6 @@ gst_jack_ring_buffer_class_init (GstJackRingBufferClass * klass) ring_parent_class = g_type_class_peek_parent (klass); - gobject_class->dispose = gst_jack_ring_buffer_dispose; - gobject_class->finalize = gst_jack_ring_buffer_finalize; - gstringbuffer_class->open_device = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_open_device); gstringbuffer_class->close_device = @@ -322,20 +319,6 @@ gst_jack_ring_buffer_init (GstJackRingBuffer * buf, buf->sample_rate = -1; } -static void -gst_jack_ring_buffer_dispose (GObject * object) -{ - G_OBJECT_CLASS (ring_parent_class)->dispose (object); -} - -static void -gst_jack_ring_buffer_finalize (GObject * object) -{ - GstJackRingBuffer *ringbuffer; - ringbuffer = GST_JACK_RING_BUFFER_CAST (object); - G_OBJECT_CLASS (ring_parent_class)->finalize (object); -} - /* the _open_device method should make a connection with the server */ static gboolean diff --git a/ext/jack/gstjackringbuffer.h b/ext/jack/gstjackringbuffer.h index dfe21b1a9c..266fdfa31d 100644 --- a/ext/jack/gstjackringbuffer.h +++ b/ext/jack/gstjackringbuffer.h @@ -73,8 +73,6 @@ struct _GstJackRingBufferClass static void gst_jack_ring_buffer_class_init(GstJackRingBufferClass * klass); static void gst_jack_ring_buffer_init(GstJackRingBuffer * ringbuffer, GstJackRingBufferClass * klass); -static void gst_jack_ring_buffer_dispose(GObject * object); -static void gst_jack_ring_buffer_finalize(GObject * object); static GstRingBufferClass *ring_parent_class = NULL; From 55f1ea09b04baf03996753ed0a3080563e4f3d6d Mon Sep 17 00:00:00 2001 From: Tristan Matthews Date: Tue, 27 Apr 2010 10:48:32 -0400 Subject: [PATCH 196/351] jack: implement multichannel support correctly for jackaudiosrc Fixes parts of bug #616541. --- ext/jack/Makefile.am | 4 +- ext/jack/gstjackaudiosrc.c | 3 + ext/jack/gstjackutil.c | 114 +++++++++++++++++++++++++++++++++++++ ext/jack/gstjackutil.h | 30 ++++++++++ 4 files changed, 149 insertions(+), 2 deletions(-) create mode 100644 ext/jack/gstjackutil.c create mode 100644 ext/jack/gstjackutil.h diff --git a/ext/jack/Makefile.am b/ext/jack/Makefile.am index 5e9c8af61d..dd31d77e72 100644 --- a/ext/jack/Makefile.am +++ b/ext/jack/Makefile.am @@ -1,12 +1,12 @@ plugin_LTLIBRARIES = libgstjack.la -libgstjack_la_SOURCES = gstjack.c gstjackaudiosrc.c gstjackaudiosink.c gstjackaudioclient.c +libgstjack_la_SOURCES = gstjackutil.c gstjack.c gstjackaudiosrc.c gstjackaudiosink.c gstjackaudioclient.c libgstjack_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) $(JACK_CFLAGS) libgstjack_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR) $(JACK_LIBS) libgstjack_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) libgstjack_la_LIBTOOLFLAGS = --tag=disable-static -noinst_HEADERS = gstjackaudiosrc.h gstjackaudiosink.h gstjackaudioclient.h gstjack.h gstjackringbuffer.h +noinst_HEADERS = gstjackutil.h gstjackaudiosrc.h gstjackaudiosink.h gstjackaudioclient.h gstjack.h gstjackringbuffer.h EXTRA_DIST = README diff --git a/ext/jack/gstjackaudiosrc.c b/ext/jack/gstjackaudiosrc.c index f817c2bc1f..f140206614 100644 --- a/ext/jack/gstjackaudiosrc.c +++ b/ext/jack/gstjackaudiosrc.c @@ -80,6 +80,7 @@ #include "gstjackaudiosrc.h" #include "gstjackringbuffer.h" +#include "gstjackutil.h" GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_src_debug); #define GST_CAT_DEFAULT gst_jack_audio_src_debug @@ -418,6 +419,8 @@ gst_jack_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec) if (!gst_jack_audio_src_allocate_channels (src, channels)) goto out_of_ports; + gst_jack_set_layout_on_caps (&spec->caps, channels); + buffer_size = jack_get_buffer_size (client); /* the segment size in bytes, this is large enough to hold a buffer of 32bit floats diff --git a/ext/jack/gstjackutil.c b/ext/jack/gstjackutil.c new file mode 100644 index 0000000000..cde84d8e85 --- /dev/null +++ b/ext/jack/gstjackutil.c @@ -0,0 +1,114 @@ +/* GStreamer Jack utility functions + * Copyright (C) 2010 Tristan Matthews + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#include "gstjackutil.h" +#include + +static const GstAudioChannelPosition default_positions[8][8] = { + /* 1 channel */ + { + GST_AUDIO_CHANNEL_POSITION_FRONT_MONO, + }, + /* 2 channels */ + { + GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, + GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, + }, + /* 3 channels (2.1) */ + { + GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, + GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, + GST_AUDIO_CHANNEL_POSITION_LFE, /* or FRONT_CENTER for 3.0? */ + }, + /* 4 channels (4.0 or 3.1?) */ + { + GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, + GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, + GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, + GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, + }, + /* 5 channels */ + { + GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, + GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, + GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, + GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, + GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, + }, + /* 6 channels */ + { + GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, + GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, + GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, + GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, + GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, + GST_AUDIO_CHANNEL_POSITION_LFE, + }, + /* 7 channels */ + { + GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, + GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, + GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, + GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, + GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, + GST_AUDIO_CHANNEL_POSITION_LFE, + GST_AUDIO_CHANNEL_POSITION_REAR_CENTER, + }, + /* 8 channels */ + { + GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, + GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, + GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, + GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, + GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, + GST_AUDIO_CHANNEL_POSITION_LFE, + GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT, + GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT, + } +}; + + +/* if channels are less than or equal to 8, we set a default layout, + * otherwise set layout to an array of GST_AUDIO_CHANNEL_POSITION_NONE */ +void +gst_jack_set_layout_on_caps (GstCaps ** caps, gint channels) +{ + int c; + GValue pos = { 0 }; + GValue chanpos = { 0 }; + gst_caps_unref (*caps); + + if (channels <= 8) { + g_assert (channels >= 1); + gst_audio_set_channel_positions (gst_caps_get_structure (*caps, 0), + default_positions[channels - 1]); + } else { + g_value_init (&chanpos, GST_TYPE_ARRAY); + g_value_init (&pos, GST_TYPE_AUDIO_CHANNEL_POSITION); + for (c = 0; c < channels; c++) { + g_value_set_enum (&pos, GST_AUDIO_CHANNEL_POSITION_NONE); + gst_value_array_append_value (&chanpos, &pos); + } + g_value_unset (&pos); + gst_structure_set_value (gst_caps_get_structure (*caps, 0), + "channel-positions", &chanpos); + g_value_unset (&chanpos); + } + gst_caps_ref (*caps); +} diff --git a/ext/jack/gstjackutil.h b/ext/jack/gstjackutil.h new file mode 100644 index 0000000000..e330afd5e5 --- /dev/null +++ b/ext/jack/gstjackutil.h @@ -0,0 +1,30 @@ +/* GStreamer + * Copyright (C) 2010 Tristan Matthews + * + * gstjackutil.h: + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifndef _GST_JACK_UTIL_H_ +#define _GST_JACK_UTIL_H_ + +#include + +void +gst_jack_set_layout_on_caps (GstCaps **caps, gint channels); + +#endif // _GST_JACK_UTIL_H_ From e05b723274a24aa20c599eedfc179648306db851 Mon Sep 17 00:00:00 2001 From: Tristan Matthews Date: Tue, 11 May 2010 17:56:31 -0400 Subject: [PATCH 197/351] jack: improve process_cb --- ext/jack/gstjackaudiosrc.c | 40 +++++++++++++++----------------------- ext/jack/gstjackaudiosrc.h | 1 + 2 files changed, 17 insertions(+), 24 deletions(-) diff --git a/ext/jack/gstjackaudiosrc.c b/ext/jack/gstjackaudiosrc.c index f140206614..d5483debe9 100644 --- a/ext/jack/gstjackaudiosrc.c +++ b/ext/jack/gstjackaudiosrc.c @@ -98,6 +98,7 @@ gst_jack_audio_src_allocate_channels (GstJackAudioSrc * src, gint channels) /* alloc enough input ports */ src->ports = g_realloc (src->ports, sizeof (jack_port_t *) * channels); + src->buffers = g_realloc (src->buffers, sizeof (sample_t *) * channels); /* create an input port for each channel */ while (src->port_count < channels) { @@ -138,6 +139,8 @@ gst_jack_audio_src_free_channels (GstJackAudioSrc * src) } g_free (src->ports); src->ports = NULL; + g_free (src->buffers); + src->buffers = NULL; } /* ringbuffer abstract base class */ @@ -203,47 +206,35 @@ jack_process_cb (jack_nframes_t nframes, void *arg) { GstJackAudioSrc *src; GstRingBuffer *buf; - GstJackRingBuffer *abuf; gint len, givenLen; - guint8 *writeptr, *dataStart; + guint8 *writeptr; gint writeseg; gint channels, i, j; - sample_t **buffers, *data; + sample_t *data; buf = GST_RING_BUFFER_CAST (arg); - abuf = GST_JACK_RING_BUFFER_CAST (arg); src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf)); channels = buf->spec.channels; len = sizeof (sample_t) * nframes * channels; - - /* alloc pointers to samples */ - buffers = g_alloca (sizeof (sample_t *) * channels); - data = g_alloca (len); - /* get input buffers */ for (i = 0; i < channels; i++) - buffers[i] = (sample_t *) jack_port_get_buffer (src->ports[i], nframes); - - //writeptr = data; - dataStart = (guint8 *) data; - - /* the samples in the jack input buffers have to be interleaved into the - * ringbuffer - */ - for (i = 0; i < nframes; ++i) - for (j = 0; j < channels; ++j) - *data++ = buffers[j][i]; + src->buffers[i] = + (sample_t *) jack_port_get_buffer (src->ports[i], nframes); if (gst_ring_buffer_prepare_read (buf, &writeseg, &writeptr, &givenLen)) { - memcpy (writeptr, (char *) dataStart, givenLen); + /* the samples in the jack input buffers have to be interleaved into the + * ringbuffer + */ + data = (sample_t *) writeptr; + for (i = 0; i < nframes; ++i) + for (j = 0; j < channels; ++j) + *data++ = src->buffers[j][i]; + GST_DEBUG ("copy %d frames: %p, %d bytes, %d channels", nframes, writeptr, len / channels, channels); - /* clear written samples in the ringbuffer */ - // gst_ring_buffer_clear(buf, 0); - /* we wrote one segment */ gst_ring_buffer_advance (buf, 1); } @@ -728,6 +719,7 @@ gst_jack_audio_src_init (GstJackAudioSrc * src, GstJackAudioSrcClass * gclass) src->server = g_strdup (DEFAULT_PROP_SERVER); src->ports = NULL; src->port_count = 0; + src->buffers = NULL; } static void diff --git a/ext/jack/gstjackaudiosrc.h b/ext/jack/gstjackaudiosrc.h index b8d451aebb..e81b421422 100644 --- a/ext/jack/gstjackaudiosrc.h +++ b/ext/jack/gstjackaudiosrc.h @@ -81,6 +81,7 @@ struct _GstJackAudioSrc /* our ports */ jack_port_t **ports; int port_count; + sample_t **buffers; }; struct _GstJackAudioSrcClass From 71b06e49575d75f0390565dc9b08dafb73c9e44c Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Thu, 13 May 2010 12:55:29 +0200 Subject: [PATCH 198/351] jacksrc: make sure we always read nframes Error out when we are asked to read a different size that what was configured as the jack period size because that would mean something else is wrong. Fixes #618409 --- ext/jack/gstjackaudiosrc.c | 28 ++++++++++++++++++++-------- 1 file changed, 20 insertions(+), 8 deletions(-) diff --git a/ext/jack/gstjackaudiosrc.c b/ext/jack/gstjackaudiosrc.c index d5483debe9..22ba291b86 100644 --- a/ext/jack/gstjackaudiosrc.c +++ b/ext/jack/gstjackaudiosrc.c @@ -206,32 +206,36 @@ jack_process_cb (jack_nframes_t nframes, void *arg) { GstJackAudioSrc *src; GstRingBuffer *buf; - gint len, givenLen; + gint len; guint8 *writeptr; gint writeseg; - gint channels, i, j; + gint channels, i, j, flen; sample_t *data; buf = GST_RING_BUFFER_CAST (arg); src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf)); channels = buf->spec.channels; - len = sizeof (sample_t) * nframes * channels; + /* get input buffers */ for (i = 0; i < channels; i++) src->buffers[i] = (sample_t *) jack_port_get_buffer (src->ports[i], nframes); - if (gst_ring_buffer_prepare_read (buf, &writeseg, &writeptr, &givenLen)) { - /* the samples in the jack input buffers have to be interleaved into the - * ringbuffer - */ + if (gst_ring_buffer_prepare_read (buf, &writeseg, &writeptr, &len)) { + flen = len / channels; + + /* the number of samples must be exactly the segment size */ + if (nframes * sizeof (sample_t) != flen) + goto wrong_size; + + /* the samples in the jack input buffers have to be interleaved into the + * ringbuffer */ data = (sample_t *) writeptr; for (i = 0; i < nframes; ++i) for (j = 0; j < channels; ++j) *data++ = src->buffers[j][i]; - GST_DEBUG ("copy %d frames: %p, %d bytes, %d channels", nframes, writeptr, len / channels, channels); @@ -239,6 +243,14 @@ jack_process_cb (jack_nframes_t nframes, void *arg) gst_ring_buffer_advance (buf, 1); } return 0; + + /* ERRORS */ +wrong_size: + { + GST_ERROR_OBJECT (src, "nbytes (%d) != flen (%d)", + (gint) (nframes * sizeof (sample_t)), flen); + return 1; + } } /* we error out */ From a110fe3343a11edb42ee4abcfc63e5b80a4f49af Mon Sep 17 00:00:00 2001 From: Tristan Matthews Date: Thu, 17 Jun 2010 16:26:07 -0400 Subject: [PATCH 199/351] jack: removed unused file gstjackbin.c This is a 0.8 leftover. --- ext/jack/gstjackbin.c | 349 ------------------------------------------ 1 file changed, 349 deletions(-) delete mode 100644 ext/jack/gstjackbin.c diff --git a/ext/jack/gstjackbin.c b/ext/jack/gstjackbin.c deleted file mode 100644 index 08fed371e0..0000000000 --- a/ext/jack/gstjackbin.c +++ /dev/null @@ -1,349 +0,0 @@ -/* -*- Mode: C; c-basic-offset: 4 -*- */ -/* - Copyright (C) 2002, 2003 Andy Wingo - - This library is free software; you can redistribute it and/or - modify it under the terms of the GNU General Public - License as published by the Free Software Foundation; either - version 2 of the License, or (at your option) any later version. - - This library is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the implied warranty of - MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - General Public License for more details. - - You should have received a copy of the GNU General Public - License along with this library; if not, write to the Free - Software Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. -*/ - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif - -#include -#include -#include -#include "gstjack.h" - - -static GstBinClass *parent_class = NULL; - -static void gst_jack_bin_init (GstJackBin * this); -static void gst_jack_bin_class_init (GstJackBinClass * klass); - -static GstStateChangeReturn gst_jack_bin_change_state (GstElement * element, - GstStateChange transition); - -/* jack callbacks */ -static int process (jack_nframes_t nframes, void *arg); -static int sample_rate (jack_nframes_t nframes, void *arg); -static int buffer_size (jack_nframes_t nframes, void *arg); -static void shutdown (void *arg); - -static void sighup_handler (int sig); -static GstJackBin *_jackbin = NULL; -static gboolean watchdog_please_set_the_jackbin_to_ready = FALSE; - -/* fixme: we need a watchdog thread to see if we have received a SIGHUP, and if - * so set the state of the bin to READY. */ - -GType -gst_jack_bin_get_type (void) -{ - static GType jack_bin_type = 0; - - if (!jack_bin_type) { - static const GTypeInfo jack_bin_info = { - sizeof (GstJackBinClass), - NULL, - NULL, - (GClassInitFunc) gst_jack_bin_class_init, - NULL, - NULL, - sizeof (GstJackBin), - 0, - (GInstanceInitFunc) gst_jack_bin_init, - }; - - jack_bin_type = - g_type_register_static (GST_TYPE_BIN, "GstJackBin", &jack_bin_info, 0); - } - return jack_bin_type; -} - -static void -gst_jack_bin_class_init (GstJackBinClass * klass) -{ - GObjectClass *object_class; - GstElementClass *element_class; - - object_class = (GObjectClass *) klass; - element_class = (GstElementClass *) klass; - - parent_class = g_type_class_peek_parent (klass); - - element_class->change_state = gst_jack_bin_change_state; -} - -static void -gst_jack_bin_init (GstJackBin * this) -{ - GST_DEBUG ("initializing jack bin"); - - /* jack bins are managing bins and iterate themselves */ - GST_OBJECT_FLAG_SET (this, GST_BIN_FLAG_MANAGER); - GST_OBJECT_FLAG_SET (this, GST_BIN_SELF_SCHEDULABLE); - - /* make a new scheduler and associate it with the bin */ - gst_scheduler_factory_make (NULL, GST_ELEMENT (this)); -} - -static GstStateChangeReturn -gst_jack_bin_change_state (GstElement * element, GstStateChange transition) -{ - GstJackBin *this; - GList *l = NULL; - GstJackPad *pad; - - g_return_val_if_fail (element != NULL, FALSE); - this = GST_JACK_BIN (element); - - switch (GST_STATE_PENDING (element)) { - case GST_STATE_NULL: - JACK_DEBUG ("jackbin: NULL state"); - if (this->client) { - JACK_DEBUG ("jackbin: closing client"); - jack_client_close (this->client); - this->client = NULL; - } - - if (_jackbin) - signal (SIGHUP, SIG_DFL); - _jackbin = NULL; - - if (GST_ELEMENT_CLASS (parent_class)->change_state) - return GST_ELEMENT_CLASS (parent_class)->change_state (element, - transition); - break; - - case GST_STATE_READY: - JACK_DEBUG ("jackbin: READY"); - - _jackbin = this; - signal (SIGHUP, sighup_handler); - - if (!this->client) { - if (!(this->client = jack_client_new ("gst-jack"))) { - g_warning ("jack server not running?"); - return GST_STATE_CHANGE_FAILURE; - } - - gst_scheduler_setup (GST_ELEMENT_SCHED (this)); - - jack_set_process_callback (this->client, process, this); - jack_set_sample_rate_callback (this->client, sample_rate, this); - jack_set_buffer_size_callback (this->client, buffer_size, this); - this->nframes = jack_get_buffer_size (this->client); - jack_on_shutdown (this->client, shutdown, this); - } - - if (GST_OBJECT_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_OPEN)) { - l = this->src_pads; - while (l) { - JACK_DEBUG ("jackbin: unregistering pad %s:%s", - GST_JACK_PAD (l)->name, GST_JACK_PAD (l)->peer_name); - jack_port_unregister (this->client, GST_JACK_PAD (l)->port); - l = g_list_next (l); - } - l = this->sink_pads; - while (l) { - JACK_DEBUG ("jackbin: unregistering pad %s:%s", - GST_JACK_PAD (l)->name, GST_JACK_PAD (l)->peer_name); - jack_port_unregister (this->client, GST_JACK_PAD (l)->port); - l = g_list_next (l); - } - GST_OBJECT_FLAG_UNSET (GST_OBJECT (this), GST_JACK_OPEN); - - if (GST_OBJECT_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_ACTIVE)) { - JACK_DEBUG ("jackbin: deactivating client"); - jack_deactivate (this->client); - GST_OBJECT_FLAG_UNSET (GST_OBJECT (this), GST_JACK_ACTIVE); - } - } - - if (GST_ELEMENT_CLASS (parent_class)->change_state) - return GST_ELEMENT_CLASS (parent_class)->change_state (element, - transition); - break; - - case GST_STATE_PAUSED: - JACK_DEBUG ("jackbin: PAUSED"); - - if (!GST_OBJECT_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_OPEN)) { - l = this->src_pads; - while (l) { - pad = GST_JACK_PAD (l); - JACK_DEBUG ("jackbin: registering input port %s (peer %s)", pad->name, - pad->peer_name); - pad->port = - jack_port_register (this->client, pad->name, - JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput | JackPortIsTerminal, 0); - l = g_list_next (l); - } - l = this->sink_pads; - while (l) { - pad = GST_JACK_PAD (l); - JACK_DEBUG ("jackbin: registering output port %s (peer %s)", - pad->name, pad->peer_name); - pad->port = - jack_port_register (this->client, pad->name, - JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput | JackPortIsTerminal, - 0); - l = g_list_next (l); - } - - /* must activate before connecting */ - if (!GST_OBJECT_FLAG_IS_SET (GST_OBJECT (this), GST_JACK_ACTIVE)) { - JACK_DEBUG ("jackbin: activating client"); - jack_activate (this->client); - GST_OBJECT_FLAG_SET (GST_OBJECT (this), GST_JACK_ACTIVE); - } - - l = this->src_pads; - while (l) { - pad = GST_JACK_PAD (l); - JACK_DEBUG ("connecting jack port %s to gst jack port %s", - pad->peer_name, jack_port_name (pad->port)); - if (jack_connect (this->client, pad->peer_name, - jack_port_name (pad->port))) { - g_warning ("jackbin: could not connect %s and %s", pad->peer_name, - jack_port_name (pad->port)); - return GST_STATE_CHANGE_FAILURE; - } - l = g_list_next (l); - } - l = this->sink_pads; - while (l) { - pad = GST_JACK_PAD (l); - JACK_DEBUG ("connecting gst jack port %s to jack port %s", - jack_port_name (pad->port), pad->peer_name); - if (jack_connect (this->client, jack_port_name (pad->port), - pad->peer_name)) { - g_warning ("jackbin: could not connect %s and %s", pad->peer_name, - jack_port_name (pad->port)); - return GST_STATE_CHANGE_FAILURE; - } - l = g_list_next (l); - } - - JACK_DEBUG ("jackbin: setting OPEN flag"); - GST_OBJECT_FLAG_SET (GST_OBJECT (this), GST_JACK_OPEN); - - if (GST_ELEMENT_CLASS (parent_class)->change_state) - return GST_ELEMENT_CLASS (parent_class)->change_state (element, - transition); - } else { - if (GST_ELEMENT_CLASS (parent_class)->change_state) - return GST_ELEMENT_CLASS (parent_class)->change_state (element, - transition); - } - - break; - case GST_STATE_PLAYING: - JACK_DEBUG ("jackbin: PLAYING"); - - if (GST_ELEMENT_CLASS (parent_class)->change_state) - return GST_ELEMENT_CLASS (parent_class)->change_state (element, - transition); - break; - } - - JACK_DEBUG ("jackbin: state change finished"); - - return GST_STATE_CHANGE_SUCCESS; -} - -/* jack callbacks */ - -/* keep in mind that these run in another thread, mm-kay? */ - -static int -process (jack_nframes_t nframes, void *arg) -{ - GstJackBin *bin = (GstJackBin *) arg; - GstJackPad *pad; - GList *l; - - g_assert (bin); - - JACK_DEBUG ("jackbin: process()"); - - if (GST_STATE (bin) != GST_STATE_PLAYING) { - JACK_DEBUG ("jackbin: bin is not PLAYING yet, returning"); - return 0; - } else { - JACK_DEBUG ("jackbin: we are PLAYING, let's process()"); - } - - l = bin->src_pads; - while (l) { - pad = GST_JACK_PAD (l); - pad->data = jack_port_get_buffer (pad->port, nframes); - l = g_list_next (l); - } - - l = bin->sink_pads; - while (l) { - pad = GST_JACK_PAD (l); - pad->data = jack_port_get_buffer (pad->port, nframes); - l = g_list_next (l); - } - - bin->nframes = nframes; - - JACK_DEBUG ("jackbin: iterating to process %ld frames of audio...", nframes); - if (!gst_bin_iterate (GST_BIN (bin))) { - g_warning ("bin failed to iterate"); - return -1; - } - - /* that's all folks */ - - return 0; -} - -static int -sample_rate (jack_nframes_t nframes, void *arg) -{ - GstJackBin *bin = (GstJackBin *) arg; - - JACK_DEBUG ("the sample rate is now %lu/sec\n", nframes); - bin->rate = nframes; - return 0; -} - -static int -buffer_size (jack_nframes_t nframes, void *arg) -{ - GstJackBin *bin = (GstJackBin *) arg; - - JACK_DEBUG ("the buffer size is now %lu\n", nframes); - bin->nframes = nframes; - return 0; -} - -static void -shutdown (void *arg) -{ -/* GstJackClient *client = (GstJackClient*) arg; */ - JACK_DEBUG ("shutdown %p\n", arg); -/* gst_element_set_state (GST_ELEMENT (client->manager), GST_STATE_READY); */ -} - -static void -sighup_handler (int sig) -{ - g_message ("got sighup, setting state to READY"); - watchdog_please_set_the_jackbin_to_ready = TRUE; -} From c070cfa7c832757a2e9b8a8f8114c661138761dd Mon Sep 17 00:00:00 2001 From: Tristan Matthews Date: Mon, 6 Sep 2010 16:11:31 -0400 Subject: [PATCH 200/351] jack: added client property --- ext/jack/gstjack.c | 29 +++++++++++++++++++++++++++++ ext/jack/gstjack.h | 3 +++ ext/jack/gstjackaudioclient.c | 22 +++++++++++----------- ext/jack/gstjackaudioclient.h | 1 + ext/jack/gstjackaudiosink.c | 18 ++++++++++++++++++ ext/jack/gstjackaudiosink.h | 1 + ext/jack/gstjackaudiosrc.c | 18 ++++++++++++++++++ ext/jack/gstjackaudiosrc.h | 1 + 8 files changed, 82 insertions(+), 11 deletions(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 083025016d..371a9e934f 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -46,6 +46,35 @@ gst_jack_connect_get_type (void) } +static gpointer +gst_jack_client_copy (gpointer jclient) +{ + return jclient; +} + + +static void +gst_jack_client_free (gpointer jclient) +{ + return; +} + + +GType +gst_jack_client_get_type (void) +{ + static GType type; /* 0 */ + + if (type == 0) { + /* hackish, but makes it show up nicely in gst-inspect */ + type = g_boxed_type_register_static ("JackClient", + (GBoxedCopyFunc) gst_jack_client_copy, + (GBoxedFreeFunc) gst_jack_client_free); + } + + return type; +} + static gboolean plugin_init (GstPlugin * plugin) { diff --git a/ext/jack/gstjack.h b/ext/jack/gstjack.h index 4565e067d5..d923866dfa 100644 --- a/ext/jack/gstjack.h +++ b/ext/jack/gstjack.h @@ -47,6 +47,9 @@ typedef enum { typedef jack_default_audio_sample_t sample_t; #define GST_TYPE_JACK_CONNECT (gst_jack_connect_get_type()) +#define GST_TYPE_JACK_CLIENT (gst_jack_client_get_type ()) + +GType gst_jack_client_get_type(void); GType gst_jack_connect_get_type(void); #endif // _GST_JACK_H_ diff --git a/ext/jack/gstjackaudioclient.c b/ext/jack/gstjackaudioclient.c index 60e413819d..1789edb60d 100644 --- a/ext/jack/gstjackaudioclient.c +++ b/ext/jack/gstjackaudioclient.c @@ -189,11 +189,10 @@ connection_find (GstJackAudioConnection * conn, FindData * data) * status set. */ static GstJackAudioConnection * gst_jack_audio_make_connection (const gchar * id, const gchar * server, - jack_status_t * status) + jack_client_t * jclient, jack_status_t * status) { GstJackAudioConnection *conn; jack_options_t options; - jack_client_t *jclient; gint res; *status = 0; @@ -207,7 +206,8 @@ gst_jack_audio_make_connection (const gchar * id, const gchar * server, if (server != NULL) options |= JackServerName; /* open the client */ - jclient = jack_client_open (id, options, status, server); + if (jclient == NULL) + jclient = jack_client_open (id, options, status, server); if (jclient == NULL) goto could_not_open; @@ -258,7 +258,7 @@ could_not_activate: static GstJackAudioConnection * gst_jack_audio_get_connection (const gchar * id, const gchar * server, - jack_status_t * status) + jack_client_t * jclient, jack_status_t * status) { GstJackAudioConnection *conn; GList *found; @@ -273,7 +273,7 @@ gst_jack_audio_get_connection (const gchar * id, const gchar * server, G_LOCK (connections_lock); found = g_list_find_custom (connections, &data, (GCompareFunc) connection_find); - if (found != NULL) { + if (found != NULL && jclient != NULL) { /* we found it, increase refcount and return it */ conn = (GstJackAudioConnection *) found->data; conn->refcount++; @@ -281,7 +281,7 @@ gst_jack_audio_get_connection (const gchar * id, const gchar * server, GST_DEBUG ("found connection %p", conn); } else { /* make new connection */ - conn = gst_jack_audio_make_connection (id, server, status); + conn = gst_jack_audio_make_connection (id, server, jclient, status); if (conn != NULL) { GST_DEBUG ("created connection %p", conn); /* add to list on success */ @@ -407,10 +407,10 @@ gst_jack_audio_connection_remove_client (GstJackAudioConnection * conn, */ GstJackAudioClient * gst_jack_audio_client_new (const gchar * id, const gchar * server, - GstJackClientType type, void (*shutdown) (void *arg), - JackProcessCallback process, JackBufferSizeCallback buffer_size, - JackSampleRateCallback sample_rate, gpointer user_data, - jack_status_t * status) + jack_client_t * jclient, GstJackClientType type, + void (*shutdown) (void *arg), JackProcessCallback process, + JackBufferSizeCallback buffer_size, JackSampleRateCallback sample_rate, + gpointer user_data, jack_status_t * status) { GstJackAudioClient *client; GstJackAudioConnection *conn; @@ -419,7 +419,7 @@ gst_jack_audio_client_new (const gchar * id, const gchar * server, g_return_val_if_fail (status != NULL, NULL); /* first get a connection for the id/server pair */ - conn = gst_jack_audio_get_connection (id, server, status); + conn = gst_jack_audio_get_connection (id, server, jclient, status); if (conn == NULL) goto no_connection; diff --git a/ext/jack/gstjackaudioclient.h b/ext/jack/gstjackaudioclient.h index 9092e7ded5..5fb7e35444 100644 --- a/ext/jack/gstjackaudioclient.h +++ b/ext/jack/gstjackaudioclient.h @@ -40,6 +40,7 @@ void gst_jack_audio_client_init (void); GstJackAudioClient * gst_jack_audio_client_new (const gchar *id, const gchar *server, + jack_client_t *jclient, GstJackClientType type, void (*shutdown) (void *arg), JackProcessCallback process, diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c index ed11ca2d38..e90282351f 100644 --- a/ext/jack/gstjackaudiosink.c +++ b/ext/jack/gstjackaudiosink.c @@ -334,6 +334,7 @@ gst_jack_ring_buffer_open_device (GstRingBuffer * buf) name = "GStreamer"; sink->client = gst_jack_audio_client_new (name, sink->server, + sink->jclient, GST_JACK_CLIENT_SINK, jack_shutdown_cb, jack_process_cb, jack_buffer_size_cb, jack_sample_rate_cb, buf, &status); @@ -627,6 +628,7 @@ enum PROP_0, PROP_CONNECT, PROP_SERVER, + PROP_CLIENT, PROP_LAST }; @@ -685,6 +687,12 @@ gst_jack_audio_sink_class_init (GstJackAudioSinkClass * klass) "The Jack server to connect to (NULL = default)", DEFAULT_PROP_SERVER, G_PARAM_READWRITE)); + g_object_class_install_property (gobject_class, PROP_CLIENT, + g_param_spec_boxed ("client", "JackClient", "Handle for jack client", + GST_TYPE_JACK_CLIENT, + GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE | + G_PARAM_STATIC_STRINGS)); + gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_jack_audio_sink_getcaps); gstbaseaudiosink_class->create_ringbuffer = @@ -703,6 +711,7 @@ gst_jack_audio_sink_init (GstJackAudioSink * sink, { sink->connect = DEFAULT_PROP_CONNECT; sink->server = g_strdup (DEFAULT_PROP_SERVER); + sink->jclient = NULL; sink->ports = NULL; sink->port_count = 0; } @@ -732,6 +741,12 @@ gst_jack_audio_sink_set_property (GObject * object, guint prop_id, g_free (sink->server); sink->server = g_value_dup_string (value); break; + case PROP_CLIENT: + if (GST_STATE (sink) == GST_STATE_NULL || + GST_STATE (sink) == GST_STATE_READY) { + sink->jclient = g_value_get_boxed (value); + } + break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; @@ -753,6 +768,9 @@ gst_jack_audio_sink_get_property (GObject * object, guint prop_id, case PROP_SERVER: g_value_set_string (value, sink->server); break; + case PROP_CLIENT: + g_value_set_boxed (value, sink->jclient); + break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; diff --git a/ext/jack/gstjackaudiosink.h b/ext/jack/gstjackaudiosink.h index b4a770332f..def423329d 100644 --- a/ext/jack/gstjackaudiosink.h +++ b/ext/jack/gstjackaudiosink.h @@ -57,6 +57,7 @@ struct _GstJackAudioSink { /* properties */ GstJackConnect connect; gchar *server; + jack_client_t *jclient; /* our client */ GstJackAudioClient *client; diff --git a/ext/jack/gstjackaudiosrc.c b/ext/jack/gstjackaudiosrc.c index 22ba291b86..1492b3ec04 100644 --- a/ext/jack/gstjackaudiosrc.c +++ b/ext/jack/gstjackaudiosrc.c @@ -341,6 +341,7 @@ gst_jack_ring_buffer_open_device (GstRingBuffer * buf) name = "GStreamer"; src->client = gst_jack_audio_client_new (name, src->server, + src->jclient, GST_JACK_CLIENT_SOURCE, jack_shutdown_cb, jack_process_cb, jack_buffer_size_cb, jack_sample_rate_cb, buf, &status); @@ -629,6 +630,7 @@ enum PROP_0, PROP_CONNECT, PROP_SERVER, + PROP_CLIENT, PROP_LAST }; @@ -707,6 +709,12 @@ gst_jack_audio_src_class_init (GstJackAudioSrcClass * klass) "The Jack server to connect to (NULL = default)", DEFAULT_PROP_SERVER, G_PARAM_READWRITE)); + g_object_class_install_property (gobject_class, PROP_CLIENT, + g_param_spec_boxed ("client", "JackClient", "Handle for jack client", + GST_TYPE_JACK_CLIENT, + GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE | + G_PARAM_STATIC_STRINGS)); + gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_jack_audio_src_getcaps); gstbaseaudiosrc_class->create_ringbuffer = GST_DEBUG_FUNCPTR (gst_jack_audio_src_create_ringbuffer); @@ -729,6 +737,7 @@ gst_jack_audio_src_init (GstJackAudioSrc * src, GstJackAudioSrcClass * gclass) //gst_base_src_set_live(GST_BASE_SRC (src), TRUE); src->connect = DEFAULT_PROP_CONNECT; src->server = g_strdup (DEFAULT_PROP_SERVER); + src->jclient = NULL; src->ports = NULL; src->port_count = 0; src->buffers = NULL; @@ -757,6 +766,12 @@ gst_jack_audio_src_set_property (GObject * object, guint prop_id, g_free (src->server); src->server = g_value_dup_string (value); break; + case PROP_CLIENT: + if (GST_STATE (src) == GST_STATE_NULL || + GST_STATE (src) == GST_STATE_READY) { + src->jclient = g_value_get_boxed (value); + } + break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; @@ -776,6 +791,9 @@ gst_jack_audio_src_get_property (GObject * object, guint prop_id, case PROP_SERVER: g_value_set_string (value, src->server); break; + case PROP_CLIENT: + g_value_set_boxed (value, src->jclient); + break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; diff --git a/ext/jack/gstjackaudiosrc.h b/ext/jack/gstjackaudiosrc.h index e81b421422..7e99b69df9 100644 --- a/ext/jack/gstjackaudiosrc.h +++ b/ext/jack/gstjackaudiosrc.h @@ -74,6 +74,7 @@ struct _GstJackAudioSrc /* properties */ GstJackConnect connect; gchar *server; + jack_client_t *jclient; /* our client */ GstJackAudioClient *client; From 13957deeed8348f10e8e8ec79f43105395dd26d5 Mon Sep 17 00:00:00 2001 From: Tristan Matthews Date: Mon, 6 Sep 2010 17:17:54 -0400 Subject: [PATCH 201/351] examples: add test to demonstrate jack_client_t usage --- tests/examples/jack/Makefile.am | 6 +++ tests/examples/jack/jack_client.c | 79 +++++++++++++++++++++++++++++++ 2 files changed, 85 insertions(+) create mode 100644 tests/examples/jack/Makefile.am create mode 100644 tests/examples/jack/jack_client.c diff --git a/tests/examples/jack/Makefile.am b/tests/examples/jack/Makefile.am new file mode 100644 index 0000000000..4adfd13142 --- /dev/null +++ b/tests/examples/jack/Makefile.am @@ -0,0 +1,6 @@ +noinst_PROGRAMS = jack_client + +jack_client_SOURCES = jack_client.c +jack_client_CFLAGS = $(GST_CFLAGS) $(GTK_CFLAGS) $(JACK_CFLAGS) +jack_client_LDFLAGS = $(GST_LIBS) $(GTK_LIBS) $(JACK_LIBS) + diff --git a/tests/examples/jack/jack_client.c b/tests/examples/jack/jack_client.c new file mode 100644 index 0000000000..99599ab5cb --- /dev/null +++ b/tests/examples/jack/jack_client.c @@ -0,0 +1,79 @@ +/* This app demonstrates the creation and use of a jack client in conjunction + * with the jack plugins. This way, an application can control the jack client + * directly. + */ + +#include +#include +#include + +static gboolean +quit_cb (gpointer data) +{ + gtk_main_quit (); + return FALSE; +} + +int +main (int argc, char **argv) +{ + jack_client_t *src_client, *sink_client; + jack_status_t status; + GstElement *pipeline, *src, *sink; + GstStateChangeReturn ret; + + gst_init (&argc, &argv); + + /* create jack clients */ + src_client = jack_client_open ("src_client", JackNoStartServer, &status); + if (src_client == NULL) { + if (status & JackServerFailed) + g_print ("JACK server not running\n"); + else + g_print ("jack_client_open() failed, status = 0x%2.0x\n", status); + return 1; + } + + sink_client = jack_client_open ("sink_client", JackNoStartServer, &status); + if (sink_client == NULL) { + if (status & JackServerFailed) + g_print ("JACK server not running\n"); + else + g_print ("jack_client_open() failed, status = 0x%2.0x\n", status); + return 1; + } + + /* create gst elements */ + pipeline = gst_pipeline_new ("my_pipeline"); + + src = gst_element_factory_make ("jackaudiosrc", NULL); + sink = gst_element_factory_make ("jackaudiosink", NULL); + + g_object_set (src, "client", src_client, NULL); + g_object_set (sink, "client", sink_client, NULL); + + gst_bin_add_many (GST_BIN (pipeline), src, sink, NULL); + + /* link everything together */ + if (!gst_element_link (src, sink)) { + g_print ("Failed to link elements!\n"); + return 1; + } + + /* run */ + ret = gst_element_set_state (pipeline, GST_STATE_PLAYING); + if (ret == GST_STATE_CHANGE_FAILURE) { + g_print ("Failed to start up pipeline!\n"); + return 1; + } + + /* quit after 5 seconds */ + g_timeout_add (5000, (GSourceFunc) quit_cb, NULL); + gtk_main (); + + /* clean up */ + gst_element_set_state (pipeline, GST_STATE_NULL); + gst_object_unref (pipeline); + + return 0; +} From 11c4157af41f1f05a6edca4f8de910146770b351 Mon Sep 17 00:00:00 2001 From: Tristan Matthews Date: Thu, 9 Sep 2010 14:49:06 -0400 Subject: [PATCH 202/351] jack: added translatable text for server not found error --- ext/jack/Makefile.am | 2 +- ext/jack/gstjackaudiosink.c | 8 +++++++- ext/jack/gstjackaudiosrc.c | 10 +++++++--- 3 files changed, 15 insertions(+), 5 deletions(-) diff --git a/ext/jack/Makefile.am b/ext/jack/Makefile.am index dd31d77e72..ede70594bd 100644 --- a/ext/jack/Makefile.am +++ b/ext/jack/Makefile.am @@ -2,7 +2,7 @@ plugin_LTLIBRARIES = libgstjack.la libgstjack_la_SOURCES = gstjackutil.c gstjack.c gstjackaudiosrc.c gstjackaudiosink.c gstjackaudioclient.c -libgstjack_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) $(JACK_CFLAGS) +libgstjack_la_CFLAGS = $(GST_PLUGINS_BAD_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) $(JACK_CFLAGS) libgstjack_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR) $(JACK_LIBS) libgstjack_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) libgstjack_la_LIBTOOLFLAGS = --tag=disable-static diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c index e90282351f..8f23791386 100644 --- a/ext/jack/gstjackaudiosink.c +++ b/ext/jack/gstjackaudiosink.c @@ -55,6 +55,11 @@ * Last reviewed on 2006-11-30 (0.10.4) */ +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include #include #include @@ -350,7 +355,8 @@ could_not_open: { if (status & JackServerFailed) { GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND, - (NULL), ("Cannot connect to the Jack server (status %d)", status)); + (_("Jack server not found")), + ("Cannot connect to the Jack server (status %d)", status)); } else { GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE, (NULL), ("Jack client open error (status %d)", status)); diff --git a/ext/jack/gstjackaudiosrc.c b/ext/jack/gstjackaudiosrc.c index 1492b3ec04..a97cea03e4 100644 --- a/ext/jack/gstjackaudiosrc.c +++ b/ext/jack/gstjackaudiosrc.c @@ -74,7 +74,11 @@ * Last reviewed on 2008-07-22 (0.10.4) */ -#include +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include #include #include @@ -357,7 +361,8 @@ could_not_open: { if (status & JackServerFailed) { GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, - (NULL), ("Cannot connect to the Jack server (status %d)", status)); + (_("Jack server not found")), + ("Cannot connect to the Jack server (status %d)", status)); } else { GST_ELEMENT_ERROR (src, RESOURCE, OPEN_WRITE, (NULL), ("Jack client open error (status %d)", status)); @@ -670,7 +675,6 @@ static GstRingBuffer *gst_jack_audio_src_create_ringbuffer (GstBaseAudioSrc * static void gst_jack_audio_src_base_init (gpointer gclass) { - GstElementClass *element_class = GST_ELEMENT_CLASS (gclass); gst_element_class_add_pad_template (element_class, From 3c43a9e102043b265d75674acaf3f34f2368731c Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Tue, 19 Oct 2010 16:23:23 +0300 Subject: [PATCH 203/351] various (ext): add missing G_PARAM_STATIC_STRINGS flags Canonicalize property names as needed. --- ext/jack/gstjackaudiosink.c | 5 +++-- ext/jack/gstjackaudiosrc.c | 5 +++-- 2 files changed, 6 insertions(+), 4 deletions(-) diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c index 8f23791386..32bf1af3e4 100644 --- a/ext/jack/gstjackaudiosink.c +++ b/ext/jack/gstjackaudiosink.c @@ -686,12 +686,13 @@ gst_jack_audio_sink_class_init (GstJackAudioSinkClass * klass) g_object_class_install_property (gobject_class, PROP_CONNECT, g_param_spec_enum ("connect", "Connect", "Specify how the output ports will be connected", - GST_TYPE_JACK_CONNECT, DEFAULT_PROP_CONNECT, G_PARAM_READWRITE)); + GST_TYPE_JACK_CONNECT, DEFAULT_PROP_CONNECT, + G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_SERVER, g_param_spec_string ("server", "Server", "The Jack server to connect to (NULL = default)", - DEFAULT_PROP_SERVER, G_PARAM_READWRITE)); + DEFAULT_PROP_SERVER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_CLIENT, g_param_spec_boxed ("client", "JackClient", "Handle for jack client", diff --git a/ext/jack/gstjackaudiosrc.c b/ext/jack/gstjackaudiosrc.c index a97cea03e4..0ffdb23985 100644 --- a/ext/jack/gstjackaudiosrc.c +++ b/ext/jack/gstjackaudiosrc.c @@ -706,12 +706,13 @@ gst_jack_audio_src_class_init (GstJackAudioSrcClass * klass) g_object_class_install_property (gobject_class, PROP_CONNECT, g_param_spec_enum ("connect", "Connect", "Specify how the input ports will be connected", - GST_TYPE_JACK_CONNECT, DEFAULT_PROP_CONNECT, G_PARAM_READWRITE)); + GST_TYPE_JACK_CONNECT, DEFAULT_PROP_CONNECT, + G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_SERVER, g_param_spec_string ("server", "Server", "The Jack server to connect to (NULL = default)", - DEFAULT_PROP_SERVER, G_PARAM_READWRITE)); + DEFAULT_PROP_SERVER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_CLIENT, g_param_spec_boxed ("client", "JackClient", "Handle for jack client", From 67f4489cd0cf5e0b0f031b19f2b44ceca1f3687f Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Sun, 2 Jan 2011 15:25:41 +0000 Subject: [PATCH 204/351] jack: new jackaudiosrc and jackaudiosink elements, moved from gst-plugins-bad https://bugzilla.gnome.org/show_bug.cgi?id=621929 --- .gitignore | 2 + configure.ac | 13 +++++- docs/plugins/Makefile.am | 2 + .../gst-plugins-good-plugins-docs.sgml | 3 ++ .../gst-plugins-good-plugins-sections.txt | 30 +++++++++++++ docs/plugins/inspect/plugin-jack.xml | 43 +++++++++++++++++++ ext/Makefile.am | 8 ++++ gst-plugins-good.spec.in | 1 + tests/examples/Makefile.am | 10 ++++- tests/examples/jack/Makefile.am | 8 +++- 10 files changed, 116 insertions(+), 4 deletions(-) create mode 100644 docs/plugins/inspect/plugin-jack.xml diff --git a/.gitignore b/.gitignore index 2ff4d5a7d5..91867b2857 100644 --- a/.gitignore +++ b/.gitignore @@ -43,3 +43,5 @@ gst/deinterlace/tvtime.h tmp-orc.c *orc.h + +/tests/examples/jack/jack_client diff --git a/configure.ac b/configure.ac index f510f70673..246e1f2b58 100644 --- a/configure.ac +++ b/configure.ac @@ -748,6 +748,14 @@ AG_GST_CHECK_FEATURE(HAL, [HAL libraries], halelements, [ AG_GST_PKG_CHECK_MODULES(HAL, [hal >= 0.5.6, dbus-1 >= 0.32]) ]) +dnl *** Jack *** +translit(dnm, m, l) AM_CONDITIONAL(USE_JACK, true) +AG_GST_CHECK_FEATURE(JACK, Jack, jack, [ + PKG_CHECK_MODULES(JACK, jack >= 0.99.10, HAVE_JACK="yes", HAVE_JACK="no") + AC_SUBST(JACK_CFLAGS) + AC_SUBST(JACK_LIBS) +]) + dnl *** jpeg *** dnl FIXME: we could use header checks here as well IMO translit(dnm, m, l) AM_CONDITIONAL(USE_JPEG, true) @@ -1020,6 +1028,7 @@ AM_CONDITIONAL(USE_GCONFTOOL, false) AM_CONDITIONAL(USE_GDK_PIXBUF, false) AM_CONDITIONAL(USE_GST_V4L2, false) AM_CONDITIONAL(USE_HAL, false) +AM_CONDITIONAL(USE_JACK, false) AM_CONDITIONAL(USE_JPEG, false) AM_CONDITIONAL(USE_LIBCACA, false) AM_CONDITIONAL(USE_LIBDV, false) @@ -1144,7 +1153,6 @@ gst/wavenc/Makefile gst/wavparse/Makefile gst/flx/Makefile gst/y4m/Makefile -ext/jpeg/Makefile ext/Makefile ext/aalib/Makefile ext/annodex/Makefile @@ -1155,6 +1163,8 @@ ext/flac/Makefile ext/gconf/Makefile ext/gdk_pixbuf/Makefile ext/hal/Makefile +ext/jack/Makefile +ext/jpeg/Makefile ext/libcaca/Makefile ext/libpng/Makefile ext/pulse/Makefile @@ -1180,6 +1190,7 @@ tests/check/Makefile tests/examples/Makefile tests/examples/audiofx/Makefile tests/examples/equalizer/Makefile +tests/examples/jack/Makefile tests/examples/level/Makefile tests/examples/pulse/Makefile tests/examples/rtp/Makefile diff --git a/docs/plugins/Makefile.am b/docs/plugins/Makefile.am index 6598f4db50..975e465ab9 100644 --- a/docs/plugins/Makefile.am +++ b/docs/plugins/Makefile.am @@ -94,6 +94,8 @@ EXTRA_HFILES = \ $(top_srcdir)/ext/gdk_pixbuf/gstgdkpixbufsink.h \ $(top_srcdir)/ext/hal/gsthalaudiosink.h \ $(top_srcdir)/ext/hal/gsthalaudiosrc.h \ + $(top_srcdir)/ext/jack/gstjackaudiosrc.h \ + $(top_srcdir)/ext/jack/gstjackaudiosink.h \ $(top_srcdir)/ext/jpeg/gstjpegdec.h \ $(top_srcdir)/ext/jpeg/gstjpegenc.h \ $(top_srcdir)/ext/jpeg/gstsmokedec.h \ diff --git a/docs/plugins/gst-plugins-good-plugins-docs.sgml b/docs/plugins/gst-plugins-good-plugins-docs.sgml index 1b7136b796..ade8dfde20 100644 --- a/docs/plugins/gst-plugins-good-plugins-docs.sgml +++ b/docs/plugins/gst-plugins-good-plugins-docs.sgml @@ -94,6 +94,8 @@ + + @@ -206,6 +208,7 @@ + diff --git a/docs/plugins/gst-plugins-good-plugins-sections.txt b/docs/plugins/gst-plugins-good-plugins-sections.txt index c643d47713..60d2c45a2d 100644 --- a/docs/plugins/gst-plugins-good-plugins-sections.txt +++ b/docs/plugins/gst-plugins-good-plugins-sections.txt @@ -1113,6 +1113,36 @@ GstInterleaveFunc gst_interleave_get_type +
+element-jackaudiosrc +jackaudiosrc +GstJackAudioSrc + +GstJackAudioSrcClass +GST_JACK_AUDIO_SRC +GST_JACK_AUDIO_SRC_CLASS +GST_JACK_AUDIO_SRC_GET_CLASS +GST_IS_JACK_AUDIO_SRC +GST_IS_JACK_AUDIO_SRC_CLASS +GST_TYPE_JACK_AUDIO_SRC +gst_jack_audio_src_get_type +
+ +
+element-jackaudiosink +jackaudiosink +GstJackAudioSink + +GstJackAudioSinkClass +GST_JACK_AUDIO_SINK +GST_JACK_AUDIO_SINK_CLASS +GST_JACK_AUDIO_SINK_GET_CLASS +GST_IS_JACK_AUDIO_SINK +GST_IS_JACK_AUDIO_SINK_CLASS +GST_TYPE_JACK_AUDIO_SINK +gst_jack_audio_sink_get_type +
+
element-jpegdec jpegdec diff --git a/docs/plugins/inspect/plugin-jack.xml b/docs/plugins/inspect/plugin-jack.xml new file mode 100644 index 0000000000..dcead7b27c --- /dev/null +++ b/docs/plugins/inspect/plugin-jack.xml @@ -0,0 +1,43 @@ + + jack + Jack elements + ../../ext/jack/.libs/libgstjack.so + libgstjack.so + 0.10.26.1 + LGPL + gst-plugins-good + GStreamer Good Plug-ins git + Unknown package origin + + + jackaudiosink + Audio Sink (Jack) + Sink/Audio + Output to Jack + Wim Taymans <wim@fluendo.com> + + + sink + sink + always +
audio/x-raw-float, endianness=(int){ 1234 }, width=(int)32, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]
+ + + + + jackaudiosrc + Audio Source (Jack) + Source/Audio + Input from Jack + Tristan Matthews <tristan@sat.qc.ca> + + + src + source + always +
audio/x-raw-float, endianness=(int){ 1234 }, width=(int)32, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]
+ + + + + \ No newline at end of file diff --git a/ext/Makefile.am b/ext/Makefile.am index 1dc2e2abca..409408ea8f 100644 --- a/ext/Makefile.am +++ b/ext/Makefile.am @@ -46,6 +46,12 @@ else HAL_DIR = endif +if USE_JACK +JACK_DIR=jack +else +JACK_DIR= +endif + if USE_JPEG JPEG_DIR = jpeg else @@ -135,6 +141,7 @@ SUBDIRS = \ $(GCONF_DIR) \ $(GDK_PIXBUF_DIR) \ $(HAL_DIR) \ + $(JACK_DIR) \ $(JPEG_DIR) \ $(LIBCACA_DIR) \ $(LIBDV_DIR) \ @@ -158,6 +165,7 @@ DIST_SUBDIRS = \ gconf \ gdk_pixbuf \ hal \ + jack \ jpeg \ libcaca \ libpng \ diff --git a/gst-plugins-good.spec.in b/gst-plugins-good.spec.in index 6cdd96c4c9..88dc858ace 100644 --- a/gst-plugins-good.spec.in +++ b/gst-plugins-good.spec.in @@ -147,6 +147,7 @@ rm -rf $RPM_BUILD_ROOT @USE_LIBCACA_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstcacasink.so @USE_ESD_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstesd.so @USE_FLAC_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstflac.so +@USE_JACK_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstjack.so @USE_JPEG_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstjpeg.so @USE_LIBPNG_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstpng.so @USE_OSS_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstossaudio.so diff --git a/tests/examples/Makefile.am b/tests/examples/Makefile.am index d3d93d4af3..54e53bafb7 100644 --- a/tests/examples/Makefile.am +++ b/tests/examples/Makefile.am @@ -1,5 +1,11 @@ -SUBDIRS = audiofx equalizer level pulse rtp shapewipe spectrum v4l2 +if USE_JACK +JACK_DIR=jack +else +JACK_DIR= +endif -DIST_SUBDIRS = audiofx equalizer level pulse rtp shapewipe spectrum v4l2 +SUBDIRS = audiofx equalizer $(JACK_DIR) level pulse rtp shapewipe spectrum v4l2 + +DIST_SUBDIRS = audiofx equalizer jack level pulse rtp shapewipe spectrum v4l2 include $(top_srcdir)/common/parallel-subdirs.mak diff --git a/tests/examples/jack/Makefile.am b/tests/examples/jack/Makefile.am index 4adfd13142..ad61cbd876 100644 --- a/tests/examples/jack/Makefile.am +++ b/tests/examples/jack/Makefile.am @@ -1,4 +1,10 @@ -noinst_PROGRAMS = jack_client +if HAVE_GTK +GTK_EXAMPLES=jack_client +else +GTK_EXAMPLES= +endif + +noinst_PROGRAMS = $(GTK_EXAMPLES) jack_client_SOURCES = jack_client.c jack_client_CFLAGS = $(GST_CFLAGS) $(GTK_CFLAGS) $(JACK_CFLAGS) From 1f6a83ce1db6e29ef86af431419fc41fa1d2b189 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Sun, 2 Jan 2011 15:27:19 +0000 Subject: [PATCH 205/351] docs: update plugin docs --- docs/plugins/gst-plugins-good-plugins.args | 178 +++++++++++++++- .../gst-plugins-good-plugins.hierarchy | 10 + .../gst-plugins-good-plugins.interfaces | 4 +- .../gst-plugins-good-plugins.prerequisites | 1 + docs/plugins/inspect/plugin-deinterlace.xml | 2 +- docs/plugins/inspect/plugin-matroska.xml | 4 +- docs/plugins/inspect/plugin-monoscope.xml | 4 +- docs/plugins/inspect/plugin-rtp.xml | 197 ++++++++++++------ 8 files changed, 322 insertions(+), 78 deletions(-) diff --git a/docs/plugins/gst-plugins-good-plugins.args b/docs/plugins/gst-plugins-good-plugins.args index d0ac7fe073..7e277a5ca7 100644 --- a/docs/plugins/gst-plugins-good-plugins.args +++ b/docs/plugins/gst-plugins-good-plugins.args @@ -745,7 +745,7 @@ rw Buffer Mode Control the buffering algorithm in use. -Slave receiver to sender clock +Choose mode depending on stream live @@ -1998,6 +1998,16 @@ "auto" + +GstProgressReport::do-query +gboolean + +rw +Use a query instead of buffer metadata to determine stream position +Use a query instead of buffer metadata to determine stream position. +TRUE + + GstNavSeek::seek-offset gdouble @@ -2688,6 +2698,16 @@ TRUE + +GstXImageSrc::remote +gboolean + +rw +Remote dispay +Whether the display is remote. +FALSE + + GstVideoBalance::brightness gdouble @@ -2768,6 +2788,16 @@ Faster, less accurate integer method + +GstJpegDec::max-errors +gint +>= G_MAXULONG +rw +Maximum Consecutive Decoding Errors +Error out after receiving N consecutive decoding errors (-1 = never fail, 0 = automatic, 1 = fail on first error). +0 + + GstRTPiLBCDepay::mode iLBCMode @@ -19674,7 +19704,7 @@ rw Byte Stream -Generate byte stream format of NALU. +Generate byte stream format of NALU (deprecated; use caps). TRUE @@ -19684,7 +19714,7 @@ rw Access Unit -Merge NALU into AU (picture). +Merge NALU into AU (picture) (deprecated; use caps). FALSE @@ -19838,6 +19868,16 @@ + +GstPulseSrc::client +gchar* + +rw +Client +The PulseAudio client_name_to_use. +"" + + GstPulseMixer::device gchar* @@ -20805,7 +20845,7 @@ rw Queue size Number of buffers to be enqueud in the driver in streaming mode. -8 +12 @@ -20848,6 +20888,56 @@ 0 + +GstV4l2Sink::crop-height +guint + +rw +Crop height +The height of the video crop; default is equal to negotiated image height. +0 + + + +GstV4l2Sink::crop-left +gint + +rw +Crop left +The leftmost (x) coordinate of the video crop; top left corner of image is 0,0. +0 + + + +GstV4l2Sink::crop-top +gint + +rw +Crop top +The topmost (y) coordinate of the video crop; top left corner of image is 0,0. +0 + + + +GstV4l2Sink::crop-width +guint + +rw +Crop width +The width of the video crop; default is equal to negotiated image width. +0 + + + +GstV4l2Sink::min-queued-bufs +guint +<= 16 +rw +Minimum queued bufs +Minimum number of queued bufs; v4l2sink won't dqbuf if the driver doesn't have more than this number (which normally you shouldn't change). +1 + + GstShapeWipe::border gfloat @@ -21018,3 +21108,83 @@ Checker pattern + +GstRtpJ2KPay::buffer-list +gboolean + +rw +Buffer List +Use Buffer Lists. +TRUE + + + +GstRtpJ2KDepay::buffer-list +gboolean + +rw +Buffer List +Use Buffer Lists. +TRUE + + + +GstJackAudioSrc::client +JackClient* + +rw +JackClient +Handle for jack client. + + + + +GstJackAudioSrc::connect +GstJackConnect + +rw +Connect +Specify how the input ports will be connected. +Automatically connect ports to physical ports + + + +GstJackAudioSrc::server +gchar* + +rw +Server +The Jack server to connect to (NULL = default). +NULL + + + +GstJackAudioSink::client +JackClient* + +rw +JackClient +Handle for jack client. + + + + +GstJackAudioSink::connect +GstJackConnect + +rw +Connect +Specify how the output ports will be connected. +Automatically connect ports to physical ports + + + +GstJackAudioSink::server +gchar* + +rw +Server +The Jack server to connect to (NULL = default). +NULL + + diff --git a/docs/plugins/gst-plugins-good-plugins.hierarchy b/docs/plugins/gst-plugins-good-plugins.hierarchy index 334128a2bc..5585c4a857 100644 --- a/docs/plugins/gst-plugins-good-plugins.hierarchy +++ b/docs/plugins/gst-plugins-good-plugins.hierarchy @@ -30,6 +30,7 @@ GObject GstRtpG723Depay GstRtpG726Depay GstRtpG729Depay + GstRtpGSTDepay GstRtpH263Depay GstRtpH263PDepay GstRtpH264Depay @@ -70,8 +71,10 @@ GObject GstRTPGSMPay GstRTPMP2TPay GstRTPMPVPay + GstRtpAC3Pay GstRtpAMRPay GstRtpCELTPay + GstRtpGSTPay GstRtpH263PPay GstRtpH263Pay GstRtpH264Pay @@ -92,6 +95,7 @@ GObject GstEsdSink GstOss4Sink GstOssSink + GstJackAudioSink GstPulseSink GstCACASink GstDynUDPSink @@ -110,6 +114,7 @@ GObject GstOss4Source GstOssSrc GstPulseSrc + GstJackAudioSrc GstDV1394Src GstHDV1394Src GstMultiFileSrc @@ -218,6 +223,7 @@ GObject GstMatroskaDemux GstMatroskaMux GstWebMMux + GstMonoscope GstMuLawDec GstMuLawEnc GstMultipartDemux @@ -269,6 +275,8 @@ GObject GstRingBuffer GstAudioSinkRingBuffer GstAudioSrcRingBuffer + GstJackAudioSinkRingBuffer + GstJackAudioSrcRingBuffer GstTask GstTaskPool GstSignalObject @@ -282,6 +290,7 @@ GInterface GstColorBalance GstImplementsInterface GstMixer + GstNavigation GstPreset GstPropertyProbe GstStreamVolume @@ -289,3 +298,4 @@ GInterface GstTuner GstURIHandler GstVideoOrientation + GstXOverlay diff --git a/docs/plugins/gst-plugins-good-plugins.interfaces b/docs/plugins/gst-plugins-good-plugins.interfaces index a7edf25999..d66eca41ca 100644 --- a/docs/plugins/gst-plugins-good-plugins.interfaces +++ b/docs/plugins/gst-plugins-good-plugins.interfaces @@ -19,7 +19,7 @@ GstRgVolume GstChildProxy GstAspectRatioCrop GstChildProxy GstPulseSink GstStreamVolume GstImplementsInterface GstPropertyProbe GstOss4Sink GstStreamVolume GstPropertyProbe -GstV4l2Sink GstImplementsInterface GstColorBalance GstVideoOrientation GstPropertyProbe +GstV4l2Sink GstImplementsInterface GstXOverlay GstNavigation GstColorBalance GstVideoOrientation GstPropertyProbe GstShout2send GstTagSetter GstUDPSink GstURIHandler GstDV1394Src GstURIHandler GstPropertyProbe @@ -28,7 +28,7 @@ GstSoupHTTPSrc GstURIHandler GstPulseSrc GstImplementsInterface GstMixer GstPropertyProbe GstOssSrc GstImplementsInterface GstMixer GstOss4Source GstImplementsInterface GstMixer GstPropertyProbe -GstV4l2Src GstURIHandler GstImplementsInterface GstTuner GstColorBalance GstVideoOrientation GstPropertyProbe +GstV4l2Src GstURIHandler GstImplementsInterface GstTuner GstXOverlay GstColorBalance GstVideoOrientation GstPropertyProbe GstUDPSrc GstURIHandler GstWavpackEnc GstPreset GstTagLibMux GstTagSetter diff --git a/docs/plugins/gst-plugins-good-plugins.prerequisites b/docs/plugins/gst-plugins-good-plugins.prerequisites index fa0a1d51c6..1f92056772 100644 --- a/docs/plugins/gst-plugins-good-plugins.prerequisites +++ b/docs/plugins/gst-plugins-good-plugins.prerequisites @@ -4,6 +4,7 @@ GstStreamVolume GObject GstImplementsInterface GstElement GstMixer GstImplementsInterface GstElement GstTuner GstImplementsInterface GstElement +GstXOverlay GstImplementsInterface GstElement GstColorBalance GstImplementsInterface GstElement GstVideoOrientation GstImplementsInterface GstElement GIcon GObject diff --git a/docs/plugins/inspect/plugin-deinterlace.xml b/docs/plugins/inspect/plugin-deinterlace.xml index 8c375e2405..1a0c5f2dc8 100644 --- a/docs/plugins/inspect/plugin-deinterlace.xml +++ b/docs/plugins/inspect/plugin-deinterlace.xml @@ -12,7 +12,7 @@ deinterlace Deinterlacer - Filter/Video + Filter/Effect/Video/Deinterlace Deinterlace Methods ported from DScaler/TvTime Martin Eikermann <meiker@upb.de>, Sebastian Dröge <sebastian.droege@collabora.co.uk> diff --git a/docs/plugins/inspect/plugin-matroska.xml b/docs/plugins/inspect/plugin-matroska.xml index 7e09a94a9d..09397b88fb 100644 --- a/docs/plugins/inspect/plugin-matroska.xml +++ b/docs/plugins/inspect/plugin-matroska.xml @@ -53,13 +53,13 @@ audio_%d sink request -
audio/mpeg, mpegversion=(int)1, layer=(int)[ 1, 3 ], stream-format=(string){ raw }, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/mpeg, mpegversion=(int){ 2, 4 }, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-ac3, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-vorbis, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-flac, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-speex, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw-int, width=(int)8, depth=(int)8, signed=(boolean)false, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw-int, width=(int)16, depth=(int)16, endianness=(int){ 4321, 1234 }, signed=(boolean)true, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw-int, width=(int)24, depth=(int)24, endianness=(int){ 4321, 1234 }, signed=(boolean)true, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw-int, width=(int)32, depth=(int)32, endianness=(int){ 4321, 1234 }, signed=(boolean)true, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw-float, width=(int)[ 32, 64 ], endianness=(int)1234, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-tta, width=(int){ 8, 16, 24 }, channels=(int){ 1, 2 }, rate=(int)[ 8000, 96000 ]; audio/x-pn-realaudio, raversion=(int){ 1, 2, 8 }, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-wma, wmaversion=(int)[ 1, 3 ], block_align=(int)[ 0, 65535 ], bitrate=(int)[ 0, 524288 ], channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]
+
audio/mpeg, mpegversion=(int)1, layer=(int)[ 1, 3 ], stream-format=(string){ raw }, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/mpeg, mpegversion=(int){ 2, 4 }, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-ac3, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-eac3, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-dts, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-vorbis, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-flac, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-speex, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw-int, width=(int)8, depth=(int)8, signed=(boolean)false, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw-int, width=(int)16, depth=(int)16, endianness=(int){ 4321, 1234 }, signed=(boolean)true, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw-int, width=(int)24, depth=(int)24, endianness=(int){ 4321, 1234 }, signed=(boolean)true, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw-int, width=(int)32, depth=(int)32, endianness=(int){ 4321, 1234 }, signed=(boolean)true, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw-float, width=(int)[ 32, 64 ], endianness=(int)1234, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-tta, width=(int){ 8, 16, 24 }, channels=(int){ 1, 2 }, rate=(int)[ 8000, 96000 ]; audio/x-pn-realaudio, raversion=(int){ 1, 2, 8 }, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-wma, wmaversion=(int)[ 1, 3 ], block_align=(int)[ 0, 65535 ], bitrate=(int)[ 0, 524288 ], channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]
subtitle_%d sink request -
ANY
+
subtitle/x-kate
 video_%d diff --git a/docs/plugins/inspect/plugin-monoscope.xml b/docs/plugins/inspect/plugin-monoscope.xml index 8d6951fdd4..05d39c6be3 100644 --- a/docs/plugins/inspect/plugin-monoscope.xml +++ b/docs/plugins/inspect/plugin-monoscope.xml @@ -3,10 +3,10 @@ Monoscope visualization ../../gst/monoscope/.libs/libgstmonoscope.so libgstmonoscope.so - 0.10.24.5 + 0.10.26.1 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-rtp.xml b/docs/plugins/inspect/plugin-rtp.xml index d3c27203f2..a471bd9ad3 100644 --- a/docs/plugins/inspect/plugin-rtp.xml +++ b/docs/plugins/inspect/plugin-rtp.xml @@ -12,7 +12,7 @@ asteriskh263 RTP Asterisk H263 depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts H263 video from RTP and encodes in Asterisk H263 format Neil Stratford <neils@vipadia.com> @@ -33,7 +33,7 @@ rtpL16depay RTP audio depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts raw audio from RTP packets Zeeshan Ali <zak147@yahoo.com>,Wim Taymans <wim.taymans@gmail.com> @@ -54,7 +54,7 @@ rtpL16pay RTP audio payloader - Codec/Payloader/Network + Codec/Payloader/Network/RTP Payload-encode Raw audio into RTP packets (RFC 3551) Wim Taymans <wim.taymans@gmail.com> @@ -75,7 +75,7 @@ rtpac3depay RTP AC3 depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts AC3 audio from RTP packets (RFC 4184) Wim Taymans <wim.taymans@gmail.com> @@ -93,10 +93,31 @@ + + rtpac3pay + RTP AC3 audio payloader + Codec/Payloader/Network/RTP + Payload AC3 audio as RTP packets (RFC 4184) + Wim Taymans <wim.taymans@gmail.com> + + + sink + sink + always +
audio/ac3; audio/x-ac3
+ + + src + source + always +
application/x-rtp, media=(string)audio, payload=(int)[ 96, 127 ], clock-rate=(int){ 32000, 44100, 48000 }, encoding-name=(string)AC3
+ + + rtpamrdepay RTP AMR depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts AMR or AMR-WB audio from RTP packets (RFC 3267) Wim Taymans <wim.taymans@gmail.com> @@ -117,7 +138,7 @@ rtpamrpay RTP AMR payloader - Codec/Payloader/Network + Codec/Payloader/Network/RTP Payload-encode AMR or AMR-WB audio into RTP packets (RFC 3267) Wim Taymans <wim.taymans@gmail.com> @@ -138,7 +159,7 @@ rtpbvdepay RTP BroadcomVoice depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts BroadcomVoice audio from RTP packets (RFC 4298) Wim Taymans <wim.taymans@collabora.co.uk> @@ -159,7 +180,7 @@ rtpbvpay RTP BV Payloader - Codec/Payloader/Network + Codec/Payloader/Network/RTP Packetize BroadcomVoice audio streams into RTP packets (RFC 4298) Wim Taymans <wim.taymans@collabora.co.uk> @@ -180,7 +201,7 @@ rtpceltdepay RTP CELT depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts CELT audio from RTP packets Wim Taymans <wim.taymans@gmail.com> @@ -201,7 +222,7 @@ rtpceltpay RTP CELT payloader - Codec/Payloader/Network + Codec/Payloader/Network/RTP Payload-encodes CELT audio into a RTP packet Wim Taymans <wim.taymans@gmail.com> @@ -222,7 +243,7 @@ rtpdepay Dummy RTP session manager - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Accepts raw RTP and RTCP packets and sends them forward Wim Taymans <wim.taymans@gmail.com> @@ -255,7 +276,7 @@ rtpdvdepay RTP DV Depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Depayloads DV from RTP packets (RFC 3189) Marcel Moreaux <marcelm@spacelabs.nl>, Wim Taymans <wim.taymans@gmail.com> @@ -276,7 +297,7 @@ rtpdvpay RTP DV Payloader - Codec/Payloader/Network + Codec/Payloader/Network/RTP Payloads DV into RTP packets (RFC 3189) Marcel Moreaux <marcelm@spacelabs.nl>, Wim Taymans <wim.taymans@gmail.com> @@ -297,7 +318,7 @@ rtpg722depay RTP audio depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts G722 audio from RTP packets Wim Taymans <wim.taymans@gmail.com> @@ -318,7 +339,7 @@ rtpg722pay RTP audio payloader - Codec/Payloader/Network + Codec/Payloader/Network/RTP Payload-encode Raw audio into RTP packets (RFC 3551) Wim Taymans <wim.taymans@gmail.com> @@ -339,7 +360,7 @@ rtpg723depay RTP G.723 depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts G.723 audio from RTP packets (RFC 3551) Wim Taymans <wim.taymans@gmail.com> @@ -360,7 +381,7 @@ rtpg723pay RTP G.723 payloader - Codec/Payloader/Network + Codec/Payloader/Network/RTP Packetize G.723 audio into RTP packets Wim Taymans <wim.taymans@gmail.com> @@ -381,7 +402,7 @@ rtpg726depay RTP G.726 depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts G.726 audio from RTP packets Axis Communications <dev-gstreamer@axis.com> @@ -402,7 +423,7 @@ rtpg726pay RTP G.726 payloader - Codec/Payloader/Network + Codec/Payloader/Network/RTP Payload-encodes G.726 audio into a RTP packet Axis Communications <dev-gstreamer@axis.com> @@ -423,7 +444,7 @@ rtpg729depay RTP G.729 depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts G.729 audio from RTP packets (RFC 3551) Laurent Glayal <spglegle@yahoo.fr> @@ -444,7 +465,7 @@ rtpg729pay RTP G.729 payloader - Codec/Payloader/Network + Codec/Payloader/Network/RTP Packetize G.729 audio into RTP packets Olivier Crete <olivier.crete@collabora.co.uk> @@ -465,7 +486,7 @@ rtpgsmdepay RTP GSM depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts GSM audio from RTP packets Zeeshan Ali <zeenix@gmail.com> @@ -486,7 +507,7 @@ rtpgsmpay RTP GSM payloader - Codec/Payloader/Network + Codec/Payloader/Network/RTP Payload-encodes GSM audio into a RTP packet Zeeshan Ali <zeenix@gmail.com> @@ -504,10 +525,52 @@ + + rtpgstdepay + GStreamer depayloader + Codec/Depayloader/Network + Extracts GStreamer buffers from RTP packets + Wim Taymans <wim.taymans@gmail.com> + + + sink + sink + always +
application/x-rtp, media=(string)application, payload=(int)[ 96, 127 ], clock-rate=(int)90000, encoding-name=(string)X-GST
+ + + src + source + always +
ANY
+ + + + + rtpgstpay + RTP GStreamer payloader + Codec/Payloader/Network/RTP + Payload GStreamer buffers as RTP packets + Wim Taymans <wim.taymans@gmail.com> + + + sink + sink + always +
ANY
+ + + src + source + always +
application/x-rtp, media=(string)application, payload=(int)[ 96, 127 ], clock-rate=(int)90000, encoding-name=(string)X-GST
+ + + rtph263depay RTP H263 depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts H263 video from RTP packets (RFC 2190) Philippe Kalaf <philippe.kalaf@collabora.co.uk>, Edward Hervey <bilboed@bilboed.com> @@ -528,7 +591,7 @@ rtph263pay RTP H263 packet payloader - Codec/Payloader/Network + Codec/Payloader/Network/RTP Payload-encodes H263 video in RTP packets (RFC 2190) Neil Stratford <neils@vipadia.com>Dejan Sakelsak <dejan.sakelsak@marand.si> @@ -549,7 +612,7 @@ rtph263pdepay RTP H263 depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts H263/+/++ video from RTP packets (RFC 4629) Wim Taymans <wim.taymans@gmail.com> @@ -570,7 +633,7 @@ rtph263ppay RTP H263 payloader - Codec/Payloader/Network + Codec/Payloader/Network/RTP Payload-encodes H263/+/++ video in RTP packets (RFC 4629) Wim Taymans <wim.taymans@gmail.com> @@ -591,7 +654,7 @@ rtph264depay RTP H264 depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts H264 video from RTP packets (RFC 3984) Wim Taymans <wim.taymans@gmail.com> @@ -612,7 +675,7 @@ rtph264pay RTP H264 payloader - Codec/Payloader/Network + Codec/Payloader/Network/RTP Payload-encode H264 video into RTP packets (RFC 3984) Laurent Glayal <spglegle@yahoo.fr> @@ -633,7 +696,7 @@ rtpilbcdepay RTP iLBC depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts iLBC audio from RTP packets (RFC 3952) Philippe Kalaf <philippe.kalaf@collabora.co.uk> @@ -654,7 +717,7 @@ rtpilbcpay RTP iLBC Payloader - Codec/Payloader/Network + Codec/Payloader/Network/RTP Packetize iLBC audio streams into RTP packets Philippe Kalaf <philippe.kalaf@collabora.co.uk> @@ -675,7 +738,7 @@ rtpj2kdepay RTP JPEG 2000 depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts JPEG 2000 video from RTP packets (RFC 5371) Wim Taymans <wim.taymans@gmail.com> @@ -696,7 +759,7 @@ rtpj2kpay RTP JPEG 2000 payloader - Codec/Payloader/Network + Codec/Payloader/Network/RTP Payload-encodes JPEG 2000 pictures into RTP packets (RFC 5371) Wim Taymans <wim.taymans@gmail.com> @@ -717,7 +780,7 @@ rtpjpegdepay RTP JPEG depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts JPEG video from RTP packets (RFC 2435) Wim Taymans <wim.taymans@gmail.com> @@ -738,7 +801,7 @@ rtpjpegpay RTP JPEG payloader - Codec/Payloader/Network + Codec/Payloader/Network/RTP Payload-encodes JPEG pictures into RTP packets (RFC 2435) Axis Communications <dev-gstreamer@axis.com> @@ -759,7 +822,7 @@ rtpmp1sdepay RTP MPEG1 System Stream depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts MPEG1 System Streams from RTP packets (RFC 3555) Wim Taymans <wim.taymans@gmail.com> @@ -780,7 +843,7 @@ rtpmp2tdepay RTP MPEG Transport Stream depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts MPEG2 TS from RTP packets (RFC 2250) Wim Taymans <wim.taymans@gmail.com>, Thijs Vermeir <thijs.vermeir@barco.com> @@ -801,7 +864,7 @@ rtpmp2tpay RTP MPEG2 Transport Stream payloader - Codec/Payloader/Network + Codec/Payloader/Network/RTP Payload-encodes MPEG2 TS into RTP packets (RFC 2250) Wim Taymans <wim.taymans@gmail.com> @@ -822,7 +885,7 @@ rtpmp4adepay RTP MPEG4 audio depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts MPEG4 audio from RTP packets (RFC 3016) Nokia Corporation (contact <stefan.kost@nokia.com>), Wim Taymans <wim.taymans@gmail.com> @@ -843,7 +906,7 @@ rtpmp4apay RTP MPEG4 audio payloader - Codec/Payloader/Network + Codec/Payloader/Network/RTP Payload MPEG4 audio as RTP packets (RFC 3016) Wim Taymans <wim.taymans@gmail.com> @@ -864,7 +927,7 @@ rtpmp4gdepay RTP MPEG4 ES depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts MPEG4 elementary streams from RTP packets (RFC 3640) Wim Taymans <wim.taymans@gmail.com> @@ -885,7 +948,7 @@ rtpmp4gpay RTP MPEG4 ES payloader - Codec/Payloader/Network + Codec/Payloader/Network/RTP Payload MPEG4 elementary streams as RTP packets (RFC 3640) Wim Taymans <wim.taymans@gmail.com> @@ -906,7 +969,7 @@ rtpmp4vdepay RTP MPEG4 video depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts MPEG4 video from RTP packets (RFC 3016) Wim Taymans <wim.taymans@gmail.com> @@ -927,7 +990,7 @@ rtpmp4vpay RTP MPEG4 Video payloader - Codec/Payloader/Network + Codec/Payloader/Network/RTP Payload MPEG-4 video as RTP packets (RFC 3016) Wim Taymans <wim.taymans@gmail.com> @@ -935,7 +998,7 @@ sink sink always -
video/mpeg, mpegversion=(int)4, systemstream=(boolean)false
+
video/mpeg, mpegversion=(int)4, systemstream=(boolean)false; video/x-xvid
src @@ -948,7 +1011,7 @@ rtpmpadepay RTP MPEG audio depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts MPEG audio from RTP packets (RFC 2038) Wim Taymans <wim.taymans@gmail.com> @@ -969,7 +1032,7 @@ rtpmpapay RTP MPEG audio payloader - Codec/Payloader/Network + Codec/Payloader/Network/RTP Payload MPEG audio as RTP packets (RFC 2038) Wim Taymans <wim.taymans@gmail.com> @@ -990,7 +1053,7 @@ rtpmparobustdepay RTP MPEG audio depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts MPEG audio from RTP packets (RFC 5219) Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk> @@ -1011,7 +1074,7 @@ rtpmpvdepay RTP MPEG video depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts MPEG video from RTP packets (RFC 2250) Wim Taymans <wim.taymans@gmail.com> @@ -1032,7 +1095,7 @@ rtpmpvpay RTP MPEG2 ES video payloader - Codec/Payloader/Network + Codec/Payloader/Network/RTP Payload-encodes MPEG2 ES into RTP packets (RFC 2250) Thijs Vermeir <thijsvermeir@gmail.com> @@ -1053,7 +1116,7 @@ rtppcmadepay RTP PCMA depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts PCMA audio from RTP packets Edgard Lima <edgard.lima@indt.org.br>, Zeeshan Ali <zeenix@gmail.com> @@ -1074,7 +1137,7 @@ rtppcmapay RTP PCMA payloader - Codec/Payloader/Network + Codec/Payloader/Network/RTP Payload-encodes PCMA audio into a RTP packet Edgard Lima <edgard.lima@indt.org.br> @@ -1095,7 +1158,7 @@ rtppcmudepay RTP PCMU depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts PCMU audio from RTP packets Edgard Lima <edgard.lima@indt.org.br>, Zeeshan Ali <zeenix@gmail.com> @@ -1116,7 +1179,7 @@ rtppcmupay RTP PCMU payloader - Codec/Payloader/Network + Codec/Payloader/Network/RTP Payload-encodes PCMU audio into a RTP packet Edgard Lima <edgard.lima@indt.org.br> @@ -1137,7 +1200,7 @@ rtpqcelpdepay RTP QCELP depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts QCELP (PureVoice) audio from RTP packets (RFC 2658) Wim Taymans <wim.taymans@gmail.com> @@ -1158,7 +1221,7 @@ rtpqdm2depay RTP QDM2 depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts QDM2 audio from RTP packets (no RFC) Edward Hervey <bilboed@bilboed.com> @@ -1179,7 +1242,7 @@ rtpsirendepay RTP Siren packet depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts Siren audio from RTP packets Philippe Kalaf <philippe.kalaf@collabora.co.uk> @@ -1200,7 +1263,7 @@ rtpsirenpay RTP Payloader for Siren Audio - Codec/Payloader/Network + Codec/Payloader/Network/RTP Packetize Siren audio streams into RTP packets Youness Alaoui <kakaroto@kakaroto.homelinux.net> @@ -1221,7 +1284,7 @@ rtpspeexdepay RTP Speex depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts Speex audio from RTP packets Edgard Lima <edgard.lima@indt.org.br> @@ -1242,7 +1305,7 @@ rtpspeexpay RTP Speex payloader - Codec/Payloader/Network + Codec/Payloader/Network/RTP Payload-encodes Speex audio into a RTP packet Edgard Lima <edgard.lima@indt.org.br> @@ -1263,7 +1326,7 @@ rtpsv3vdepay RTP SVQ3 depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts SVQ3 video from RTP packets (no RFC) Wim Taymans <wim.taymans@gmail.com> @@ -1284,7 +1347,7 @@ rtptheoradepay RTP Theora depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts Theora video from RTP packets (draft-01 of RFC XXXX) Wim Taymans <wim.taymans@gmail.com> @@ -1305,7 +1368,7 @@ rtptheorapay RTP Theora payloader - Codec/Payloader/Network + Codec/Payloader/Network/RTP Payload-encode Theora video into RTP packets (draft-01 RFC XXXX) Wim Taymans <wim.taymans@gmail.com> @@ -1326,7 +1389,7 @@ rtpvorbisdepay RTP Vorbis depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts Vorbis Audio from RTP packets (RFC 5215) Wim Taymans <wim.taymans@gmail.com> @@ -1347,7 +1410,7 @@ rtpvorbispay RTP Vorbis depayloader - Codec/Payloader/Network + Codec/Payloader/Network/RTP Payload-encode Vorbis audio into RTP packets (RFC 5215) Wim Taymans <wimi.taymans@gmail.com> @@ -1368,7 +1431,7 @@ rtpvrawdepay RTP Raw Video depayloader - Codec/Depayloader/Network + Codec/Depayloader/Network/RTP Extracts raw video from RTP packets (RFC 4175) Wim Taymans <wim.taymans@gmail.com> @@ -1389,7 +1452,7 @@ rtpvrawpay RTP Raw Video payloader - Codec/Payloader/Network + Codec/Payloader/Network/RTP Payload raw video as RTP packets (RFC 4175) Wim Taymans <wim.taymans@gmail.com> From 1f5083a17cf48c0197ceac71c52d46d1831b3388 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Sun, 2 Jan 2011 17:07:19 +0000 Subject: [PATCH 206/351] jack: make get_type functions thread-safe Because we can (shouldn't be needed with other workarounds still there). --- ext/jack/gstjack.c | 38 +++++++++++++++++++------------------ ext/jack/gstjackaudiosink.c | 12 ++++++------ ext/jack/gstjackaudiosrc.c | 12 ++++++------ 3 files changed, 32 insertions(+), 30 deletions(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 371a9e934f..74b8ddf079 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -27,22 +27,23 @@ GType gst_jack_connect_get_type (void) { - static GType jack_connect_type = 0; - static const GEnumValue jack_connect[] = { - {GST_JACK_CONNECT_NONE, - "Don't automatically connect ports to physical ports", "none"}, - {GST_JACK_CONNECT_AUTO, - "Automatically connect ports to physical ports", "auto"}, - {GST_JACK_CONNECT_AUTO_FORCED, - "Automatically connect ports to as many physical ports as possible", - "auto-forced"}, - {0, NULL, NULL}, - }; + static volatile gsize jack_connect_type = 0; - if (!jack_connect_type) { - jack_connect_type = g_enum_register_static ("GstJackConnect", jack_connect); + if (g_once_init_enter (&jack_connect_type)) { + static const GEnumValue jack_connect_enums[] = { + {GST_JACK_CONNECT_NONE, + "Don't automatically connect ports to physical ports", "none"}, + {GST_JACK_CONNECT_AUTO, + "Automatically connect ports to physical ports", "auto"}, + {GST_JACK_CONNECT_AUTO_FORCED, + "Automatically connect ports to as many physical ports as possible", + "auto-forced"}, + {0, NULL, NULL}, + }; + GType tmp = g_enum_register_static ("GstJackConnect", jack_connect_enums); + g_once_init_leave (&jack_connect_type, tmp); } - return jack_connect_type; + return (GType) jack_connect_type; } @@ -63,16 +64,17 @@ gst_jack_client_free (gpointer jclient) GType gst_jack_client_get_type (void) { - static GType type; /* 0 */ + static volatile gsize jack_client_type = 0; - if (type == 0) { + if (g_once_init_enter (&jack_client_type)) { /* hackish, but makes it show up nicely in gst-inspect */ - type = g_boxed_type_register_static ("JackClient", + GType tmp = g_boxed_type_register_static ("JackClient", (GBoxedCopyFunc) gst_jack_client_copy, (GBoxedFreeFunc) gst_jack_client_free); + g_once_init_leave (&jack_client_type, tmp); } - return type; + return (GType) jack_client_type; } static gboolean diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c index 32bf1af3e4..ea75b9cd45 100644 --- a/ext/jack/gstjackaudiosink.c +++ b/ext/jack/gstjackaudiosink.c @@ -129,9 +129,9 @@ gst_jack_audio_sink_free_channels (GstJackAudioSink * sink) static GType gst_jack_ring_buffer_get_type (void) { - static GType ringbuffer_type = 0; + static volatile gsize ringbuffer_type = 0; - if (!ringbuffer_type) { + if (g_once_init_enter (&ringbuffer_type)) { static const GTypeInfo ringbuffer_info = { sizeof (GstJackRingBufferClass), NULL, @@ -144,12 +144,12 @@ gst_jack_ring_buffer_get_type (void) (GInstanceInitFunc) gst_jack_ring_buffer_init, NULL }; - - ringbuffer_type = - g_type_register_static (GST_TYPE_RING_BUFFER, + GType tmp = g_type_register_static (GST_TYPE_RING_BUFFER, "GstJackAudioSinkRingBuffer", &ringbuffer_info, 0); + g_once_init_leave (&ringbuffer_type, tmp); } - return ringbuffer_type; + + return (GType) ringbuffer_type; } static void diff --git a/ext/jack/gstjackaudiosrc.c b/ext/jack/gstjackaudiosrc.c index 0ffdb23985..57ec66ab85 100644 --- a/ext/jack/gstjackaudiosrc.c +++ b/ext/jack/gstjackaudiosrc.c @@ -151,9 +151,9 @@ gst_jack_audio_src_free_channels (GstJackAudioSrc * src) static GType gst_jack_ring_buffer_get_type (void) { - static GType ringbuffer_type = 0; + static volatile gsize ringbuffer_type = 0; - if (!ringbuffer_type) { + if (g_once_init_enter (&ringbuffer_type)) { static const GTypeInfo ringbuffer_info = { sizeof (GstJackRingBufferClass), NULL, NULL, @@ -165,12 +165,12 @@ gst_jack_ring_buffer_get_type (void) (GInstanceInitFunc) gst_jack_ring_buffer_init, NULL }; - - ringbuffer_type = - g_type_register_static (GST_TYPE_RING_BUFFER, + GType tmp = g_type_register_static (GST_TYPE_RING_BUFFER, "GstJackAudioSrcRingBuffer", &ringbuffer_info, 0); + g_once_init_leave (&ringbuffer_type, tmp); } - return ringbuffer_type; + + return (GType) ringbuffer_type; } static void From 79a8e0ee4298c029b2959d4e2ab0a04ba4a63456 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Sun, 2 Jan 2011 17:40:41 +0000 Subject: [PATCH 207/351] jack: template caps had lists with one value, just use value directly --- ext/jack/gstjackaudiosink.c | 2 +- ext/jack/gstjackaudiosrc.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c index ea75b9cd45..b160bbfd34 100644 --- a/ext/jack/gstjackaudiosink.c +++ b/ext/jack/gstjackaudiosink.c @@ -614,7 +614,7 @@ GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, " - "endianness = (int) { " G_STRINGIFY (G_BYTE_ORDER) " }, " + "endianness = (int) BYTE_ORDER, " "width = (int) 32, " "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]") ); diff --git a/ext/jack/gstjackaudiosrc.c b/ext/jack/gstjackaudiosrc.c index 57ec66ab85..f96fb813af 100644 --- a/ext/jack/gstjackaudiosrc.c +++ b/ext/jack/gstjackaudiosrc.c @@ -649,7 +649,7 @@ static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, " - "endianness = (int) { " G_STRINGIFY (G_BYTE_ORDER) " }, " + "endianness = (int) BYTE_ORDER, " "width = (int) 32, " "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]") ); From 5b79a3d2f89eb8ed155268b9507125008b288a1a Mon Sep 17 00:00:00 2001 From: Erich Schubert Date: Sun, 2 Jan 2011 19:23:51 +0000 Subject: [PATCH 208/351] id3demux: fix parsing of ID3v2.4 genre frames with multiple genres We'd only extract the first genre (multiple times) instead of all genres. https://bugzilla.gnome.org/show_bug.cgi?id=638535 --- gst/id3demux/id3v2frames.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/gst/id3demux/id3v2frames.c b/gst/id3demux/id3v2frames.c index 021a3d173c..e51bbb78ec 100644 --- a/gst/id3demux/id3v2frames.c +++ b/gst/id3demux/id3v2frames.c @@ -875,7 +875,7 @@ id3v2_genre_fields_to_taglist (ID3TagsWorking * work, const gchar * tag_name, for (i = 0; i < tag_fields->len; i++) { gint len; - tag_str = g_array_index (tag_fields, gchar *, 0); + tag_str = g_array_index (tag_fields, gchar *, i); if (tag_str == NULL) continue; From 57e35559c814527333de3d5e13be6a74a423dc5a Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Sun, 2 Jan 2011 19:43:02 +0000 Subject: [PATCH 209/351] jack: fix up element details and some other minor clean-ups --- ext/jack/Makefile.am | 4 +--- ext/jack/README | 4 ---- ext/jack/gstjack.c | 2 +- ext/jack/gstjackaudiosink.c | 3 ++- ext/jack/gstjackaudiosrc.c | 4 ++-- 5 files changed, 6 insertions(+), 11 deletions(-) delete mode 100644 ext/jack/README diff --git a/ext/jack/Makefile.am b/ext/jack/Makefile.am index ede70594bd..cf7789971a 100644 --- a/ext/jack/Makefile.am +++ b/ext/jack/Makefile.am @@ -2,11 +2,9 @@ plugin_LTLIBRARIES = libgstjack.la libgstjack_la_SOURCES = gstjackutil.c gstjack.c gstjackaudiosrc.c gstjackaudiosink.c gstjackaudioclient.c -libgstjack_la_CFLAGS = $(GST_PLUGINS_BAD_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) $(JACK_CFLAGS) +libgstjack_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) $(JACK_CFLAGS) libgstjack_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR) $(JACK_LIBS) libgstjack_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) libgstjack_la_LIBTOOLFLAGS = --tag=disable-static noinst_HEADERS = gstjackutil.h gstjackaudiosrc.h gstjackaudiosink.h gstjackaudioclient.h gstjack.h gstjackringbuffer.h - -EXTRA_DIST = README diff --git a/ext/jack/README b/ext/jack/README deleted file mode 100644 index 2bb97bebad..0000000000 --- a/ext/jack/README +++ /dev/null @@ -1,4 +0,0 @@ -to be written, la dee da - -jackit.sf.net - diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 74b8ddf079..8180afbb3c 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -93,5 +93,5 @@ plugin_init (GstPlugin * plugin) GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, "jack", - "Jack elements", + "JACK audio elements", plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN) diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c index b160bbfd34..4620bce712 100644 --- a/ext/jack/gstjackaudiosink.c +++ b/ext/jack/gstjackaudiosink.c @@ -660,7 +660,8 @@ gst_jack_audio_sink_base_init (gpointer g_class) GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); gst_element_class_set_details_simple (element_class, "Audio Sink (Jack)", - "Sink/Audio", "Output to Jack", "Wim Taymans "); + "Sink/Audio", "Output audio to a JACK server", + "Wim Taymans "); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&jackaudiosink_sink_factory)); diff --git a/ext/jack/gstjackaudiosrc.c b/ext/jack/gstjackaudiosrc.c index f96fb813af..08b325ead7 100644 --- a/ext/jack/gstjackaudiosrc.c +++ b/ext/jack/gstjackaudiosrc.c @@ -680,8 +680,8 @@ gst_jack_audio_src_base_init (gpointer gclass) gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&src_factory)); gst_element_class_set_details_simple (element_class, "Audio Source (Jack)", - "Source/Audio", - "Input from Jack", "Tristan Matthews "); + "Source/Audio", "Captures audio from a JACK server", + "Tristan Matthews "); } /* initialize the jack_audio_src's class */ From 6ab508dfa56f92c86c3b379b6cfe4ff28f2551e4 Mon Sep 17 00:00:00 2001 From: David Schleef Date: Sun, 2 Jan 2011 19:19:27 -0800 Subject: [PATCH 210/351] multifilesink: send stream headers in key-frame mode --- gst/multifile/gstmultifilesink.c | 63 ++++++++++++++++++++++++++++++++ gst/multifile/gstmultifilesink.h | 3 ++ 2 files changed, 66 insertions(+) diff --git a/gst/multifile/gstmultifilesink.c b/gst/multifile/gstmultifilesink.c index 2d4ab5b487..b333e809d1 100644 --- a/gst/multifile/gstmultifilesink.c +++ b/gst/multifile/gstmultifilesink.c @@ -146,6 +146,8 @@ static void gst_multi_file_sink_get_property (GObject * object, guint prop_id, static gboolean gst_multi_file_sink_stop (GstBaseSink * sink); static GstFlowReturn gst_multi_file_sink_render (GstBaseSink * sink, GstBuffer * buffer); +static gboolean gst_multi_file_sink_set_caps (GstBaseSink * sink, + GstCaps * caps); #define GST_TYPE_MULTI_FILE_SINK_NEXT (gst_multi_file_sink_next_get_type ()) static GType @@ -237,6 +239,8 @@ gst_multi_file_sink_class_init (GstMultiFileSinkClass * klass) gstbasesink_class->get_times = NULL; gstbasesink_class->stop = GST_DEBUG_FUNCPTR (gst_multi_file_sink_stop); gstbasesink_class->render = GST_DEBUG_FUNCPTR (gst_multi_file_sink_render); + gstbasesink_class->set_caps = + GST_DEBUG_FUNCPTR (gst_multi_file_sink_set_caps); } static void @@ -327,6 +331,7 @@ static gboolean gst_multi_file_sink_stop (GstBaseSink * sink) { GstMultiFileSink *multifilesink; + int i; multifilesink = GST_MULTI_FILE_SINK (sink); @@ -335,6 +340,13 @@ gst_multi_file_sink_stop (GstBaseSink * sink) multifilesink->file = NULL; } + if (multifilesink->streamheaders) { + for (i = 0; i < multifilesink->n_streamheaders; i++) { + gst_buffer_unref (multifilesink->streamheaders[i]); + } + g_free (multifilesink->streamheaders); + } + return TRUE; } @@ -462,6 +474,8 @@ gst_multi_file_sink_render (GstBaseSink * sink, GstBuffer * buffer) } if (multifilesink->file == NULL) { + int i; + filename = g_strdup_printf (multifilesink->filename, multifilesink->index); multifilesink->file = g_fopen (filename, "wb"); @@ -469,6 +483,16 @@ gst_multi_file_sink_render (GstBaseSink * sink, GstBuffer * buffer) if (multifilesink->file == NULL) goto stdio_write_error; + + if (multifilesink->streamheaders) { + for (i = 0; i < multifilesink->n_streamheaders; i++) { + ret = fwrite (GST_BUFFER_DATA (multifilesink->streamheaders[i]), + GST_BUFFER_SIZE (multifilesink->streamheaders[i]), 1, + multifilesink->file); + if (ret != 1) + goto stdio_write_error; + } + } } ret = fwrite (GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer), 1, @@ -508,3 +532,42 @@ stdio_write_error: ("Error while writing to file."), (NULL)); return GST_FLOW_ERROR; } + +static gboolean +gst_multi_file_sink_set_caps (GstBaseSink * sink, GstCaps * caps) +{ + GstMultiFileSink *multifilesink; + GstStructure *structure; + + multifilesink = GST_MULTI_FILE_SINK (sink); + + structure = gst_caps_get_structure (caps, 0); + if (structure) { + const GValue *value; + + value = gst_structure_get_value (structure, "streamheader"); + + if (GST_VALUE_HOLDS_ARRAY (value)) { + int i; + + if (multifilesink->streamheaders) { + for (i = 0; i < multifilesink->n_streamheaders; i++) { + gst_buffer_unref (multifilesink->streamheaders[i]); + } + g_free (multifilesink->streamheaders); + } + + multifilesink->n_streamheaders = gst_value_array_get_size (value); + multifilesink->streamheaders = + g_malloc (sizeof (GstBuffer *) * multifilesink->n_streamheaders); + + for (i = 0; i < multifilesink->n_streamheaders; i++) { + multifilesink->streamheaders[i] = + gst_buffer_ref (gst_value_get_buffer (gst_value_array_get_value + (value, i))); + } + } + } + + return TRUE; +} diff --git a/gst/multifile/gstmultifilesink.h b/gst/multifile/gstmultifilesink.h index cc4a4ddf45..c5515706a8 100644 --- a/gst/multifile/gstmultifilesink.h +++ b/gst/multifile/gstmultifilesink.h @@ -69,6 +69,9 @@ struct _GstMultiFileSink FILE *file; gint64 next_segment; + + int n_streamheaders; + GstBuffer **streamheaders; }; struct _GstMultiFileSinkClass From dc221c02193bb4740be7cdd3fc2addbcc5e358ae Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Mon, 3 Jan 2011 15:40:11 +0100 Subject: [PATCH 211/351] rtspsrc: increase udp buffer size Set a bigger UDP buffer size by default to reduce packet loss with high bitrate streams. --- gst/rtsp/gstrtspsrc.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c index 7e003d3c87..15bf5cfc7f 100644 --- a/gst/rtsp/gstrtspsrc.c +++ b/gst/rtsp/gstrtspsrc.c @@ -161,7 +161,7 @@ gst_rtsp_src_buffer_mode_get_type (void) #define DEFAULT_DEBUG FALSE #define DEFAULT_RETRY 20 #define DEFAULT_TIMEOUT 5000000 -#define DEFAULT_UDP_BUFFER_SIZE 0 +#define DEFAULT_UDP_BUFFER_SIZE 0x80000 #define DEFAULT_TCP_TIMEOUT 20000000 #define DEFAULT_LATENCY_MS 2000 #define DEFAULT_CONNECTION_SPEED 0 @@ -1494,6 +1494,10 @@ again: goto no_udp_protocol; g_object_set (G_OBJECT (udpsrc0), "port", tmp_rtp, "reuse", FALSE, NULL); + if (src->udp_buffer_size != 0) + g_object_set (G_OBJECT (udpsrc0), "buffer-size", src->udp_buffer_size, + NULL); + ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED); if (ret == GST_STATE_CHANGE_FAILURE) { if (tmp_rtp != 0) { @@ -2720,8 +2724,6 @@ gst_rtspsrc_stream_configure_udp (GstRTSPSrc * src, GstRTSPStream * stream, * if we can. */ g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", src->udp_timeout, NULL); - g_object_set (G_OBJECT (stream->udpsrc[0]), "buffer-size", - src->udp_buffer_size, NULL); /* get output pad of the UDP source. */ *outpad = gst_element_get_static_pad (stream->udpsrc[0], "src"); From 06b68c81b4f8e62c9b8942f0ea4ce7bcadfcbef0 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Mon, 3 Jan 2011 01:18:06 +0000 Subject: [PATCH 212/351] v4l2src: undo presumably accidental enablement of the GstXOverlay interface Looks like this got enabled by accident when adding it to v4l2sink, so undo this for now. Not sure it makes much sense in a GStreamer context with current hardware. --- sys/v4l2/gstv4l2src.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sys/v4l2/gstv4l2src.c b/sys/v4l2/gstv4l2src.c index ed3b2ac6ef..4a37d354b9 100644 --- a/sys/v4l2/gstv4l2src.c +++ b/sys/v4l2/gstv4l2src.c @@ -44,6 +44,8 @@ #include #endif +#undef HAVE_XVIDEO + #include #include #include "v4l2src_calls.h" @@ -143,6 +145,7 @@ gst_v4l2src_init_interfaces (GType type) NULL, }; #ifdef HAVE_XVIDEO + /* FIXME: does GstXOverlay for v4l2src make sense in a GStreamer context? */ static const GInterfaceInfo v4l2_xoverlay_info = { (GInterfaceInitFunc) gst_v4l2src_xoverlay_interface_init, NULL, From c17f7a943d54c7a3855a25818bf1b4cc79300406 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Mon, 3 Jan 2011 18:05:15 +0100 Subject: [PATCH 213/351] cacasink: fix masks and strides Use the right endianness to read the masks. Use the right strides for the bitmap. Fixes #638569 --- ext/libcaca/gstcacasink.c | 18 +++++++++++++----- 1 file changed, 13 insertions(+), 5 deletions(-) diff --git a/ext/libcaca/gstcacasink.c b/ext/libcaca/gstcacasink.c index ee05b36cc4..1846c26d48 100644 --- a/ext/libcaca/gstcacasink.c +++ b/ext/libcaca/gstcacasink.c @@ -209,12 +209,14 @@ gst_cacasink_setcaps (GstBaseSink * basesink, GstCaps * caps) { GstCACASink *cacasink; GstStructure *structure; + gint endianness; cacasink = GST_CACASINK (basesink); structure = gst_caps_get_structure (caps, 0); gst_structure_get_int (structure, "width", &(cacasink->width)); gst_structure_get_int (structure, "height", &(cacasink->height)); + gst_structure_get_int (structure, "endianness", &endianness); gst_structure_get_int (structure, "bpp", (int *) &cacasink->bpp); gst_structure_get_int (structure, "red_mask", (int *) &cacasink->red_mask); gst_structure_get_int (structure, "green_mask", @@ -233,10 +235,16 @@ gst_cacasink_setcaps (GstBaseSink * basesink, GstCaps * caps) cacasink->blue_mask = GUINT32_FROM_BE (cacasink->blue_mask); } - else if (cacasink->bpp == 16 || cacasink->bpp == 15) { - cacasink->red_mask = GUINT16_FROM_BE (cacasink->red_mask); - cacasink->green_mask = GUINT16_FROM_BE (cacasink->green_mask); - cacasink->blue_mask = GUINT16_FROM_BE (cacasink->blue_mask); + else if (cacasink->bpp == 16) { + if (endianness == G_BIG_ENDIAN) { + cacasink->red_mask = GUINT16_FROM_BE (cacasink->red_mask); + cacasink->green_mask = GUINT16_FROM_BE (cacasink->green_mask); + cacasink->blue_mask = GUINT16_FROM_BE (cacasink->blue_mask); + } else { + cacasink->red_mask = GUINT16_FROM_LE (cacasink->red_mask); + cacasink->green_mask = GUINT16_FROM_LE (cacasink->green_mask); + cacasink->blue_mask = GUINT16_FROM_LE (cacasink->blue_mask); + } } if (cacasink->bitmap) { @@ -246,7 +254,7 @@ gst_cacasink_setcaps (GstBaseSink * basesink, GstCaps * caps) cacasink->bitmap = caca_create_bitmap (cacasink->bpp, cacasink->width, cacasink->height, - cacasink->width * cacasink->bpp / 8, + GST_ROUND_UP_4 (cacasink->width * cacasink->bpp / 8), cacasink->red_mask, cacasink->green_mask, cacasink->blue_mask, 0); if (!cacasink->bitmap) { From a570310cdc82c339c579ef42b9e7dbd23fc13c72 Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Mon, 3 Jan 2011 20:06:35 +0200 Subject: [PATCH 214/351] matroska-demux: remove unused variables when debug-logging disabled --- gst/matroska/matroska-demux.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/gst/matroska/matroska-demux.c b/gst/matroska/matroska-demux.c index 29ba781ad0..7bca5d19fa 100644 --- a/gst/matroska/matroska-demux.c +++ b/gst/matroska/matroska-demux.c @@ -4386,7 +4386,6 @@ static GstFlowReturn gst_matroska_demux_add_mpeg_seq_header (GstElement * element, GstMatroskaTrackContext * stream, GstBuffer ** buf) { - GstMatroskaDemux *demux = GST_MATROSKA_DEMUX (element); guint8 *seq_header; guint seq_header_len; guint32 header; @@ -4416,7 +4415,7 @@ gst_matroska_demux_add_mpeg_seq_header (GstElement * element, newbuf = gst_buffer_new_and_alloc (GST_BUFFER_SIZE (*buf) + seq_header_len); gst_buffer_set_caps (newbuf, stream->caps); - GST_DEBUG_OBJECT (demux, "Prepending MPEG sequence header"); + GST_DEBUG_OBJECT (element, "Prepending MPEG sequence header"); gst_buffer_copy_metadata (newbuf, *buf, GST_BUFFER_COPY_TIMESTAMPS | GST_BUFFER_COPY_FLAGS); g_memmove (GST_BUFFER_DATA (newbuf), seq_header, seq_header_len); @@ -4433,7 +4432,6 @@ static GstFlowReturn gst_matroska_demux_add_wvpk_header (GstElement * element, GstMatroskaTrackContext * stream, GstBuffer ** buf) { - GstMatroskaDemux *demux = GST_MATROSKA_DEMUX (element); GstMatroskaTrackAudioContext *audiocontext = (GstMatroskaTrackAudioContext *) stream; GstBuffer *newbuf = NULL; @@ -4495,7 +4493,7 @@ gst_matroska_demux_add_wvpk_header (GstElement * element, size = GST_BUFFER_SIZE (*buf); if (size < 4) { - GST_ERROR_OBJECT (demux, "Too small wavpack buffer"); + GST_ERROR_OBJECT (element, "Too small wavpack buffer"); return GST_FLOW_ERROR; } From c9e0db646955b80c8073f2a257e40da86be12d48 Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Mon, 3 Jan 2011 20:16:22 +0200 Subject: [PATCH 215/351] rtspsrc: remove unused variables when debug-logging disabled --- gst/rtsp/gstrtspsrc.c | 11 ++++------- 1 file changed, 4 insertions(+), 7 deletions(-) diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c index 15bf5cfc7f..92c4efc560 100644 --- a/gst/rtsp/gstrtspsrc.c +++ b/gst/rtsp/gstrtspsrc.c @@ -2247,9 +2247,7 @@ unknown_stream: static void gst_rtspsrc_do_stream_eos (GstRTSPSrc * src, GstRTSPStream * stream) { - guint session = stream->id; - - GST_DEBUG_OBJECT (src, "setting stream for session %u to EOS", session); + GST_DEBUG_OBJECT (src, "setting stream for session %u to EOS", stream->id); if (stream->eos) goto was_eos; @@ -2261,7 +2259,7 @@ gst_rtspsrc_do_stream_eos (GstRTSPSrc * src, GstRTSPStream * stream) /* ERRORS */ was_eos: { - GST_DEBUG_OBJECT (src, "stream for session %u was already EOS", session); + GST_DEBUG_OBJECT (src, "stream for session %u was already EOS", stream->id); return; } } @@ -2303,9 +2301,8 @@ on_npt_stop (GstElement * rtpbin, guint session, guint ssrc, GstRTSPSrc * src) static void on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream) { - GstRTSPSrc *src = stream->parent; - - GST_DEBUG_OBJECT (src, "source in session %u is active", stream->id); + GST_DEBUG_OBJECT (stream->parent, "source in session %u is active", + stream->id); } /* try to get and configure a manager */ From 43c6d0c72afba986a3642e1bda825461c7a971a2 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Wed, 5 Jan 2011 12:04:03 +0100 Subject: [PATCH 216/351] multiudpsink: add buffer-size property Add buffer-size property to configure the kernel send buffer. --- gst/udp/gstmultiudpsink.c | 53 +++++++++++++++++++++++++++++++++++---- gst/udp/gstmultiudpsink.h | 1 + 2 files changed, 49 insertions(+), 5 deletions(-) diff --git a/gst/udp/gstmultiudpsink.c b/gst/udp/gstmultiudpsink.c index 8a24275225..78e491b2b6 100644 --- a/gst/udp/gstmultiudpsink.c +++ b/gst/udp/gstmultiudpsink.c @@ -80,6 +80,7 @@ enum #define DEFAULT_LOOP TRUE #define DEFAULT_QOS_DSCP -1 #define DEFAULT_SEND_DUPLICATES TRUE +#define DEFAULT_BUFFER_SIZE 0 enum { @@ -96,6 +97,7 @@ enum PROP_LOOP, PROP_QOS_DSCP, PROP_SEND_DUPLICATES, + PROP_BUFFER_SIZE, PROP_LAST }; @@ -347,6 +349,11 @@ gst_multiudpsink_class_init (GstMultiUDPSinkClass * klass) "multiple times as well", DEFAULT_SEND_DUPLICATES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BUFFER_SIZE, + g_param_spec_int ("buffer-size", "Buffer Size", + "Size of the kernel send buffer in bytes, 0=default", 0, G_MAXINT, + DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + gstelement_class->change_state = gst_multiudpsink_change_state; gstbasesink_class->render = gst_multiudpsink_render; @@ -773,6 +780,9 @@ gst_multiudpsink_set_property (GObject * object, guint prop_id, case PROP_SEND_DUPLICATES: udpsink->send_duplicates = g_value_get_boolean (value); break; + case PROP_BUFFER_SIZE: + udpsink->buffer_size = g_value_get_int (value); + break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; @@ -825,6 +835,9 @@ gst_multiudpsink_get_property (GObject * object, guint prop_id, GValue * value, case PROP_SEND_DUPLICATES: g_value_set_boolean (value, udpsink->send_duplicates); break; + case PROP_BUFFER_SIZE: + g_value_set_int (value, udpsink->buffer_size); + break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; @@ -899,6 +912,12 @@ gst_multiudpsink_init_send (GstMultiUDPSink * sink) guint bc_val; GList *clients; GstUDPClient *client; + int sndsize, ret; +#ifdef G_OS_WIN32 + gint len; +#else + guint len; +#endif if (sink->sockfd == -1) { GST_DEBUG_OBJECT (sink, "creating sockets"); @@ -914,11 +933,6 @@ gst_multiudpsink_init_send (GstMultiUDPSink * sink) sink->externalfd = FALSE; } else { struct sockaddr_storage myaddr; -#ifdef G_OS_WIN32 - gint len; -#else - guint len; -#endif GST_DEBUG_OBJECT (sink, "using configured socket"); /* we use the configured socket, try to get some info about it */ @@ -932,6 +946,35 @@ gst_multiudpsink_init_send (GstMultiUDPSink * sink) sink->externalfd = TRUE; } + len = sizeof (sndsize); + if (sink->buffer_size != 0) { + sndsize = sink->buffer_size; + + GST_DEBUG_OBJECT (sink, "setting udp buffer of %d bytes", sndsize); + /* set buffer size, Note that on Linux this is typically limited to a + * maximum of around 100K. Also a minimum of 128 bytes is required on + * Linux. */ + ret = + setsockopt (sink->sockfd, SOL_SOCKET, SO_SNDBUF, (void *) &sndsize, + len); + if (ret != 0) { + GST_ELEMENT_WARNING (sink, RESOURCE, SETTINGS, (NULL), + ("Could not create a buffer of requested %d bytes, %d: %s (%d)", + sndsize, ret, g_strerror (errno), errno)); + } + } + + /* read the value of the receive buffer. Note that on linux this returns 2x the + * value we set because the kernel allocates extra memory for metadata. + * The default on Linux is about 100K (which is about 50K without metadata) */ + ret = + getsockopt (sink->sockfd, SOL_SOCKET, SO_SNDBUF, (void *) &sndsize, &len); + if (ret == 0) + GST_DEBUG_OBJECT (sink, "have udp buffer of %d bytes", sndsize); + else + GST_DEBUG_OBJECT (sink, "could not get udp buffer size"); + + bc_val = 1; if (setsockopt (sink->sock, SOL_SOCKET, SO_BROADCAST, &bc_val, sizeof (bc_val)) < 0) diff --git a/gst/udp/gstmultiudpsink.h b/gst/udp/gstmultiudpsink.h index 6d4da77e95..c31dbad83a 100644 --- a/gst/udp/gstmultiudpsink.h +++ b/gst/udp/gstmultiudpsink.h @@ -80,6 +80,7 @@ struct _GstMultiUDPSink { guint16 ss_family; gboolean send_duplicates; + gint buffer_size; }; struct _GstMultiUDPSinkClass { From 6b91c5f6e765e8522199d6b33b23dfd3d69d06b1 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Wed, 5 Jan 2011 15:03:32 +0100 Subject: [PATCH 217/351] vrawdepay: fix length check Add some more debugging. Add the length check so we don't cause unneeded warnings. --- gst/rtp/gstrtpvrawdepay.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/gst/rtp/gstrtpvrawdepay.c b/gst/rtp/gstrtpvrawdepay.c index 8e1af75e72..5919a5bcb0 100644 --- a/gst/rtp/gstrtpvrawdepay.c +++ b/gst/rtp/gstrtpvrawdepay.c @@ -242,6 +242,7 @@ gst_rtp_vraw_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps) GST_DEBUG_OBJECT (depayload, "width %d, height %d, format %d", width, height, format); GST_DEBUG_OBJECT (depayload, "yp %d, up %d, vp %d", yp, up, vp); + GST_DEBUG_OBJECT (depayload, "xinc %d, yinc %d", xinc, yinc); GST_DEBUG_OBJECT (depayload, "pgroup %d, ystride %d, uvstride %d", pgroup, ystride, uvstride); GST_DEBUG_OBJECT (depayload, "outsize %u", outsize); @@ -386,10 +387,10 @@ gst_rtp_vraw_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf) } /* calculate the maximim amount of bytes we can use per line */ - if (offs + ((length / pgroup) * xinc) > (width - xinc)) { + if (offs + ((length / pgroup) * xinc) > width) { plen = ((width - offs) * pgroup) / xinc; - GST_WARNING_OBJECT (depayload, "clipping length %d, offset %d", length, - offs); + GST_WARNING_OBJECT (depayload, "clipping length %d, offset %d, plen %d", + length, offs, plen); } else plen = length; From ee2a615851ac361030d4af8c93d485e91da23c89 Mon Sep 17 00:00:00 2001 From: Edward Hervey Date: Wed, 5 Jan 2011 16:24:13 +0100 Subject: [PATCH 218/351] qtdemux: Fix unitialized variables on macosx --- gst/qtdemux/qtdemux.c | 20 ++++++++++---------- gst/qtdemux/qtdemux_dump.c | 8 ++++---- 2 files changed, 14 insertions(+), 14 deletions(-) diff --git a/gst/qtdemux/qtdemux.c b/gst/qtdemux/qtdemux.c index dac0db10e2..80fd5a0313 100644 --- a/gst/qtdemux/qtdemux.c +++ b/gst/qtdemux/qtdemux.c @@ -1965,8 +1965,8 @@ extract_initial_length_and_fourcc (const guint8 * data, guint size, static gboolean qtdemux_parse_mehd (GstQTDemux * qtdemux, GstByteReader * br) { - guint32 version; - guint64 duration; + guint32 version = 0; + guint64 duration = 0; if (!gst_byte_reader_get_uint32_be (br, &version)) goto failed; @@ -1976,7 +1976,7 @@ qtdemux_parse_mehd (GstQTDemux * qtdemux, GstByteReader * br) if (!gst_byte_reader_get_uint64_be (br, &duration)) goto failed; } else { - guint32 dur; + guint32 dur = 0; if (!gst_byte_reader_get_uint32_be (br, &dur)) goto failed; @@ -2008,7 +2008,7 @@ qtdemux_parse_trex (GstQTDemux * qtdemux, QtDemuxStream * stream, trex = qtdemux_tree_get_child_by_type_full (mvex, FOURCC_trex, &trex_data); while (trex) { - guint32 id, dur, size, flags; + guint32 id = 0, dur = 0, size = 0, flags = 0; /* skip version/flags */ if (!gst_byte_reader_skip (&trex_data, 4)) @@ -2066,7 +2066,7 @@ qtdemux_parse_trun (GstQTDemux * qtdemux, GstByteReader * trun, { guint64 timestamp; gint32 data_offset = 0; - guint32 flags, first_flags = 0, samples_count; + guint32 flags = 0, first_flags = 0, samples_count = 0; gint i; guint8 *data; guint entry_size, dur_offset, size_offset, flags_offset = 0, ct_offset = 0; @@ -2290,8 +2290,8 @@ qtdemux_parse_tfhd (GstQTDemux * qtdemux, GstByteReader * tfhd, guint32 * default_sample_size, guint32 * default_sample_flags, gint64 * base_offset) { - guint32 flags; - guint32 track_id; + guint32 flags = 0; + guint32 track_id = 0; if (!gst_byte_reader_skip (tfhd, 1) || !gst_byte_reader_get_uint24_be (tfhd, &flags)) @@ -5201,7 +5201,7 @@ qtdemux_stbl_init (GstQTDemux * qtdemux, QtDemuxStream * stream, GNode * stbl) /* sync sample atom */ stream->stps_present = FALSE; if ((stream->stss_present = - ! !qtdemux_tree_get_child_by_type_full (stbl, FOURCC_stss, + !!qtdemux_tree_get_child_by_type_full (stbl, FOURCC_stss, &stream->stss) ? TRUE : FALSE) == TRUE) { /* copy atom data into a new buffer for later use */ stream->stss.data = g_memdup (stream->stss.data, stream->stss.size); @@ -5219,7 +5219,7 @@ qtdemux_stbl_init (GstQTDemux * qtdemux, QtDemuxStream * stream, GNode * stbl) /* partial sync sample atom */ if ((stream->stps_present = - ! !qtdemux_tree_get_child_by_type_full (stbl, FOURCC_stps, + !!qtdemux_tree_get_child_by_type_full (stbl, FOURCC_stps, &stream->stps) ? TRUE : FALSE) == TRUE) { /* copy atom data into a new buffer for later use */ stream->stps.data = g_memdup (stream->stps.data, stream->stps.size); @@ -5339,7 +5339,7 @@ qtdemux_stbl_init (GstQTDemux * qtdemux, QtDemuxStream * stream, GNode * stbl) /* composition time-to-sample */ if ((stream->ctts_present = - ! !qtdemux_tree_get_child_by_type_full (stbl, FOURCC_ctts, + !!qtdemux_tree_get_child_by_type_full (stbl, FOURCC_ctts, &stream->ctts) ? TRUE : FALSE) == TRUE) { /* copy atom data into a new buffer for later use */ stream->ctts.data = g_memdup (stream->ctts.data, stream->ctts.size); diff --git a/gst/qtdemux/qtdemux_dump.c b/gst/qtdemux/qtdemux_dump.c index b384fcd4dc..d7fa62ab7d 100644 --- a/gst/qtdemux/qtdemux_dump.c +++ b/gst/qtdemux/qtdemux_dump.c @@ -523,7 +523,7 @@ gboolean qtdemux_dump_tfra (GstQTDemux * qtdemux, GstByteReader * data, int depth) { guint64 time = 0, moof_offset = 0; - guint32 len = 0, num_entries = 0, ver_flags, track_id, i; + guint32 len = 0, num_entries = 0, ver_flags = 0, track_id = 0, i; guint value_size, traf_size, trun_size, sample_size; if (!gst_byte_reader_get_uint32_be (data, &ver_flags)) @@ -569,8 +569,8 @@ qtdemux_dump_tfra (GstQTDemux * qtdemux, GstByteReader * data, int depth) gboolean qtdemux_dump_tfhd (GstQTDemux * qtdemux, GstByteReader * data, int depth) { - guint32 flags, n, track_id; - guint64 base_data_offset; + guint32 flags = 0, n = 0, track_id = 0; + guint64 base_data_offset = 0; if (!gst_byte_reader_skip (data, 1) || !gst_byte_reader_get_uint24_be (data, &flags)) @@ -700,7 +700,7 @@ qtdemux_dump_trex (GstQTDemux * qtdemux, GstByteReader * data, int depth) gboolean qtdemux_dump_mehd (GstQTDemux * qtdemux, GstByteReader * data, int depth) { - guint32 version; + guint32 version = 0; guint64 fragment_duration; guint value_size; From bc0824181bc24ef34e221602400c8b1a80d559b2 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Wed, 5 Jan 2011 18:32:58 +0100 Subject: [PATCH 219/351] rtspsrc: don't confuse return values Return a return value of the right type. --- gst/rtsp/gstrtspsrc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c index 92c4efc560..a507d4e747 100644 --- a/gst/rtsp/gstrtspsrc.c +++ b/gst/rtsp/gstrtspsrc.c @@ -4397,7 +4397,7 @@ server_eof: GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL), ("The server closed the connection.")); gst_rtsp_message_unset (response); - return GST_FLOW_UNEXPECTED; + return res; } } From 97d74f293c7810d828426461c95e5821debc5dbb Mon Sep 17 00:00:00 2001 From: David Schleef Date: Sat, 11 Dec 2010 12:42:10 -0800 Subject: [PATCH 220/351] deinterlace: Change the default to linear The previous default, greedyh, takes 4 times as long as MPEG-2 video decoding, and is unlikely fast enough on any current CPU to play 1080i video in real-time. greedyl isn't much faster. linear was chosen over vfir, since the quality advantage of vfir is minimal compared to the occasional visual artifacts and slower processing. --- gst/deinterlace/gstdeinterlace.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/gst/deinterlace/gstdeinterlace.c b/gst/deinterlace/gstdeinterlace.c index ed752b20d0..1ad960e87d 100644 --- a/gst/deinterlace/gstdeinterlace.c +++ b/gst/deinterlace/gstdeinterlace.c @@ -52,7 +52,7 @@ GST_DEBUG_CATEGORY_STATIC (deinterlace_debug); /* Properties */ #define DEFAULT_MODE GST_DEINTERLACE_MODE_AUTO -#define DEFAULT_METHOD GST_DEINTERLACE_GREEDY_H +#define DEFAULT_METHOD GST_DEINTERLACE_LINEAR #define DEFAULT_FIELDS GST_DEINTERLACE_ALL #define DEFAULT_FIELD_LAYOUT GST_DEINTERLACE_LAYOUT_AUTO From 088ae75681bf9e803fc7b7a10f2d2a6c23379dd5 Mon Sep 17 00:00:00 2001 From: David Schleef Date: Wed, 5 Jan 2011 17:49:16 -0800 Subject: [PATCH 221/351] capsdebug: Add capdebug debug category --- gst/debugutils/gstcapsdebug.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/gst/debugutils/gstcapsdebug.c b/gst/debugutils/gstcapsdebug.c index 469408c56c..3806bae6a8 100644 --- a/gst/debugutils/gstcapsdebug.c +++ b/gst/debugutils/gstcapsdebug.c @@ -26,6 +26,9 @@ #include #include "gstcapsdebug.h" +GST_DEBUG_CATEGORY_STATIC (gst_caps_debug_debug); +#define GST_CAT_DEFAULT gst_caps_debug_debug + /* prototypes */ @@ -67,7 +70,12 @@ GST_STATIC_PAD_TEMPLATE ("src", /* class initialization */ -GST_BOILERPLATE (GstCapsDebug, gst_caps_debug, GstElement, GST_TYPE_ELEMENT); +#define DEBUG_INIT(bla) \ + GST_DEBUG_CATEGORY_INIT (gst_caps_debug_debug, "capsdebug", 0, \ + "debug category for capsdebug element"); + +GST_BOILERPLATE_FULL (GstCapsDebug, gst_caps_debug, GstElement, + GST_TYPE_ELEMENT, DEBUG_INIT); static void gst_caps_debug_base_init (gpointer g_class) From 74360c22be94be528e8b016813a3eb54b82c2bdf Mon Sep 17 00:00:00 2001 From: Edward Hervey Date: Thu, 6 Jan 2011 12:28:58 +0100 Subject: [PATCH 222/351] qtdemux: Fix unitialized variables on macosx --- gst/qtdemux/qtdemux_dump.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/gst/qtdemux/qtdemux_dump.c b/gst/qtdemux/qtdemux_dump.c index d7fa62ab7d..fa66767583 100644 --- a/gst/qtdemux/qtdemux_dump.c +++ b/gst/qtdemux/qtdemux_dump.c @@ -621,8 +621,9 @@ qtdemux_dump_tfhd (GstQTDemux * qtdemux, GstByteReader * data, int depth) gboolean qtdemux_dump_trun (GstQTDemux * qtdemux, GstByteReader * data, int depth) { - guint32 flags, samples_count, data_offset, first_sample_flags; - guint32 sample_duration, sample_size, sample_flags, composition_time_offsets; + guint32 flags = 0, samples_count = 0, data_offset = 0, first_sample_flags = 0; + guint32 sample_duration = 0, sample_size = 0, sample_flags = + 0, composition_time_offsets = 0; int i = 0; if (!gst_byte_reader_skip (data, 1) || From 4decc3aaea1cf927fa22529960cfe89aba6644cd Mon Sep 17 00:00:00 2001 From: Edward Hervey Date: Thu, 6 Jan 2011 12:29:21 +0100 Subject: [PATCH 223/351] rtp: Fix unitialized variables on macosx --- gst/rtp/gstrtpmp4adepay.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/gst/rtp/gstrtpmp4adepay.c b/gst/rtp/gstrtpmp4adepay.c index 394fc0d288..18ebfd2190 100644 --- a/gst/rtp/gstrtpmp4adepay.c +++ b/gst/rtp/gstrtpmp4adepay.c @@ -170,8 +170,8 @@ gst_rtp_mp4a_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps) guint8 *data; guint size; gint i; - guint32 rate; - guint8 obj_type, sr_idx, channels; + guint32 rate = 0; + guint8 obj_type = 0, sr_idx = 0, channels = 0; GstBitReader br; buffer = gst_value_get_buffer (&v); From 8f501521c76e47b5029df253072c706550b25ec3 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Fri, 7 Jan 2011 01:11:02 +0000 Subject: [PATCH 224/351] tests: never disable g_assert() and cast checks for the unit tests The unit tests are riddled with g_assert() and friends, make sure we don't disable assert and cast checks for the unit tests even if this has been specified for the rest of the code base, e.g. via --disable-glib-asserts. --- tests/check/Makefile.am | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/tests/check/Makefile.am b/tests/check/Makefile.am index 5079bc7274..ccce33cf03 100644 --- a/tests/check/Makefile.am +++ b/tests/check/Makefile.am @@ -164,7 +164,8 @@ noinst_PROGRAMS = \ elements/autodetect AM_CFLAGS = $(GST_OBJ_CFLAGS) $(GST_CHECK_CFLAGS) $(CHECK_CFLAGS) \ - $(GST_OPTION_CFLAGS) -DGST_TEST_FILES_PATH="\"$(TEST_FILES_DIRECTORY)\"" + $(GST_OPTION_CFLAGS) -DGST_TEST_FILES_PATH="\"$(TEST_FILES_DIRECTORY)\"" \ + -UG_DISABLE_ASSERT -UG_DISABLE_CAST_CHECKS LDADD = $(GST_OBJ_LIBS) $(GST_CHECK_LIBS) $(CHECK_LIBS) # valgrind testing @@ -210,7 +211,7 @@ elements_level_LDADD = $(LDADD) $(LIBM) elements_matroskamux_LDADD = $(GST_BASE_LIBS) $(LDADD) $(LIBM) elements_rtpbin_buffer_list_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) \ - $(WARNING_CFLAGS) $(ERROR_CFLAGS) $(GST_CHECK_CFLAGS) + $(WARNING_CFLAGS) $(ERROR_CFLAGS) $(GST_CHECK_CFLAGS) $(AM_CFLAGS) elements_rtpbin_buffer_list_LDADD = $(GST_PLUGINS_BASE_LIBS) \ -lgstnetbuffer-@GST_MAJORMINOR@ -lgstrtp-@GST_MAJORMINOR@ \ $(GST_BASE_LIBS) $(GST_LIBS_LIBS) $(GST_CHECK_LIBS) From 09d19e7dcd4d9903422c5118be114a342a0a25a2 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Fri, 7 Jan 2011 01:35:45 +0000 Subject: [PATCH 225/351] v4l2sink: don't put functional code like ioctl calls into g_return_if_fail() These macros will expand to NOOPs given the right defines. Also, g_return_if_fail() and friends are meant to be used to catch programming errors (like invalid input to functions), not runtime error handling. --- sys/v4l2/gstv4l2sink.c | 20 ++++++++++++++++---- 1 file changed, 16 insertions(+), 4 deletions(-) diff --git a/sys/v4l2/gstv4l2sink.c b/sys/v4l2/gstv4l2sink.c index d938ba3e34..f3eba3b90d 100644 --- a/sys/v4l2/gstv4l2sink.c +++ b/sys/v4l2/gstv4l2sink.c @@ -400,7 +400,10 @@ gst_v4l2sink_sync_overlay_fields (GstV4l2Sink * v4l2sink) memset (&format, 0x00, sizeof (struct v4l2_format)); format.type = V4L2_BUF_TYPE_VIDEO_OVERLAY; - g_return_if_fail (v4l2_ioctl (fd, VIDIOC_G_FMT, &format) >= 0); + if (v4l2_ioctl (fd, VIDIOC_G_FMT, &format) < 0) { + GST_WARNING_OBJECT (v4l2sink, "VIDIOC_G_FMT failed"); + return; + } GST_DEBUG_OBJECT (v4l2sink, "setting overlay: overlay_fields_set=0x%02x, top=%d, left=%d, width=%d, height=%d", @@ -417,7 +420,10 @@ gst_v4l2sink_sync_overlay_fields (GstV4l2Sink * v4l2sink) if (v4l2sink->overlay_fields_set & RECT_HEIGHT_SET) format.fmt.win.w.height = v4l2sink->overlay.height; - g_return_if_fail (v4l2_ioctl (fd, VIDIOC_S_FMT, &format) >= 0); + if (v4l2_ioctl (fd, VIDIOC_S_FMT, &format) < 0) { + GST_WARNING_OBJECT (v4l2sink, "VIDIOC_S_FMT failed"); + return; + } v4l2sink->overlay_fields_set = 0; v4l2sink->overlay = format.fmt.win.w; @@ -438,7 +444,10 @@ gst_v4l2sink_sync_crop_fields (GstV4l2Sink * v4l2sink) memset (&crop, 0x00, sizeof (struct v4l2_crop)); crop.type = V4L2_BUF_TYPE_VIDEO_OUTPUT; - g_return_if_fail (v4l2_ioctl (fd, VIDIOC_G_CROP, &crop) >= 0); + if (v4l2_ioctl (fd, VIDIOC_G_CROP, &crop) < 0) { + GST_WARNING_OBJECT (v4l2sink, "VIDIOC_G_CROP failed"); + return; + } GST_DEBUG_OBJECT (v4l2sink, "setting crop: crop_fields_set=0x%02x, top=%d, left=%d, width=%d, height=%d", @@ -455,7 +464,10 @@ gst_v4l2sink_sync_crop_fields (GstV4l2Sink * v4l2sink) if (v4l2sink->crop_fields_set & RECT_HEIGHT_SET) crop.c.height = v4l2sink->crop.height; - g_return_if_fail (v4l2_ioctl (fd, VIDIOC_S_CROP, &crop) >= 0); + if (v4l2_ioctl (fd, VIDIOC_S_CROP, &crop) < 0) { + GST_WARNING_OBJECT (v4l2sink, "VIDIOC_S_CROP failed"); + return; + } v4l2sink->crop_fields_set = 0; v4l2sink->crop = crop.c; From 68fa8f341750bc6b84ff72816d9ea8af8b6ddc42 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Fri, 7 Jan 2011 02:06:51 +0000 Subject: [PATCH 226/351] matroska: don't put essential function calls into g_assert() g_assert() will expand to NOOPs if -DG_DISABLE_ASSERT is passed. --- gst/matroska/ebml-read.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/gst/matroska/ebml-read.c b/gst/matroska/ebml-read.c index 379c882d08..24780e7775 100644 --- a/gst/matroska/ebml-read.c +++ b/gst/matroska/ebml-read.c @@ -262,7 +262,8 @@ gst_ebml_read_master (GstEbmlRead * ebml, guint32 * id) return ret; /* we just at least peeked the id */ - g_assert (gst_byte_reader_skip (gst_ebml_read_br (ebml), prefix)); + if (!gst_byte_reader_skip (gst_ebml_read_br (ebml), prefix)) + return GST_FLOW_ERROR; /* FIXME: do proper error handling */ m.offset = gst_ebml_read_get_pos (ebml); if (!gst_byte_reader_get_data (gst_ebml_read_br (ebml), length, &data)) @@ -329,7 +330,8 @@ gst_ebml_read_buffer (GstEbmlRead * ebml, guint32 * id, GstBuffer ** buf) return ret; /* we just at least peeked the id */ - g_assert (gst_byte_reader_skip (gst_ebml_read_br (ebml), prefix)); + if (!gst_byte_reader_skip (gst_ebml_read_br (ebml), prefix)) + return GST_FLOW_ERROR; /* FIXME: do proper error handling */ if (G_LIKELY (length > 0)) { guint offset; @@ -367,7 +369,8 @@ gst_ebml_read_bytes (GstEbmlRead * ebml, guint32 * id, const guint8 ** data, return ret; /* we just at least peeked the id */ - g_assert (gst_byte_reader_skip (gst_ebml_read_br (ebml), prefix)); + if (!gst_byte_reader_skip (gst_ebml_read_br (ebml), prefix)) + return GST_FLOW_ERROR; /* FIXME: do proper error handling */ *data = NULL; if (G_LIKELY (length >= 0)) { From 4a7a78dc9e8049756960c21d296596616046f512 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Fri, 7 Jan 2011 02:32:20 +0000 Subject: [PATCH 227/351] alpha: fix compiler warnings caused by -DG_DISABLE_ASSERT --- gst/alpha/gstalpha.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/gst/alpha/gstalpha.c b/gst/alpha/gstalpha.c index fd9b851d68..d72c8c5b1e 100644 --- a/gst/alpha/gstalpha.c +++ b/gst/alpha/gstalpha.c @@ -397,7 +397,7 @@ gst_alpha_set_property (GObject * object, guint prop_id, case PROP_PREFER_PASSTHROUGH:{ gboolean prefer_passthrough = g_value_get_boolean (value); - reconfigure = ((!!prefer_passthrough) != (!!alpha->prefer_passthrough)) + reconfigure = ((! !prefer_passthrough) != (! !alpha->prefer_passthrough)) && (alpha->method == ALPHA_METHOD_SET) && (alpha->alpha == 1.0); alpha->prefer_passthrough = prefer_passthrough; break; @@ -1425,7 +1425,7 @@ gst_alpha_set_planar_yuv_ayuv (const guint8 * src, guint8 * dest, gint width, break; default: g_assert_not_reached (); - break; + return; } if (alpha->in_sdtv == alpha->out_sdtv) { @@ -1544,7 +1544,7 @@ gst_alpha_chroma_key_planar_yuv_ayuv (const guint8 * src, guint8 * dest, break; default: g_assert_not_reached (); - break; + return; } if (alpha->in_sdtv == alpha->out_sdtv) { @@ -1685,7 +1685,7 @@ gst_alpha_set_planar_yuv_argb (const guint8 * src, guint8 * dest, gint width, break; default: g_assert_not_reached (); - break; + return; } memcpy (matrix, @@ -1793,7 +1793,7 @@ gst_alpha_chroma_key_planar_yuv_argb (const guint8 * src, guint8 * dest, break; default: g_assert_not_reached (); - break; + return; } memcpy (matrix, From fcbfd8909bd08c874c711277f887ba9638cd919d Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Fri, 7 Jan 2011 13:06:38 +0000 Subject: [PATCH 228/351] po: update translations --- po/af.po | 18 +++++++++++++++--- po/az.po | 18 +++++++++++++++--- po/bg.po | 21 +++++++++++++++++---- po/ca.po | 18 +++++++++++++++--- po/cs.po | 22 ++++++++++++++++++---- po/da.po | 21 +++++++++++++++++---- po/de.po | 23 +++++++++++++++++++---- po/el.po | 22 ++++++++++++++++++---- po/en_GB.po | 18 +++++++++++++++--- po/es.po | 22 ++++++++++++++++++---- po/eu.po | 22 ++++++++++++++++++---- po/fi.po | 22 ++++++++++++++++++---- po/fr.po | 22 ++++++++++++++++++---- po/gl.po | 22 ++++++++++++++++++---- po/hu.po | 22 ++++++++++++++++++---- po/id.po | 21 +++++++++++++++++---- po/it.po | 22 ++++++++++++++++++---- po/ja.po | 22 ++++++++++++++++++---- po/lt.po | 22 ++++++++++++++++++---- po/lv.po | 22 ++++++++++++++++++---- po/mt.po | 22 ++++++++++++++++++---- po/nb.po | 18 +++++++++++++++--- po/nl.po | 22 ++++++++++++++++++---- po/or.po | 18 +++++++++++++++--- po/pl.po | 21 +++++++++++++++++---- po/pt_BR.po | 22 ++++++++++++++++++---- po/ro.po | 22 ++++++++++++++++++---- po/ru.po | 22 ++++++++++++++++++---- po/sk.po | 21 +++++++++++++++++---- po/sl.po | 22 ++++++++++++++++++---- po/sq.po | 18 +++++++++++++++--- po/sr.po | 18 +++++++++++++++--- po/sv.po | 26 +++++++++++++++++++------- po/tr.po | 21 +++++++++++++++++---- po/uk.po | 22 ++++++++++++++++++---- po/vi.po | 21 +++++++++++++++++---- po/zh_CN.po | 20 ++++++++++++++++---- po/zh_HK.po | 18 +++++++++++++++--- po/zh_TW.po | 18 +++++++++++++++--- 39 files changed, 665 insertions(+), 149 deletions(-) diff --git a/po/af.po b/po/af.po index b1df978f63..138c08481f 100644 --- a/po/af.po +++ b/po/af.po @@ -7,7 +7,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins 0.7.6\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-09-12 00:08+0100\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2004-03-18 14:16+0200\n" "Last-Translator: Petri Jooste \n" "Language-Team: Afrikaans \n" @@ -70,10 +70,13 @@ msgstr "" msgid "This file is invalid and cannot be played." msgstr "" -msgid "This file is incomplete and cannot be played." +msgid "This file is corrupt and cannot be played." msgstr "" -msgid "This file is corrupt and cannot be played." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." msgstr "" msgid "The video in this file might not play correctly." @@ -626,6 +629,15 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Kon nie oudio-toestel \"%s\" toemaak nie." +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "Kon nie genoeg buffers vanaf toestel \"%s\" kry nie." + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "Kon nie oudio-toestel \"%s\" toemaak nie." + #, fuzzy, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "Kon nie buffers vanaf toestel \"%s\" verkry nie." diff --git a/po/az.po b/po/az.po index d51b991e8a..f51c1cb74c 100644 --- a/po/az.po +++ b/po/az.po @@ -7,7 +7,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-0.8.0\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-09-12 00:08+0100\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2004-03-19 18:29+0200\n" "Last-Translator: Metin Amiroff \n" "Language-Team: Azerbaijani \n" @@ -71,10 +71,13 @@ msgstr "" msgid "This file is invalid and cannot be played." msgstr "" -msgid "This file is incomplete and cannot be played." +msgid "This file is corrupt and cannot be played." msgstr "" -msgid "This file is corrupt and cannot be played." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." msgstr "" msgid "The video in this file might not play correctly." @@ -627,6 +630,15 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "\"%s\" audio avadanlığı bağlana bilmədi." +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "\"%s\" avadanlığından kifayət qədər bufferlər alına bilmədi." + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "\"%s\" audio avadanlığı bağlana bilmədi." + #, fuzzy, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "\"%s\" avadanlığından bufferlər alına bilmədi." diff --git a/po/bg.po b/po/bg.po index 32c4cacacf..da3786aa97 100644 --- a/po/bg.po +++ b/po/bg.po @@ -8,7 +8,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-good 0.10.25.3\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-11-18 00:43+0000\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2010-11-04 14:23+0200\n" "Last-Translator: Alexander Shopov \n" "Language-Team: Bulgarian \n" @@ -69,12 +69,15 @@ msgstr "Този файл не съдържа изпълними потоци." msgid "This file is invalid and cannot be played." msgstr "Този файл е повреден и не може да бъде изпълнен." -msgid "This file is incomplete and cannot be played." -msgstr "Този файл е непълен и не може да бъде изпълнен." - msgid "This file is corrupt and cannot be played." msgstr "Този файл е повреден и не може да бъде изпълнен." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." +msgstr "Този файл е непълен и не може да бъде изпълнен." + msgid "The video in this file might not play correctly." msgstr "Видео потокът в този файл може да не се покаже правилно." @@ -627,6 +630,16 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Неуспех при задаването на вход %d на устройство „%s“." +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "" +"Неуспех при получаването на текущия вход на устройство „%s“. Може би е радио." + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "Неуспех при задаването на вход %d на устройство „%s“." + #, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "Буферите в устройство „%s“ не могат да се подредят в опашка." diff --git a/po/ca.po b/po/ca.po index df0a28115e..8ceac115f9 100644 --- a/po/ca.po +++ b/po/ca.po @@ -7,7 +7,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-good 0.9.7\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-09-12 00:08+0100\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2005-12-04 21:54+0100\n" "Last-Translator: Jordi Mallach \n" "Language-Team: Catalan \n" @@ -67,10 +67,13 @@ msgstr "" msgid "This file is invalid and cannot be played." msgstr "" -msgid "This file is incomplete and cannot be played." +msgid "This file is corrupt and cannot be played." msgstr "" -msgid "This file is corrupt and cannot be played." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." msgstr "" msgid "The video in this file might not play correctly." @@ -619,6 +622,15 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "" +#, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "" + +#, c-format +msgid "Failed to set output %d on device %s." +msgstr "" + #, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "" diff --git a/po/cs.po b/po/cs.po index 4158ec74fc..d91a5ee220 100644 --- a/po/cs.po +++ b/po/cs.po @@ -8,7 +8,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-good-0.10.25.3\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-11-01 17:27+0000\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2010-10-31 10:21+0100\n" "Last-Translator: Petr Kovar \n" "Language-Team: Czech \n" @@ -70,12 +70,15 @@ msgstr "Tento soubor neobsahuje hratelné proudy." msgid "This file is invalid and cannot be played." msgstr "Tento soubor je neplatný a nelze jej přehrát." -msgid "This file is incomplete and cannot be played." -msgstr "Tento soubor není úplný a nelze jej přehrát." - msgid "This file is corrupt and cannot be played." msgstr "Tento soubor je poškozen a nelze jej přehrát." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." +msgstr "Tento soubor není úplný a nelze jej přehrát." + msgid "The video in this file might not play correctly." msgstr "Video v tomto souboru se nemusí přehrát správně." @@ -629,6 +632,17 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Nezdařilo se nastavení vstupu \"%d\" na zařízení \"%s\"." +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "" +"Nezdařilo se získání aktuálního vstupu na zařízení \"%s\". Možná se jedná o " +"radiopřijímač." + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "Nezdařilo se nastavení vstupu \"%d\" na zařízení \"%s\"." + #, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "Nezdařilo se zařazení vyrovnávací paměti na zařízení \"%s\"." diff --git a/po/da.po b/po/da.po index 2d5704e21f..0959902e26 100644 --- a/po/da.po +++ b/po/da.po @@ -12,7 +12,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-good-0.10.25.3\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-11-01 17:27+0000\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2010-10-26 23:54+0200\n" "Last-Translator: Joe Hansen \n" "Language-Team: Danish \n" @@ -73,12 +73,15 @@ msgstr "Denne fil indeholder ingen spilbar strøm." msgid "This file is invalid and cannot be played." msgstr "Denne fil er ufuldstændig og kan ikke afspilles." -msgid "This file is incomplete and cannot be played." -msgstr "Denne fil er ufuldstændig og kan ikke afspilles." - msgid "This file is corrupt and cannot be played." msgstr "Denne fil er ødelagt og kan ikke afspilles." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." +msgstr "Denne fil er ufuldstændig og kan ikke afspilles." + msgid "The video in this file might not play correctly." msgstr "Videoen i denne fil afspilles måske ikke korrekt." @@ -634,6 +637,16 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Kunne ikke sætte inddata %d for enhed %s." +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "" +"Kunne ikke hente nuværende inddata for enhed '%s'. Måske er det en radioenhed" + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "Kunne ikke sætte inddata %d for enhed %s." + #, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "Kunne ikke sætte mellemlager fra enhed '%s' i kø." diff --git a/po/de.po b/po/de.po index 13a0ad6344..12e2c13286 100644 --- a/po/de.po +++ b/po/de.po @@ -14,7 +14,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-good 0.10.25.3\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-11-01 17:27+0000\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2010-10-25 00:33+0200\n" "Last-Translator: Christian Kirbach \n" "Language-Team: German \n" @@ -79,12 +79,15 @@ msgstr "Diese Datei enthält keine abspielbaren Ströme." msgid "This file is invalid and cannot be played." msgstr "Diese Datei ist ungültig und kann nicht wiedergegeben werden." -msgid "This file is incomplete and cannot be played." -msgstr "Diese Datei ist unvollständig und kann nicht wiedergegeben werden." - msgid "This file is corrupt and cannot be played." msgstr "Diese Datei ist beschädigt und kann nicht wiedergegeben werden." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." +msgstr "Diese Datei ist unvollständig und kann nicht wiedergegeben werden." + msgid "The video in this file might not play correctly." msgstr "Das Video in dieser Datei wird vielleicht nicht korrekt wiedergegeben." @@ -643,6 +646,18 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Festlegen der Eingabe »%d« des Geräts »%s« schlug fehl." +# Sendegerät sicherlich nicht, eben eher ein Radio. Warum diese Fehlermeldung beie einem Radiogerät kommen könnte, weiß ich allerdings auch nicht. +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "" +"Auslesen der aktuellen Eingabe auf dem Gerät »%s« schlug fehl. Vielleicht " +"ist es ein Radiogerät." + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "Festlegen der Eingabe »%d« des Geräts »%s« schlug fehl." + #, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "Auf Gerät »%s« konnten keine Puffer eingereiht werden." diff --git a/po/el.po b/po/el.po index e14b1e8944..1423f2d221 100644 --- a/po/el.po +++ b/po/el.po @@ -7,7 +7,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-good 0.10.25.3\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-11-01 17:27+0000\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2010-10-27 12:16+0200\n" "Last-Translator: Michael Kotsarinis \n" "Language-Team: Greek \n" @@ -70,12 +70,15 @@ msgstr "Αυτό το αρχείο δεν περιέχει αναπαραγώγ msgid "This file is invalid and cannot be played." msgstr "Το αρχείο αυτό δεν είναι έγκυρο και δεν μπορεί να αναπαραχθεί." -msgid "This file is incomplete and cannot be played." -msgstr "Το αρχείο αυτό είναι ανολοκλήρωτο και δεν μπορεί να αναπαραχθεί." - msgid "This file is corrupt and cannot be played." msgstr "Το αρχείο αυτό είναι κατεστραμμένο και δεν μπορεί να αναπαραχθεί. " +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." +msgstr "Το αρχείο αυτό είναι ανολοκλήρωτο και δεν μπορεί να αναπαραχθεί." + msgid "The video in this file might not play correctly." msgstr "Το βίντεο σε αυτό το αρχείο μπορεί να μην αναπαραχθεί σωστά." @@ -631,6 +634,17 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Αποτυχία ρύθμισης εισαγωγής %d στην συσκευή %s." +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "" +"Αποτυχία ανάγνωσης της τρέχουσας εισαγωγής στην συσκευή '%s'. Πιθανόν να " +"είναι μια συσκευή ράδιο" + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "Αποτυχία ρύθμισης εισαγωγής %d στην συσκευή %s." + #, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "Αδυναμία θέσης σε σειρά των buffer στη συσκευή '%s'." diff --git a/po/en_GB.po b/po/en_GB.po index 156b7fc141..d3fe92c4cc 100644 --- a/po/en_GB.po +++ b/po/en_GB.po @@ -6,7 +6,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins 0.8.1\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-09-12 00:08+0100\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2004-04-26 10:41-0400\n" "Last-Translator: Gareth Owen \n" "Language-Team: English (British) \n" @@ -70,10 +70,13 @@ msgstr "" msgid "This file is invalid and cannot be played." msgstr "" -msgid "This file is incomplete and cannot be played." +msgid "This file is corrupt and cannot be played." msgstr "" -msgid "This file is corrupt and cannot be played." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." msgstr "" msgid "The video in this file might not play correctly." @@ -626,6 +629,15 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Could not close audio device \"%s\"." +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "Could not get enough buffers from device \"%s\"." + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "Could not close audio device \"%s\"." + #, fuzzy, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "Could not get buffers from device \"%s\"." diff --git a/po/es.po b/po/es.po index bf412870b7..bb8ac8a2a1 100644 --- a/po/es.po +++ b/po/es.po @@ -7,7 +7,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-good 0.10.25.3\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-11-01 17:27+0000\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2010-11-01 18:05+0100\n" "Last-Translator: Jorge González González \n" "Language-Team: Spanish \n" @@ -70,12 +70,15 @@ msgstr "Este archivo no contiene flujos reproducibles." msgid "This file is invalid and cannot be played." msgstr "Este archivo no es válido y no se puede reproducir." -msgid "This file is incomplete and cannot be played." -msgstr "Este archivo está incompleto y no se puede reproducir." - msgid "This file is corrupt and cannot be played." msgstr "Este archivo está corrupto y no se puede reproducir." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." +msgstr "Este archivo está incompleto y no se puede reproducir." + msgid "The video in this file might not play correctly." msgstr "" "Puede que el vídeo en este archivo no se pueda reproducir correctamente." @@ -636,6 +639,17 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Falló al establecer la entrada %d en el dispositivo %s." +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "" +"Falló al obtener la entrada actual en el dispositivo «%s». Quizá sea un " +"dispositivo de radio." + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "Falló al establecer la entrada %d en el dispositivo %s." + #, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "No se pueden encolar los búferes en el dispositivo «%s»." diff --git a/po/eu.po b/po/eu.po index 283db27ce5..02d9f8965b 100644 --- a/po/eu.po +++ b/po/eu.po @@ -8,7 +8,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-good-0.10.18.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-09-12 00:08+0100\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2010-03-25 12:37+0100\n" "Last-Translator: Mikel Olasagasti Uranga \n" "Language-Team: Basque \n" @@ -74,12 +74,15 @@ msgstr "Fitxategi horretan ez dago erreproduzi daitekeen korronterik." msgid "This file is invalid and cannot be played." msgstr "Fitxategi hau ez da baliozkoa eta ezin da erreproduzitu." -msgid "This file is incomplete and cannot be played." -msgstr "Fitxategi hau osatu gabea dago, eta ezin da erreproduzitu." - msgid "This file is corrupt and cannot be played." msgstr "Fitxategi hau hondatua dago, eta ezin da erreproduzitu." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." +msgstr "Fitxategi hau osatu gabea dago, eta ezin da erreproduzitu." + msgid "The video in this file might not play correctly." msgstr "" "Litekeena da fitxategi honetako bideoa ez behar bezala erreproduzitzea." @@ -634,6 +637,17 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Huts egin du '%2$s' gailuko %1$d. sarrera ezartzean." +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "" +"Huts egin du '%s' gailuko uneko sarrera eskuratzean. Litekeena da irrati-" +"gailu bat izatea." + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "Huts egin du '%2$s' gailuko %1$d. sarrera ezartzean." + #, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "Ezin izan dira bufferrak ilaran jarri '%s' gailuan." diff --git a/po/fi.po b/po/fi.po index 3e0104a5f3..a659e8d0eb 100644 --- a/po/fi.po +++ b/po/fi.po @@ -11,7 +11,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-good 0.10.25.3\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-11-18 00:43+0000\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2010-11-17 23:03+0200\n" "Last-Translator: Tommi Vainikainen \n" "Language-Team: Finnish \n" @@ -74,12 +74,15 @@ msgstr "Tiedosto ei sisällä soitettavia virtoja." msgid "This file is invalid and cannot be played." msgstr "Tiedosto on virheellinen eikä sitä voida esittää." -msgid "This file is incomplete and cannot be played." -msgstr "Tiedosto on vajaa eikä sitä voida esittää." - msgid "This file is corrupt and cannot be played." msgstr "Tiedosto on vioittunut eikä sitä voida näyttää." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." +msgstr "Tiedosto on vajaa eikä sitä voida esittää." + msgid "The video in this file might not play correctly." msgstr "Tiedostossa olevaa videota ei ehkä voida näyttää oikein." @@ -628,6 +631,17 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Sisääntuloa %d ei voitu asettaa laitteelle %s." +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "" +"Laitteen ”%s” tämänhetkistä sisääntuloa ei voitu lukea, se ei ehkä ole " +"radiolaite" + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "Sisääntuloa %d ei voitu asettaa laitteelle %s." + #, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "Puskureita ei voitu laittaa jonoon laitteella ”%s”." diff --git a/po/fr.po b/po/fr.po index 6e527dc448..a3220e74d3 100644 --- a/po/fr.po +++ b/po/fr.po @@ -9,7 +9,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-good 0.10.25.3\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-11-01 17:27+0000\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2010-10-27 09:48+0200\n" "Last-Translator: Claude Paroz \n" "Language-Team: French \n" @@ -72,12 +72,15 @@ msgstr "Ce fichier ne contient aucun flux exploitable." msgid "This file is invalid and cannot be played." msgstr "Ce fichier n'est pas valide et ne peut donc pas être lu." -msgid "This file is incomplete and cannot be played." -msgstr "Ce fichier n'est pas complet et ne peut donc pas être lu." - msgid "This file is corrupt and cannot be played." msgstr "Ce fichier est corrompu et ne peut pas être lu." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." +msgstr "Ce fichier n'est pas complet et ne peut donc pas être lu." + msgid "The video in this file might not play correctly." msgstr "" "Il est possible que la vidéo de ce fichier ne puisse pas être lue " @@ -637,6 +640,17 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Impossible de définir l'entrée %d du périphérique %s." +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "" +"Impossible d'obtenir l'entrée actuelle du périphérique « %s ». C'est peut-" +"être un périphérique radio" + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "Impossible de définir l'entrée %d du périphérique %s." + #, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "" diff --git a/po/gl.po b/po/gl.po index f1753f619e..01937dc019 100644 --- a/po/gl.po +++ b/po/gl.po @@ -7,7 +7,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-good 0.10.23.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-09-12 00:08+0100\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2010-08-12 15:46+0200\n" "Last-Translator: Fran Diéguez \n" "Language-Team: Galician \n" @@ -70,12 +70,15 @@ msgstr "Este ficheiro non contén ningún fluxo reproducíbel." msgid "This file is invalid and cannot be played." msgstr "Este ficheiro é incorrecto e non pode reproducirse." -msgid "This file is incomplete and cannot be played." -msgstr "Este ficheiro está incompleto e non pode reproducirse." - msgid "This file is corrupt and cannot be played." msgstr "Este ficheiro está danado e non pode reproducirse." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." +msgstr "Este ficheiro está incompleto e non pode reproducirse." + msgid "The video in this file might not play correctly." msgstr "Este vídeo neste ficheiro podería non reproducirse correctamente." @@ -663,6 +666,17 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Produciuse un fallo ao estabelecer a entrada %d no dispositivo %s." +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "" +"Produciuse un fallo ao obter a entrada actual no dispositivo «%s». Cicáis " +"sexa un dispositivo de radio." + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "Produciuse un fallo ao estabelecer a entrada %d no dispositivo %s." + #, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "Non é posíbel meter na cola os búferes no dispositivo «%s»." diff --git a/po/hu.po b/po/hu.po index 8c43eb4446..4b92d18916 100644 --- a/po/hu.po +++ b/po/hu.po @@ -8,7 +8,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-good 0.10.25.3\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-11-18 00:43+0000\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2010-11-04 01:21+0100\n" "Last-Translator: Gabor Kelemen \n" "Language-Team: Hungarian \n" @@ -73,12 +73,15 @@ msgstr "A fájl nem tartalmaz lejátszható adatfolyamokat." msgid "This file is invalid and cannot be played." msgstr "A fájl nem érvényes és nem játszható le." -msgid "This file is incomplete and cannot be played." -msgstr "A fájl nem teljes és nem játszható le." - msgid "This file is corrupt and cannot be played." msgstr "A fájl sérült és nem játszható le." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." +msgstr "A fájl nem teljes és nem játszható le." + msgid "The video in this file might not play correctly." msgstr "A fájlban található videó lehet, hogy nem játszható le megfelelően." @@ -631,6 +634,17 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "A(z) %d. bemenet beállítása meghiúsult a(z) „%s” eszközön." +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "" +"Az aktuális bemenet lekérése meghiúsult a(z) „%s” eszközről. Lehet, hogy ez " +"egy rádióeszköz." + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "A(z) %d. bemenet beállítása meghiúsult a(z) „%s” eszközön." + #, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "Nem állíthatók sorba a pufferek a(z) „%s” eszközben." diff --git a/po/id.po b/po/id.po index aa92786aac..ab9e8132d4 100644 --- a/po/id.po +++ b/po/id.po @@ -7,7 +7,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-good 0.10.23.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-09-12 00:08+0100\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2010-06-29 22:42+0700\n" "Last-Translator: Andhika Padmawan \n" "Language-Team: Indonesian \n" @@ -70,12 +70,15 @@ msgstr "Berkas ini tidak berisi arus yang dapat diputar." msgid "This file is invalid and cannot be played." msgstr "Berkas ini tidak sah dan tak dapat diputar." -msgid "This file is incomplete and cannot be played." -msgstr "Berkas ini tidak lengkap dan tak dapat diputar." - msgid "This file is corrupt and cannot be played." msgstr "Berkas ini rusak dan tak dapat diputar." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." +msgstr "Berkas ini tidak lengkap dan tak dapat diputar." + msgid "The video in this file might not play correctly." msgstr "Video di berkas ini mungkin tak dapat diputar dengan benar." @@ -623,6 +626,16 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Gagal mengatur masukan %d di divais %s." +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "" +"Gagal mendapatkan masukan terkini di divais '%s'. Mungkin itu divais radio" + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "Gagal mengatur masukan %d di divais %s." + #, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "Tak dapat mengantrekan penyangga di divais '%s'." diff --git a/po/it.po b/po/it.po index 798a735b48..63f9bf0c23 100644 --- a/po/it.po +++ b/po/it.po @@ -7,7 +7,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-good 0.10.25.3\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-11-01 17:27+0000\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2010-10-25 10:11+0200\n" "Last-Translator: Luca Ferretti \n" "Language-Team: Italian \n" @@ -70,12 +70,15 @@ msgstr "Questo file non contiene alcuno stream riproducibile." msgid "This file is invalid and cannot be played." msgstr "Questo file non è valido e non può essere riprodotto." -msgid "This file is incomplete and cannot be played." -msgstr "Questo file è incompleto e non può essere riprodotto." - msgid "This file is corrupt and cannot be played." msgstr "Questo file è alterato e non può essere riprodotto." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." +msgstr "Questo file è incompleto e non può essere riprodotto." + msgid "The video in this file might not play correctly." msgstr "Il video in questo file potrebbe non essere riprodotto correttamente." @@ -645,6 +648,17 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Impostazione dell'ingresso %d sul device «%s» non riuscita." +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "" +"Recupero dell'attuale ingresso sul device «%s» non riuscito. Forse è un " +"dispositivo radio" + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "Impostazione dell'ingresso %d sul device «%s» non riuscita." + #, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "Impossibile accodare i buffer nel device «%s»." diff --git a/po/ja.po b/po/ja.po index 2e2b19c696..45d44cf5ec 100644 --- a/po/ja.po +++ b/po/ja.po @@ -7,7 +7,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-good 0.10.14.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-09-12 00:08+0100\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2009-06-01 19:29+0900\n" "Last-Translator: Makoto Kato \n" "Language-Team: Japanese \n" @@ -73,12 +73,15 @@ msgstr "このファイルには再生不可能なストリームが含まれて msgid "This file is invalid and cannot be played." msgstr "このファイルは不正なため再生することができません" -msgid "This file is incomplete and cannot be played." -msgstr "このファイルは正常ではないため、再生することができません" - msgid "This file is corrupt and cannot be played." msgstr "このファイルは壊れているため再生することができません" +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." +msgstr "このファイルは正常ではないため、再生することができません" + msgid "The video in this file might not play correctly." msgstr "このファイルの中の動画は正しく再生できないかもしれません" @@ -697,6 +700,17 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "" +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "" +"デバイス '%s' 上の現在の入力の取得に失敗しました。ラジオデバイスかもしれませ" +"ん" + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "デバイス '%2$s' 上のチューナー %1$d の設定の取得に失敗しました。" + #, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "" diff --git a/po/lt.po b/po/lt.po index 2a213c69b0..29d9afb4ca 100644 --- a/po/lt.po +++ b/po/lt.po @@ -7,7 +7,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-good 0.10.23.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-09-12 00:08+0100\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2010-07-16 19:34+0300\n" "Last-Translator: Žygimantas Beručka \n" "Language-Team: Lithuanian \n" @@ -72,12 +72,15 @@ msgstr "Šiame faile nėra atkurtinų srautų." msgid "This file is invalid and cannot be played." msgstr "Failas netinkamas ir negali būti atkurtas." -msgid "This file is incomplete and cannot be played." -msgstr "Failas nebaigtas ir negali būti atkurtas." - msgid "This file is corrupt and cannot be played." msgstr "Failas sugadintas ir negali būti atkurtas." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." +msgstr "Failas nebaigtas ir negali būti atkurtas." + msgid "The video in this file might not play correctly." msgstr "Šiame faile esantis vaizdo įrašas gali būti atkurtas nekorektiškai." @@ -624,6 +627,17 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Nepavyko nustatyti įvesties %d įrenginyje %s." +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "" +"Nepavyko gauti dabartinės įvesties įrenginyje „%s“. Galbūt tai radijo " +"įrenginys" + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "Nepavyko nustatyti įvesties %d įrenginyje %s." + #, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "Nepavyko sustatyti į eilė buferių įrenginyje „%s“." diff --git a/po/lv.po b/po/lv.po index 1b3f6916c3..f2ee50cca1 100644 --- a/po/lv.po +++ b/po/lv.po @@ -7,7 +7,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-good 0.10.23.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-09-12 00:08+0100\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2010-07-07 11:53+0100\n" "Last-Translator: Rihards Priedītis \n" "Language-Team: Latvian \n" @@ -72,12 +72,15 @@ msgstr "Šis fails nesatur nevienu atskaņojamu straumi." msgid "This file is invalid and cannot be played." msgstr "Šis fails ir nederīgs un nevar tikt atskaņots." -msgid "This file is incomplete and cannot be played." -msgstr "Šis pails ir nepabeigts un nevar tikt atskaņots." - msgid "This file is corrupt and cannot be played." msgstr "Šis fails ir bojāts un nevar tikt atskaņots." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." +msgstr "Šis pails ir nepabeigts un nevar tikt atskaņots." + msgid "The video in this file might not play correctly." msgstr "Video šajā failā var tikt atskaņots nepareizi." @@ -626,6 +629,17 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Neizdevās uzstādīt ievadi %d uz ierīces %s." +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "" +"Neizdevās saņemt pašreizējo ievadi no ierīces \"%s\". Iespējams tā ir radio " +"ierīce" + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "Neizdevās uzstādīt ievadi %d uz ierīces %s." + #, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "Nevar ierindod buferus ierīcē \"%s\"." diff --git a/po/mt.po b/po/mt.po index b8ab7c36bd..9b847571a3 100644 --- a/po/mt.po +++ b/po/mt.po @@ -5,7 +5,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-good-0.10.10.3\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-09-12 00:08+0100\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2008-10-26 19:09+0100\n" "Last-Translator: Michel Bugeja \n" "Language-Team: Maltese \n" @@ -70,12 +70,15 @@ msgstr "Il-fajl ma fiħ l-ebda stream li tista tindaqq." msgid "This file is invalid and cannot be played." msgstr "Il-fajl mhux validu u ma jistax jindaqq" -msgid "This file is incomplete and cannot be played." -msgstr "Il-fajl mhux komplut u ma jistax jindaqq," - msgid "This file is corrupt and cannot be played." msgstr "Il-fajl huwa korrott u ma jistax jinfetaħ." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." +msgstr "Il-fajl mhux komplut u ma jistax jindaqq," + msgid "The video in this file might not play correctly." msgstr "Jista jkun illi l-vidjo ġo dan il-fajl ma jidhirx sewwa." @@ -658,6 +661,17 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Problema biex nissettja input %d fuq apparat %s." +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "" +"Problema biex inġib current input fuq apparat '%s'. Jista jkun li huwa " +"apparat tar-radju." + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "Problema biex nissettja input %d fuq apparat %s." + #, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "Could not enqueue buffers in device '%s'." diff --git a/po/nb.po b/po/nb.po index fd4cb7bbf8..f37e3b0b0a 100644 --- a/po/nb.po +++ b/po/nb.po @@ -6,7 +6,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-good 0.10.25.3\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-11-01 17:27+0000\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2010-10-24 21:53+0200\n" "Last-Translator: Kjartan Maraas \n" "Language-Team: Norwegian Bokmaal \n" @@ -66,10 +66,13 @@ msgstr "" msgid "This file is invalid and cannot be played." msgstr "" -msgid "This file is incomplete and cannot be played." +msgid "This file is corrupt and cannot be played." msgstr "" -msgid "This file is corrupt and cannot be played." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." msgstr "" msgid "The video in this file might not play correctly." @@ -602,6 +605,15 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "" +#, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "" + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "Kunne ikke lukke VFS-fil «%s»." + #, fuzzy, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "Kunne ikke lukke VFS-fil «%s»." diff --git a/po/nl.po b/po/nl.po index 6c3939c899..5f6f242e56 100644 --- a/po/nl.po +++ b/po/nl.po @@ -7,7 +7,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-good 0.10.25.3\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-11-01 17:27+0000\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2010-10-25 20:15+0200\n" "Last-Translator: Freek de Kruijf \n" "Language-Team: Dutch \n" @@ -71,12 +71,15 @@ msgstr "Dit bestand bevat geen afspeelbare stromen." msgid "This file is invalid and cannot be played." msgstr "Dit bestand is ongeldig en kan niet afgespeeld worden." -msgid "This file is incomplete and cannot be played." -msgstr "Dit bestand is incompleet en kan niet afgespeeld worden." - msgid "This file is corrupt and cannot be played." msgstr "Dit bestand is beschadigd en kan niet afgespeeld worden." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." +msgstr "Dit bestand is incompleet en kan niet afgespeeld worden." + msgid "The video in this file might not play correctly." msgstr "De video in dit bestand zal mogelijk niet correct afgespeeld worden." @@ -627,6 +630,17 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Kan invoer %d op apparaat '%s' niet instellen." +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "" +"Kan de huidige instellingen van apparaat '%s' niet verkrijgen. Het is " +"mogelijk een radio" + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "Kan invoer %d op apparaat '%s' niet instellen." + #, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "Kan geen buffers toekennen in apparaat '%s'." diff --git a/po/or.po b/po/or.po index 0f4ebbc6bb..f91b2daf39 100644 --- a/po/or.po +++ b/po/or.po @@ -8,7 +8,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-0.8.3\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-09-12 00:08+0100\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2004-09-27 13:32+0530\n" "Last-Translator: Gora Mohanty \n" "Language-Team: Oriya \n" @@ -72,10 +72,13 @@ msgstr "" msgid "This file is invalid and cannot be played." msgstr "" -msgid "This file is incomplete and cannot be played." +msgid "This file is corrupt and cannot be played." msgstr "" -msgid "This file is corrupt and cannot be played." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." msgstr "" msgid "The video in this file might not play correctly." @@ -628,6 +631,15 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "\"%s\" ଧ୍ବନି ଯନ୍ତ୍ର ବନ୍ଦ କରିହେଲା ନାହିଁ." +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "\"%s\" ଯନ୍ତ୍ରରୁ ପର୍ଯ୍ଯାପ୍ତ ଅସ୍ଥାୟୀ ସଞ୍ଚୟ ସ୍ଥାନ ଆଣିହେଲା ନାହିଁ." + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "\"%s\" ଧ୍ବନି ଯନ୍ତ୍ର ବନ୍ଦ କରିହେଲା ନାହିଁ." + #, fuzzy, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "\"%s\" ଯନ୍ତ୍ରରୁ ଅସ୍ଥାୟୀ ସଞ୍ଚୟ ସ୍ଥାନ ଆଣିହେଲା ନାହିଁ." diff --git a/po/pl.po b/po/pl.po index 59347df17f..af5af264f8 100644 --- a/po/pl.po +++ b/po/pl.po @@ -6,7 +6,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-good 0.10.25.3\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-11-01 17:27+0000\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2010-10-25 07:27+0200\n" "Last-Translator: Jakub Bogusz \n" "Language-Team: Polish \n" @@ -66,12 +66,15 @@ msgstr "Ten plik nie zawiera strumieni nadających się do odtworzenia." msgid "This file is invalid and cannot be played." msgstr "Ten plik jest błędny i nie może być odtworzony." -msgid "This file is incomplete and cannot be played." -msgstr "Ten plik jest niekompletny i nie może być odtworzony." - msgid "This file is corrupt and cannot be played." msgstr "Ten plik jest uszkodzony i nie może być odtworzony." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." +msgstr "Ten plik jest niekompletny i nie może być odtworzony." + msgid "The video in this file might not play correctly." msgstr "Obraz w tym pliku może nie być odtwarzany prawidłowo." @@ -622,6 +625,16 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Nie udało się ustawić wejścia %d urządzenia %s." +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "" +"Nie udało się uzyskać aktualnego wejścia urządzenia '%s'. Może to radio" + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "Nie udało się ustawić wejścia %d urządzenia %s." + #, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "Nie udało się skolejkować buforów urządzenia '%s'." diff --git a/po/pt_BR.po b/po/pt_BR.po index 5a80f5c25b..0c0af7d32f 100644 --- a/po/pt_BR.po +++ b/po/pt_BR.po @@ -11,7 +11,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-good 0.10.23.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-09-12 00:08+0100\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2010-06-28 21:00-0300\n" "Last-Translator: Fabrício Godoy \n" "Language-Team: Brazilian Portuguese \n" @@ -75,12 +75,15 @@ msgstr "Este arquivo não contêm fluxos reproduzíveis." msgid "This file is invalid and cannot be played." msgstr "Este arquivo é inválido e não pôde ser reproduzido." -msgid "This file is incomplete and cannot be played." -msgstr "Este arquivo está incompleto e não pôde ser reproduzido." - msgid "This file is corrupt and cannot be played." msgstr "Este arquivo está corrompido e não pôde ser reproduzido." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." +msgstr "Este arquivo está incompleto e não pôde ser reproduzido." + msgid "The video in this file might not play correctly." msgstr "O vídeo neste arquivo pode não ser reproduzido corretamente." @@ -634,6 +637,17 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Falha ao definir a entrada %d no dispositivo %s." +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "" +"Falha ao obter a entrada atual no dispositivo \"%s\". Talvez seja um " +"dispositivo de rádio" + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "Falha ao definir a entrada %d no dispositivo %s." + #, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "Não possível adicionar buffers à fila no dispositivo \"%s\"." diff --git a/po/ro.po b/po/ro.po index 896616e532..97ac758708 100644 --- a/po/ro.po +++ b/po/ro.po @@ -5,7 +5,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-good 0.10.23.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-09-12 00:08+0100\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2010-08-16 03:22+0300\n" "Last-Translator: Lucian Adrian Grijincu \n" "Language-Team: Romanian \n" @@ -72,12 +72,15 @@ msgstr "Acest fișier nu conține fluxuri ce pot fi redate." msgid "This file is invalid and cannot be played." msgstr "Fișierul nu este valid și nu poate fi redat." -msgid "This file is incomplete and cannot be played." -msgstr "Acest fișier nu este complet și nu poate fi redat." - msgid "This file is corrupt and cannot be played." msgstr "Acest fișier este corupt și nu poate fi redat." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." +msgstr "Acest fișier nu este complet și nu poate fi redat." + msgid "The video in this file might not play correctly." msgstr "Este posibil ca fișierul video să nu fie redat corect." @@ -633,6 +636,17 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Definirea valorii %d pentru dispozitivul „%s” a eșuat." +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "" +"Obținerea intrări curente pentru dispozitivul „%s” a eșuat. Posibil să fie " +"un dispozitiv radio." + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "Definirea valorii %d pentru dispozitivul „%s” a eșuat." + #, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "Nu s-a putut programa memoria tampon în dispozitivul „%s”." diff --git a/po/ru.po b/po/ru.po index 28bf04077d..0428d5fe67 100644 --- a/po/ru.po +++ b/po/ru.po @@ -7,7 +7,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-good 0.10.13.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-09-12 00:08+0100\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2009-02-12 14:35+0200\n" "Last-Translator: Pavel Maryanov \n" "Language-Team: Russian \n" @@ -72,12 +72,15 @@ msgstr "Файл не содержит воспроизводимых поток msgid "This file is invalid and cannot be played." msgstr "Файл некорректен и не может быть воспроизведён." -msgid "This file is incomplete and cannot be played." -msgstr "Файл не полон и не может быть воспроизведён." - msgid "This file is corrupt and cannot be played." msgstr "Файл повреждён и не может быть воспроизведён." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." +msgstr "Файл не полон и не может быть воспроизведён." + msgid "The video in this file might not play correctly." msgstr "Видео из этого файла может быть воспроизведено некорректно." @@ -663,6 +666,17 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Не удалось выбрать вход %d для устройства %s" +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "" +"Не удалось определить активный вход устройства «%s». Возможно, это радио-" +"устройство" + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "Не удалось выбрать вход %d для устройства %s" + #, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "Не удалось добавить в очередь буферы устройства «%s»" diff --git a/po/sk.po b/po/sk.po index b9048a1e1d..c1304b08ab 100644 --- a/po/sk.po +++ b/po/sk.po @@ -6,7 +6,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-good 0.10.25.3\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-11-18 00:43+0000\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2010-11-08 15:48+0100\n" "Last-Translator: Peter Tuhársky \n" "Language-Team: Slovak \n" @@ -69,12 +69,15 @@ msgstr "Tento súbor neobsahuje žiadne prehrateľné prúdy údajov." msgid "This file is invalid and cannot be played." msgstr "Tento súbor je chybný a nedá sa prehrať." -msgid "This file is incomplete and cannot be played." -msgstr "Tento súbor je neúplný a nedá sa prehrať." - msgid "This file is corrupt and cannot be played." msgstr "Tento súbor je poškodený a nedá sa prehrať." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." +msgstr "Tento súbor je neúplný a nedá sa prehrať." + msgid "The video in this file might not play correctly." msgstr "Video v tomto súbore možno nebude hrať korektne." @@ -626,6 +629,16 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Nepodarilo sa nastaviť vstup %d na zariadení %s." +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "" +"Nepodarilo sa zistiť súčasný vstup na zariadení '%s'. Možno je to rádio." + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "Nepodarilo sa nastaviť vstup %d na zariadení %s." + #, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "Nepodarilo sa zaradiť vyrovnávaciu pamäť na zariadení '%s'." diff --git a/po/sl.po b/po/sl.po index c07e0af215..4472288878 100644 --- a/po/sl.po +++ b/po/sl.po @@ -9,7 +9,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-good 0.10.25.3\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-11-01 17:27+0000\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2010-10-25 12:23+0100\n" "Last-Translator: Matej Urbančič \n" "Language-Team: Slovenian \n" @@ -75,12 +75,15 @@ msgstr "Datoteka ne vsebuje pretokov za predvajanje." msgid "This file is invalid and cannot be played." msgstr "Datoteka je napačna, zato je ni mogoče predvajati." -msgid "This file is incomplete and cannot be played." -msgstr "Datoteka ni popolna, zato je ni mogoče predvajati." - msgid "This file is corrupt and cannot be played." msgstr "Datoteka je pokvarjena, zato je ni mogoče predvajati." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." +msgstr "Datoteka ni popolna, zato je ni mogoče predvajati." + msgid "The video in this file might not play correctly." msgstr "Video zapis v tej datoteki morda ne bo predvajan pravilno." @@ -631,6 +634,17 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Napaka med določanjem vhoda %d naprave %s." +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "" +"Napaka med pridobivanjem vhodnega signala naprave '%s'. Morda je radijska " +"naprava." + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "Napaka med določanjem vhoda %d naprave %s." + #, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "Ni mogoče uvrstiti medpomnilnika naprave '%s'." diff --git a/po/sq.po b/po/sq.po index 9c53eda84b..014e9cbb82 100644 --- a/po/sq.po +++ b/po/sq.po @@ -6,7 +6,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins 0.8.3\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-09-12 00:08+0100\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2004-08-07 20:29+0200\n" "Last-Translator: Laurent Dhima \n" "Language-Team: Albanian \n" @@ -70,10 +70,13 @@ msgstr "" msgid "This file is invalid and cannot be played." msgstr "" -msgid "This file is incomplete and cannot be played." +msgid "This file is corrupt and cannot be played." msgstr "" -msgid "This file is corrupt and cannot be played." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." msgstr "" msgid "The video in this file might not play correctly." @@ -626,6 +629,15 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "E pamundur mbyllja e dispozitivit audio \"%s\"." +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "E pamundur marrja e buffers të mjaftueshëm nga dispozitivi \"%s\"." + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "E pamundur mbyllja e dispozitivit audio \"%s\"." + #, fuzzy, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "E pamundur marrja e buffers nga dispozitivi \"%s\"." diff --git a/po/sr.po b/po/sr.po index 81f69de103..528ffe5b24 100644 --- a/po/sr.po +++ b/po/sr.po @@ -6,7 +6,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins 0.7.6\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-09-12 00:08+0100\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2004-03-13 00:18+0100\n" "Last-Translator: Danilo Segan \n" "Language-Team: Serbian \n" @@ -71,10 +71,13 @@ msgstr "" msgid "This file is invalid and cannot be played." msgstr "" -msgid "This file is incomplete and cannot be played." +msgid "This file is corrupt and cannot be played." msgstr "" -msgid "This file is corrupt and cannot be played." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." msgstr "" msgid "The video in this file might not play correctly." @@ -627,6 +630,15 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Не могу да отворим радио уређај '%s'" +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "Не могу да примим довољно бафера са уређаја „%s“." + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "Не могу да отворим радио уређај '%s'" + #, fuzzy, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "Не могу да примим бафере са уређаја „%s“." diff --git a/po/sv.po b/po/sv.po index 99f25c7ade..aaf8a30710 100644 --- a/po/sv.po +++ b/po/sv.po @@ -8,7 +8,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-good 0.10.25.3\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-11-01 17:27+0000\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2010-10-26 08:50+0100\n" "Last-Translator: Daniel Nylander \n" "Language-Team: Swedish \n" @@ -68,12 +68,16 @@ msgstr "Den här filen innehåller inga uppspelningsbara strömmar." msgid "This file is invalid and cannot be played." msgstr "Den här filen är ogiltig och kan inte spelas upp." -msgid "This file is incomplete and cannot be played." -msgstr "Den här filen är inte fullständig och kan inte spelas upp." - msgid "This file is corrupt and cannot be played." msgstr "Den här filen är skadad och kan inte spelas upp." +#, fuzzy +msgid "Invalid atom size." +msgstr "Ogiltig cd-enhet" + +msgid "This file is incomplete and cannot be played." +msgstr "Den här filen är inte fullständig och kan inte spelas upp." + msgid "The video in this file might not play correctly." msgstr "Videon i den här filen kanske inte kan spelas upp korrekt." @@ -630,6 +634,17 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Kunde inte ställa in ingång %d på enheten %s." +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "" +"Misslyckades med att få tag på aktuell ingång på enheten \"%s\". Kan vara en " +"radioenhet" + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "Kunde inte ställa in ingång %d på enheten %s." + #, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "Kunde inte kölägga buffertar i enheten \"%s\"." @@ -1486,9 +1501,6 @@ msgstr "Kan inte fungera utan en klocka" #~ msgid "_Set device" #~ msgstr "_Ställ in enhet" -#~ msgid "Invalid CD device" -#~ msgstr "Ogiltig cd-enhet" - #~ msgid "Volume control" #~ msgstr "Volymkontroll" diff --git a/po/tr.po b/po/tr.po index d780abdfbd..d4b3230b5a 100644 --- a/po/tr.po +++ b/po/tr.po @@ -5,7 +5,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-good 0.10.25.3\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-11-18 00:43+0000\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2010-11-16 16:41+0200\n" "Last-Translator: Server Acim \n" "Language-Team: Turkish \n" @@ -67,12 +67,15 @@ msgstr "Dosya çalınabilir akışlar içermiyor." msgid "This file is invalid and cannot be played." msgstr "Bu dosya geçersiz ve oynatılamaz." -msgid "This file is incomplete and cannot be played." -msgstr "Dosya eksik ve oynatılamaz." - msgid "This file is corrupt and cannot be played." msgstr "Bu dosya bozuk ve oynatılamaz." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." +msgstr "Dosya eksik ve oynatılamaz." + msgid "The video in this file might not play correctly." msgstr "Bu dosyadaki vidyo doğru oynatılamıyabilir." @@ -615,6 +618,16 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Giriş değeri olarak bu %d şu aygıtta %s ayarlanamadı." +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "" +"Şu aygıtta '%s' geçerli giriş elde edilemedi. O bir radyo aygıtı olabilir." + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "Giriş değeri olarak bu %d şu aygıtta %s ayarlanamadı." + #, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "Aygıtta '%s' arabellek kuyruğa sokulamıyor." diff --git a/po/uk.po b/po/uk.po index 6c54efa4c6..34eafceb87 100644 --- a/po/uk.po +++ b/po/uk.po @@ -7,7 +7,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-good 0.10.6\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-09-12 00:08+0100\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2007-07-05 15:40+0200\n" "Last-Translator: Maxim V. Dziumanenko \n" "Language-Team: Ukrainian \n" @@ -71,12 +71,15 @@ msgstr "Файл містить потоки, які неможливо відт msgid "This file is invalid and cannot be played." msgstr "Файл неповний або не може відтворюватись." -msgid "This file is incomplete and cannot be played." -msgstr "Файл неповний або не може відтворюватись." - msgid "This file is corrupt and cannot be played." msgstr "Файл пошкоджений та не може бути відтворений." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." +msgstr "Файл неповний або не може відтворюватись." + msgid "The video in this file might not play correctly." msgstr "Потік відео у цьому файлі неможливо коректно відтворити." @@ -616,6 +619,17 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Не вдається встановити ввід %d пристрою \"%s\"." +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "" +"Не вдається отримати поточний ввід пристрою \"%s\". Можливо цей пристрій - " +"радіо." + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "Не вдається встановити ввід %d пристрою \"%s\"." + #, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "Не вдається опитати буфери від пристрою \"%s\"." diff --git a/po/vi.po b/po/vi.po index bb2a94d0ba..d3d7f5e41a 100644 --- a/po/vi.po +++ b/po/vi.po @@ -7,7 +7,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-good 0.10.23.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-10-16 01:23+0100\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2010-10-03 18:48+1030\n" "Last-Translator: Clytie Siddall \n" "Language-Team: Vietnamese \n" @@ -73,12 +73,15 @@ msgstr "Tập tin này không chứa luồng có thể phát." msgid "This file is invalid and cannot be played." msgstr "Tập tin này không hợp lệ nên không thể phát được." -msgid "This file is incomplete and cannot be played." -msgstr "Tập tin này chưa hoàn thành nên không thể được phát." - msgid "This file is corrupt and cannot be played." msgstr "Tập tin này bị hỏng nên không thể phát." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." +msgstr "Tập tin này chưa hoàn thành nên không thể được phát." + msgid "The video in this file might not play correctly." msgstr "Ảnh động trong tập tin này có thể không phát đúng." @@ -626,6 +629,16 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Lỗi đặt dữ liệu nhập %d vào thiết bị « %s »." +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "" +"Lỗi lấy kết nhập hiện thời vào thiết bị « %s ». Có thể là thiết bị thu thanh." + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "Lỗi đặt dữ liệu nhập %d vào thiết bị « %s »." + #, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "Không thể phụ thêm các bộ đệm vào hàng đợi trên thiết bị « %s »." diff --git a/po/zh_CN.po b/po/zh_CN.po index 154344a167..20354bb49f 100644 --- a/po/zh_CN.po +++ b/po/zh_CN.po @@ -7,7 +7,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-good 0.10.16.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-09-12 00:08+0100\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2009-11-13 22:20+0800\n" "Last-Translator: Ji ZhengYu \n" "Language-Team: Chinese (simplified) \n" @@ -70,12 +70,15 @@ msgstr "此文件不包含可播放的流。" msgid "This file is invalid and cannot be played." msgstr "此文件无效，无法播放。" -msgid "This file is incomplete and cannot be played." -msgstr "此文件不完整且无法播放。" - msgid "This file is corrupt and cannot be played." msgstr "此文件已损坏，无法播放。" +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." +msgstr "此文件不完整且无法播放。" + msgid "The video in this file might not play correctly." msgstr "此文件中的视频可能无法正确播放。" @@ -633,6 +636,15 @@ msgstr "获取设备‘%s’上的当前输入出错。也许它是一个广播 msgid "Failed to set input %d on device %s." msgstr "设置设备 %2$s 上的输入 %1$d 时出错。" +#, fuzzy, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "获取设备‘%s’上的当前输入出错。也许它是一个广播设备" + +#, fuzzy, c-format +msgid "Failed to set output %d on device %s." +msgstr "设置设备 %2$s 上的输入 %1$d 时出错。" + #, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "无法对设备‘%s’中的缓冲区进行排序。" diff --git a/po/zh_HK.po b/po/zh_HK.po index 2d3c2400a6..1049ff8182 100644 --- a/po/zh_HK.po +++ b/po/zh_HK.po @@ -6,7 +6,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-good-0.10.2 0.10.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-09-12 00:08+0100\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2006-08-29 01:08+0800\n" "Last-Translator: Abel Cheung \n" "Language-Team: Chinese (Hong Kong) \n" @@ -66,10 +66,13 @@ msgstr "" msgid "This file is invalid and cannot be played." msgstr "" -msgid "This file is incomplete and cannot be played." +msgid "This file is corrupt and cannot be played." msgstr "" -msgid "This file is corrupt and cannot be played." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." msgstr "" msgid "The video in this file might not play correctly." @@ -623,6 +626,15 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "" +#, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "" + +#, c-format +msgid "Failed to set output %d on device %s." +msgstr "" + #, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "" diff --git a/po/zh_TW.po b/po/zh_TW.po index 7c944e3042..eb1b3534ea 100644 --- a/po/zh_TW.po +++ b/po/zh_TW.po @@ -6,7 +6,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-good-0.10.2 0.10.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2010-09-12 00:08+0100\n" +"POT-Creation-Date: 2011-01-07 01:15+0000\n" "PO-Revision-Date: 2006-08-29 01:08+0800\n" "Last-Translator: Abel Cheung \n" "Language-Team: Chinese (traditional) \n" @@ -66,10 +66,13 @@ msgstr "" msgid "This file is invalid and cannot be played." msgstr "" -msgid "This file is incomplete and cannot be played." +msgid "This file is corrupt and cannot be played." msgstr "" -msgid "This file is corrupt and cannot be played." +msgid "Invalid atom size." +msgstr "" + +msgid "This file is incomplete and cannot be played." msgstr "" msgid "The video in this file might not play correctly." @@ -623,6 +626,15 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "" +#, c-format +msgid "" +"Failed to get current output on device '%s'. May be it is a radio device" +msgstr "" + +#, c-format +msgid "Failed to set output %d on device %s." +msgstr "" + #, c-format msgid "Could not enqueue buffers in device '%s'." msgstr "" From 99d01f2e0fc4e0b08bb1cb1d9599da30f4357ff8 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Fri, 7 Jan 2011 13:24:02 +0000 Subject: [PATCH 229/351] 0.10.26.2 pre-release --- configure.ac | 6 +++--- win32/common/config.h | 13 ++++++++----- 2 files changed, 11 insertions(+), 8 deletions(-) diff --git a/configure.ac b/configure.ac index 246e1f2b58..5ee8e87b37 100644 --- a/configure.ac +++ b/configure.ac @@ -5,7 +5,7 @@ dnl please read gstreamer/docs/random/autotools before changing this file dnl initialize autoconf dnl releases only do -Wall, git and prerelease does -Werror too dnl use a three digit version number for releases, and four for git/pre -AC_INIT(GStreamer Good Plug-ins, 0.10.26.1, +AC_INIT(GStreamer Good Plug-ins, 0.10.26.2, http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer, gst-plugins-good) @@ -52,8 +52,8 @@ AC_LIBTOOL_WIN32_DLL AM_PROG_LIBTOOL dnl *** required versions of GStreamer stuff *** -GST_REQ=0.10.31.1 -GSTPB_REQ=0.10.31.1 +GST_REQ=0.10.31.2 +GSTPB_REQ=0.10.31.2 dnl *** autotools stuff **** diff --git a/win32/common/config.h b/win32/common/config.h index 521659e1d1..da86c35383 100644 --- a/win32/common/config.h +++ b/win32/common/config.h @@ -43,13 +43,13 @@ #define GST_LICENSE "LGPL" /* package name in plugins */ -#define GST_PACKAGE_NAME "GStreamer Good Plug-ins git" +#define GST_PACKAGE_NAME "GStreamer Good Plug-ins prerelease" /* package origin */ #define GST_PACKAGE_ORIGIN "Unknown package origin" /* GStreamer package release date/time for plugins as YYYY-MM-DD */ -#define GST_PACKAGE_RELEASE_DATETIME "2010-12-02T18:08Z" +#define GST_PACKAGE_RELEASE_DATETIME "2011-01-07T01:34Z" /* struct v4l2_buffer missing */ #undef GST_V4L2_MISSING_BUFDECL @@ -204,6 +204,9 @@ /* Define to 1 if you have the `isinf' function. */ #undef HAVE_ISINF +/* Define to enable Jack (used by jack). */ +#undef HAVE_JACK + /* Define to enable jpeg library (used by jpeg). */ #undef HAVE_JPEG @@ -383,7 +386,7 @@ #define PACKAGE_NAME "GStreamer Good Plug-ins" /* Define to the full name and version of this package. */ -#define PACKAGE_STRING "GStreamer Good Plug-ins 0.10.26.1" +#define PACKAGE_STRING "GStreamer Good Plug-ins 0.10.26.2" /* Define to the one symbol short name of this package. */ #define PACKAGE_TARNAME "gst-plugins-good" @@ -392,7 +395,7 @@ #undef PACKAGE_URL /* Define to the version of this package. */ -#define PACKAGE_VERSION "0.10.26.1" +#define PACKAGE_VERSION "0.10.26.2" /* directory where plugins are located */ #ifdef _DEBUG @@ -423,7 +426,7 @@ #undef STDC_HEADERS /* Version number of package */ -#define VERSION "0.10.26.1" +#define VERSION "0.10.26.2" /* old wavpack API */ #undef WAVPACK_OLD_API From 2cb2fa729a54586c00875539c92ca893e0d7f5ea Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Mon, 10 Jan 2011 14:53:39 +0000 Subject: [PATCH 230/351] Automatic update of common submodule From 46445ad to ccbaa85 --- common | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/common b/common index 46445ad50f..ccbaa85fc5 160000 --- a/common +++ b/common @@ -1 +1 @@ -Subproject commit 46445ad50f184d2640dd205361e0446e9121ba5f +Subproject commit ccbaa85fc58099fb90f1b5a0fad4d92d45500d19 From b6a962e6eb9f417e3d29c8d74f000f9ffbc57ea8 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Mon, 10 Jan 2011 16:36:19 +0000 Subject: [PATCH 231/351] Automatic update of common submodule From ccbaa85 to e572c87 --- common | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/common b/common index ccbaa85fc5..e572c87c58 160000 --- a/common +++ b/common @@ -1 +1 @@ -Subproject commit ccbaa85fc58099fb90f1b5a0fad4d92d45500d19 +Subproject commit e572c87c582166a6c65edf2a6a0fa73e31a3b04c From 696767c39c6e01b14b40e4150ca4edd5b5939951 Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Tue, 11 Jan 2011 15:50:28 +0200 Subject: [PATCH 232/351] Automatic update of common submodule From e572c87 to f94d739 --- common | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/common b/common index e572c87c58..f94d739915 160000 --- a/common +++ b/common @@ -1 +1 @@ -Subproject commit e572c87c582166a6c65edf2a6a0fa73e31a3b04c +Subproject commit f94d73991563ea2dcae2218b4847e32998f06816 From c046dca80a9321501ea9f9b8cefbb14adf7f49ac Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Tue, 11 Jan 2011 19:33:16 +0000 Subject: [PATCH 233/351] po: update translations --- po/da.po | 18 ++-- po/gl.po | 261 ++++++++++++++++++++++++---------------------------- po/pl.po | 18 ++-- po/pt_BR.po | 91 ++++-------------- po/sl.po | 24 ++--- po/sv.po | 21 ++--- po/tr.po | 17 ++-- 7 files changed, 184 insertions(+), 266 deletions(-) diff --git a/po/da.po b/po/da.po index 0959902e26..8d48f151e6 100644 --- a/po/da.po +++ b/po/da.po @@ -3,17 +3,17 @@ # This file is distributed under the same license as the gst-plugins-good package. # # Mogens Jaeger , 2007. -# Joe Hansen , 2008, 2009, 2010. +# Joe Hansen , 2008, 2009, 2010, 2011. # # gain -> forhøjelse # boost -> øgning? # msgid "" msgstr "" -"Project-Id-Version: gst-plugins-good-0.10.25.3\n" +"Project-Id-Version: gst-plugins-good-0.10.26.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-01-07 01:15+0000\n" -"PO-Revision-Date: 2010-10-26 23:54+0200\n" +"POT-Creation-Date: 2011-01-11 19:32+0000\n" +"PO-Revision-Date: 2011-01-07 23:54+0200\n" "Last-Translator: Joe Hansen \n" "Language-Team: Danish \n" "Language: da\n" @@ -77,7 +77,7 @@ msgid "This file is corrupt and cannot be played." msgstr "Denne fil er ødelagt og kan ikke afspilles." msgid "Invalid atom size." -msgstr "" +msgstr "Ugyldig atomstørrelse." msgid "This file is incomplete and cannot be played." msgstr "Denne fil er ufuldstændig og kan ikke afspilles." @@ -637,15 +637,15 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Kunne ikke sætte inddata %d for enhed %s." -#, fuzzy, c-format +#, c-format msgid "" "Failed to get current output on device '%s'. May be it is a radio device" msgstr "" -"Kunne ikke hente nuværende inddata for enhed '%s'. Måske er det en radioenhed" +"Kunne ikke hente nuværende uddata for enhed '%s'. Måske er det en radioenhed" -#, fuzzy, c-format +#, c-format msgid "Failed to set output %d on device %s." -msgstr "Kunne ikke sætte inddata %d for enhed %s." +msgstr "Kunne ikke sætte uddata %d for enhed %s." #, c-format msgid "Could not enqueue buffers in device '%s'." diff --git a/po/gl.po b/po/gl.po index 01937dc019..b34bcbaafe 100644 --- a/po/gl.po +++ b/po/gl.po @@ -1,21 +1,21 @@ # Galician translation of gst-plugins-good. # Copyright (C) 2009 gst-plugins-good's COPYRIGHT HOLDER # This file is distributed under the same license as the gst-plugins-good package. -# Fran Diéguez , 2009, 2010. +# Fran Diéguez , 2009, 2010, 2011. # msgid "" msgstr "" -"Project-Id-Version: gst-plugins-good 0.10.23.2\n" +"Project-Id-Version: gst-plugins-good 0.10.26.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-01-07 01:15+0000\n" -"PO-Revision-Date: 2010-08-12 15:46+0200\n" +"POT-Creation-Date: 2011-01-11 19:32+0000\n" +"PO-Revision-Date: 2011-01-09 21:20+0100\n" "Last-Translator: Fran Diéguez \n" "Language-Team: Galician \n" "Language: gl\n" "MIME-Version: 1.0\n" "Content-Type: text/plain; charset=UTF-8\n" "Content-Transfer-Encoding: 8bit\n" -"Plural-Forms: nplurals=2; plural=(n!=1);\\\n" +"Plural-Forms: nplurals=2; plural=(n!=1);\n" msgid "Could not establish connection to sound server" msgstr "Non foi posíbel estabelecer a conexión co servidor de son" @@ -38,28 +38,28 @@ msgid "Could not connect to server" msgstr "Non foi posíbel conectarse ao servidor" msgid "Server does not support seeking." -msgstr "" +msgstr "O servidor non admite a busca." -#, fuzzy msgid "Could not resolve server name." -msgstr "Non foi posíbel conectarse ao servidor" +msgstr "Non foi posíbel resolver o nome do servidor." -#, fuzzy msgid "Could not establish connection to server." -msgstr "Non foi posíbel estabelecer a conexión co servidor de son" +msgstr "Non foi posíbel estabelecer a conexión co servidor." msgid "Secure connection setup failed." -msgstr "" +msgstr "Produciuse un fallo de configuración da conexión segura." msgid "" "A network error occured, or the server closed the connection unexpectedly." msgstr "" +"Produciuse un erro de rede ou o servidor pechou a conexión de forma " +"inesperada." msgid "Server sent bad data." -msgstr "" +msgstr "O servidor enviou datos erróneos." msgid "No URL set." -msgstr "" +msgstr "No existe un URL estabelecido." msgid "No or invalid input audio, AVI stream will be corrupt." msgstr "O audio non existe ou non é válido, o fluxo AVI está corrompido." @@ -74,7 +74,7 @@ msgid "This file is corrupt and cannot be played." msgstr "Este ficheiro está danado e non pode reproducirse." msgid "Invalid atom size." -msgstr "" +msgstr "Tamaño atom non válido." msgid "This file is incomplete and cannot be played." msgstr "Este ficheiro está incompleto e non pode reproducirse." @@ -208,44 +208,41 @@ msgstr "" msgid "Could not open audio device for recording." msgstr "Non foi posíbel abrir o dispositivo de son para a gravación." -#, fuzzy msgid "Could not open audio device for mixer control handling." -msgstr "Non foi posíbel abrir o dispositivo de son para a gravación." +msgstr "" +"Non foi posíbel abrir o dispositivo de son para a súa xestión polo control " +"de misturado." -#, fuzzy msgid "" "Could not open audio device for mixer control handling. This version of the " "Open Sound System is not supported by this element." msgstr "" -"Non foi posíbel abrir o dispositivo de son para a gravación. Vostede non ten " -"permisos para abrir o dispositivo." +"Non foi posíbel abrir o dispositivo de son para a súa xestión polo control " +"de misturado. Este elemento non admite esta versión do Open Sound System." msgid "Master" -msgstr "" +msgstr "Mestre" msgid "Front" -msgstr "" +msgstr "Frontal" -#, fuzzy msgid "Rear" -msgstr "Gravar" +msgstr "Treseiro" -#, fuzzy msgid "Headphones" -msgstr "Cascos" +msgstr "Auriculares" msgid "Center" -msgstr "" +msgstr "Centro" msgid "LFE" -msgstr "" +msgstr "LFE" msgid "Surround" -msgstr "" +msgstr "Envolvente" -#, fuzzy msgid "Side" -msgstr "Vídeo" +msgstr "Lateral" msgid "Built-in Speaker" msgstr "Altofalante interno" @@ -256,308 +253,286 @@ msgstr "Saída AUX 1" msgid "AUX 2 Out" msgstr "Saída AUX 2" -#, fuzzy msgid "AUX Out" -msgstr "Saída AUX 1" +msgstr "Saída AUX" msgid "3D Depth" -msgstr "" +msgstr "Profundidade 3D" msgid "3D Center" -msgstr "" +msgstr "Centro 3D" msgid "3D Enhance" -msgstr "" +msgstr "Mellora 3D" -#, fuzzy msgid "Telephone" -msgstr "Cascos" +msgstr "Teléfono" msgid "Line Out" msgstr "Liña de saída" -#, fuzzy msgid "Line In" msgstr "Liña de entrada" msgid "Internal CD" -msgstr "" +msgstr "CD interno" -#, fuzzy msgid "Video In" -msgstr "Vídeo" +msgstr "Entrada de vídeo" -#, fuzzy msgid "AUX 1 In" -msgstr "Saída AUX 1" +msgstr "Entrada AUX 1" -#, fuzzy msgid "AUX 2 In" -msgstr "Saída AUX 2" +msgstr "Entrada AUX 2" msgid "AUX In" -msgstr "" +msgstr "Entrada AUX" -#, fuzzy msgid "Record Gain" -msgstr "Gravar" +msgstr "Ganancia de gravación" -#, fuzzy msgid "Output Gain" msgstr "Ganancia de saída" -#, fuzzy msgid "Microphone Boost" -msgstr "Micrófono" +msgstr "Aumento do micrófono" msgid "Loopback" -msgstr "" +msgstr "Bucle local" msgid "Diagnostic" -msgstr "" +msgstr "Diagnóstico" msgid "Bass Boost" -msgstr "" +msgstr "Aumento de baixos" msgid "Playback Ports" -msgstr "" +msgstr "Portos de reprodución" msgid "Input" -msgstr "" +msgstr "Entrada" -#, fuzzy msgid "Record Source" -msgstr "Gravar" +msgstr "Orixe da gravación" -#, fuzzy msgid "Monitor Source" -msgstr "Monitor" +msgstr "Monitor de orixe" msgid "Keyboard Beep" -msgstr "" +msgstr "Pitido de teclado" msgid "Simulate Stereo" -msgstr "" +msgstr "Simular estéreo" msgid "Stereo" -msgstr "" +msgstr "Estéreo" msgid "Surround Sound" -msgstr "" +msgstr "Son envolvente" -#, fuzzy msgid "Microphone Gain" -msgstr "Micrófono" +msgstr "Ganancia do micrófono" -#, fuzzy msgid "Speaker Source" -msgstr "Altofalante" +msgstr "Altofalante de orixe" -#, fuzzy msgid "Microphone Source" -msgstr "Micrófono" +msgstr "Micrófono de orixe" msgid "Jack" -msgstr "" +msgstr "Jack" msgid "Center / LFE" -msgstr "" +msgstr "Centro / LFE" msgid "Stereo Mix" -msgstr "" +msgstr "Misturador estéreo" msgid "Mono Mix" -msgstr "" +msgstr "Misturador mono" msgid "Input Mix" -msgstr "" +msgstr "Misturador de entrada" -#, fuzzy msgid "SPDIF In" -msgstr "Saída SPDIF" +msgstr "Entrada SPDIF" msgid "SPDIF Out" msgstr "Saída SPDIF" -#, fuzzy msgid "Microphone 1" -msgstr "Micrófono" +msgstr "Micrófono 1" -#, fuzzy msgid "Microphone 2" -msgstr "Micrófono" +msgstr "Micrófono 2" -#, fuzzy msgid "Digital Out" -msgstr "Dixital 1" +msgstr "Saída dixital" -#, fuzzy msgid "Digital In" -msgstr "Dixital 1" +msgstr "Entrada dixital" msgid "HDMI" -msgstr "" +msgstr "HDMI" msgid "Modem" -msgstr "" +msgstr "Módem" msgid "Handset" -msgstr "" +msgstr "Auriculares" msgid "Other" -msgstr "" +msgstr "Outro" msgid "None" -msgstr "" +msgstr "Ningún" msgid "On" -msgstr "" +msgstr "Acendido" msgid "Off" -msgstr "" +msgstr "Apagado" msgid "Mute" -msgstr "" +msgstr "Silenciar" msgid "Fast" -msgstr "" +msgstr "Rápido" #. TRANSLATORS: "Very Low" is a quality setting here msgid "Very Low" -msgstr "" +msgstr "Moi baixo" #. TRANSLATORS: "Low" is a quality setting here msgid "Low" -msgstr "" +msgstr "Baixo" #. TRANSLATORS: "Medium" is a quality setting here msgid "Medium" -msgstr "" +msgstr "Medio" #. TRANSLATORS: "High" is a quality setting here msgid "High" -msgstr "" +msgstr "Alto" #. TRANSLATORS: "Very High" is a quality setting here msgid "Very High" -msgstr "" +msgstr "Moi alto" #. TRANSLATORS: "Production" is a quality setting here msgid "Production" -msgstr "" +msgstr "Produción" -#, fuzzy msgid "Front Panel Microphone" -msgstr "Micrófono" +msgstr "Micrófono do panel frontal" msgid "Front Panel Line In" -msgstr "" +msgstr "Entrada do panel frontal" msgid "Front Panel Headphones" -msgstr "" +msgstr "Auriculares do panel frontal" msgid "Front Panel Line Out" -msgstr "" +msgstr "Saída do panel frontal" msgid "Green Connector" -msgstr "" +msgstr "Conectador verde" msgid "Pink Connector" -msgstr "" +msgstr "Conectador rosa" msgid "Blue Connector" -msgstr "" +msgstr "Conectador azul" msgid "White Connector" -msgstr "" +msgstr "Conectador branco" msgid "Black Connector" -msgstr "" +msgstr "Conectador negro" msgid "Gray Connector" -msgstr "" +msgstr "Conectador gris" msgid "Orange Connector" -msgstr "" +msgstr "Conectador laranxa" msgid "Red Connector" -msgstr "" +msgstr "Conectador vermello" msgid "Yellow Connector" -msgstr "" +msgstr "Conectador amarelo" msgid "Green Front Panel Connector" -msgstr "" +msgstr "Conectador verde do panel frontal" msgid "Pink Front Panel Connector" -msgstr "" +msgstr "Conectador rosa do panel frontal" msgid "Blue Front Panel Connector" -msgstr "" +msgstr "Conectador azul do panel frontal" msgid "White Front Panel Connector" -msgstr "" +msgstr "Conectador branco do panel frontal" msgid "Black Front Panel Connector" -msgstr "" +msgstr "Conectador negro do panel frontal" msgid "Gray Front Panel Connector" -msgstr "" +msgstr "Conectador gris do panel frontal" msgid "Orange Front Panel Connector" -msgstr "" +msgstr "Conectador laranxado panel frontal" msgid "Red Front Panel Connector" -msgstr "" +msgstr "Conectador vermello do panel frontal" msgid "Yellow Front Panel Connector" -msgstr "" +msgstr "Conectador amarelo do panel frontal" msgid "Spread Output" -msgstr "" +msgstr "Expandir saída" msgid "Downmix" -msgstr "" +msgstr "Redución de canles" msgid "Virtual Mixer Input" -msgstr "" +msgstr "Entrada do misturador virtual" msgid "Virtual Mixer Output" -msgstr "" +msgstr "Saída do misturador virtual" msgid "Virtual Mixer Channels" -msgstr "" +msgstr "Canles do misturador virtual" #. TRANSLATORS: name + number of a volume mixer control #, c-format msgid "%s %d Function" -msgstr "" +msgstr "Función %s %d" #. TRANSLATORS: name of a volume mixer control #, c-format msgid "%s Function" -msgstr "" +msgstr "Función %s" -#, fuzzy msgid "" "Could not open audio device for playback. This version of the Open Sound " "System is not supported by this element." msgstr "" -"Non foi posíbel abrir o dispositivo de son. Vostede non ten permisos para " -"abrir o dispositivo." +"Non foi posíbel abrir o dispositivo de son para a reprodución. Esta versión " +"do Open Sound System non está admitida por este elemento." msgid "Playback is not supported by this audio device." -msgstr "" +msgstr "Este dispositivo de son non admite a reprodución." msgid "Audio playback error." -msgstr "" +msgstr "Erro de reprodución de son." msgid "Recording is not supported by this audio device." -msgstr "" +msgstr "Este dispositivo de son non admite a gravación." -#, fuzzy msgid "Error recording from audio device." -msgstr "Produciuse un erro ao ler %d bytes desde o dispositivo «%s»." +msgstr "Erro ao gravar do dispositivo de son." msgid "Gain" msgstr "Ganancia" @@ -666,16 +641,16 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Produciuse un fallo ao estabelecer a entrada %d no dispositivo %s." -#, fuzzy, c-format +#, c-format msgid "" "Failed to get current output on device '%s'. May be it is a radio device" msgstr "" -"Produciuse un fallo ao obter a entrada actual no dispositivo «%s». Cicáis " -"sexa un dispositivo de radio." +"Produciuse un fallo ao obter a entrada actual no dispositivo «%s». Quizais " +"sexa un dispositivo de radio" -#, fuzzy, c-format +#, c-format msgid "Failed to set output %d on device %s." -msgstr "Produciuse un fallo ao estabelecer a entrada %d no dispositivo %s." +msgstr "Produciuse un fallo ao estabelecer a saída %d no dispositivo %s." #, c-format msgid "Could not enqueue buffers in device '%s'." diff --git a/po/pl.po b/po/pl.po index af5af264f8..2ba0bc1384 100644 --- a/po/pl.po +++ b/po/pl.po @@ -1,13 +1,13 @@ # Polish translation for gst-plugins-good. # This file is distributed under the same license as the gst-plugins-good package. -# Jakub Bogusz , 2007-2010. +# Jakub Bogusz , 2007-2011. # msgid "" msgstr "" -"Project-Id-Version: gst-plugins-good 0.10.25.3\n" +"Project-Id-Version: gst-plugins-good 0.10.26.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-01-07 01:15+0000\n" -"PO-Revision-Date: 2010-10-25 07:27+0200\n" +"POT-Creation-Date: 2011-01-11 19:32+0000\n" +"PO-Revision-Date: 2011-01-07 21:53+0100\n" "Last-Translator: Jakub Bogusz \n" "Language-Team: Polish \n" "Language: pl\n" @@ -70,7 +70,7 @@ msgid "This file is corrupt and cannot be played." msgstr "Ten plik jest uszkodzony i nie może być odtworzony." msgid "Invalid atom size." -msgstr "" +msgstr "Błędny rozmiar atomu." msgid "This file is incomplete and cannot be played." msgstr "Ten plik jest niekompletny i nie może być odtworzony." @@ -625,15 +625,15 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Nie udało się ustawić wejścia %d urządzenia %s." -#, fuzzy, c-format +#, c-format msgid "" "Failed to get current output on device '%s'. May be it is a radio device" msgstr "" -"Nie udało się uzyskać aktualnego wejścia urządzenia '%s'. Może to radio" +"Nie udało się uzyskać aktualnego wyjścia urządzenia '%s'. Może to radio" -#, fuzzy, c-format +#, c-format msgid "Failed to set output %d on device %s." -msgstr "Nie udało się ustawić wejścia %d urządzenia %s." +msgstr "Nie udało się ustawić wyjścia %d urządzenia %s." #, c-format msgid "Could not enqueue buffers in device '%s'." diff --git a/po/pt_BR.po b/po/pt_BR.po index 0c0af7d32f..6a669ff87c 100644 --- a/po/pt_BR.po +++ b/po/pt_BR.po @@ -1,7 +1,7 @@ # Brazilian Portuguese translation of gst-plugins-good. # This file is distributed under the same license as the gst-plugins-good package. -# Copyright (C) 2008-2010 Free Software Foundation, Inc. -# Fabrício Godoy , 2008-2010. +# Copyright (C) 2008-2011 Free Software Foundation, Inc. +# Fabrício Godoy , 2008-2011. # # data flow -> fluxo de dados # streaming -> fluxo contínuo @@ -9,10 +9,10 @@ # msgid "" msgstr "" -"Project-Id-Version: gst-plugins-good 0.10.23.2\n" +"Project-Id-Version: gst-plugins-good 0.10.26.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-01-07 01:15+0000\n" -"PO-Revision-Date: 2010-06-28 21:00-0300\n" +"POT-Creation-Date: 2011-01-11 19:32+0000\n" +"PO-Revision-Date: 2011-01-08 01:28-0300\n" "Last-Translator: Fabrício Godoy \n" "Language-Team: Brazilian Portuguese \n" "Language: pt_BR\n" @@ -41,30 +41,28 @@ msgstr "Falha ao decodificar a imagem JPEG" msgid "Could not connect to server" msgstr "Não foi possível conectar ao servidor" -#, fuzzy msgid "Server does not support seeking." -msgstr "Não há suporte a captura de vídeo no dispositivo \"%s\"" +msgstr "Nâo há suporte a busca pelo servidor." -#, fuzzy msgid "Could not resolve server name." -msgstr "Não foi possível conectar ao servidor" +msgstr "Não foi possível resolver o nome do servidor." -#, fuzzy msgid "Could not establish connection to server." -msgstr "Não foi possível estabelecer uma conexão com servidor de som" +msgstr "Não foi possível estabelecer uma conexão com servidor." msgid "Secure connection setup failed." -msgstr "" +msgstr "Configuração de conexão segura falhou." msgid "" "A network error occured, or the server closed the connection unexpectedly." msgstr "" +"Um erro de rede ocorreu, ou o servidor fechou a conexão inesperadamente." msgid "Server sent bad data." -msgstr "" +msgstr "O servidor enviou dados ruins." msgid "No URL set." -msgstr "" +msgstr "Nenhum URL definido." msgid "No or invalid input audio, AVI stream will be corrupt." msgstr "Entrada de áudio nula ou inválida, o fluxo AVI pode estar corrompido." @@ -79,7 +77,7 @@ msgid "This file is corrupt and cannot be played." msgstr "Este arquivo está corrompido e não pôde ser reproduzido." msgid "Invalid atom size." -msgstr "" +msgstr "Tamanho de Atom inválido." msgid "This file is incomplete and cannot be played." msgstr "Este arquivo está incompleto e não pôde ser reproduzido." @@ -637,16 +635,16 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Falha ao definir a entrada %d no dispositivo %s." -#, fuzzy, c-format +#, c-format msgid "" "Failed to get current output on device '%s'. May be it is a radio device" msgstr "" -"Falha ao obter a entrada atual no dispositivo \"%s\". Talvez seja um " +"Falha ao obter a saída atual no dispositivo \"%s\". Talvez seja um " "dispositivo de rádio" -#, fuzzy, c-format +#, c-format msgid "Failed to set output %d on device %s." -msgstr "Falha ao definir a entrada %d no dispositivo %s." +msgstr "Falha ao definir a saída %d no dispositivo %s." #, c-format msgid "Could not enqueue buffers in device '%s'." @@ -684,58 +682,3 @@ msgstr "Ainda não há suporte a mudança de resolução enquanto está executan msgid "Cannot operate without a clock" msgstr "Não é possível operar sem um temporizador" - -#~ msgid "Failed to enumerate possible video formats device '%s' can work with" -#~ msgstr "" -#~ "Falha ao especificar os formatos possíveis de vídeo que o dispositivo \"%s" -#~ "\" pode funcionar" - -#~ msgid "" -#~ "The buffer type is not supported, or the index is out of bounds, or no " -#~ "buffers have been allocated yet, or the userptr or length are invalid. " -#~ "device %s" -#~ msgstr "" -#~ "Não há suporte ao tipo de buffer, ou o índice está fora dos limites, ou " -#~ "nenhum buffer foi definido ainda, ou o userptr ou tamanho são inválidos. " -#~ "Dispositivo %s" - -#~ msgid "" -#~ "Failed trying to get video frames from device '%s'. Not enough memory." -#~ msgstr "" -#~ "Falha ao tentar obter os quadros de vídeo do dispositivo \"%s\". Memória " -#~ "insuficiente." - -#~ msgid "insufficient memory to enqueue a user pointer buffer. device %s." -#~ msgstr "" -#~ "memória insuficiente para adicionar à fila um buffer de ponteiro do " -#~ "usuário. Dispositivo %s." - -#~ msgid "No free buffers found in the pool at index %d." -#~ msgstr "Nenhum buffer livre encontrado no reservatório, no índice %d." - -#~ msgid "Device '%s' cannot capture at %dx%d" -#~ msgstr "O dispositivo \"%s\" não captura à %dx%d" - -#~ msgid "Device '%s' cannot capture in the specified format" -#~ msgstr "O dispositivo \"%s\" não captura no formato especificado" - -#~ msgid "Could not get buffers from device '%s'." -#~ msgstr "Não foi possível obter buffers do dispositivo \"%s\"." - -#~ msgid "Could not get enough buffers from device '%s'." -#~ msgstr "Não foi possível obter buffers suficientes do dispositivo \"%s\"." - -#~ msgid "Error starting streaming capture from device '%s'." -#~ msgstr "Erro ao iniciar um fluxo contínuo de captura do dispositivo \"%s\"." - -#~ msgid "Error stopping streaming capture from device '%s'." -#~ msgstr "Erro ao parar um fluxo contínuo de captura do dispositivo \"%s\"." - -#~ msgid "Failed getting controls attributes on device '%s.'" -#~ msgstr "Falha ao obter atributos de controle no dispositivo \"%s\"." - -#~ msgid "Could not read from CD." -#~ msgstr "Não foi possível ler o CD." - -#~ msgid "Disc is not an Audio CD." -#~ msgstr "O disco não é um CD de áudio." diff --git a/po/sl.po b/po/sl.po index 4472288878..cb587ace1d 100644 --- a/po/sl.po +++ b/po/sl.po @@ -2,15 +2,15 @@ # This file is distributed under the same license as the gst-plugins-good package. # Copyright (C) 2005 Free Software Foundation, Inc. # -# Matej Urbančič , 2010. # Robert Horvat , 2010. +# Matej Urbančič , 2010 - 2011. # msgid "" msgstr "" -"Project-Id-Version: gst-plugins-good 0.10.25.3\n" +"Project-Id-Version: gst-plugins-good 0.10.26.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-01-07 01:15+0000\n" -"PO-Revision-Date: 2010-10-25 12:23+0100\n" +"POT-Creation-Date: 2011-01-11 19:32+0000\n" +"PO-Revision-Date: 2011-01-07 20:28+0100\n" "Last-Translator: Matej Urbančič \n" "Language-Team: Slovenian \n" "Language: sl\n" @@ -64,7 +64,7 @@ msgid "Server sent bad data." msgstr "Strežnik je poslal slabe podatke." msgid "No URL set." -msgstr "Ni nastavljen naslov URL." +msgstr "Ni nastavljenega naslova URL." msgid "No or invalid input audio, AVI stream will be corrupt." msgstr "Napaka vhodnega zvoka, AVI pretok bo pokvarjen." @@ -79,7 +79,7 @@ msgid "This file is corrupt and cannot be played." msgstr "Datoteka je pokvarjena, zato je ni mogoče predvajati." msgid "Invalid atom size." -msgstr "" +msgstr "Neveljavna velikost atoma" msgid "This file is incomplete and cannot be played." msgstr "Datoteka ni popolna, zato je ni mogoče predvajati." @@ -89,7 +89,7 @@ msgstr "Video zapis v tej datoteki morda ne bo predvajan pravilno." #, c-format msgid "This file contains too many streams. Only playing first %d" -msgstr "Datoteka vsebuje več pretokov. Predvajanje le prvih %d" +msgstr "Datoteka vsebuje več pretokov. Predvajano bo le začetnih %d" msgid "" "No supported stream was found. You might need to install a GStreamer RTSP " @@ -634,16 +634,16 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Napaka med določanjem vhoda %d naprave %s." -#, fuzzy, c-format +#, c-format msgid "" "Failed to get current output on device '%s'. May be it is a radio device" msgstr "" -"Napaka med pridobivanjem vhodnega signala naprave '%s'. Morda je radijska " +"Napaka med pridobivanjem odvodnega signala naprave '%s'. Morda je radijska " "naprava." -#, fuzzy, c-format +#, c-format msgid "Failed to set output %d on device %s." -msgstr "Napaka med določanjem vhoda %d naprave %s." +msgstr "Napaka med določanjem odvoda %d naprave %s." #, c-format msgid "Could not enqueue buffers in device '%s'." @@ -679,7 +679,7 @@ msgid "Changing resolution at runtime is not yet supported." msgstr "Spreminjanje ločljivosti med delovanjem še ni podprto." msgid "Cannot operate without a clock" -msgstr "Ni mogoče izvajati opravil brez ure" +msgstr "Izvajanje opravil brez ure ni mogoče" #~ msgid "Describes the selected input element." #~ msgstr "Opisuje izbran vnosni predmet." diff --git a/po/sv.po b/po/sv.po index aaf8a30710..8f4509afd2 100644 --- a/po/sv.po +++ b/po/sv.po @@ -1,15 +1,15 @@ # Swedish messages for gst-plugins-good. -# Copyright (C) 2007, 2008, 2009, 2010 Free Software Foundation, Inc. +# Copyright (C) 2007-2011 Free Software Foundation, Inc. # This file is distributed under the same license as the gst-plugins-good package. # Christian Rose , 2004. -# Daniel Nylander , 2007, 2008, 2009, 2010. +# Daniel Nylander , 2007, 2008, 2009, 2010, 2011. # msgid "" msgstr "" -"Project-Id-Version: gst-plugins-good 0.10.25.3\n" +"Project-Id-Version: gst-plugins-good 0.10.26.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-01-07 01:15+0000\n" -"PO-Revision-Date: 2010-10-26 08:50+0100\n" +"POT-Creation-Date: 2011-01-11 19:32+0000\n" +"PO-Revision-Date: 2011-01-09 19:36+0100\n" "Last-Translator: Daniel Nylander \n" "Language-Team: Swedish \n" "Language: sv\n" @@ -71,9 +71,8 @@ msgstr "Den här filen är ogiltig och kan inte spelas upp." msgid "This file is corrupt and cannot be played." msgstr "Den här filen är skadad och kan inte spelas upp." -#, fuzzy msgid "Invalid atom size." -msgstr "Ogiltig cd-enhet" +msgstr "Ogiltig atomstorlek." msgid "This file is incomplete and cannot be played." msgstr "Den här filen är inte fullständig och kan inte spelas upp." @@ -634,16 +633,16 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Kunde inte ställa in ingång %d på enheten %s." -#, fuzzy, c-format +#, c-format msgid "" "Failed to get current output on device '%s'. May be it is a radio device" msgstr "" -"Misslyckades med att få tag på aktuell ingång på enheten \"%s\". Kan vara en " +"Misslyckades med att få tag på aktuell utgång på enheten \"%s\". Kan vara en " "radioenhet" -#, fuzzy, c-format +#, c-format msgid "Failed to set output %d on device %s." -msgstr "Kunde inte ställa in ingång %d på enheten %s." +msgstr "Misslyckades med att ställa in utgång %d på enheten %s." #, c-format msgid "Could not enqueue buffers in device '%s'." diff --git a/po/tr.po b/po/tr.po index d4b3230b5a..c0823489e2 100644 --- a/po/tr.po +++ b/po/tr.po @@ -1,12 +1,13 @@ -# translation of gst-plugins-good-0.10.25.3.po to Turkish +# translation of gst-plugins-good-0.10.26.2.po to Turkish # This file is put in the public domain. # Server Acim , 2010. +# Server Acim , 2011. msgid "" msgstr "" -"Project-Id-Version: gst-plugins-good 0.10.25.3\n" +"Project-Id-Version: gst-plugins-good 0.10.26.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-01-07 01:15+0000\n" -"PO-Revision-Date: 2010-11-16 16:41+0200\n" +"POT-Creation-Date: 2011-01-11 19:32+0000\n" +"PO-Revision-Date: 2011-01-08 00:03+0200\n" "Last-Translator: Server Acim \n" "Language-Team: Turkish \n" "Language: tr\n" @@ -71,7 +72,7 @@ msgid "This file is corrupt and cannot be played." msgstr "Bu dosya bozuk ve oynatılamaz." msgid "Invalid atom size." -msgstr "" +msgstr "Geçersiz atom boyutu" msgid "This file is incomplete and cannot be played." msgstr "Dosya eksik ve oynatılamaz." @@ -618,15 +619,15 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Giriş değeri olarak bu %d şu aygıtta %s ayarlanamadı." -#, fuzzy, c-format +#, c-format msgid "" "Failed to get current output on device '%s'. May be it is a radio device" msgstr "" "Şu aygıtta '%s' geçerli giriş elde edilemedi. O bir radyo aygıtı olabilir." -#, fuzzy, c-format +#, c-format msgid "Failed to set output %d on device %s." -msgstr "Giriş değeri olarak bu %d şu aygıtta %s ayarlanamadı." +msgstr "Çıkış değeri olarak %d şu aygıtta %s elde edilemedi." #, c-format msgid "Could not enqueue buffers in device '%s'." From 7280ea1f68fc43ed4850278b0d75c4ba0e61effe Mon Sep 17 00:00:00 2001 From: Arun Raghavan Date: Tue, 11 Jan 2011 23:39:12 +0530 Subject: [PATCH 234/351] pulsesink: Make corking during pause synchronous This makes the call to pa_stream_cork() during ringbuffer pause() synchronous, which makes sure that the clock does not advance after we take a snapshot for start_time. https://bugzilla.gnome.org/show_bug.cgi?id=639240 --- ext/pulse/pulsesink.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/ext/pulse/pulsesink.c b/ext/pulse/pulsesink.c index 796c7b7234..f885843c85 100644 --- a/ext/pulse/pulsesink.c +++ b/ext/pulse/pulsesink.c @@ -1092,7 +1092,7 @@ gst_pulseringbuffer_pause (GstRingBuffer * buf) GST_DEBUG_OBJECT (psink, "pausing and corking"); /* make sure the commit method stops writing */ pbuf->paused = TRUE; - res = gst_pulsering_set_corked (pbuf, TRUE, FALSE); + res = gst_pulsering_set_corked (pbuf, TRUE, TRUE); if (pbuf->in_commit) { /* we are waiting in a commit, signal */ GST_DEBUG_OBJECT (psink, "signal commit"); From a9852ac57f219f5fc111bffe8910c6b3b4003460 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Tue, 11 Jan 2011 22:42:42 +0000 Subject: [PATCH 235/351] docs: update docs --- docs/plugins/gst-plugins-good-plugins.args | 14 ++++++++++++-- .../gst-plugins-good-plugins.hierarchy | 6 ++---- .../gst-plugins-good-plugins.interfaces | 3 +-- .../gst-plugins-good-plugins.prerequisites | 1 - docs/plugins/inspect/plugin-1394.xml | 4 ++-- docs/plugins/inspect/plugin-aasink.xml | 4 ++-- docs/plugins/inspect/plugin-alaw.xml | 4 ++-- docs/plugins/inspect/plugin-alpha.xml | 4 ++-- docs/plugins/inspect/plugin-alphacolor.xml | 4 ++-- docs/plugins/inspect/plugin-annodex.xml | 4 ++-- docs/plugins/inspect/plugin-apetag.xml | 4 ++-- docs/plugins/inspect/plugin-audiofx.xml | 4 ++-- docs/plugins/inspect/plugin-auparse.xml | 4 ++-- docs/plugins/inspect/plugin-autodetect.xml | 4 ++-- docs/plugins/inspect/plugin-avi.xml | 4 ++-- docs/plugins/inspect/plugin-cacasink.xml | 4 ++-- docs/plugins/inspect/plugin-cairo.xml | 4 ++-- docs/plugins/inspect/plugin-cutter.xml | 4 ++-- docs/plugins/inspect/plugin-debug.xml | 4 ++-- docs/plugins/inspect/plugin-deinterlace.xml | 4 ++-- docs/plugins/inspect/plugin-dv.xml | 4 ++-- docs/plugins/inspect/plugin-efence.xml | 4 ++-- docs/plugins/inspect/plugin-effectv.xml | 4 ++-- docs/plugins/inspect/plugin-equalizer.xml | 4 ++-- docs/plugins/inspect/plugin-esdsink.xml | 4 ++-- docs/plugins/inspect/plugin-flac.xml | 4 ++-- docs/plugins/inspect/plugin-flv.xml | 4 ++-- docs/plugins/inspect/plugin-flxdec.xml | 4 ++-- docs/plugins/inspect/plugin-gconfelements.xml | 4 ++-- docs/plugins/inspect/plugin-gdkpixbuf.xml | 4 ++-- docs/plugins/inspect/plugin-goom.xml | 4 ++-- docs/plugins/inspect/plugin-goom2k1.xml | 4 ++-- docs/plugins/inspect/plugin-gstrtpmanager.xml | 4 ++-- docs/plugins/inspect/plugin-halelements.xml | 4 ++-- docs/plugins/inspect/plugin-icydemux.xml | 4 ++-- docs/plugins/inspect/plugin-id3demux.xml | 4 ++-- docs/plugins/inspect/plugin-imagefreeze.xml | 4 ++-- docs/plugins/inspect/plugin-interleave.xml | 4 ++-- docs/plugins/inspect/plugin-jack.xml | 16 ++++++++-------- docs/plugins/inspect/plugin-jpeg.xml | 4 ++-- docs/plugins/inspect/plugin-level.xml | 4 ++-- docs/plugins/inspect/plugin-matroska.xml | 4 ++-- docs/plugins/inspect/plugin-mulaw.xml | 4 ++-- docs/plugins/inspect/plugin-multifile.xml | 4 ++-- docs/plugins/inspect/plugin-multipart.xml | 4 ++-- .../plugins/inspect/plugin-navigationtest.xml | 4 ++-- docs/plugins/inspect/plugin-oss4.xml | 4 ++-- docs/plugins/inspect/plugin-ossaudio.xml | 4 ++-- docs/plugins/inspect/plugin-png.xml | 4 ++-- docs/plugins/inspect/plugin-pulseaudio.xml | 4 ++-- docs/plugins/inspect/plugin-quicktime.xml | 4 ++-- docs/plugins/inspect/plugin-replaygain.xml | 4 ++-- docs/plugins/inspect/plugin-rtp.xml | 4 ++-- docs/plugins/inspect/plugin-rtsp.xml | 4 ++-- docs/plugins/inspect/plugin-shapewipe.xml | 4 ++-- docs/plugins/inspect/plugin-shout2send.xml | 2 +- docs/plugins/inspect/plugin-smpte.xml | 4 ++-- docs/plugins/inspect/plugin-soup.xml | 4 ++-- docs/plugins/inspect/plugin-spectrum.xml | 4 ++-- docs/plugins/inspect/plugin-speex.xml | 4 ++-- docs/plugins/inspect/plugin-taglib.xml | 4 ++-- docs/plugins/inspect/plugin-udp.xml | 4 ++-- docs/plugins/inspect/plugin-video4linux2.xml | 19 ++----------------- docs/plugins/inspect/plugin-videobox.xml | 4 ++-- docs/plugins/inspect/plugin-videocrop.xml | 4 ++-- docs/plugins/inspect/plugin-videofilter.xml | 4 ++-- docs/plugins/inspect/plugin-videomixer.xml | 4 ++-- docs/plugins/inspect/plugin-wavenc.xml | 4 ++-- docs/plugins/inspect/plugin-wavpack.xml | 4 ++-- docs/plugins/inspect/plugin-wavparse.xml | 4 ++-- docs/plugins/inspect/plugin-ximagesrc.xml | 4 ++-- docs/plugins/inspect/plugin-y4menc.xml | 4 ++-- 72 files changed, 156 insertions(+), 165 deletions(-) diff --git a/docs/plugins/gst-plugins-good-plugins.args b/docs/plugins/gst-plugins-good-plugins.args index 7e277a5ca7..3553b914d4 100644 --- a/docs/plugins/gst-plugins-good-plugins.args +++ b/docs/plugins/gst-plugins-good-plugins.args @@ -765,7 +765,7 @@ rw UDP Buffer Size Size of the kernel UDP receive buffer in bytes, 0=default. -0 +524288 @@ -2528,6 +2528,16 @@ TRUE + +GstMultiUDPSink::buffer-size +gint +>= 0 +rw +Buffer Size +Size of the kernel send buffer in bytes, 0=default. +0 + + GstCmmlDec::wait-clip-end-time gboolean @@ -20355,7 +20365,7 @@ rw Method Deinterlace Method. -Motion Adaptive: Advanced Detection +Television: Full resolution diff --git a/docs/plugins/gst-plugins-good-plugins.hierarchy b/docs/plugins/gst-plugins-good-plugins.hierarchy index 5585c4a857..7bb718ceef 100644 --- a/docs/plugins/gst-plugins-good-plugins.hierarchy +++ b/docs/plugins/gst-plugins-good-plugins.hierarchy @@ -4,6 +4,8 @@ GObject GstCmmlTagHead GstCmmlTagStream GstColorBalanceChannel + GstMixerTrack + GstMixerOptions GstObject GstBus GstClock @@ -106,7 +108,6 @@ GObject GstTest GstVideoSink GstGdkPixbufSink - GstV4l2Sink GstBaseSrc GstPushSrc GstBaseAudioSrc @@ -223,7 +224,6 @@ GObject GstMatroskaDemux GstMatroskaMux GstWebMMux - GstMonoscope GstMuLawDec GstMuLawEnc GstMultipartDemux @@ -290,7 +290,6 @@ GInterface GstColorBalance GstImplementsInterface GstMixer - GstNavigation GstPreset GstPropertyProbe GstStreamVolume @@ -298,4 +297,3 @@ GInterface GstTuner GstURIHandler GstVideoOrientation - GstXOverlay diff --git a/docs/plugins/gst-plugins-good-plugins.interfaces b/docs/plugins/gst-plugins-good-plugins.interfaces index d66eca41ca..dc5c5915e6 100644 --- a/docs/plugins/gst-plugins-good-plugins.interfaces +++ b/docs/plugins/gst-plugins-good-plugins.interfaces @@ -19,7 +19,6 @@ GstRgVolume GstChildProxy GstAspectRatioCrop GstChildProxy GstPulseSink GstStreamVolume GstImplementsInterface GstPropertyProbe GstOss4Sink GstStreamVolume GstPropertyProbe -GstV4l2Sink GstImplementsInterface GstXOverlay GstNavigation GstColorBalance GstVideoOrientation GstPropertyProbe GstShout2send GstTagSetter GstUDPSink GstURIHandler GstDV1394Src GstURIHandler GstPropertyProbe @@ -28,7 +27,7 @@ GstSoupHTTPSrc GstURIHandler GstPulseSrc GstImplementsInterface GstMixer GstPropertyProbe GstOssSrc GstImplementsInterface GstMixer GstOss4Source GstImplementsInterface GstMixer GstPropertyProbe -GstV4l2Src GstURIHandler GstImplementsInterface GstTuner GstXOverlay GstColorBalance GstVideoOrientation GstPropertyProbe +GstV4l2Src GstURIHandler GstImplementsInterface GstTuner GstColorBalance GstVideoOrientation GstPropertyProbe GstUDPSrc GstURIHandler GstWavpackEnc GstPreset GstTagLibMux GstTagSetter diff --git a/docs/plugins/gst-plugins-good-plugins.prerequisites b/docs/plugins/gst-plugins-good-plugins.prerequisites index 1f92056772..fa0a1d51c6 100644 --- a/docs/plugins/gst-plugins-good-plugins.prerequisites +++ b/docs/plugins/gst-plugins-good-plugins.prerequisites @@ -4,7 +4,6 @@ GstStreamVolume GObject GstImplementsInterface GstElement GstMixer GstImplementsInterface GstElement GstTuner GstImplementsInterface GstElement -GstXOverlay GstImplementsInterface GstElement GstColorBalance GstImplementsInterface GstElement GstVideoOrientation GstImplementsInterface GstElement GIcon GObject diff --git a/docs/plugins/inspect/plugin-1394.xml b/docs/plugins/inspect/plugin-1394.xml index 971756f303..36459372b0 100644 --- a/docs/plugins/inspect/plugin-1394.xml +++ b/docs/plugins/inspect/plugin-1394.xml @@ -3,10 +3,10 @@ Source for video data via IEEE1394 interface ../../ext/raw1394/.libs/libgst1394.so libgst1394.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-aasink.xml b/docs/plugins/inspect/plugin-aasink.xml index bb4c9f8562..58881197e1 100644 --- a/docs/plugins/inspect/plugin-aasink.xml +++ b/docs/plugins/inspect/plugin-aasink.xml @@ -3,10 +3,10 @@ ASCII Art video sink ../../ext/aalib/.libs/libgstaasink.so libgstaasink.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-alaw.xml b/docs/plugins/inspect/plugin-alaw.xml index c1fd5fbb52..f2f3a63022 100644 --- a/docs/plugins/inspect/plugin-alaw.xml +++ b/docs/plugins/inspect/plugin-alaw.xml @@ -3,10 +3,10 @@ ALaw audio conversion routines ../../gst/law/.libs/libgstalaw.so libgstalaw.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-alpha.xml b/docs/plugins/inspect/plugin-alpha.xml index 57fbba62e2..e406b9e8c8 100644 --- a/docs/plugins/inspect/plugin-alpha.xml +++ b/docs/plugins/inspect/plugin-alpha.xml @@ -3,10 +3,10 @@ adds an alpha channel to video - constant or via chroma-keying ../../gst/alpha/.libs/libgstalpha.so libgstalpha.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-alphacolor.xml b/docs/plugins/inspect/plugin-alphacolor.xml index 73c2daf9d8..652430e630 100644 --- a/docs/plugins/inspect/plugin-alphacolor.xml +++ b/docs/plugins/inspect/plugin-alphacolor.xml @@ -3,10 +3,10 @@ RGBA from/to AYUV colorspace conversion preserving the alpha channel ../../gst/alpha/.libs/libgstalphacolor.so libgstalphacolor.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-annodex.xml b/docs/plugins/inspect/plugin-annodex.xml index 4b6a03d197..3cc312eb1e 100644 --- a/docs/plugins/inspect/plugin-annodex.xml +++ b/docs/plugins/inspect/plugin-annodex.xml @@ -3,10 +3,10 @@ annodex stream manipulation (info about annodex: http://www.annodex.net) ../../ext/annodex/.libs/libgstannodex.so libgstannodex.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-apetag.xml b/docs/plugins/inspect/plugin-apetag.xml index c6a1042120..2ea6f67ac4 100644 --- a/docs/plugins/inspect/plugin-apetag.xml +++ b/docs/plugins/inspect/plugin-apetag.xml @@ -3,10 +3,10 @@ APEv1/2 tag reader ../../gst/apetag/.libs/libgstapetag.so libgstapetag.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-audiofx.xml b/docs/plugins/inspect/plugin-audiofx.xml index b7e9c553d5..46904fdc6c 100644 --- a/docs/plugins/inspect/plugin-audiofx.xml +++ b/docs/plugins/inspect/plugin-audiofx.xml @@ -3,10 +3,10 @@ Audio effects plugin ../../gst/audiofx/.libs/libgstaudiofx.so libgstaudiofx.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-auparse.xml b/docs/plugins/inspect/plugin-auparse.xml index 53141b1b76..374a368aaf 100644 --- a/docs/plugins/inspect/plugin-auparse.xml +++ b/docs/plugins/inspect/plugin-auparse.xml @@ -3,10 +3,10 @@ parses au streams ../../gst/auparse/.libs/libgstauparse.so libgstauparse.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-autodetect.xml b/docs/plugins/inspect/plugin-autodetect.xml index fb0d15f61f..9b2ba39a4e 100644 --- a/docs/plugins/inspect/plugin-autodetect.xml +++ b/docs/plugins/inspect/plugin-autodetect.xml @@ -3,10 +3,10 @@ Plugin contains auto-detection plugins for video/audio in- and outputs ../../gst/autodetect/.libs/libgstautodetect.so libgstautodetect.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-avi.xml b/docs/plugins/inspect/plugin-avi.xml index 3baa80cf69..e3287a14ca 100644 --- a/docs/plugins/inspect/plugin-avi.xml +++ b/docs/plugins/inspect/plugin-avi.xml @@ -3,10 +3,10 @@ AVI stream handling ../../gst/avi/.libs/libgstavi.so libgstavi.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-cacasink.xml b/docs/plugins/inspect/plugin-cacasink.xml index 7efbee0825..40b8216164 100644 --- a/docs/plugins/inspect/plugin-cacasink.xml +++ b/docs/plugins/inspect/plugin-cacasink.xml @@ -3,10 +3,10 @@ Colored ASCII Art video sink ../../ext/libcaca/.libs/libgstcacasink.so libgstcacasink.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-cairo.xml b/docs/plugins/inspect/plugin-cairo.xml index 9e4291190c..b5b1758efa 100644 --- a/docs/plugins/inspect/plugin-cairo.xml +++ b/docs/plugins/inspect/plugin-cairo.xml @@ -3,10 +3,10 @@ Cairo-based elements ../../ext/cairo/.libs/libgstcairo.so libgstcairo.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-cutter.xml b/docs/plugins/inspect/plugin-cutter.xml index defe275701..e004a021d8 100644 --- a/docs/plugins/inspect/plugin-cutter.xml +++ b/docs/plugins/inspect/plugin-cutter.xml @@ -3,10 +3,10 @@ Audio Cutter to split audio into non-silent bits ../../gst/cutter/.libs/libgstcutter.so libgstcutter.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-debug.xml b/docs/plugins/inspect/plugin-debug.xml index f1f4127b33..da8d22b9f0 100644 --- a/docs/plugins/inspect/plugin-debug.xml +++ b/docs/plugins/inspect/plugin-debug.xml @@ -3,10 +3,10 @@ elements for testing and debugging ../../gst/debugutils/.libs/libgstdebug.so libgstdebug.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-deinterlace.xml b/docs/plugins/inspect/plugin-deinterlace.xml index 1a0c5f2dc8..6414aa706b 100644 --- a/docs/plugins/inspect/plugin-deinterlace.xml +++ b/docs/plugins/inspect/plugin-deinterlace.xml @@ -3,10 +3,10 @@ Deinterlacer ../../gst/deinterlace/.libs/libgstdeinterlace.so libgstdeinterlace.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-dv.xml b/docs/plugins/inspect/plugin-dv.xml index 0cee0f907c..a9e7a7c119 100644 --- a/docs/plugins/inspect/plugin-dv.xml +++ b/docs/plugins/inspect/plugin-dv.xml @@ -3,10 +3,10 @@ DV demuxer and decoder based on libdv (libdv.sf.net) ../../ext/dv/.libs/libgstdv.so libgstdv.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-efence.xml b/docs/plugins/inspect/plugin-efence.xml index 64d81053cd..af1edacf26 100644 --- a/docs/plugins/inspect/plugin-efence.xml +++ b/docs/plugins/inspect/plugin-efence.xml @@ -3,10 +3,10 @@ This element converts a stream of normal GStreamer buffers into a stream of buffers that are allocated in such a way that out-of-bounds access to data in the buffer is more likely to cause segmentation faults. This allocation method is very similar to the debugging tool "Electric Fence". ../../gst/debugutils/.libs/libgstefence.so libgstefence.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-effectv.xml b/docs/plugins/inspect/plugin-effectv.xml index bcc608be43..21b4b3ebf7 100644 --- a/docs/plugins/inspect/plugin-effectv.xml +++ b/docs/plugins/inspect/plugin-effectv.xml @@ -3,10 +3,10 @@ effect plugins from the effectv project ../../gst/effectv/.libs/libgsteffectv.so libgsteffectv.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-equalizer.xml b/docs/plugins/inspect/plugin-equalizer.xml index 5cda20e0f3..6cd90a49d3 100644 --- a/docs/plugins/inspect/plugin-equalizer.xml +++ b/docs/plugins/inspect/plugin-equalizer.xml @@ -3,10 +3,10 @@ GStreamer audio equalizers ../../gst/equalizer/.libs/libgstequalizer.so libgstequalizer.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-esdsink.xml b/docs/plugins/inspect/plugin-esdsink.xml index 2c089284ad..cdea83b642 100644 --- a/docs/plugins/inspect/plugin-esdsink.xml +++ b/docs/plugins/inspect/plugin-esdsink.xml @@ -3,10 +3,10 @@ ESD Element Plugins ../../ext/esd/.libs/libgstesd.so libgstesd.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-flac.xml b/docs/plugins/inspect/plugin-flac.xml index d09a9974b3..32d6614df9 100644 --- a/docs/plugins/inspect/plugin-flac.xml +++ b/docs/plugins/inspect/plugin-flac.xml @@ -3,10 +3,10 @@ The FLAC Lossless compressor Codec ../../ext/flac/.libs/libgstflac.so libgstflac.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-flv.xml b/docs/plugins/inspect/plugin-flv.xml index 0d5547fe9b..f01c04c410 100644 --- a/docs/plugins/inspect/plugin-flv.xml +++ b/docs/plugins/inspect/plugin-flv.xml @@ -3,10 +3,10 @@ FLV muxing and demuxing plugin ../../gst/flv/.libs/libgstflv.so libgstflv.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-flxdec.xml b/docs/plugins/inspect/plugin-flxdec.xml index ec51525cfb..e0c36f65d1 100644 --- a/docs/plugins/inspect/plugin-flxdec.xml +++ b/docs/plugins/inspect/plugin-flxdec.xml @@ -3,10 +3,10 @@ FLC/FLI/FLX video decoder ../../gst/flx/.libs/libgstflxdec.so libgstflxdec.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-gconfelements.xml b/docs/plugins/inspect/plugin-gconfelements.xml index e90f8e1b7b..8367254b41 100644 --- a/docs/plugins/inspect/plugin-gconfelements.xml +++ b/docs/plugins/inspect/plugin-gconfelements.xml @@ -3,10 +3,10 @@ elements wrapping the GStreamer/GConf audio/video output settings ../../ext/gconf/.libs/libgstgconfelements.so libgstgconfelements.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-gdkpixbuf.xml b/docs/plugins/inspect/plugin-gdkpixbuf.xml index cfe5c2b643..87627d80f1 100644 --- a/docs/plugins/inspect/plugin-gdkpixbuf.xml +++ b/docs/plugins/inspect/plugin-gdkpixbuf.xml @@ -3,10 +3,10 @@ GdkPixbuf-based image decoder, scaler and sink ../../ext/gdk_pixbuf/.libs/libgstgdkpixbuf.so libgstgdkpixbuf.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-goom.xml b/docs/plugins/inspect/plugin-goom.xml index 9c1a9e5097..e6f39049b7 100644 --- a/docs/plugins/inspect/plugin-goom.xml +++ b/docs/plugins/inspect/plugin-goom.xml @@ -3,10 +3,10 @@ GOOM visualization filter ../../gst/goom/.libs/libgstgoom.so libgstgoom.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-goom2k1.xml b/docs/plugins/inspect/plugin-goom2k1.xml index d4b090ce62..fa0cc637ad 100644 --- a/docs/plugins/inspect/plugin-goom2k1.xml +++ b/docs/plugins/inspect/plugin-goom2k1.xml @@ -3,10 +3,10 @@ GOOM 2k1 visualization filter ../../gst/goom2k1/.libs/libgstgoom2k1.so libgstgoom2k1.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-gstrtpmanager.xml b/docs/plugins/inspect/plugin-gstrtpmanager.xml index 90932beeac..769a97c34e 100644 --- a/docs/plugins/inspect/plugin-gstrtpmanager.xml +++ b/docs/plugins/inspect/plugin-gstrtpmanager.xml @@ -3,10 +3,10 @@ RTP session management plugin library ../../gst/rtpmanager/.libs/libgstrtpmanager.so libgstrtpmanager.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-halelements.xml b/docs/plugins/inspect/plugin-halelements.xml index 0a44fc15f4..61bb87cfff 100644 --- a/docs/plugins/inspect/plugin-halelements.xml +++ b/docs/plugins/inspect/plugin-halelements.xml @@ -3,10 +3,10 @@ elements wrapping the GStreamer/HAL audio input/output devices ../../ext/hal/.libs/libgsthalelements.so libgsthalelements.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-icydemux.xml b/docs/plugins/inspect/plugin-icydemux.xml index 2c14795f34..8ac3061619 100644 --- a/docs/plugins/inspect/plugin-icydemux.xml +++ b/docs/plugins/inspect/plugin-icydemux.xml @@ -3,10 +3,10 @@ Demux ICY tags from a stream ../../gst/icydemux/.libs/libgsticydemux.so libgsticydemux.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-id3demux.xml b/docs/plugins/inspect/plugin-id3demux.xml index 6abe75fcf4..82b2a560fb 100644 --- a/docs/plugins/inspect/plugin-id3demux.xml +++ b/docs/plugins/inspect/plugin-id3demux.xml @@ -3,10 +3,10 @@ Demux ID3v1 and ID3v2 tags from a file ../../gst/id3demux/.libs/libgstid3demux.so libgstid3demux.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-imagefreeze.xml b/docs/plugins/inspect/plugin-imagefreeze.xml index 26d92271af..90b764513b 100644 --- a/docs/plugins/inspect/plugin-imagefreeze.xml +++ b/docs/plugins/inspect/plugin-imagefreeze.xml @@ -3,10 +3,10 @@ Still frame stream generator ../../gst/imagefreeze/.libs/libgstimagefreeze.so libgstimagefreeze.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-interleave.xml b/docs/plugins/inspect/plugin-interleave.xml index 7c99bd0dc1..e60911ddfd 100644 --- a/docs/plugins/inspect/plugin-interleave.xml +++ b/docs/plugins/inspect/plugin-interleave.xml @@ -3,10 +3,10 @@ Audio interleaver/deinterleaver ../../gst/interleave/.libs/libgstinterleave.so libgstinterleave.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-jack.xml b/docs/plugins/inspect/plugin-jack.xml index dcead7b27c..c8cc852806 100644 --- a/docs/plugins/inspect/plugin-jack.xml +++ b/docs/plugins/inspect/plugin-jack.xml @@ -1,26 +1,26 @@ jack - Jack elements + JACK audio elements ../../ext/jack/.libs/libgstjack.so libgstjack.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin jackaudiosink Audio Sink (Jack) Sink/Audio - Output to Jack - Wim Taymans <wim@fluendo.com> + Output audio to a JACK server + Wim Taymans <wim.taymans@gmail.com> sink sink always -
audio/x-raw-float, endianness=(int){ 1234 }, width=(int)32, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]
+
audio/x-raw-float, endianness=(int)1234, width=(int)32, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]
 @@ -28,14 +28,14 @@ jackaudiosrc Audio Source (Jack) Source/Audio - Input from Jack + Captures audio from a JACK server Tristan Matthews <tristan@sat.qc.ca> src source always -
audio/x-raw-float, endianness=(int){ 1234 }, width=(int)32, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]
+
audio/x-raw-float, endianness=(int)1234, width=(int)32, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]
 diff --git a/docs/plugins/inspect/plugin-jpeg.xml b/docs/plugins/inspect/plugin-jpeg.xml index 88ac09b67b..fa8d04c3b2 100644 --- a/docs/plugins/inspect/plugin-jpeg.xml +++ b/docs/plugins/inspect/plugin-jpeg.xml @@ -3,10 +3,10 @@ JPeg plugin library ../../ext/jpeg/.libs/libgstjpeg.so libgstjpeg.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-level.xml b/docs/plugins/inspect/plugin-level.xml index a45421ebcc..f2f62c1c88 100644 --- a/docs/plugins/inspect/plugin-level.xml +++ b/docs/plugins/inspect/plugin-level.xml @@ -3,10 +3,10 @@ Audio level plugin ../../gst/level/.libs/libgstlevel.so libgstlevel.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-matroska.xml b/docs/plugins/inspect/plugin-matroska.xml index 09397b88fb..1a1c3b684b 100644 --- a/docs/plugins/inspect/plugin-matroska.xml +++ b/docs/plugins/inspect/plugin-matroska.xml @@ -3,10 +3,10 @@ Matroska and WebM stream handling ../../gst/matroska/.libs/libgstmatroska.so libgstmatroska.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-mulaw.xml b/docs/plugins/inspect/plugin-mulaw.xml index e7b2ccb0b5..a76367cae0 100644 --- a/docs/plugins/inspect/plugin-mulaw.xml +++ b/docs/plugins/inspect/plugin-mulaw.xml @@ -3,10 +3,10 @@ MuLaw audio conversion routines ../../gst/law/.libs/libgstmulaw.so libgstmulaw.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-multifile.xml b/docs/plugins/inspect/plugin-multifile.xml index dc9da5e01d..ee32044194 100644 --- a/docs/plugins/inspect/plugin-multifile.xml +++ b/docs/plugins/inspect/plugin-multifile.xml @@ -3,10 +3,10 @@ Reads/Writes buffers from/to sequentially named files ../../gst/multifile/.libs/libgstmultifile.so libgstmultifile.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-multipart.xml b/docs/plugins/inspect/plugin-multipart.xml index d3dcea5bed..f18f90c6ec 100644 --- a/docs/plugins/inspect/plugin-multipart.xml +++ b/docs/plugins/inspect/plugin-multipart.xml @@ -3,10 +3,10 @@ multipart stream manipulation ../../gst/multipart/.libs/libgstmultipart.so libgstmultipart.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-navigationtest.xml b/docs/plugins/inspect/plugin-navigationtest.xml index 2c40feca8a..a4acdd8ff9 100644 --- a/docs/plugins/inspect/plugin-navigationtest.xml +++ b/docs/plugins/inspect/plugin-navigationtest.xml @@ -3,10 +3,10 @@ Template for a video filter ../../gst/debugutils/.libs/libgstnavigationtest.so libgstnavigationtest.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-oss4.xml b/docs/plugins/inspect/plugin-oss4.xml index ed11ad234c..69fc251634 100644 --- a/docs/plugins/inspect/plugin-oss4.xml +++ b/docs/plugins/inspect/plugin-oss4.xml @@ -3,10 +3,10 @@ Open Sound System (OSS) version 4 support for GStreamer ../../sys/oss4/.libs/libgstoss4audio.so libgstoss4audio.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-ossaudio.xml b/docs/plugins/inspect/plugin-ossaudio.xml index 5ce429813c..51483d1ee6 100644 --- a/docs/plugins/inspect/plugin-ossaudio.xml +++ b/docs/plugins/inspect/plugin-ossaudio.xml @@ -3,10 +3,10 @@ OSS (Open Sound System) support for GStreamer ../../sys/oss/.libs/libgstossaudio.so libgstossaudio.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-png.xml b/docs/plugins/inspect/plugin-png.xml index 1e98203d73..ee491d8829 100644 --- a/docs/plugins/inspect/plugin-png.xml +++ b/docs/plugins/inspect/plugin-png.xml @@ -3,10 +3,10 @@ PNG plugin library ../../ext/libpng/.libs/libgstpng.so libgstpng.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-pulseaudio.xml b/docs/plugins/inspect/plugin-pulseaudio.xml index a453c9dc22..0c4250fcdb 100644 --- a/docs/plugins/inspect/plugin-pulseaudio.xml +++ b/docs/plugins/inspect/plugin-pulseaudio.xml @@ -3,10 +3,10 @@ PulseAudio plugin library ../../ext/pulse/.libs/libgstpulse.so libgstpulse.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-quicktime.xml b/docs/plugins/inspect/plugin-quicktime.xml index 9f7889f850..018278c500 100644 --- a/docs/plugins/inspect/plugin-quicktime.xml +++ b/docs/plugins/inspect/plugin-quicktime.xml @@ -3,10 +3,10 @@ Quicktime support ../../gst/qtdemux/.libs/libgstqtdemux.so libgstqtdemux.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-replaygain.xml b/docs/plugins/inspect/plugin-replaygain.xml index 6749df4393..fd75b24cfe 100644 --- a/docs/plugins/inspect/plugin-replaygain.xml +++ b/docs/plugins/inspect/plugin-replaygain.xml @@ -3,10 +3,10 @@ ReplayGain volume normalization ../../gst/replaygain/.libs/libgstreplaygain.so libgstreplaygain.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-rtp.xml b/docs/plugins/inspect/plugin-rtp.xml index a471bd9ad3..1a5a20f56e 100644 --- a/docs/plugins/inspect/plugin-rtp.xml +++ b/docs/plugins/inspect/plugin-rtp.xml @@ -3,10 +3,10 @@ Real-time protocol plugins ../../gst/rtp/.libs/libgstrtp.so libgstrtp.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-rtsp.xml b/docs/plugins/inspect/plugin-rtsp.xml index 621dc7a07b..83c4e466e1 100644 --- a/docs/plugins/inspect/plugin-rtsp.xml +++ b/docs/plugins/inspect/plugin-rtsp.xml @@ -3,10 +3,10 @@ transfer data via RTSP ../../gst/rtsp/.libs/libgstrtsp.so libgstrtsp.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-shapewipe.xml b/docs/plugins/inspect/plugin-shapewipe.xml index e8ea7b2abe..a322f8d69c 100644 --- a/docs/plugins/inspect/plugin-shapewipe.xml +++ b/docs/plugins/inspect/plugin-shapewipe.xml @@ -3,10 +3,10 @@ Shape Wipe transition filter ../../gst/shapewipe/.libs/libgstshapewipe.so libgstshapewipe.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-shout2send.xml b/docs/plugins/inspect/plugin-shout2send.xml index b2c96942a5..3b483c4b62 100644 --- a/docs/plugins/inspect/plugin-shout2send.xml +++ b/docs/plugins/inspect/plugin-shout2send.xml @@ -3,7 +3,7 @@ Sends data to an icecast server using libshout2 ../../ext/shout2/.libs/libgstshout2.so libgstshout2.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good libshout2 diff --git a/docs/plugins/inspect/plugin-smpte.xml b/docs/plugins/inspect/plugin-smpte.xml index f7c22d45ac..a2e30a118b 100644 --- a/docs/plugins/inspect/plugin-smpte.xml +++ b/docs/plugins/inspect/plugin-smpte.xml @@ -3,10 +3,10 @@ Apply the standard SMPTE transitions on video images ../../gst/smpte/.libs/libgstsmpte.so libgstsmpte.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-soup.xml b/docs/plugins/inspect/plugin-soup.xml index a9a826ce13..772fb6ec2d 100644 --- a/docs/plugins/inspect/plugin-soup.xml +++ b/docs/plugins/inspect/plugin-soup.xml @@ -3,10 +3,10 @@ libsoup HTTP client src ../../ext/soup/.libs/libgstsouphttpsrc.so libgstsouphttpsrc.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-spectrum.xml b/docs/plugins/inspect/plugin-spectrum.xml index 7310f1226e..9034dbd7a9 100644 --- a/docs/plugins/inspect/plugin-spectrum.xml +++ b/docs/plugins/inspect/plugin-spectrum.xml @@ -3,10 +3,10 @@ Run an FFT on the audio signal, output spectrum data ../../gst/spectrum/.libs/libgstspectrum.so libgstspectrum.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-speex.xml b/docs/plugins/inspect/plugin-speex.xml index c45f6dba50..50dcc6d699 100644 --- a/docs/plugins/inspect/plugin-speex.xml +++ b/docs/plugins/inspect/plugin-speex.xml @@ -3,10 +3,10 @@ Speex plugin library ../../ext/speex/.libs/libgstspeex.so libgstspeex.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-taglib.xml b/docs/plugins/inspect/plugin-taglib.xml index 5e6eed40c3..478a066ffc 100644 --- a/docs/plugins/inspect/plugin-taglib.xml +++ b/docs/plugins/inspect/plugin-taglib.xml @@ -3,10 +3,10 @@ Tag writing plug-in based on taglib ../../ext/taglib/.libs/libgsttaglib.so libgsttaglib.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-udp.xml b/docs/plugins/inspect/plugin-udp.xml index 768d1b9249..734322c32a 100644 --- a/docs/plugins/inspect/plugin-udp.xml +++ b/docs/plugins/inspect/plugin-udp.xml @@ -3,10 +3,10 @@ transfer data via UDP ../../gst/udp/.libs/libgstudp.so libgstudp.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-video4linux2.xml b/docs/plugins/inspect/plugin-video4linux2.xml index 598ae23c17..11cdf58804 100644 --- a/docs/plugins/inspect/plugin-video4linux2.xml +++ b/docs/plugins/inspect/plugin-video4linux2.xml @@ -3,27 +3,12 @@ elements for Video 4 Linux ../../sys/v4l2/.libs/libgstvideo4linux2.so libgstvideo4linux2.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin - - v4l2sink - Video (video4linux2) Sink - Sink/Video - Displays frames on a video4linux2 device - Rob Clark <rob@ti.com>, - - - sink - sink - always -
video/x-raw-rgb, bpp=(int)8, depth=(int)8, red_mask=(int)224, green_mask=(int)28, blue_mask=(int)3, endianness=(int)1234, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-rgb, bpp=(int)16, depth=(int)15, red_mask=(int)31744, green_mask=(int)992, blue_mask=(int)31, endianness=(int)1234, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-rgb, bpp=(int)16, depth=(int)16, red_mask=(int)63488, green_mask=(int)2016, blue_mask=(int)31, endianness=(int)1234, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-rgb, bpp=(int)16, depth=(int)15, red_mask=(int)31744, green_mask=(int)992, blue_mask=(int)31, endianness=(int)4321, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-rgb, bpp=(int)16, depth=(int)16, red_mask=(int)63488, green_mask=(int)2016, blue_mask=(int)31, endianness=(int)4321, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-rgb, bpp=(int)24, depth=(int)24, red_mask=(int)255, green_mask=(int)65280, blue_mask=(int)16711680, endianness=(int)4321, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-rgb, bpp=(int)24, depth=(int)24, red_mask=(int)16711680, green_mask=(int)65280, blue_mask=(int)255, endianness=(int)4321, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-rgb, bpp=(int)32, depth=(int)32, red_mask=(int)255, green_mask=(int)65280, blue_mask=(int)16711680, endianness=(int)4321, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-rgb, bpp=(int)32, depth=(int)32, red_mask=(int)-16777216, green_mask=(int)16711680, blue_mask=(int)65280, endianness=(int)4321, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-gray, bpp=(int)8, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)YVU9, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)YV12, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)YUY2, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)UYVY, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)Y42B, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)Y41B, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)Y41P, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)NV12, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)NV21, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)YUV9, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)I420, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-bayer, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; image/jpeg, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; image/jpeg, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-dv, systemstream=(boolean)true, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/mpegts; video/x-sonix, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-pwc1, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-pwc2, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)YVYU, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]
- - - v4l2src Video (video4linux2) Source diff --git a/docs/plugins/inspect/plugin-videobox.xml b/docs/plugins/inspect/plugin-videobox.xml index 13c9bedd8f..9b870c1d61 100644 --- a/docs/plugins/inspect/plugin-videobox.xml +++ b/docs/plugins/inspect/plugin-videobox.xml @@ -3,10 +3,10 @@ resizes a video by adding borders or cropping ../../gst/videobox/.libs/libgstvideobox.so libgstvideobox.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-videocrop.xml b/docs/plugins/inspect/plugin-videocrop.xml index f87c2989e7..1fff2d7440 100644 --- a/docs/plugins/inspect/plugin-videocrop.xml +++ b/docs/plugins/inspect/plugin-videocrop.xml @@ -3,10 +3,10 @@ Crops video into a user-defined region ../../gst/videocrop/.libs/libgstvideocrop.so libgstvideocrop.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-videofilter.xml b/docs/plugins/inspect/plugin-videofilter.xml index fcd028ca47..d098c80c23 100644 --- a/docs/plugins/inspect/plugin-videofilter.xml +++ b/docs/plugins/inspect/plugin-videofilter.xml @@ -3,10 +3,10 @@ Video filters plugin ../../gst/videofilter/.libs/libgstvideofilter.so libgstvideofilter.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-videomixer.xml b/docs/plugins/inspect/plugin-videomixer.xml index c1b460d3b0..34a97e5d8d 100644 --- a/docs/plugins/inspect/plugin-videomixer.xml +++ b/docs/plugins/inspect/plugin-videomixer.xml @@ -3,10 +3,10 @@ Video mixer ../../gst/videomixer/.libs/libgstvideomixer.so libgstvideomixer.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-wavenc.xml b/docs/plugins/inspect/plugin-wavenc.xml index bb3c6125ec..01011207b1 100644 --- a/docs/plugins/inspect/plugin-wavenc.xml +++ b/docs/plugins/inspect/plugin-wavenc.xml @@ -3,10 +3,10 @@ Encode raw audio into WAV ../../gst/wavenc/.libs/libgstwavenc.so libgstwavenc.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-wavpack.xml b/docs/plugins/inspect/plugin-wavpack.xml index 987e6b96b7..4c2a092959 100644 --- a/docs/plugins/inspect/plugin-wavpack.xml +++ b/docs/plugins/inspect/plugin-wavpack.xml @@ -3,10 +3,10 @@ Wavpack lossless/lossy audio format handling ../../ext/wavpack/.libs/libgstwavpack.so libgstwavpack.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-wavparse.xml b/docs/plugins/inspect/plugin-wavparse.xml index b05e4eb86d..c9e62769b1 100644 --- a/docs/plugins/inspect/plugin-wavparse.xml +++ b/docs/plugins/inspect/plugin-wavparse.xml @@ -3,10 +3,10 @@ Parse a .wav file into raw audio ../../gst/wavparse/.libs/libgstwavparse.so libgstwavparse.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-ximagesrc.xml b/docs/plugins/inspect/plugin-ximagesrc.xml index 8173a1b22a..5490f007d3 100644 --- a/docs/plugins/inspect/plugin-ximagesrc.xml +++ b/docs/plugins/inspect/plugin-ximagesrc.xml @@ -3,10 +3,10 @@ X11 video input plugin using standard Xlib calls ../../sys/ximage/.libs/libgstximagesrc.so libgstximagesrc.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-y4menc.xml b/docs/plugins/inspect/plugin-y4menc.xml index f48e39cd76..08063283e6 100644 --- a/docs/plugins/inspect/plugin-y4menc.xml +++ b/docs/plugins/inspect/plugin-y4menc.xml @@ -3,10 +3,10 @@ Encodes a YUV frame into the yuv4mpeg format (mjpegtools) ../../gst/y4m/.libs/libgsty4menc.so libgsty4menc.so - 0.10.26.1 + 0.10.26.3 LGPL gst-plugins-good - GStreamer Good Plug-ins git + GStreamer Good Plug-ins prerelease Unknown package origin From 0520052540f3c736e8e6694cb26a392fa2d260f4 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Tue, 11 Jan 2011 23:18:59 +0000 Subject: [PATCH 236/351] 0.10.26.3 pre-release --- configure.ac | 6 +++--- win32/common/config.h | 8 ++++---- 2 files changed, 7 insertions(+), 7 deletions(-) diff --git a/configure.ac b/configure.ac index 5ee8e87b37..5e5dbd0a18 100644 --- a/configure.ac +++ b/configure.ac @@ -5,7 +5,7 @@ dnl please read gstreamer/docs/random/autotools before changing this file dnl initialize autoconf dnl releases only do -Wall, git and prerelease does -Werror too dnl use a three digit version number for releases, and four for git/pre -AC_INIT(GStreamer Good Plug-ins, 0.10.26.2, +AC_INIT(GStreamer Good Plug-ins, 0.10.26.3, http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer, gst-plugins-good) @@ -52,8 +52,8 @@ AC_LIBTOOL_WIN32_DLL AM_PROG_LIBTOOL dnl *** required versions of GStreamer stuff *** -GST_REQ=0.10.31.2 -GSTPB_REQ=0.10.31.2 +GST_REQ=0.10.31.3 +GSTPB_REQ=0.10.31.3 dnl *** autotools stuff **** diff --git a/win32/common/config.h b/win32/common/config.h index da86c35383..e45f46beb1 100644 --- a/win32/common/config.h +++ b/win32/common/config.h @@ -49,7 +49,7 @@ #define GST_PACKAGE_ORIGIN "Unknown package origin" /* GStreamer package release date/time for plugins as YYYY-MM-DD */ -#define GST_PACKAGE_RELEASE_DATETIME "2011-01-07T01:34Z" +#define GST_PACKAGE_RELEASE_DATETIME "2011-01-11T20:04Z" /* struct v4l2_buffer missing */ #undef GST_V4L2_MISSING_BUFDECL @@ -386,7 +386,7 @@ #define PACKAGE_NAME "GStreamer Good Plug-ins" /* Define to the full name and version of this package. */ -#define PACKAGE_STRING "GStreamer Good Plug-ins 0.10.26.2" +#define PACKAGE_STRING "GStreamer Good Plug-ins 0.10.26.3" /* Define to the one symbol short name of this package. */ #define PACKAGE_TARNAME "gst-plugins-good" @@ -395,7 +395,7 @@ #undef PACKAGE_URL /* Define to the version of this package. */ -#define PACKAGE_VERSION "0.10.26.2" +#define PACKAGE_VERSION "0.10.26.3" /* directory where plugins are located */ #ifdef _DEBUG @@ -426,7 +426,7 @@ #undef STDC_HEADERS /* Version number of package */ -#define VERSION "0.10.26.2" +#define VERSION "0.10.26.3" /* old wavpack API */ #undef WAVPACK_OLD_API From 9dcb60822f2befa8d5eeee7d927681a7406427fc Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Wed, 12 Jan 2011 20:38:59 +0000 Subject: [PATCH 237/351] v4l2: define V4L2_FIELD_INTERLACED_{TB,BT} if not available in header Older kernels don't have these, and there's no easy way to check for the existance of enums that doesn't involve a configure check, so just define these if the V4L2_CAP_VIDEO_OUTPUT_OVERLAY define is not there, which was added in the same commit as the TB/BT enum. Fixes compilation on CentOS 5. https://bugzilla.gnome.org/show_bug.cgi?id=639339 --- sys/v4l2/gstv4l2bufferpool.c | 8 ++++++++ sys/v4l2/gstv4l2object.c | 7 +++++++ 2 files changed, 15 insertions(+) diff --git a/sys/v4l2/gstv4l2bufferpool.c b/sys/v4l2/gstv4l2bufferpool.c index 505689a66e..e976ec5079 100644 --- a/sys/v4l2/gstv4l2bufferpool.c +++ b/sys/v4l2/gstv4l2bufferpool.c @@ -40,6 +40,14 @@ #include "v4l2_calls.h" #include "gst/gst-i18n-plugin.h" +/* videodev2.h is not versioned and we can't easily check for the presence + * of enum values at compile time, but the V4L2_CAP_VIDEO_OUTPUT_OVERLAY define + * was added in the same commit as V4L2_FIELD_INTERLACED_{TB,BT} (b2787845) */ +#ifndef V4L2_CAP_VIDEO_OUTPUT_OVERLAY +#define V4L2_FIELD_INTERLACED_TB 8 +#define V4L2_FIELD_INTERLACED_BT 9 +#endif + GST_DEBUG_CATEGORY_EXTERN (v4l2_debug); #define GST_CAT_DEFAULT v4l2_debug diff --git a/sys/v4l2/gstv4l2object.c b/sys/v4l2/gstv4l2object.c index 21ec2bc622..7e15489c4e 100644 --- a/sys/v4l2/gstv4l2object.c +++ b/sys/v4l2/gstv4l2object.c @@ -41,6 +41,13 @@ #include "gst/gst-i18n-plugin.h" +/* videodev2.h is not versioned and we can't easily check for the presence + * of enum values at compile time, but the V4L2_CAP_VIDEO_OUTPUT_OVERLAY define + * was added in the same commit as V4L2_FIELD_INTERLACED_{TB,BT} (b2787845) */ +#ifndef V4L2_CAP_VIDEO_OUTPUT_OVERLAY +#define V4L2_FIELD_INTERLACED_TB 8 +#define V4L2_FIELD_INTERLACED_BT 9 +#endif GST_DEBUG_CATEGORY_EXTERN (v4l2_debug); #define GST_CAT_DEFAULT v4l2_debug From a49a248fff74f4a87bdecd8975f6a136055d1109 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Fri, 14 Jan 2011 14:48:49 +0100 Subject: [PATCH 238/351] avimux: set src pad caps when starting file Fixes #639516. --- gst/avi/gstavimux.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/gst/avi/gstavimux.c b/gst/avi/gstavimux.c index 83dbbbd2f2..6a92a2b168 100644 --- a/gst/avi/gstavimux.c +++ b/gst/avi/gstavimux.c @@ -1673,6 +1673,7 @@ gst_avi_mux_start_file (GstAviMux * avimux) GstFlowReturn res; GstBuffer *header; GSList *node; + GstCaps *caps; avimux->total_data = 0; avimux->total_frames = 0; @@ -1715,6 +1716,10 @@ gst_avi_mux_start_file (GstAviMux * avimux) } } + caps = gst_caps_copy (gst_pad_get_pad_template_caps (avimux->srcpad)); + gst_pad_set_caps (avimux->srcpad, caps); + gst_caps_unref (caps); + /* let downstream know we think in BYTES and expect to do seeking later on */ gst_pad_push_event (avimux->srcpad, gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_BYTES, 0, -1, 0)); From 0124302b58c2c575245bcec36cc9d12dd75761e1 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Fri, 14 Jan 2011 14:51:51 +0100 Subject: [PATCH 239/351] matroskamux: avoid creating caps from string when possible Fixes #639516. --- gst/matroska/matroska-mux.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/gst/matroska/matroska-mux.c b/gst/matroska/matroska-mux.c index 7da4e6fa92..7cea073179 100644 --- a/gst/matroska/matroska-mux.c +++ b/gst/matroska/matroska-mux.c @@ -2025,9 +2025,9 @@ gst_matroska_mux_start (GstMatroskaMux * mux) GTimeVal time = { 0, 0 }; if (!strcmp (mux->doctype, GST_MATROSKA_DOCTYPE_WEBM)) { - ebml->caps = gst_caps_from_string ("video/webm"); + ebml->caps = gst_caps_new_simple ("video/webm", NULL); } else { - ebml->caps = gst_caps_from_string ("video/x-matroska"); + ebml->caps = gst_caps_new_simple ("video/x-matroska", NULL); } /* we start with a EBML header */ doctype = mux->doctype; From e18e25a324c6367210f0210f48afeab7ca38c529 Mon Sep 17 00:00:00 2001 From: Arun Raghavan Date: Mon, 17 Jan 2011 20:09:16 +0530 Subject: [PATCH 240/351] pulsesink: Uncork stream while flushing the ringbuffer After starting the ringbuffer, we wait for enough data to arrive before uncorking the stream. This will cause the pipeline to stall if we get an EOS (or otherwise need to flush the stream) before sufficient data becomes available. This patch makes sure that the stream is uncorked while flushing to avoid this problem. Fixes issue with a webkit unit test testing reverse playback of an MP4 H.264/AAC file. https://bugzilla.gnome.org/show_bug.cgi?id=639740 --- ext/pulse/pulsesink.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/ext/pulse/pulsesink.c b/ext/pulse/pulsesink.c index f885843c85..9bebfec3bb 100644 --- a/ext/pulse/pulsesink.c +++ b/ext/pulse/pulsesink.c @@ -2637,6 +2637,11 @@ gst_pulsesink_flush_ringbuffer (GstPulseSink * psink) gst_pulsering_flush (pbuf); + /* Uncork if we haven't already (happens when waiting to get enough data + * to send out the first time) */ + if (pbuf->corked) + gst_pulsering_set_corked (pbuf, FALSE, FALSE); + /* We're not interested if this operation failed or not */ unlock: pa_threaded_mainloop_unlock (mainloop); From 7d9b7abbd45947d107da19f9e5248d8884cfeb5d Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Tue, 18 Jan 2011 09:33:06 +0000 Subject: [PATCH 241/351] avisubtitle: set caps on srcpad to fix issue with discoverer Set caps from the start so discoverer doesn't blow up on seeing no negotiated caps between elements on preroll, which might happen if no subtitle buffers have been pushed yet at the time. See file from bug #603308. --- gst/avi/gstavisubtitle.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/gst/avi/gstavisubtitle.c b/gst/avi/gstavisubtitle.c index 38771c38f6..10c4fafe7d 100644 --- a/gst/avi/gstavisubtitle.c +++ b/gst/avi/gstavisubtitle.c @@ -333,12 +333,20 @@ gst_avi_subtitle_class_init (GstAviSubtitleClass * klass) static void gst_avi_subtitle_init (GstAviSubtitle * self, GstAviSubtitleClass * klass) { + GstCaps *caps; + self->src = gst_pad_new_from_static_template (&src_template, "src"); gst_element_add_pad (GST_ELEMENT (self), self->src); self->sink = gst_pad_new_from_static_template (&sink_template, "sink"); gst_pad_set_chain_function (self->sink, GST_DEBUG_FUNCPTR (gst_avi_subtitle_chain)); + + caps = gst_static_pad_template_get_caps (&src_template); + gst_pad_set_caps (self->src, caps); + gst_caps_unref (caps); + + gst_pad_use_fixed_caps (self->src); gst_element_add_pad (GST_ELEMENT (self), self->sink); self->subfile = NULL; From c2b343cd7694822f0c2a5a77ace37cf9a0691e17 Mon Sep 17 00:00:00 2001 From: David Schleef Date: Wed, 12 Jan 2011 14:03:57 -0800 Subject: [PATCH 242/351] deinterlace: rewrite how neighboring scan lines are calculated Old code was difficult to understand exactly how the neighboring scan lines are calculated, and it appeared that some were off by +2 or -2, depending on the field flag. Fixes #639321. --- gst/deinterlace/gstdeinterlacemethod.c | 292 ++++++++----------------- 1 file changed, 93 insertions(+), 199 deletions(-) diff --git a/gst/deinterlace/gstdeinterlacemethod.c b/gst/deinterlace/gstdeinterlacemethod.c index ac653dff89..7228f01676 100644 --- a/gst/deinterlace/gstdeinterlacemethod.c +++ b/gst/deinterlace/gstdeinterlacemethod.c @@ -307,132 +307,84 @@ gst_deinterlace_simple_method_deinterlace_frame_packed (GstDeinterlaceMethod * GstDeinterlaceSimpleMethod *self = GST_DEINTERLACE_SIMPLE_METHOD (method); GstDeinterlaceMethodClass *dm_class = GST_DEINTERLACE_METHOD_GET_CLASS (self); GstDeinterlaceScanlineData scanlines; - guint8 *out = GST_BUFFER_DATA (outbuf); - const guint8 *field0 = NULL, *field1 = NULL, *field2 = NULL, *field3 = NULL; + guint8 *dest; + const guint8 *field0, *field1, *field2, *field3; gint cur_field_idx = history_count - dm_class->fields_required; guint cur_field_flags = history[cur_field_idx].flags; - gint line; - gint field_height = self->parent.frame_height / 2; - gint row_stride = self->parent.row_stride[0]; - gint field_stride = self->parent.row_stride[0] * 2; + gint i; + gint frame_height = self->parent.frame_height; + gint stride = self->parent.row_stride[0]; g_assert (self->interpolate_scanline_packed != NULL); g_assert (self->copy_scanline_packed != NULL); + dest = GST_BUFFER_DATA (outbuf); field0 = GST_BUFFER_DATA (history[cur_field_idx].buf); - if (history[cur_field_idx].flags & PICTURE_INTERLACED_BOTTOM) - field0 += row_stride; g_assert (dm_class->fields_required <= 4); if (dm_class->fields_required >= 2) { field1 = GST_BUFFER_DATA (history[cur_field_idx + 1].buf); - if (history[cur_field_idx + 1].flags & PICTURE_INTERLACED_BOTTOM) - field1 += row_stride; + } else { + field1 = NULL; } if (dm_class->fields_required >= 3) { field2 = GST_BUFFER_DATA (history[cur_field_idx + 2].buf); - if (history[cur_field_idx + 2].flags & PICTURE_INTERLACED_BOTTOM) - field2 += row_stride; + } else { + field2 = NULL; } if (dm_class->fields_required >= 4) { field3 = GST_BUFFER_DATA (history[cur_field_idx + 3].buf); - if (history[cur_field_idx + 3].flags & PICTURE_INTERLACED_BOTTOM) - field3 += row_stride; + } else { + field3 = NULL; } - if (cur_field_flags == PICTURE_INTERLACED_BOTTOM) { - /* double the first scanline of the bottom field */ - memcpy (out, field0, row_stride); - out += row_stride; - } - - memcpy (out, field0, row_stride); - out += row_stride; - - for (line = 2; line <= field_height; line++) { +#define CLAMP_LOW(i) (((i)<0) ? (i+2) : (i)) +#define CLAMP_HI(i) (((i)>=(frame_height)) ? (i-2) : (i)) +#define LINE(x,i) ((x) + CLAMP_HI(CLAMP_LOW(i)) * (stride)) +#define LINE2(x,i) ((x) ? LINE(x,i) : NULL) + for (i = 0; i < frame_height; i++) { memset (&scanlines, 0, sizeof (scanlines)); scanlines.bottom_field = (cur_field_flags == PICTURE_INTERLACED_BOTTOM); - /* interp. scanline */ - scanlines.t0 = field0; - scanlines.b0 = field0 + field_stride; + if (!((i & 1) ^ (cur_field_flags == PICTURE_INTERLACED_BOTTOM))) { + /* copying */ + scanlines.tt0 = LINE2 (field0, (i - 2 >= 0) ? i - 2 : i); + scanlines.m0 = LINE2 (field0, i); + scanlines.bb0 = LINE2 (field0, (i + 2 < frame_height ? i + 2 : i)); - if (field1 != NULL) { - scanlines.tt1 = field1; - scanlines.m1 = field1 + field_stride; - scanlines.bb1 = field1 + field_stride * 2; - field1 += field_stride; + scanlines.t1 = LINE2 (field1, i - 1); + scanlines.b1 = LINE2 (field1, i + 1); + + scanlines.tt2 = LINE2 (field2, (i - 2 >= 0) ? i - 2 : i); + scanlines.m2 = LINE2 (field2, i); + scanlines.bb2 = LINE2 (field2, (i + 2 < frame_height ? i + 2 : i)); + + scanlines.t3 = LINE2 (field3, i - 1); + scanlines.b3 = LINE2 (field3, i + 1); + + self->copy_scanline_packed (self, LINE (dest, i), &scanlines); + } else { + /* interpolating */ + scanlines.t0 = LINE2 (field0, i - 1); + scanlines.b0 = LINE2 (field0, i + 1); + + scanlines.tt1 = LINE2 (field1, (i - 2 >= 0) ? i - 2 : i); + scanlines.m1 = LINE2 (field1, i); + scanlines.bb1 = LINE2 (field1, (i + 2 < frame_height ? i + 2 : i)); + + scanlines.t2 = LINE2 (field2, i - 1); + scanlines.b2 = LINE2 (field2, i + 1); + + scanlines.tt3 = LINE2 (field3, (i - 2 >= 0) ? i - 2 : i); + scanlines.m3 = LINE2 (field3, i); + scanlines.bb3 = LINE2 (field3, (i + 2 < frame_height ? i + 2 : i)); + + self->interpolate_scanline_packed (self, LINE (dest, i), &scanlines); } - - if (field2 != NULL) { - scanlines.t2 = field2; - scanlines.b2 = field2 + field_stride; - } - - if (field3 != NULL) { - scanlines.tt3 = field3; - scanlines.m3 = field3 + field_stride; - scanlines.bb3 = field3 + field_stride * 2; - field3 += field_stride; - } - - /* set valid data for corner cases */ - if (line == 2) { - scanlines.tt1 = scanlines.bb1; - scanlines.tt3 = scanlines.bb3; - } else if (line == field_height) { - scanlines.bb1 = scanlines.tt1; - scanlines.bb3 = scanlines.tt3; - } - - self->interpolate_scanline_packed (self, out, &scanlines); - out += row_stride; - - memset (&scanlines, 0, sizeof (scanlines)); - scanlines.bottom_field = (cur_field_flags == PICTURE_INTERLACED_BOTTOM); - - /* copy a scanline */ - scanlines.tt0 = field0; - scanlines.m0 = field0 + field_stride; - scanlines.bb0 = field0 + field_stride * 2; - field0 += field_stride; - - if (field1 != NULL) { - scanlines.t1 = field1; - scanlines.b1 = field1 + field_stride; - } - - if (field2 != NULL) { - scanlines.tt2 = field2; - scanlines.m2 = field2 + field_stride; - scanlines.bb2 = field2 + field_stride * 2; - field2 += field_stride; - } - - if (field3 != NULL) { - scanlines.t3 = field3; - scanlines.b3 = field3 + field_stride; - } - - /* set valid data for corner cases */ - if (line == field_height) { - scanlines.bb0 = scanlines.tt0; - scanlines.b1 = scanlines.t1; - scanlines.bb2 = scanlines.tt2; - scanlines.b3 = scanlines.t3; - } - - self->copy_scanline_packed (self, out, &scanlines); - out += row_stride; - } - - if (cur_field_flags == PICTURE_INTERLACED_TOP) { - /* double the last scanline of the top field */ - memcpy (out, field0, row_stride); } } @@ -483,111 +435,59 @@ gst_deinterlace_simple_method_copy_scanline_planar_v (GstDeinterlaceSimpleMethod static void gst_deinterlace_simple_method_deinterlace_frame_planar_plane - (GstDeinterlaceSimpleMethod * self, guint8 * out, const guint8 * field0, + (GstDeinterlaceSimpleMethod * self, guint8 * dest, const guint8 * field0, const guint8 * field1, const guint8 * field2, const guint8 * field3, guint cur_field_flags, gint plane, GstDeinterlaceSimpleMethodFunction copy_scanline, GstDeinterlaceSimpleMethodFunction interpolate_scanline) { GstDeinterlaceScanlineData scanlines; - gint line; - gint field_height = self->parent.height[plane] / 2; - gint row_stride = self->parent.row_stride[plane]; - gint field_stride = self->parent.row_stride[plane] * 2; + gint i; + gint frame_height = self->parent.height[plane]; + gint stride = self->parent.row_stride[plane]; g_assert (interpolate_scanline != NULL); g_assert (copy_scanline != NULL); - if (cur_field_flags == PICTURE_INTERLACED_BOTTOM) { - /* double the first scanline of the bottom field */ - memcpy (out, field0, row_stride); - out += row_stride; - } - - memcpy (out, field0, row_stride); - out += row_stride; - - for (line = 2; line <= field_height; line++) { - + for (i = 0; i < frame_height; i++) { memset (&scanlines, 0, sizeof (scanlines)); scanlines.bottom_field = (cur_field_flags == PICTURE_INTERLACED_BOTTOM); - /* interp. scanline */ - scanlines.t0 = field0; - scanlines.b0 = field0 + field_stride; + if (!((i & 1) ^ (cur_field_flags == PICTURE_INTERLACED_BOTTOM))) { + /* copying */ + scanlines.tt0 = LINE2 (field0, (i - 2 >= 0) ? i - 2 : i); + scanlines.m0 = LINE2 (field0, i); + scanlines.bb0 = LINE2 (field0, (i + 2 < frame_height ? i + 2 : i)); - if (field1 != NULL) { - scanlines.tt1 = field1; - scanlines.m1 = field1 + field_stride; - scanlines.bb1 = field1 + field_stride * 2; - field1 += field_stride; + scanlines.t1 = LINE2 (field1, i - 1); + scanlines.b1 = LINE2 (field1, i + 1); + + scanlines.tt2 = LINE2 (field2, (i - 2 >= 0) ? i - 2 : i); + scanlines.m2 = LINE2 (field2, i); + scanlines.bb2 = LINE2 (field2, (i + 2 < frame_height ? i + 2 : i)); + + scanlines.t3 = LINE2 (field3, i - 1); + scanlines.b3 = LINE2 (field3, i + 1); + + copy_scanline (self, LINE (dest, i), &scanlines); + } else { + /* interpolating */ + scanlines.t0 = LINE2 (field0, i - 1); + scanlines.b0 = LINE2 (field0, i + 1); + + scanlines.tt1 = LINE2 (field1, (i - 2 >= 0) ? i - 2 : i); + scanlines.m1 = LINE2 (field1, i); + scanlines.bb1 = LINE2 (field1, (i + 2 < frame_height ? i + 2 : i)); + + scanlines.t2 = LINE2 (field2, i - 1); + scanlines.b2 = LINE2 (field2, i + 1); + + scanlines.tt3 = LINE2 (field3, (i - 2 >= 0) ? i - 2 : i); + scanlines.m3 = LINE2 (field3, i); + scanlines.bb3 = LINE2 (field3, (i + 2 < frame_height ? i + 2 : i)); + + interpolate_scanline (self, LINE (dest, i), &scanlines); } - - if (field2 != NULL) { - scanlines.t2 = field2; - scanlines.b2 = field2 + field_stride; - } - - if (field3 != NULL) { - scanlines.tt3 = field3; - scanlines.m3 = field3 + field_stride; - scanlines.bb3 = field3 + field_stride * 2; - field3 += field_stride; - } - - /* set valid data for corner cases */ - if (line == 2) { - scanlines.tt1 = scanlines.bb1; - scanlines.tt3 = scanlines.bb3; - } else if (line == field_height) { - scanlines.bb1 = scanlines.tt1; - scanlines.bb3 = scanlines.tt3; - } - - interpolate_scanline (self, out, &scanlines); - out += row_stride; - - memset (&scanlines, 0, sizeof (scanlines)); - scanlines.bottom_field = (cur_field_flags == PICTURE_INTERLACED_BOTTOM); - - /* copy a scanline */ - scanlines.tt0 = field0; - scanlines.m0 = field0 + field_stride; - scanlines.bb0 = field0 + field_stride * 2; - field0 += field_stride; - - if (field1 != NULL) { - scanlines.t1 = field1; - scanlines.b1 = field1 + field_stride; - } - - if (field2 != NULL) { - scanlines.tt2 = field2; - scanlines.m2 = field2 + field_stride; - scanlines.bb2 = field2 + field_stride * 2; - field2 += field_stride; - } - - if (field3 != NULL) { - scanlines.t3 = field3; - scanlines.b3 = field3 + field_stride; - } - - /* set valid data for corner cases */ - if (line == field_height) { - scanlines.bb0 = scanlines.tt0; - scanlines.b1 = scanlines.t1; - scanlines.bb2 = scanlines.tt2; - scanlines.b3 = scanlines.t3; - } - - copy_scanline (self, out, &scanlines); - out += row_stride; - } - - if (cur_field_flags == PICTURE_INTERLACED_TOP) { - /* double the last scanline of the top field */ - memcpy (out, field0, row_stride); } } @@ -599,10 +499,10 @@ gst_deinterlace_simple_method_deinterlace_frame_planar (GstDeinterlaceMethod * GstDeinterlaceSimpleMethod *self = GST_DEINTERLACE_SIMPLE_METHOD (method); GstDeinterlaceMethodClass *dm_class = GST_DEINTERLACE_METHOD_GET_CLASS (self); guint8 *out; - const guint8 *field0 = NULL, *field1 = NULL, *field2 = NULL, *field3 = NULL; + const guint8 *field0, *field1, *field2, *field3; gint cur_field_idx = history_count - dm_class->fields_required; guint cur_field_flags = history[cur_field_idx].flags; - gint i, row_stride, offset; + gint i, offset; GstDeinterlaceSimpleMethodFunction copy_scanline; GstDeinterlaceSimpleMethodFunction interpolate_scanline; @@ -614,7 +514,6 @@ gst_deinterlace_simple_method_deinterlace_frame_planar (GstDeinterlaceMethod * g_assert (self->copy_scanline_planar[2] != NULL); for (i = 0; i < 3; i++) { - row_stride = self->parent.row_stride[i]; offset = self->parent.offset[i]; copy_scanline = self->copy_scanline_planar[i]; interpolate_scanline = self->interpolate_scanline_planar[i]; @@ -622,27 +521,22 @@ gst_deinterlace_simple_method_deinterlace_frame_planar (GstDeinterlaceMethod * out = GST_BUFFER_DATA (outbuf) + offset; field0 = GST_BUFFER_DATA (history[cur_field_idx].buf) + offset; - if (history[cur_field_idx].flags & PICTURE_INTERLACED_BOTTOM) - field0 += row_stride; g_assert (dm_class->fields_required <= 4); + field1 = NULL; if (dm_class->fields_required >= 2) { field1 = GST_BUFFER_DATA (history[cur_field_idx + 1].buf) + offset; - if (history[cur_field_idx + 1].flags & PICTURE_INTERLACED_BOTTOM) - field1 += row_stride; } + field2 = NULL; if (dm_class->fields_required >= 3) { field2 = GST_BUFFER_DATA (history[cur_field_idx + 2].buf) + offset; - if (history[cur_field_idx + 2].flags & PICTURE_INTERLACED_BOTTOM) - field2 += row_stride; } + field3 = NULL; if (dm_class->fields_required >= 4) { field3 = GST_BUFFER_DATA (history[cur_field_idx + 3].buf) + offset; - if (history[cur_field_idx + 3].flags & PICTURE_INTERLACED_BOTTOM) - field3 += row_stride; } gst_deinterlace_simple_method_deinterlace_frame_planar_plane (self, out, From 84311e58d3e61753bec23657638b7205d268a8bc Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Tue, 18 Jan 2011 12:58:29 +0000 Subject: [PATCH 243/351] 0.10.26.4 pre-release --- configure.ac | 2 +- win32/common/config.h | 8 ++++---- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/configure.ac b/configure.ac index 5e5dbd0a18..61043a31a6 100644 --- a/configure.ac +++ b/configure.ac @@ -5,7 +5,7 @@ dnl please read gstreamer/docs/random/autotools before changing this file dnl initialize autoconf dnl releases only do -Wall, git and prerelease does -Werror too dnl use a three digit version number for releases, and four for git/pre -AC_INIT(GStreamer Good Plug-ins, 0.10.26.3, +AC_INIT(GStreamer Good Plug-ins, 0.10.26.4, http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer, gst-plugins-good) diff --git a/win32/common/config.h b/win32/common/config.h index e45f46beb1..d62372eb1e 100644 --- a/win32/common/config.h +++ b/win32/common/config.h @@ -49,7 +49,7 @@ #define GST_PACKAGE_ORIGIN "Unknown package origin" /* GStreamer package release date/time for plugins as YYYY-MM-DD */ -#define GST_PACKAGE_RELEASE_DATETIME "2011-01-11T20:04Z" +#define GST_PACKAGE_RELEASE_DATETIME "2011-01-18T11:02Z" /* struct v4l2_buffer missing */ #undef GST_V4L2_MISSING_BUFDECL @@ -386,7 +386,7 @@ #define PACKAGE_NAME "GStreamer Good Plug-ins" /* Define to the full name and version of this package. */ -#define PACKAGE_STRING "GStreamer Good Plug-ins 0.10.26.3" +#define PACKAGE_STRING "GStreamer Good Plug-ins 0.10.26.4" /* Define to the one symbol short name of this package. */ #define PACKAGE_TARNAME "gst-plugins-good" @@ -395,7 +395,7 @@ #undef PACKAGE_URL /* Define to the version of this package. */ -#define PACKAGE_VERSION "0.10.26.3" +#define PACKAGE_VERSION "0.10.26.4" /* directory where plugins are located */ #ifdef _DEBUG @@ -426,7 +426,7 @@ #undef STDC_HEADERS /* Version number of package */ -#define VERSION "0.10.26.3" +#define VERSION "0.10.26.4" /* old wavpack API */ #undef WAVPACK_OLD_API From 9acda034f89a8d362ee5ba47a4f2a063aa270b8b Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Thu, 20 Jan 2011 00:07:33 +0000 Subject: [PATCH 244/351] v4l2sink: fix accidental breakage of navigation interface support --- sys/v4l2/gstv4l2sink.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sys/v4l2/gstv4l2sink.c b/sys/v4l2/gstv4l2sink.c index f3eba3b90d..aa6785cfea 100644 --- a/sys/v4l2/gstv4l2sink.c +++ b/sys/v4l2/gstv4l2sink.c @@ -103,6 +103,7 @@ gst_v4l2sink_iface_supported (GstImplementsInterface * iface, GType iface_type) #ifdef HAVE_XVIDEO g_assert (iface_type == GST_TYPE_X_OVERLAY || + iface_type == GST_TYPE_NAVIGATION || iface_type == GST_TYPE_COLOR_BALANCE || iface_type == GST_TYPE_VIDEO_ORIENTATION); #else From 6d545e29add17f1c02ee4fe5623409ed63369c17 Mon Sep 17 00:00:00 2001 From: Edward Hervey Date: Thu, 20 Jan 2011 11:45:47 +0100 Subject: [PATCH 245/351] qtdemux: Gracefully handle mov files misusing the WAVE atoms Check that the WAVEHEADER node is present instead of blindly using it. If not present we won't be able to provide a more refined caps, but at least we won't crash. https://bugzilla.gnome.org/show_bug.cgi?id=640028 --- gst/qtdemux/qtdemux.c | 41 ++++++++++++++++++++++------------------- 1 file changed, 22 insertions(+), 19 deletions(-) diff --git a/gst/qtdemux/qtdemux.c b/gst/qtdemux/qtdemux.c index 80fd5a0313..b4f9cbf164 100644 --- a/gst/qtdemux/qtdemux.c +++ b/gst/qtdemux/qtdemux.c @@ -7004,31 +7004,34 @@ qtdemux_parse_trak (GstQTDemux * qtdemux, GNode * trak) guint32 headerlen; waveheadernode = qtdemux_tree_get_child_by_type (wavenode, fourcc); - waveheader = (const guint8 *) waveheadernode->data; - headerlen = QT_UINT32 (waveheader); + if (waveheadernode) { + waveheader = (const guint8 *) waveheadernode->data; + headerlen = QT_UINT32 (waveheader); - if (headerlen > 8) { - gst_riff_strf_auds *header = NULL; - GstBuffer *headerbuf; - GstBuffer *extra; + if (headerlen > 8) { + gst_riff_strf_auds *header = NULL; + GstBuffer *headerbuf; + GstBuffer *extra; - waveheader += 8; - headerlen -= 8; + waveheader += 8; + headerlen -= 8; - headerbuf = gst_buffer_new (); - GST_BUFFER_DATA (headerbuf) = (guint8 *) waveheader; - GST_BUFFER_SIZE (headerbuf) = headerlen; + headerbuf = gst_buffer_new (); + GST_BUFFER_DATA (headerbuf) = (guint8 *) waveheader; + GST_BUFFER_SIZE (headerbuf) = headerlen; - if (gst_riff_parse_strf_auds (GST_ELEMENT_CAST (qtdemux), - headerbuf, &header, &extra)) { - gst_caps_unref (stream->caps); - stream->caps = gst_riff_create_audio_caps (header->format, NULL, - header, extra, NULL, NULL); + if (gst_riff_parse_strf_auds (GST_ELEMENT_CAST (qtdemux), + headerbuf, &header, &extra)) { + gst_caps_unref (stream->caps); + stream->caps = gst_riff_create_audio_caps (header->format, NULL, + header, extra, NULL, NULL); - if (extra) - gst_buffer_unref (extra); + if (extra) + gst_buffer_unref (extra); + } } - } + } else + GST_DEBUG ("Didn't find waveheadernode for this codec"); } g_node_destroy (wavenode); } From 08855b45b6fddd4f675191f1ebec012917fb1664 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Thu, 20 Jan 2011 13:41:33 +0000 Subject: [PATCH 246/351] rtspsrc: don't leak url string https://bugzilla.gnome.org/show_bug.cgi?id=640064 --- gst/rtsp/gstrtspsrc.c | 1 + 1 file changed, 1 insertion(+) diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c index a507d4e747..e4ec109766 100644 --- a/gst/rtsp/gstrtspsrc.c +++ b/gst/rtsp/gstrtspsrc.c @@ -528,6 +528,7 @@ gst_rtspsrc_finalize (GObject * object) gst_rtsp_ext_list_free (rtspsrc->extensions); g_free (rtspsrc->conninfo.location); gst_rtsp_url_free (rtspsrc->conninfo.url); + g_free (rtspsrc->conninfo.url_str); g_free (rtspsrc->user_id); g_free (rtspsrc->user_pw); From 693b3b7e0bc16604321724686c1e174c6b6fb45b Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Thu, 20 Jan 2011 14:10:55 +0000 Subject: [PATCH 247/351] h264depay: don't leak codec data buffer in byte-stream=true mode https://bugzilla.gnome.org/show_bug.cgi?id=640063 --- gst/rtp/gstrtph264depay.c | 1 + 1 file changed, 1 insertion(+) diff --git a/gst/rtp/gstrtph264depay.c b/gst/rtp/gstrtph264depay.c index 20261c5803..0d62628384 100644 --- a/gst/rtp/gstrtph264depay.c +++ b/gst/rtp/gstrtph264depay.c @@ -445,6 +445,7 @@ gst_rtp_h264_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps) gst_caps_set_simple (srccaps, "codec_data", GST_TYPE_BUFFER, codec_data, NULL); + gst_buffer_unref (codec_data); } gst_caps_set_simple (srccaps, "stream-format", G_TYPE_STRING, From 43505244cbcb7fc57f5667cc5276f0d2f296f731 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Fri, 21 Jan 2011 12:54:16 +0000 Subject: [PATCH 248/351] Release 0.10.27 --- ChangeLog | 3298 ++++++++++++++++- NEWS | 105 +- RELEASE | 255 +- configure.ac | 6 +- docs/plugins/inspect/plugin-1394.xml | 4 +- docs/plugins/inspect/plugin-aasink.xml | 4 +- docs/plugins/inspect/plugin-alaw.xml | 4 +- docs/plugins/inspect/plugin-alpha.xml | 4 +- docs/plugins/inspect/plugin-alphacolor.xml | 4 +- docs/plugins/inspect/plugin-annodex.xml | 4 +- docs/plugins/inspect/plugin-apetag.xml | 4 +- docs/plugins/inspect/plugin-audiofx.xml | 4 +- docs/plugins/inspect/plugin-auparse.xml | 4 +- docs/plugins/inspect/plugin-autodetect.xml | 4 +- docs/plugins/inspect/plugin-avi.xml | 4 +- docs/plugins/inspect/plugin-cacasink.xml | 4 +- docs/plugins/inspect/plugin-cairo.xml | 4 +- docs/plugins/inspect/plugin-cutter.xml | 4 +- docs/plugins/inspect/plugin-debug.xml | 4 +- docs/plugins/inspect/plugin-deinterlace.xml | 4 +- docs/plugins/inspect/plugin-dv.xml | 4 +- docs/plugins/inspect/plugin-efence.xml | 4 +- docs/plugins/inspect/plugin-effectv.xml | 4 +- docs/plugins/inspect/plugin-equalizer.xml | 4 +- docs/plugins/inspect/plugin-esdsink.xml | 4 +- docs/plugins/inspect/plugin-flac.xml | 4 +- docs/plugins/inspect/plugin-flv.xml | 4 +- docs/plugins/inspect/plugin-flxdec.xml | 4 +- docs/plugins/inspect/plugin-gconfelements.xml | 4 +- docs/plugins/inspect/plugin-gdkpixbuf.xml | 4 +- docs/plugins/inspect/plugin-goom.xml | 4 +- docs/plugins/inspect/plugin-goom2k1.xml | 4 +- docs/plugins/inspect/plugin-gstrtpmanager.xml | 4 +- docs/plugins/inspect/plugin-halelements.xml | 4 +- docs/plugins/inspect/plugin-icydemux.xml | 4 +- docs/plugins/inspect/plugin-id3demux.xml | 4 +- docs/plugins/inspect/plugin-imagefreeze.xml | 4 +- docs/plugins/inspect/plugin-interleave.xml | 4 +- docs/plugins/inspect/plugin-jack.xml | 4 +- docs/plugins/inspect/plugin-jpeg.xml | 4 +- docs/plugins/inspect/plugin-level.xml | 4 +- docs/plugins/inspect/plugin-matroska.xml | 4 +- docs/plugins/inspect/plugin-mulaw.xml | 4 +- docs/plugins/inspect/plugin-multifile.xml | 4 +- docs/plugins/inspect/plugin-multipart.xml | 4 +- .../plugins/inspect/plugin-navigationtest.xml | 4 +- docs/plugins/inspect/plugin-oss4.xml | 4 +- docs/plugins/inspect/plugin-ossaudio.xml | 4 +- docs/plugins/inspect/plugin-png.xml | 4 +- docs/plugins/inspect/plugin-pulseaudio.xml | 4 +- docs/plugins/inspect/plugin-quicktime.xml | 4 +- docs/plugins/inspect/plugin-replaygain.xml | 4 +- docs/plugins/inspect/plugin-rtp.xml | 4 +- docs/plugins/inspect/plugin-rtsp.xml | 4 +- docs/plugins/inspect/plugin-shapewipe.xml | 4 +- docs/plugins/inspect/plugin-shout2send.xml | 2 +- docs/plugins/inspect/plugin-smpte.xml | 4 +- docs/plugins/inspect/plugin-soup.xml | 4 +- docs/plugins/inspect/plugin-spectrum.xml | 4 +- docs/plugins/inspect/plugin-speex.xml | 4 +- docs/plugins/inspect/plugin-taglib.xml | 4 +- docs/plugins/inspect/plugin-udp.xml | 4 +- docs/plugins/inspect/plugin-video4linux2.xml | 4 +- docs/plugins/inspect/plugin-videobox.xml | 4 +- docs/plugins/inspect/plugin-videocrop.xml | 4 +- docs/plugins/inspect/plugin-videofilter.xml | 4 +- docs/plugins/inspect/plugin-videomixer.xml | 4 +- docs/plugins/inspect/plugin-wavenc.xml | 4 +- docs/plugins/inspect/plugin-wavpack.xml | 4 +- docs/plugins/inspect/plugin-wavparse.xml | 4 +- docs/plugins/inspect/plugin-ximagesrc.xml | 4 +- docs/plugins/inspect/plugin-y4menc.xml | 4 +- gst-plugins-good.doap | 11 + win32/common/config.h | 10 +- 74 files changed, 3672 insertions(+), 283 deletions(-) diff --git a/ChangeLog b/ChangeLog index 72ede1114b..941cfb4198 100644 --- a/ChangeLog +++ b/ChangeLog @@ -1,9 +1,3301 @@ -=== release 0.10.26 === +=== release 0.10.27 === -2010-12-01 Tim-Philipp Müller +2011-01-21 Tim-Philipp Müller * configure.ac: - releasing 0.10.26, "Escapades" + releasing 0.10.27, "Some Kind of Temporal Blend" + +2011-01-20 14:10:55 +0000 Tim-Philipp Müller + + * gst/rtp/gstrtph264depay.c: + h264depay: don't leak codec data buffer in byte-stream=true mode + https://bugzilla.gnome.org/show_bug.cgi?id=640063 + +2011-01-20 13:41:33 +0000 Tim-Philipp Müller + + * gst/rtsp/gstrtspsrc.c: + rtspsrc: don't leak url string + https://bugzilla.gnome.org/show_bug.cgi?id=640064 + +2011-01-20 11:45:47 +0100 Edward Hervey + + * gst/qtdemux/qtdemux.c: + qtdemux: Gracefully handle mov files misusing the WAVE atoms + Check that the WAVEHEADER node is present instead of blindly using it. + If not present we won't be able to provide a more refined caps, but at + least we won't crash. + https://bugzilla.gnome.org/show_bug.cgi?id=640028 + +2011-01-20 00:07:33 +0000 Tim-Philipp Müller + + * sys/v4l2/gstv4l2sink.c: + v4l2sink: fix accidental breakage of navigation interface support + +2011-01-18 12:58:29 +0000 Tim-Philipp Müller + + * configure.ac: + * win32/common/config.h: + 0.10.26.4 pre-release + +2011-01-12 14:03:57 -0800 David Schleef + + * gst/deinterlace/gstdeinterlacemethod.c: + deinterlace: rewrite how neighboring scan lines are calculated + Old code was difficult to understand exactly how the neighboring + scan lines are calculated, and it appeared that some were off by + +2 or -2, depending on the field flag. Fixes #639321. + +2011-01-18 09:33:06 +0000 Tim-Philipp Müller + + * gst/avi/gstavisubtitle.c: + avisubtitle: set caps on srcpad to fix issue with discoverer + Set caps from the start so discoverer doesn't blow up on + seeing no negotiated caps between elements on preroll, + which might happen if no subtitle buffers have been + pushed yet at the time. See file from bug #603308. + +2011-01-17 20:09:16 +0530 Arun Raghavan + + * ext/pulse/pulsesink.c: + pulsesink: Uncork stream while flushing the ringbuffer + After starting the ringbuffer, we wait for enough data to arrive before + uncorking the stream. This will cause the pipeline to stall if we get an + EOS (or otherwise need to flush the stream) before sufficient data + becomes available. This patch makes sure that the stream is uncorked + while flushing to avoid this problem. + Fixes issue with a webkit unit test testing reverse playback of + an MP4 H.264/AAC file. + https://bugzilla.gnome.org/show_bug.cgi?id=639740 + +2011-01-14 14:51:51 +0100 Mark Nauwelaerts + + * gst/matroska/matroska-mux.c: + matroskamux: avoid creating caps from string when possible + Fixes #639516. + +2011-01-14 14:48:49 +0100 Mark Nauwelaerts + + * gst/avi/gstavimux.c: + avimux: set src pad caps when starting file + Fixes #639516. + +2011-01-12 20:38:59 +0000 Tim-Philipp Müller + + * sys/v4l2/gstv4l2bufferpool.c: + * sys/v4l2/gstv4l2object.c: + v4l2: define V4L2_FIELD_INTERLACED_{TB,BT} if not available in header + Older kernels don't have these, and there's no easy way to check for the + existance of enums that doesn't involve a configure check, so just define + these if the V4L2_CAP_VIDEO_OUTPUT_OVERLAY define is not there, which was + added in the same commit as the TB/BT enum. Fixes compilation on CentOS 5. + https://bugzilla.gnome.org/show_bug.cgi?id=639339 + +2011-01-11 23:18:59 +0000 Tim-Philipp Müller + + * configure.ac: + * win32/common/config.h: + 0.10.26.3 pre-release + +2011-01-11 22:42:42 +0000 Tim-Philipp Müller + + * docs/plugins/gst-plugins-good-plugins.args: + * docs/plugins/gst-plugins-good-plugins.hierarchy: + * docs/plugins/gst-plugins-good-plugins.interfaces: + * docs/plugins/gst-plugins-good-plugins.prerequisites: + * docs/plugins/inspect/plugin-1394.xml: + * docs/plugins/inspect/plugin-aasink.xml: + * docs/plugins/inspect/plugin-alaw.xml: + * docs/plugins/inspect/plugin-alpha.xml: + * docs/plugins/inspect/plugin-alphacolor.xml: + * docs/plugins/inspect/plugin-annodex.xml: + * docs/plugins/inspect/plugin-apetag.xml: + * docs/plugins/inspect/plugin-audiofx.xml: + * docs/plugins/inspect/plugin-auparse.xml: + * docs/plugins/inspect/plugin-autodetect.xml: + * docs/plugins/inspect/plugin-avi.xml: + * docs/plugins/inspect/plugin-cacasink.xml: + * docs/plugins/inspect/plugin-cairo.xml: + * docs/plugins/inspect/plugin-cutter.xml: + * docs/plugins/inspect/plugin-debug.xml: + * docs/plugins/inspect/plugin-deinterlace.xml: + * docs/plugins/inspect/plugin-dv.xml: + * docs/plugins/inspect/plugin-efence.xml: + * docs/plugins/inspect/plugin-effectv.xml: + * docs/plugins/inspect/plugin-equalizer.xml: + * docs/plugins/inspect/plugin-esdsink.xml: + * docs/plugins/inspect/plugin-flac.xml: + * docs/plugins/inspect/plugin-flv.xml: + * docs/plugins/inspect/plugin-flxdec.xml: + * docs/plugins/inspect/plugin-gconfelements.xml: + * docs/plugins/inspect/plugin-gdkpixbuf.xml: + * docs/plugins/inspect/plugin-goom.xml: + * docs/plugins/inspect/plugin-goom2k1.xml: + * docs/plugins/inspect/plugin-gstrtpmanager.xml: + * docs/plugins/inspect/plugin-halelements.xml: + * docs/plugins/inspect/plugin-icydemux.xml: + * docs/plugins/inspect/plugin-id3demux.xml: + * docs/plugins/inspect/plugin-imagefreeze.xml: + * docs/plugins/inspect/plugin-interleave.xml: + * docs/plugins/inspect/plugin-jack.xml: + * docs/plugins/inspect/plugin-jpeg.xml: + * docs/plugins/inspect/plugin-level.xml: + * docs/plugins/inspect/plugin-matroska.xml: + * docs/plugins/inspect/plugin-mulaw.xml: + * docs/plugins/inspect/plugin-multifile.xml: + * docs/plugins/inspect/plugin-multipart.xml: + * docs/plugins/inspect/plugin-navigationtest.xml: + * docs/plugins/inspect/plugin-oss4.xml: + * docs/plugins/inspect/plugin-ossaudio.xml: + * docs/plugins/inspect/plugin-png.xml: + * docs/plugins/inspect/plugin-pulseaudio.xml: + * docs/plugins/inspect/plugin-quicktime.xml: + * docs/plugins/inspect/plugin-replaygain.xml: + * docs/plugins/inspect/plugin-rtp.xml: + * docs/plugins/inspect/plugin-rtsp.xml: + * docs/plugins/inspect/plugin-shapewipe.xml: + * docs/plugins/inspect/plugin-shout2send.xml: + * docs/plugins/inspect/plugin-smpte.xml: + * docs/plugins/inspect/plugin-soup.xml: + * docs/plugins/inspect/plugin-spectrum.xml: + * docs/plugins/inspect/plugin-speex.xml: + * docs/plugins/inspect/plugin-taglib.xml: + * docs/plugins/inspect/plugin-udp.xml: + * docs/plugins/inspect/plugin-video4linux2.xml: + * docs/plugins/inspect/plugin-videobox.xml: + * docs/plugins/inspect/plugin-videocrop.xml: + * docs/plugins/inspect/plugin-videofilter.xml: + * docs/plugins/inspect/plugin-videomixer.xml: + * docs/plugins/inspect/plugin-wavenc.xml: + * docs/plugins/inspect/plugin-wavpack.xml: + * docs/plugins/inspect/plugin-wavparse.xml: + * docs/plugins/inspect/plugin-ximagesrc.xml: + * docs/plugins/inspect/plugin-y4menc.xml: + docs: update docs + +2011-01-11 23:39:12 +0530 Arun Raghavan + + * ext/pulse/pulsesink.c: + pulsesink: Make corking during pause synchronous + This makes the call to pa_stream_cork() during ringbuffer pause() + synchronous, which makes sure that the clock does not advance after we + take a snapshot for start_time. + https://bugzilla.gnome.org/show_bug.cgi?id=639240 + +2011-01-11 19:33:16 +0000 Tim-Philipp Müller + + * po/da.po: + * po/gl.po: + * po/pl.po: + * po/pt_BR.po: + * po/sl.po: + * po/sv.po: + * po/tr.po: + po: update translations + +2011-01-11 15:50:28 +0200 Stefan Kost + + * common: + Automatic update of common submodule + From e572c87 to f94d739 + +2011-01-10 16:36:19 +0000 Tim-Philipp Müller + + * common: + Automatic update of common submodule + From ccbaa85 to e572c87 + +2011-01-10 14:53:39 +0000 Tim-Philipp Müller + + * common: + Automatic update of common submodule + From 46445ad to ccbaa85 + +2011-01-07 13:24:02 +0000 Tim-Philipp Müller + + * configure.ac: + * win32/common/config.h: + 0.10.26.2 pre-release + +2011-01-07 13:06:38 +0000 Tim-Philipp Müller + + * po/af.po: + * po/az.po: + * po/bg.po: + * po/ca.po: + * po/cs.po: + * po/da.po: + * po/de.po: + * po/el.po: + * po/en_GB.po: + * po/es.po: + * po/eu.po: + * po/fi.po: + * po/fr.po: + * po/gl.po: + * po/hu.po: + * po/id.po: + * po/it.po: + * po/ja.po: + * po/lt.po: + * po/lv.po: + * po/mt.po: + * po/nb.po: + * po/nl.po: + * po/or.po: + * po/pl.po: + * po/pt_BR.po: + * po/ro.po: + * po/ru.po: + * po/sk.po: + * po/sl.po: + * po/sq.po: + * po/sr.po: + * po/sv.po: + * po/tr.po: + * po/uk.po: + * po/vi.po: + * po/zh_CN.po: + * po/zh_HK.po: + * po/zh_TW.po: + po: update translations + +2011-01-07 02:32:20 +0000 Tim-Philipp Müller + + * gst/alpha/gstalpha.c: + alpha: fix compiler warnings caused by -DG_DISABLE_ASSERT + +2011-01-07 02:06:51 +0000 Tim-Philipp Müller + + * gst/matroska/ebml-read.c: + matroska: don't put essential function calls into g_assert() + g_assert() will expand to NOOPs if -DG_DISABLE_ASSERT is passed. + +2011-01-07 01:35:45 +0000 Tim-Philipp Müller + + * sys/v4l2/gstv4l2sink.c: + v4l2sink: don't put functional code like ioctl calls into g_return_if_fail() + These macros will expand to NOOPs given the right defines. Also, + g_return_if_fail() and friends are meant to be used to catch programming + errors (like invalid input to functions), not runtime error handling. + +2011-01-07 01:11:02 +0000 Tim-Philipp Müller + + * tests/check/Makefile.am: + tests: never disable g_assert() and cast checks for the unit tests + The unit tests are riddled with g_assert() and friends, make sure we + don't disable assert and cast checks for the unit tests even if + this has been specified for the rest of the code base, e.g. via + --disable-glib-asserts. + +2011-01-06 12:29:21 +0100 Edward Hervey + + * gst/rtp/gstrtpmp4adepay.c: + rtp: Fix unitialized variables on macosx + +2011-01-06 12:28:58 +0100 Edward Hervey + + * gst/qtdemux/qtdemux_dump.c: + qtdemux: Fix unitialized variables on macosx + +2011-01-05 17:49:16 -0800 David Schleef + + * gst/debugutils/gstcapsdebug.c: + capsdebug: Add capdebug debug category + +2010-12-11 12:42:10 -0800 David Schleef + + * gst/deinterlace/gstdeinterlace.c: + deinterlace: Change the default to linear + The previous default, greedyh, takes 4 times as long as MPEG-2 + video decoding, and is unlikely fast enough on any current CPU + to play 1080i video in real-time. greedyl isn't much faster. + linear was chosen over vfir, since the quality advantage of vfir + is minimal compared to the occasional visual artifacts and slower + processing. + +2011-01-05 18:32:58 +0100 Wim Taymans + + * gst/rtsp/gstrtspsrc.c: + rtspsrc: don't confuse return values + Return a return value of the right type. + +2011-01-05 16:24:13 +0100 Edward Hervey + + * gst/qtdemux/qtdemux.c: + * gst/qtdemux/qtdemux_dump.c: + qtdemux: Fix unitialized variables on macosx + +2011-01-05 15:03:32 +0100 Wim Taymans + + * gst/rtp/gstrtpvrawdepay.c: + vrawdepay: fix length check + Add some more debugging. + Add the length check so we don't cause unneeded warnings. + +2011-01-05 12:04:03 +0100 Wim Taymans + + * gst/udp/gstmultiudpsink.c: + * gst/udp/gstmultiudpsink.h: + multiudpsink: add buffer-size property + Add buffer-size property to configure the kernel send buffer. + +2011-01-03 20:16:22 +0200 Stefan Kost + + * gst/rtsp/gstrtspsrc.c: + rtspsrc: remove unused variables when debug-logging disabled + +2011-01-03 20:06:35 +0200 Stefan Kost + + * gst/matroska/matroska-demux.c: + matroska-demux: remove unused variables when debug-logging disabled + +2011-01-03 18:05:15 +0100 Wim Taymans + + * ext/libcaca/gstcacasink.c: + cacasink: fix masks and strides + Use the right endianness to read the masks. + Use the right strides for the bitmap. + Fixes #638569 + +2011-01-03 01:18:06 +0000 Tim-Philipp Müller + + * sys/v4l2/gstv4l2src.c: + v4l2src: undo presumably accidental enablement of the GstXOverlay interface + Looks like this got enabled by accident when adding it to v4l2sink, + so undo this for now. Not sure it makes much sense in a GStreamer + context with current hardware. + +2011-01-03 15:40:11 +0100 Wim Taymans + + * gst/rtsp/gstrtspsrc.c: + rtspsrc: increase udp buffer size + Set a bigger UDP buffer size by default to reduce packet loss with + high bitrate streams. + +2011-01-02 19:19:27 -0800 David Schleef + + * gst/multifile/gstmultifilesink.c: + * gst/multifile/gstmultifilesink.h: + multifilesink: send stream headers in key-frame mode + +2011-01-02 19:43:02 +0000 Tim-Philipp Müller + + * ext/jack/Makefile.am: + * ext/jack/README: + * ext/jack/gstjack.c: + * ext/jack/gstjackaudiosink.c: + * ext/jack/gstjackaudiosrc.c: + jack: fix up element details and some other minor clean-ups + +2011-01-02 19:23:51 +0000 Erich Schubert + + * gst/id3demux/id3v2frames.c: + id3demux: fix parsing of ID3v2.4 genre frames with multiple genres + We'd only extract the first genre (multiple times) instead of all + genres. + https://bugzilla.gnome.org/show_bug.cgi?id=638535 + +2011-01-02 17:40:41 +0000 Tim-Philipp Müller + + * ext/jack/gstjackaudiosink.c: + * ext/jack/gstjackaudiosrc.c: + jack: template caps had lists with one value, just use value directly + +2011-01-02 17:07:19 +0000 Tim-Philipp Müller + + * ext/jack/gstjack.c: + * ext/jack/gstjackaudiosink.c: + * ext/jack/gstjackaudiosrc.c: + jack: make get_type functions thread-safe + Because we can (shouldn't be needed with other workarounds still there). + +2011-01-02 15:27:19 +0000 Tim-Philipp Müller + + * docs/plugins/gst-plugins-good-plugins.args: + * docs/plugins/gst-plugins-good-plugins.hierarchy: + * docs/plugins/gst-plugins-good-plugins.interfaces: + * docs/plugins/gst-plugins-good-plugins.prerequisites: + * docs/plugins/inspect/plugin-deinterlace.xml: + * docs/plugins/inspect/plugin-matroska.xml: + * docs/plugins/inspect/plugin-monoscope.xml: + * docs/plugins/inspect/plugin-rtp.xml: + docs: update plugin docs + +2011-01-02 15:25:41 +0000 Tim-Philipp Müller + + * .gitignore: + * configure.ac: + * docs/plugins/Makefile.am: + * docs/plugins/gst-plugins-good-plugins-docs.sgml: + * docs/plugins/gst-plugins-good-plugins-sections.txt: + * docs/plugins/inspect/plugin-jack.xml: + * ext/Makefile.am: + * gst-plugins-good.spec.in: + * tests/examples/Makefile.am: + * tests/examples/jack/Makefile.am: + jack: new jackaudiosrc and jackaudiosink elements, moved from gst-plugins-bad + https://bugzilla.gnome.org/show_bug.cgi?id=621929 + +2010-10-19 16:23:23 +0300 Stefan Kost + + * ext/jack/gstjackaudiosink.c: + * ext/jack/gstjackaudiosrc.c: + various (ext): add missing G_PARAM_STATIC_STRINGS flags + Canonicalize property names as needed. + +2010-09-09 14:49:06 -0400 Tristan Matthews + + * ext/jack/Makefile.am: + * ext/jack/gstjackaudiosink.c: + * ext/jack/gstjackaudiosrc.c: + jack: added translatable text for server not found error + +2010-09-06 17:17:54 -0400 Tristan Matthews + + * tests/examples/jack/Makefile.am: + * tests/examples/jack/jack_client.c: + examples: add test to demonstrate jack_client_t usage + +2010-09-06 16:11:31 -0400 Tristan Matthews + + * ext/jack/gstjack.c: + * ext/jack/gstjack.h: + * ext/jack/gstjackaudioclient.c: + * ext/jack/gstjackaudioclient.h: + * ext/jack/gstjackaudiosink.c: + * ext/jack/gstjackaudiosink.h: + * ext/jack/gstjackaudiosrc.c: + * ext/jack/gstjackaudiosrc.h: + jack: added client property + +2010-06-17 16:26:07 -0400 Tristan Matthews + + * ext/jack/gstjackbin.c: + jack: removed unused file gstjackbin.c + This is a 0.8 leftover. + +2010-05-13 12:55:29 +0200 Wim Taymans + + * ext/jack/gstjackaudiosrc.c: + jacksrc: make sure we always read nframes + Error out when we are asked to read a different size that what was configured as + the jack period size because that would mean something else is wrong. + Fixes #618409 + +2010-05-11 17:56:31 -0400 Tristan Matthews + + * ext/jack/gstjackaudiosrc.c: + * ext/jack/gstjackaudiosrc.h: + jack: improve process_cb + +2010-04-27 10:48:32 -0400 Tristan Matthews + + * ext/jack/Makefile.am: + * ext/jack/gstjackaudiosrc.c: + * ext/jack/gstjackutil.c: + * ext/jack/gstjackutil.h: + jack: implement multichannel support correctly for jackaudiosrc + Fixes parts of bug #616541. + +2010-04-27 11:21:16 +0300 Stefan Kost + + * ext/jack/gstjackaudiosink.c: + * ext/jack/gstjackaudiosrc.c: + * ext/jack/gstjackringbuffer.h: + jack: remove empty dispose and finalize methods + +2010-04-27 10:59:00 +0300 Stefan Kost + + * ext/jack/gstjackaudiosink.c: + * ext/jack/gstjackaudiosrc.c: + jack: don't leak caps + Add dispose methods to clear caps. + +2010-04-27 10:34:24 +0300 Stefan Kost + + * ext/jack/gstjackaudiosink.c: + * ext/jack/gstjackaudiosrc.c: + jack: don't use GST_DEBUG_FUNCPTR for gobject vmethods + +2010-03-24 15:59:53 +0200 Stefan Kost + + * ext/jack/gstjackaudiosrc.c: + jack: fix element name in section doc blob + +2010-03-22 16:56:03 +0100 Benjamin Otte + + * ext/jack/gstjackaudiosrc.c: + Add -Wold-style-definition + and fix the warnings + +2010-03-21 21:39:18 +0100 Benjamin Otte + + * ext/jack/gstjack.h: + Add -Wmissing-declarations -Wmissing-prototypes to configure flags + And fix all warnings + +2010-03-18 17:30:26 +0100 Benjamin Otte + + * ext/jack/gstjackaudiosink.c: + * ext/jack/gstjackaudiosrc.c: + gst_element_class_set_details => gst_element_class_set_details_simple + +2009-10-12 09:06:37 +0300 Stefan Kost + + * ext/jack/gstjackaudiosink.c: + * ext/jack/gstjackaudiosrc.c: + jack: ensure segtotal is at least 2 + Not only adjust buffer-time and avoid segtotal=0, but instead ensure segtotal is + atleast 2. Do same change on jacksrc. We could also check the latency and buffer + time configured by the client and adjust buffer-time so that we get to the same + number of segments. + +2009-10-12 00:51:27 +0300 Stefan Kost + + * ext/jack/gstjackaudiosink.c: + jack: don't crash in ringbuffer with SIGFPE on small buffer-times + Jack overrides user-specified latency-time with the one it gets from jack + itself. It also needs to adjust buffer-time somewhat to avoid segtotal being 0 + +2009-05-11 16:12:54 +0300 Stefan Kost + + * ext/jack/gstjackaudioclient.c: + * ext/jack/gstjackaudiosink.c: + jack: when stopping playback, do one more cycle to flush the port. Fixes #582167 + The gst_jack_audio_client_set_active() flags the port as deactivating and uses + a GCond to wait until the jack_process_cb() has run once more and cleared the + flag. This way the client zero's the buffer. This happens if one manyally go + to PAUSED and then to READY, while leting the mainloop run inbetween. + +2009-03-16 11:21:02 +0100 Wim Taymans + + * ext/jack/gstjack.c: + * ext/jack/gstjack.h: + * ext/jack/gstjackaudiosink.c: + * ext/jack/gstjackaudiosrc.c: + jack: Add new connection mode + Add a new connection mode to jacksrc and jacksink. In this new auto-force + connection mode jack will create as many ports as requested/needed in the + pipeline and will then connect as many physical ports as possible, possibly + leaving some ports unconnected. + Also get rid of some leftover g_print. + Fixes #575284. + +2008-11-23 17:50:08 +0000 Stefan Kost + + ext/jack/: Query port latencies for sink/src delays. + Original commit message from CVS: + * ext/jack/gstjackaudiosink.c: + * ext/jack/gstjackaudiosrc.c: + Query port latencies for sink/src delays. + * ext/jack/gstjackbin.c: + No printf please. + +2008-11-04 12:42:30 +0000 Stefan Kost + + Don't install static libs for plugins. Fixes #550851 for -bad. + Original commit message from CVS: + * ext/alsaspdif/Makefile.am: + * ext/amrwb/Makefile.am: + * ext/apexsink/Makefile.am: + * ext/arts/Makefile.am: + * ext/artsd/Makefile.am: + * ext/audiofile/Makefile.am: + * ext/audioresample/Makefile.am: + * ext/bz2/Makefile.am: + * ext/cdaudio/Makefile.am: + * ext/celt/Makefile.am: + * ext/dc1394/Makefile.am: + * ext/dirac/Makefile.am: + * ext/directfb/Makefile.am: + * ext/divx/Makefile.am: + * ext/dts/Makefile.am: + * ext/faac/Makefile.am: + * ext/faad/Makefile.am: + * ext/gsm/Makefile.am: + * ext/hermes/Makefile.am: + * ext/ivorbis/Makefile.am: + * ext/jack/Makefile.am: + * ext/jp2k/Makefile.am: + * ext/ladspa/Makefile.am: + * ext/lcs/Makefile.am: + * ext/libfame/Makefile.am: + * ext/libmms/Makefile.am: + * ext/metadata/Makefile.am: + * ext/mpeg2enc/Makefile.am: + * ext/mplex/Makefile.am: + * ext/musepack/Makefile.am: + * ext/musicbrainz/Makefile.am: + * ext/mythtv/Makefile.am: + * ext/nas/Makefile.am: + * ext/neon/Makefile.am: + * ext/ofa/Makefile.am: + * ext/polyp/Makefile.am: + * ext/resindvd/Makefile.am: + * ext/sdl/Makefile.am: + * ext/shout/Makefile.am: + * ext/snapshot/Makefile.am: + * ext/sndfile/Makefile.am: + * ext/soundtouch/Makefile.am: + * ext/spc/Makefile.am: + * ext/swfdec/Makefile.am: + * ext/tarkin/Makefile.am: + * ext/theora/Makefile.am: + * ext/timidity/Makefile.am: + * ext/twolame/Makefile.am: + * ext/x264/Makefile.am: + * ext/xine/Makefile.am: + * ext/xvid/Makefile.am: + * gst-libs/gst/app/Makefile.am: + * gst-libs/gst/dshow/Makefile.am: + * gst/aiffparse/Makefile.am: + * gst/app/Makefile.am: + * gst/audiobuffer/Makefile.am: + * gst/bayer/Makefile.am: + * gst/cdxaparse/Makefile.am: + * gst/chart/Makefile.am: + * gst/colorspace/Makefile.am: + * gst/dccp/Makefile.am: + * gst/deinterlace/Makefile.am: + * gst/deinterlace2/Makefile.am: + * gst/dvdspu/Makefile.am: + * gst/festival/Makefile.am: + * gst/filter/Makefile.am: + * gst/flacparse/Makefile.am: + * gst/flv/Makefile.am: + * gst/games/Makefile.am: + * gst/h264parse/Makefile.am: + * gst/librfb/Makefile.am: + * gst/mixmatrix/Makefile.am: + * gst/modplug/Makefile.am: + * gst/mpeg1sys/Makefile.am: + * gst/mpeg4videoparse/Makefile.am: + * gst/mpegdemux/Makefile.am: + * gst/mpegtsmux/Makefile.am: + * gst/mpegvideoparse/Makefile.am: + * gst/mve/Makefile.am: + * gst/nsf/Makefile.am: + * gst/nuvdemux/Makefile.am: + * gst/overlay/Makefile.am: + * gst/passthrough/Makefile.am: + * gst/pcapparse/Makefile.am: + * gst/playondemand/Makefile.am: + * gst/rawparse/Makefile.am: + * gst/real/Makefile.am: + * gst/rtjpeg/Makefile.am: + * gst/rtpmanager/Makefile.am: + * gst/scaletempo/Makefile.am: + * gst/sdp/Makefile.am: + * gst/selector/Makefile.am: + * gst/smooth/Makefile.am: + * gst/smoothwave/Makefile.am: + * gst/speed/Makefile.am: + * gst/speexresample/Makefile.am: + * gst/stereo/Makefile.am: + * gst/subenc/Makefile.am: + * gst/tta/Makefile.am: + * gst/vbidec/Makefile.am: + * gst/videodrop/Makefile.am: + * gst/videosignal/Makefile.am: + * gst/virtualdub/Makefile.am: + * gst/vmnc/Makefile.am: + * gst/y4m/Makefile.am: + * sys/acmenc/Makefile.am: + * sys/cdrom/Makefile.am: + * sys/dshowdecwrapper/Makefile.am: + * sys/dshowsrcwrapper/Makefile.am: + * sys/dvb/Makefile.am: + * sys/dxr3/Makefile.am: + * sys/fbdev/Makefile.am: + * sys/oss4/Makefile.am: + * sys/qcam/Makefile.am: + * sys/qtwrapper/Makefile.am: + * sys/vcd/Makefile.am: + * sys/wininet/Makefile.am: + * win32/common/config.h: + Don't install static libs for plugins. Fixes #550851 for -bad. + +2008-09-17 13:59:21 +0000 Jan Schmidt + + Fix compiler warnings on OS/X + Original commit message from CVS: + * ext/jack/gstjackaudiosink.c: (jack_process_cb): + * gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew): + Fix compiler warnings on OS/X + +2008-08-07 13:15:21 +0000 Stefan Kost + + ext/jack/gstjackaudiosrc.c: Try committing this once again. Now properly renamed. + Original commit message from CVS: + * ext/jack/gstjackaudiosrc.c: + Try committing this once again. Now properly renamed. + +2008-08-07 09:09:44 +0000 Stefan Kost + + docs/plugins/: docs/plugins/inspect/plugin-jack.xml + Original commit message from CVS: + * docs/plugins/Makefile.am: + * docs/plugins/gst-plugins-bad-plugins-docs.sgml: + * docs/plugins/gst-plugins-bad-plugins-sections.txt: + * docs/plugins/gst-plugins-bad-plugins.args: + * docs/plugins/gst-plugins-bad-plugins.hierarchy: + * docs/plugins/gst-plugins-bad-plugins.interfaces: + * docs/plugins/gst-plugins-bad-plugins.prerequisites: + * docs/plugins/inspect/plugin-jack.xml + Add new element to docs. + * ext/jack/gstjack.h + Add missing file. + * ext/jack/gstjackaudiosrc.c: + * ext/jack/gstjackaudiosrc.h: + Rename jackaudiosrc to jack_audio_src. + +2008-08-07 08:47:40 +0000 Tristan Matthews + + ext/jack/: Add a jackaudiosrc. Refactor sink slightly for better code reuse. + Original commit message from CVS: + patch by: Tristan Matthews + * ext/jack/Makefile.am: + * ext/jack/gstjack.c: + * ext/jack/gstjackaudioclient.c: + * ext/jack/gstjackaudiosink.c: + * ext/jack/gstjackaudiosink.h: + * ext/jack/gstjackaudiosrc.c: + * ext/jack/gstjackaudiosrc.h: + * ext/jack/gstjackringbuffer.h: + Add a jackaudiosrc. Refactor sink slightly for better code reuse. + Fixes #545197. + +2008-06-13 11:59:23 +0000 Stefan Kost + + docs/plugins/: docs/plugins/inspect/plugin-mythtv.xml + Original commit message from CVS: + * docs/plugins/Makefile.am: + * docs/plugins/gst-plugins-bad-plugins-docs.sgml: + * docs/plugins/gst-plugins-bad-plugins-sections.txt: + * docs/plugins/gst-plugins-bad-plugins.args: + * docs/plugins/gst-plugins-bad-plugins.hierarchy: + * docs/plugins/gst-plugins-bad-plugins.interfaces: + * docs/plugins/gst-plugins-bad-plugins.prerequisites: + * docs/plugins/gst-plugins-bad-plugins.signals: + * docs/plugins/inspect/plugin-alsaspdif.xml: + * docs/plugins/inspect/plugin-amrwb.xml: + * docs/plugins/inspect/plugin-app.xml: + * docs/plugins/inspect/plugin-bayer.xml: + * docs/plugins/inspect/plugin-bz2.xml: + * docs/plugins/inspect/plugin-cdaudio.xml: + * docs/plugins/inspect/plugin-cdxaparse.xml: + * docs/plugins/inspect/plugin-dtsdec.xml: + * docs/plugins/inspect/plugin-dvb.xml: + * docs/plugins/inspect/plugin-dvdspu.xml: + * docs/plugins/inspect/plugin-faac.xml: + * docs/plugins/inspect/plugin-faad.xml: + * docs/plugins/inspect/plugin-fbdevsink.xml: + * docs/plugins/inspect/plugin-festival.xml: + * docs/plugins/inspect/plugin-filter.xml: + * docs/plugins/inspect/plugin-flvdemux.xml: + * docs/plugins/inspect/plugin-freeze.xml: + * docs/plugins/inspect/plugin-gsm.xml: + * docs/plugins/inspect/plugin-gstinterlace.xml: + * docs/plugins/inspect/plugin-gstrtpmanager.xml: + * docs/plugins/inspect/plugin-h264parse.xml: + * docs/plugins/inspect/plugin-interleave.xml: + * docs/plugins/inspect/plugin-jack.xml: + * docs/plugins/inspect/plugin-ladspa.xml: + * docs/plugins/inspect/plugin-metadata.xml: + * docs/plugins/inspect/plugin-mms.xml: + * docs/plugins/inspect/plugin-modplug.xml: + * docs/plugins/inspect/plugin-mpeg2enc.xml: + * docs/plugins/inspect/plugin-mpeg4videoparse.xml: + * docs/plugins/inspect/plugin-mpegtsparse.xml: + * docs/plugins/inspect/plugin-mpegvideoparse.xml: + * docs/plugins/inspect/plugin-musepack.xml: + * docs/plugins/inspect/plugin-musicbrainz.xml: + * docs/plugins/inspect/plugin-mve.xml: + * docs/plugins/inspect/plugin-mythtv.xml + * docs/plugins/inspect/plugin-nas.xml: + * docs/plugins/inspect/plugin-neon.xml: + * docs/plugins/inspect/plugin-nsfdec.xml: + * docs/plugins/inspect/plugin-nuvdemux.xml: + * docs/plugins/inspect/plugin-oss4.xml + * docs/plugins/inspect/plugin-rawparse.xml: + * docs/plugins/inspect/plugin-real.xml: + * docs/plugins/inspect/plugin-replaygain.xml: + * docs/plugins/inspect/plugin-rfbsrc.xml: + * docs/plugins/inspect/plugin-sdl.xml: + * docs/plugins/inspect/plugin-sdp.xml: + * docs/plugins/inspect/plugin-selector.xml: + * docs/plugins/inspect/plugin-sndfile.xml: + * docs/plugins/inspect/plugin-soundtouch.xml: + * docs/plugins/inspect/plugin-spcdec.xml: + * docs/plugins/inspect/plugin-speed.xml: + * docs/plugins/inspect/plugin-speexresample.xml: + * docs/plugins/inspect/plugin-stereo.xml: + * docs/plugins/inspect/plugin-subenc.xml + * docs/plugins/inspect/plugin-timidity.xml: + * docs/plugins/inspect/plugin-tta.xml: + * docs/plugins/inspect/plugin-vcdsrc.xml: + * docs/plugins/inspect/plugin-videosignal.xml: + * docs/plugins/inspect/plugin-vmnc.xml: + * docs/plugins/inspect/plugin-wildmidi.xml: + * docs/plugins/inspect/plugin-x264.xml: + * docs/plugins/inspect/plugin-xvid.xml: + * docs/plugins/inspect/plugin-y4menc.xml: + * ext/amrwb/gstamrwbdec.c: + * ext/amrwb/gstamrwbenc.c: + * ext/amrwb/gstamrwbparse.c: + * ext/dc1394/gstdc1394.c: + * ext/directfb/dfbvideosink.c: + * ext/ivorbis/vorbisdec.c: + * ext/jack/gstjackaudiosink.c: + * ext/mpeg2enc/gstmpeg2enc.cc: + * ext/mplex/gstmplex.cc: + * ext/musicbrainz/gsttrm.c: + * ext/mythtv/gstmythtvsrc.c: + * ext/theora/theoradec.c: + * ext/timidity/gsttimidity.c: + * ext/timidity/gstwildmidi.c: + * gst-libs/gst/app/gstappsink.c: + * gst/deinterlace/gstdeinterlace.c: + * gst/dvdspu/gstdvdspu.c: + * gst/festival/gstfestival.c: + * gst/freeze/gstfreeze.c: + * gst/interleave/deinterleave.c: + * gst/interleave/interleave.c: + * gst/modplug/gstmodplug.cc: + * gst/nuvdemux/gstnuvdemux.c: + Add missing elements to docs. Fix doc-markup: use convinience syntax + for examples (produces valid docbook), add several refsec2 when we + have several titles. Fix some types. + +2008-06-12 14:49:18 +0000 Stefan Kost + + Do not use short_description in section docs for elements. We extract them from element details and there will be war... + Original commit message from CVS: + * ext/dc1394/gstdc1394.c: + * ext/ivorbis/vorbisdec.c: + * ext/jack/gstjackaudiosink.c: + * ext/metadata/gstmetadatademux.c: + * ext/mythtv/gstmythtvsrc.c: + * ext/theora/theoradec.c: + * gst-libs/gst/app/gstappsink.c: + * gst/bayer/gstbayer2rgb.c: + * gst/deinterlace/gstdeinterlace.c: + * gst/rawparse/gstaudioparse.c: + * gst/rawparse/gstvideoparse.c: + * gst/rtpmanager/gstrtpbin.c: + * gst/rtpmanager/gstrtpclient.c: + * gst/rtpmanager/gstrtpjitterbuffer.c: + * gst/rtpmanager/gstrtpptdemux.c: + * gst/rtpmanager/gstrtpsession.c: + * gst/rtpmanager/gstrtpssrcdemux.c: + * gst/selector/gstinputselector.c: + * gst/selector/gstoutputselector.c: + * gst/videosignal/gstvideoanalyse.c: + * gst/videosignal/gstvideodetect.c: + * gst/videosignal/gstvideomark.c: + * sys/oss4/oss4-mixer.c: + * sys/oss4/oss4-sink.c: + * sys/oss4/oss4-source.c: + Do not use short_description in section docs for elements. We extract + them from element details and there will be warnings if they differ. + Also fixing up the ChangeLog order. + +2008-05-26 17:52:21 +0000 Wim Taymans + + ext/jack/gstjackaudiosink.c: Include the element name in the port name to avoid duplicate port names. + Original commit message from CVS: + * ext/jack/gstjackaudiosink.c: + (gst_jack_audio_sink_allocate_channels): + Include the element name in the port name to avoid duplicate port names. + +2008-04-06 20:18:16 +0000 Tim-Philipp Müller + + ext/jack/gstjackaudiosink.c: Work around missing bits of thread-safety on older GLibs some more to avoid assertions w... + Original commit message from CVS: + * ext/jack/gstjackaudiosink.c: (gst_jack_audio_sink_class_init): + Work around missing bits of thread-safety on older GLibs some + more to avoid assertions when starting up multiple playbin + objects concurrently (see #512382). + +2008-03-13 14:25:20 +0000 Sebastian Dröge + + Use GST_LICENSE, GST_PACKAGE_NAME and GST_PACKAGE_ORIGIN instead of hardcoding values where possible. Fixes bug #522212. + Original commit message from CVS: + * ext/alsaspdif/alsaspdifsink.c: + * ext/gsm/gstgsm.c: + * ext/jack/gstjack.c: + * ext/libmms/gstmms.c: + * ext/neon/gstneonhttpsrc.c: + * ext/shout/gstshout.c: + * ext/timidity/gsttimidity.c: + * ext/timidity/gstwildmidi.c: + * gst/nuvdemux/gstnuvdemux.c: + * gst/tta/gsttta.c: + Use GST_LICENSE, GST_PACKAGE_NAME and GST_PACKAGE_ORIGIN instead + of hardcoding values where possible. Fixes bug #522212. + +2007-07-18 07:42:47 +0000 Stefan Kost + + ext/jack/gstjackaudiosink.c: Add stdlib include here too. + Original commit message from CVS: + * ext/jack/gstjackaudiosink.c: (gst_jack_ring_buffer_open_device), + (gst_jack_ring_buffer_acquire): + Add stdlib include here too. + +2007-04-04 07:36:28 +0000 Stefan Kost + + ext/jack/gstjackaudiosink.c: Try t better name clients. properly handle return codes when re- establishing links. + Original commit message from CVS: + * ext/jack/gstjackaudiosink.c: (gst_jack_ring_buffer_open_device), + (gst_jack_ring_buffer_acquire): + Try t better name clients. properly handle return codes when re- + establishing links. + +2007-03-18 17:57:48 +0000 Paul Davis + + ext/jack/gstjackaudioclient.c: Don't need to take the connection lock, it will not be used and could cause deadlocks. + Original commit message from CVS: + Based on patch by: Paul Davis + * ext/jack/gstjackaudioclient.c: (gst_jack_audio_unref_connection): + Don't need to take the connection lock, it will not be used and could + cause deadlocks. + +2007-03-08 15:24:52 +0000 Paul Davis + + ext/jack/: Make an object to manage client connections to the jack server which we will use in the future to run sele... + Original commit message from CVS: + Includes patch by: Paul Davis + * ext/jack/Makefile.am: + * ext/jack/gstjackaudioclient.c: (gst_jack_audio_client_init), + (jack_process_cb), (jack_sample_rate_cb), (jack_buffer_size_cb), + (jack_shutdown_cb), (connection_find), + (gst_jack_audio_make_connection), (gst_jack_audio_get_connection), + (gst_jack_audio_unref_connection), + (gst_jack_audio_connection_add_client), + (gst_jack_audio_connection_remove_client), + (gst_jack_audio_client_new), (gst_jack_audio_client_free), + (gst_jack_audio_client_get_client), + (gst_jack_audio_client_set_active): + * ext/jack/gstjackaudioclient.h: + Make an object to manage client connections to the jack server which we + will use in the future to run selected jack elements with the same jack + connection. + Make some stuff a bit more threadsafe. + Activate the jack client ASAP. + * ext/jack/gstjackaudiosink.c: + (gst_jack_audio_sink_allocate_channels), + (gst_jack_audio_sink_free_channels), (jack_process_cb), + (gst_jack_ring_buffer_open_device), + (gst_jack_ring_buffer_close_device), + (gst_jack_ring_buffer_acquire), (gst_jack_ring_buffer_release), + (gst_jack_audio_sink_class_init), (gst_jack_audio_sink_init), + (gst_jack_audio_sink_getcaps): + * ext/jack/gstjackaudiosink.h: + Use new client object to manage connections. + Don't remove and recreate all ports, try to reuse them. + +2007-01-12 10:25:40 +0000 Wim Taymans + + ext/jack/gstjackaudiosink.*: Improve docs. + Original commit message from CVS: + * ext/jack/gstjackaudiosink.c: (jack_sample_rate_cb), + (jack_buffer_size_cb), (jack_shutdown_cb), + (gst_jack_ring_buffer_acquire): + * ext/jack/gstjackaudiosink.h: + Improve docs. + +2006-12-06 16:57:17 +0000 Jan Schmidt + + ext/jack/.cvsignore: Ignore old files as requested by the build slave. + Original commit message from CVS: + * ext/jack/.cvsignore: + Ignore old files as requested by the build slave. + +2006-11-30 11:59:04 +0000 Wim Taymans + + ext/Makefile.am: Fix build. + Original commit message from CVS: + * ext/Makefile.am: + Fix build. + * ext/jack/gstjackaudiosink.c: (jack_process_cb), + (jack_sample_rate_cb), (jack_buffer_size_cb), (jack_shutdown_cb), + (gst_jack_ring_buffer_acquire): + Small cleanups. + +2006-11-30 11:49:36 +0000 Wim Taymans + + Added fully functional jackaudiosink. + Original commit message from CVS: + * configure.ac: + * ext/Makefile.am: + * ext/jack/Makefile.am: + * ext/jack/gstjack.c: (plugin_init): + * ext/jack/gstjack.h: + * ext/jack/gstjackaudiosink.c: (gst_jack_ring_buffer_get_type), + (gst_jack_ring_buffer_class_init), (jack_process_cb), + (jack_sample_rate_cb), (jack_buffer_size_cb), (jack_shutdown_cb), + (gst_jack_ring_buffer_init), (gst_jack_ring_buffer_dispose), + (gst_jack_ring_buffer_finalize), + (gst_jack_ring_buffer_open_device), + (gst_jack_ring_buffer_close_device), + (gst_jack_ring_buffer_acquire), (gst_jack_ring_buffer_release), + (gst_jack_ring_buffer_start), (gst_jack_ring_buffer_pause), + (gst_jack_ring_buffer_stop), (gst_jack_ring_buffer_delay), + (gst_jack_connect_get_type), (gst_jack_audio_sink_base_init), + (gst_jack_audio_sink_class_init), (gst_jack_audio_sink_init), + (gst_jack_audio_sink_set_property), + (gst_jack_audio_sink_get_property), (gst_jack_audio_sink_getcaps), + (gst_jack_audio_sink_create_ringbuffer): + * ext/jack/gstjackaudiosink.h: + Added fully functional jackaudiosink. + +2006-04-08 21:48:01 +0000 Stefan Kost + + Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent) + Original commit message from CVS: + * ext/amrwb/gstamrwbdec.c: (gst_amrwbdec_class_init): + * ext/amrwb/gstamrwbenc.c: (gst_amrwbenc_class_init): + * ext/amrwb/gstamrwbparse.c: (gst_amrwbparse_class_init): + * ext/arts/gst_arts.c: (gst_arts_class_init): + * ext/artsd/gstartsdsink.c: (gst_artsdsink_class_init): + * ext/audiofile/gstafsink.c: (gst_afsink_class_init): + * ext/audiofile/gstafsrc.c: (gst_afsrc_class_init): + * ext/audioresample/gstaudioresample.c: + * ext/cdaudio/gstcdaudio.c: (gst_cdaudio_class_init): + * ext/directfb/dfbvideosink.c: (gst_dfbvideosink_class_init): + * ext/divx/gstdivxdec.c: (gst_divxdec_class_init): + * ext/hermes/gsthermescolorspace.c: + (gst_hermes_colorspace_class_init): + * ext/ivorbis/vorbisfile.c: (gst_ivorbisfile_class_init): + * ext/jack/gstjack.c: (gst_jack_class_init): + * ext/jack/gstjackbin.c: (gst_jack_bin_class_init): + * ext/lcs/gstcolorspace.c: (gst_colorspace_class_init): + * ext/libfame/gstlibfame.c: (gst_fameenc_class_init): + * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_class_init): + * ext/nas/nassink.c: (gst_nassink_class_init): + * ext/shout/gstshout.c: (gst_icecastsend_class_init): + * ext/snapshot/gstsnapshot.c: (gst_snapshot_class_init): + * ext/sndfile/gstsf.c: (gst_sf_class_init): + * ext/swfdec/gstswfdec.c: (gst_swfdecbuffer_class_init), + (gst_swfdec_class_init): + * ext/tarkin/gsttarkindec.c: (gst_tarkindec_class_init): + * ext/tarkin/gsttarkinenc.c: (gst_tarkinenc_class_init): + * gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_class_init): + * gst/chart/gstchart.c: (gst_chart_class_init): + * gst/colorspace/gstcolorspace.c: (gst_colorspace_class_init): + * gst/deinterlace/gstdeinterlace.c: (gst_deinterlace_class_init): + * gst/festival/gstfestival.c: (gst_festival_class_init): + * gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init): + * gst/filter/gstiir.c: (gst_iir_class_init): + * gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init): + * gst/librfb/gstrfbsrc.c: (gst_rfbsrc_class_init): + * gst/mixmatrix/mixmatrix.c: (gst_mixmatrix_class_init): + * gst/mpeg1sys/gstmpeg1systemencode.c: + (gst_system_encode_class_init): + * gst/mpeg1videoparse/gstmp1videoparse.c: + (gst_mp1videoparse_class_init): + * gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_class_init): + * gst/mpegaudioparse/gstmpegaudioparse.c: + (gst_mp3parse_class_init): + * gst/overlay/gstoverlay.c: (gst_overlay_class_init): + * gst/passthrough/gstpassthrough.c: (passthrough_class_init): + * gst/playondemand/gstplayondemand.c: (play_on_demand_class_init): + * gst/rtjpeg/gstrtjpegdec.c: (gst_rtjpegdec_class_init): + * gst/rtjpeg/gstrtjpegenc.c: (gst_rtjpegenc_class_init): + * gst/smooth/gstsmooth.c: (gst_smooth_class_init): + * gst/smoothwave/gstsmoothwave.c: (gst_smoothwave_class_init): + * gst/spectrum/gstspectrum.c: (gst_spectrum_class_init): + * gst/stereo/gststereo.c: (gst_stereo_class_init): + * gst/switch/gstswitch.c: (gst_switch_class_init): + * gst/tta/gstttadec.c: (gst_tta_dec_class_init): + * gst/tta/gstttaparse.c: (gst_tta_parse_class_init): + * gst/vbidec/gstvbidec.c: (gst_vbidec_class_init): + * gst/videocrop/gstvideocrop.c: (gst_video_crop_class_init): + * gst/virtualdub/gstxsharpen.c: (gst_xsharpen_class_init): + * gst/y4m/gsty4mencode.c: (gst_y4mencode_class_init): + * sys/cdrom/gstcdplayer.c: (cdplayer_class_init): + * sys/directsound/gstdirectsoundsink.c: + (gst_directsoundsink_class_init): + * sys/dxr3/dxr3audiosink.c: (dxr3audiosink_class_init): + * sys/dxr3/dxr3spusink.c: (dxr3spusink_class_init): + * sys/dxr3/dxr3videosink.c: (dxr3videosink_class_init): + * sys/qcam/gstqcamsrc.c: (gst_qcamsrc_class_init): + * sys/v4l2/gstv4l2colorbalance.c: + (gst_v4l2_color_balance_channel_class_init): + * sys/v4l2/gstv4l2tuner.c: (gst_v4l2_tuner_channel_class_init), + (gst_v4l2_tuner_norm_class_init): + * sys/ximagesrc/ximagesrc.c: (gst_ximagesrc_class_init): + Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent) + +2006-04-01 10:09:11 +0000 Thomas Vander Stichele + + * ext/jack/gstjack.c: + rework build; add translations for v4l2 + Original commit message from CVS: + rework build; add translations for v4l2 + +2005-10-12 14:29:55 +0000 Stefan Kost + + renamed GST_FLAGS macros to GST_OBJECT_FLAGS moved bitshift from macro to enum definition + Original commit message from CVS: + * examples/indexing/indexmpeg.c: (main): + * ext/artsd/gstartsdsink.c: (gst_artsdsink_open_audio), + (gst_artsdsink_close_audio), (gst_artsdsink_change_state): + * ext/artsd/gstartsdsink.h: + * ext/audiofile/gstafparse.c: (gst_afparse_open_file), + (gst_afparse_close_file): + * ext/audiofile/gstafparse.h: + * ext/audiofile/gstafsink.c: (gst_afsink_open_file), + (gst_afsink_close_file), (gst_afsink_chain), + (gst_afsink_change_state): + * ext/audiofile/gstafsink.h: + * ext/audiofile/gstafsrc.c: (gst_afsrc_open_file), + (gst_afsrc_close_file), (gst_afsrc_change_state): + * ext/audiofile/gstafsrc.h: + * ext/cdaudio/gstcdaudio.c: (gst_cdaudio_init): + * ext/directfb/directfbvideosink.c: (gst_directfbvideosink_init): + * ext/dts/gstdtsdec.c: (gst_dtsdec_init): + * ext/jack/gstjack.h: + * ext/jack/gstjackbin.c: (gst_jack_bin_init), + (gst_jack_bin_change_state): + * ext/musepack/gstmusepackdec.c: (gst_musepackdec_init): + * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_init): + * ext/nas/nassink.c: (gst_nassink_open_audio), + (gst_nassink_close_audio), (gst_nassink_change_state): + * ext/nas/nassink.h: + * ext/polyp/polypsink.c: (gst_polypsink_init): + * ext/sdl/sdlvideosink.c: (gst_sdlvideosink_change_state): + * ext/sdl/sdlvideosink.h: + * ext/smoothwave/gstsmoothwave.c: (gst_smoothwave_init): + * ext/sndfile/gstsf.c: (gst_sf_set_property), + (gst_sf_change_state), (gst_sf_release_request_pad), + (gst_sf_open_file), (gst_sf_close_file), (gst_sf_loop): + * ext/sndfile/gstsf.h: + * ext/swfdec/gstswfdec.c: (gst_swfdec_init): + * ext/tarkin/gsttarkindec.c: (gst_tarkindec_init): + * gst/apetag/apedemux.c: (gst_ape_demux_init): + * gst/cdxaparse/gstcdxaparse.c: (gst_cdxaparse_init): + * gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_init): + * gst/festival/gstfestival.c: (gst_festival_change_state): + * gst/festival/gstfestival.h: + * gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_init): + * gst/multifilesink/gstmultifilesink.c: (gst_multifilesink_init), + (gst_multifilesink_set_location), (gst_multifilesink_open_file), + (gst_multifilesink_close_file), (gst_multifilesink_next_file), + (gst_multifilesink_pad_query), (gst_multifilesink_handle_event), + (gst_multifilesink_chain), (gst_multifilesink_change_state): + * gst/multifilesink/gstmultifilesink.h: + * gst/videodrop/gstvideodrop.c: (gst_videodrop_init): + * sys/cdrom/gstcdplayer.c: (cdplayer_init): + * sys/dxr3/dxr3audiosink.c: (dxr3audiosink_init), + (dxr3audiosink_open), (dxr3audiosink_close), + (dxr3audiosink_chain_pcm), (dxr3audiosink_chain_ac3), + (dxr3audiosink_change_state): + * sys/dxr3/dxr3audiosink.h: + * sys/dxr3/dxr3spusink.c: (dxr3spusink_init), (dxr3spusink_open), + (dxr3spusink_close), (dxr3spusink_chain), + (dxr3spusink_change_state): + * sys/dxr3/dxr3spusink.h: + * sys/dxr3/dxr3videosink.c: (dxr3videosink_init), + (dxr3videosink_open), (dxr3videosink_close), + (dxr3videosink_write_data), (dxr3videosink_change_state): + * sys/dxr3/dxr3videosink.h: + * sys/glsink/glimagesink.c: (gst_glimagesink_init): + * sys/qcam/gstqcamsrc.c: (gst_qcamsrc_change_state), + (gst_qcamsrc_open), (gst_qcamsrc_close): + * sys/qcam/gstqcamsrc.h: + * sys/v4l2/gstv4l2src.c: (gst_v4l2src_init): + * sys/vcd/vcdsrc.c: (gst_vcdsrc_set_property), (gst_vcdsrc_get), + (gst_vcdsrc_open_file), (gst_vcdsrc_close_file), + (gst_vcdsrc_change_state), (gst_vcdsrc_recalculate): + * sys/vcd/vcdsrc.h: + renamed GST_FLAGS macros to GST_OBJECT_FLAGS + moved bitshift from macro to enum definition + +2005-09-05 17:20:29 +0000 Jan Schmidt + + * ext/jack/gstjack.c: + * ext/jack/gstjackbin.c: + Fix up all the state change functions. + Original commit message from CVS: + Fix up all the state change functions. + +2004-08-03 14:28:12 +0000 Benjamin Otte + + fixes for G_DISABLE_ASSERT and friends + Original commit message from CVS: + * examples/dynparams/filter.c: (ui_control_create): + * examples/gstplay/player.c: (print_tag): + * ext/alsa/gstalsa.c: (gst_alsa_request_new_pad): + * ext/gdk_pixbuf/gstgdkanimation.c: + (gst_gdk_animation_iter_may_advance): + * ext/jack/gstjack.c: (gst_jack_request_new_pad): + * ext/mad/gstid3tag.c: (gst_mad_id3_to_tag_list), + (tag_list_to_id3_tag_foreach), (gst_id3_tag_handle_event): + * ext/vorbis/oggvorbisenc.c: (gst_oggvorbisenc_get_tag_value): + * ext/vorbis/vorbisenc.c: (gst_vorbisenc_get_tag_value): + * ext/xine/xineaudiodec.c: (gst_xine_audio_dec_chain): + * gst-libs/gst/media-info/media-info-test.c: (print_tag): + * gst/sine/demo-dparams.c: (main): + * gst/tags/gstvorbistag.c: (gst_tag_to_vorbis_comments): + * testsuite/alsa/formats.c: (create_pipeline): + * testsuite/alsa/sinesrc.c: (sinesrc_force_caps), (sinesrc_get): + fixes for G_DISABLE_ASSERT and friends + * gst/typefind/gsttypefindfunctions.c: (aac_type_find), + (mp3_type_frame_length_from_header), (mp3_type_find), + (plugin_init): + require mp3 typefinding to have at least MIN_HEADERS valid headers + add typefinding for AAC adts files + +2004-05-21 23:28:57 +0000 Stéphane Loeuillet + + * ext/jack/gstjack.c: + * ext/jack/gstjack.h: + second batch : remove ',' at end of enums as they could confuse older gcc, foreign compilers (forte) and gtk-doc (in ... + Original commit message from CVS: + second batch : + remove ',' at end of enums as they could confuse older gcc, foreign compilers (forte) and gtk-doc + (in gst-plugins/ext/ this time) + +2004-03-15 19:32:27 +0000 Thomas Vander Stichele + + * ext/jack/gstjack.c: + * ext/jack/gstjackbin.c: + don't mix tabs and spaces + Original commit message from CVS: + don't mix tabs and spaces + +2004-03-15 16:32:54 +0000 Johan Dahlin + + *.h: Revert indenting + Original commit message from CVS: + * *.h: Revert indenting + +2004-03-14 22:34:33 +0000 Thomas Vander Stichele + + * ext/jack/gstjack.c: + * ext/jack/gstjack.h: + * ext/jack/gstjackbin.c: + gst-indent + Original commit message from CVS: + gst-indent + +2004-01-12 03:40:18 +0000 David Schleef + + * ext/jack/gstjack.c: + Remove all usage of gst_pad_get_caps(), and replace it with gst_pad_get_allowed_caps() or gst_pad_get_negotiated_cap(). + Original commit message from CVS: + Remove all usage of gst_pad_get_caps(), and replace it with + gst_pad_get_allowed_caps() or gst_pad_get_negotiated_cap(). + +2003-12-22 01:47:09 +0000 David Schleef + + * ext/jack/gstjack.c: + Merge CAPS branch + Original commit message from CVS: + Merge CAPS branch + +2003-12-13 16:59:51 +0000 Benjamin Otte + + * ext/jack/gstjackbin.c: + removed GST_*_CAST. Disabling of type checking is done in glib. + Original commit message from CVS: + removed GST_*_CAST. Disabling of type checking is done in glib. + +2003-12-04 10:37:38 +0000 Andy Wingo + + * ext/jack/gstjack.c: + remove copyright field from plugins + Original commit message from CVS: + remove copyright field from plugins + +2003-11-07 12:47:02 +0000 Ronald S. Bultje + + * ext/jack/gstjackbin.c: + Remove all config.h includes from header files, add it to each source file and remove duplicate config.h includes fro... + Original commit message from CVS: + Remove all config.h includes from header files, add it to each source file and remove duplicate config.h includes from several source files + +2003-11-01 23:43:13 +0000 Iain Holmes + + * ext/jack/gstjack.c: + Jack fixed too + Original commit message from CVS: + Jack fixed too + +2003-10-29 03:15:55 +0000 David Schleef + + * ext/jack/gstjack.h: + change gst/bytestream.h to gst/bytestream/bytestream.h + Original commit message from CVS: + change gst/bytestream.h to gst/bytestream/bytestream.h + +2003-10-28 20:52:41 +0000 Benjamin Otte + + * ext/jack/gstjack.h: + merge TYPEFIND branch. Major changes: + Original commit message from CVS: + merge TYPEFIND branch. Major changes: + - totally reworked type(find) system + - all typefind functions are in gst/typefind now + - more typefind functions then before + - some plugins might fail to compile now because I don't have them installed and they + a) require bytestream or + b) haven't had their typefind fixed. + Please fix those plugins and put the typefind functions into gst/typefind if they don't have dependencies + +2003-10-08 16:08:19 +0000 Andy Wingo + + * ext/jack/gstjack.c: + /GstBuffer/GstData/ in the API where you can pass events. Fix the plugins to deal with that. Fixes #113488. + Original commit message from CVS: + /GstBuffer/GstData/ in the API where you can pass events. Fix the plugins to deal with that. Fixes #113488. + +2003-10-01 13:14:50 +0000 Ronald S. Bultje + + * ext/jack/gstjack.h: + New typefind system: bytestream is now part of the core all plugins have been modified to use this new typefind syste... + Original commit message from CVS: + New typefind system: + * bytestream is now part of the core + * all plugins have been modified to use this new typefind system + * asf typefinding added + * mpeg video stream typefiding removed because it's broken + * duplicate typefind entries removed + * extra id3 typefinding added, because we've seen 4 types of files + (riff/wav, flac, vorbis, mp3) with id3 headers and each of these needs + to work. Instead, I've added an id3 element and let it redo typefiding + after the id3 header. this needs a hack because spider only typefinds + once. We can remove this hack once spider supports multiple typefinds. + * with all this, mp3 typefinding is semi-rewritten + * id3 typefinding in flac/vorbis is removed, it's no longer needed + * fixed spider and gst-typefind to use this, too. + * Other general cleanups + +2003-09-30 12:56:27 +0000 Andy Wingo + + * ext/jack/gstjack.c: + * ext/jack/gstjack.h: + * ext/jack/gstjackbin.c: + conform to the buffer-frames props entry -- much nicer now... + Original commit message from CVS: + conform to the buffer-frames props entry -- much nicer now... + +2003-08-10 00:01:58 +0000 David Schleef + + * ext/jack/Makefile.am: + Remove redundant plugindir definition + Original commit message from CVS: + Remove redundant plugindir definition + +2003-07-19 23:25:25 +0000 Leif Johnson + + * ext/jack/gstjack.c: + * ext/jack/gstjack.h: + + changes for new float caps without slope/intercept + some category changes for plugins + Original commit message from CVS: + + changes for new float caps without slope/intercept + + some category changes for plugins + +2003-07-06 20:49:52 +0000 Ronald S. Bultje + + * ext/jack/gstjack.c: + New mimetypes gone into effect today - this commit changes all old mimetypes over to the new mimetypes spec as descri... + Original commit message from CVS: + New mimetypes gone into effect today - this commit changes all old mimetypes over to the new mimetypes spec as described in the previous commit's document. Note: some plugins will break, some pipelines will break, expect HEAD to be broken or at least not 100% working for a few days, but don't forget to report bugs + +2003-07-01 02:27:06 +0000 David Schleef + + * ext/jack/gstjack.c: + fix type punning + Original commit message from CVS: + fix type punning + +2003-06-29 19:46:13 +0000 Benjamin Otte + + * ext/jack/gstjack.c: + * ext/jack/gstjackbin.c: + compatibility fix for new GST_DEBUG stuff. + Original commit message from CVS: + compatibility fix for new GST_DEBUG stuff. + Includes fixes for missing includes for config.h and unistd.h + I only ensured for plugins I can build that they work, so if some of them are still broken, you gotta fix them yourselves unfortunately. + +2003-06-13 21:21:17 +0000 Wim Taymans + + * ext/jack/gstjack.c: + Removed ugly caps fixed flag hack, will be done automatically in core soon + Original commit message from CVS: + Removed ugly caps fixed flag hack, will be done automatically in + core soon + +2003-03-04 15:34:20 +0000 Andy Wingo + + * ext/jack/gstjack.c: + * ext/jack/gstjack.h: + * ext/jack/gstjackbin.c: + update for the latest jack cvs and non-cothreaded gst scheduler + Original commit message from CVS: + update for the latest jack cvs and non-cothreaded gst scheduler + +2003-02-05 20:38:41 +0000 Jan Schmidt + + * ext/jack/gstjack.c: + Changed caps->fixed to use FLAG_SET + Original commit message from CVS: + Changed caps->fixed to use FLAG_SET + +2003-01-10 13:38:32 +0000 Thomas Vander Stichele + + * ext/jack/gstjack.c: + PadConnect -> PadLink + Original commit message from CVS: + PadConnect -> PadLink + +2003-01-10 10:22:25 +0000 Thomas Vander Stichele + + * ext/jack/gstjack.c: + another batch of connect->link fixes please let me know about issues and please refrain of making them yourself, so t... + Original commit message from CVS: + another batch of connect->link fixes + please let me know about issues + and please refrain of making them yourself, so that I don't spend double + the time resolving conflicts + +2002-12-08 14:50:10 +0000 Thomas Vander Stichele + + * ext/jack/Makefile.am: + parallel install fixes + Original commit message from CVS: + parallel install fixes + +2002-09-29 18:12:18 +0000 Andy Wingo + + * ext/jack/gstjack.c: + * ext/jack/gstjackbin.c: + licenses again + Original commit message from CVS: + licenses again + +2002-09-18 19:02:52 +0000 Christian Schaller + + * ext/jack/gstjack.c: + plugins part of license field patch + Original commit message from CVS: + plugins part of license field patch + +2002-09-10 09:31:40 +0000 Ronald S. Bultje + + * ext/jack/gstjack.c: + This updates all plugins to the new API for gst_pad_try_set_caps + Original commit message from CVS: + This updates all plugins to the new API for gst_pad_try_set_caps + +2002-09-09 23:27:38 +0000 Thomas Vander Stichele + + * ext/jack/gstjack.c: + removing warnings as approved by wim + Original commit message from CVS: + removing warnings as approved by wim + +2002-08-23 04:04:11 +0000 Andy Wingo + + * ext/jack/gstjack.c: + * ext/jack/gstjackbin.c: + fix jack input port connection + Original commit message from CVS: + fix jack input port connection + +2002-07-09 17:39:17 +0000 Andy Wingo + + * ext/jack/gstjack.c: + compile fixen, and prepare to move MAINTAINER_MODE to as-version.m4 + Original commit message from CVS: + compile fixen, and prepare to move MAINTAINER_MODE to as-version.m4 + +2002-07-02 23:35:07 +0000 Andy Wingo + + * ext/jack/gstjack.c: + * ext/jack/gstjackbin.c: + make jack work in all its full duplex glory + Original commit message from CVS: + make jack work in all its full duplex glory + +2002-06-12 03:32:02 +0000 Andy Wingo + + * ext/jack/gstjack.c: + * ext/jack/gstjackbin.c: + working jack elements (fixed a problem in upstream jack) random other fixen... + Original commit message from CVS: + * working jack elements (fixed a problem in upstream jack) + * random other fixen... + +2002-05-15 19:08:49 +0000 Steve Baker + + * ext/jack/gstjack.c: + use new bytestream api + Original commit message from CVS: + use new bytestream api + +2002-05-13 18:08:33 +0000 Andy Wingo + + * ext/jack/gstjack.c: + * ext/jack/gstjack.h: + * ext/jack/gstjackbin.c: + update to new jack api + Original commit message from CVS: + update to new jack api + +2002-05-05 19:39:17 +0000 Andy Wingo + + * ext/jack/gstjack.c: + add some includes + Original commit message from CVS: + add some includes + +2002-05-05 01:08:05 +0000 Andy Wingo + + * ext/jack/gstjack.c: + * ext/jack/gstjack.h: + * ext/jack/gstjackbin.c: + better initialization. it doesn't work over here, though. + Original commit message from CVS: + better initialization. it doesn't work over here, though. + +2002-05-04 21:38:56 +0000 Andy Wingo + + * ext/jack/gstjackbin.c: + a commit so that jack will build without errors on Uraeus's system ;) + Original commit message from CVS: + a commit so that jack will build without errors on Uraeus's system ;) + +2002-05-04 20:53:35 +0000 Andy Wingo + + * ext/jack/gstjack.c: + set caps once we know the sample rate of the system + Original commit message from CVS: + set caps once we know the sample rate of the system + +2002-05-04 18:57:44 +0000 Andy Wingo + + * ext/jack/gstjack.c: + * ext/jack/gstjack.h: + * ext/jack/gstjackbin.c: + some jack fixes, alsa touchups, and add rtp by default to the build if there are any problems building rtp, we're mov... + Original commit message from CVS: + some jack fixes, alsa touchups, and add rtp by default to the build + if there are any problems building rtp, we're moving it back to experimental ;) + +2002-04-20 21:42:51 +0000 Andy Wingo + + * ext/jack/gstjack.c: + a hack to work around intltool's brokenness a current check for mpeg2dec details->klass reorganizations an element br... + Original commit message from CVS: + * a hack to work around intltool's brokenness + * a current check for mpeg2dec + * details->klass reorganizations + * an element browser that uses details->klass + * separated cdxa parse out from the avi directory + +2002-04-16 17:14:05 +0000 Andy Wingo + + * ext/jack/Makefile.am: + * ext/jack/gstjack.c: + * ext/jack/gstjack.h: + * ext/jack/gstjackbin.c: + Finally we're on to a proper jack setup, with a specialized bin and elements that can only go in a jack bin. I had to... + Original commit message from CVS: + Finally we're on to a proper jack setup, with a specialized bin and elements + that can only go in a jack bin. I had to fix the parser first to do this, but + to run it, the syntax is like so: + gst-launch jackbin.( filesrc ! mad ! jacksink ) + But of course it's not fully functional yet. Sigh. + +2002-04-11 20:42:26 +0000 Andy Wingo + + * ext/jack/gstjack.c: + GstPadTemplate <-> gst_pad_template <-> GST_PAD_TEMPLATE same with *factory and typefind. + Original commit message from CVS: + GstPadTemplate <-> gst_pad_template <-> GST_PAD_TEMPLATE + same with *factory and typefind. + also, some -Werror fixes. + +2002-03-30 21:07:51 +0000 Andy Wingo + + * ext/jack/gstjack.c: + alphabetization fixen a jack caps fix + Original commit message from CVS: + * alphabetization fixen + * a jack caps fix + +2002-03-30 19:31:13 +0000 Andy Wingo + + * ext/jack/gstjack.c: + add notify back to filesrc, it's needed for MVC applications remove notify printouts from gst-launch cleanup in gst-p... + Original commit message from CVS: + * add notify back to filesrc, it's needed for MVC applications + * remove notify printouts from gst-launch + * cleanup in gst-plugins configure.ac + * some jack updates + * remove SELF_ITERATING flag in favor of SEF_SCHEDULABLE (not a clear name, + but it's what we have for the moment) + * improve parsing of request pad names, no more sscanf + * fixes to the fastscheduler Makefile.am + +2002-03-20 21:45:04 +0000 Andy Wingo + + * ext/jack/gstjack.c: + s/Gnome-Streamer/GStreamer/ + Original commit message from CVS: + s/Gnome-Streamer/GStreamer/ + +2002-03-19 04:10:06 +0000 Andy Wingo + + * ext/jack/Makefile.am: + * ext/jack/gstjack.c: + removal of //-style comments don't link plugins to core libs -- the versioning is done internally to the plugins with... + Original commit message from CVS: + * removal of //-style comments + * don't link plugins to core libs -- the versioning is done internally to the plugins with the plugin_info struct, + and symbol resolution is lazy, so we can always know if a plugin can be loaded by the plugin_info data. in theory. + +2002-03-19 01:39:43 +0000 Andy Wingo + + * ext/jack/Makefile.am: + s/@GST_PLUGIN_LDFLAGS@/$(GST_PLUGIN_LDFLAGS)/ @-substitued variables variables are defined as make variables automagi... + Original commit message from CVS: + s/@GST_PLUGIN_LDFLAGS@/$(GST_PLUGIN_LDFLAGS)/ + @-substitued variables variables are defined as make variables automagically, + and this gives the user the freedom to say make GST_PLUGIN_LDFLAGS=-myflag + +2002-03-18 04:41:35 +0000 Andy Wingo + + * ext/jack/Makefile.am: + * ext/jack/README: + * ext/jack/gstjack.c: + * ext/jack/gstjack.h: + s/gst_element_install_std_props/gst_element_class_install_std_props/ -- it just makes more sense that way added jack ... + Original commit message from CVS: + * s/gst_element_install_std_props/gst_element_class_install_std_props/ -- it just makes more sense that way + * added jack element, doesn't quite work right yet but i didn't want to lose the work -- it does build, register, + and attempt to run though + * imposed some restrictions on the naming of request pads to better allow for reverse parsing + * added '%s' to reverse parsing + * added new bin flag to indicate that it is self-iterating, and some lame code in gst-launch to test it out + * fixen on launch-gui + * added pkg-config stuff for the editor's libs + +2011-01-02 11:37:14 +0000 Tim-Philipp Müller + + * sys/v4l2/Makefile.am: + * sys/v4l2/gstv4l2.c: + * sys/v4l2/gstv4l2bufferpool.c: + * sys/v4l2/v4l2_calls.c: + v4l2: mark v4l2sink as experimental and build only if --enable-experimental is passed + It's not really of 'good' quality yet, but there's a lot of + code shared with v4l2src, so not so easy to move it elswhere. + https://bugzilla.gnome.org/show_bug.cgi?id=612244 + +2011-01-02 01:24:21 +0000 Tim-Philipp Müller + + * sys/v4l2/gstv4l2object.c: + * sys/v4l2/gstv4l2object.h: + * sys/v4l2/gstv4l2sink.c: + * sys/v4l2/gstv4l2tuner.c: + * sys/v4l2/gstv4l2tuner.h: + * sys/v4l2/v4l2_calls.c: + Revert "v4l2: add norm property" + This reverts commit 9e1d419d07337e6db2cc3936472be205ce927e54. + Reverting this since it adds unreviewed and bad API to v4l2src + (property of type enum, with seemingly random and unsorted values). + +2011-01-01 23:26:33 +0000 Tim-Philipp Müller + + * tools/.gitignore: + * tools/Makefile.am: + * tools/README.filterstamp: + * tools/filterstamp.sh: + * tools/gst-launch-ext-m.m: + * tools/gst-launch-ext.1.in: + * tools/gst-visualise-m.m: + * tools/gst-visualise.1.in: + tools: remove unused left-over directory + These are all in -base/tools. + +2010-12-31 13:57:05 +0100 Wim Taymans + + * gst/rtp/gstrtpmp4adepay.c: + * gst/rtp/gstrtpmp4adepay.h: + mp4adepay: improve timestamps on outgoing packets + Improve parsing of the samplerate. + Parse the framelen so that we can calculate timestamps. + When interpollate the incomming timestamp on outgoing buffers when there are + multiple subframes. + fixes #625825 + +2010-12-31 02:16:54 +0000 Tim-Philipp Müller + + * ext/cairo/gsttimeoverlay.c: + * gst/videofilter/gstvideobalance.c: + cairo, videofilter: use gst/math-compat.h header for rint + +2010-12-30 14:30:27 -0800 David Schleef + + * gst/videofilter/gstvideobalance.c: + videobalance: Check for HAVE_RINT instead + Also change M_PI to G_PI for giggles. + +2010-12-30 14:21:37 -0800 David Schleef + + * ext/cairo/gstcairorender.c: + cairo: Don't use #ifdefs inside macros + +2010-12-30 14:20:52 -0800 David Schleef + + * gst/audiofx/audiochebband.c: + * gst/audiofx/audiocheblimit.c: + * gst/audiofx/audiokaraoke.c: + * gst/audiofx/audiowsincband.c: + * gst/audiofx/audiowsinclimit.c: + * gst/effectv/gstop.c: + * gst/equalizer/gstiirequalizer.c: + * gst/goom/convolve_fx.c: + * gst/goom/ifs.c: + * gst/goom/lines.c: + * gst/goom/tentacle3d.c: + * tests/examples/audiofx/firfilter-example.c: + * tests/examples/audiofx/iirfilter-example.c: + Change M_PI to G_PI + +2010-12-30 12:07:52 -0800 David Schleef + + * gst/videofilter/gstvideobalance.c: + videobalance: use G_OS_WIN32 for windows check + +2010-12-30 16:24:16 +0100 Wim Taymans + + * gst/rtp/gstrtpmp4adepay.c: + mp4adepay: fix timestamps on buffers + +2010-12-30 16:22:48 +0100 Wim Taymans + + * gst/rtp/gstrtpmpvpay.c: + mpvpay: fix flushing and discont + Fix flushing and disconts. + Clean up in state changes. + +2010-12-29 23:38:18 +0000 Tim-Philipp Müller + + * gst/matroska/matroska-demux.c: + matroska-demux: increase allowed max. block size for push mode from 10M to 15M + It was an arbitrary limit from the start, meant as a basic sanity check, + so may just as well increase it a little. Would be good to provide + progress reporting while completing the block in any case.. + https://bugzilla.gnome.org/show_bug.cgi?id=637060 + +2010-12-29 23:09:04 +0000 Tim-Philipp Müller + + * gst/matroska/matroska-demux.c: + matroska-demux: assume matroska if no doctype is specified + https://bugzilla.gnome.org/show_bug.cgi?id=638019 + +2010-12-04 13:43:11 -0600 Rob Clark + + * sys/v4l2/gstv4l2object.c: + * sys/v4l2/gstv4l2object.h: + * sys/v4l2/gstv4l2sink.c: + * sys/v4l2/gstv4l2src.c: + * sys/v4l2/v4l2src_calls.c: + * sys/v4l2/v4l2src_calls.h: + v4l2: add interlaced support + +2010-10-02 14:45:14 -0500 Rob Clark + + * sys/v4l2/gstv4l2sink.c: + * sys/v4l2/gstv4l2sink.h: + * sys/v4l2/gstv4l2xoverlay.c: + * sys/v4l2/gstv4l2xoverlay.h: + v4l2sink: add navigation support + +2010-04-04 06:43:41 -0500 Rob Clark + + * sys/v4l2/gstv4l2object.c: + * sys/v4l2/gstv4l2object.h: + * sys/v4l2/gstv4l2sink.c: + * sys/v4l2/gstv4l2tuner.c: + * sys/v4l2/gstv4l2tuner.h: + * sys/v4l2/v4l2_calls.c: + v4l2: add norm property + Based on a patch by Guennadi Liakhovetski. + +2010-07-13 10:03:51 -0500 Rob Clark + + * sys/v4l2/gstv4l2sink.c: + * sys/v4l2/v4l2_calls.c: + * sys/v4l2/v4l2_calls.h: + v4l2: cleanup get/set input/output + output devices should use get/set output, and in either case we should + not print a warning message if the ioctl fails but the device does not + claim to support the tuner interface + +2010-06-10 11:15:46 -0500 Rob Clark + + * sys/v4l2/gstv4l2sink.c: + * sys/v4l2/gstv4l2xoverlay.c: + * sys/v4l2/gstv4l2xoverlay.h: + v4l2xoverlay: add support to create window + If xoverlay is available, v4l2sink should create a window for the overlay to + display in. + The window automatically tries to make itself as large as possible. + This works well on a small screen, but perhaps should first attempt to use + the size of the video that is played (no scaling). + +2010-04-04 06:41:28 -0500 Rob Clark + + * sys/v4l2/gstv4l2sink.c: + v4l2sink: special handling for cases gst_buffer_make_metadata_writable() + Special case check for sub-buffers: In certain cases, places like + GstBaseTransform, which might check that the buffer is writable before copying + metadata, timestamp, and such, will find that the buffer has more than one + reference to it. In these cases, they will create a sub-buffer with an offset=0 + and length equal to the original buffer size. + This could happen in two scenarios: (1) a tee in the pipeline, and (2) because + the refcnt is incremented in gst_mini_object_free() before the finalize function + is called, and decremented after it returns.. but returning this buffer to the + buffer pool in the finalize function, could wake up a thread blocked in + _buffer_alloc() which could run and get a buffer w/ refcnt==2 before the thread + originally unref'ing the buffer returns from finalize function and decrements + the refcnt back to 1! + This is related to issue #545501 + +2010-04-04 06:39:52 -0500 Rob Clark + + * sys/v4l2/gstv4l2bufferpool.c: + v4l2: fix race condition + The size of the buffer would be zero'd out in gst_v4l2_buffer_finalize() + after the buffer is qbuf'd or pushed onto the queue of available buffers.. + leaving a race condition where the thread waiting for the buffer could awake + and set back a valid size before the finalizing thread zeros out the length. + This would result that the newly allocated buffer has length of zero. + +2010-04-04 06:39:08 -0500 Rob Clark + + * sys/v4l2/gstv4l2sink.c: + * sys/v4l2/gstv4l2sink.h: + v4l2sink: add properties to control crop + +2010-04-04 06:37:16 -0500 Rob Clark + + * sys/v4l2/Makefile.am: + * sys/v4l2/gstv4l2object.c: + * sys/v4l2/gstv4l2sink.c: + * sys/v4l2/gstv4l2src.c: + * sys/v4l2/gstv4l2xoverlay.c: + v4l2: re-enable x-overlay support + +2010-12-25 11:52:36 -0600 Rob Clark + + * sys/v4l2/gstv4l2sink.c: + v4l2sink: fix for PAUSED->READY->PAUSED state transitions + When v4l2sink goes to PAUSED->READY it only stops streaming, so the state + should be set to STATE_PENDING_STREAMON in case the element transitions + back to PLAYING. + +2010-04-04 06:28:51 -0500 Rob Clark + + * sys/v4l2/gstv4l2sink.c: + * sys/v4l2/gstv4l2sink.h: + v4l2sink: add "min-queued-bufs" property + +2010-04-04 06:26:50 -0500 Rob Clark + + * sys/v4l2/gstv4l2bufferpool.c: + * sys/v4l2/gstv4l2bufferpool.h: + * sys/v4l2/gstv4l2sink.c: + * sys/v4l2/v4l2src_calls.c: + v4l2sink: Add support for blocking dequeue. + We'd prefer to throttle the decoder if we run out of buffers, to keep a bound + on memory usage. Also, for OMAP4 it is a requirement of the decoder to not + alternate between memory alloced by the display driver and malloc'd userspace + memory. + +2010-04-04 06:24:41 -0500 Rob Clark + + * sys/v4l2/gstv4l2bufferpool.c: + v4l2: clear flags before reusing buffer from buffer pool + note: this really only affects v4l2sink since gst_v4l2_buffer_pool_get() is + only called once per buffer in the v4l2src case (in + gst_v4l2src_buffer_pool_activate()) + +2010-04-04 06:23:31 -0500 Rob Clark + + * sys/v4l2/gstv4l2sink.c: + v4l2sink: don't render preroll buffers + Most v4l2 drivers will get upset when you queue the same buffer twice in a + row without first dequeueing it. + Rendering of pre-roll buffers can be re-introduced later, but will require + tracking the state of the buffer, and avoiding to re-QBUF if the buffer has + already been passed to the driver. + +2010-04-04 06:22:43 -0500 Rob Clark + + * sys/v4l2/gstv4l2sink.c: + v4l2sink: Improve behavior for shared buffers. + When the decoder is using pad_alloc(), v4l2sink would behave badly if + the number of buffers ('queue-size' property) was not high enough to + account for all the buffers needed by the decoder, and other elements + (such as queues) between the decoder and v4l2sink. This patch + slightly increases the default number of buffers, and changes v4l2sink + to drop frames rather than return an error in case the number of + buffers is not high enough. + +2010-11-15 15:58:28 +0100 Andy Wingo + + * ext/pulse/pulsesrc.c: + * ext/pulse/pulsesrc.h: + add "client" property + * ext/pulse/pulsesrc.c (gst_pulsesrc_class_init, gst_pulsesrc_init) + (gst_pulsesrc_set_property, gst_pulsesrc_get_property) + (gst_pulsesrc_open): Add a "client" property, as in pulsesink. + Fixes #634914 + +2010-12-29 15:54:46 +0000 Tim-Philipp Müller + + * gst/rtsp/gstrtspsrc.c: + rtspsrc: serialise/deserialise floats without changing locale + Use g_ascii_dtostr() and g_ascii_strtod() to serialise/deserialise + floating point numbers, instead of ugly hacks that switch locale + before and after calling libc functions (which is not a good idea + in a multi-threaded application). + +2010-12-29 14:40:05 +0000 Tim-Philipp Müller + + * gst/rtp/gstrtpjpegdepay.c: + rtpjpegdepay: fix framerate parsing for locales that use a comma as floating point + atof() converts strings according to the current locale, but the + framerate string will likely always use a dot as floating point + separator, so use g_ascii_strtod() instead (but also canonicalise + the string before, so we can handle both formats as input). + +2010-12-27 13:11:59 +0100 Wim Taymans + + * gst/rtpmanager/rtpsource.c: + rtpsource: use the right variable + Use the right variable for specifying that we sent a receiver report. + +2010-12-23 16:42:29 -0600 Rob Clark + + * sys/v4l2/gstv4l2bufferpool.c: + v4l2: fix typo + +2010-12-23 16:03:00 -0600 Rob Clark + + * gst/matroska/matroska-demux.c: + matroska-demux: add stream-format and alignment properties for h264 + +2010-12-22 11:41:59 +0100 Wim Taymans + + * gst/rtp/gstrtpgstpay.c: + gstpay: fix klass, add RTP as a use case + +2010-12-12 15:10:47 +0100 Wim Taymans + + * gst/rtp/gstrtpgstdepay.c: + gstdepay: cleanup the cache + +2010-12-12 05:10:01 +0100 Wim Taymans + + * gst/rtp/Makefile.am: + * gst/rtp/gstrtp.c: + * gst/rtp/gstrtpgstdepay.c: + * gst/rtp/gstrtpgstdepay.h: + * gst/rtp/gstrtpgstpay.c: + * gst/rtp/gstrtpgstpay.h: + gstpay/depay: add generic gstreamer payloader + Add the beginnings of a generic GStreamer buffers payloader. + +2010-12-23 17:06:58 +0100 Wim Taymans + + * gst/rtp/gstrtpmp4gpay.c: + mp4gpay: reset state on flush-stop + +2010-12-23 16:26:07 +0100 Wim Taymans + + * gst/rtp/gstrtpmp4gdepay.c: + mp4gdepay: flush state on flush-stop + +2010-12-23 16:25:15 +0100 Wim Taymans + + * gst/rtsp/gstrtspsrc.c: + rtspsrc: on-npt-stop is a manager signal + +2010-12-23 15:24:29 +0100 Wim Taymans + + * gst/rtsp/gstrtspsrc.c: + * gst/rtsp/gstrtspsrc.h: + rtspsrc: improve RTP session handling + Store the RTP session in the stream so that we can more efficiently + perform actions on the stream based on RTP signals. + +2010-12-23 13:55:31 +0100 Wim Taymans + + * gst/rtpmanager/rtpsource.c: + rtpsource: include last send RB block + Only report RB values for non-internal sources. + Report not only the RB blocks we last received from but also the last RB + block we sent to a source. + +2010-12-23 13:52:57 +0100 Wim Taymans + + * gst/rtpmanager/rtpsession.c: + * gst/rtpmanager/rtpsource.h: + rtpsession: remember last sent RB values. + +2010-12-23 13:00:49 +0100 Wim Taymans + + * gst/rtpmanager/rtpsource.c: + rtpsource: include all stats and document + Include all possible stats of a source in the stats structure because we might + be interested in what happened in the past. + Document the stats property and the fields. + +2010-12-23 12:59:59 +0100 Wim Taymans + + * tests/examples/rtp/client-PCMA.c: + examples: add example RTP stats + Add some more RTP examples for how to retrieve RTP stats in a receiver. + +2010-12-23 12:58:05 +0100 Wim Taymans + + * gst/rtpmanager/rtpsession.c: + rtpsession: also emit RTCP activity on SR + Also emit RTCP activity signals when we receive an SR packet without RB blocks, + such as from a sender that is not receiving anything. + +2010-12-23 11:10:55 +0100 Wim Taymans + + * gst/rtpmanager/gstrtpbin.c: + docs: add some more gstrtpbin docs + +2010-12-22 21:27:11 +0100 Edward Hervey + + * sys/ximage/gstximagesrc.c: + ximagesrc: remote is a boolean (and not uint) property + +2010-12-22 19:58:21 +0100 Sebastian Dröge + + * gst/matroska/matroska-demux.c: + matroskademux: Don't use gst_pad_alloc_buffer() + Using this in a demuxer will cause deadlocks if there's + a pad with a pending pad-block downstream, no matter if + there is a queue between the pad or not. Queues pass + bufferalloc downstream from the same thread and only + act as a thread boundary for events and buffers. + +2010-12-22 14:14:08 +0000 Tim-Philipp Müller + + * gst/matroska/matroska-mux.c: + matroskamux: fix subtitle pad template, we only handle kate for now + +2010-12-16 11:44:44 +0000 Tim-Philipp Müller + + * gst/rtsp/gstrtspsrc.c: + docs: update rtspsrc docs, rtpbin is not in -bad any more + +2010-12-22 11:42:31 +0100 Wim Taymans + + * gst/rtpmanager/gstrtpsession.c: + rtpsession: unlock before emitting signals + +2010-12-21 22:34:49 +0100 Wim Taymans + + * gst/rtp/Makefile.am: + * gst/rtp/gstrtp.c: + * gst/rtp/gstrtpac3pay.c: + * gst/rtp/gstrtpac3pay.h: + rtpac3pay: add AC3 payloader + +2010-12-21 22:17:19 +0100 Wim Taymans + + * gst/rtp/gstrtpac3depay.c: + ac3depay: fix debug category description + +2010-12-21 22:16:42 +0100 Wim Taymans + + * gst/rtp/gstrtpmpapay.c: + mpapay: add debug category + +2010-12-20 14:49:02 -0300 Thiago Santos + + * tests/check/Makefile.am: + * tests/check/elements/jpegenc.c: + jpegenc: Adds another test case + Adds a test for jpegenc to check that is possible to negotiate and + push buffers with different resolution one after another. + https://bugzilla.gnome.org/show_bug.cgi?id=637686 + +2010-12-21 13:37:40 -0300 Thiago Santos + + * ext/jpeg/gstjpegenc.c: + jpegenc: sink pad's getcaps shouldn't use the src pad getcaps + Instead of using get_allowed_caps on the srcpad, the sinkpad getcaps + should use the getcaps of the srcpad's peer. This way the srcpad + can keep using fixed_caps and sinkpad getcaps exposes all caps + that can be negotiated + https://bugzilla.gnome.org/show_bug.cgi?id=637686 + +2010-12-21 16:58:47 +0100 Wim Taymans + + * gst/rtp/gstasteriskh263.c: + * gst/rtp/gstrtpL16depay.c: + * gst/rtp/gstrtpL16pay.c: + * gst/rtp/gstrtpac3depay.c: + * gst/rtp/gstrtpamrdepay.c: + * gst/rtp/gstrtpamrpay.c: + * gst/rtp/gstrtpbvdepay.c: + * gst/rtp/gstrtpbvpay.c: + * gst/rtp/gstrtpceltdepay.c: + * gst/rtp/gstrtpceltpay.c: + * gst/rtp/gstrtpdepay.c: + * gst/rtp/gstrtpdvdepay.c: + * gst/rtp/gstrtpdvpay.c: + * gst/rtp/gstrtpg722depay.c: + * gst/rtp/gstrtpg722pay.c: + * gst/rtp/gstrtpg723depay.c: + * gst/rtp/gstrtpg723pay.c: + * gst/rtp/gstrtpg726depay.c: + * gst/rtp/gstrtpg726pay.c: + * gst/rtp/gstrtpg729depay.c: + * gst/rtp/gstrtpg729pay.c: + * gst/rtp/gstrtpgsmdepay.c: + * gst/rtp/gstrtpgsmpay.c: + * gst/rtp/gstrtph263depay.c: + * gst/rtp/gstrtph263pay.c: + * gst/rtp/gstrtph263pdepay.c: + * gst/rtp/gstrtph263ppay.c: + * gst/rtp/gstrtph264depay.c: + * gst/rtp/gstrtph264pay.c: + * gst/rtp/gstrtpilbcdepay.c: + * gst/rtp/gstrtpilbcpay.c: + * gst/rtp/gstrtpj2kdepay.c: + * gst/rtp/gstrtpj2kpay.c: + * gst/rtp/gstrtpjpegdepay.c: + * gst/rtp/gstrtpjpegpay.c: + * gst/rtp/gstrtpmp1sdepay.c: + * gst/rtp/gstrtpmp2tdepay.c: + * gst/rtp/gstrtpmp2tpay.c: + * gst/rtp/gstrtpmp4adepay.c: + * gst/rtp/gstrtpmp4apay.c: + * gst/rtp/gstrtpmp4gdepay.c: + * gst/rtp/gstrtpmp4gpay.c: + * gst/rtp/gstrtpmp4vdepay.c: + * gst/rtp/gstrtpmp4vpay.c: + * gst/rtp/gstrtpmpadepay.c: + * gst/rtp/gstrtpmpapay.c: + * gst/rtp/gstrtpmparobustdepay.c: + * gst/rtp/gstrtpmpvdepay.c: + * gst/rtp/gstrtpmpvpay.c: + * gst/rtp/gstrtppcmadepay.c: + * gst/rtp/gstrtppcmapay.c: + * gst/rtp/gstrtppcmudepay.c: + * gst/rtp/gstrtppcmupay.c: + * gst/rtp/gstrtpqcelpdepay.c: + * gst/rtp/gstrtpqdmdepay.c: + * gst/rtp/gstrtpsirendepay.c: + * gst/rtp/gstrtpsirenpay.c: + * gst/rtp/gstrtpspeexdepay.c: + * gst/rtp/gstrtpspeexpay.c: + * gst/rtp/gstrtpsv3vdepay.c: + * gst/rtp/gstrtptheoradepay.c: + * gst/rtp/gstrtptheorapay.c: + * gst/rtp/gstrtpvorbisdepay.c: + * gst/rtp/gstrtpvorbispay.c: + * gst/rtp/gstrtpvrawdepay.c: + * gst/rtp/gstrtpvrawpay.c: + rtp: add RTP hint to the klass + +2010-12-21 16:49:28 +0100 Wim Taymans + + * gst/rtp/gstasteriskh263.c: + * gst/rtp/gstrtpL16depay.c: + * gst/rtp/gstrtpL16pay.c: + * gst/rtp/gstrtpac3depay.c: + * gst/rtp/gstrtpamrdepay.c: + * gst/rtp/gstrtpamrpay.c: + * gst/rtp/gstrtpbvdepay.c: + * gst/rtp/gstrtpbvpay.c: + * gst/rtp/gstrtpceltdepay.c: + * gst/rtp/gstrtpceltpay.c: + * gst/rtp/gstrtpdepay.c: + * gst/rtp/gstrtpdvdepay.c: + * gst/rtp/gstrtpdvpay.c: + * gst/rtp/gstrtpg722depay.c: + * gst/rtp/gstrtpg722pay.c: + * gst/rtp/gstrtpg723depay.c: + * gst/rtp/gstrtpg723pay.c: + * gst/rtp/gstrtpg726depay.c: + * gst/rtp/gstrtpg726pay.c: + * gst/rtp/gstrtpg729depay.c: + * gst/rtp/gstrtpg729pay.c: + * gst/rtp/gstrtpgsmdepay.c: + * gst/rtp/gstrtpgsmpay.c: + * gst/rtp/gstrtph263depay.c: + * gst/rtp/gstrtph263pay.c: + * gst/rtp/gstrtph263pdepay.c: + * gst/rtp/gstrtph263ppay.c: + * gst/rtp/gstrtph264depay.c: + * gst/rtp/gstrtph264pay.c: + * gst/rtp/gstrtpilbcdepay.c: + * gst/rtp/gstrtpilbcpay.c: + * gst/rtp/gstrtpj2kdepay.c: + * gst/rtp/gstrtpj2kpay.c: + * gst/rtp/gstrtpjpegdepay.c: + * gst/rtp/gstrtpjpegpay.c: + * gst/rtp/gstrtpmp1sdepay.c: + * gst/rtp/gstrtpmp2tdepay.c: + * gst/rtp/gstrtpmp2tpay.c: + * gst/rtp/gstrtpmp4adepay.c: + * gst/rtp/gstrtpmp4apay.c: + * gst/rtp/gstrtpmp4gdepay.c: + * gst/rtp/gstrtpmp4gpay.c: + * gst/rtp/gstrtpmp4vdepay.c: + * gst/rtp/gstrtpmp4vpay.c: + * gst/rtp/gstrtpmpadepay.c: + * gst/rtp/gstrtpmpapay.c: + * gst/rtp/gstrtpmparobustdepay.c: + * gst/rtp/gstrtpmpvdepay.c: + * gst/rtp/gstrtpmpvpay.c: + * gst/rtp/gstrtppcmadepay.c: + * gst/rtp/gstrtppcmapay.c: + * gst/rtp/gstrtppcmudepay.c: + * gst/rtp/gstrtppcmupay.c: + * gst/rtp/gstrtpqcelpdepay.c: + * gst/rtp/gstrtpqdmdepay.c: + * gst/rtp/gstrtpsirendepay.c: + * gst/rtp/gstrtpsirenpay.c: + * gst/rtp/gstrtpspeexdepay.c: + * gst/rtp/gstrtpspeexpay.c: + * gst/rtp/gstrtpsv3vdepay.c: + * gst/rtp/gstrtptheoradepay.c: + * gst/rtp/gstrtptheorapay.c: + * gst/rtp/gstrtpvorbisdepay.c: + * gst/rtp/gstrtpvorbispay.c: + * gst/rtp/gstrtpvrawdepay.c: + * gst/rtp/gstrtpvrawpay.c: + rtp: fix rank of payloaders and depayloaders + Set the payloaders and depayloaders to a reasonable rank. + +2010-12-21 15:24:18 +0100 Wim Taymans + + * gst/rtp/gstrtpvrawdepay.c: + vrawdepay: reset depayloader state + Reset the depayloader state on flush-stop. + +2010-12-21 15:07:14 +0100 Wim Taymans + + * gst/rtp/gstrtpmp4vpay.c: + * gst/rtp/gstrtpmp4vpay.h: + mp4pay: use vmethod for intercepting events + +2010-12-21 13:55:40 +0100 Wim Taymans + + * gst/rtp/gstrtptheorapay.c: + theorapay: clear packet on flush-stop + +2010-12-21 13:49:41 +0100 Wim Taymans + + * gst/rtp/gstrtpvorbispay.c: + vorbispay: clear packet on flush-stop + +2010-12-21 12:31:44 +0100 Wim Taymans + + * gst/rtp/gstrtpmp4gdepay.c: + mp4gdepay: reset depayloader state + +2010-12-21 12:29:58 +0100 Wim Taymans + + * gst/rtp/gstrtph264pay.c: + h264pay: flush adapter on flush-stop + +2010-12-20 18:49:49 +0100 Wim Taymans + + * gst/rtp/gstrtpmpapay.c: + mpapay: flush last packets on EOS + +2010-12-20 17:47:05 +0100 Edward Hervey + + * common: + Automatic update of common submodule + From 169462a to 46445ad + +2010-12-20 16:51:47 +0100 Wim Taymans + + * gst/rtp/gstrtpmpapay.c: + mpapay: reset payloader on state change + +2010-12-20 16:05:36 +0100 Wim Taymans + + * gst/rtp/gstrtpmpapay.c: + mpapay: reset payloader on flush + Reset the payloader on a flush event. + Handle DISCONT better. + +2010-12-20 15:54:45 +0100 Wim Taymans + + * gst/rtpmanager/rtpjitterbuffer.c: + jitterbuffer: get better buffering level + When the jitterbuffer contains -1 timestamps, make sure we still calculate the + buffer fill level by skipping the -1 buffers. + Try to be more resilient to weird input timestamps. + +2010-12-20 11:10:22 +0100 Wim Taymans + + * gst/rtpmanager/gstrtpjitterbuffer.c: + jitterbuffer: provide a clock. + since we are using the clock for sync, we need to also provide a clock for good + measure. The reason is that even if downstream elements provide a clock, we + don't want to have that clock selected because it might not be running yet. + +2010-12-20 10:49:56 +0100 Wim Taymans + + * gst/rtpmanager/gstrtpbin.c: + rtpbin: copy buffering stats + when we create an aggregate buffering message, copy the buffering stats form the + last message. At least we get correct buffering mode then. + +2010-12-19 11:02:41 +0100 Sebastian Dröge + + * tests/check/pipelines/wavenc.c: + wavenc: Fix memory leaks in the unit test + +2010-12-19 10:58:16 +0100 Sebastian Dröge + + * gst/effectv/gstradioac.c: + * gst/effectv/gstradioac.h: + radioactv: Prevent use of uninitialized values + Fixes bug #618652. + +2010-12-19 10:22:29 +0100 Sebastian Dröge + + * gst/debugutils/gstcapsdebug.c: + capsdebug: Don't leak pad templates created from static pad templates + +2010-11-29 12:36:06 +0000 Vincent Penquerc'h + + * sys/ximage/gstximagesrc.c: + * sys/ximage/gstximagesrc.h: + ximagesrc: change from XGetImage to XGetSubImage dependant on a property + ximagesrc: change from XGetImage to XGetSubImage dependant on a property + to avoid unnecessary performance hits by default. + +2010-11-28 16:04:35 +0000 Vincent Penquerc'h + + * sys/ximage/gstximagesrc.c: + ximagesrc: use XGetSubImage instead of XGetImage, works with remote X + ximagesrc: use XGetSubImage instead of XGetImage, works with remote X + (on my setup anyway...) + +2010-11-27 17:15:32 +0000 Vincent Penquerc'h + + * sys/ximage/gstximagesrc.c: + ximagesrc: fix various width/height calculations being off by one, + ximagesrc: fix various width/height calculations being off by one, + and make it so a single pixel width/height can be captured (except + the top left one, as 0,0,0,0 is reserved for full screen as per + the property comments). + +2010-12-17 19:19:35 -0600 Rob Clark + + * sys/v4l2/gstv4l2object.c: + fix compile errors on macosx + with i686-apple-darwin10-gcc-4.2.1: + gstv4l2object.c: In function 'gst_v4l2_object_get_nearest_size': + gstv4l2object.c:1988: warning: format '%u' expects type 'unsigned int', but argument 12 has type 'gint *' + gstv4l2object.c:1988: warning: format '%u' expects type 'unsigned int', but argument 13 has type 'gint *' + +2010-12-17 15:38:15 +0100 Mark Nauwelaerts + + * gst/rtp/gstrtph264depay.c: + rtph264depay: determine output h264 layout using caps negotiation + ... thereby (partially) deprecating properties currently controlling whether + or not byte-stream output or NAL/AU alignment (though properties still determine + fallback if nothing specified in caps). + Fixes #606662. + +2010-12-16 18:55:43 +0100 Wim Taymans + + * gst/rtp/gstrtpj2kpay.c: + j2kpay: handle EOC correctly + Don't include the next 2 bytes when we are at the end of the data and there are + no more bytes left. + +2010-12-16 15:15:49 +0100 Mark Nauwelaerts + + * ext/pulse/pulsesink.c: + pulsesink: flush remaining buffered samples on EOS + ... which can make a difference between all or nothing when dealing + with short streams and relatively large ringbuffer segment. + +2010-12-16 10:04:19 +0100 Sebastian Dröge + + * gst/deinterlace/gstdeinterlace.c: + deinterlace: Change classification to Filter/Effect/Video/Deinterlace + +2010-12-15 18:21:34 +0100 Edward Hervey + + * gst/rtp/gstrtpj2kpay.c: + rtpj2kpay: Initialize all fields + Makes sad compliers happy + +2010-12-15 16:22:54 +0100 Wim Taymans + + * gst/rtp/gstrtpj2kpay.c: + j2kpay: cleanup header construction + Use a simpler way of constructing the header that doesn't depend on + the endianness. + +2010-12-15 13:30:50 +0000 Tim-Philipp Müller + + * configure.ac: + configure: depend on -base from git for new rtp base depayloader features + This is ok in this case, since the plan is to release core/base again + along with good/ugly/bad in the next cycle. + +2010-12-15 14:55:58 +0200 Stefan Kost + + * common: + Automatic update of common submodule + From 20742ae to 169462a + +2010-12-15 13:12:09 +0100 Wim Taymans + + * gst/rtp/gstrtpj2kdepay.c: + * gst/rtp/gstrtpj2kdepay.h: + j2kdepay: add support for buffer lists + +2010-12-14 18:12:43 +0100 Wim Taymans + + * gst/rtpmanager/rtpsession.c: + session: fix average RTCP packet size some more. + Fix stupid error in averaging macro. + Include udp headers in packet length estimation. + +2010-12-14 17:15:23 +0100 Wim Taymans + + * gst/rtpmanager/rtpsession.c: + * gst/rtpmanager/rtpstats.c: + rtpbin: correctly calculate RTCP packet size + +2010-12-14 15:27:52 +0100 Wim Taymans + + * gst/rtp/gstrtpj2kpay.c: + j2kpay: stop scanning when we reached the end + Stop scanning for markers when we reached the end of the data. + +2010-12-13 16:23:24 +0200 Stefan Kost + + * common: + Automatic update of common submodule + From 011bcc8 to 20742ae + +2010-12-13 12:56:12 +0100 Wim Taymans + + * gst/rtpmanager/gstrtpjitterbuffer.c: + jitterbuffer: avoid leaking sink events + Avoid leaking the newsegment event when it has the wrong format. + +2010-12-12 14:53:17 +0100 Wim Taymans + + * gst/rtp/gstrtpmp4vpay.c: + mp4vpay: we can also accept xvid caps + +2010-12-12 01:39:06 +1100 Jan Schmidt + + * gst/deinterlace/gstdeinterlace.c: + deinterlace: Avoid infinite loop draining frames + When the pipeline is flushed just as we're draining history, + don't loop infinitely, just discard the history and abort. + +2010-12-11 17:39:20 +0000 Tim-Philipp Müller + + * ext/jpeg/gstjpegdec.c: + * ext/jpeg/gstjpegdec.h: + jpegdec: add "max-errors" property to ignore decoding errors + Add property to ignore decoding errors. Default is to ignore a few + decoding errors if the input is packetized, but error out immediately + if the input is not packetized. + Ignoring errors for packetized input most likely doesn't work + properly yet, so don't do that for now. + https://bugzilla.gnome.org/show_bug.cgi?id=623063 + +2010-05-28 15:27:14 +0100 Tim-Philipp Müller + + * ext/jpeg/gstjpegenc.c: + jpegenc: free/malloc instead of realloc, avoids memcpy + +2010-12-11 17:49:03 +0100 Sebastian Dröge + + * gst/qtdemux/qtdemux.c: + qtdemux: Check if there's actually a seek table before parsing it + +2010-12-11 17:46:17 +0100 Kishore Arepalli + + * gst/qtdemux/qtdemux.c: + qtdemux: Implement CONVERT and FORMATS query + Fixes bug #636784. + +2010-07-01 00:22:07 +0100 Tim-Philipp Müller + + * gst/matroska/matroska-demux.c: + matroska-demux: put unrecognised RIFF format IDs into the unknown caps + Extra info can't hurt. Field names aren't necessarily consistent with + what's used elsewhere though (e.g. avidemux), but then neither are the + caps. + https://bugzilla.gnome.org/show_bug.cgi?id=623178 + +2010-10-29 22:50:14 +0100 Jan Schmidt + + * ext/pulse/pulsemixerctrl.c: + * ext/pulse/pulsemixerctrl.h: + pulsemixer: Implement MIXER_FLAG_AUTO_NOTIFICATIONS + Add the mixer flag and send notifications when either the volume or muted + status changes. + https://bugzilla.gnome.org/show_bug.cgi?id=618389 + +2010-02-08 21:41:29 +0100 Mark Nauwelaerts + + * gst/rtsp/gstrtspsrc.c: + rtspsrc: mark DISCONT when resuming PLAY + In particular, when streaming interleaved, this arranges for setting a new + timestamp on outgoing buffer so downstream can appropriate reset + to a change in (rtp)time. + +2010-12-02 16:08:34 +0100 Mark Nauwelaerts + + * gst/rtsp/gstrtspsrc.c: + * gst/rtsp/gstrtspsrc.h: + rtspsrc: degrade gracefully upon failing seek and tweak QUERY_SEEKING response + +2010-10-25 11:51:06 +0200 Mark Nauwelaerts + + * gst/rtsp/gstrtspsrc.c: + rtspsrc: add and use auto buffering mode + ... which selects BUFFER for a non-live stream, and otherwise SLAVE. + Fixes #633088. + +2010-12-06 12:16:12 +0100 Wim Taymans + + * gst/rtp/gstrtpj2kdepay.c: + * gst/rtp/gstrtpj2kdepay.h: + j2kdepay: make the depayloader more resilient + Use 3 adapters, one to accumulate paketization units, another on to accumulate + tiles and a last one to accumulate the final frame. + Don't just blindly flush the adapter on DISCONT but only discard the current + packetization unit. + When we dropped jpeg2000 packets between SOP markers, adjust the SOT header with + the new lenght. + +2010-12-09 13:49:04 +0100 Mark Nauwelaerts + + * gst/qtdemux/qtdemux.c: + qtdemux: fix flow return aggregation + +2010-12-08 11:35:33 +0100 Mark Nauwelaerts + + * gst/qtdemux/qtdemux.c: + qtdemux: fix handling near end-of-file corner cases + Also, relax some error handling to not bail out completely when something + feels amiss, but consider this EOF and continue with was obtained so far. + +2010-12-07 17:19:00 +0100 Mark Nauwelaerts + + * gst/qtdemux/qtdemux.c: + qtdemux: fragmented support; fix offset handling and relax error raising + In particular, accept unknown stream in track fragment, and only error out + if that raises problems later on with respect to offset tracking. + Fixes #620283. + +2010-12-07 13:11:48 +0100 Mark Nauwelaerts + + * gst/flv/Makefile.am: + * gst/flv/gstflvdemux.c: + flvdemux: use aac codec-data to adjust samplerate if needed + Based on patch by Fabien Lebaillif-Delamare + Fixes #636621. + +2010-12-07 11:43:13 +0100 Wim Taymans + + * ext/pulse/pulsesink.c: + pulsesink: don't uncork in _start + Don't uncork in the _start method just yet but wait until we have written some + samples to pulseaudio. This avoid underruns on pulseaudio and less crackling + noises when starting. + +2010-12-06 19:59:49 +0100 Alessandro Decina + + * gst/qtdemux/qtdemux.c: + qtdemux: fix compiler warnings on OSX. + +2010-12-06 18:17:24 +0100 Mark Nauwelaerts + + * ext/jpeg/gstjpegdec.c: + jpegdec: add debug to notify when skipping to jpeg header + +2010-12-06 18:16:19 +0100 Mark Nauwelaerts + + * ext/jpeg/gstjpegdec.c: + jpegdec: discard incomplete image + ... as determined when finding SOI next image before an EOI. + Based on patch by David Hoyt + Fixes #635734. + +2010-12-06 17:45:38 +0100 Mark Nauwelaerts + + * ext/jpeg/gstjpegdec.c: + jpegdec: avoid infinite loop when resyncing + Fixes #635734 (partly). + +2010-12-06 15:21:53 +0100 David Hoyt + + * gst/imagefreeze/gstimagefreeze.c: + imagefreeze: pass along eos if received before buffer arrives + Fixes #636172. + +2010-10-20 11:05:49 +0200 Andoni Morales Alastruey + + * gst/matroska/ebml-write.c: + * gst/matroska/ebml-write.h: + * gst/matroska/matroska-mux.c: + matroskamux: try to write timestamps in all the outgoing buffers + Fixes #632654. + +2010-12-06 12:17:21 +0100 Mark Nauwelaerts + + * gst/debugutils/progressreport.c: + * gst/debugutils/progressreport.h: + progressreport: optionally determine progress using buffer metadata + Based on patch by Leo Singer + Fixes #629418. + +2010-12-05 14:39:19 +0100 Edward Hervey + + * tests/check/elements/interleave.c: + check: Fixup the shutting down order + First bring down everything to NULL before attempting to unlink + or unref anything. + Avoids the tests just hanging there for ever waiting to acquire a + lock that doesn't exist anymore. + +2010-11-04 19:31:45 +0100 Janne Grunau + + * sys/v4l2/gstv4l2bufferpool.c: + v4l2src: set top field first for interlaced buffers if v4l2 exports it + https://bugzilla.gnome.org/show_bug.cgi?id=634393 + +2010-11-04 18:36:09 +0100 Janne Grunau + + * sys/v4l2/gstv4l2object.c: + v4l2src: check field information and set interlaced caps accordingly + Reject the format if the field type is not supported. + https://bugzilla.gnome.org/show_bug.cgi?id=634391 + +2010-12-03 17:42:14 +0100 Benjamin Gaignard + + * Android.mk: + * android/NOTICE: + * android/apetag.mk: + * android/avi.mk: + * android/flv.mk: + * android/gst/rtpmanager/gstrtpbin-marshal.c: + * android/gst/rtpmanager/gstrtpbin-marshal.h: + * android/gst/udp/gstudp-enumtypes.c: + * android/gst/udp/gstudp-enumtypes.h: + * android/gst/udp/gstudp-marshal.c: + * android/gst/udp/gstudp-marshal.h: + * android/icydemux.mk: + * android/id3demux.mk: + * android/qtdemux.mk: + * android/rtp.mk: + * android/rtpmanager.mk: + * android/rtsp.mk: + * android/soup.mk: + * android/udp.mk: + * android/wavenc.mk: + * android/wavparse.mk: + Add build system for Android + +2010-03-26 13:51:58 +0100 Guillaume Emont + + * gst/debugutils/gstnavseek.c: + navseek: add basic support to change playback rate + The following keys will now be interpreted by navseek: + 'f' means fast forward: the stream gets played at rate 2.0 + 'r' means rewind: the stream gets played at rate -2.0 + 'n' means normal: the stream gets played at rate 1.0 + Fixes #631516. + +2010-12-01 13:12:04 +0100 Mark Nauwelaerts + + * gst/qtdemux/qtdemux.c: + qtdemux: add support for e(a)c-3 audio + +2010-11-19 12:44:35 +0100 Mark Nauwelaerts + + * gst/qtdemux/qtdemux.c: + qtdemux: avoid sending EOS event twice + +2010-11-19 12:44:18 +0100 Mark Nauwelaerts + + * gst/qtdemux/qtdemux.c: + qtdemux: remove dead code trying to update stream duration + On the one hand, it insufficiently checks whether it only updates a dummy + segment. On the other hand, only doing this at the time the last sampled is + prepared (and sent downstream) is too little too late. + +2010-11-09 10:58:57 +0100 Mark Nauwelaerts + + * gst/qtdemux/qtdemux.c: + qtdemux: fragmented support; handle ismv sample flags + +2010-11-08 11:41:21 +0100 Mark Nauwelaerts + + * gst/qtdemux/qtdemux.c: + qtdemux: fragmented support; handle ismv stbl atoms + ... or lack of some thereof, such as mandatory stsz. Shuffle some code + in _stbl_init to detect this early enough. + +2010-11-08 11:39:37 +0100 Mark Nauwelaerts + + * gst/qtdemux/qtdemux.c: + qtdemux: fragmented support; compensate for ismv offset handling + ... or lack thereof, which according to specs would put media data in + unlikely position. + +2010-11-04 14:07:56 +0100 Mark Nauwelaerts + + * gst/qtdemux/qtdemux.c: + * gst/qtdemux/qtdemux.h: + qtdemux: fragmented support for push mode + +2010-11-04 10:17:37 +0100 Mark Nauwelaerts + + * gst/qtdemux/qtdemux.c: + * gst/qtdemux/qtdemux.h: + qtdemux: fragmented support; proper and incremental moof parsing + That is, parse each moof in one pass (considering all contained streams' + metadata), and do so incrementally as needed for playback rather than + an initial complete scan of all moof (though all moov sample metadata + is fully parsed at startup). + +2010-11-04 10:06:30 +0100 Mark Nauwelaerts + + * gst/qtdemux/qtdemux.c: + qtdemux: refactor stream freeing + +2010-11-04 10:05:15 +0100 Mark Nauwelaerts + + * gst/qtdemux/qtdemux.c: + qtdemux: delegate linear search for sample to binary search when possible + Also arrange for parsing a sample prior to taking a reference to it, + which requires less memory layout assumptions for correctness. + +2010-11-01 15:52:29 +0100 Mark Nauwelaerts + + * gst/qtdemux/qtdemux.c: + qtdemux: fragmented support; handle moov samples and proper stream duration + +2010-11-01 13:40:05 +0100 Mark Nauwelaerts + + * gst/qtdemux/qtdemux.c: + qtdemux: fragmented support; consider mvex and handle flags and offset fields + +2010-10-28 16:49:41 +0200 Mark Nauwelaerts + + * gst/qtdemux/qtdemux.c: + qtdemux: fragmented support; forego check for short streams + ... as some bogus files may indicate streams of 0 duration in moov, + while indicating the complete movie duration in mvhd (the latter should + be in mehd). + +2010-10-28 16:46:48 +0200 Mark Nauwelaerts + + * gst/qtdemux/qtdemux.c: + * gst/qtdemux/qtdemux_types.h: + qtdemux: fragmented support; code cleanups and optimizations in atom parsing + Avoid extra allocation in _parse_trun, add more checks for parsing errors, + add or adjust some debug statement, fix comments, sprinkle some branch + prediction. + +2010-09-13 23:19:44 -0300 Thiago Santos + + * gst/qtdemux/qtdemux.c: + qtdemux: parse_moof should return TRUE on success + +2010-09-10 22:41:03 -0300 Thiago Santos + + * gst/qtdemux/qtdemux.c: + qtdemux: Fix iteration bug + Avoid infinite loop when iterating traf + +2010-09-10 21:32:26 -0300 Thiago Santos + + * gst/qtdemux/qtdemux.c: + qtdemux: Refactor trun parsing + The allocation of the samples can be placed out of the loop. + Makes the code clearer. + Also avoid relying on traf information as it is placed on the + end of the file and might not be acessible on push mode. + +2010-09-10 00:29:26 -0300 Thiago Santos + + * gst/qtdemux/qtdemux.c: + qtdemux: Remove parsing of unused atom + sdtp atom is parsed but not used, so we don't have to + parse it. + +2010-11-09 11:45:00 +0100 Mark Nauwelaerts + + * gst/qtdemux/qtdemux.c: + qtdemux: tweak wam support + ... with some comment and portability macros. + +2009-09-23 18:47:42 +0200 Marc-André Lureau + + * gst/qtdemux/qtdemux.c: + * gst/qtdemux/qtdemux_fourcc.h: + * gst/qtdemux/qtdemux_types.c: + qtdemux: support wma & vc-1 + https://bugzilla.gnome.org/show_bug.cgi?id=596321 + +2010-03-11 09:56:04 +0100 Andoni Morales Alastruey + + * gst/qtdemux/qtdemux.c: + * gst/qtdemux/qtdemux.h: + qtdemux: parse fmp4 samples information + The fragmented mp4 format stores the tracks and samples information in the + 'moof' boxes, which are appended before each fragment (fragment->'moof'+'mdat'). + The 'mfra' box stores the offset of each 'moof' box and their presentation + time. The location of this box can be retrieved from the 'mfro' box, which is + located at the end of the file. + The 'mfra' box is parsed to get the offset of each 'moof' box and their + presentation time. + Each 'moof' box can contain information for one or more tracks inside + 'tfhd' boxes. For each track in a 'moof', we have a 'trun' box, which + contains information of each sample (offset and duration) used to build + the samples table. + Based on patch by Marc-André Lureau + https://bugzilla.gnome.org/show_bug.cgi?id=596321 + +2010-03-11 15:34:49 +0100 Marc-André Lureau + + * gst/qtdemux/qtatomparser.h: + * gst/qtdemux/qtdemux_dump.c: + * gst/qtdemux/qtdemux_dump.h: + * gst/qtdemux/qtdemux_fourcc.h: + * gst/qtdemux/qtdemux_types.c: + * gst/qtdemux/qtdemux_types.h: + qtdemux: add fragmented mp4 fourccs + Adds fourcc's for tfra, tfhd, trun, sdtp, trex, mehd and + their dumps + https://bugzilla.gnome.org/show_bug.cgi?id=596321 + +2010-03-11 10:24:56 +0100 Marc-André Lureau + + * gst/qtdemux/qtdemux.c: + qtdemux: parse the track id from the track header + Signed-off-by: Andoni Morales Alastruey + https://bugzilla.gnome.org/show_bug.cgi?id=596321 + +2010-03-11 14:10:12 +0100 Marc-André Lureau + + * gst/qtdemux/qtdemux.c: + qtdemux: allow pulling atoms with unknown size + Signed-off-by: Andoni Morales Alastruey + https://bugzilla.gnome.org/show_bug.cgi?id=596321 + +2010-07-14 20:13:55 +0200 Marc-André Lureau + + * gst/qtdemux/qtdemux_dump.c: + qtdemux: make qtdemux_dump_mvhd parse version 1 correctly + Versions 0 and 1 of mvhd have different sizes of its values + (32bits/64bits). This patch makes it dump them correctly. + Also use the right node in the parameter and not the root node. + https://bugzilla.gnome.org/show_bug.cgi?id=596321 + +2010-11-19 12:45:00 +0100 Mark Nauwelaerts + + * gst/matroska/matroska-mux.c: + matroskademux: minor cleanups in setting streamheader on caps + +2010-11-02 17:04:04 +0100 Mark Nauwelaerts + + * gst/matroska/matroska-demux.c: + matroskademux: normalize empty Cues to no Cues + ... to trigger indexless seeking. + +2010-10-26 11:15:49 +0200 Mark Nauwelaerts + + * gst/avi/gstavidemux.c: + avidemux: add workaround for buggy list size + Fixes truncated extra-data in hdrl/strl/strf due to buggy containing + list size not accounting for padding in contained chunks. + +2010-12-02 16:11:01 +0100 Mark Nauwelaerts + + * gst/rtpmanager/gstrtpssrcdemux.c: + rtpssrcdemux: do not hold custom PAD_LOCK when pushing downstream + +2010-12-02 16:10:14 +0100 Mark Nauwelaerts + + * gst/rtsp/gstrtspsrc.c: + rtspsrc: reset session manager base time when flushing + ... as rtpbin uses running time to handle rtpjitterbuffer's buffer mode pauses. + +2010-12-01 16:51:33 +0100 Mark Nauwelaerts + + * gst/rtsp/gstrtspsrc.c: + rtspsrc: include range request for all streams with non-aggregate control + +2010-10-07 14:50:53 +0200 Mark Nauwelaerts + + * gst/rtsp/gstrtspsrc.c: + rtspsrc: fix debug statement + +2010-12-03 15:38:00 +0100 Edward Hervey + + * gst/avi/gstavidemux.c: + avidemux: Parse more variants of numerical IDIT tag + +2010-05-07 17:30:30 +0200 Edward Hervey + + * ext/libpng/gstpngenc.c: + pngenc: Use proper framerate range in caps + +2010-12-03 15:04:26 +0100 Edward Hervey + + * tests/check/pipelines/wavenc.c: + tests: Fix previously unbuildable/untested wavenc test + +2010-10-24 15:21:08 +0200 Edward Hervey + + * gst/flv/gstflvdemux.c: + flvdemux: Refactor tag pushing logic + The logic of when to push was wrong also (resulting in some tags never + being pushed). + +2010-10-24 15:20:27 +0200 Edward Hervey + + * gst/flv/Makefile.am: + * gst/flv/gstflvdemux.c: + flvdemux: Use pbutils for codec descriptions + +2010-04-13 11:29:30 +0200 Edward Hervey + + * tests/check/elements/udpsink.c: + check: Use fail_unless_equals_int instead of fail_if + Makes the error message more interesting + +2010-11-30 19:22:11 +0100 Edward Hervey + + * gst/avi/gstavidemux.c: + avidemux: Also extract IDIT tags present too early + https://bugzilla.gnome.org/show_bug.cgi?id=636143 + +2010-11-30 19:21:23 +0100 Edward Hervey + + * gst/avi/gstavidemux.c: + avidemux: Also emit DateTime tag + https://bugzilla.gnome.org/show_bug.cgi?id=636143 + +2010-12-03 00:22:48 +0000 Tim-Philipp Müller + + * gst/wavparse/gstwavparse.c: + wavparse: detect DTS advertised as PCM correctly in some more cases + The DTS typefinder may return a lower probability for frames that start + at non-zero offsets and where there's no second frame sync in the first + buffer. It's fairly unlikely that we'll acidentally identify PCM data + as DTS, so we don't do additional checks for now. + https://bugzilla.gnome.org/show_bug.cgi?id=636234 + +2010-11-08 17:11:42 +0200 Stefan Kost + + * tests/check/Makefile.am: + tests: makefile cleanup + Fix indentation. Use $(GST_MAJORMINOR) instead of hardcoded 0.10. + +2010-11-08 17:02:56 +0200 Stefan Kost + + * tests/check/Makefile.am: + * tests/check/pipelines/.gitignore: + * tests/check/pipelines/wavenc.c: + tests: add a test for wav muxing + +2010-11-08 16:57:17 +0200 Stefan Kost + + * tests/check/elements/interleave.c: + * tests/check/pipelines/wavpack.c: + tests: remove newlines between variable decls (old gst-indent failure) + +2010-11-08 14:47:04 +0200 Stefan Kost + + * ext/libpng/gstpngdec.c: + pngdec: use png_error() as recommended by libpng docs to signal an error + Without that the element loops endlessly on broekn pngs. Fixes #634314 + +2010-11-16 17:48:16 -0300 Thiago Santos + + * gst/qtdemux/qtdemux.c: + qtdemux: Parse and use creation time tag from mvhd + Expose creation time from mvhd as a datetime tag + Fixes #634928 + +2010-10-27 19:15:20 +0200 Andoni Morales Alastruey + + * gst/icydemux/gsticydemux.c: + icydemux: Add 'StreamUrl' metadata as GST_TAG_HOMEPAGE tag + +2010-10-23 19:34:00 -0400 Tom Janiszewski + + * gst/flv/gstflvmux.c: + flvmux: Fix for nellymoser codecid setting + Fixes bug #632897. + +2010-10-21 16:15:08 +0200 Sebastian Dröge + + * gst/matroska/matroska-mux.c: + matroskamux: Add support for E-AC3 + +2010-10-21 16:14:44 +0200 Sebastian Dröge + + * gst/matroska/matroska-mux.c: + matroskamux: Add support for DTS + +2010-10-31 18:08:17 +0100 Sebastian Dröge + + * ext/soup/gstsouphttpsrc.c: + souphttpsrc: Don't send seeks behind the end of file to the server + Also improve debug output, re-initialize the content size and let the + seek handler error out on invalid seek segments. + Fixes bug #632977. + +2010-12-02 17:53:42 +0100 Wim Taymans + + * gst/rtp/gstrtpj2kpay.c: + j2kpay: use SOP markers to split bitstream + When parsing the bitstream, look for SOP markers because we are allowed to split + packets on those marker boundaries. + Rework the parsing code a little so that we can pack multiple Packetization + units in one RTP packet. + +2010-11-18 12:49:47 +0100 Wim Taymans + + * gst/rtp/gstrtpj2kpay.c: + * gst/rtp/gstrtpj2kpay.h: + rtpj2kpay: use buffer lists + Use buffer lists for doing zerocopy payloading. + Add property to disable buffer lists. + +2010-11-16 16:54:25 +0100 Wim Taymans + + * gst/rtp/gstrtph264pay.c: + h264pay: small cleanups + Allocate adapter only once. + Make some guint8 * const. + +2010-11-16 15:39:24 +0100 Tambet Ingo + + * gst/rtp/gstrtph264pay.c: + * gst/rtp/gstrtph264pay.h: + rtph264pay: implement full bytestream scan mode. + Implement the full bytestream scan mode. + Fixes #634910 + +2010-11-15 10:52:31 +0100 Wim Taymans + + * tests/examples/rtp/client-H263p-AMR.sh: + * tests/examples/rtp/client-H263p-PCMA.sh: + * tests/examples/rtp/client-H263p.sh: + * tests/examples/rtp/client-H264-PCMA.sh: + * tests/examples/rtp/client-H264.sh: + * tests/examples/rtp/client-PCMA.sh: + * tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh: + examples: improve RTP examples + Make the examples use autovideosink and ffmpegcolorspace for better + compàtibility. + Make some more variables for the sink and the decoders. + Set zerolatency tuning on x264enc for better realtime results. + +2010-11-10 11:04:48 +0100 Wim Taymans + + * gst/rtsp/gstrtspsrc.c: + * gst/rtsp/gstrtspsrc.h: + rtspsrc: select multicast transports in a smarter way + When we see a multicast address in the SDP connection, only try to negotiate a + multicast transport with the server. + Fixes #634093 + +2010-12-02 18:14:16 +0000 Tim-Philipp Müller + + * configure.ac: + Bump GLib requirement to implicit requirement + ie. >= 2.20 while we depend on core/base 0.10.31 + +2010-12-02 18:13:57 +0000 Tim-Philipp Müller + + * configure.ac: + * docs/plugins/gst-plugins-good-plugins.hierarchy: + * docs/plugins/inspect/plugin-1394.xml: + * docs/plugins/inspect/plugin-aasink.xml: + * docs/plugins/inspect/plugin-alaw.xml: + * docs/plugins/inspect/plugin-alpha.xml: + * docs/plugins/inspect/plugin-alphacolor.xml: + * docs/plugins/inspect/plugin-annodex.xml: + * docs/plugins/inspect/plugin-apetag.xml: + * docs/plugins/inspect/plugin-audiofx.xml: + * docs/plugins/inspect/plugin-auparse.xml: + * docs/plugins/inspect/plugin-autodetect.xml: + * docs/plugins/inspect/plugin-avi.xml: + * docs/plugins/inspect/plugin-cacasink.xml: + * docs/plugins/inspect/plugin-cairo.xml: + * docs/plugins/inspect/plugin-cutter.xml: + * docs/plugins/inspect/plugin-debug.xml: + * docs/plugins/inspect/plugin-deinterlace.xml: + * docs/plugins/inspect/plugin-dv.xml: + * docs/plugins/inspect/plugin-efence.xml: + * docs/plugins/inspect/plugin-effectv.xml: + * docs/plugins/inspect/plugin-equalizer.xml: + * docs/plugins/inspect/plugin-esdsink.xml: + * docs/plugins/inspect/plugin-flac.xml: + * docs/plugins/inspect/plugin-flv.xml: + * docs/plugins/inspect/plugin-flxdec.xml: + * docs/plugins/inspect/plugin-gconfelements.xml: + * docs/plugins/inspect/plugin-gdkpixbuf.xml: + * docs/plugins/inspect/plugin-goom.xml: + * docs/plugins/inspect/plugin-goom2k1.xml: + * docs/plugins/inspect/plugin-gstrtpmanager.xml: + * docs/plugins/inspect/plugin-halelements.xml: + * docs/plugins/inspect/plugin-icydemux.xml: + * docs/plugins/inspect/plugin-id3demux.xml: + * docs/plugins/inspect/plugin-imagefreeze.xml: + * docs/plugins/inspect/plugin-interleave.xml: + * docs/plugins/inspect/plugin-jpeg.xml: + * docs/plugins/inspect/plugin-level.xml: + * docs/plugins/inspect/plugin-matroska.xml: + * docs/plugins/inspect/plugin-mulaw.xml: + * docs/plugins/inspect/plugin-multifile.xml: + * docs/plugins/inspect/plugin-multipart.xml: + * docs/plugins/inspect/plugin-navigationtest.xml: + * docs/plugins/inspect/plugin-oss4.xml: + * docs/plugins/inspect/plugin-ossaudio.xml: + * docs/plugins/inspect/plugin-png.xml: + * docs/plugins/inspect/plugin-pulseaudio.xml: + * docs/plugins/inspect/plugin-quicktime.xml: + * docs/plugins/inspect/plugin-replaygain.xml: + * docs/plugins/inspect/plugin-rtp.xml: + * docs/plugins/inspect/plugin-rtsp.xml: + * docs/plugins/inspect/plugin-shapewipe.xml: + * docs/plugins/inspect/plugin-shout2send.xml: + * docs/plugins/inspect/plugin-smpte.xml: + * docs/plugins/inspect/plugin-soup.xml: + * docs/plugins/inspect/plugin-spectrum.xml: + * docs/plugins/inspect/plugin-speex.xml: + * docs/plugins/inspect/plugin-taglib.xml: + * docs/plugins/inspect/plugin-udp.xml: + * docs/plugins/inspect/plugin-video4linux2.xml: + * docs/plugins/inspect/plugin-videobox.xml: + * docs/plugins/inspect/plugin-videocrop.xml: + * docs/plugins/inspect/plugin-videofilter.xml: + * docs/plugins/inspect/plugin-videomixer.xml: + * docs/plugins/inspect/plugin-wavenc.xml: + * docs/plugins/inspect/plugin-wavpack.xml: + * docs/plugins/inspect/plugin-wavparse.xml: + * docs/plugins/inspect/plugin-ximagesrc.xml: + * docs/plugins/inspect/plugin-y4menc.xml: + * win32/common/config.h: + Back to development + +=== release 0.10.26 === + +2010-12-01 21:15:09 +0000 Tim-Philipp Müller + + * ChangeLog: + * NEWS: + * RELEASE: + * configure.ac: + * docs/plugins/gst-plugins-good-plugins.args: + * docs/plugins/gst-plugins-good-plugins.hierarchy: + * docs/plugins/gst-plugins-good-plugins.interfaces: + * docs/plugins/gst-plugins-good-plugins.prerequisites: + * docs/plugins/inspect/plugin-1394.xml: + * docs/plugins/inspect/plugin-aasink.xml: + * docs/plugins/inspect/plugin-alaw.xml: + * docs/plugins/inspect/plugin-alpha.xml: + * docs/plugins/inspect/plugin-alphacolor.xml: + * docs/plugins/inspect/plugin-annodex.xml: + * docs/plugins/inspect/plugin-apetag.xml: + * docs/plugins/inspect/plugin-audiofx.xml: + * docs/plugins/inspect/plugin-auparse.xml: + * docs/plugins/inspect/plugin-autodetect.xml: + * docs/plugins/inspect/plugin-avi.xml: + * docs/plugins/inspect/plugin-cacasink.xml: + * docs/plugins/inspect/plugin-cairo.xml: + * docs/plugins/inspect/plugin-cutter.xml: + * docs/plugins/inspect/plugin-debug.xml: + * docs/plugins/inspect/plugin-deinterlace.xml: + * docs/plugins/inspect/plugin-dv.xml: + * docs/plugins/inspect/plugin-efence.xml: + * docs/plugins/inspect/plugin-effectv.xml: + * docs/plugins/inspect/plugin-equalizer.xml: + * docs/plugins/inspect/plugin-esdsink.xml: + * docs/plugins/inspect/plugin-flac.xml: + * docs/plugins/inspect/plugin-flv.xml: + * docs/plugins/inspect/plugin-flxdec.xml: + * docs/plugins/inspect/plugin-gconfelements.xml: + * docs/plugins/inspect/plugin-gdkpixbuf.xml: + * docs/plugins/inspect/plugin-goom.xml: + * docs/plugins/inspect/plugin-goom2k1.xml: + * docs/plugins/inspect/plugin-gstrtpmanager.xml: + * docs/plugins/inspect/plugin-halelements.xml: + * docs/plugins/inspect/plugin-icydemux.xml: + * docs/plugins/inspect/plugin-id3demux.xml: + * docs/plugins/inspect/plugin-imagefreeze.xml: + * docs/plugins/inspect/plugin-interleave.xml: + * docs/plugins/inspect/plugin-jpeg.xml: + * docs/plugins/inspect/plugin-level.xml: + * docs/plugins/inspect/plugin-matroska.xml: + * docs/plugins/inspect/plugin-mulaw.xml: + * docs/plugins/inspect/plugin-multifile.xml: + * docs/plugins/inspect/plugin-multipart.xml: + * docs/plugins/inspect/plugin-navigationtest.xml: + * docs/plugins/inspect/plugin-oss4.xml: + * docs/plugins/inspect/plugin-ossaudio.xml: + * docs/plugins/inspect/plugin-png.xml: + * docs/plugins/inspect/plugin-pulseaudio.xml: + * docs/plugins/inspect/plugin-quicktime.xml: + * docs/plugins/inspect/plugin-replaygain.xml: + * docs/plugins/inspect/plugin-rtp.xml: + * docs/plugins/inspect/plugin-rtsp.xml: + * docs/plugins/inspect/plugin-shapewipe.xml: + * docs/plugins/inspect/plugin-shout2send.xml: + * docs/plugins/inspect/plugin-smpte.xml: + * docs/plugins/inspect/plugin-soup.xml: + * docs/plugins/inspect/plugin-spectrum.xml: + * docs/plugins/inspect/plugin-speex.xml: + * docs/plugins/inspect/plugin-taglib.xml: + * docs/plugins/inspect/plugin-udp.xml: + * docs/plugins/inspect/plugin-video4linux2.xml: + * docs/plugins/inspect/plugin-videobox.xml: + * docs/plugins/inspect/plugin-videocrop.xml: + * docs/plugins/inspect/plugin-videofilter.xml: + * docs/plugins/inspect/plugin-videomixer.xml: + * docs/plugins/inspect/plugin-wavenc.xml: + * docs/plugins/inspect/plugin-wavpack.xml: + * docs/plugins/inspect/plugin-wavparse.xml: + * docs/plugins/inspect/plugin-ximagesrc.xml: + * docs/plugins/inspect/plugin-y4menc.xml: + * gst-plugins-good.doap: + * win32/common/config.h: + Release 0.10.26 2010-11-30 15:28:50 -0800 David Schleef diff --git a/NEWS b/NEWS index 7c3431d968..5eb6f34475 100644 --- a/NEWS +++ b/NEWS @@ -1,4 +1,107 @@ -This is GStreamer Good Plug-ins 0.10.26, "Escapades" +This is GStreamer Good Plug-ins 0.10.27, "Some Kind of Temporal Blend" + +Changes since 0.10.26: + + * avidemux: add workaround for buggy list size; extract datetime tags + * cacasink: fix masks and strides + * deinterlace: change the default to linear + * deinterlace: avoid infinite loop draining + * deinterlace: rewrite/fix how neighboring scan lines are calculated + * flvdemux: use aac codec-data to adjust samplerate if needed + * flvmux: Fix for nellymoser codecid setting + * icydemux: Add 'StreamUrl' metadata as GST_TAG_HOMEPAGE tag + * id3demux: fix parsing of ID3v2.4 genre frames with multiple genres + * imagefreeze: pass along eos if received before buffer arrives + * jpegdec: add "max-errors" property to ignore decoding errors + * jpegdec: avoid infinite loop when resyncing; discard incomplete image + * matroskademux: add stream-format and alignment properties for h264 + * matroskademux: assume matroska if no doctype is specified + * matroskademux: increase allowed max. block size for push mode from 10M to 15M + * matroskademux: normalize empty Cues to no Cues + * matroskamux: add support for DTS and E-AC3 audio + * matroskamux: try to write timestamps in all the outgoing buffers + * multifilesink: send stream headers in key-frame mode + * multiudpsink: add buffer-size property + * navseek: add basic support to change playback rate + * pulsemixer: Implement MIXER_FLAG_AUTO_NOTIFICATIONS + * pulsesink: flush remaining buffered samples on EOS + * pulsesink: make corking during pause synchronous; don't uncork in _start + * pulsesink: Uncork stream while flushing the ringbuffer + * pulsesrc: add "client" property + * qtdemux: add support for fragmented mp4 + * qtdemux: add support for (E)AC-3, WMA and VC-1 audio + * qtdemux: allow pulling atoms with unknown size + * qtdemux: fix flow return aggregation and handling of near end-of-file corner cases + * qtdemux: parse and use creation time tag from mvhd + * rtpbin: copy buffering stats + * rtpbin: correctly calculate RTCP packet size + * rtp: fix rank of payloaders and depayloaders + * rtp: flush state on flush-stop for seek handling for many (de)payloaders + * rtp ac3pay: add AC3 payloader + * rtp h264depay: determine output h264 layout using caps negotiation + * rtp h264pay: implement full bytestream scan mode + * rtp j2kdepay: add support for buffer lists; make depayloader more resilient + * rtp j2kpay: use buffer lists for better performance + * rtp j2kpay: handle EOC correctly; stop scanning when we reached the end + * rtp j2kpay: use SOP markers to split bitstream + * rtp jitterbuffer: provide a clock; get better buffering level + * rtp jpegdepay: fix framerate parsing for locales that use a comma as floating point + * rtp mp4adepay: improve timestamps on outgoing packets + * rtpsession: also emit RTCP activity on SR + * rtpsession: remember last sent RB values + * rtspsrc: add and use auto buffering mode + * rtspsrc: degrade gracefully upon failing seek and tweak QUERY_SEEKING response + * rtspsrc: include range request for all streams with non-aggregate control + * rtspsrc: increase udp buffer size + * rtspsrc: reset session manager base time when flushing + * rtspsrc: select multicast transports in a smarter way + * souphttpsrc: don't send seeks behind the end of file to the server + * v4l2sink: add navigation support; properties to control crop + * vrawdepay: fix length check + * wavparse: detect DTS advertised as PCM correctly in some more cases + * ximagesrc: change from XGetImage to XGetSubImage dependant on a property + +Bugs fixed since 0.10.26: + + * 596321 : qtdemux: add support for fragmented MP4 and " mfra " boxes + * 618389 : [pulsemixer] Should implement MIXER_FLAG_AUTO_NOTIFICATIONS interface + * 618652 : [effectv] Use of uninitialised value in unit test + * 620283 : Support for Adobe's F4F missing + * 621929 : [PLUGIN-MOVE] move jack plugin from -bad to -good + * 623178 : [matroskademux] error message for unrecognised FOURCC codes should be improved + * 625825 : cannot link rtpmp4adepay ! aacparse + * 629418 : progressreport: add support for determining stream position from buffer timestamps instead of using queries + * 631516 : [navseek] Add support to change playback rate + * 632654 : [matroskamux] try to write timestamps in most of the outgoing buffers + * 632897 : flvmux does not set the correct nellymoser codec id + * 633280 : [icydemux][PATCH] icydemux: Send 'StreamUrl' metadata as GST_TAG_HOMEPAGE tag + * 634314 : pngdec hangs on faulty pngs + * 634391 : [v4l2src] add interlaced field to caps + * 634393 : v4l2src: Set top field first for interlaced captures + * 634910 : [rtph264pay] Implement bytestream scan mode + * 634928 : [qtdemux] report creation/modification time via metadata tag + * 635734 : jpegdec: infinite loop when playing back motion jpeg stream + * 636049 : ximagesrc: fix remote X and off by ones + * 636172 : imagefreeze: eos is not passed before a buffer arrives + * 636234 : [wavparse] dts 6ch played as stereo 16 bit pcm if DTS frame starts at non-zero offset + * 636621 : flvdemux: doesn't set the right sample rate for aac audio + * 636784 : [qtdemux] GST_QUERY_CONVERT implementation for qtdemux + * 637060 : matroskademux: errors out on 13MB blocks when streaming + * 637686 : [jpegenc] Improve sinkpad getcaps results + * 638019 : [matroskademux] some matroska files are not specifying DocType + * 638072 : build failure: rtpsource.c: error: 'have_rb' may be used uninitialized in this function + * 638535 : id3demux: multiple genres as per ID3v2.4 not supported correctly + * 638569 : cacasink crashes when given 15-bit video. + * 639240 : pulsesink: PLAYING- > PAUSED- > PLAYING transition causes dropout + * 639321 : deinterlace: field{1,3} scanline pointers seem to be off by one field line + * 639339 : v4l2: fails to build with older kernels due to missing V4L_FIELD_INTERLACED_{TB,BT} + * 639516 : muxers: fix setting src pad caps + * 639740 : [pulsesink] doesn't uncork in some cases during reverse playback + * 640028 : [qtdemux] crash on malformed mov stream + * 640063 : rtph264depay: leaks codec data buffer in byte-stream=false mode + * 640064 : rtspsrc memory leak + * 640080 : rtspsrc: fails to error out properly on network failure + * 623063 : [jpegdec] add " max-errors " property Changes since 0.10.25: diff --git a/RELEASE b/RELEASE index 7ed324d26c..83a14b184e 100644 --- a/RELEASE +++ b/RELEASE @@ -1,5 +1,5 @@ -Release notes for GStreamer Good Plug-ins 0.10.26 "Escapades" +Release notes for GStreamer Good Plug-ins 0.10.27 "Some Kind of Temporal Blend" @@ -9,8 +9,6 @@ GStreamer Good Plug-ins. The 0.10.x series is a stable series targeted at end users. -It is not API or ABI compatible with the stable 0.8.x series. -It is, however, parallel installable with the 0.8.x series. @@ -54,125 +52,106 @@ contains a set of less supported plug-ins that haven't passed the Features of this release - * alphacolor: make passthrough work - * avidemux: reverse playback fixes; prevent overlap of subsequent fragments - * deinterlace: remove assembly code in favor of orc - * dvdemux: parse SMPTE time codes - * flvdemux: parse and use cts (fixes jittery H.264 playback in some cases) - * flvmux: resend onMetada tag when tags changes in streamable mode - * g729pay: extend from right parent - * gconf: Don't install schemas when GConf is disabled - * goom, goom2k1: add latency compensation code, report latency correctly - * gstrtpjpegpay: Added Define Restart Interval (DRI) Marker - * h264depay: always mark the codec_data as keyframe - * icydemux: forward tag events - * id3v2mux: Add mapping for album artist - * imagefreeze: generate a perfectly timestamped stream - * level: avoid division by zero on silence - * matroskademux: more robustness for parse errors and corner-cases - * matroskademux: extract H.264 profile and level and set on caps - * matroskamux: reduce newsegment event spam and set discont flag where needed - * pulse: allow setting of pulse stream properties - * pulse: fix device_description in READY - * pulsesink: Add "client" property to set the PA client name - * pulsesink: share the PA context between all clients with the same name - * qtdemux: export AAC/MPEG-4/H.264 profile and level in caps - * rtp: add G722 payloader and depayloader elements - * rtpamr(de)pay: support AMR-WB SID frame - * rtpamrpay: proper duration for multiple frame payload; properly support perfect-rtptime - * rtpbin: add "ntp-sync" property and "use-pipeline-clock" properties - * rtpg729pay: properly support perfect-rtptime - * rtph264depay: only set delta unit on all-non-key units - * rtpmanager: provide additional statistics - * rtpmp4adepay: grab the sampling rate and put into caps - * rtpmparobustdepay: properly insert dummy buffers; use valid bitrate for dummy frame - * rtpmpvpay: fix timestamping of rtp buffers - * rtpsession: Add the option to auto-discover the RTP bandwidth - * rtpsession: Calculate RTCP bandwidth as a fraction of the RTP bandwidth - * rtpsession: Count sent RTCP packets after they have been finished - * rtpsession: relax third-party collision detection - * rtpstats: Rectify description of current_time in RTPArrivalStats - * rtspext: stop configuration on first failure - * rtspsrc: Add property to configure udpsrc buffer size - * rtspsrc: add rtsp-sdp protocol support - * rtspsrc: don't add /UDP in the transport, it's the default - * rtspsrc: fix duration reporting - * rtspsrc: handle stale digest authentication session data - * rtspsrc: use sdp uri parse method - * shapewipe: add optional border parameter and slowdown animation - * shapewipe: Force format to AYUV in the example pipeline for the same reason - * shapewipe: Force the input to AYUV to prevent negotiation failures in videomixer - * spectrum: only aggregate magnitude/phase if user asks for it, performance fixes - * v4l2src: add controllable colorbalance parameters, add decimate property - * v4l2src: fix using mpegts via the mmap interface; use GstBaseSrc::block-size as fallback size - * videomixer2: new videomixer2 element that behaves better than videomixer - * vrawdepay: handle invalid payload better + * avidemux: add workaround for buggy list size; extract datetime tags + * cacasink: fix masks and strides + * deinterlace: change the default to linear + * deinterlace: avoid infinite loop draining + * deinterlace: rewrite/fix how neighboring scan lines are calculated + * flvdemux: use aac codec-data to adjust samplerate if needed + * flvmux: Fix for nellymoser codecid setting + * icydemux: Add 'StreamUrl' metadata as GST_TAG_HOMEPAGE tag + * id3demux: fix parsing of ID3v2.4 genre frames with multiple genres + * imagefreeze: pass along eos if received before buffer arrives + * jpegdec: add "max-errors" property to ignore decoding errors + * jpegdec: avoid infinite loop when resyncing; discard incomplete image + * matroskademux: add stream-format and alignment properties for h264 + * matroskademux: assume matroska if no doctype is specified + * matroskademux: increase allowed max. block size for push mode from 10M to 15M + * matroskademux: normalize empty Cues to no Cues + * matroskamux: add support for DTS and E-AC3 audio + * matroskamux: try to write timestamps in all the outgoing buffers + * multifilesink: send stream headers in key-frame mode + * multiudpsink: add buffer-size property + * navseek: add basic support to change playback rate + * pulsemixer: Implement MIXER_FLAG_AUTO_NOTIFICATIONS + * pulsesink: flush remaining buffered samples on EOS + * pulsesink: make corking during pause synchronous; don't uncork in _start + * pulsesink: Uncork stream while flushing the ringbuffer + * pulsesrc: add "client" property + * qtdemux: add support for fragmented mp4 + * qtdemux: add support for (E)AC-3, WMA and VC-1 audio + * qtdemux: allow pulling atoms with unknown size + * qtdemux: fix flow return aggregation and handling of near end-of-file corner cases + * qtdemux: parse and use creation time tag from mvhd + * rtpbin: copy buffering stats + * rtpbin: correctly calculate RTCP packet size + * rtp: fix rank of payloaders and depayloaders + * rtp: flush state on flush-stop for seek handling for many (de)payloaders + * rtp ac3pay: add AC3 payloader + * rtp h264depay: determine output h264 layout using caps negotiation + * rtp h264pay: implement full bytestream scan mode + * rtp j2kdepay: add support for buffer lists; make depayloader more resilient + * rtp j2kpay: use buffer lists for better performance + * rtp j2kpay: handle EOC correctly; stop scanning when we reached the end + * rtp j2kpay: use SOP markers to split bitstream + * rtp jitterbuffer: provide a clock; get better buffering level + * rtp jpegdepay: fix framerate parsing for locales that use a comma as floating point + * rtp mp4adepay: improve timestamps on outgoing packets + * rtpsession: also emit RTCP activity on SR + * rtpsession: remember last sent RB values + * rtspsrc: add and use auto buffering mode + * rtspsrc: degrade gracefully upon failing seek and tweak QUERY_SEEKING response + * rtspsrc: include range request for all streams with non-aggregate control + * rtspsrc: increase udp buffer size + * rtspsrc: reset session manager base time when flushing + * rtspsrc: select multicast transports in a smarter way + * souphttpsrc: don't send seeks behind the end of file to the server + * v4l2sink: add navigation support; properties to control crop + * vrawdepay: fix length check + * wavparse: detect DTS advertised as PCM correctly in some more cases + * ximagesrc: change from XGetImage to XGetSubImage dependant on a property Bugs fixed in this release + * 596321 : qtdemux: add support for fragmented MP4 and " mfra " boxes + * 618389 : [pulsemixer] Should implement MIXER_FLAG_AUTO_NOTIFICATIONS interface + * 618652 : [effectv] Use of uninitialised value in unit test + * 620283 : Support for Adobe's F4F missing + * 621929 : [PLUGIN-MOVE] move jack plugin from -bad to -good + * 623178 : [matroskademux] error message for unrecognised FOURCC codes should be improved * 625825 : cannot link rtpmp4adepay ! aacparse - * 629047 : segfault in seek matroskademux - * 537544 : [pulse] allow setting pa context properties - * 628996 : pulsesink broken after shared context patch (bug #624338) - * 529672 : Big latency and bad framerate while mixing multiple live streams - * 581294 : rtspext: extensions configure_stream methods conflict - * 598915 : qtdemux: propagate jpeg2000 header data in image/x-j2c - * 612313 : qtdemux: Post AAC profile/level in caps - * 616521 : qtdemux: Export MPEG-4 video profile and level in stream caps - * 617318 : matroskademux, qtdemux: Use pbutils for H.264 profile/level extraction - * 620790 : [matroskademux] general stream error when trying to play certain .mkv file - * 622390 : [v4l2] add controllable color balance properties / programmable camera - * 624338 : [pulsesink] Handle pulse context separate from the ringbuffers and share them - * 625547 : imagefreeze unit test fails occasionally - * 626048 : [videomixer] needs mode that syncs streams based on timestamps - * 626518 : [imagefreeze] better caps negotiation - * 627162 : [pulse] better fallback return value for gst_pulse_client_name() - * 627174 : [pulsesink] new property to tune the PA client name - * 627289 : souphttpsrc: tweak error messages - * 627341 : wavparse: strange handling of files less than 12 bytes - * 627796 : rtpbin: add ntp clock sync - * 628020 : [pulsesink] assertion failure in change_state NULL- > READY - * 628058 : Need a way to set the SO_RCVBUF property on rtsp-based sockets. - * 628127 : jpeg rtp payloader crashes when there is corruption in the jpeg byte stream. - * 628214 : Add support to RTSP initiation through SDP files - * 628349 : [v4l2src] Doesn't support capturing mpegts using mmap - * 628454 : Matroska demuxer doesn't handle DATE tag if it contains only a year number - * 628608 : [alphacolor] element classification is wrong - * 629018 : rtpjpegpay: unable to build because of uninitialized variable warning - * 629522 : [rtpjpegpay] add support for Define Restart Interval (DRI) - * 629839 : [qtdemux] Update xmp tags parsing - * 629896 : Error compiling raw1394 (without iec61883) - * 630088 : [flvdemux] jerky h.264 video playback - * 630205 : [icydemux] Forward tag events downstrem - * 630256 : rtph264-pay/depay: doesn't respect timestamps from incomming buffers - * 630317 : Getting pulsesink device names doesn't work like for alsasink - * 630378 : speexenc/speexdec crash with MSVC - * 630446 : rtpmanager: provide additional statistics - * 630447 : rtpsession: relax third-party collision detection - * 630449 : rtpbin: Unlock before adding pad in new_payload_found - * 630451 : rtpbin: Handle rysnc of iterator when looking for free pad name - * 630452 : rtpbin: Make cleaning up sources in rtp_session_on_timeout MT safe - * 630457 : rtpmanager: packet lost should not be a warning. - * 630458 : level: avoid division by zero on silence - * 630500 : [rtspsrc] does rtsp setup message always need " /UDP " string? - * 630888 : v4l2sink does not cope with v4l2loopback kernel module - * 631082 : rtpjitterbuffer: improve document reference - * 631303 : [goom] qos warnings if source is GstAudioSrc - * 631330 : [flvmux][PATCH] Resend updated onMetada tag when tags changes in streamable mode - * 631996 : [h264depay] regression: rtsp://stream.zoovision.com/KibaEp1n900.3gp - * 632548 : [rtspsrc] regression; fails to report duration - * 632553 : --disable-gconf still tries to install schemas - * 632682 : [matroskademux] Handle missing CodecPrivate for Vorbis/Theora - * 632945 : rtph264depay in access-unit=true mode does not aggregate the delta unit flag correctly - * 633205 : Fix for navigation events in videoflip - * 633212 : [goom] Return not-negotiated when bps is unknown - * 633970 : [icydemux] broken taglist handling - * 635532 : rtspsrc: unexpected eos when using authentication (regression) - * 635843 : [rtph264depay] segfault on empty payload - * 636179 : [deinterlace] Fields in wrong order - * 626463 : [matroskademux] " reading large block of size 14688496 not supported " - * 628894 : [matroskademux] sloppy reverse playback - * 633294 : deinterlace breaks some DVD menu scenarios + * 629418 : progressreport: add support for determining stream position from buffer timestamps instead of using queries + * 631516 : [navseek] Add support to change playback rate + * 632654 : [matroskamux] try to write timestamps in most of the outgoing buffers + * 632897 : flvmux does not set the correct nellymoser codec id + * 633280 : [icydemux][PATCH] icydemux: Send 'StreamUrl' metadata as GST_TAG_HOMEPAGE tag + * 634314 : pngdec hangs on faulty pngs + * 634391 : [v4l2src] add interlaced field to caps + * 634393 : v4l2src: Set top field first for interlaced captures + * 634910 : [rtph264pay] Implement bytestream scan mode + * 634928 : [qtdemux] report creation/modification time via metadata tag + * 635734 : jpegdec: infinite loop when playing back motion jpeg stream + * 636049 : ximagesrc: fix remote X and off by ones + * 636172 : imagefreeze: eos is not passed before a buffer arrives + * 636234 : [wavparse] dts 6ch played as stereo 16 bit pcm if DTS frame starts at non-zero offset + * 636621 : flvdemux: doesn't set the right sample rate for aac audio + * 636784 : [qtdemux] GST_QUERY_CONVERT implementation for qtdemux + * 637060 : matroskademux: errors out on 13MB blocks when streaming + * 637686 : [jpegenc] Improve sinkpad getcaps results + * 638019 : [matroskademux] some matroska files are not specifying DocType + * 638072 : build failure: rtpsource.c: error: 'have_rb' may be used uninitialized in this function + * 638535 : id3demux: multiple genres as per ID3v2.4 not supported correctly + * 638569 : cacasink crashes when given 15-bit video. + * 639240 : pulsesink: PLAYING- > PAUSED- > PLAYING transition causes dropout + * 639321 : deinterlace: field{1,3} scanline pointers seem to be off by one field line + * 639339 : v4l2: fails to build with older kernels due to missing V4L_FIELD_INTERLACED_{TB,BT} + * 639516 : muxers: fix setting src pad caps + * 639740 : [pulsesink] doesn't uncork in some cases during reverse playback + * 640028 : [qtdemux] crash on malformed mov stream + * 640063 : rtph264depay: leaks codec data buffer in byte-stream=false mode + * 640064 : rtspsrc memory leak + * 640080 : rtspsrc: fails to error out properly on network failure + * 623063 : [jpegdec] add " max-errors " property Download @@ -202,34 +181,38 @@ Applications Contributors to this release * Alessandro Decina - * American Dynamics * Andoni Morales Alastruey + * Andy Wingo * Arun Raghavan - * Bastien Nocera + * Benjamin Gaignard + * Benjamin Otte + * Christian Schaller * David Hoyt * David Schleef * Edward Hervey - * Havard Graff - * IOhannes m zmölnig + * Erich Schubert + * Guillaume Emont + * Iain Holmes * Jan Schmidt - * Jonathan Matthew + * Janne Grunau + * Johan Dahlin + * Kishore Arepalli + * Leif Johnson + * Marc-André Lureau * Mark Nauwelaerts - * Olivier Crête - * Pascal Buhler - * Pavel Kostyuchenko - * Philip Jägenstedt - * Philippe Normand - * René Stadler - * Robert Swain + * Paul Davis + * Rob Clark + * Ronald S. Bultje * Sebastian Dröge - * Sjoerd Simons * Stefan Kost + * Steve Baker + * Stéphane Loeuillet + * Tambet Ingo * Thiago Santos - * Thibault Saunier - * Thijs Vermeir + * Thomas Vander Stichele * Tim-Philipp Müller - * Trond Andersen - * Vladimir Eremeev + * Tom Janiszewski + * Tristan Matthews + * Vincent Penquerc'h * Wim Taymans - * Zaheer Abbas Merali   \ No newline at end of file diff --git a/configure.ac b/configure.ac index 61043a31a6..c7757bbee1 100644 --- a/configure.ac +++ b/configure.ac @@ -5,7 +5,7 @@ dnl please read gstreamer/docs/random/autotools before changing this file dnl initialize autoconf dnl releases only do -Wall, git and prerelease does -Werror too dnl use a three digit version number for releases, and four for git/pre -AC_INIT(GStreamer Good Plug-ins, 0.10.26.4, +AC_INIT(GStreamer Good Plug-ins, 0.10.27, http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer, gst-plugins-good) @@ -52,8 +52,8 @@ AC_LIBTOOL_WIN32_DLL AM_PROG_LIBTOOL dnl *** required versions of GStreamer stuff *** -GST_REQ=0.10.31.3 -GSTPB_REQ=0.10.31.3 +GST_REQ=0.10.32 +GSTPB_REQ=0.10.32 dnl *** autotools stuff **** diff --git a/docs/plugins/inspect/plugin-1394.xml b/docs/plugins/inspect/plugin-1394.xml index 36459372b0..e70a8f365e 100644 --- a/docs/plugins/inspect/plugin-1394.xml +++ b/docs/plugins/inspect/plugin-1394.xml @@ -3,10 +3,10 @@ Source for video data via IEEE1394 interface ../../ext/raw1394/.libs/libgst1394.so libgst1394.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-aasink.xml b/docs/plugins/inspect/plugin-aasink.xml index 58881197e1..7805c93198 100644 --- a/docs/plugins/inspect/plugin-aasink.xml +++ b/docs/plugins/inspect/plugin-aasink.xml @@ -3,10 +3,10 @@ ASCII Art video sink ../../ext/aalib/.libs/libgstaasink.so libgstaasink.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-alaw.xml b/docs/plugins/inspect/plugin-alaw.xml index f2f3a63022..3f8d8d25d4 100644 --- a/docs/plugins/inspect/plugin-alaw.xml +++ b/docs/plugins/inspect/plugin-alaw.xml @@ -3,10 +3,10 @@ ALaw audio conversion routines ../../gst/law/.libs/libgstalaw.so libgstalaw.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-alpha.xml b/docs/plugins/inspect/plugin-alpha.xml index e406b9e8c8..73972fca65 100644 --- a/docs/plugins/inspect/plugin-alpha.xml +++ b/docs/plugins/inspect/plugin-alpha.xml @@ -3,10 +3,10 @@ adds an alpha channel to video - constant or via chroma-keying ../../gst/alpha/.libs/libgstalpha.so libgstalpha.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-alphacolor.xml b/docs/plugins/inspect/plugin-alphacolor.xml index 652430e630..24e394ee4b 100644 --- a/docs/plugins/inspect/plugin-alphacolor.xml +++ b/docs/plugins/inspect/plugin-alphacolor.xml @@ -3,10 +3,10 @@ RGBA from/to AYUV colorspace conversion preserving the alpha channel ../../gst/alpha/.libs/libgstalphacolor.so libgstalphacolor.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-annodex.xml b/docs/plugins/inspect/plugin-annodex.xml index 3cc312eb1e..5e709d2a54 100644 --- a/docs/plugins/inspect/plugin-annodex.xml +++ b/docs/plugins/inspect/plugin-annodex.xml @@ -3,10 +3,10 @@ annodex stream manipulation (info about annodex: http://www.annodex.net) ../../ext/annodex/.libs/libgstannodex.so libgstannodex.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-apetag.xml b/docs/plugins/inspect/plugin-apetag.xml index 2ea6f67ac4..67707f6b08 100644 --- a/docs/plugins/inspect/plugin-apetag.xml +++ b/docs/plugins/inspect/plugin-apetag.xml @@ -3,10 +3,10 @@ APEv1/2 tag reader ../../gst/apetag/.libs/libgstapetag.so libgstapetag.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-audiofx.xml b/docs/plugins/inspect/plugin-audiofx.xml index 46904fdc6c..3e5e84a310 100644 --- a/docs/plugins/inspect/plugin-audiofx.xml +++ b/docs/plugins/inspect/plugin-audiofx.xml @@ -3,10 +3,10 @@ Audio effects plugin ../../gst/audiofx/.libs/libgstaudiofx.so libgstaudiofx.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-auparse.xml b/docs/plugins/inspect/plugin-auparse.xml index 374a368aaf..b6ed29c959 100644 --- a/docs/plugins/inspect/plugin-auparse.xml +++ b/docs/plugins/inspect/plugin-auparse.xml @@ -3,10 +3,10 @@ parses au streams ../../gst/auparse/.libs/libgstauparse.so libgstauparse.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-autodetect.xml b/docs/plugins/inspect/plugin-autodetect.xml index 9b2ba39a4e..15be26f8dd 100644 --- a/docs/plugins/inspect/plugin-autodetect.xml +++ b/docs/plugins/inspect/plugin-autodetect.xml @@ -3,10 +3,10 @@ Plugin contains auto-detection plugins for video/audio in- and outputs ../../gst/autodetect/.libs/libgstautodetect.so libgstautodetect.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-avi.xml b/docs/plugins/inspect/plugin-avi.xml index e3287a14ca..e4a30b8e16 100644 --- a/docs/plugins/inspect/plugin-avi.xml +++ b/docs/plugins/inspect/plugin-avi.xml @@ -3,10 +3,10 @@ AVI stream handling ../../gst/avi/.libs/libgstavi.so libgstavi.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-cacasink.xml b/docs/plugins/inspect/plugin-cacasink.xml index 40b8216164..25d58662db 100644 --- a/docs/plugins/inspect/plugin-cacasink.xml +++ b/docs/plugins/inspect/plugin-cacasink.xml @@ -3,10 +3,10 @@ Colored ASCII Art video sink ../../ext/libcaca/.libs/libgstcacasink.so libgstcacasink.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-cairo.xml b/docs/plugins/inspect/plugin-cairo.xml index b5b1758efa..26483ab38c 100644 --- a/docs/plugins/inspect/plugin-cairo.xml +++ b/docs/plugins/inspect/plugin-cairo.xml @@ -3,10 +3,10 @@ Cairo-based elements ../../ext/cairo/.libs/libgstcairo.so libgstcairo.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-cutter.xml b/docs/plugins/inspect/plugin-cutter.xml index e004a021d8..f5edf8fb2f 100644 --- a/docs/plugins/inspect/plugin-cutter.xml +++ b/docs/plugins/inspect/plugin-cutter.xml @@ -3,10 +3,10 @@ Audio Cutter to split audio into non-silent bits ../../gst/cutter/.libs/libgstcutter.so libgstcutter.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-debug.xml b/docs/plugins/inspect/plugin-debug.xml index da8d22b9f0..5d307928d9 100644 --- a/docs/plugins/inspect/plugin-debug.xml +++ b/docs/plugins/inspect/plugin-debug.xml @@ -3,10 +3,10 @@ elements for testing and debugging ../../gst/debugutils/.libs/libgstdebug.so libgstdebug.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-deinterlace.xml b/docs/plugins/inspect/plugin-deinterlace.xml index 6414aa706b..455df7df26 100644 --- a/docs/plugins/inspect/plugin-deinterlace.xml +++ b/docs/plugins/inspect/plugin-deinterlace.xml @@ -3,10 +3,10 @@ Deinterlacer ../../gst/deinterlace/.libs/libgstdeinterlace.so libgstdeinterlace.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-dv.xml b/docs/plugins/inspect/plugin-dv.xml index a9e7a7c119..25d098a85c 100644 --- a/docs/plugins/inspect/plugin-dv.xml +++ b/docs/plugins/inspect/plugin-dv.xml @@ -3,10 +3,10 @@ DV demuxer and decoder based on libdv (libdv.sf.net) ../../ext/dv/.libs/libgstdv.so libgstdv.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-efence.xml b/docs/plugins/inspect/plugin-efence.xml index af1edacf26..9ab33e6568 100644 --- a/docs/plugins/inspect/plugin-efence.xml +++ b/docs/plugins/inspect/plugin-efence.xml @@ -3,10 +3,10 @@ This element converts a stream of normal GStreamer buffers into a stream of buffers that are allocated in such a way that out-of-bounds access to data in the buffer is more likely to cause segmentation faults. This allocation method is very similar to the debugging tool "Electric Fence". ../../gst/debugutils/.libs/libgstefence.so libgstefence.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-effectv.xml b/docs/plugins/inspect/plugin-effectv.xml index 21b4b3ebf7..99c3e59f72 100644 --- a/docs/plugins/inspect/plugin-effectv.xml +++ b/docs/plugins/inspect/plugin-effectv.xml @@ -3,10 +3,10 @@ effect plugins from the effectv project ../../gst/effectv/.libs/libgsteffectv.so libgsteffectv.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-equalizer.xml b/docs/plugins/inspect/plugin-equalizer.xml index 6cd90a49d3..334f21bd0f 100644 --- a/docs/plugins/inspect/plugin-equalizer.xml +++ b/docs/plugins/inspect/plugin-equalizer.xml @@ -3,10 +3,10 @@ GStreamer audio equalizers ../../gst/equalizer/.libs/libgstequalizer.so libgstequalizer.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-esdsink.xml b/docs/plugins/inspect/plugin-esdsink.xml index cdea83b642..feb9021f06 100644 --- a/docs/plugins/inspect/plugin-esdsink.xml +++ b/docs/plugins/inspect/plugin-esdsink.xml @@ -3,10 +3,10 @@ ESD Element Plugins ../../ext/esd/.libs/libgstesd.so libgstesd.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-flac.xml b/docs/plugins/inspect/plugin-flac.xml index 32d6614df9..a4133713a8 100644 --- a/docs/plugins/inspect/plugin-flac.xml +++ b/docs/plugins/inspect/plugin-flac.xml @@ -3,10 +3,10 @@ The FLAC Lossless compressor Codec ../../ext/flac/.libs/libgstflac.so libgstflac.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-flv.xml b/docs/plugins/inspect/plugin-flv.xml index f01c04c410..e1afab1a18 100644 --- a/docs/plugins/inspect/plugin-flv.xml +++ b/docs/plugins/inspect/plugin-flv.xml @@ -3,10 +3,10 @@ FLV muxing and demuxing plugin ../../gst/flv/.libs/libgstflv.so libgstflv.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-flxdec.xml b/docs/plugins/inspect/plugin-flxdec.xml index e0c36f65d1..2ec08d559f 100644 --- a/docs/plugins/inspect/plugin-flxdec.xml +++ b/docs/plugins/inspect/plugin-flxdec.xml @@ -3,10 +3,10 @@ FLC/FLI/FLX video decoder ../../gst/flx/.libs/libgstflxdec.so libgstflxdec.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-gconfelements.xml b/docs/plugins/inspect/plugin-gconfelements.xml index 8367254b41..1e477bc9a9 100644 --- a/docs/plugins/inspect/plugin-gconfelements.xml +++ b/docs/plugins/inspect/plugin-gconfelements.xml @@ -3,10 +3,10 @@ elements wrapping the GStreamer/GConf audio/video output settings ../../ext/gconf/.libs/libgstgconfelements.so libgstgconfelements.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-gdkpixbuf.xml b/docs/plugins/inspect/plugin-gdkpixbuf.xml index 87627d80f1..aabb20f220 100644 --- a/docs/plugins/inspect/plugin-gdkpixbuf.xml +++ b/docs/plugins/inspect/plugin-gdkpixbuf.xml @@ -3,10 +3,10 @@ GdkPixbuf-based image decoder, scaler and sink ../../ext/gdk_pixbuf/.libs/libgstgdkpixbuf.so libgstgdkpixbuf.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-goom.xml b/docs/plugins/inspect/plugin-goom.xml index e6f39049b7..8bf4060781 100644 --- a/docs/plugins/inspect/plugin-goom.xml +++ b/docs/plugins/inspect/plugin-goom.xml @@ -3,10 +3,10 @@ GOOM visualization filter ../../gst/goom/.libs/libgstgoom.so libgstgoom.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-goom2k1.xml b/docs/plugins/inspect/plugin-goom2k1.xml index fa0cc637ad..d70137e95e 100644 --- a/docs/plugins/inspect/plugin-goom2k1.xml +++ b/docs/plugins/inspect/plugin-goom2k1.xml @@ -3,10 +3,10 @@ GOOM 2k1 visualization filter ../../gst/goom2k1/.libs/libgstgoom2k1.so libgstgoom2k1.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-gstrtpmanager.xml b/docs/plugins/inspect/plugin-gstrtpmanager.xml index 769a97c34e..aa2a7538ee 100644 --- a/docs/plugins/inspect/plugin-gstrtpmanager.xml +++ b/docs/plugins/inspect/plugin-gstrtpmanager.xml @@ -3,10 +3,10 @@ RTP session management plugin library ../../gst/rtpmanager/.libs/libgstrtpmanager.so libgstrtpmanager.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-halelements.xml b/docs/plugins/inspect/plugin-halelements.xml index 61bb87cfff..224abf3ad5 100644 --- a/docs/plugins/inspect/plugin-halelements.xml +++ b/docs/plugins/inspect/plugin-halelements.xml @@ -3,10 +3,10 @@ elements wrapping the GStreamer/HAL audio input/output devices ../../ext/hal/.libs/libgsthalelements.so libgsthalelements.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-icydemux.xml b/docs/plugins/inspect/plugin-icydemux.xml index 8ac3061619..d30ca100bc 100644 --- a/docs/plugins/inspect/plugin-icydemux.xml +++ b/docs/plugins/inspect/plugin-icydemux.xml @@ -3,10 +3,10 @@ Demux ICY tags from a stream ../../gst/icydemux/.libs/libgsticydemux.so libgsticydemux.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-id3demux.xml b/docs/plugins/inspect/plugin-id3demux.xml index 82b2a560fb..937ae2e982 100644 --- a/docs/plugins/inspect/plugin-id3demux.xml +++ b/docs/plugins/inspect/plugin-id3demux.xml @@ -3,10 +3,10 @@ Demux ID3v1 and ID3v2 tags from a file ../../gst/id3demux/.libs/libgstid3demux.so libgstid3demux.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-imagefreeze.xml b/docs/plugins/inspect/plugin-imagefreeze.xml index 90b764513b..54f5b85893 100644 --- a/docs/plugins/inspect/plugin-imagefreeze.xml +++ b/docs/plugins/inspect/plugin-imagefreeze.xml @@ -3,10 +3,10 @@ Still frame stream generator ../../gst/imagefreeze/.libs/libgstimagefreeze.so libgstimagefreeze.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-interleave.xml b/docs/plugins/inspect/plugin-interleave.xml index e60911ddfd..4b3c60af0c 100644 --- a/docs/plugins/inspect/plugin-interleave.xml +++ b/docs/plugins/inspect/plugin-interleave.xml @@ -3,10 +3,10 @@ Audio interleaver/deinterleaver ../../gst/interleave/.libs/libgstinterleave.so libgstinterleave.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-jack.xml b/docs/plugins/inspect/plugin-jack.xml index c8cc852806..0e92a21760 100644 --- a/docs/plugins/inspect/plugin-jack.xml +++ b/docs/plugins/inspect/plugin-jack.xml @@ -3,10 +3,10 @@ JACK audio elements ../../ext/jack/.libs/libgstjack.so libgstjack.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-jpeg.xml b/docs/plugins/inspect/plugin-jpeg.xml index fa8d04c3b2..9ae11c5394 100644 --- a/docs/plugins/inspect/plugin-jpeg.xml +++ b/docs/plugins/inspect/plugin-jpeg.xml @@ -3,10 +3,10 @@ JPeg plugin library ../../ext/jpeg/.libs/libgstjpeg.so libgstjpeg.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-level.xml b/docs/plugins/inspect/plugin-level.xml index f2f62c1c88..198cb7fb88 100644 --- a/docs/plugins/inspect/plugin-level.xml +++ b/docs/plugins/inspect/plugin-level.xml @@ -3,10 +3,10 @@ Audio level plugin ../../gst/level/.libs/libgstlevel.so libgstlevel.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-matroska.xml b/docs/plugins/inspect/plugin-matroska.xml index 1a1c3b684b..9604442a95 100644 --- a/docs/plugins/inspect/plugin-matroska.xml +++ b/docs/plugins/inspect/plugin-matroska.xml @@ -3,10 +3,10 @@ Matroska and WebM stream handling ../../gst/matroska/.libs/libgstmatroska.so libgstmatroska.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-mulaw.xml b/docs/plugins/inspect/plugin-mulaw.xml index a76367cae0..49429116e5 100644 --- a/docs/plugins/inspect/plugin-mulaw.xml +++ b/docs/plugins/inspect/plugin-mulaw.xml @@ -3,10 +3,10 @@ MuLaw audio conversion routines ../../gst/law/.libs/libgstmulaw.so libgstmulaw.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-multifile.xml b/docs/plugins/inspect/plugin-multifile.xml index ee32044194..bb2d80adb9 100644 --- a/docs/plugins/inspect/plugin-multifile.xml +++ b/docs/plugins/inspect/plugin-multifile.xml @@ -3,10 +3,10 @@ Reads/Writes buffers from/to sequentially named files ../../gst/multifile/.libs/libgstmultifile.so libgstmultifile.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-multipart.xml b/docs/plugins/inspect/plugin-multipart.xml index f18f90c6ec..e5ef7b422e 100644 --- a/docs/plugins/inspect/plugin-multipart.xml +++ b/docs/plugins/inspect/plugin-multipart.xml @@ -3,10 +3,10 @@ multipart stream manipulation ../../gst/multipart/.libs/libgstmultipart.so libgstmultipart.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-navigationtest.xml b/docs/plugins/inspect/plugin-navigationtest.xml index a4acdd8ff9..0dc2377fe5 100644 --- a/docs/plugins/inspect/plugin-navigationtest.xml +++ b/docs/plugins/inspect/plugin-navigationtest.xml @@ -3,10 +3,10 @@ Template for a video filter ../../gst/debugutils/.libs/libgstnavigationtest.so libgstnavigationtest.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-oss4.xml b/docs/plugins/inspect/plugin-oss4.xml index 69fc251634..0a59daa9f7 100644 --- a/docs/plugins/inspect/plugin-oss4.xml +++ b/docs/plugins/inspect/plugin-oss4.xml @@ -3,10 +3,10 @@ Open Sound System (OSS) version 4 support for GStreamer ../../sys/oss4/.libs/libgstoss4audio.so libgstoss4audio.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-ossaudio.xml b/docs/plugins/inspect/plugin-ossaudio.xml index 51483d1ee6..ddc69591c3 100644 --- a/docs/plugins/inspect/plugin-ossaudio.xml +++ b/docs/plugins/inspect/plugin-ossaudio.xml @@ -3,10 +3,10 @@ OSS (Open Sound System) support for GStreamer ../../sys/oss/.libs/libgstossaudio.so libgstossaudio.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-png.xml b/docs/plugins/inspect/plugin-png.xml index ee491d8829..409b496a99 100644 --- a/docs/plugins/inspect/plugin-png.xml +++ b/docs/plugins/inspect/plugin-png.xml @@ -3,10 +3,10 @@ PNG plugin library ../../ext/libpng/.libs/libgstpng.so libgstpng.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-pulseaudio.xml b/docs/plugins/inspect/plugin-pulseaudio.xml index 0c4250fcdb..f3e1647405 100644 --- a/docs/plugins/inspect/plugin-pulseaudio.xml +++ b/docs/plugins/inspect/plugin-pulseaudio.xml @@ -3,10 +3,10 @@ PulseAudio plugin library ../../ext/pulse/.libs/libgstpulse.so libgstpulse.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-quicktime.xml b/docs/plugins/inspect/plugin-quicktime.xml index 018278c500..44edb00923 100644 --- a/docs/plugins/inspect/plugin-quicktime.xml +++ b/docs/plugins/inspect/plugin-quicktime.xml @@ -3,10 +3,10 @@ Quicktime support ../../gst/qtdemux/.libs/libgstqtdemux.so libgstqtdemux.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-replaygain.xml b/docs/plugins/inspect/plugin-replaygain.xml index fd75b24cfe..b8d2e3f65b 100644 --- a/docs/plugins/inspect/plugin-replaygain.xml +++ b/docs/plugins/inspect/plugin-replaygain.xml @@ -3,10 +3,10 @@ ReplayGain volume normalization ../../gst/replaygain/.libs/libgstreplaygain.so libgstreplaygain.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-rtp.xml b/docs/plugins/inspect/plugin-rtp.xml index 1a5a20f56e..4023abc33e 100644 --- a/docs/plugins/inspect/plugin-rtp.xml +++ b/docs/plugins/inspect/plugin-rtp.xml @@ -3,10 +3,10 @@ Real-time protocol plugins ../../gst/rtp/.libs/libgstrtp.so libgstrtp.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-rtsp.xml b/docs/plugins/inspect/plugin-rtsp.xml index 83c4e466e1..f5b2259591 100644 --- a/docs/plugins/inspect/plugin-rtsp.xml +++ b/docs/plugins/inspect/plugin-rtsp.xml @@ -3,10 +3,10 @@ transfer data via RTSP ../../gst/rtsp/.libs/libgstrtsp.so libgstrtsp.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-shapewipe.xml b/docs/plugins/inspect/plugin-shapewipe.xml index a322f8d69c..2e06ea3b2e 100644 --- a/docs/plugins/inspect/plugin-shapewipe.xml +++ b/docs/plugins/inspect/plugin-shapewipe.xml @@ -3,10 +3,10 @@ Shape Wipe transition filter ../../gst/shapewipe/.libs/libgstshapewipe.so libgstshapewipe.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-shout2send.xml b/docs/plugins/inspect/plugin-shout2send.xml index 3b483c4b62..25ff258009 100644 --- a/docs/plugins/inspect/plugin-shout2send.xml +++ b/docs/plugins/inspect/plugin-shout2send.xml @@ -3,7 +3,7 @@ Sends data to an icecast server using libshout2 ../../ext/shout2/.libs/libgstshout2.so libgstshout2.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good libshout2 diff --git a/docs/plugins/inspect/plugin-smpte.xml b/docs/plugins/inspect/plugin-smpte.xml index a2e30a118b..c84436523d 100644 --- a/docs/plugins/inspect/plugin-smpte.xml +++ b/docs/plugins/inspect/plugin-smpte.xml @@ -3,10 +3,10 @@ Apply the standard SMPTE transitions on video images ../../gst/smpte/.libs/libgstsmpte.so libgstsmpte.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-soup.xml b/docs/plugins/inspect/plugin-soup.xml index 772fb6ec2d..92cb65e7a3 100644 --- a/docs/plugins/inspect/plugin-soup.xml +++ b/docs/plugins/inspect/plugin-soup.xml @@ -3,10 +3,10 @@ libsoup HTTP client src ../../ext/soup/.libs/libgstsouphttpsrc.so libgstsouphttpsrc.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-spectrum.xml b/docs/plugins/inspect/plugin-spectrum.xml index 9034dbd7a9..6cb6a0145b 100644 --- a/docs/plugins/inspect/plugin-spectrum.xml +++ b/docs/plugins/inspect/plugin-spectrum.xml @@ -3,10 +3,10 @@ Run an FFT on the audio signal, output spectrum data ../../gst/spectrum/.libs/libgstspectrum.so libgstspectrum.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-speex.xml b/docs/plugins/inspect/plugin-speex.xml index 50dcc6d699..b4bfb0ea92 100644 --- a/docs/plugins/inspect/plugin-speex.xml +++ b/docs/plugins/inspect/plugin-speex.xml @@ -3,10 +3,10 @@ Speex plugin library ../../ext/speex/.libs/libgstspeex.so libgstspeex.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-taglib.xml b/docs/plugins/inspect/plugin-taglib.xml index 478a066ffc..ccf5bf98d8 100644 --- a/docs/plugins/inspect/plugin-taglib.xml +++ b/docs/plugins/inspect/plugin-taglib.xml @@ -3,10 +3,10 @@ Tag writing plug-in based on taglib ../../ext/taglib/.libs/libgsttaglib.so libgsttaglib.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-udp.xml b/docs/plugins/inspect/plugin-udp.xml index 734322c32a..eb4e29096f 100644 --- a/docs/plugins/inspect/plugin-udp.xml +++ b/docs/plugins/inspect/plugin-udp.xml @@ -3,10 +3,10 @@ transfer data via UDP ../../gst/udp/.libs/libgstudp.so libgstudp.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-video4linux2.xml b/docs/plugins/inspect/plugin-video4linux2.xml index 11cdf58804..2ab987c17c 100644 --- a/docs/plugins/inspect/plugin-video4linux2.xml +++ b/docs/plugins/inspect/plugin-video4linux2.xml @@ -3,10 +3,10 @@ elements for Video 4 Linux ../../sys/v4l2/.libs/libgstvideo4linux2.so libgstvideo4linux2.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-videobox.xml b/docs/plugins/inspect/plugin-videobox.xml index 9b870c1d61..45cb7aed0c 100644 --- a/docs/plugins/inspect/plugin-videobox.xml +++ b/docs/plugins/inspect/plugin-videobox.xml @@ -3,10 +3,10 @@ resizes a video by adding borders or cropping ../../gst/videobox/.libs/libgstvideobox.so libgstvideobox.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-videocrop.xml b/docs/plugins/inspect/plugin-videocrop.xml index 1fff2d7440..07753e93ba 100644 --- a/docs/plugins/inspect/plugin-videocrop.xml +++ b/docs/plugins/inspect/plugin-videocrop.xml @@ -3,10 +3,10 @@ Crops video into a user-defined region ../../gst/videocrop/.libs/libgstvideocrop.so libgstvideocrop.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-videofilter.xml b/docs/plugins/inspect/plugin-videofilter.xml index d098c80c23..9576ac8d30 100644 --- a/docs/plugins/inspect/plugin-videofilter.xml +++ b/docs/plugins/inspect/plugin-videofilter.xml @@ -3,10 +3,10 @@ Video filters plugin ../../gst/videofilter/.libs/libgstvideofilter.so libgstvideofilter.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-videomixer.xml b/docs/plugins/inspect/plugin-videomixer.xml index 34a97e5d8d..127d84e157 100644 --- a/docs/plugins/inspect/plugin-videomixer.xml +++ b/docs/plugins/inspect/plugin-videomixer.xml @@ -3,10 +3,10 @@ Video mixer ../../gst/videomixer/.libs/libgstvideomixer.so libgstvideomixer.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-wavenc.xml b/docs/plugins/inspect/plugin-wavenc.xml index 01011207b1..37cdd9fe25 100644 --- a/docs/plugins/inspect/plugin-wavenc.xml +++ b/docs/plugins/inspect/plugin-wavenc.xml @@ -3,10 +3,10 @@ Encode raw audio into WAV ../../gst/wavenc/.libs/libgstwavenc.so libgstwavenc.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-wavpack.xml b/docs/plugins/inspect/plugin-wavpack.xml index 4c2a092959..c08c8af74b 100644 --- a/docs/plugins/inspect/plugin-wavpack.xml +++ b/docs/plugins/inspect/plugin-wavpack.xml @@ -3,10 +3,10 @@ Wavpack lossless/lossy audio format handling ../../ext/wavpack/.libs/libgstwavpack.so libgstwavpack.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-wavparse.xml b/docs/plugins/inspect/plugin-wavparse.xml index c9e62769b1..3512a6ea5c 100644 --- a/docs/plugins/inspect/plugin-wavparse.xml +++ b/docs/plugins/inspect/plugin-wavparse.xml @@ -3,10 +3,10 @@ Parse a .wav file into raw audio ../../gst/wavparse/.libs/libgstwavparse.so libgstwavparse.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-ximagesrc.xml b/docs/plugins/inspect/plugin-ximagesrc.xml index 5490f007d3..2d672d8542 100644 --- a/docs/plugins/inspect/plugin-ximagesrc.xml +++ b/docs/plugins/inspect/plugin-ximagesrc.xml @@ -3,10 +3,10 @@ X11 video input plugin using standard Xlib calls ../../sys/ximage/.libs/libgstximagesrc.so libgstximagesrc.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-y4menc.xml b/docs/plugins/inspect/plugin-y4menc.xml index 08063283e6..4ea18da5db 100644 --- a/docs/plugins/inspect/plugin-y4menc.xml +++ b/docs/plugins/inspect/plugin-y4menc.xml @@ -3,10 +3,10 @@ Encodes a YUV frame into the yuv4mpeg format (mjpegtools) ../../gst/y4m/.libs/libgsty4menc.so libgsty4menc.so - 0.10.26.3 + 0.10.27 LGPL gst-plugins-good - GStreamer Good Plug-ins prerelease + GStreamer Good Plug-ins source release Unknown package origin diff --git a/gst-plugins-good.doap b/gst-plugins-good.doap index 5e1323d39b..5345f171f1 100644 --- a/gst-plugins-good.doap +++ b/gst-plugins-good.doap @@ -32,6 +32,17 @@ the plug-in code, LGPL or LGPL-compatible for the supporting library). + + + 0.10.27 + 0.10 + Some Kind of Temporal Blend + 2011-01-21 + + + + + 0.10.26 diff --git a/win32/common/config.h b/win32/common/config.h index d62372eb1e..387257fa49 100644 --- a/win32/common/config.h +++ b/win32/common/config.h @@ -43,13 +43,13 @@ #define GST_LICENSE "LGPL" /* package name in plugins */ -#define GST_PACKAGE_NAME "GStreamer Good Plug-ins prerelease" +#define GST_PACKAGE_NAME "GStreamer Good Plug-ins source release" /* package origin */ #define GST_PACKAGE_ORIGIN "Unknown package origin" /* GStreamer package release date/time for plugins as YYYY-MM-DD */ -#define GST_PACKAGE_RELEASE_DATETIME "2011-01-18T11:02Z" +#define GST_PACKAGE_RELEASE_DATETIME "2011-01-21" /* struct v4l2_buffer missing */ #undef GST_V4L2_MISSING_BUFDECL @@ -386,7 +386,7 @@ #define PACKAGE_NAME "GStreamer Good Plug-ins" /* Define to the full name and version of this package. */ -#define PACKAGE_STRING "GStreamer Good Plug-ins 0.10.26.4" +#define PACKAGE_STRING "GStreamer Good Plug-ins 0.10.27" /* Define to the one symbol short name of this package. */ #define PACKAGE_TARNAME "gst-plugins-good" @@ -395,7 +395,7 @@ #undef PACKAGE_URL /* Define to the version of this package. */ -#define PACKAGE_VERSION "0.10.26.4" +#define PACKAGE_VERSION "0.10.27" /* directory where plugins are located */ #ifdef _DEBUG @@ -426,7 +426,7 @@ #undef STDC_HEADERS /* Version number of package */ -#define VERSION "0.10.26.4" +#define VERSION "0.10.27" /* old wavpack API */ #undef WAVPACK_OLD_API From cb1019875ae523803afd2505c1de1039157d3ca7 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Mon, 24 Jan 2011 00:32:41 +0000 Subject: [PATCH 249/351] Back to development --- configure.ac | 2 +- .../gst-plugins-good-plugins.hierarchy | 6 ++++-- .../gst-plugins-good-plugins.interfaces | 1 + .../gst-plugins-good-plugins.prerequisites | 1 + docs/plugins/inspect/plugin-1394.xml | 4 ++-- docs/plugins/inspect/plugin-aasink.xml | 4 ++-- docs/plugins/inspect/plugin-alaw.xml | 4 ++-- docs/plugins/inspect/plugin-alpha.xml | 4 ++-- docs/plugins/inspect/plugin-alphacolor.xml | 4 ++-- docs/plugins/inspect/plugin-annodex.xml | 4 ++-- docs/plugins/inspect/plugin-apetag.xml | 4 ++-- docs/plugins/inspect/plugin-audiofx.xml | 4 ++-- docs/plugins/inspect/plugin-auparse.xml | 4 ++-- docs/plugins/inspect/plugin-autodetect.xml | 4 ++-- docs/plugins/inspect/plugin-avi.xml | 4 ++-- docs/plugins/inspect/plugin-cacasink.xml | 4 ++-- docs/plugins/inspect/plugin-cairo.xml | 4 ++-- docs/plugins/inspect/plugin-cutter.xml | 4 ++-- docs/plugins/inspect/plugin-debug.xml | 4 ++-- docs/plugins/inspect/plugin-deinterlace.xml | 4 ++-- docs/plugins/inspect/plugin-dv.xml | 4 ++-- docs/plugins/inspect/plugin-efence.xml | 4 ++-- docs/plugins/inspect/plugin-effectv.xml | 4 ++-- docs/plugins/inspect/plugin-equalizer.xml | 4 ++-- docs/plugins/inspect/plugin-esdsink.xml | 4 ++-- docs/plugins/inspect/plugin-flac.xml | 4 ++-- docs/plugins/inspect/plugin-flv.xml | 4 ++-- docs/plugins/inspect/plugin-flxdec.xml | 4 ++-- docs/plugins/inspect/plugin-gconfelements.xml | 4 ++-- docs/plugins/inspect/plugin-gdkpixbuf.xml | 4 ++-- docs/plugins/inspect/plugin-goom.xml | 4 ++-- docs/plugins/inspect/plugin-goom2k1.xml | 4 ++-- docs/plugins/inspect/plugin-gstrtpmanager.xml | 4 ++-- docs/plugins/inspect/plugin-halelements.xml | 4 ++-- docs/plugins/inspect/plugin-icydemux.xml | 4 ++-- docs/plugins/inspect/plugin-id3demux.xml | 4 ++-- docs/plugins/inspect/plugin-imagefreeze.xml | 4 ++-- docs/plugins/inspect/plugin-interleave.xml | 4 ++-- docs/plugins/inspect/plugin-jack.xml | 4 ++-- docs/plugins/inspect/plugin-jpeg.xml | 4 ++-- docs/plugins/inspect/plugin-level.xml | 4 ++-- docs/plugins/inspect/plugin-matroska.xml | 4 ++-- docs/plugins/inspect/plugin-monoscope.xml | 2 +- docs/plugins/inspect/plugin-mulaw.xml | 4 ++-- docs/plugins/inspect/plugin-multifile.xml | 4 ++-- docs/plugins/inspect/plugin-multipart.xml | 4 ++-- .../plugins/inspect/plugin-navigationtest.xml | 4 ++-- docs/plugins/inspect/plugin-oss4.xml | 4 ++-- docs/plugins/inspect/plugin-ossaudio.xml | 4 ++-- docs/plugins/inspect/plugin-png.xml | 4 ++-- docs/plugins/inspect/plugin-pulseaudio.xml | 4 ++-- docs/plugins/inspect/plugin-quicktime.xml | 4 ++-- docs/plugins/inspect/plugin-replaygain.xml | 4 ++-- docs/plugins/inspect/plugin-rtp.xml | 4 ++-- docs/plugins/inspect/plugin-rtsp.xml | 4 ++-- docs/plugins/inspect/plugin-shapewipe.xml | 4 ++-- docs/plugins/inspect/plugin-shout2send.xml | 2 +- docs/plugins/inspect/plugin-smpte.xml | 4 ++-- docs/plugins/inspect/plugin-soup.xml | 4 ++-- docs/plugins/inspect/plugin-spectrum.xml | 4 ++-- docs/plugins/inspect/plugin-speex.xml | 4 ++-- docs/plugins/inspect/plugin-taglib.xml | 4 ++-- docs/plugins/inspect/plugin-udp.xml | 4 ++-- docs/plugins/inspect/plugin-video4linux2.xml | 19 +++++++++++++++++-- docs/plugins/inspect/plugin-videobox.xml | 4 ++-- docs/plugins/inspect/plugin-videocrop.xml | 4 ++-- docs/plugins/inspect/plugin-videofilter.xml | 4 ++-- docs/plugins/inspect/plugin-videomixer.xml | 4 ++-- docs/plugins/inspect/plugin-wavenc.xml | 4 ++-- docs/plugins/inspect/plugin-wavpack.xml | 4 ++-- docs/plugins/inspect/plugin-wavparse.xml | 4 ++-- docs/plugins/inspect/plugin-ximagesrc.xml | 4 ++-- docs/plugins/inspect/plugin-y4menc.xml | 4 ++-- win32/common/config.h | 10 +++++----- 74 files changed, 163 insertions(+), 144 deletions(-) diff --git a/configure.ac b/configure.ac index c7757bbee1..5e93f5d8c9 100644 --- a/configure.ac +++ b/configure.ac @@ -5,7 +5,7 @@ dnl please read gstreamer/docs/random/autotools before changing this file dnl initialize autoconf dnl releases only do -Wall, git and prerelease does -Werror too dnl use a three digit version number for releases, and four for git/pre -AC_INIT(GStreamer Good Plug-ins, 0.10.27, +AC_INIT(GStreamer Good Plug-ins, 0.10.27.1, http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer, gst-plugins-good) diff --git a/docs/plugins/gst-plugins-good-plugins.hierarchy b/docs/plugins/gst-plugins-good-plugins.hierarchy index 7bb718ceef..5585c4a857 100644 --- a/docs/plugins/gst-plugins-good-plugins.hierarchy +++ b/docs/plugins/gst-plugins-good-plugins.hierarchy @@ -4,8 +4,6 @@ GObject GstCmmlTagHead GstCmmlTagStream GstColorBalanceChannel - GstMixerTrack - GstMixerOptions GstObject GstBus GstClock @@ -108,6 +106,7 @@ GObject GstTest GstVideoSink GstGdkPixbufSink + GstV4l2Sink GstBaseSrc GstPushSrc GstBaseAudioSrc @@ -224,6 +223,7 @@ GObject GstMatroskaDemux GstMatroskaMux GstWebMMux + GstMonoscope GstMuLawDec GstMuLawEnc GstMultipartDemux @@ -290,6 +290,7 @@ GInterface GstColorBalance GstImplementsInterface GstMixer + GstNavigation GstPreset GstPropertyProbe GstStreamVolume @@ -297,3 +298,4 @@ GInterface GstTuner GstURIHandler GstVideoOrientation + GstXOverlay diff --git a/docs/plugins/gst-plugins-good-plugins.interfaces b/docs/plugins/gst-plugins-good-plugins.interfaces index dc5c5915e6..7d98456987 100644 --- a/docs/plugins/gst-plugins-good-plugins.interfaces +++ b/docs/plugins/gst-plugins-good-plugins.interfaces @@ -19,6 +19,7 @@ GstRgVolume GstChildProxy GstAspectRatioCrop GstChildProxy GstPulseSink GstStreamVolume GstImplementsInterface GstPropertyProbe GstOss4Sink GstStreamVolume GstPropertyProbe +GstV4l2Sink GstImplementsInterface GstXOverlay GstNavigation GstColorBalance GstVideoOrientation GstPropertyProbe GstShout2send GstTagSetter GstUDPSink GstURIHandler GstDV1394Src GstURIHandler GstPropertyProbe diff --git a/docs/plugins/gst-plugins-good-plugins.prerequisites b/docs/plugins/gst-plugins-good-plugins.prerequisites index fa0a1d51c6..3f9042f552 100644 --- a/docs/plugins/gst-plugins-good-plugins.prerequisites +++ b/docs/plugins/gst-plugins-good-plugins.prerequisites @@ -6,4 +6,5 @@ GstMixer GstImplementsInterface GstElement GstTuner GstImplementsInterface GstElement GstColorBalance GstImplementsInterface GstElement GstVideoOrientation GstImplementsInterface GstElement +GstXOverlay GstImplementsInterface GstElement GIcon GObject diff --git a/docs/plugins/inspect/plugin-1394.xml b/docs/plugins/inspect/plugin-1394.xml index e70a8f365e..7cef67174a 100644 --- a/docs/plugins/inspect/plugin-1394.xml +++ b/docs/plugins/inspect/plugin-1394.xml @@ -3,10 +3,10 @@ Source for video data via IEEE1394 interface ../../ext/raw1394/.libs/libgst1394.so libgst1394.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-aasink.xml b/docs/plugins/inspect/plugin-aasink.xml index 7805c93198..1ed508bf95 100644 --- a/docs/plugins/inspect/plugin-aasink.xml +++ b/docs/plugins/inspect/plugin-aasink.xml @@ -3,10 +3,10 @@ ASCII Art video sink ../../ext/aalib/.libs/libgstaasink.so libgstaasink.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-alaw.xml b/docs/plugins/inspect/plugin-alaw.xml index 3f8d8d25d4..8e0ace5bec 100644 --- a/docs/plugins/inspect/plugin-alaw.xml +++ b/docs/plugins/inspect/plugin-alaw.xml @@ -3,10 +3,10 @@ ALaw audio conversion routines ../../gst/law/.libs/libgstalaw.so libgstalaw.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-alpha.xml b/docs/plugins/inspect/plugin-alpha.xml index 73972fca65..5e72d1580b 100644 --- a/docs/plugins/inspect/plugin-alpha.xml +++ b/docs/plugins/inspect/plugin-alpha.xml @@ -3,10 +3,10 @@ adds an alpha channel to video - constant or via chroma-keying ../../gst/alpha/.libs/libgstalpha.so libgstalpha.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-alphacolor.xml b/docs/plugins/inspect/plugin-alphacolor.xml index 24e394ee4b..5145144005 100644 --- a/docs/plugins/inspect/plugin-alphacolor.xml +++ b/docs/plugins/inspect/plugin-alphacolor.xml @@ -3,10 +3,10 @@ RGBA from/to AYUV colorspace conversion preserving the alpha channel ../../gst/alpha/.libs/libgstalphacolor.so libgstalphacolor.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-annodex.xml b/docs/plugins/inspect/plugin-annodex.xml index 5e709d2a54..757c17b3a3 100644 --- a/docs/plugins/inspect/plugin-annodex.xml +++ b/docs/plugins/inspect/plugin-annodex.xml @@ -3,10 +3,10 @@ annodex stream manipulation (info about annodex: http://www.annodex.net) ../../ext/annodex/.libs/libgstannodex.so libgstannodex.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-apetag.xml b/docs/plugins/inspect/plugin-apetag.xml index 67707f6b08..1dc3d8e1b5 100644 --- a/docs/plugins/inspect/plugin-apetag.xml +++ b/docs/plugins/inspect/plugin-apetag.xml @@ -3,10 +3,10 @@ APEv1/2 tag reader ../../gst/apetag/.libs/libgstapetag.so libgstapetag.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-audiofx.xml b/docs/plugins/inspect/plugin-audiofx.xml index 3e5e84a310..2441f9387f 100644 --- a/docs/plugins/inspect/plugin-audiofx.xml +++ b/docs/plugins/inspect/plugin-audiofx.xml @@ -3,10 +3,10 @@ Audio effects plugin ../../gst/audiofx/.libs/libgstaudiofx.so libgstaudiofx.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-auparse.xml b/docs/plugins/inspect/plugin-auparse.xml index b6ed29c959..5981acefb5 100644 --- a/docs/plugins/inspect/plugin-auparse.xml +++ b/docs/plugins/inspect/plugin-auparse.xml @@ -3,10 +3,10 @@ parses au streams ../../gst/auparse/.libs/libgstauparse.so libgstauparse.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-autodetect.xml b/docs/plugins/inspect/plugin-autodetect.xml index 15be26f8dd..338849f3cb 100644 --- a/docs/plugins/inspect/plugin-autodetect.xml +++ b/docs/plugins/inspect/plugin-autodetect.xml @@ -3,10 +3,10 @@ Plugin contains auto-detection plugins for video/audio in- and outputs ../../gst/autodetect/.libs/libgstautodetect.so libgstautodetect.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-avi.xml b/docs/plugins/inspect/plugin-avi.xml index e4a30b8e16..8407f16c7d 100644 --- a/docs/plugins/inspect/plugin-avi.xml +++ b/docs/plugins/inspect/plugin-avi.xml @@ -3,10 +3,10 @@ AVI stream handling ../../gst/avi/.libs/libgstavi.so libgstavi.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-cacasink.xml b/docs/plugins/inspect/plugin-cacasink.xml index 25d58662db..7fd0297df0 100644 --- a/docs/plugins/inspect/plugin-cacasink.xml +++ b/docs/plugins/inspect/plugin-cacasink.xml @@ -3,10 +3,10 @@ Colored ASCII Art video sink ../../ext/libcaca/.libs/libgstcacasink.so libgstcacasink.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-cairo.xml b/docs/plugins/inspect/plugin-cairo.xml index 26483ab38c..e240b369b2 100644 --- a/docs/plugins/inspect/plugin-cairo.xml +++ b/docs/plugins/inspect/plugin-cairo.xml @@ -3,10 +3,10 @@ Cairo-based elements ../../ext/cairo/.libs/libgstcairo.so libgstcairo.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-cutter.xml b/docs/plugins/inspect/plugin-cutter.xml index f5edf8fb2f..cf7d7202cd 100644 --- a/docs/plugins/inspect/plugin-cutter.xml +++ b/docs/plugins/inspect/plugin-cutter.xml @@ -3,10 +3,10 @@ Audio Cutter to split audio into non-silent bits ../../gst/cutter/.libs/libgstcutter.so libgstcutter.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-debug.xml b/docs/plugins/inspect/plugin-debug.xml index 5d307928d9..a3bac01696 100644 --- a/docs/plugins/inspect/plugin-debug.xml +++ b/docs/plugins/inspect/plugin-debug.xml @@ -3,10 +3,10 @@ elements for testing and debugging ../../gst/debugutils/.libs/libgstdebug.so libgstdebug.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-deinterlace.xml b/docs/plugins/inspect/plugin-deinterlace.xml index 455df7df26..35a5156783 100644 --- a/docs/plugins/inspect/plugin-deinterlace.xml +++ b/docs/plugins/inspect/plugin-deinterlace.xml @@ -3,10 +3,10 @@ Deinterlacer ../../gst/deinterlace/.libs/libgstdeinterlace.so libgstdeinterlace.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-dv.xml b/docs/plugins/inspect/plugin-dv.xml index 25d098a85c..772493b8b3 100644 --- a/docs/plugins/inspect/plugin-dv.xml +++ b/docs/plugins/inspect/plugin-dv.xml @@ -3,10 +3,10 @@ DV demuxer and decoder based on libdv (libdv.sf.net) ../../ext/dv/.libs/libgstdv.so libgstdv.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-efence.xml b/docs/plugins/inspect/plugin-efence.xml index 9ab33e6568..a0d91632d7 100644 --- a/docs/plugins/inspect/plugin-efence.xml +++ b/docs/plugins/inspect/plugin-efence.xml @@ -3,10 +3,10 @@ This element converts a stream of normal GStreamer buffers into a stream of buffers that are allocated in such a way that out-of-bounds access to data in the buffer is more likely to cause segmentation faults. This allocation method is very similar to the debugging tool "Electric Fence". ../../gst/debugutils/.libs/libgstefence.so libgstefence.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-effectv.xml b/docs/plugins/inspect/plugin-effectv.xml index 99c3e59f72..5011b2ef06 100644 --- a/docs/plugins/inspect/plugin-effectv.xml +++ b/docs/plugins/inspect/plugin-effectv.xml @@ -3,10 +3,10 @@ effect plugins from the effectv project ../../gst/effectv/.libs/libgsteffectv.so libgsteffectv.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-equalizer.xml b/docs/plugins/inspect/plugin-equalizer.xml index 334f21bd0f..17ff647440 100644 --- a/docs/plugins/inspect/plugin-equalizer.xml +++ b/docs/plugins/inspect/plugin-equalizer.xml @@ -3,10 +3,10 @@ GStreamer audio equalizers ../../gst/equalizer/.libs/libgstequalizer.so libgstequalizer.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-esdsink.xml b/docs/plugins/inspect/plugin-esdsink.xml index feb9021f06..b1ad8174f5 100644 --- a/docs/plugins/inspect/plugin-esdsink.xml +++ b/docs/plugins/inspect/plugin-esdsink.xml @@ -3,10 +3,10 @@ ESD Element Plugins ../../ext/esd/.libs/libgstesd.so libgstesd.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-flac.xml b/docs/plugins/inspect/plugin-flac.xml index a4133713a8..e7c7860a38 100644 --- a/docs/plugins/inspect/plugin-flac.xml +++ b/docs/plugins/inspect/plugin-flac.xml @@ -3,10 +3,10 @@ The FLAC Lossless compressor Codec ../../ext/flac/.libs/libgstflac.so libgstflac.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-flv.xml b/docs/plugins/inspect/plugin-flv.xml index e1afab1a18..a251587ae9 100644 --- a/docs/plugins/inspect/plugin-flv.xml +++ b/docs/plugins/inspect/plugin-flv.xml @@ -3,10 +3,10 @@ FLV muxing and demuxing plugin ../../gst/flv/.libs/libgstflv.so libgstflv.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-flxdec.xml b/docs/plugins/inspect/plugin-flxdec.xml index 2ec08d559f..d2402ed306 100644 --- a/docs/plugins/inspect/plugin-flxdec.xml +++ b/docs/plugins/inspect/plugin-flxdec.xml @@ -3,10 +3,10 @@ FLC/FLI/FLX video decoder ../../gst/flx/.libs/libgstflxdec.so libgstflxdec.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-gconfelements.xml b/docs/plugins/inspect/plugin-gconfelements.xml index 1e477bc9a9..49d897b667 100644 --- a/docs/plugins/inspect/plugin-gconfelements.xml +++ b/docs/plugins/inspect/plugin-gconfelements.xml @@ -3,10 +3,10 @@ elements wrapping the GStreamer/GConf audio/video output settings ../../ext/gconf/.libs/libgstgconfelements.so libgstgconfelements.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-gdkpixbuf.xml b/docs/plugins/inspect/plugin-gdkpixbuf.xml index aabb20f220..75e6154910 100644 --- a/docs/plugins/inspect/plugin-gdkpixbuf.xml +++ b/docs/plugins/inspect/plugin-gdkpixbuf.xml @@ -3,10 +3,10 @@ GdkPixbuf-based image decoder, scaler and sink ../../ext/gdk_pixbuf/.libs/libgstgdkpixbuf.so libgstgdkpixbuf.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-goom.xml b/docs/plugins/inspect/plugin-goom.xml index 8bf4060781..a45d8552ff 100644 --- a/docs/plugins/inspect/plugin-goom.xml +++ b/docs/plugins/inspect/plugin-goom.xml @@ -3,10 +3,10 @@ GOOM visualization filter ../../gst/goom/.libs/libgstgoom.so libgstgoom.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-goom2k1.xml b/docs/plugins/inspect/plugin-goom2k1.xml index d70137e95e..f443b7bb8f 100644 --- a/docs/plugins/inspect/plugin-goom2k1.xml +++ b/docs/plugins/inspect/plugin-goom2k1.xml @@ -3,10 +3,10 @@ GOOM 2k1 visualization filter ../../gst/goom2k1/.libs/libgstgoom2k1.so libgstgoom2k1.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-gstrtpmanager.xml b/docs/plugins/inspect/plugin-gstrtpmanager.xml index aa2a7538ee..bc8012f16c 100644 --- a/docs/plugins/inspect/plugin-gstrtpmanager.xml +++ b/docs/plugins/inspect/plugin-gstrtpmanager.xml @@ -3,10 +3,10 @@ RTP session management plugin library ../../gst/rtpmanager/.libs/libgstrtpmanager.so libgstrtpmanager.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-halelements.xml b/docs/plugins/inspect/plugin-halelements.xml index 224abf3ad5..68d00d5749 100644 --- a/docs/plugins/inspect/plugin-halelements.xml +++ b/docs/plugins/inspect/plugin-halelements.xml @@ -3,10 +3,10 @@ elements wrapping the GStreamer/HAL audio input/output devices ../../ext/hal/.libs/libgsthalelements.so libgsthalelements.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-icydemux.xml b/docs/plugins/inspect/plugin-icydemux.xml index d30ca100bc..5189eff17d 100644 --- a/docs/plugins/inspect/plugin-icydemux.xml +++ b/docs/plugins/inspect/plugin-icydemux.xml @@ -3,10 +3,10 @@ Demux ICY tags from a stream ../../gst/icydemux/.libs/libgsticydemux.so libgsticydemux.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-id3demux.xml b/docs/plugins/inspect/plugin-id3demux.xml index 937ae2e982..c3a3c79d18 100644 --- a/docs/plugins/inspect/plugin-id3demux.xml +++ b/docs/plugins/inspect/plugin-id3demux.xml @@ -3,10 +3,10 @@ Demux ID3v1 and ID3v2 tags from a file ../../gst/id3demux/.libs/libgstid3demux.so libgstid3demux.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-imagefreeze.xml b/docs/plugins/inspect/plugin-imagefreeze.xml index 54f5b85893..b2ba988d37 100644 --- a/docs/plugins/inspect/plugin-imagefreeze.xml +++ b/docs/plugins/inspect/plugin-imagefreeze.xml @@ -3,10 +3,10 @@ Still frame stream generator ../../gst/imagefreeze/.libs/libgstimagefreeze.so libgstimagefreeze.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-interleave.xml b/docs/plugins/inspect/plugin-interleave.xml index 4b3c60af0c..9d9f136852 100644 --- a/docs/plugins/inspect/plugin-interleave.xml +++ b/docs/plugins/inspect/plugin-interleave.xml @@ -3,10 +3,10 @@ Audio interleaver/deinterleaver ../../gst/interleave/.libs/libgstinterleave.so libgstinterleave.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-jack.xml b/docs/plugins/inspect/plugin-jack.xml index 0e92a21760..4d63a04c15 100644 --- a/docs/plugins/inspect/plugin-jack.xml +++ b/docs/plugins/inspect/plugin-jack.xml @@ -3,10 +3,10 @@ JACK audio elements ../../ext/jack/.libs/libgstjack.so libgstjack.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-jpeg.xml b/docs/plugins/inspect/plugin-jpeg.xml index 9ae11c5394..4f897e60d1 100644 --- a/docs/plugins/inspect/plugin-jpeg.xml +++ b/docs/plugins/inspect/plugin-jpeg.xml @@ -3,10 +3,10 @@ JPeg plugin library ../../ext/jpeg/.libs/libgstjpeg.so libgstjpeg.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-level.xml b/docs/plugins/inspect/plugin-level.xml index 198cb7fb88..b7e9bb87bf 100644 --- a/docs/plugins/inspect/plugin-level.xml +++ b/docs/plugins/inspect/plugin-level.xml @@ -3,10 +3,10 @@ Audio level plugin ../../gst/level/.libs/libgstlevel.so libgstlevel.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-matroska.xml b/docs/plugins/inspect/plugin-matroska.xml index 9604442a95..ac63701a3f 100644 --- a/docs/plugins/inspect/plugin-matroska.xml +++ b/docs/plugins/inspect/plugin-matroska.xml @@ -3,10 +3,10 @@ Matroska and WebM stream handling ../../gst/matroska/.libs/libgstmatroska.so libgstmatroska.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-monoscope.xml b/docs/plugins/inspect/plugin-monoscope.xml index 05d39c6be3..0771febb2b 100644 --- a/docs/plugins/inspect/plugin-monoscope.xml +++ b/docs/plugins/inspect/plugin-monoscope.xml @@ -3,7 +3,7 @@ Monoscope visualization ../../gst/monoscope/.libs/libgstmonoscope.so libgstmonoscope.so - 0.10.26.1 + 0.10.27.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-mulaw.xml b/docs/plugins/inspect/plugin-mulaw.xml index 49429116e5..2b9320a635 100644 --- a/docs/plugins/inspect/plugin-mulaw.xml +++ b/docs/plugins/inspect/plugin-mulaw.xml @@ -3,10 +3,10 @@ MuLaw audio conversion routines ../../gst/law/.libs/libgstmulaw.so libgstmulaw.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-multifile.xml b/docs/plugins/inspect/plugin-multifile.xml index bb2d80adb9..434b1c0944 100644 --- a/docs/plugins/inspect/plugin-multifile.xml +++ b/docs/plugins/inspect/plugin-multifile.xml @@ -3,10 +3,10 @@ Reads/Writes buffers from/to sequentially named files ../../gst/multifile/.libs/libgstmultifile.so libgstmultifile.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-multipart.xml b/docs/plugins/inspect/plugin-multipart.xml index e5ef7b422e..7ee0e82268 100644 --- a/docs/plugins/inspect/plugin-multipart.xml +++ b/docs/plugins/inspect/plugin-multipart.xml @@ -3,10 +3,10 @@ multipart stream manipulation ../../gst/multipart/.libs/libgstmultipart.so libgstmultipart.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-navigationtest.xml b/docs/plugins/inspect/plugin-navigationtest.xml index 0dc2377fe5..3ab99095db 100644 --- a/docs/plugins/inspect/plugin-navigationtest.xml +++ b/docs/plugins/inspect/plugin-navigationtest.xml @@ -3,10 +3,10 @@ Template for a video filter ../../gst/debugutils/.libs/libgstnavigationtest.so libgstnavigationtest.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-oss4.xml b/docs/plugins/inspect/plugin-oss4.xml index 0a59daa9f7..1a53012ea2 100644 --- a/docs/plugins/inspect/plugin-oss4.xml +++ b/docs/plugins/inspect/plugin-oss4.xml @@ -3,10 +3,10 @@ Open Sound System (OSS) version 4 support for GStreamer ../../sys/oss4/.libs/libgstoss4audio.so libgstoss4audio.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-ossaudio.xml b/docs/plugins/inspect/plugin-ossaudio.xml index ddc69591c3..37ba51bad6 100644 --- a/docs/plugins/inspect/plugin-ossaudio.xml +++ b/docs/plugins/inspect/plugin-ossaudio.xml @@ -3,10 +3,10 @@ OSS (Open Sound System) support for GStreamer ../../sys/oss/.libs/libgstossaudio.so libgstossaudio.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-png.xml b/docs/plugins/inspect/plugin-png.xml index 409b496a99..698e4ba215 100644 --- a/docs/plugins/inspect/plugin-png.xml +++ b/docs/plugins/inspect/plugin-png.xml @@ -3,10 +3,10 @@ PNG plugin library ../../ext/libpng/.libs/libgstpng.so libgstpng.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-pulseaudio.xml b/docs/plugins/inspect/plugin-pulseaudio.xml index f3e1647405..d907f1068e 100644 --- a/docs/plugins/inspect/plugin-pulseaudio.xml +++ b/docs/plugins/inspect/plugin-pulseaudio.xml @@ -3,10 +3,10 @@ PulseAudio plugin library ../../ext/pulse/.libs/libgstpulse.so libgstpulse.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-quicktime.xml b/docs/plugins/inspect/plugin-quicktime.xml index 44edb00923..d71f20e15f 100644 --- a/docs/plugins/inspect/plugin-quicktime.xml +++ b/docs/plugins/inspect/plugin-quicktime.xml @@ -3,10 +3,10 @@ Quicktime support ../../gst/qtdemux/.libs/libgstqtdemux.so libgstqtdemux.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-replaygain.xml b/docs/plugins/inspect/plugin-replaygain.xml index b8d2e3f65b..5cf289d496 100644 --- a/docs/plugins/inspect/plugin-replaygain.xml +++ b/docs/plugins/inspect/plugin-replaygain.xml @@ -3,10 +3,10 @@ ReplayGain volume normalization ../../gst/replaygain/.libs/libgstreplaygain.so libgstreplaygain.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-rtp.xml b/docs/plugins/inspect/plugin-rtp.xml index 4023abc33e..dd4c0051bf 100644 --- a/docs/plugins/inspect/plugin-rtp.xml +++ b/docs/plugins/inspect/plugin-rtp.xml @@ -3,10 +3,10 @@ Real-time protocol plugins ../../gst/rtp/.libs/libgstrtp.so libgstrtp.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-rtsp.xml b/docs/plugins/inspect/plugin-rtsp.xml index f5b2259591..c643cdd17a 100644 --- a/docs/plugins/inspect/plugin-rtsp.xml +++ b/docs/plugins/inspect/plugin-rtsp.xml @@ -3,10 +3,10 @@ transfer data via RTSP ../../gst/rtsp/.libs/libgstrtsp.so libgstrtsp.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-shapewipe.xml b/docs/plugins/inspect/plugin-shapewipe.xml index 2e06ea3b2e..fdf823145e 100644 --- a/docs/plugins/inspect/plugin-shapewipe.xml +++ b/docs/plugins/inspect/plugin-shapewipe.xml @@ -3,10 +3,10 @@ Shape Wipe transition filter ../../gst/shapewipe/.libs/libgstshapewipe.so libgstshapewipe.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-shout2send.xml b/docs/plugins/inspect/plugin-shout2send.xml index 25ff258009..3927945275 100644 --- a/docs/plugins/inspect/plugin-shout2send.xml +++ b/docs/plugins/inspect/plugin-shout2send.xml @@ -3,7 +3,7 @@ Sends data to an icecast server using libshout2 ../../ext/shout2/.libs/libgstshout2.so libgstshout2.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good libshout2 diff --git a/docs/plugins/inspect/plugin-smpte.xml b/docs/plugins/inspect/plugin-smpte.xml index c84436523d..0d24a5afef 100644 --- a/docs/plugins/inspect/plugin-smpte.xml +++ b/docs/plugins/inspect/plugin-smpte.xml @@ -3,10 +3,10 @@ Apply the standard SMPTE transitions on video images ../../gst/smpte/.libs/libgstsmpte.so libgstsmpte.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-soup.xml b/docs/plugins/inspect/plugin-soup.xml index 92cb65e7a3..0cbc37192a 100644 --- a/docs/plugins/inspect/plugin-soup.xml +++ b/docs/plugins/inspect/plugin-soup.xml @@ -3,10 +3,10 @@ libsoup HTTP client src ../../ext/soup/.libs/libgstsouphttpsrc.so libgstsouphttpsrc.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-spectrum.xml b/docs/plugins/inspect/plugin-spectrum.xml index 6cb6a0145b..1df70e15f6 100644 --- a/docs/plugins/inspect/plugin-spectrum.xml +++ b/docs/plugins/inspect/plugin-spectrum.xml @@ -3,10 +3,10 @@ Run an FFT on the audio signal, output spectrum data ../../gst/spectrum/.libs/libgstspectrum.so libgstspectrum.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-speex.xml b/docs/plugins/inspect/plugin-speex.xml index b4bfb0ea92..959df3cad0 100644 --- a/docs/plugins/inspect/plugin-speex.xml +++ b/docs/plugins/inspect/plugin-speex.xml @@ -3,10 +3,10 @@ Speex plugin library ../../ext/speex/.libs/libgstspeex.so libgstspeex.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-taglib.xml b/docs/plugins/inspect/plugin-taglib.xml index ccf5bf98d8..271141b0b3 100644 --- a/docs/plugins/inspect/plugin-taglib.xml +++ b/docs/plugins/inspect/plugin-taglib.xml @@ -3,10 +3,10 @@ Tag writing plug-in based on taglib ../../ext/taglib/.libs/libgsttaglib.so libgsttaglib.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-udp.xml b/docs/plugins/inspect/plugin-udp.xml index eb4e29096f..8870867133 100644 --- a/docs/plugins/inspect/plugin-udp.xml +++ b/docs/plugins/inspect/plugin-udp.xml @@ -3,10 +3,10 @@ transfer data via UDP ../../gst/udp/.libs/libgstudp.so libgstudp.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-video4linux2.xml b/docs/plugins/inspect/plugin-video4linux2.xml index 2ab987c17c..6d5fd2fddd 100644 --- a/docs/plugins/inspect/plugin-video4linux2.xml +++ b/docs/plugins/inspect/plugin-video4linux2.xml @@ -3,12 +3,27 @@ elements for Video 4 Linux ../../sys/v4l2/.libs/libgstvideo4linux2.so libgstvideo4linux2.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin + + v4l2sink + Video (video4linux2) Sink + Sink/Video + Displays frames on a video4linux2 device + Rob Clark <rob@ti.com>, + + + sink + sink + always +
video/x-raw-rgb, bpp=(int)8, depth=(int)8, red_mask=(int)224, green_mask=(int)28, blue_mask=(int)3, endianness=(int)1234, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-rgb, bpp=(int)16, depth=(int)15, red_mask=(int)31744, green_mask=(int)992, blue_mask=(int)31, endianness=(int)1234, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-rgb, bpp=(int)16, depth=(int)16, red_mask=(int)63488, green_mask=(int)2016, blue_mask=(int)31, endianness=(int)1234, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-rgb, bpp=(int)16, depth=(int)15, red_mask=(int)31744, green_mask=(int)992, blue_mask=(int)31, endianness=(int)4321, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-rgb, bpp=(int)16, depth=(int)16, red_mask=(int)63488, green_mask=(int)2016, blue_mask=(int)31, endianness=(int)4321, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-rgb, bpp=(int)24, depth=(int)24, red_mask=(int)255, green_mask=(int)65280, blue_mask=(int)16711680, endianness=(int)4321, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-rgb, bpp=(int)24, depth=(int)24, red_mask=(int)16711680, green_mask=(int)65280, blue_mask=(int)255, endianness=(int)4321, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-rgb, bpp=(int)32, depth=(int)32, red_mask=(int)255, green_mask=(int)65280, blue_mask=(int)16711680, endianness=(int)4321, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-rgb, bpp=(int)32, depth=(int)32, red_mask=(int)-16777216, green_mask=(int)16711680, blue_mask=(int)65280, endianness=(int)4321, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-gray, bpp=(int)8, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)YVU9, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)YV12, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)YUY2, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)UYVY, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)Y42B, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)Y41B, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)Y41P, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)NV12, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)NV21, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)YUV9, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)I420, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-bayer, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; image/jpeg, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; image/jpeg, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-dv, systemstream=(boolean)true, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/mpegts; video/x-sonix, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-pwc1, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-pwc2, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)YVYU, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]
+ + + v4l2src Video (video4linux2) Source diff --git a/docs/plugins/inspect/plugin-videobox.xml b/docs/plugins/inspect/plugin-videobox.xml index 45cb7aed0c..121ff06ed2 100644 --- a/docs/plugins/inspect/plugin-videobox.xml +++ b/docs/plugins/inspect/plugin-videobox.xml @@ -3,10 +3,10 @@ resizes a video by adding borders or cropping ../../gst/videobox/.libs/libgstvideobox.so libgstvideobox.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-videocrop.xml b/docs/plugins/inspect/plugin-videocrop.xml index 07753e93ba..4460f5c1f4 100644 --- a/docs/plugins/inspect/plugin-videocrop.xml +++ b/docs/plugins/inspect/plugin-videocrop.xml @@ -3,10 +3,10 @@ Crops video into a user-defined region ../../gst/videocrop/.libs/libgstvideocrop.so libgstvideocrop.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-videofilter.xml b/docs/plugins/inspect/plugin-videofilter.xml index 9576ac8d30..5325b1d56d 100644 --- a/docs/plugins/inspect/plugin-videofilter.xml +++ b/docs/plugins/inspect/plugin-videofilter.xml @@ -3,10 +3,10 @@ Video filters plugin ../../gst/videofilter/.libs/libgstvideofilter.so libgstvideofilter.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-videomixer.xml b/docs/plugins/inspect/plugin-videomixer.xml index 127d84e157..49b1bfaa67 100644 --- a/docs/plugins/inspect/plugin-videomixer.xml +++ b/docs/plugins/inspect/plugin-videomixer.xml @@ -3,10 +3,10 @@ Video mixer ../../gst/videomixer/.libs/libgstvideomixer.so libgstvideomixer.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-wavenc.xml b/docs/plugins/inspect/plugin-wavenc.xml index 37cdd9fe25..f021a046ae 100644 --- a/docs/plugins/inspect/plugin-wavenc.xml +++ b/docs/plugins/inspect/plugin-wavenc.xml @@ -3,10 +3,10 @@ Encode raw audio into WAV ../../gst/wavenc/.libs/libgstwavenc.so libgstwavenc.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-wavpack.xml b/docs/plugins/inspect/plugin-wavpack.xml index c08c8af74b..89f3d67343 100644 --- a/docs/plugins/inspect/plugin-wavpack.xml +++ b/docs/plugins/inspect/plugin-wavpack.xml @@ -3,10 +3,10 @@ Wavpack lossless/lossy audio format handling ../../ext/wavpack/.libs/libgstwavpack.so libgstwavpack.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-wavparse.xml b/docs/plugins/inspect/plugin-wavparse.xml index 3512a6ea5c..c47fab1f7d 100644 --- a/docs/plugins/inspect/plugin-wavparse.xml +++ b/docs/plugins/inspect/plugin-wavparse.xml @@ -3,10 +3,10 @@ Parse a .wav file into raw audio ../../gst/wavparse/.libs/libgstwavparse.so libgstwavparse.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-ximagesrc.xml b/docs/plugins/inspect/plugin-ximagesrc.xml index 2d672d8542..9e575bd9e5 100644 --- a/docs/plugins/inspect/plugin-ximagesrc.xml +++ b/docs/plugins/inspect/plugin-ximagesrc.xml @@ -3,10 +3,10 @@ X11 video input plugin using standard Xlib calls ../../sys/ximage/.libs/libgstximagesrc.so libgstximagesrc.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/docs/plugins/inspect/plugin-y4menc.xml b/docs/plugins/inspect/plugin-y4menc.xml index 4ea18da5db..b988cfd944 100644 --- a/docs/plugins/inspect/plugin-y4menc.xml +++ b/docs/plugins/inspect/plugin-y4menc.xml @@ -3,10 +3,10 @@ Encodes a YUV frame into the yuv4mpeg format (mjpegtools) ../../gst/y4m/.libs/libgsty4menc.so libgsty4menc.so - 0.10.27 + 0.10.27.1 LGPL gst-plugins-good - GStreamer Good Plug-ins source release + GStreamer Good Plug-ins git Unknown package origin diff --git a/win32/common/config.h b/win32/common/config.h index 387257fa49..d2f750caec 100644 --- a/win32/common/config.h +++ b/win32/common/config.h @@ -43,13 +43,13 @@ #define GST_LICENSE "LGPL" /* package name in plugins */ -#define GST_PACKAGE_NAME "GStreamer Good Plug-ins source release" +#define GST_PACKAGE_NAME "GStreamer Good Plug-ins git" /* package origin */ #define GST_PACKAGE_ORIGIN "Unknown package origin" /* GStreamer package release date/time for plugins as YYYY-MM-DD */ -#define GST_PACKAGE_RELEASE_DATETIME "2011-01-21" +#define GST_PACKAGE_RELEASE_DATETIME "2011-01-24T00:02Z" /* struct v4l2_buffer missing */ #undef GST_V4L2_MISSING_BUFDECL @@ -386,7 +386,7 @@ #define PACKAGE_NAME "GStreamer Good Plug-ins" /* Define to the full name and version of this package. */ -#define PACKAGE_STRING "GStreamer Good Plug-ins 0.10.27" +#define PACKAGE_STRING "GStreamer Good Plug-ins 0.10.27.1" /* Define to the one symbol short name of this package. */ #define PACKAGE_TARNAME "gst-plugins-good" @@ -395,7 +395,7 @@ #undef PACKAGE_URL /* Define to the version of this package. */ -#define PACKAGE_VERSION "0.10.27" +#define PACKAGE_VERSION "0.10.27.1" /* directory where plugins are located */ #ifdef _DEBUG @@ -426,7 +426,7 @@ #undef STDC_HEADERS /* Version number of package */ -#define VERSION "0.10.27" +#define VERSION "0.10.27.1" /* old wavpack API */ #undef WAVPACK_OLD_API From 285235a10a18f72b239385829e4963a221c7a897 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Mon, 24 Jan 2011 13:39:58 +0000 Subject: [PATCH 250/351] examples: autoaudisink -> autoaudiosink in RTP examples --- tests/examples/rtp/client-H263p-AMR.sh | 2 +- tests/examples/rtp/client-H263p-PCMA.sh | 2 +- tests/examples/rtp/client-H264-PCMA.sh | 2 +- tests/examples/rtp/client-PCMA.sh | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/tests/examples/rtp/client-H263p-AMR.sh b/tests/examples/rtp/client-H263p-AMR.sh index 07a466fa81..bb0b795fee 100755 --- a/tests/examples/rtp/client-H263p-AMR.sh +++ b/tests/examples/rtp/client-H263p-AMR.sh @@ -10,7 +10,7 @@ VIDEO_DEC="rtph263pdepay ! ffdec_h263" AUDIO_DEC="rtpamrdepay ! amrnbdec" VIDEO_SINK="ffmpegcolorspace ! autovideosink" -AUDIO_SINK="audioconvert ! audioresample ! autoaudisink" +AUDIO_SINK="audioconvert ! audioresample ! autoaudiosink" gst-launch -v gstrtpbin name=rtpbin latency=100 \ udpsrc caps=$VIDEO_CAPS port=5000 ! rtpbin.recv_rtp_sink_0 \ diff --git a/tests/examples/rtp/client-H263p-PCMA.sh b/tests/examples/rtp/client-H263p-PCMA.sh index 8204c1f11d..9db0d7ecf0 100755 --- a/tests/examples/rtp/client-H263p-PCMA.sh +++ b/tests/examples/rtp/client-H263p-PCMA.sh @@ -13,7 +13,7 @@ VIDEO_DEC="rtph263pdepay ! ffdec_h263" AUDIO_DEC="rtppcmadepay ! alawdec" VIDEO_SINK="ffmpegcolorspace ! autovideosink" -AUDIO_SINK="audioconvert ! audioresample ! autoaudisink" +AUDIO_SINK="audioconvert ! audioresample ! autoaudiosink" LATENCY=100 diff --git a/tests/examples/rtp/client-H264-PCMA.sh b/tests/examples/rtp/client-H264-PCMA.sh index 600cefe935..9f007f1f9a 100755 --- a/tests/examples/rtp/client-H264-PCMA.sh +++ b/tests/examples/rtp/client-H264-PCMA.sh @@ -51,7 +51,7 @@ VIDEO_DEC="rtph264depay ! ffdec_h264" AUDIO_DEC="rtppcmadepay ! alawdec" VIDEO_SINK="ffmpegcolorspace ! autovideosink" -AUDIO_SINK="audioconvert ! audioresample ! autoaudisink" +AUDIO_SINK="audioconvert ! audioresample ! autoaudiosink" gst-launch -v gstrtpbin name=rtpbin latency=$LATENCY \ udpsrc caps=$VIDEO_CAPS port=5000 ! rtpbin.recv_rtp_sink_0 \ diff --git a/tests/examples/rtp/client-PCMA.sh b/tests/examples/rtp/client-PCMA.sh index 5d473404bd..b2c92f7d9e 100755 --- a/tests/examples/rtp/client-PCMA.sh +++ b/tests/examples/rtp/client-PCMA.sh @@ -25,7 +25,7 @@ AUDIO_CAPS="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding- AUDIO_DEC="rtppcmadepay ! alawdec" -AUDIO_SINK="audioconvert ! audioresample ! autoaudisink" +AUDIO_SINK="audioconvert ! audioresample ! autoaudiosink" # the destination machine to send RTCP to. This is the address of the sender and # is used to send back the RTCP reports of this receiver. If the data is sent From 06364a54ceea96cef0e814539a88da656c10bd93 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Mon, 24 Jan 2011 15:11:02 +0000 Subject: [PATCH 251/351] qtdemux: fix printf format warning on mingw32 Make win32 build bot happy again, and nicefy output while we're at it. qtdemux.c: In function 'qtdemux_parse_trun': qtdemux.c:2162:3: error: format '%lu' expects type 'long unsigned int', but argument 9 has type 'guint32' --- gst/qtdemux/qtdemux.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/gst/qtdemux/qtdemux.c b/gst/qtdemux/qtdemux.c index b4f9cbf164..dbb822863d 100644 --- a/gst/qtdemux/qtdemux.c +++ b/gst/qtdemux/qtdemux.c @@ -2159,8 +2159,9 @@ qtdemux_parse_trun (GstQTDemux * qtdemux, GstByteReader * trun, QTDEMUX_MAX_SAMPLE_INDEX_SIZE / sizeof (QtDemuxSample)) goto index_too_big; - GST_DEBUG_OBJECT (qtdemux, "allocating n_samples %u (%lu MB)", - stream->n_samples, (stream->n_samples * sizeof (QtDemuxSample)) >> 20); + GST_DEBUG_OBJECT (qtdemux, "allocating n_samples %u * %u (%.2f MB)", + stream->n_samples, (guint) sizeof (QtDemuxSample), + stream->n_samples * sizeof (QtDemuxSample) / (1024.0 * 1024.0)); /* create a new array of samples if it's the first sample parsed */ if (stream->n_samples == 0) @@ -5201,7 +5202,7 @@ qtdemux_stbl_init (GstQTDemux * qtdemux, QtDemuxStream * stream, GNode * stbl) /* sync sample atom */ stream->stps_present = FALSE; if ((stream->stss_present = - !!qtdemux_tree_get_child_by_type_full (stbl, FOURCC_stss, + ! !qtdemux_tree_get_child_by_type_full (stbl, FOURCC_stss, &stream->stss) ? TRUE : FALSE) == TRUE) { /* copy atom data into a new buffer for later use */ stream->stss.data = g_memdup (stream->stss.data, stream->stss.size); @@ -5219,7 +5220,7 @@ qtdemux_stbl_init (GstQTDemux * qtdemux, QtDemuxStream * stream, GNode * stbl) /* partial sync sample atom */ if ((stream->stps_present = - !!qtdemux_tree_get_child_by_type_full (stbl, FOURCC_stps, + ! !qtdemux_tree_get_child_by_type_full (stbl, FOURCC_stps, &stream->stps) ? TRUE : FALSE) == TRUE) { /* copy atom data into a new buffer for later use */ stream->stps.data = g_memdup (stream->stps.data, stream->stps.size); @@ -5316,10 +5317,9 @@ qtdemux_stbl_init (GstQTDemux * qtdemux, QtDemuxStream * stream, GNode * stbl) goto corrupt_file; } - GST_DEBUG_OBJECT (qtdemux, - "allocating n_samples %u * %" G_GSIZE_FORMAT " = (%u MB)", - stream->n_samples, sizeof (QtDemuxSample), - (guint) (stream->n_samples * sizeof (QtDemuxSample)) >> 20); + GST_DEBUG_OBJECT (qtdemux, "allocating n_samples %u * %u (%.2f MB)", + stream->n_samples, (guint) sizeof (QtDemuxSample), + stream->n_samples * sizeof (QtDemuxSample) / (1024.0 * 1024.0)); if (stream->n_samples >= QTDEMUX_MAX_SAMPLE_INDEX_SIZE / sizeof (QtDemuxSample)) { @@ -5339,7 +5339,7 @@ qtdemux_stbl_init (GstQTDemux * qtdemux, QtDemuxStream * stream, GNode * stbl) /* composition time-to-sample */ if ((stream->ctts_present = - !!qtdemux_tree_get_child_by_type_full (stbl, FOURCC_ctts, + ! !qtdemux_tree_get_child_by_type_full (stbl, FOURCC_ctts, &stream->ctts) ? TRUE : FALSE) == TRUE) { /* copy atom data into a new buffer for later use */ stream->ctts.data = g_memdup (stream->ctts.data, stream->ctts.size); From 899dad89498bb00da02ea1163370e69b082bf57b Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Thu, 20 Jan 2011 12:14:08 +0100 Subject: [PATCH 252/351] qtdemux: take configured start time into account when creating the newsegment event, take the configured start time into account. --- gst/qtdemux/qtdemux.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/gst/qtdemux/qtdemux.c b/gst/qtdemux/qtdemux.c index dbb822863d..01701a2e4a 100644 --- a/gst/qtdemux/qtdemux.c +++ b/gst/qtdemux/qtdemux.c @@ -3018,9 +3018,16 @@ gst_qtdemux_activate_segment (GstQTDemux * qtdemux, QtDemuxStream * stream, start = MIN (segment->media_start + seg_time, stop); time = offset; } else { - start = segment->media_start; + if (segment->media_start >= qtdemux->segment.start) { + start = segment->media_start; + time = segment->time; + } else { + start = qtdemux->segment.start; + time = segment->time + (qtdemux->segment.start - segment->media_start); + } + + start = MAX (segment->media_start, qtdemux->segment.start); stop = MIN (segment->media_start + seg_time, stop); - time = segment->time; } GST_DEBUG_OBJECT (qtdemux, "newsegment %d from %" GST_TIME_FORMAT From c6b9f5b25ab435669816a07049b0e5a8f01e09ca Mon Sep 17 00:00:00 2001 From: Edward Hervey Date: Mon, 24 Jan 2011 18:27:52 +0100 Subject: [PATCH 253/351] avidemux: Don't consider 0 fcc_handler as uncompressed. Just avoids a warning --- gst/avi/gstavidemux.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/gst/avi/gstavidemux.c b/gst/avi/gstavidemux.c index 5e4c1be984..c2741e32bf 100644 --- a/gst/avi/gstavidemux.c +++ b/gst/avi/gstavidemux.c @@ -4417,9 +4417,10 @@ swap_line (guint8 * d1, guint8 * d2, guint8 * tmp, gint bytes) #define gst_avi_demux_is_uncompressed(fourcc) \ - (fourcc == GST_RIFF_DIB || \ - fourcc == GST_RIFF_rgb || \ - fourcc == GST_RIFF_RGB || fourcc == GST_RIFF_RAW) + (fourcc && \ + (fourcc == GST_RIFF_DIB || \ + fourcc == GST_RIFF_rgb || \ + fourcc == GST_RIFF_RGB || fourcc == GST_RIFF_RAW)) /* * Invert DIB buffers... Takes existing buffer and From 662ec520ae6a3f411741695a546193f454ce3dc6 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Mon, 10 Jan 2011 12:34:03 +0100 Subject: [PATCH 254/351] qtdemux: simplify retrieving stsd child entry atom --- gst/qtdemux/qtdemux.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/gst/qtdemux/qtdemux.c b/gst/qtdemux/qtdemux.c index 01701a2e4a..209f9e22f7 100644 --- a/gst/qtdemux/qtdemux.c +++ b/gst/qtdemux/qtdemux.c @@ -6328,13 +6328,8 @@ qtdemux_parse_trak (GstQTDemux * qtdemux, GNode * trak) esds = NULL; pasp = NULL; - mp4v = qtdemux_tree_get_child_by_type (stsd, FOURCC_mp4v); - if (!mp4v) - mp4v = qtdemux_tree_get_child_by_type (stsd, FOURCC_MP4V); - /* H264 is MPEG-4 after all, - * and qt seems to put MPEG-4 stuff in there as well */ - if (!mp4v) - mp4v = qtdemux_tree_get_child_by_type (stsd, FOURCC_avc1); + /* pick 'the' stsd child */ + mp4v = qtdemux_tree_get_child_by_type (stsd, fourcc); if (mp4v) { esds = qtdemux_tree_get_child_by_type (mp4v, FOURCC_esds); pasp = qtdemux_tree_get_child_by_type (mp4v, FOURCC_pasp); From 9f7570f132fa3bd38a2c5ae3ef5b5d7071fc382a Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Mon, 10 Jan 2011 12:34:22 +0100 Subject: [PATCH 255/351] qtdemux: support some more mpeg-4 fourcc variants --- gst/qtdemux/qtdemux.c | 14 +++++++++++--- gst/qtdemux/qtdemux_fourcc.h | 2 ++ 2 files changed, 13 insertions(+), 3 deletions(-) diff --git a/gst/qtdemux/qtdemux.c b/gst/qtdemux/qtdemux.c index 209f9e22f7..2076ba9272 100644 --- a/gst/qtdemux/qtdemux.c +++ b/gst/qtdemux/qtdemux.c @@ -4689,12 +4689,15 @@ qtdemux_parse_node (GstQTDemux * qtdemux, GNode * node, const guint8 * buffer, } case FOURCC_mp4v: case FOURCC_MP4V: + case FOURCC_fmp4: + case FOURCC_FMP4: { const guint8 *buf; guint32 version; int tlen; - GST_DEBUG_OBJECT (qtdemux, "parsing in mp4v"); + GST_DEBUG_OBJECT (qtdemux, "parsing in %" GST_FOURCC_FORMAT, + GST_FOURCC_ARGS (fourcc)); version = QT_UINT32 (buffer + 16); GST_DEBUG_OBJECT (qtdemux, "version %08x", version); if (1 || version == 0x00000000) { @@ -6439,13 +6442,17 @@ qtdemux_parse_trak (GstQTDemux * qtdemux, GNode * trak) } case FOURCC_mp4v: case FOURCC_MP4V: + case FOURCC_fmp4: + case FOURCC_FMP4: { GNode *glbl; - GST_DEBUG_OBJECT (qtdemux, "found mp4v"); + GST_DEBUG_OBJECT (qtdemux, "found %" GST_FOURCC_FORMAT, + GST_FOURCC_ARGS (fourcc)); /* codec data might be in glbl extension atom */ - glbl = qtdemux_tree_get_child_by_type (mp4v, FOURCC_glbl); + glbl = mp4v ? + qtdemux_tree_get_child_by_type (mp4v, FOURCC_glbl) : NULL; if (glbl) { guint8 *data; GstBuffer *buf; @@ -6463,6 +6470,7 @@ qtdemux_parse_trak (GstQTDemux * qtdemux, GNode * trak) gst_buffer_unref (buf); } } + break; } case FOURCC_mjp2: { diff --git a/gst/qtdemux/qtdemux_fourcc.h b/gst/qtdemux/qtdemux_fourcc.h index d7875cc668..29ad155a3f 100644 --- a/gst/qtdemux/qtdemux_fourcc.h +++ b/gst/qtdemux/qtdemux_fourcc.h @@ -75,6 +75,8 @@ G_BEGIN_DECLS #define FOURCC_mp4a GST_MAKE_FOURCC('m','p','4','a') #define FOURCC_mp4v GST_MAKE_FOURCC('m','p','4','v') #define FOURCC_MP4V GST_MAKE_FOURCC('M','P','4','V') +#define FOURCC_fmp4 GST_MAKE_FOURCC('f','m','p','4') +#define FOURCC_FMP4 GST_MAKE_FOURCC('F','M','P','4') #define FOURCC_glbl GST_MAKE_FOURCC('g','l','b','l') #define FOURCC_wave GST_MAKE_FOURCC('w','a','v','e') #define FOURCC_appl GST_MAKE_FOURCC('a','p','p','l') From 68fc03fd76059c525cdb287ff51170737febbb52 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Tue, 18 Jan 2011 14:48:04 +0100 Subject: [PATCH 256/351] matroskademux: pull mode should always report seekable ... as it no longer requires an index, but can seek by scanning as well. --- gst/matroska/matroska-demux.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/gst/matroska/matroska-demux.c b/gst/matroska/matroska-demux.c index 7bca5d19fa..4541b44c0d 100644 --- a/gst/matroska/matroska-demux.c +++ b/gst/matroska/matroska-demux.c @@ -2071,7 +2071,7 @@ gst_matroska_demux_query (GstMatroskaDemux * demux, GstPad * pad, /* assuming we'll be able to get an index ... */ seekable = demux->seekable; } else { - seekable = ! !demux->index; + seekable = TRUE; } gst_query_set_seeking (query, GST_FORMAT_TIME, seekable, From 9f34b89245a642b874939bbf2107e5c43978c5f2 Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Tue, 25 Jan 2011 21:56:19 +0200 Subject: [PATCH 257/351] rtpjitterbuffer: don't divide by 0 --- gst/rtpmanager/rtpjitterbuffer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/gst/rtpmanager/rtpjitterbuffer.c b/gst/rtpmanager/rtpjitterbuffer.c index 59307df358..341388bd25 100644 --- a/gst/rtpmanager/rtpjitterbuffer.c +++ b/gst/rtpmanager/rtpjitterbuffer.c @@ -283,7 +283,7 @@ update_buffer_level (RTPJitterBuffer * jbuf, gint * percent) if (post) { gint perc; - if (jbuf->buffering) { + if (jbuf->buffering && (jbuf->high_level != 0)) { perc = (level * 100 / jbuf->high_level); perc = MIN (perc, 100); } else { From 0add79cbf1612b8a3cdcb8ea0ad41f1e6059282e Mon Sep 17 00:00:00 2001 From: Ognyan Tonchev Date: Wed, 26 Jan 2011 08:48:43 +0000 Subject: [PATCH 258/351] matroskamux: don't leak ebml writer caps when re-using matroskamux https://bugzilla.gnome.org/show_bug.cgi?id=640542 --- gst/matroska/ebml-write.c | 6 ++++ tests/check/elements/matroskamux.c | 51 ++++++++++++++++++++++++++++++ 2 files changed, 57 insertions(+) diff --git a/gst/matroska/ebml-write.c b/gst/matroska/ebml-write.c index ed39c51e16..b66098709d 100644 --- a/gst/matroska/ebml-write.c +++ b/gst/matroska/ebml-write.c @@ -133,6 +133,12 @@ gst_ebml_write_reset (GstEbmlWrite * ebml) gst_byte_writer_free (ebml->cache); ebml->cache = NULL; } + + if (ebml->caps) { + gst_caps_unref (ebml->caps); + ebml->caps = NULL; + } + ebml->last_write_result = GST_FLOW_OK; ebml->timestamp = GST_CLOCK_TIME_NONE; } diff --git a/tests/check/elements/matroskamux.c b/tests/check/elements/matroskamux.c index 9d6f3752d3..bbc1862e38 100644 --- a/tests/check/elements/matroskamux.c +++ b/tests/check/elements/matroskamux.c @@ -387,6 +387,56 @@ GST_START_TEST (test_block_group) GST_END_TEST; +GST_START_TEST (test_reset) +{ + GstElement *matroskamux; + GstBuffer *inbuffer; + GstBuffer *outbuffer; + int num_buffers; + int i; + + matroskamux = setup_matroskamux (&srcac3template); + fail_unless (gst_element_set_state (matroskamux, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, + "could not set to playing"); + + inbuffer = gst_buffer_new_and_alloc (1); + ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1); + fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK); + num_buffers = g_list_length (buffers); + fail_unless (num_buffers >= 1, + "expected at least 1 buffer, but got only %d", num_buffers); + + fail_unless (gst_element_set_state (matroskamux, + GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS, "could not set to null"); + + fail_unless (gst_element_set_state (matroskamux, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, + "could not set to playing"); + + inbuffer = gst_buffer_new_and_alloc (1); + ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1); + fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK); + num_buffers = g_list_length (buffers); + fail_unless (num_buffers >= 2, + "expected at least 2 buffers, but got only %d", num_buffers); + + for (i = 0; i < num_buffers; ++i) { + outbuffer = GST_BUFFER (buffers->data); + fail_if (outbuffer == NULL); + buffers = g_list_remove (buffers, outbuffer); + + ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1); + gst_buffer_unref (outbuffer); + } + + cleanup_matroskamux (matroskamux); + g_list_free (buffers); + buffers = NULL; +} + +GST_END_TEST; + static Suite * matroskamux_suite (void) { @@ -397,6 +447,7 @@ matroskamux_suite (void) tcase_add_test (tc_chain, test_ebml_header); tcase_add_test (tc_chain, test_vorbis_header); tcase_add_test (tc_chain, test_block_group); + tcase_add_test (tc_chain, test_reset); return s; } From cb76a13cc98367b955c8e2a02c4d4dacd2a5bd04 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Thu, 27 Jan 2011 12:16:46 +0100 Subject: [PATCH 259/351] qtdemux: mind rounding issues when converting from global time to mov time In particular, this avoids missing the intended keyframe when first converting from the frame's mov time to global segment time, and then back from global time to mov time when activating the segment. --- gst/qtdemux/qtdemux.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/gst/qtdemux/qtdemux.c b/gst/qtdemux/qtdemux.c index 2076ba9272..b6457ce28e 100644 --- a/gst/qtdemux/qtdemux.c +++ b/gst/qtdemux/qtdemux.c @@ -848,7 +848,8 @@ gst_qtdemux_find_index (GstQTDemux * qtdemux, QtDemuxStream * str, guint32 index; /* convert media_time to mov format */ - media_time = gst_util_uint64_scale (media_time, str->timescale, GST_SECOND); + media_time = + gst_util_uint64_scale_ceil (media_time, str->timescale, GST_SECOND); result = gst_util_array_binary_search (str->samples, str->stbl_index + 1, sizeof (QtDemuxSample), (GCompareDataFunc) find_func, From 7def8c9b9860be41ffdfa8436b7442e36be6cea8 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Wed, 26 Jan 2011 22:21:31 +0100 Subject: [PATCH 260/351] udp: use socklen_t where appropriate rather than custom type In particular, fixes Cygwin build where socklen_t is defined as int in line with native win32 api definition. --- gst/udp/gstmultiudpsink.c | 6 +----- gst/udp/gstudpnetutils.h | 3 +++ gst/udp/gstudpsrc.c | 9 +-------- 3 files changed, 5 insertions(+), 13 deletions(-) diff --git a/gst/udp/gstmultiudpsink.c b/gst/udp/gstmultiudpsink.c index 78e491b2b6..dd93e3f9d0 100644 --- a/gst/udp/gstmultiudpsink.c +++ b/gst/udp/gstmultiudpsink.c @@ -913,11 +913,7 @@ gst_multiudpsink_init_send (GstMultiUDPSink * sink) GList *clients; GstUDPClient *client; int sndsize, ret; -#ifdef G_OS_WIN32 - gint len; -#else - guint len; -#endif + socklen_t len; if (sink->sockfd == -1) { GST_DEBUG_OBJECT (sink, "creating sockets"); diff --git a/gst/udp/gstudpnetutils.h b/gst/udp/gstudpnetutils.h index 6dd8d75a79..ec863bd7db 100644 --- a/gst/udp/gstudpnetutils.h +++ b/gst/udp/gstudpnetutils.h @@ -32,6 +32,9 @@ #define WINVER 0x0501 #include #include +#ifndef socklen_t +#define socklen_t int +#endif /* Needed for GstObject and GST_WARNING_OBJECT */ #include diff --git a/gst/udp/gstudpsrc.c b/gst/udp/gstudpsrc.c index b76385b069..f00c74b364 100644 --- a/gst/udp/gstudpsrc.c +++ b/gst/udp/gstudpsrc.c @@ -117,9 +117,6 @@ #endif #include -#ifdef G_OS_WIN32 -typedef int socklen_t; -#endif #ifdef HAVE_FIONREAD_IN_SYS_FILIO #include @@ -778,11 +775,7 @@ gst_udpsrc_start (GstBaseSrc * bsrc) gint ret; int rcvsize; struct sockaddr_storage bind_address; -#ifdef G_OS_WIN32 - gint len; -#else - guint len; -#endif + socklen_t len; src = GST_UDPSRC (bsrc); if (src->sockfd == -1) { From 8d90dba55e8c4517e54b8ea900fee24145d75c7c Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Thu, 27 Jan 2011 16:02:46 +0100 Subject: [PATCH 261/351] avidemux: initialize local variable to please mingw32 compiler --- gst/avi/gstavidemux.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/gst/avi/gstavidemux.c b/gst/avi/gstavidemux.c index c2741e32bf..a4a0525aa6 100644 --- a/gst/avi/gstavidemux.c +++ b/gst/avi/gstavidemux.c @@ -696,7 +696,7 @@ gst_avi_demux_seek_streams_index (GstAviDemux * avi, guint64 offset, GstAviIndexEntry *entry; gint i; gint64 val, min = offset; - guint index; + guint index = 0; for (i = 0; i < avi->num_streams; i++) { stream = &avi->stream[i]; From b0b124ec902ae1c5e643726e4d68f71becc680fd Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Thu, 27 Jan 2011 15:35:06 +0000 Subject: [PATCH 262/351] win32: fix DEFAULT_AUDIOSINK, should be direct*sound*sink https://bugzilla.gnome.org/show_bug.cgi?id=640705 --- configure.ac | 2 +- win32/common/config.h | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/configure.ac b/configure.ac index 5e93f5d8c9..d86ffde1af 100644 --- a/configure.ac +++ b/configure.ac @@ -1247,7 +1247,7 @@ sed \ -e 's/.* PLUGINDIR$/#ifdef _DEBUG\n# define PLUGINDIR PREFIX "\\\\debug\\\\lib\\\\gstreamer-0.10"\n#else\n# define PLUGINDIR PREFIX "\\\\lib\\\\gstreamer-0.10"\n#endif/' \ -e 's/.* USE_BINARY_REGISTRY$/#define USE_BINARY_REGISTRY/' \ -e 's/.* VERSION$/#define VERSION "'$VERSION'"/' \ - -e "s/.* DEFAULT_AUDIOSINK$/#define DEFAULT_AUDIOSINK \"directaudiosink\"/" \ + -e "s/.* DEFAULT_AUDIOSINK$/#define DEFAULT_AUDIOSINK \"directsoundsink\"/" \ -e "s/.* DEFAULT_AUDIOSRC$/#define DEFAULT_AUDIOSRC \"audiotestsrc\"/" \ -e "s/.* DEFAULT_VIDEOSRC$/#define DEFAULT_VIDEOSRC \"videotestsrc\"/" \ -e "s/.* DEFAULT_VISUALIZER$/#define DEFAULT_VISUALIZER \"goom\"/" \ diff --git a/win32/common/config.h b/win32/common/config.h index d2f750caec..8abfbb58dd 100644 --- a/win32/common/config.h +++ b/win32/common/config.h @@ -9,7 +9,7 @@ #undef AC_APPLE_UNIVERSAL_BUILD /* Default audio sink */ -#define DEFAULT_AUDIOSINK "directaudiosink" +#define DEFAULT_AUDIOSINK "directsoundsink" /* Default audio source */ #define DEFAULT_AUDIOSRC "audiotestsrc" @@ -49,7 +49,7 @@ #define GST_PACKAGE_ORIGIN "Unknown package origin" /* GStreamer package release date/time for plugins as YYYY-MM-DD */ -#define GST_PACKAGE_RELEASE_DATETIME "2011-01-24T00:02Z" +#define GST_PACKAGE_RELEASE_DATETIME "2011-01-27T15:30Z" /* struct v4l2_buffer missing */ #undef GST_V4L2_MISSING_BUFDECL From ede0635d45342f6e06aa213350ce006c2dcb6628 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Mon, 31 Jan 2011 13:44:45 +0000 Subject: [PATCH 263/351] deinterlace: simplify template caps We can merge all the YUV variants into one single structure. --- gst/deinterlace/gstdeinterlace.c | 30 ++++++++++-------------------- 1 file changed, 10 insertions(+), 20 deletions(-) diff --git a/gst/deinterlace/gstdeinterlace.c b/gst/deinterlace/gstdeinterlace.c index 1ad960e87d..300f23e7fd 100644 --- a/gst/deinterlace/gstdeinterlace.c +++ b/gst/deinterlace/gstdeinterlace.c @@ -160,34 +160,24 @@ gst_deinterlace_modes_get_type (void) return deinterlace_modes_type; } +#define DEINTERLACE_CAPS \ + GST_VIDEO_CAPS_YUV ("{ AYUV, Y444, YUY2, YVYU, UYVY, Y42B, I420, YV12, Y41B }") ";" \ + GST_VIDEO_CAPS_ARGB ";" GST_VIDEO_CAPS_ABGR ";" \ + GST_VIDEO_CAPS_RGBA ";" GST_VIDEO_CAPS_BGRA ";" \ + GST_VIDEO_CAPS_xRGB ";" GST_VIDEO_CAPS_xBGR ";" \ + GST_VIDEO_CAPS_RGBx ";" GST_VIDEO_CAPS_BGRx ";" \ + GST_VIDEO_CAPS_RGB ";" GST_VIDEO_CAPS_BGR + static GstStaticPadTemplate src_templ = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, - GST_STATIC_CAPS (GST_VIDEO_CAPS_YUV ("AYUV") ";" GST_VIDEO_CAPS_YUV ("Y444") - ";" GST_VIDEO_CAPS_YUV ("YUY2") ";" GST_VIDEO_CAPS_YUV ("YVYU") ";" - GST_VIDEO_CAPS_YUV ("UYVY") ";" - GST_VIDEO_CAPS_YUV ("Y42B") ";" GST_VIDEO_CAPS_YUV ("I420") ";" - GST_VIDEO_CAPS_YUV ("YV12") ";" GST_VIDEO_CAPS_YUV ("Y41B") ";" - GST_VIDEO_CAPS_ARGB ";" GST_VIDEO_CAPS_ABGR ";" - GST_VIDEO_CAPS_RGBA ";" GST_VIDEO_CAPS_BGRA ";" - GST_VIDEO_CAPS_xRGB ";" GST_VIDEO_CAPS_xBGR ";" - GST_VIDEO_CAPS_RGBx ";" GST_VIDEO_CAPS_BGRx ";" - GST_VIDEO_CAPS_RGB ";" GST_VIDEO_CAPS_BGR) + GST_STATIC_CAPS (DEINTERLACE_CAPS) ); static GstStaticPadTemplate sink_templ = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, - GST_STATIC_CAPS (GST_VIDEO_CAPS_YUV ("AYUV") ";" GST_VIDEO_CAPS_YUV ("Y444") - ";" GST_VIDEO_CAPS_YUV ("YUY2") ";" GST_VIDEO_CAPS_YUV ("YVYU") ";" - GST_VIDEO_CAPS_YUV ("UYVY") ";" - GST_VIDEO_CAPS_YUV ("Y42B") ";" GST_VIDEO_CAPS_YUV ("I420") ";" - GST_VIDEO_CAPS_YUV ("YV12") ";" GST_VIDEO_CAPS_YUV ("Y41B") ";" - GST_VIDEO_CAPS_ARGB ";" GST_VIDEO_CAPS_ABGR ";" - GST_VIDEO_CAPS_RGBA ";" GST_VIDEO_CAPS_BGRA ";" - GST_VIDEO_CAPS_xRGB ";" GST_VIDEO_CAPS_xBGR ";" - GST_VIDEO_CAPS_RGBx ";" GST_VIDEO_CAPS_BGRx ";" - GST_VIDEO_CAPS_RGB ";" GST_VIDEO_CAPS_BGR) + GST_STATIC_CAPS (DEINTERLACE_CAPS) ); static void gst_deinterlace_finalize (GObject * self); From 1e2c1467ae042a3c6bb1a6bc0c07aeff13ec5edb Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Mon, 31 Jan 2011 05:58:36 +0100 Subject: [PATCH 264/351] Pulsesink: Allow chunks up to bufsize instead of segsize By allowing larger chunks to be sent, PulseAudio will have a lower CPU usage. This is especially important on low-end machines, where PulseAudio can crash if packets are coming in at a higher rate than PulseAudio can process them. Signed-off-by: David Henningsson --- ext/pulse/pulsesink.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/ext/pulse/pulsesink.c b/ext/pulse/pulsesink.c index 9bebfec3bb..295d93f494 100644 --- a/ext/pulse/pulsesink.c +++ b/ext/pulse/pulsesink.c @@ -1339,11 +1339,11 @@ gst_pulseringbuffer_commit (GstRingBuffer * buf, guint64 * sample, towrite = out_samples * bps; - /* Only ever write segsize bytes at once. This will - * also limit the PA shm buffer to segsize + /* Only ever write bufsize bytes at once. This will + * also limit the PA shm buffer to bufsize */ - if (towrite > buf->spec.segsize) - towrite = buf->spec.segsize; + if (towrite > bufsize) + towrite = bufsize; if ((pbuf->m_writable < towrite) || (offset != pbuf->m_lastoffset)) { /* if no room left or discontinuity in offset, @@ -1392,9 +1392,9 @@ gst_pulseringbuffer_commit (GstRingBuffer * buf, guint64 * sample, } /* make sure we only buffer up latency-time samples */ - if (pbuf->m_writable > buf->spec.segsize) { + if (pbuf->m_writable > bufsize) { /* limit buffering to latency-time value */ - pbuf->m_writable = buf->spec.segsize; + pbuf->m_writable = bufsize; GST_LOG_OBJECT (psink, "Limiting buffering to %" G_GSIZE_FORMAT, pbuf->m_writable); @@ -1413,9 +1413,9 @@ gst_pulseringbuffer_commit (GstRingBuffer * buf, guint64 * sample, pbuf->m_writable); /* Just to make sure that we didn't get more than requested */ - if (pbuf->m_writable > buf->spec.segsize) { + if (pbuf->m_writable > bufsize) { /* limit buffering to latency-time value */ - pbuf->m_writable = buf->spec.segsize; + pbuf->m_writable = bufsize; } } From 07ebec51f50a1f3269654791bcfb3f5b6854b7fd Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Olivier=20Cr=C3=AAte?= Date: Mon, 31 Jan 2011 17:56:18 -0500 Subject: [PATCH 265/351] rtppcmapay: Rename the class to have the right name It was name pmca instead of pcma and made debug logs hard to search. --- gst/rtp/gstrtppcmapay.c | 6 +++--- gst/rtp/gstrtppcmapay.h | 12 ++++++------ 2 files changed, 9 insertions(+), 9 deletions(-) diff --git a/gst/rtp/gstrtppcmapay.c b/gst/rtp/gstrtppcmapay.c index 97b31a9d31..d897ce1289 100644 --- a/gst/rtp/gstrtppcmapay.c +++ b/gst/rtp/gstrtppcmapay.c @@ -53,7 +53,7 @@ static GstStaticPadTemplate gst_rtp_pcma_pay_src_template = static gboolean gst_rtp_pcma_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps); -GST_BOILERPLATE (GstRtpPmcaPay, gst_rtp_pcma_pay, GstBaseRTPAudioPayload, +GST_BOILERPLATE (GstRtpPcmaPay, gst_rtp_pcma_pay, GstBaseRTPAudioPayload, GST_TYPE_BASE_RTP_AUDIO_PAYLOAD); static void @@ -72,7 +72,7 @@ gst_rtp_pcma_pay_base_init (gpointer klass) } static void -gst_rtp_pcma_pay_class_init (GstRtpPmcaPayClass * klass) +gst_rtp_pcma_pay_class_init (GstRtpPcmaPayClass * klass) { GstBaseRTPPayloadClass *gstbasertppayload_class; @@ -82,7 +82,7 @@ gst_rtp_pcma_pay_class_init (GstRtpPmcaPayClass * klass) } static void -gst_rtp_pcma_pay_init (GstRtpPmcaPay * rtppcmapay, GstRtpPmcaPayClass * klass) +gst_rtp_pcma_pay_init (GstRtpPcmaPay * rtppcmapay, GstRtpPcmaPayClass * klass) { GstBaseRTPAudioPayload *basertpaudiopayload; diff --git a/gst/rtp/gstrtppcmapay.h b/gst/rtp/gstrtppcmapay.h index 7a39d7dd7f..a51e975e4b 100644 --- a/gst/rtp/gstrtppcmapay.h +++ b/gst/rtp/gstrtppcmapay.h @@ -21,26 +21,26 @@ G_BEGIN_DECLS -typedef struct _GstRtpPmcaPay GstRtpPmcaPay; -typedef struct _GstRtpPmcaPayClass GstRtpPmcaPayClass; +typedef struct _GstRtpPcmaPay GstRtpPcmaPay; +typedef struct _GstRtpPcmaPayClass GstRtpPcmaPayClass; #define GST_TYPE_RTP_PCMA_PAY \ (gst_rtp_pcma_pay_get_type()) #define GST_RTP_PCMA_PAY(obj) \ - (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_PCMA_PAY,GstRtpPmcaPay)) + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_PCMA_PAY,GstRtpPcmaPay)) #define GST_RTP_PCMA_PAY_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_PCMA_PAY,GstRtpPmcaPayClass)) + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_PCMA_PAY,GstRtpPcmaPayClass)) #define GST_IS_RTP_PCMA_PAY(obj) \ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_PCMA_PAY)) #define GST_IS_RTP_PCMA_PAY_CLASS(klass) \ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_PCMA_PAY)) -struct _GstRtpPmcaPay +struct _GstRtpPcmaPay { GstBaseRTPAudioPayload audiopayload; }; -struct _GstRtpPmcaPayClass +struct _GstRtpPcmaPayClass { GstBaseRTPAudioPayloadClass parent_class; }; From f95c30a4131a009c9b1e19ed94200fb2aa71f6bd Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Tue, 1 Feb 2011 16:38:20 +0100 Subject: [PATCH 266/351] j2kpay: skip EPH packets Include EPH markers into the previous chunk of packets. --- gst/rtp/gstrtpj2kpay.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/gst/rtp/gstrtpj2kpay.c b/gst/rtp/gstrtpj2kpay.c index 7385d3bdcb..3734534f8b 100644 --- a/gst/rtp/gstrtpj2kpay.c +++ b/gst/rtp/gstrtpj2kpay.c @@ -73,6 +73,7 @@ typedef enum J2K_MARKER_SOC = 0x4F, J2K_MARKER_SOT = 0x90, J2K_MARKER_SOP = 0x91, + J2K_MARKER_EPH = 0x92, J2K_MARKER_SOD = 0x93, J2K_MARKER_EOC = 0xD9 } RtpJ2KMarker; @@ -236,6 +237,10 @@ find_pu_end (GstRtpJ2KPay * pay, const guint8 * data, guint size, return offset - 2; cut_sop = TRUE; break; + case J2K_MARKER_EPH: + /* just skip over EPH */ + GST_LOG_OBJECT (pay, "found EPH at %u", offset); + break; default: if (offset >= state->next_sot) { GST_LOG_OBJECT (pay, "reached next SOT at %u", offset); From 9f073459e092097aec6f2d98189fbb0e8e72931e Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Olivier=20Cr=C3=AAte?= Date: Fri, 1 Oct 2010 17:19:16 -0400 Subject: [PATCH 267/351] rtpsession: Emit "on-ssrc-validated" when validating by RTCP Emit "on-ssrc-validated" if the SSRC is validated by receiving a RTCP SDES packet. --- gst/rtpmanager/rtpsession.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c index 2698f3c7a9..176a86f498 100644 --- a/gst/rtpmanager/rtpsession.c +++ b/gst/rtpmanager/rtpsession.c @@ -1749,7 +1749,7 @@ rtp_session_process_sdes (RTPSession * sess, GstRTCPPacket * packet, i = 0; while (more_items) { guint32 ssrc; - gboolean changed, created; + gboolean changed, created, validated; RTPSource *source; GstStructure *sdes; @@ -1802,10 +1802,13 @@ rtp_session_process_sdes (RTPSession * sess, GstRTCPPacket * packet, /* takes ownership of sdes */ changed = rtp_source_set_sdes_struct (source, sdes); + validated = !RTP_SOURCE_IS_ACTIVE (source); source->validated = TRUE; if (created) on_new_ssrc (sess, source); + if (validated) + on_ssrc_validated (sess, source); if (changed) on_ssrc_sdes (sess, source); From c7b1ce7310d903fe265a8b5e7c773b17790334b4 Mon Sep 17 00:00:00 2001 From: Olivier Crete Date: Fri, 18 Jun 2010 19:12:40 -0400 Subject: [PATCH 268/351] rtpssrcdemux: Tag upstream custom events with SSRC --- gst/rtpmanager/gstrtpssrcdemux.c | 30 ++++++++++++++++++++++++++---- 1 file changed, 26 insertions(+), 4 deletions(-) diff --git a/gst/rtpmanager/gstrtpssrcdemux.c b/gst/rtpmanager/gstrtpssrcdemux.c index c8348da51e..65d4475202 100644 --- a/gst/rtpmanager/gstrtpssrcdemux.c +++ b/gst/rtpmanager/gstrtpssrcdemux.c @@ -208,6 +208,7 @@ create_demux_pad_for_ssrc (GstRtpSsrcDemux * demux, guint32 ssrc, gst_rtp_ssrc_demux_iterate_internal_links); gst_pad_set_active (rtp_pad, TRUE); + gst_pad_set_event_function (rtcp_pad, gst_rtp_ssrc_demux_src_event); gst_pad_set_iterate_internal_links_function (rtcp_pad, gst_rtp_ssrc_demux_iterate_internal_links); gst_pad_set_active (rtcp_pad, TRUE); @@ -621,18 +622,39 @@ static gboolean gst_rtp_ssrc_demux_src_event (GstPad * pad, GstEvent * event) { GstRtpSsrcDemux *demux; - gboolean res = FALSE; + const GstStructure *s; demux = GST_RTP_SSRC_DEMUX (gst_pad_get_parent (pad)); switch (GST_EVENT_TYPE (event)) { - case GST_EVENT_SEEK: + case GST_EVENT_CUSTOM_UPSTREAM: + case GST_EVENT_CUSTOM_BOTH: + case GST_EVENT_CUSTOM_BOTH_OOB: + s = gst_event_get_structure (event); + if (s && !gst_structure_has_field (s, "ssrc")) { + GSList *walk; + + for (walk = demux->srcpads; walk; walk = g_slist_next (walk)) { + GstRtpSsrcDemuxPad *dpad = (GstRtpSsrcDemuxPad *) walk->data; + + if (dpad->rtp_pad == pad || dpad->rtcp_pad == pad) { + event = + GST_EVENT_CAST (gst_mini_object_make_writable + (GST_MINI_OBJECT_CAST (event))); + gst_structure_set (event->structure, "ssrc", G_TYPE_UINT, + dpad->ssrc, NULL); + break; + } + } + } + break; default: - res = gst_pad_event_default (pad, event); break; } + gst_object_unref (demux); - return res; + + return gst_pad_event_default (pad, event); } static GstIterator * From 589b254ce594abc0763a731fc1293b63e9fd0b06 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Olivier=20Cr=C3=AAte?= Date: Sat, 19 Jun 2010 19:11:06 -0400 Subject: [PATCH 269/351] rtpptdemux: Tag upstream custom events with payload type --- gst/rtpmanager/gstrtpptdemux.c | 45 ++++++++++++++++++++++++++++++++++ 1 file changed, 45 insertions(+) diff --git a/gst/rtpmanager/gstrtpptdemux.c b/gst/rtpmanager/gstrtpptdemux.c index 6b49483fb9..78c4956d38 100644 --- a/gst/rtpmanager/gstrtpptdemux.c +++ b/gst/rtpmanager/gstrtpptdemux.c @@ -132,6 +132,9 @@ static void gst_rtp_pt_demux_clear_pt_map (GstRtpPtDemux * rtpdemux); static GstRtpPtDemuxPad *find_pad_for_pt (GstRtpPtDemux * rtpdemux, guint8 pt); +static gboolean gst_rtp_pt_demux_src_event (GstPad * pad, GstEvent * event); + + static guint gst_rtp_pt_demux_signals[LAST_SIGNAL] = { 0 }; static void @@ -329,6 +332,7 @@ gst_rtp_pt_demux_chain (GstPad * pad, GstBuffer * buf) srcpad = gst_pad_new_from_template (templ, padname); gst_pad_use_fixed_caps (srcpad); g_free (padname); + gst_pad_set_event_function (srcpad, gst_rtp_pt_demux_src_event); caps = gst_rtp_pt_demux_get_caps (rtpdemux, pt); if (!caps) @@ -462,6 +466,47 @@ gst_rtp_pt_demux_sink_event (GstPad * pad, GstEvent * event) } +static gboolean +gst_rtp_pt_demux_src_event (GstPad * pad, GstEvent * event) +{ + GstRtpPtDemux *demux; + const GstStructure *s; + + demux = GST_RTP_PT_DEMUX (gst_pad_get_parent (pad)); + + switch (GST_EVENT_TYPE (event)) { + case GST_EVENT_CUSTOM_UPSTREAM: + case GST_EVENT_CUSTOM_BOTH: + case GST_EVENT_CUSTOM_BOTH_OOB: + s = gst_event_get_structure (event); + if (s && !gst_structure_has_field (s, "payload")) { + GSList *walk; + + for (walk = demux->srcpads; walk; walk = g_slist_next (walk)) { + GstRtpPtDemuxPad *dpad = (GstRtpPtDemuxPad *) walk->data; + + if (dpad->pad == pad) { + event = + GST_EVENT_CAST (gst_mini_object_make_writable + (GST_MINI_OBJECT_CAST (event))); + gst_structure_set (event->structure, + "payload", G_TYPE_UINT, dpad->pt, NULL); + break; + } + } + } + break; + default: + break; + } + + gst_object_unref (demux); + + return gst_pad_event_default (pad, event); +} + + + /* * Reserves resources for the object. */ From cdb546574177d88bcc2b2456fa2e87755db98ed6 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Olivier=20Cr=C3=AAte?= Date: Mon, 14 Jun 2010 18:40:33 -0400 Subject: [PATCH 270/351] rtpsession: Emit signal when sending a compound RTCP packet This allows users to add extra RTCP packets to the compound RTCP packet. --- gst/rtpmanager/gstrtpbin-marshal.list | 1 + gst/rtpmanager/rtpsession.c | 33 +++++++++++++++++++++++++++ gst/rtpmanager/rtpsession.h | 2 ++ 3 files changed, 36 insertions(+) diff --git a/gst/rtpmanager/gstrtpbin-marshal.list b/gst/rtpmanager/gstrtpbin-marshal.list index 01b530be41..0b0d81ff3a 100644 --- a/gst/rtpmanager/gstrtpbin-marshal.list +++ b/gst/rtpmanager/gstrtpbin-marshal.list @@ -7,3 +7,4 @@ VOID:UINT VOID:UINT,UINT VOID:OBJECT,OBJECT UINT64:BOOL,UINT64 +BOOL:POINTER,BOOL diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c index 176a86f498..4b1513d180 100644 --- a/gst/rtpmanager/rtpsession.c +++ b/gst/rtpmanager/rtpsession.c @@ -42,6 +42,7 @@ enum SIGNAL_ON_BYE_TIMEOUT, SIGNAL_ON_TIMEOUT, SIGNAL_ON_SENDER_TIMEOUT, + SIGNAL_ON_SENDING_RTCP, LAST_SIGNAL }; @@ -107,6 +108,16 @@ static GstFlowReturn rtp_session_schedule_bye_locked (RTPSession * sess, static GstClockTime calculate_rtcp_interval (RTPSession * sess, gboolean deterministic, gboolean first); +static gboolean +accumulate_trues (GSignalInvocationHint * ihint, GValue * return_accu, + const GValue * handler_return, gpointer data) +{ + if (g_value_get_boolean (handler_return)) + g_value_set_boolean (return_accu, TRUE); + + return TRUE; +} + static void rtp_session_class_init (RTPSessionClass * klass) { @@ -240,6 +251,24 @@ rtp_session_class_init (RTPSessionClass * klass) NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1, RTP_TYPE_SOURCE); + /** + * RTPSession::on-sending-rtcp + * @session: the object which received the signal + * @buffer: the #GstBuffer containing the RTCP packet about to be sent + * @early: %TRUE if the packet is early, %FALSE if it is regular + * + * This signal is emitted before sending an RTCP packet, it can be used + * to add extra RTCP Packets. + * + * Returns: %TRUE if the RTCP buffer should NOT be suppressed, %FALSE + * if suppressing it is acceptable + */ + rtp_session_signals[SIGNAL_ON_SENDING_RTCP] = + g_signal_new ("on-sending-rtcp", G_TYPE_FROM_CLASS (klass), + G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sending_rtcp), + accumulate_trues, NULL, gst_rtp_bin_marshal_BOOLEAN__POINTER_BOOLEAN, + G_TYPE_BOOLEAN, 2, G_TYPE_POINTER, G_TYPE_BOOLEAN); + g_object_class_install_property (gobject_class, PROP_INTERNAL_SSRC, g_param_spec_uint ("internal-ssrc", "Internal SSRC", "The internal SSRC used for the session", @@ -2704,6 +2733,10 @@ rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time, /* push out the RTCP packet */ if (data.rtcp) { + /* Give the user a change to add its own packet */ + g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDING_RTCP], 0, + data.rtcp, FALSE, NULL); + /* close the RTCP packet */ gst_rtcp_buffer_end (data.rtcp); diff --git a/gst/rtpmanager/rtpsession.h b/gst/rtpmanager/rtpsession.h index 3fc9874675..0f86856cd7 100644 --- a/gst/rtpmanager/rtpsession.h +++ b/gst/rtpmanager/rtpsession.h @@ -224,6 +224,8 @@ struct _RTPSessionClass { void (*on_bye_timeout) (RTPSession *sess, RTPSource *source); void (*on_timeout) (RTPSession *sess, RTPSource *source); void (*on_sender_timeout) (RTPSession *sess, RTPSource *source); + gboolean (*on_sending_rtcp) (RTPSession *sess, GstBuffer *buffer, + gboolean early); }; GType rtp_session_get_type (void); From 975e1fecb3047212c06ffc4b1ab99c469a17b2e4 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Olivier=20Cr=C3=AAte?= Date: Tue, 1 Jun 2010 19:28:01 -0400 Subject: [PATCH 271/351] rtpsession: Add property for minimum interval between Regular RTCP messages This can be changed according to RFC 4585 --- gst/rtpmanager/gstrtpsession.c | 16 ++++++++++++++++ gst/rtpmanager/rtpsession.c | 14 ++++++++++++++ gst/rtpmanager/rtpstats.c | 6 ++++++ gst/rtpmanager/rtpstats.h | 3 +++ 4 files changed, 39 insertions(+) diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c index 751b7f8cce..f10ab92e65 100644 --- a/gst/rtpmanager/gstrtpsession.c +++ b/gst/rtpmanager/gstrtpsession.c @@ -199,6 +199,7 @@ enum #define DEFAULT_NUM_SOURCES 0 #define DEFAULT_NUM_ACTIVE_SOURCES 0 #define DEFAULT_USE_PIPELINE_CLOCK FALSE +#define DEFAULT_RTCP_MIN_INTERVAL (RTP_STATS_MIN_INTERVAL * GST_SECOND) enum { @@ -213,6 +214,7 @@ enum PROP_NUM_ACTIVE_SOURCES, PROP_INTERNAL_SESSION, PROP_USE_PIPELINE_CLOCK, + PROP_RTCP_MIN_INTERVAL, PROP_LAST }; @@ -588,6 +590,12 @@ gst_rtp_session_class_init (GstRtpSessionClass * klass) DEFAULT_USE_PIPELINE_CLOCK, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + g_object_class_install_property (gobject_class, PROP_RTCP_MIN_INTERVAL, + g_param_spec_uint64 ("rtcp-min-interval", "Minimum RTCP interval", + "Minimum interval between Regular RTCP packet (in ns)", + 0, G_MAXUINT64, DEFAULT_RTCP_MIN_INTERVAL, + G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_rtp_session_change_state); gstelement_class->request_new_pad = @@ -693,6 +701,10 @@ gst_rtp_session_set_property (GObject * object, guint prop_id, case PROP_USE_PIPELINE_CLOCK: priv->use_pipeline_clock = g_value_get_boolean (value); break; + case PROP_RTCP_MIN_INTERVAL: + g_object_set_property (G_OBJECT (priv->session), "rtcp-min-interval", + value); + break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; @@ -745,6 +757,10 @@ gst_rtp_session_get_property (GObject * object, guint prop_id, case PROP_USE_PIPELINE_CLOCK: g_value_set_boolean (value, priv->use_pipeline_clock); break; + case PROP_RTCP_MIN_INTERVAL: + g_object_get_property (G_OBJECT (priv->session), "rtcp-min-interval", + value); + break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c index 4b1513d180..43a4e2e54f 100644 --- a/gst/rtpmanager/rtpsession.c +++ b/gst/rtpmanager/rtpsession.c @@ -56,6 +56,7 @@ enum #define DEFAULT_NUM_SOURCES 0 #define DEFAULT_NUM_ACTIVE_SOURCES 0 #define DEFAULT_SOURCES NULL +#define DEFAULT_RTCP_MIN_INTERVAL (RTP_STATS_MIN_INTERVAL * GST_SECOND) enum { @@ -72,6 +73,7 @@ enum PROP_NUM_ACTIVE_SOURCES, PROP_SOURCES, PROP_FAVOR_NEW, + PROP_RTCP_MIN_INTERVAL, PROP_LAST }; @@ -360,6 +362,11 @@ rtp_session_class_init (RTPSessionClass * klass) "Resolve SSRC conflict in favor of new sources", FALSE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + g_object_class_install_property (gobject_class, PROP_RTCP_MIN_INTERVAL, + g_param_spec_uint64 ("rtcp-min-interval", "Minimum RTCP interval", + "Minimum interval between Regular RTCP packet (in ns)", + 0, G_MAXUINT64, DEFAULT_RTCP_MIN_INTERVAL, + G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); klass->get_source_by_ssrc = GST_DEBUG_FUNCPTR (rtp_session_get_source_by_ssrc); @@ -505,6 +512,10 @@ rtp_session_set_property (GObject * object, guint prop_id, case PROP_FAVOR_NEW: sess->favor_new = g_value_get_boolean (value); break; + case PROP_RTCP_MIN_INTERVAL: + rtp_stats_set_min_interval (&sess->stats, + (gdouble) g_value_get_uint64 (value) / GST_SECOND); + break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; @@ -556,6 +567,9 @@ rtp_session_get_property (GObject * object, guint prop_id, case PROP_FAVOR_NEW: g_value_set_boolean (value, sess->favor_new); break; + case PROP_RTCP_MIN_INTERVAL: + g_value_set_uint64 (value, sess->stats.min_interval * GST_SECOND); + break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; diff --git a/gst/rtpmanager/rtpstats.c b/gst/rtpmanager/rtpstats.c index a31e7e52ff..8fe1d1fce6 100644 --- a/gst/rtpmanager/rtpstats.c +++ b/gst/rtpmanager/rtpstats.c @@ -286,3 +286,9 @@ rtp_stats_get_packets_lost (const RTPSourceStats * stats) return lost; } + +void +rtp_stats_set_min_interval (RTPSessionStats * stats, gdouble min_interval) +{ + stats->min_interval = min_interval; +} diff --git a/gst/rtpmanager/rtpstats.h b/gst/rtpmanager/rtpstats.h index d657cdfed0..643da004ab 100644 --- a/gst/rtpmanager/rtpstats.h +++ b/gst/rtpmanager/rtpstats.h @@ -195,4 +195,7 @@ GstClockTime rtp_stats_calculate_rtcp_interval (RTPSessionStats *stats, gbo GstClockTime rtp_stats_add_rtcp_jitter (RTPSessionStats *stats, GstClockTime interval); GstClockTime rtp_stats_calculate_bye_interval (RTPSessionStats *stats); gint64 rtp_stats_get_packets_lost (const RTPSourceStats *stats); + +void rtp_stats_set_min_interval (RTPSessionStats *stats, + gdouble min_interval); #endif /* __RTP_STATS_H__ */ From 1bde4272509c30f0818af865c2f8f51014f7ac14 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Olivier=20Cr=C3=AAte?= Date: Tue, 15 Jun 2010 18:39:47 -0400 Subject: [PATCH 272/351] rtpsession: Add method to request early RTCP packet Implement the early mode defined in RFC 4585. In this mode, RTCP feedback packets are sent early to notifier. --- gst/rtpmanager/rtpsession.c | 153 ++++++++++++++++++++++++++++++++---- gst/rtpmanager/rtpsession.h | 8 ++ 2 files changed, 145 insertions(+), 16 deletions(-) diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c index 43a4e2e54f..7d806eec04 100644 --- a/gst/rtpmanager/rtpsession.c +++ b/gst/rtpmanager/rtpsession.c @@ -421,6 +421,7 @@ rtp_session_init (RTPSession * sess) rtp_source_set_sdes_string (sess->source, GST_RTCP_SDES_TOOL, "GStreamer"); sess->first_rtcp = TRUE; + sess->allow_early = TRUE; GST_DEBUG ("%p: session using SSRC: %08x", sess, sess->source->ssrc); } @@ -2223,6 +2224,7 @@ rtp_session_schedule_bye_locked (RTPSession * sess, const gchar * reason, sess->stats.bye_members = 1; sess->first_rtcp = TRUE; sess->sent_bye = FALSE; + sess->allow_early = TRUE; /* reschedule transmission */ sess->last_rtcp_send_time = current_time; @@ -2286,6 +2288,11 @@ rtp_session_next_timeout (RTPSession * sess, GstClockTime current_time) RTP_SESSION_LOCK (sess); + if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) { + result = sess->next_early_rtcp_time; + goto early_exit; + } + result = sess->next_rtcp_check_time; GST_DEBUG ("current time: %" GST_TIME_FORMAT ", next :%" GST_TIME_FORMAT, @@ -2326,7 +2333,11 @@ rtp_session_next_timeout (RTPSession * sess, GstClockTime current_time) sess->next_rtcp_check_time = result; - GST_DEBUG ("next timeout: %" GST_TIME_FORMAT, GST_TIME_ARGS (result)); +early_exit: + + GST_DEBUG ("current time: %" GST_TIME_FORMAT + ", next time: %" GST_TIME_FORMAT, + GST_TIME_ARGS (current_time), GST_TIME_ARGS (result)); RTP_SESSION_UNLOCK (sess); return result; @@ -2343,6 +2354,8 @@ typedef struct GstRTCPPacket packet; gboolean is_bye; gboolean has_sdes; + gboolean is_early; + gboolean may_suppress; } ReportData; static void @@ -2390,6 +2403,9 @@ session_report_blocks (const gchar * key, RTPSource * source, ReportData * data) /* create a new buffer if needed */ if (data->rtcp == NULL) { session_start_rtcp (sess, data); + } else if (data->is_early) { + /* Put a single RR or SR in minimal compound packets */ + return; } if (gst_rtcp_packet_get_rb_count (packet) < GST_RTCP_MAX_RB_COUNT) { /* only report about other sender sources */ @@ -2521,6 +2537,10 @@ session_sdes (RTPSession * sess, ReportData * data) continue; type = gst_rtcp_sdes_name_to_type (field); + /* Early packets are minimal and only include the CNAME */ + if (data->is_early && type != GST_RTCP_SDES_CNAME) + continue; + if (type > GST_RTCP_SDES_END && type < GST_RTCP_SDES_PRIV) { gst_rtcp_packet_sdes_add_entry (packet, type, strlen (value), (const guint8 *) value); @@ -2578,7 +2598,9 @@ static gboolean is_rtcp_time (RTPSession * sess, GstClockTime current_time, ReportData * data) { GstClockTime new_send_time, elapsed; - gboolean result; + + if (data->is_early && sess->next_early_rtcp_time < current_time) + goto early; /* no need to check yet */ if (sess->next_rtcp_check_time > current_time) { @@ -2603,18 +2625,40 @@ is_rtcp_time (RTPSession * sess, GstClockTime current_time, ReportData * data) if (current_time < new_send_time) { GST_DEBUG ("reconsider RTCP for %" GST_TIME_FORMAT, GST_TIME_ARGS (new_send_time)); - result = FALSE; /* store new check time */ sess->next_rtcp_check_time = new_send_time; - } else { - result = TRUE; - new_send_time = calculate_rtcp_interval (sess, FALSE, FALSE); - - GST_DEBUG ("can send RTCP now, next interval %" GST_TIME_FORMAT, - GST_TIME_ARGS (new_send_time)); - sess->next_rtcp_check_time = current_time + new_send_time; + return FALSE; } - return result; + +early: + + new_send_time = calculate_rtcp_interval (sess, FALSE, FALSE); + + GST_DEBUG ("can send RTCP now, next interval %" GST_TIME_FORMAT, + GST_TIME_ARGS (new_send_time)); + sess->next_rtcp_check_time = current_time + new_send_time; + + /* Apply the rules from RFC 4585 section 3.5.3 */ + if (sess->stats.min_interval != 0 && !sess->first_rtcp) { + GstClockTimeDiff T_rr_current_interval = g_random_double_range (0.5, 1.5) * + sess->stats.min_interval; + + /* This will caused the RTCP to be suppressed if no FB packets are added */ + if (sess->last_rtcp_send_time + T_rr_current_interval > + sess->next_rtcp_check_time) { + GST_DEBUG ("RTCP packet could be suppressed min: %" GST_TIME_FORMAT + " last: %" GST_TIME_FORMAT + " + T_rr_current_interval: %" GST_TIME_FORMAT + " > sess->next_rtcp_check_time: %" GST_TIME_FORMAT, + GST_TIME_ARGS (sess->stats.min_interval), + GST_TIME_ARGS (sess->last_rtcp_send_time), + GST_TIME_ARGS (T_rr_current_interval), + GST_TIME_ARGS (sess->next_rtcp_check_time)); + data->may_suppress = TRUE; + } + } + + return TRUE; } static void @@ -2668,6 +2712,7 @@ rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time, data.ntpnstime = ntpnstime; data.is_bye = FALSE; data.has_sdes = FALSE; + data.may_suppress = FALSE; data.running_time = running_time; own = sess->source; @@ -2692,6 +2737,11 @@ rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time, g_hash_table_foreach_remove (sess->ssrcs[sess->mask_idx], (GHRFunc) remove_closing_sources, NULL); + if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) + data.is_early = TRUE; + else + data.is_early = FALSE; + /* see if we need to generate SR or RR packets */ if (is_rtcp_time (sess, current_time, &data)) { if (own->received_bye) { @@ -2709,8 +2759,10 @@ rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time, if (data.rtcp) { /* we keep track of the last report time in order to timeout inactive * receivers or senders */ - sess->last_rtcp_send_time = data.current_time; + if (!data.is_early && !data.may_suppress) + sess->last_rtcp_send_time = data.current_time; sess->first_rtcp = FALSE; + sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE; /* add SDES for this source when not already added */ if (!data.has_sdes) @@ -2740,6 +2792,9 @@ rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time, notify = TRUE; GST_DEBUG ("changed our SSRC to %08x", own->ssrc); } + + sess->allow_early = TRUE; + RTP_SESSION_UNLOCK (sess); if (notify) @@ -2747,12 +2802,15 @@ rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time, /* push out the RTCP packet */ if (data.rtcp) { + gboolean do_not_suppress; + /* Give the user a change to add its own packet */ g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDING_RTCP], 0, - data.rtcp, FALSE, NULL); + data.rtcp, data.is_early, &do_not_suppress); - /* close the RTCP packet */ - gst_rtcp_buffer_end (data.rtcp); + if (sess->callbacks.send_rtcp && (do_not_suppress || !data.may_suppress)) { + /* close the RTCP packet */ + gst_rtcp_buffer_end (data.rtcp); if (sess->callbacks.send_rtcp) { guint packet_size; @@ -2766,10 +2824,73 @@ rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time, sess->callbacks.send_rtcp (sess, own, data.rtcp, sess->sent_bye, sess->send_rtcp_user_data); } else { - GST_DEBUG ("freeing packet"); + GST_DEBUG ("freeing packet callback: %p" + " do_not_suppress: %d may_suppress: %d", + sess->callbacks.send_rtcp, do_not_suppress, data.may_suppress); gst_buffer_unref (data.rtcp); } } return result; } + +void +rtp_session_request_early_rtcp (RTPSession * sess, GstClockTime current_time, + GstClockTimeDiff max_delay) +{ + GstClockTime T_dither_max; + + /* Implements the algorithm described in RFC 4585 section 3.5.2 */ + + RTP_SESSION_LOCK (sess); + + /* Check if already requested */ + /* RFC 4585 section 3.5.2 step 2 */ + if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) + goto dont_send; + + /* Ignore the request a scheduled packet will be in time anyway */ + if (current_time + max_delay > sess->next_rtcp_check_time) + goto dont_send; + + /* RFC 4585 section 3.5.2 step 2b */ + /* If the total sources is <=2, then there is only us and one peer */ + if (sess->total_sources <= 2) { + T_dither_max = 0; + } else { + /* Divide by 2 because l = 0.5 */ + T_dither_max = sess->next_rtcp_check_time - sess->last_rtcp_send_time; + T_dither_max /= 2; + } + + /* RFC 4585 section 3.5.2 step 3 */ + if (current_time + T_dither_max > sess->next_rtcp_check_time) + goto dont_send; + + /* RFC 4585 section 3.5.2 step 4 */ + if (sess->allow_early == FALSE) + goto dont_send; + + if (T_dither_max) { + /* Schedule an early transmission later */ + sess->next_early_rtcp_time = g_random_double () * T_dither_max + + current_time; + } else { + /* If no dithering, schedule it for NOW */ + sess->next_early_rtcp_time = current_time; + } + + RTP_SESSION_UNLOCK (sess); + + /* notify app of need to send packet early + * and therefore of timeout change */ + if (sess->callbacks.reconsider) + sess->callbacks.reconsider (sess, sess->reconsider_user_data); + + return; + +dont_send: + + RTP_SESSION_UNLOCK (sess); + +} diff --git a/gst/rtpmanager/rtpsession.h b/gst/rtpmanager/rtpsession.h index 0f86856cd7..a2d5d571b1 100644 --- a/gst/rtpmanager/rtpsession.h +++ b/gst/rtpmanager/rtpsession.h @@ -183,6 +183,9 @@ struct _RTPSession { GstClockTime next_rtcp_check_time; GstClockTime last_rtcp_send_time; gboolean first_rtcp; + gboolean allow_early; + + GstClockTime next_early_rtcp_time; gchar *bye_reason; gboolean sent_bye; @@ -298,4 +301,9 @@ GstClockTime rtp_session_next_timeout (RTPSession *sess, GstClockTi GstFlowReturn rtp_session_on_timeout (RTPSession *sess, GstClockTime current_time, guint64 ntpnstime, GstClockTime running_time); +/* request the transmittion of an early RTCP packet */ +void rtp_session_request_early_rtcp (RTPSession * sess, GstClockTime current_time, + GstClockTimeDiff max_delay); + + #endif /* __RTP_SESSION_H__ */ From f399b6a641faf8dea373bf39f17b774be51924cf Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Tue, 1 Feb 2011 18:17:13 +0100 Subject: [PATCH 273/351] rtpsession: fix compilation --- gst/rtpmanager/rtpsession.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c index 7d806eec04..a0ac833313 100644 --- a/gst/rtpmanager/rtpsession.c +++ b/gst/rtpmanager/rtpsession.c @@ -2809,12 +2809,11 @@ rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time, data.rtcp, data.is_early, &do_not_suppress); if (sess->callbacks.send_rtcp && (do_not_suppress || !data.may_suppress)) { + guint packet_size; + /* close the RTCP packet */ gst_rtcp_buffer_end (data.rtcp); - if (sess->callbacks.send_rtcp) { - guint packet_size; - packet_size = GST_BUFFER_SIZE (data.rtcp) + sess->header_len; UPDATE_AVG (sess->stats.avg_rtcp_packet_size, packet_size); From 1643f427db8c3de8cb8dd952eb408c076a4223ee Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Olivier=20Cr=C3=AAte?= Date: Thu, 17 Jun 2010 17:34:19 -0400 Subject: [PATCH 274/351] rtpsession: Emit signal on incoming RTCP FB packet --- gst/rtpmanager/gstrtpbin-marshal.list | 1 + gst/rtpmanager/rtpsession.c | 57 +++++++++++++++++++++++++++ gst/rtpmanager/rtpsession.h | 2 + 3 files changed, 60 insertions(+) diff --git a/gst/rtpmanager/gstrtpbin-marshal.list b/gst/rtpmanager/gstrtpbin-marshal.list index 0b0d81ff3a..5176b334d4 100644 --- a/gst/rtpmanager/gstrtpbin-marshal.list +++ b/gst/rtpmanager/gstrtpbin-marshal.list @@ -8,3 +8,4 @@ VOID:UINT,UINT VOID:OBJECT,OBJECT UINT64:BOOL,UINT64 BOOL:POINTER,BOOL +VOID:UINT,UINT,UINT,UINT,POINTER diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c index a0ac833313..d4bbf08796 100644 --- a/gst/rtpmanager/rtpsession.c +++ b/gst/rtpmanager/rtpsession.c @@ -43,6 +43,7 @@ enum SIGNAL_ON_TIMEOUT, SIGNAL_ON_SENDER_TIMEOUT, SIGNAL_ON_SENDING_RTCP, + SIGNAL_ON_FEEDBACK_RTCP, LAST_SIGNAL }; @@ -271,6 +272,27 @@ rtp_session_class_init (RTPSessionClass * klass) accumulate_trues, NULL, gst_rtp_bin_marshal_BOOLEAN__POINTER_BOOLEAN, G_TYPE_BOOLEAN, 2, G_TYPE_POINTER, G_TYPE_BOOLEAN); + /** + * RTPSession::on-feedback-rtcp: + * @session: the object which received the signal + * @type: Type of RTCP packet, will be %GST_RTCP_TYPE_RTPFB or + * %GST_RTCP_TYPE_RTPFB + * @fbtype: The type of RTCP FB packet, probably part of #GstRTCPFBType + * @sender_ssrc: The SSRC of the sender + * @media_ssrc: The SSRC of the media this refers to + * @fci: a #GstBuffer with the FCI data from the FB packet or %NULL if + * there was no FCI + * + * Notify that a RTCP feedback packet has been received + */ + + rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP] = + g_signal_new ("on-feedback-rtcp", G_TYPE_FROM_CLASS (klass), + G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_feedback_rtcp), + NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT_UINT_UINT_POINTER, + G_TYPE_NONE, 4, G_TYPE_UINT, G_TYPE_UINT, G_TYPE_UINT, G_TYPE_UINT, + G_TYPE_POINTER); + g_object_class_install_property (gobject_class, PROP_INTERNAL_SSRC, g_param_spec_uint ("internal-ssrc", "Internal SSRC", "The internal SSRC used for the session", @@ -1960,6 +1982,37 @@ rtp_session_process_app (RTPSession * sess, GstRTCPPacket * packet, GST_DEBUG ("received APP"); } + +static void +rtp_session_process_feedback (RTPSession * sess, GstRTCPPacket * packet, + RTPArrivalStats * arrival) +{ + GstRTCPType type = gst_rtcp_packet_get_type (packet); + GstRTCPFBType fbtype = gst_rtcp_packet_fb_get_type (packet); + guint32 sender_ssrc = gst_rtcp_packet_fb_get_sender_ssrc (packet); + guint32 media_ssrc = gst_rtcp_packet_fb_get_media_ssrc (packet); + guint length = 4 * (gst_rtcp_packet_get_length (packet) - 2); + + GST_DEBUG ("received feedback %d:%d from %08X about %08X" + " with FCI of length %d", type, fbtype, sender_ssrc, media_ssrc, length); + + if (g_signal_has_handler_pending (sess, + rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0, TRUE)) { + GstBuffer *fci = NULL; + + if (length) { + fci = gst_buffer_create_sub (packet->buffer, packet->offset + 72, length); + GST_BUFFER_TIMESTAMP (fci) = arrival->running_time; + } + + g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0, + type, fbtype, sender_ssrc, media_ssrc, fci); + + if (fci) + gst_buffer_unref (fci); + } +} + /** * rtp_session_process_rtcp: * @sess: and #RTPSession @@ -2030,6 +2083,10 @@ rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer, case GST_RTCP_TYPE_APP: rtp_session_process_app (sess, &packet, &arrival); break; + case GST_RTCP_TYPE_RTPFB: + case GST_RTCP_TYPE_PSFB: + rtp_session_process_feedback (sess, &packet, &arrival); + break; default: GST_WARNING ("got unknown RTCP packet"); break; diff --git a/gst/rtpmanager/rtpsession.h b/gst/rtpmanager/rtpsession.h index a2d5d571b1..b63bbcc0a2 100644 --- a/gst/rtpmanager/rtpsession.h +++ b/gst/rtpmanager/rtpsession.h @@ -229,6 +229,8 @@ struct _RTPSessionClass { void (*on_sender_timeout) (RTPSession *sess, RTPSource *source); gboolean (*on_sending_rtcp) (RTPSession *sess, GstBuffer *buffer, gboolean early); + void (*on_feedback_rtcp) (RTPSession *sess, guint type, guint fbtype, + guint sender_ssrc, guint media_ssrc, GstBuffer *fci); }; GType rtp_session_get_type (void); From 90354ecb49096408ba6e21b9e050f4c61c9ddcb0 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Olivier=20Cr=C3=AAte?= Date: Tue, 7 Sep 2010 13:35:16 +0300 Subject: [PATCH 275/351] rtpsession: Make rtcp buffer metadata writable after processing it Functions that process the rtcp buffer could decide to keep a ref on the buffer for further processing. So make the metadata writable only after they are done. --- gst/rtpmanager/rtpsession.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c index d4bbf08796..5938d55802 100644 --- a/gst/rtpmanager/rtpsession.c +++ b/gst/rtpmanager/rtpsession.c @@ -2048,9 +2048,6 @@ rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer, if (sess->sent_bye) goto ignore; - /* make writable, we might want to change the buffer */ - buffer = gst_buffer_make_metadata_writable (buffer); - /* start processing the compound packet */ more = gst_rtcp_buffer_get_first_packet (buffer, &packet); while (more) { @@ -2111,10 +2108,13 @@ rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer, RTP_SESSION_UNLOCK (sess); /* notify caller of sr packets in the callback */ - if (do_sync && sess->callbacks.sync_rtcp) + if (do_sync && sess->callbacks.sync_rtcp) { + /* make writable, we might want to change the buffer */ + buffer = gst_buffer_make_metadata_writable (buffer); + result = sess->callbacks.sync_rtcp (sess, sess->source, buffer, sess->sync_rtcp_user_data); - else + } else gst_buffer_unref (buffer); return result; From db5150a23a7eb99033a89045be4a3d3f59b201f1 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Olivier=20Cr=C3=AAte?= Date: Tue, 22 Jun 2010 13:33:32 -0400 Subject: [PATCH 276/351] rtpsource: Retain RTCP Feedback packets for a specified amount of time --- gst/rtpmanager/rtpsession.c | 25 ++++++++++++++++++- gst/rtpmanager/rtpsession.h | 1 + gst/rtpmanager/rtpsource.c | 49 ++++++++++++++++++++++++++++++++++++- gst/rtpmanager/rtpsource.h | 13 +++++++++- 4 files changed, 85 insertions(+), 3 deletions(-) diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c index 5938d55802..ec40115756 100644 --- a/gst/rtpmanager/rtpsession.c +++ b/gst/rtpmanager/rtpsession.c @@ -58,6 +58,7 @@ enum #define DEFAULT_NUM_ACTIVE_SOURCES 0 #define DEFAULT_SOURCES NULL #define DEFAULT_RTCP_MIN_INTERVAL (RTP_STATS_MIN_INTERVAL * GST_SECOND) +#define DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW (2 * GST_SECOND) enum { @@ -75,6 +76,7 @@ enum PROP_SOURCES, PROP_FAVOR_NEW, PROP_RTCP_MIN_INTERVAL, + PROP_RTCP_FEEDBACK_RETENTION_WINDOW, PROP_LAST }; @@ -390,6 +392,15 @@ rtp_session_class_init (RTPSessionClass * klass) 0, G_MAXUINT64, DEFAULT_RTCP_MIN_INTERVAL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + g_object_class_install_property (gobject_class, + PROP_RTCP_FEEDBACK_RETENTION_WINDOW, + g_param_spec_uint64 ("rtcp-feedback-retention-window", + "RTCP Feedback retention window", + "Duration during which RTCP Feedback packets are retained (in ns)", + 0, G_MAXUINT64, DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW, + G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + + klass->get_source_by_ssrc = GST_DEBUG_FUNCPTR (rtp_session_get_source_by_ssrc); @@ -444,6 +455,7 @@ rtp_session_init (RTPSession * sess) sess->first_rtcp = TRUE; sess->allow_early = TRUE; + sess->rtcp_feedback_retention_window = DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW; GST_DEBUG ("%p: session using SSRC: %08x", sess, sess->source->ssrc); } @@ -2005,12 +2017,22 @@ rtp_session_process_feedback (RTPSession * sess, GstRTCPPacket * packet, GST_BUFFER_TIMESTAMP (fci) = arrival->running_time; } + RTP_SESSION_UNLOCK (sess); g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0, type, fbtype, sender_ssrc, media_ssrc, fci); + RTP_SESSION_LOCK (sess); if (fci) gst_buffer_unref (fci); } + + if (sess->rtcp_feedback_retention_window) { + RTPSource *src = g_hash_table_lookup (sess->ssrcs[sess->mask_idx], + GINT_TO_POINTER (media_ssrc)); + + if (src) + rtp_source_retain_rtcp_packet (src, packet, arrival->running_time); + } } /** @@ -2829,7 +2851,8 @@ rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time, /* check for outdated collisions */ GST_DEBUG ("Timing out collisions"); rtp_source_timeout (sess->source, current_time, - data.interval * RTCP_INTERVAL_COLLISION_TIMEOUT); + data.interval * RTCP_INTERVAL_COLLISION_TIMEOUT, + running_time - sess->rtcp_feedback_retention_window); if (sess->change_ssrc) { GST_DEBUG ("need to change our SSRC (%08x)", own->ssrc); diff --git a/gst/rtpmanager/rtpsession.h b/gst/rtpmanager/rtpsession.h index b63bbcc0a2..6ae318e1f5 100644 --- a/gst/rtpmanager/rtpsession.h +++ b/gst/rtpmanager/rtpsession.h @@ -202,6 +202,7 @@ struct _RTPSession { gboolean change_ssrc; gboolean favor_new; + GstClockTime rtcp_feedback_retention_window; }; /** diff --git a/gst/rtpmanager/rtpsource.c b/gst/rtpmanager/rtpsource.c index 4a3af168a7..423d2b65d2 100644 --- a/gst/rtpmanager/rtpsource.c +++ b/gst/rtpmanager/rtpsource.c @@ -238,6 +238,8 @@ rtp_source_init (RTPSource * src) src->seqnum_base = -1; src->last_rtptime = -1; + src->retained_feedback = g_queue_new (); + rtp_source_reset (src); } @@ -262,6 +264,10 @@ rtp_source_finalize (GObject * object) g_list_foreach (src->conflicting_addresses, (GFunc) g_free, NULL); g_list_free (src->conflicting_addresses); + while ((buffer = g_queue_pop_head (src->retained_feedback))) + gst_buffer_unref (buffer); + g_queue_free (src->retained_feedback); + G_OBJECT_CLASS (rtp_source_parent_class)->finalize (object); } @@ -1719,14 +1725,18 @@ rtp_source_add_conflicting_address (RTPSource * src, * @src: The #RTPSource * @current_time: The current time * @collision_timeout: The amount of time after which a collision is timed out + * @feedback_retention_window: The running time before which retained feedback + * packets have to be discarded * * This is processed on each RTCP interval. It times out old collisions. + * It also times out old retained feedback packets */ void rtp_source_timeout (RTPSource * src, GstClockTime current_time, - GstClockTime collision_timeout) + GstClockTime collision_timeout, GstClockTime feedback_retention_window) { GList *item; + GstRTCPPacket *pkt; item = g_list_first (src->conflicting_addresses); while (item) { @@ -1744,4 +1754,41 @@ rtp_source_timeout (RTPSource * src, GstClockTime current_time, } item = next_item; } + + /* Time out AVPF packets that are older than the desired length */ + while ((pkt = g_queue_peek_tail (src->retained_feedback)) && + GST_BUFFER_TIMESTAMP (pkt) < feedback_retention_window) + gst_buffer_unref (g_queue_pop_tail (src->retained_feedback)); +} + +static gint +compare_buffers (gconstpointer a, gconstpointer b, gpointer user_data) +{ + const GstBuffer *bufa = a; + const GstBuffer *bufb = b; + + return GST_BUFFER_TIMESTAMP (bufa) - GST_BUFFER_TIMESTAMP (bufb); +} + +void +rtp_source_retain_rtcp_packet (RTPSource * src, GstRTCPPacket * packet, + GstClockTime running_time) +{ + GstBuffer *buffer; + + buffer = gst_buffer_create_sub (packet->buffer, packet->offset, + (gst_rtcp_packet_get_length (packet) + 1) * 4); + + GST_BUFFER_TIMESTAMP (buffer) = running_time; + + g_queue_insert_sorted (src->retained_feedback, buffer, compare_buffers, NULL); +} + +gboolean +rtp_source_has_retained (RTPSource * src, GCompareFunc func, gconstpointer data) +{ + if (g_queue_find_custom (src->retained_feedback, data, func)) + return TRUE; + else + return FALSE; } diff --git a/gst/rtpmanager/rtpsource.h b/gst/rtpmanager/rtpsource.h index 94c22b3f5a..fadbe3068a 100644 --- a/gst/rtpmanager/rtpsource.h +++ b/gst/rtpmanager/rtpsource.h @@ -170,6 +170,8 @@ struct _RTPSource { RTPReceiverReport last_rr; GList *conflicting_addresses; + + GQueue *retained_feedback; }; struct _RTPSourceClass { @@ -248,7 +250,16 @@ void rtp_source_add_conflicting_address (RTPSource * src, void rtp_source_timeout (RTPSource * src, GstClockTime current_time, - GstClockTime collision_timeout); + GstClockTime collision_timeout, + GstClockTime feedback_retention_window); + +void rtp_source_retain_rtcp_packet (RTPSource * src, + GstRTCPPacket *pkt, + GstClockTime running_time); + +gboolean rtp_source_has_retained (RTPSource * src, + GCompareFunc func, + gconstpointer data); #endif /* __RTP_SOURCE_H__ */ From 52f95fa7ee441e842dc6ef7fb5cd28bfa1178866 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Olivier=20Cr=C3=AAte?= Date: Tue, 22 Jun 2010 19:56:50 -0400 Subject: [PATCH 277/351] rtpsession: Implement sending PLI packets in response to GstForceKeyUnit --- gst/rtpmanager/gstrtpsession.c | 73 ++++++++++++++++++++++++++++++++ gst/rtpmanager/rtpsession.c | 77 ++++++++++++++++++++++++++++++++++ gst/rtpmanager/rtpsession.h | 6 +++ 3 files changed, 156 insertions(+) diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c index f10ab92e65..6d36633038 100644 --- a/gst/rtpmanager/gstrtpsession.c +++ b/gst/rtpmanager/gstrtpsession.c @@ -1370,6 +1370,77 @@ gst_rtp_session_event_recv_rtp_sink (GstPad * pad, GstEvent * event) } +static gboolean +gst_rtp_session_request_remote_key_unit (GstRtpSession * rtpsession, + guint32 ssrc, guint payload, gboolean all_headers) +{ + GstCaps *caps; + gboolean requested = FALSE; + + caps = gst_rtp_session_get_caps_for_pt (rtpsession, payload); + + if (caps) { + gboolean fir, pli; + const GstStructure *s = gst_caps_get_structure (caps, 0); + + if (all_headers && + gst_structure_get_boolean (s, "rtcp-fb-nack-fir", &fir) && fir) { + /* 500 ms acceptable delay for FIR request is a guesstimate, it could + * be made configurable if needed + */ + rtp_session_request_early_rtcp (rtpsession->priv->session, + gst_clock_get_time (rtpsession->priv->sysclock), 500 * GST_MSECOND); + rtp_session_request_key_unit (rtpsession->priv->session, ssrc, TRUE); + requested = TRUE; + } else if (gst_structure_get_boolean (s, "rtcp-fb-nack-pli", &pli) && pli) { + rtp_session_request_key_unit (rtpsession->priv->session, ssrc, FALSE); + requested = TRUE; + } + gst_caps_unref (caps); + } + + return requested; +} + +static gboolean +gst_rtp_session_event_recv_rtp_src (GstPad * pad, GstEvent * event) +{ + GstRtpSession *rtpsession; + gboolean forward = TRUE; + gboolean ret = TRUE; + const GstStructure *s; + guint32 ssrc; + guint pt; + + rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); + + switch (GST_EVENT_TYPE (event)) { + case GST_EVENT_CUSTOM_UPSTREAM: + s = gst_event_get_structure (event); + if (gst_structure_has_name (s, "GstForceKeyUnit") && + gst_structure_get_uint (s, "ssrc", &ssrc) && + gst_structure_get_uint (s, "payload", &pt)) { + gboolean all_headers = FALSE; + + gst_structure_get_boolean (s, "all-headers", &all_headers); + if (gst_rtp_session_request_remote_key_unit (rtpsession, ssrc, pt, + all_headers)) + forward = FALSE; + } + break; + default: + break; + } + + if (forward) + ret = gst_pad_push_event (rtpsession->recv_rtp_sink, event); + + gst_object_unref (rtpsession); + + return ret; +} + + static GstIterator * gst_rtp_session_iterate_internal_links (GstPad * pad) { @@ -1780,6 +1851,8 @@ create_recv_rtp_sink (GstRtpSession * rtpsession) rtpsession->recv_rtp_src = gst_pad_new_from_static_template (&rtpsession_recv_rtp_src_template, "recv_rtp_src"); + gst_pad_set_event_function (rtpsession->recv_rtp_src, + (GstPadEventFunction) gst_rtp_session_event_recv_rtp_src); gst_pad_set_iterate_internal_links_function (rtpsession->recv_rtp_src, gst_rtp_session_iterate_internal_links); gst_pad_use_fixed_caps (rtpsession->recv_rtp_src); diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c index ec40115756..32973c0b65 100644 --- a/gst/rtpmanager/rtpsession.c +++ b/gst/rtpmanager/rtpsession.c @@ -102,6 +102,10 @@ static void rtp_session_set_property (GObject * object, guint prop_id, static void rtp_session_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); +static gboolean rtp_session_on_sending_rtcp (RTPSession * sess, + GstBuffer * buffer, gboolean early); + + static guint rtp_session_signals[LAST_SIGNAL] = { 0 }; G_DEFINE_TYPE (RTPSession, rtp_session, G_TYPE_OBJECT); @@ -403,6 +407,7 @@ rtp_session_class_init (RTPSessionClass * klass) klass->get_source_by_ssrc = GST_DEBUG_FUNCPTR (rtp_session_get_source_by_ssrc); + klass->on_sending_rtcp = GST_DEBUG_FUNCPTR (rtp_session_on_sending_rtcp); GST_DEBUG_CATEGORY_INIT (rtp_session_debug, "rtpsession", 0, "RTP Session"); } @@ -457,6 +462,8 @@ rtp_session_init (RTPSession * sess) sess->allow_early = TRUE; sess->rtcp_feedback_retention_window = DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW; + sess->rtcp_pli_requests = g_array_new (FALSE, FALSE, sizeof (guint32)); + GST_DEBUG ("%p: session using SSRC: %08x", sess, sess->source->ssrc); } @@ -477,6 +484,8 @@ rtp_session_finalize (GObject * object) g_hash_table_destroy (sess->cnames); g_object_unref (sess->source); + g_array_free (sess->rtcp_pli_requests, TRUE); + G_OBJECT_CLASS (rtp_session_parent_class)->finalize (object); } @@ -2973,3 +2982,71 @@ dont_send: RTP_SESSION_UNLOCK (sess); } + +void +rtp_session_request_key_unit (RTPSession * sess, guint32 ssrc, gboolean fir) +{ + guint i; + + if (fir) + return; + + for (i = 0; i < sess->rtcp_pli_requests->len; i++) + if (ssrc == g_array_index (sess->rtcp_pli_requests, guint32, i)) + return; + + g_array_append_val (sess->rtcp_pli_requests, ssrc); +} + +static gboolean +has_pli_compare_func (gconstpointer a, gconstpointer ignored) +{ + GstRTCPPacket packet; + + packet.buffer = (GstBuffer *) a; + packet.offset = 0; + + if (gst_rtcp_packet_get_type (&packet) == GST_RTCP_TYPE_PSFB && + gst_rtcp_packet_fb_get_type (&packet) == GST_RTCP_PSFB_TYPE_PLI) + return TRUE; + else + return FALSE; +} + +static gboolean +rtp_session_on_sending_rtcp (RTPSession * sess, GstBuffer * buffer, + gboolean early) +{ + gboolean ret = FALSE; + + RTP_SESSION_LOCK (sess); + + while (sess->rtcp_pli_requests->len) { + GstRTCPPacket rtcppacket; + guint media_ssrc = g_array_index (sess->rtcp_pli_requests, guint32, 0); + RTPSource *media_src = g_hash_table_lookup (sess->ssrcs[sess->mask_idx], + GUINT_TO_POINTER (media_ssrc)); + + if (media_src && !rtp_source_has_retained (media_src, + has_pli_compare_func, NULL)) { + if (gst_rtcp_buffer_add_packet (buffer, GST_RTCP_TYPE_PSFB, &rtcppacket)) { + gst_rtcp_packet_fb_set_type (&rtcppacket, GST_RTCP_PSFB_TYPE_PLI); + gst_rtcp_packet_fb_set_sender_ssrc (&rtcppacket, + rtp_source_get_ssrc (sess->source)); + gst_rtcp_packet_fb_set_media_ssrc (&rtcppacket, media_ssrc); + ret = TRUE; + } else { + /* Break because the packet is full, will put next request in a + * further packet + */ + break; + } + } + + g_array_remove_index (sess->rtcp_pli_requests, 0); + } + + RTP_SESSION_UNLOCK (sess); + + return ret; +} diff --git a/gst/rtpmanager/rtpsession.h b/gst/rtpmanager/rtpsession.h index 6ae318e1f5..3113247ce5 100644 --- a/gst/rtpmanager/rtpsession.h +++ b/gst/rtpmanager/rtpsession.h @@ -203,6 +203,8 @@ struct _RTPSession { gboolean change_ssrc; gboolean favor_new; GstClockTime rtcp_feedback_retention_window; + + GArray *rtcp_pli_requests; }; /** @@ -308,5 +310,9 @@ GstFlowReturn rtp_session_on_timeout (RTPSession *sess, GstClockTi void rtp_session_request_early_rtcp (RTPSession * sess, GstClockTime current_time, GstClockTimeDiff max_delay); +/* Notify session of a request for a new key unit */ +void rtp_session_request_key_unit (RTPSession * sess, + guint32 ssrc, + gboolean fir); #endif /* __RTP_SESSION_H__ */ From 7350d2adfad6d6d3bef63fb793739204048612c5 Mon Sep 17 00:00:00 2001 From: Sjoerd Simons Date: Wed, 24 Nov 2010 15:27:46 -0500 Subject: [PATCH 278/351] gstrtpsession: Fallback for FIR to PLI if PLI isn't available --- gst/rtpmanager/gstrtpsession.c | 25 +++++++++++++++++++------ 1 file changed, 19 insertions(+), 6 deletions(-) diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c index 6d36633038..cc784c83e9 100644 --- a/gst/rtpmanager/gstrtpsession.c +++ b/gst/rtpmanager/gstrtpsession.c @@ -1383,22 +1383,35 @@ gst_rtp_session_request_remote_key_unit (GstRtpSession * rtpsession, gboolean fir, pli; const GstStructure *s = gst_caps_get_structure (caps, 0); - if (all_headers && - gst_structure_get_boolean (s, "rtcp-fb-nack-fir", &fir) && fir) { - /* 500 ms acceptable delay for FIR request is a guesstimate, it could + if (!gst_structure_get_boolean (s, "rtcp-fb-nack-fir", &fir)) + fir = FALSE; + + if (!gst_structure_get_boolean (s, "rtcp-fb-nack-pli", &pli)) + pli = FALSE; + + gst_caps_unref (caps); + + if (!pli && !fir) + goto out; + + /* When we need all headers, use FIR if possible falling back to PLI if + * it's available */ + if (all_headers) { + /* 500 ms acceptable delay for urgent request is a guesstimate, it could * be made configurable if needed */ + /* If we don't have fir, fall back to pli */ + rtp_session_request_key_unit (rtpsession->priv->session, ssrc, fir); rtp_session_request_early_rtcp (rtpsession->priv->session, gst_clock_get_time (rtpsession->priv->sysclock), 500 * GST_MSECOND); - rtp_session_request_key_unit (rtpsession->priv->session, ssrc, TRUE); requested = TRUE; - } else if (gst_structure_get_boolean (s, "rtcp-fb-nack-pli", &pli) && pli) { + } else if (pli) { rtp_session_request_key_unit (rtpsession->priv->session, ssrc, FALSE); requested = TRUE; } - gst_caps_unref (caps); } +out: return requested; } From a61bb9e94b23f6dbed0319d6f018a8d857b2edcb Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Olivier=20Cr=C3=AAte?= Date: Wed, 23 Jun 2010 16:43:24 -0400 Subject: [PATCH 279/351] rtpsession: Send GstForceKeyUnit event in response to received RTCP PLI --- gst/rtpmanager/gstrtpsession.c | 18 +++++++++++++++++- gst/rtpmanager/rtpsession.c | 23 +++++++++++++++++++++++ gst/rtpmanager/rtpsession.h | 16 ++++++++++++++++ 3 files changed, 56 insertions(+), 1 deletion(-) diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c index cc784c83e9..994c53ba8a 100644 --- a/gst/rtpmanager/gstrtpsession.c +++ b/gst/rtpmanager/gstrtpsession.c @@ -257,6 +257,8 @@ static GstFlowReturn gst_rtp_session_sync_rtcp (RTPSession * sess, static gint gst_rtp_session_clock_rate (RTPSession * sess, guint8 payload, gpointer user_data); static void gst_rtp_session_reconsider (RTPSession * sess, gpointer user_data); +static void gst_rtp_session_request_key_unit (RTPSession * sess, + gboolean all_headers, gpointer user_data); static RTPSessionCallbacks callbacks = { gst_rtp_session_process_rtp, @@ -264,7 +266,8 @@ static RTPSessionCallbacks callbacks = { gst_rtp_session_sync_rtcp, gst_rtp_session_send_rtcp, gst_rtp_session_clock_rate, - gst_rtp_session_reconsider + gst_rtp_session_reconsider, + gst_rtp_session_request_key_unit }; /* GObject vmethods */ @@ -2143,3 +2146,16 @@ wrong_pad: return; } } + +static void +gst_rtp_session_request_key_unit (RTPSession * sess, + gboolean all_headers, gpointer user_data) +{ + GstRtpSession *rtpsession = GST_RTP_SESSION (user_data); + GstEvent *event; + + event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, + gst_structure_new ("GstForceKeyUnit", + "all-headers", G_TYPE_BOOLEAN, all_headers, NULL)); + gst_pad_push_event (rtpsession->send_rtp_sink, event); +} diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c index 32973c0b65..d3a7a548f1 100644 --- a/gst/rtpmanager/rtpsession.c +++ b/gst/rtpmanager/rtpsession.c @@ -769,6 +769,10 @@ rtp_session_set_callbacks (RTPSession * sess, RTPSessionCallbacks * callbacks, sess->callbacks.reconsider = callbacks->reconsider; sess->reconsider_user_data = user_data; } + if (callbacks->request_key_unit) { + sess->callbacks.request_key_unit = callbacks->request_key_unit; + sess->request_key_unit_user_data = user_data; + } } /** @@ -2042,6 +2046,25 @@ rtp_session_process_feedback (RTPSession * sess, GstRTCPPacket * packet, if (src) rtp_source_retain_rtcp_packet (src, packet, arrival->running_time); } + + if (rtp_source_get_ssrc (sess->source) == media_ssrc) { + switch (type) { + case GST_RTCP_TYPE_PSFB: + switch (fbtype) { + case GST_RTCP_PSFB_TYPE_PLI: + if (sess->callbacks.request_key_unit) + sess->callbacks.request_key_unit (sess, FALSE, + sess->request_key_unit_user_data); + break; + default: + break; + } + break; + case GST_RTCP_TYPE_RTPFB: + default: + break; + } + } } /** diff --git a/gst/rtpmanager/rtpsession.h b/gst/rtpmanager/rtpsession.h index 3113247ce5..0cc1f6bfda 100644 --- a/gst/rtpmanager/rtpsession.h +++ b/gst/rtpmanager/rtpsession.h @@ -120,6 +120,19 @@ typedef gint (*RTPSessionClockRate) (RTPSession *sess, guint8 payload, gpointer */ typedef void (*RTPSessionReconsider) (RTPSession *sess, gpointer user_data); +/** + * RTPSessionRequestKeyUnit: + * @sess: an #RTPSession + * @all_headers: %TRUE if "all-headers" property should be set on the key unit + * request + * @user_data: user data specified when registering +* + * Asks the encoder to produce a key unit as soon as possibly within the + * bandwidth constraints + */ +typedef void (*RTPSessionRequestKeyUnit) (RTPSession *sess, + gboolean all_headers, gpointer user_data); + /** * RTPSessionCallbacks: * @RTPSessionProcessRTP: callback to process RTP packets @@ -127,6 +140,7 @@ typedef void (*RTPSessionReconsider) (RTPSession *sess, gpointer user_data); * @RTPSessionSendRTCP: callback for sending RTCP packets * @RTPSessionSyncRTCP: callback for handling SR packets * @RTPSessionReconsider: callback for reconsidering the timeout + * @RTPSessionRequestKeyUnit: callback for requesting a new key unit * * These callbacks can be installed on the session manager to get notification * when RTP and RTCP packets are ready for further processing. These callbacks @@ -139,6 +153,7 @@ typedef struct { RTPSessionSendRTCP send_rtcp; RTPSessionClockRate clock_rate; RTPSessionReconsider reconsider; + RTPSessionRequestKeyUnit request_key_unit; } RTPSessionCallbacks; /** @@ -197,6 +212,7 @@ struct _RTPSession { gpointer sync_rtcp_user_data; gpointer clock_rate_user_data; gpointer reconsider_user_data; + gpointer request_key_unit_user_data; RTPSessionStats stats; From a630c68fc312d4ec79ca514f8770802da3c8ce26 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Olivier=20Cr=C3=AAte?= Date: Tue, 19 Oct 2010 22:21:54 +0200 Subject: [PATCH 280/351] rtpsession: Don't relay more than one PLI request per RTT Drop PLI requests if one was relay in the last RTT, the other side may just not have received the keyframe yet. --- gst/rtpmanager/rtpsession.c | 49 +++++++++++++++++++++++++++++++++---- gst/rtpmanager/rtpsession.h | 1 + 2 files changed, 45 insertions(+), 5 deletions(-) diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c index d3a7a548f1..cf73043f60 100644 --- a/gst/rtpmanager/rtpsession.c +++ b/gst/rtpmanager/rtpsession.c @@ -2007,10 +2007,50 @@ rtp_session_process_app (RTPSession * sess, GstRTCPPacket * packet, GST_DEBUG ("received APP"); } +static void +rtp_session_process_pli (RTPSession * sess, guint32 sender_ssrc, + guint32 media_ssrc, GstClockTime current_time) +{ + RTPSource *src; + guint32 round_trip = 0; + + if (!sess->callbacks.request_key_unit) + return; + + src = g_hash_table_lookup (sess->ssrcs[sess->mask_idx], + GINT_TO_POINTER (sender_ssrc)); + + if (!src) + return; + + if (sess->last_keyframe_request != GST_CLOCK_TIME_NONE && + rtp_source_get_last_rb (src, NULL, NULL, NULL, NULL, NULL, NULL, + &round_trip)) { + GstClockTime round_trip_in_ns = gst_util_uint64_scale (round_trip, + GST_SECOND, 65536); + + if (sess->last_keyframe_request != GST_CLOCK_TIME_NONE && + current_time - sess->last_keyframe_request < round_trip_in_ns) { + GST_DEBUG ("Ignoring PLI because one was send without one RTT (%" + GST_TIME_FORMAT " < %" GST_TIME_FORMAT ")", + GST_TIME_ARGS (current_time - sess->last_keyframe_request), + GST_TIME_ARGS (round_trip_in_ns));; + return; + } + } + + sess->last_keyframe_request = current_time; + + GST_LOG ("received PLI from %X %p(%p)", sender_ssrc, + sess->callbacks.process_rtp, sess->callbacks.request_key_unit); + + sess->callbacks.request_key_unit (sess, FALSE, + sess->request_key_unit_user_data); +} static void rtp_session_process_feedback (RTPSession * sess, GstRTCPPacket * packet, - RTPArrivalStats * arrival) + RTPArrivalStats * arrival, GstClockTime current_time) { GstRTCPType type = gst_rtcp_packet_get_type (packet); GstRTCPFBType fbtype = gst_rtcp_packet_fb_get_type (packet); @@ -2052,9 +2092,8 @@ rtp_session_process_feedback (RTPSession * sess, GstRTCPPacket * packet, case GST_RTCP_TYPE_PSFB: switch (fbtype) { case GST_RTCP_PSFB_TYPE_PLI: - if (sess->callbacks.request_key_unit) - sess->callbacks.request_key_unit (sess, FALSE, - sess->request_key_unit_user_data); + rtp_session_process_pli (sess, sender_ssrc, media_ssrc, + current_time); break; default: break; @@ -2136,7 +2175,7 @@ rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer, break; case GST_RTCP_TYPE_RTPFB: case GST_RTCP_TYPE_PSFB: - rtp_session_process_feedback (sess, &packet, &arrival); + rtp_session_process_feedback (sess, &packet, &arrival, current_time); break; default: GST_WARNING ("got unknown RTCP packet"); diff --git a/gst/rtpmanager/rtpsession.h b/gst/rtpmanager/rtpsession.h index 0cc1f6bfda..95109425d8 100644 --- a/gst/rtpmanager/rtpsession.h +++ b/gst/rtpmanager/rtpsession.h @@ -221,6 +221,7 @@ struct _RTPSession { GstClockTime rtcp_feedback_retention_window; GArray *rtcp_pli_requests; + GstClockTime last_keyframe_request; }; /** From c0996e6b903e1deaf4ef9ee08da9288bc75fac57 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Olivier=20Cr=C3=AAte?= Date: Fri, 27 Aug 2010 16:11:06 -0400 Subject: [PATCH 281/351] rtpsession: Add callback to get the current time --- gst/rtpmanager/gstrtpsession.c | 13 ++++++++++++- gst/rtpmanager/rtpsession.c | 22 ++++++++++++++++++++++ gst/rtpmanager/rtpsession.h | 17 +++++++++++++++++ 3 files changed, 51 insertions(+), 1 deletion(-) diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c index 994c53ba8a..b1223d6585 100644 --- a/gst/rtpmanager/gstrtpsession.c +++ b/gst/rtpmanager/gstrtpsession.c @@ -259,6 +259,8 @@ static gint gst_rtp_session_clock_rate (RTPSession * sess, guint8 payload, static void gst_rtp_session_reconsider (RTPSession * sess, gpointer user_data); static void gst_rtp_session_request_key_unit (RTPSession * sess, gboolean all_headers, gpointer user_data); +static GstClockTime gst_rtp_session_request_time (RTPSession * session, + gpointer user_data); static RTPSessionCallbacks callbacks = { gst_rtp_session_process_rtp, @@ -267,7 +269,8 @@ static RTPSessionCallbacks callbacks = { gst_rtp_session_send_rtcp, gst_rtp_session_clock_rate, gst_rtp_session_reconsider, - gst_rtp_session_request_key_unit + gst_rtp_session_request_key_unit, + gst_rtp_session_request_time }; /* GObject vmethods */ @@ -2159,3 +2162,11 @@ gst_rtp_session_request_key_unit (RTPSession * sess, "all-headers", G_TYPE_BOOLEAN, all_headers, NULL)); gst_pad_push_event (rtpsession->send_rtp_sink, event); } + +static GstClockTime +gst_rtp_session_request_time (RTPSession * session, gpointer user_data) +{ + GstRtpSession *rtpsession = GST_RTP_SESSION (user_data); + + return gst_clock_get_time (rtpsession->priv->sysclock); +} diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c index cf73043f60..a60e7c0945 100644 --- a/gst/rtpmanager/rtpsession.c +++ b/gst/rtpmanager/rtpsession.c @@ -773,6 +773,10 @@ rtp_session_set_callbacks (RTPSession * sess, RTPSessionCallbacks * callbacks, sess->callbacks.request_key_unit = callbacks->request_key_unit; sess->request_key_unit_user_data = user_data; } + if (callbacks->request_time) { + sess->callbacks.request_time = callbacks->request_time; + sess->request_time_user_data = user_data; + } } /** @@ -883,6 +887,24 @@ rtp_session_set_reconsider_callback (RTPSession * sess, sess->reconsider_user_data = user_data; } +/** + * rtp_session_set_request_time_callback: + * @sess: an #RTPSession + * @callback: callback to set + * @user_data: user data passed in the callback + * + * Configure only the request_time callback + */ +void +rtp_session_set_request_time_callback (RTPSession * sess, + RTPSessionRequestTime callback, gpointer user_data) +{ + g_return_if_fail (RTP_IS_SESSION (sess)); + + sess->callbacks.request_time = callback; + sess->request_time_user_data = user_data; +} + /** * rtp_session_set_bandwidth: * @sess: an #RTPSession diff --git a/gst/rtpmanager/rtpsession.h b/gst/rtpmanager/rtpsession.h index 95109425d8..3746a9b721 100644 --- a/gst/rtpmanager/rtpsession.h +++ b/gst/rtpmanager/rtpsession.h @@ -133,6 +133,17 @@ typedef void (*RTPSessionReconsider) (RTPSession *sess, gpointer user_data); typedef void (*RTPSessionRequestKeyUnit) (RTPSession *sess, gboolean all_headers, gpointer user_data); +/** + * RTPSessionRequestTime: + * @sess: an #RTPSession + * @user_data: user data specified when registering + * + * This callback will be called when @sess needs the current time. The time + * should be returned as a #GstClockTime + */ +typedef GstClockTime (*RTPSessionRequestTime) (RTPSession *sess, + gpointer user_data); + /** * RTPSessionCallbacks: * @RTPSessionProcessRTP: callback to process RTP packets @@ -154,6 +165,7 @@ typedef struct { RTPSessionClockRate clock_rate; RTPSessionReconsider reconsider; RTPSessionRequestKeyUnit request_key_unit; + RTPSessionRequestTime request_time; } RTPSessionCallbacks; /** @@ -213,6 +225,7 @@ struct _RTPSession { gpointer clock_rate_user_data; gpointer reconsider_user_data; gpointer request_key_unit_user_data; + gpointer request_time_user_data; RTPSessionStats stats; @@ -278,6 +291,10 @@ void rtp_session_set_clock_rate_callback (RTPSession * sess, void rtp_session_set_reconsider_callback (RTPSession * sess, RTPSessionReconsider callback, gpointer user_data); +void rtp_session_set_request_time_callback (RTPSession * sess, + RTPSessionRequestTime callback, + gpointer user_data); + void rtp_session_set_bandwidth (RTPSession *sess, gdouble bandwidth); gdouble rtp_session_get_bandwidth (RTPSession *sess); void rtp_session_set_rtcp_fraction (RTPSession *sess, gdouble fraction); From cd923223dd24576679ff91deda71cc8f172504c3 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Olivier=20Cr=C3=AAte?= Date: Fri, 27 Aug 2010 14:11:53 -0400 Subject: [PATCH 282/351] rtpsession: Add action signal to request early RTCP --- gst/rtpmanager/gstrtpbin-marshal.list | 1 + gst/rtpmanager/rtpsession.c | 33 +++++++++++++++++++++++++++ gst/rtpmanager/rtpsession.h | 1 + 3 files changed, 35 insertions(+) diff --git a/gst/rtpmanager/gstrtpbin-marshal.list b/gst/rtpmanager/gstrtpbin-marshal.list index 5176b334d4..7d9821f512 100644 --- a/gst/rtpmanager/gstrtpbin-marshal.list +++ b/gst/rtpmanager/gstrtpbin-marshal.list @@ -9,3 +9,4 @@ VOID:OBJECT,OBJECT UINT64:BOOL,UINT64 BOOL:POINTER,BOOL VOID:UINT,UINT,UINT,UINT,POINTER +VOID:UINT64 diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c index a60e7c0945..49ce69b613 100644 --- a/gst/rtpmanager/rtpsession.c +++ b/gst/rtpmanager/rtpsession.c @@ -44,6 +44,7 @@ enum SIGNAL_ON_SENDER_TIMEOUT, SIGNAL_ON_SENDING_RTCP, SIGNAL_ON_FEEDBACK_RTCP, + SIGNAL_SEND_RTCP, LAST_SIGNAL }; @@ -104,6 +105,8 @@ static void rtp_session_get_property (GObject * object, guint prop_id, static gboolean rtp_session_on_sending_rtcp (RTPSession * sess, GstBuffer * buffer, gboolean early); +static void rtp_session_send_rtcp (RTPSession * sess, + GstClockTimeDiff max_delay); static guint rtp_session_signals[LAST_SIGNAL] = { 0 }; @@ -299,6 +302,22 @@ rtp_session_class_init (RTPSessionClass * klass) G_TYPE_NONE, 4, G_TYPE_UINT, G_TYPE_UINT, G_TYPE_UINT, G_TYPE_UINT, G_TYPE_POINTER); + /** + * RTPSession::send-rtcp: + * @session: the object which received the signal + * @max_delay: The maximum delay after which the feedback will not be useful + * anymore + * + * Requests that the #RTPSession initiate a new RTCP packet as soon as + * possible within the requested delay. + */ + + rtp_session_signals[SIGNAL_SEND_RTCP] = + g_signal_new ("send-rtcp", G_TYPE_FROM_CLASS (klass), + G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, + G_STRUCT_OFFSET (RTPSessionClass, send_rtcp), NULL, NULL, + gst_rtp_bin_marshal_VOID__UINT64, G_TYPE_NONE, 1, G_TYPE_UINT64); + g_object_class_install_property (gobject_class, PROP_INTERNAL_SSRC, g_param_spec_uint ("internal-ssrc", "Internal SSRC", "The internal SSRC used for the session", @@ -408,6 +427,7 @@ rtp_session_class_init (RTPSessionClass * klass) klass->get_source_by_ssrc = GST_DEBUG_FUNCPTR (rtp_session_get_source_by_ssrc); klass->on_sending_rtcp = GST_DEBUG_FUNCPTR (rtp_session_on_sending_rtcp); + klass->send_rtcp = GST_DEBUG_FUNCPTR (rtp_session_send_rtcp); GST_DEBUG_CATEGORY_INIT (rtp_session_debug, "rtpsession", 0, "RTP Session"); } @@ -3134,3 +3154,16 @@ rtp_session_on_sending_rtcp (RTPSession * sess, GstBuffer * buffer, return ret; } + +static void +rtp_session_send_rtcp (RTPSession * sess, GstClockTimeDiff max_delay) +{ + GstClockTime now; + + if (!sess->callbacks.send_rtcp) + return; + + now = sess->callbacks.request_time (sess, sess->request_time_user_data); + + rtp_session_request_early_rtcp (sess, now, max_delay); +} diff --git a/gst/rtpmanager/rtpsession.h b/gst/rtpmanager/rtpsession.h index 3746a9b721..3268490cac 100644 --- a/gst/rtpmanager/rtpsession.h +++ b/gst/rtpmanager/rtpsession.h @@ -264,6 +264,7 @@ struct _RTPSessionClass { gboolean early); void (*on_feedback_rtcp) (RTPSession *sess, guint type, guint fbtype, guint sender_ssrc, guint media_ssrc, GstBuffer *fci); + void (*send_rtcp) (RTPSession *sess, GstClockTimeDiff max_delay); }; GType rtp_session_get_type (void); From 8a7a327db7dd0e677844599f4a0db76a7d06b512 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Olivier=20Cr=C3=AAte?= Date: Wed, 6 Oct 2010 21:17:28 -0400 Subject: [PATCH 283/351] rtptheoradepay: Request new keyframe on lost packets Theora can only use the last frame (or the keyframe) as a reference, so in practice. If we receive a buffer that references an unknown codebook, request new headers. It probably means that headers were lost. --- gst/rtp/gstrtptheoradepay.c | 52 ++++++++++++++++++++++++++++++++++--- gst/rtp/gstrtptheoradepay.h | 2 ++ 2 files changed, 51 insertions(+), 3 deletions(-) diff --git a/gst/rtp/gstrtptheoradepay.c b/gst/rtp/gstrtptheoradepay.c index dfe6de3d45..a21cc6c6df 100644 --- a/gst/rtp/gstrtptheoradepay.c +++ b/gst/rtp/gstrtptheoradepay.c @@ -67,6 +67,8 @@ static gboolean gst_rtp_theora_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps); static GstBuffer *gst_rtp_theora_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf); +static gboolean gst_rtp_theora_depay_packet_lost (GstBaseRTPDepayload * + depayload, GstEvent * event); static void gst_rtp_theora_depay_finalize (GObject * object); @@ -100,6 +102,7 @@ gst_rtp_theora_depay_class_init (GstRtpTheoraDepayClass * klass) gstbasertpdepayload_class->process = gst_rtp_theora_depay_process; gstbasertpdepayload_class->set_caps = gst_rtp_theora_depay_setcaps; + gstbasertpdepayload_class->packet_lost = gst_rtp_theora_depay_packet_lost; GST_DEBUG_CATEGORY_INIT (rtptheoradepay_debug, "rtptheoradepay", 0, "Theora RTP Depayloader"); @@ -308,6 +311,8 @@ gst_rtp_theora_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps) rtptheoradepay = GST_RTP_THEORA_DEPAY (depayload); + rtptheoradepay->needs_keyframe = FALSE; + structure = gst_caps_get_structure (caps, 0); /* read and parse configuration string */ @@ -554,6 +559,9 @@ gst_rtp_theora_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf) memcpy (GST_BUFFER_DATA (outbuf), payload, length); } + if ((payload[0] & 0xC0) == 0x0) + rtptheoradepay->needs_keyframe = FALSE; + payload += length; payload_len -= length; @@ -573,6 +581,9 @@ gst_rtp_theora_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf) g_free (to_free); + if (rtptheoradepay->needs_keyframe) + goto request_keyframe; + return NULL; no_output: @@ -584,13 +595,13 @@ switch_failed: { GST_ELEMENT_WARNING (rtptheoradepay, STREAM, DECODE, (NULL), ("Could not switch codebooks")); - return NULL; + goto request_config; } packet_short: { GST_ELEMENT_WARNING (rtptheoradepay, STREAM, DECODE, (NULL), ("Packet was too short (%d < 4)", payload_len)); - return NULL; + goto request_keyframe; } ignore_reserved: { @@ -601,13 +612,29 @@ length_short: { GST_ELEMENT_WARNING (rtptheoradepay, STREAM, DECODE, (NULL), ("Packet contains invalid data")); - return NULL; + goto request_keyframe; } invalid_configuration: { /* fatal, as we otherwise risk carrying on without output */ GST_ELEMENT_ERROR (rtptheoradepay, STREAM, DECODE, (NULL), ("Packet contains invalid configuration")); + goto request_config; + } +request_config: + { + gst_pad_push_event (GST_BASE_RTP_DEPAYLOAD_SINKPAD (depayload), + gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, + gst_structure_new ("GstForceKeyUnit", + "all-headers", G_TYPE_BOOLEAN, TRUE, NULL))); + return NULL; + } +request_keyframe: + { + rtptheoradepay->needs_keyframe = TRUE; + gst_pad_push_event (GST_BASE_RTP_DEPAYLOAD_SINKPAD (depayload), + gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, + gst_structure_new ("GstForceKeyUnit", NULL))); return NULL; } } @@ -618,3 +645,22 @@ gst_rtp_theora_depay_plugin_init (GstPlugin * plugin) return gst_element_register (plugin, "rtptheoradepay", GST_RANK_SECONDARY, GST_TYPE_RTP_THEORA_DEPAY); } + +static gboolean +gst_rtp_theora_depay_packet_lost (GstBaseRTPDepayload * depayload, + GstEvent * event) +{ + GstRtpTheoraDepay *rtptheoradepay = GST_RTP_THEORA_DEPAY (depayload); + guint seqnum = 0; + + gst_structure_get_uint (event->structure, "seqnum", &seqnum); + GST_LOG_OBJECT (depayload, "Requested keyframe because frame with seqnum %u" + " is missing", seqnum); + rtptheoradepay->needs_keyframe = TRUE; + + gst_pad_push_event (GST_BASE_RTP_DEPAYLOAD_SINKPAD (depayload), + gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, + gst_structure_new ("GstForceKeyUnit", NULL))); + + return TRUE; +} diff --git a/gst/rtp/gstrtptheoradepay.h b/gst/rtp/gstrtptheoradepay.h index 0d1a911f9f..dff261c43c 100644 --- a/gst/rtp/gstrtptheoradepay.h +++ b/gst/rtp/gstrtptheoradepay.h @@ -54,6 +54,8 @@ struct _GstRtpTheoraDepay GstAdapter *adapter; gboolean assembling; + + gboolean needs_keyframe; }; struct _GstRtpTheoraDepayClass From 564976b6095783d73849c51cf11e16b1a03483c4 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Tue, 1 Feb 2011 19:40:58 +0100 Subject: [PATCH 284/351] directsound: arrange for definition of _swab on Cygwin gstdirectsoundsink.c: In function 'gst_directsound_sink_write': gstdirectsoundsink.c:557: error: implicit declaration of function '_swab' gstdirectsoundsink.c:557: error: nested extern declaration of '_swab' --- sys/directsound/gstdirectsoundsink.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sys/directsound/gstdirectsoundsink.c b/sys/directsound/gstdirectsoundsink.c index e64c8ac38e..462c41210f 100644 --- a/sys/directsound/gstdirectsoundsink.c +++ b/sys/directsound/gstdirectsoundsink.c @@ -56,6 +56,13 @@ #include +#ifdef __CYGWIN__ +#include +#ifndef _swab +#define _swab swab +#endif +#endif + GST_DEBUG_CATEGORY_STATIC (directsoundsink_debug); #define GST_CAT_DEFAULT directsoundsink_debug From 8598aaf81b2d8ba601ff5407b3084e525824ce1c Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Wed, 2 Feb 2011 18:21:26 +0100 Subject: [PATCH 285/351] rtpbin: Get and use the NTP time when receiving RTCP When we receive an RTCP packet, get the current NTP time in nanseconds so that we can correctly calculate the round-trip time. --- gst/rtpmanager/gstrtpsession.c | 6 +++++- gst/rtpmanager/rtpsession.c | 13 ++++++++----- gst/rtpmanager/rtpsession.h | 3 ++- gst/rtpmanager/rtpsource.c | 14 +++++++++----- gst/rtpmanager/rtpsource.h | 2 +- gst/rtpmanager/rtpstats.h | 2 ++ 6 files changed, 27 insertions(+), 13 deletions(-) diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c index b1223d6585..64abc94ba6 100644 --- a/gst/rtpmanager/gstrtpsession.c +++ b/gst/rtpmanager/gstrtpsession.c @@ -1581,6 +1581,7 @@ gst_rtp_session_chain_recv_rtcp (GstPad * pad, GstBuffer * buffer) GstRtpSession *rtpsession; GstRtpSessionPrivate *priv; GstClockTime current_time; + guint64 ntpnstime; GstFlowReturn ret; rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); @@ -1589,7 +1590,10 @@ gst_rtp_session_chain_recv_rtcp (GstPad * pad, GstBuffer * buffer) GST_LOG_OBJECT (rtpsession, "received RTCP packet"); current_time = gst_clock_get_time (priv->sysclock); - ret = rtp_session_process_rtcp (priv->session, buffer, current_time); + get_current_times (rtpsession, NULL, &ntpnstime); + + ret = + rtp_session_process_rtcp (priv->session, buffer, current_time, ntpnstime); gst_object_unref (rtpsession); diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c index 49ce69b613..a455ad3564 100644 --- a/gst/rtpmanager/rtpsession.c +++ b/gst/rtpmanager/rtpsession.c @@ -1598,11 +1598,12 @@ rtp_session_create_source (RTPSession * sess) static void update_arrival_stats (RTPSession * sess, RTPArrivalStats * arrival, gboolean rtp, GstBuffer * buffer, GstClockTime current_time, - GstClockTime running_time) + GstClockTime running_time, guint64 ntpnstime) { /* get time of arrival */ arrival->current_time = current_time; arrival->running_time = running_time; + arrival->ntpnstime = ntpnstime; /* get packet size including header overhead */ arrival->bytes = GST_BUFFER_SIZE (buffer) + sess->header_len; @@ -1657,7 +1658,7 @@ rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer, RTP_SESSION_LOCK (sess); /* update arrival stats */ update_arrival_stats (sess, &arrival, TRUE, buffer, current_time, - running_time); + running_time, -1); /* ignore more RTP packets when we left the session */ if (sess->source->received_bye) @@ -1778,7 +1779,7 @@ rtp_session_process_rb (RTPSession * sess, RTPSource * source, /* only deal with report blocks for our session, we update the stats of * the sender of the RTCP message. We could also compare our stats against * the other sender to see if we are better or worse. */ - rtp_source_process_rb (source, arrival->current_time, fractionlost, + rtp_source_process_rb (source, arrival->ntpnstime, fractionlost, packetslost, exthighestseq, jitter, lsr, dlsr); } } @@ -2153,6 +2154,7 @@ rtp_session_process_feedback (RTPSession * sess, GstRTCPPacket * packet, * @sess: and #RTPSession * @buffer: an RTCP buffer * @current_time: the current system time + * @ntpnstime: the current NTP time in nanoseconds * * Process an RTCP buffer in the session manager. This function takes ownership * of @buffer. @@ -2161,7 +2163,7 @@ rtp_session_process_feedback (RTPSession * sess, GstRTCPPacket * packet, */ GstFlowReturn rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer, - GstClockTime current_time) + GstClockTime current_time, guint64 ntpnstime) { GstRTCPPacket packet; gboolean more, is_bye = FALSE, do_sync = FALSE; @@ -2178,7 +2180,8 @@ rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer, RTP_SESSION_LOCK (sess); /* update arrival stats */ - update_arrival_stats (sess, &arrival, FALSE, buffer, current_time, -1); + update_arrival_stats (sess, &arrival, FALSE, buffer, current_time, -1, + ntpnstime); if (sess->sent_bye) goto ignore; diff --git a/gst/rtpmanager/rtpsession.h b/gst/rtpmanager/rtpsession.h index 3268490cac..6dee1cbe28 100644 --- a/gst/rtpmanager/rtpsession.h +++ b/gst/rtpmanager/rtpsession.h @@ -326,7 +326,8 @@ GstFlowReturn rtp_session_process_rtp (RTPSession *sess, GstBuffer GstClockTime current_time, GstClockTime running_time); GstFlowReturn rtp_session_process_rtcp (RTPSession *sess, GstBuffer *buffer, - GstClockTime current_time); + GstClockTime current_time, + guint64 ntpnstime); /* processing packets for sending */ GstFlowReturn rtp_session_send_rtp (RTPSession *sess, gpointer data, gboolean is_list, diff --git a/gst/rtpmanager/rtpsource.c b/gst/rtpmanager/rtpsource.c index 423d2b65d2..e7872d8087 100644 --- a/gst/rtpmanager/rtpsource.c +++ b/gst/rtpmanager/rtpsource.c @@ -1350,7 +1350,7 @@ rtp_source_process_sr (RTPSource * src, GstClockTime time, guint64 ntptime, /** * rtp_source_process_rb: * @src: an #RTPSource - * @time: the current time in nanoseconds since 1970 + * @ntpnstime: the current time in nanoseconds since 1970 * @fractionlost: fraction lost since last SR/RR * @packetslost: the cumululative number of packets lost * @exthighestseq: the extended last sequence number received @@ -1361,13 +1361,14 @@ rtp_source_process_sr (RTPSource * src, GstClockTime time, guint64 ntptime, * Update the report block in @src. */ void -rtp_source_process_rb (RTPSource * src, GstClockTime time, guint8 fractionlost, - gint32 packetslost, guint32 exthighestseq, guint32 jitter, guint32 lsr, - guint32 dlsr) +rtp_source_process_rb (RTPSource * src, GstClockTime ntpnstime, + guint8 fractionlost, gint32 packetslost, guint32 exthighestseq, + guint32 jitter, guint32 lsr, guint32 dlsr) { RTPReceiverReport *curr; gint curridx; guint32 ntp, A; + guint64 f_ntp; g_return_if_fail (RTP_IS_SOURCE (src)); @@ -1388,8 +1389,11 @@ rtp_source_process_rb (RTPSource * src, GstClockTime time, guint8 fractionlost, curr->lsr = lsr; curr->dlsr = dlsr; + /* convert the NTP time in nanoseconds to 32.32 fixed point */ + f_ntp = gst_util_uint64_scale (ntpnstime, (1LL << 32), GST_SECOND); /* calculate round trip, round the time up */ - ntp = ((gst_rtcp_unix_to_ntp (time) + 0xffff) >> 16) & 0xffffffff; + ntp = ((f_ntp + 0xffff) >> 16) & 0xffffffff; + A = dlsr + lsr; if (A > 0 && ntp > A) A = ntp - A; diff --git a/gst/rtpmanager/rtpsource.h b/gst/rtpmanager/rtpsource.h index fadbe3068a..6db0a6a678 100644 --- a/gst/rtpmanager/rtpsource.h +++ b/gst/rtpmanager/rtpsource.h @@ -220,7 +220,7 @@ GstFlowReturn rtp_source_send_rtp (RTPSource *src, gpointer data, g void rtp_source_process_bye (RTPSource *src, const gchar *reason); void rtp_source_process_sr (RTPSource *src, GstClockTime time, guint64 ntptime, guint32 rtptime, guint32 packet_count, guint32 octet_count); -void rtp_source_process_rb (RTPSource *src, GstClockTime time, guint8 fractionlost, +void rtp_source_process_rb (RTPSource *src, guint64 ntpnstime, guint8 fractionlost, gint32 packetslost, guint32 exthighestseq, guint32 jitter, guint32 lsr, guint32 dlsr); diff --git a/gst/rtpmanager/rtpstats.h b/gst/rtpmanager/rtpstats.h index 643da004ab..4560595edb 100644 --- a/gst/rtpmanager/rtpstats.h +++ b/gst/rtpmanager/rtpstats.h @@ -58,6 +58,7 @@ typedef struct { * RTPArrivalStats: * @current_time: current time according to the system clock * @running_time: arrival time of a packet as buffer running_time + * @ntpnstime: arrival time of a packet NTP time in nanoseconds * @have_address: if the @address field contains a valid address * @address: address of the sender of the packet * @bytes: bytes of the packet including lowlevel overhead @@ -68,6 +69,7 @@ typedef struct { typedef struct { GstClockTime current_time; GstClockTime running_time; + guint64 ntpnstime; gboolean have_address; GstNetAddress address; guint bytes; From 61382aad280abfd719c5ce23b4cd0fe34552b1c1 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Wed, 2 Feb 2011 18:27:52 +0100 Subject: [PATCH 286/351] source: fix type of ntpnstime --- gst/rtpmanager/rtpsource.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/gst/rtpmanager/rtpsource.c b/gst/rtpmanager/rtpsource.c index e7872d8087..fb9e039553 100644 --- a/gst/rtpmanager/rtpsource.c +++ b/gst/rtpmanager/rtpsource.c @@ -1361,7 +1361,7 @@ rtp_source_process_sr (RTPSource * src, GstClockTime time, guint64 ntptime, * Update the report block in @src. */ void -rtp_source_process_rb (RTPSource * src, GstClockTime ntpnstime, +rtp_source_process_rb (RTPSource * src, guint64 ntpnstime, guint8 fractionlost, gint32 packetslost, guint32 exthighestseq, guint32 jitter, guint32 lsr, guint32 dlsr) { From 6b86b56abd8a28b9e94cee679e3d86367a9a0ab3 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Thu, 3 Feb 2011 23:42:59 +0100 Subject: [PATCH 287/351] v4l2: fix interlaced set_format configuration Commit 6c8268dbfd5c88fac28c882ef2e4598a6522e2d6 broke recording from interlaced v4l2 source (e.g. typical tv capture card) since V4L2_FIELD_SEQ_TB (with fields stored separately) does not map to currently defined interlaced format (fields stored interleaved). Besides this mismatch, hardware might quite likely not support or appreciate this field value, since querying supported formats mapped _INTERLACED field formats to interlaced=true caps (so the latter should not be mapped to field value that is not known to be supported). --- sys/v4l2/gstv4l2object.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sys/v4l2/gstv4l2object.c b/sys/v4l2/gstv4l2object.c index 7e15489c4e..5c9022c18b 100644 --- a/sys/v4l2/gstv4l2object.c +++ b/sys/v4l2/gstv4l2object.c @@ -2019,7 +2019,7 @@ gst_v4l2_object_set_format (GstV4l2Object * v4l2object, guint32 pixelformat, /* ideally we would differentiate between types of interlaced video * but there is not sufficient information in the caps.. */ - field = V4L2_FIELD_SEQ_TB; + field = V4L2_FIELD_INTERLACED; } else { GST_DEBUG_OBJECT (v4l2object->element, "progressive video"); field = V4L2_FIELD_NONE; From 18213aa214139de64495962761a4c3e4fad562ed Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Fri, 4 Feb 2011 12:33:09 +0200 Subject: [PATCH 288/351] docs: fix parameter name in udpsrc docs It is "buffer-size" and not "buffer". Also trim trailing whitespace. --- gst/udp/gstudpsrc.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/gst/udp/gstudpsrc.c b/gst/udp/gstudpsrc.c index f00c74b364..4363044da2 100644 --- a/gst/udp/gstudpsrc.c +++ b/gst/udp/gstudpsrc.c @@ -41,8 +41,8 @@ * for RTP implementations where the contents of the UDP packets is transfered * out-of-bounds using SDP or other means. * - * The #GstUDPSrc:buffer property is used to change the default kernel buffer - * sizes used for receiving packets. The buffer size may be increased for + * The #GstUDPSrc:buffer-size property is used to change the default kernel + * buffersizes used for receiving packets. The buffer size may be increased for * high-volume connections, or may be decreased to limit the possible backlog of * incoming data. The system places an absolute limit on these values, on Linux, * for example, the default buffer size is typically 50K and can be increased to @@ -50,7 +50,7 @@ * * The #GstUDPSrc:skip-first-bytes property is used to strip off an arbitrary * number of bytes from the start of the raw udp packet and can be used to strip - * off proprietary header, for example. + * off proprietary header, for example. * * The udpsrc is always a live source. It does however not provide a #GstClock, * this is left for upstream elements such as an RTP session manager or demuxer @@ -78,7 +78,7 @@ * because it is blocked by a firewall. * * - * A custom file descriptor can be configured with the + * A custom file descriptor can be configured with the * #GstUDPSrc:sockfd property. The socket will be closed when setting the * element to READY by default. This behaviour can be * overriden with the #GstUDPSrc:closefd property, in which case the application From cb527623a31bf9cc701630f90c8754f336dd874b Mon Sep 17 00:00:00 2001 From: Thiago Santos Date: Thu, 3 Feb 2011 16:10:49 -0300 Subject: [PATCH 289/351] deinterlace: Handle image caps without asserting Images might have framerate=0/1 in the caps, which caused an assertion on deinterlace. I don't know of interlaced image formats but deinterlace might be hardcoded on some generic pipelines and it shouldn't assert. The fix was to set field_duration to 0 if the input has a framerate with a 0 numerator. This patch also adds checks for this situation on the unit tests. https://bugzilla.gnome.org/show_bug.cgi?id=641400 --- gst/deinterlace/gstdeinterlace.c | 16 +++++++----- tests/check/elements/deinterlace.c | 40 ++++++++++++++++++++++++++++++ 2 files changed, 50 insertions(+), 6 deletions(-) diff --git a/gst/deinterlace/gstdeinterlace.c b/gst/deinterlace/gstdeinterlace.c index 300f23e7fd..2f7d2ea0e6 100644 --- a/gst/deinterlace/gstdeinterlace.c +++ b/gst/deinterlace/gstdeinterlace.c @@ -1433,12 +1433,16 @@ gst_deinterlace_setcaps (GstPad * pad, GstCaps * caps) self->frame_size = gst_video_format_get_size (self->format, self->width, self->height); - if (self->fields == GST_DEINTERLACE_ALL && otherpad == self->srcpad) - self->field_duration = - gst_util_uint64_scale (GST_SECOND, self->fps_d, self->fps_n); - else - self->field_duration = - gst_util_uint64_scale (GST_SECOND, self->fps_d, 2 * self->fps_n); + if (G_LIKELY (self->fps_n != 0)) { + if (self->fields == GST_DEINTERLACE_ALL && otherpad == self->srcpad) + self->field_duration = + gst_util_uint64_scale (GST_SECOND, self->fps_d, self->fps_n); + else + self->field_duration = + gst_util_uint64_scale (GST_SECOND, self->fps_d, 2 * self->fps_n); + } else { + self->field_duration = 0; + } if (pad == self->sinkpad) { gst_caps_replace (&self->sink_caps, caps); diff --git a/tests/check/elements/deinterlace.c b/tests/check/elements/deinterlace.c index 6606a5f7a1..b111c86f74 100644 --- a/tests/check/elements/deinterlace.c +++ b/tests/check/elements/deinterlace.c @@ -53,6 +53,9 @@ GST_END_TEST; #define CAPS_VIDEO_COMMON \ "width=(int)800, height=(int)600, framerate=(fraction)15/1" +#define CAPS_IMAGE_COMMON \ + "width=(int)3200, height=(int)3400, framerate=(fraction)0/1" + #define CAPS_YUY2 \ "video/x-raw-yuv, " \ CAPS_VIDEO_COMMON ", " \ @@ -71,6 +74,24 @@ GST_END_TEST; CAPS_YVYU ", " \ "interlaced=(boolean)true" +#define CAPS_YUY2_IMAGE \ + "video/x-raw-yuv, " \ + CAPS_IMAGE_COMMON ", " \ + "format=(fourcc)YUY2" + +#define CAPS_YUY2_INTERLACED_IMAGE \ + CAPS_YUY2_IMAGE ", " \ + "interlaced=(boolean)true" + +#define CAPS_YVYU_IMAGE \ + "video/x-raw-yuv, " \ + CAPS_IMAGE_COMMON ", " \ + "format=(fourcc)YVYU" + +#define CAPS_YVYU_INTERLACED_IMAGE \ + CAPS_YVYU_IMAGE ", " \ + "interlaced=(boolean)true" + static GstElement *deinterlace; static GstPad *srcpad; static GstPad *sinkpad; @@ -295,10 +316,14 @@ GST_START_TEST (test_mode_auto_accept_caps) /* try to set non interlaced caps */ deinterlace_set_string_caps_and_check (CAPS_YVYU, FALSE); deinterlace_set_string_caps_and_check (CAPS_YUY2, FALSE); + deinterlace_set_string_caps_and_check (CAPS_YVYU_IMAGE, FALSE); + deinterlace_set_string_caps_and_check (CAPS_YUY2_IMAGE, FALSE); /* now try to set interlaced caps */ deinterlace_set_string_caps_and_check (CAPS_YVYU_INTERLACED, TRUE); deinterlace_set_string_caps_and_check (CAPS_YUY2_INTERLACED, TRUE); + deinterlace_set_string_caps_and_check (CAPS_YVYU_INTERLACED_IMAGE, TRUE); + deinterlace_set_string_caps_and_check (CAPS_YUY2_INTERLACED_IMAGE, TRUE); /* cleanup */ gst_object_unref (sinkpad); @@ -322,10 +347,14 @@ GST_START_TEST (test_mode_forced_accept_caps) /* try to set non interlaced caps */ deinterlace_set_string_caps_and_check (CAPS_YVYU, TRUE); deinterlace_set_string_caps_and_check (CAPS_YUY2, TRUE); + deinterlace_set_string_caps_and_check (CAPS_YVYU_IMAGE, TRUE); + deinterlace_set_string_caps_and_check (CAPS_YUY2_IMAGE, TRUE); /* now try to set interlaced caps */ deinterlace_set_string_caps_and_check (CAPS_YVYU_INTERLACED, TRUE); deinterlace_set_string_caps_and_check (CAPS_YUY2_INTERLACED, TRUE); + deinterlace_set_string_caps_and_check (CAPS_YVYU_INTERLACED_IMAGE, TRUE); + deinterlace_set_string_caps_and_check (CAPS_YUY2_INTERLACED_IMAGE, TRUE); /* cleanup */ gst_object_unref (sinkpad); @@ -349,10 +378,14 @@ GST_START_TEST (test_mode_disabled_accept_caps) /* try to set non interlaced caps */ deinterlace_set_string_caps_and_check (CAPS_YVYU, FALSE); deinterlace_set_string_caps_and_check (CAPS_YUY2, FALSE); + deinterlace_set_string_caps_and_check (CAPS_YVYU_IMAGE, FALSE); + deinterlace_set_string_caps_and_check (CAPS_YUY2_IMAGE, FALSE); /* now try to set interlaced caps */ deinterlace_set_string_caps_and_check (CAPS_YVYU_INTERLACED, FALSE); deinterlace_set_string_caps_and_check (CAPS_YUY2_INTERLACED, FALSE); + deinterlace_set_string_caps_and_check (CAPS_YVYU_INTERLACED_IMAGE, FALSE); + deinterlace_set_string_caps_and_check (CAPS_YUY2_INTERLACED_IMAGE, FALSE); /* cleanup */ gst_object_unref (sinkpad); @@ -371,6 +404,11 @@ GST_START_TEST (test_mode_disabled_passthrough) deinterlace_check_passthrough (2, CAPS_YVYU_INTERLACED); deinterlace_check_passthrough (2, CAPS_YUY2); deinterlace_check_passthrough (2, CAPS_YVYU); + + deinterlace_check_passthrough (2, CAPS_YUY2_INTERLACED_IMAGE); + deinterlace_check_passthrough (2, CAPS_YVYU_INTERLACED_IMAGE); + deinterlace_check_passthrough (2, CAPS_YUY2_IMAGE); + deinterlace_check_passthrough (2, CAPS_YVYU_IMAGE); } GST_END_TEST; @@ -380,6 +418,8 @@ GST_START_TEST (test_mode_auto_deinterlaced_passthrough) /* 0 is auto mode */ deinterlace_check_passthrough (0, CAPS_YUY2); deinterlace_check_passthrough (0, CAPS_YVYU); + deinterlace_check_passthrough (0, CAPS_YUY2_IMAGE); + deinterlace_check_passthrough (0, CAPS_YVYU_IMAGE); } GST_END_TEST; From 5e071d51f26145686d90e9e188d5a7bb6d98a27c Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Mon, 7 Feb 2011 17:07:42 +0200 Subject: [PATCH 290/351] rtpsrc: set multiple properties in one go There is no need for separate g_object_set() calls here. --- gst/rtsp/gstrtspsrc.c | 29 +++++++++++------------------ 1 file changed, 11 insertions(+), 18 deletions(-) diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c index e4ec109766..dd602673a9 100644 --- a/gst/rtsp/gstrtspsrc.c +++ b/gst/rtsp/gstrtspsrc.c @@ -2806,13 +2806,11 @@ gst_rtspsrc_stream_configure_udp_sinks (GstRTSPSrc * src, goto no_sink_element; /* don't join multicast group, we will have the source socket do that */ - g_object_set (G_OBJECT (stream->udpsink[0]), "auto-multicast", FALSE, NULL); + /* no sync or async state changes needed */ + g_object_set (G_OBJECT (stream->udpsink[0]), "auto-multicast", FALSE, + "loop", FALSE, "sync", FALSE, "async", FALSE, NULL); if (ttl > 0) g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL); - g_object_set (G_OBJECT (stream->udpsink[0]), "loop", FALSE, NULL); - /* no sync or async state changes needed */ - g_object_set (G_OBJECT (stream->udpsink[0]), "sync", FALSE, "async", FALSE, - NULL); if (stream->udpsrc[0]) { /* configure socket, we give it the same UDP socket as the udpsrc for RTP @@ -2821,8 +2819,8 @@ gst_rtspsrc_stream_configure_udp_sinks (GstRTSPSrc * src, GST_DEBUG_OBJECT (src, "RTP UDP src has sock %d", sockfd); /* configure socket and make sure udpsink does not close it when shutting * down, it belongs to udpsrc after all. */ - g_object_set (G_OBJECT (stream->udpsink[0]), "sockfd", sockfd, NULL); - g_object_set (G_OBJECT (stream->udpsink[0]), "closefd", FALSE, NULL); + g_object_set (G_OBJECT (stream->udpsink[0]), "sockfd", sockfd, + "closefd", FALSE, NULL); } /* the source for the dummy packets to open up NAT */ @@ -2832,9 +2830,7 @@ gst_rtspsrc_stream_configure_udp_sinks (GstRTSPSrc * src, /* random data in 5 buffers, a size of 200 bytes should be fine */ g_object_set (G_OBJECT (stream->fakesrc), "filltype", 3, "num-buffers", 5, - NULL); - g_object_set (G_OBJECT (stream->fakesrc), "sizetype", 2, "sizemax", 200, - "silent", TRUE, NULL); + "sizetype", 2, "sizemax", 200, "silent", TRUE, NULL); /* we don't want to consider this a sink */ GST_OBJECT_FLAG_UNSET (stream->udpsink[0], GST_ELEMENT_IS_SINK); @@ -2861,14 +2857,11 @@ gst_rtspsrc_stream_configure_udp_sinks (GstRTSPSrc * src, goto no_sink_element; /* don't join multicast group, we will have the source socket do that */ - g_object_set (G_OBJECT (stream->udpsink[1]), "auto-multicast", FALSE, NULL); + /* no sync or async state changes needed */ + g_object_set (G_OBJECT (stream->udpsink[1]), "auto-multicast", FALSE, + "loop", FALSE, "sync", FALSE, "async", FALSE, NULL); if (ttl > 0) g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL); - g_object_set (G_OBJECT (stream->udpsink[1]), "loop", FALSE, NULL); - /* no sync needed */ - g_object_set (G_OBJECT (stream->udpsink[1]), "sync", FALSE, NULL); - /* no async state changes needed */ - g_object_set (G_OBJECT (stream->udpsink[1]), "async", FALSE, NULL); if (stream->udpsrc[1]) { /* configure socket, we give it the same UDP socket as the udpsrc for RTCP @@ -2878,8 +2871,8 @@ gst_rtspsrc_stream_configure_udp_sinks (GstRTSPSrc * src, GST_DEBUG_OBJECT (src, "RTCP UDP src has sock %d", sockfd); /* configure socket and make sure udpsink does not close it when shutting * down, it belongs to udpsrc after all. */ - g_object_set (G_OBJECT (stream->udpsink[1]), "sockfd", sockfd, NULL); - g_object_set (G_OBJECT (stream->udpsink[1]), "closefd", FALSE, NULL); + g_object_set (G_OBJECT (stream->udpsink[1]), "sockfd", sockfd, + "closefd", FALSE, NULL); } /* we don't want to consider this a sink */ From 6f6b2a7efce1f1368344eab4b4625c0fc30000ad Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Mon, 7 Feb 2011 17:08:47 +0200 Subject: [PATCH 291/351] rtspsrc: strip trailing spaces --- gst/rtsp/gstrtspsrc.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c index dd602673a9..89b11d90a0 100644 --- a/gst/rtsp/gstrtspsrc.c +++ b/gst/rtsp/gstrtspsrc.c @@ -56,14 +56,14 @@ * For each stream listed in the SDP a new rtp_stream%d pad will be created * with caps derived from the SDP media description. This is a caps of mime type * "application/x-rtp" that can be connected to any available RTP depayloader - * element. + * element. * * rtspsrc will internally instantiate an RTP session manager element * that will handle the RTCP messages to and from the server, jitter removal, - * packet reordering along with providing a clock for the pipeline. + * packet reordering along with providing a clock for the pipeline. * This feature is implemented using the gstrtpbin element. * - * rtspsrc acts like a live source and will therefore only generate data in the + * rtspsrc acts like a live source and will therefore only generate data in the * PLAYING state. * * @@ -1309,7 +1309,7 @@ gst_rtspsrc_sdp_attributes_to_caps (GArray * attributes, GstCaps * caps) /* * Mapping of caps to and from SDP fields: * - * m= RTP/AVP + * m= RTP/AVP * a=rtpmap: /[/] * a=fmtp: [=];... */ @@ -4133,11 +4133,11 @@ gst_rtspsrc_parse_digest_challenge (GstRTSPConnection * conn, g_slist_free (list); } -/* Parse a WWW-Authenticate Response header and determine the +/* Parse a WWW-Authenticate Response header and determine the * available authentication methods * * This code should also cope with the fact that each WWW-Authenticate - * header can contain multiple challenge methods + tokens + * header can contain multiple challenge methods + tokens * * At the moment, for Basic auth, we just do a minimal check and don't * even parse out the realm */ @@ -4166,8 +4166,8 @@ gst_rtspsrc_parse_auth_hdr (gchar * hdr, GstRTSPAuthMethod * methods, * gst_rtspsrc_setup_auth: * @src: the rtsp source * - * Configure a username and password and auth method on the - * connection object based on a response we received from the + * Configure a username and password and auth method on the + * connection object based on a response we received from the * peer. * * Currently, this requires that a username and password were supplied @@ -5672,7 +5672,7 @@ not_supported: * * url=;[seq=;rtptime=] [, url=...] * - * rtptime corresponds to the timestamp for the NPT time given in the header + * rtptime corresponds to the timestamp for the NPT time given in the header * seqbase corresponds to the next sequence number we received. This number * indicates the first seqnum after the seek and should be used to discard * packets that are from before the seek. From a86bab66893bb1a3323a756410573c117b8219ef Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Tue, 8 Feb 2011 10:07:43 +0200 Subject: [PATCH 292/351] id3demux: ensure a taglist before adding the container tag In the case of id3v1 also don't return NULL on empty tags, but also create a new taglist and add the container tag for consistency. --- gst/id3demux/gstid3demux.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/gst/id3demux/gstid3demux.c b/gst/id3demux/gstid3demux.c index 93685041ab..e91524211f 100644 --- a/gst/id3demux/gstid3demux.c +++ b/gst/id3demux/gstid3demux.c @@ -25,7 +25,7 @@ * id3demux accepts data streams with either (or both) ID3v2 regions at the * start, or ID3v1 at the end. The mime type of the data between the tag blocks * is detected using typefind functions, and the appropriate output mime type - * set on outgoing buffers. + * set on outgoing buffers. * * The element is only able to read ID3v1 tags from a seekable stream, because * they are at the end of the stream. That is, when get_range mode is supported @@ -183,6 +183,8 @@ gst_id3demux_parse_tag (GstTagDemux * demux, GstBuffer * buffer, res = id3demux_read_id3v2_tag (buffer, tag_size, tags); if (G_LIKELY (res == ID3TAGS_READ_TAG)) { + if (*tags == NULL) + *tags = gst_tag_list_new (); gst_id3demux_add_container_format (*tags); return GST_TAG_DEMUX_RESULT_OK; } else { @@ -191,9 +193,8 @@ gst_id3demux_parse_tag (GstTagDemux * demux, GstBuffer * buffer, } else { *tags = gst_tag_list_new_from_id3v1 (GST_BUFFER_DATA (buffer)); - if (G_UNLIKELY (*tags == NULL)) - return GST_TAG_DEMUX_RESULT_BROKEN_TAG; - + if (*tags == NULL) + *tags = gst_tag_list_new (); gst_id3demux_add_container_format (*tags); *tag_size = ID3V1_TAG_SIZE; return GST_TAG_DEMUX_RESULT_OK; From 9de84b45bcbd43e2e96b0287355f5695bc911280 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Wed, 9 Feb 2011 16:22:47 +0100 Subject: [PATCH 293/351] matroskademux: properly resume cluster scanning ... rather than getting offset tracking messed up, and then likely failing a subsequent assert. --- gst/matroska/matroska-demux.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/gst/matroska/matroska-demux.c b/gst/matroska/matroska-demux.c index 4541b44c0d..f1b2b783fd 100644 --- a/gst/matroska/matroska-demux.c +++ b/gst/matroska/matroska-demux.c @@ -2353,13 +2353,13 @@ gst_matroska_demux_search_cluster (GstMatroskaDemux * demux, gint64 * pos) GST_DEBUG_OBJECT (demux, "read buffer size %d at offset %" G_GINT64_FORMAT, GST_BUFFER_SIZE (buf), newpos); gst_byte_reader_init_from_buffer (&reader, buf); - cluster_pos = 0; resume: cluster_pos = gst_byte_reader_masked_scan_uint32 (&reader, 0xffffffff, - GST_MATROSKA_ID_CLUSTER, cluster_pos, - GST_BUFFER_SIZE (buf) - cluster_pos); + GST_MATROSKA_ID_CLUSTER, 0, gst_byte_reader_get_remaining (&reader)); if (cluster_pos >= 0) { newpos += cluster_pos; + /* prepare resuming at next byte */ + gst_byte_reader_skip (&reader, cluster_pos + 1); GST_DEBUG_OBJECT (demux, "found cluster ebml id at offset %" G_GINT64_FORMAT, newpos); /* extra checks whether we really sync'ed to a cluster: @@ -2399,7 +2399,7 @@ gst_matroska_demux_search_cluster (GstMatroskaDemux * demux, gint64 * pos) goto resume; } else { /* partial cluster id may have been in tail of buffer */ - newpos += MAX (GST_BUFFER_SIZE (buf), 4) - 3; + newpos += MAX (gst_byte_reader_get_remaining (&reader), 4) - 3; gst_buffer_unref (buf); buf = NULL; } From b70f4b6ffad984f76efc6f5916f7d8db42c9e496 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Thu, 10 Feb 2011 11:56:33 +0100 Subject: [PATCH 294/351] matroskademux: store cluster positions provided by SeekHead ... and use those, if available, to locate a cluster rather than scanning. --- gst/matroska/matroska-demux.c | 55 +++++++++++++++++++++++++++++++++++ gst/matroska/matroska-demux.h | 2 ++ 2 files changed, 57 insertions(+) diff --git a/gst/matroska/matroska-demux.c b/gst/matroska/matroska-demux.c index f1b2b783fd..76431193c2 100644 --- a/gst/matroska/matroska-demux.c +++ b/gst/matroska/matroska-demux.c @@ -399,6 +399,11 @@ gst_matroska_demux_reset (GstElement * element) demux->index = NULL; } + if (demux->clusters) { + g_array_free (demux->clusters, TRUE); + demux->clusters = NULL; + } + /* reset timers */ demux->clock = NULL; demux->time_scale = 1000000; @@ -2326,6 +2331,17 @@ gst_matroska_demux_move_to_entry (GstMatroskaDemux * demux, return TRUE; } +static gint +gst_matroska_cluster_compare (gint64 * i1, gint64 * i2) +{ + if (*i1 < *i2) + return -1; + else if (*i1 > *i2) + return 1; + else + return 0; +} + /* searches for a cluster start from @pos, * return GST_FLOW_OK and cluster position in @pos if found */ static GstFlowReturn @@ -2342,6 +2358,30 @@ gst_matroska_demux_search_cluster (GstMatroskaDemux * demux, gint64 * pos) orig_offset = demux->offset; + GST_LOG_OBJECT (demux, "searching cluster following offset %" G_GINT64_FORMAT, + *pos); + + if (demux->clusters) { + gint64 *cpos; + + cpos = gst_util_array_binary_search (demux->clusters->data, + demux->clusters->len, sizeof (gint64), + (GCompareDataFunc) gst_matroska_cluster_compare, + GST_SEARCH_MODE_AFTER, pos, NULL); + /* sanity check */ + if (cpos) { + GST_DEBUG_OBJECT (demux, + "cluster reported at offset %" G_GINT64_FORMAT, *cpos); + demux->offset = *cpos; + ret = + gst_matroska_demux_peek_id_length_pull (demux, &id, &length, &needed); + if (ret == GST_FLOW_OK && id == GST_MATROSKA_ID_CLUSTER) { + newpos = *cpos; + goto exit; + } + } + } + /* read in at newpos and scan for ebml cluster id */ while (1) { GstByteReader reader; @@ -2410,6 +2450,7 @@ gst_matroska_demux_search_cluster (GstMatroskaDemux * demux, gint64 * pos) buf = NULL; } +exit: demux->offset = orig_offset; *pos = newpos; return ret; @@ -5383,6 +5424,17 @@ gst_matroska_demux_parse_contents_seekentry (GstMatroskaDemux * demux, break; } + case GST_MATROSKA_ID_CLUSTER: + { + guint64 pos = seek_pos + demux->ebml_segment_start; + + GST_LOG_OBJECT (demux, "Cluster position"); + if (G_UNLIKELY (!demux->clusters)) + demux->clusters = g_array_sized_new (TRUE, TRUE, sizeof (guint64), 100); + g_array_append_val (demux->clusters, pos); + break; + } + default: GST_DEBUG_OBJECT (demux, "Ignoring Seek entry for ID=0x%x", seek_id); break; @@ -5429,6 +5481,9 @@ gst_matroska_demux_parse_contents (GstMatroskaDemux * demux, GstEbmlRead * ebml) DEBUG_ELEMENT_STOP (demux, ebml, "SeekHead", ret); + /* Sort clusters by position for easier searching */ + g_array_sort (demux->clusters, (GCompareFunc) gst_matroska_cluster_compare); + return ret; } diff --git a/gst/matroska/matroska-demux.h b/gst/matroska/matroska-demux.h index b31f79cb6a..a35a5935ac 100644 --- a/gst/matroska/matroska-demux.h +++ b/gst/matroska/matroska-demux.h @@ -92,6 +92,8 @@ typedef struct _GstMatroskaDemux { /* a cue (index) table */ GArray *index; + /* cluster positions (optional) */ + GArray *clusters; /* timescale in the file */ guint64 time_scale; From 3dc42d370437e4fc1362e861f9c44d685b262aa3 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Thu, 10 Feb 2011 18:00:11 +0100 Subject: [PATCH 295/351] qtdemux: avoid skipping exposing a stream following a removed stream --- gst/qtdemux/qtdemux.c | 1 + 1 file changed, 1 insertion(+) diff --git a/gst/qtdemux/qtdemux.c b/gst/qtdemux/qtdemux.c index b6457ce28e..7be9986f06 100644 --- a/gst/qtdemux/qtdemux.c +++ b/gst/qtdemux/qtdemux.c @@ -7362,6 +7362,7 @@ qtdemux_expose_streams (GstQTDemux * qtdemux) sizeof (QtDemuxStream *) * (GST_QTDEMUX_MAX_STREAMS - i - 1)); qtdemux->streams[GST_QTDEMUX_MAX_STREAMS - 1] = NULL; qtdemux->n_streams--; + i--; continue; } From 686189f3f0cb7c26836ea2a5b6f294faddcf9971 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Thu, 10 Feb 2011 18:11:46 +0100 Subject: [PATCH 296/351] qtdemux: propagate error during expose_streams ... as it may occur during initial parsing of fragmented file. --- gst/qtdemux/qtdemux.c | 31 +++++++++++++++++++------------ 1 file changed, 19 insertions(+), 12 deletions(-) diff --git a/gst/qtdemux/qtdemux.c b/gst/qtdemux/qtdemux.c index 7be9986f06..96e320d775 100644 --- a/gst/qtdemux/qtdemux.c +++ b/gst/qtdemux/qtdemux.c @@ -425,7 +425,7 @@ static GstCaps *qtdemux_sub_caps (GstQTDemux * qtdemux, gchar ** codec_name); static gboolean qtdemux_parse_samples (GstQTDemux * qtdemux, QtDemuxStream * stream, guint32 n); -static void qtdemux_expose_streams (GstQTDemux * qtdemux); +static GstFlowReturn qtdemux_expose_streams (GstQTDemux * qtdemux); static void @@ -2765,7 +2765,7 @@ gst_qtdemux_loop_state_header (GstQTDemux * qtdemux) beach: if (ret == GST_FLOW_UNEXPECTED && qtdemux->got_moov) { /* digested all data, show what we have */ - qtdemux_expose_streams (qtdemux); + ret = qtdemux_expose_streams (qtdemux); /* Only post, event on pads is done after newsegment */ qtdemux_post_global_tags (qtdemux); @@ -2773,7 +2773,7 @@ beach: qtdemux->state = QTDEMUX_STATE_MOVIE; GST_DEBUG_OBJECT (qtdemux, "switching state to STATE_MOVIE (%d)", qtdemux->state); - return GST_FLOW_OK; + return ret; } return ret; } @@ -5116,20 +5116,20 @@ locate_failed: /* should only do something in pull mode */ /* call with OBJECT lock */ -static gboolean +static GstFlowReturn qtdemux_add_fragmented_samples (GstQTDemux * qtdemux) { guint64 length, offset; GstBuffer *buf = NULL; GstFlowReturn ret = GST_FLOW_OK; - gboolean res = TRUE; + GstFlowReturn res = TRUE; offset = qtdemux->moof_offset; GST_DEBUG_OBJECT (qtdemux, "next moof at offset %" G_GUINT64_FORMAT, offset); if (!offset) { GST_DEBUG_OBJECT (qtdemux, "no next moof"); - return FALSE; + return GST_FLOW_UNEXPECTED; } /* best not do pull etc with lock held */ @@ -5169,7 +5169,7 @@ parse_failed: { GST_DEBUG_OBJECT (qtdemux, "failed to parse moof"); offset = 0; - res = FALSE; + res = GST_FLOW_ERROR; goto exit; } flow_failed: @@ -5182,7 +5182,7 @@ flow_failed: GST_DEBUG_OBJECT (qtdemux, "upstream WRONG_STATE"); /* resume at current position next time */ } - res = FALSE; + res = ret; goto exit; } } @@ -5772,7 +5772,7 @@ done: GST_DEBUG_OBJECT (qtdemux, "parsed all available samples; checking for more"); while (n + 1 == stream->n_samples) - if (!qtdemux_add_fragmented_samples (qtdemux)) + if (qtdemux_add_fragmented_samples (qtdemux) != GST_FLOW_OK) break; } GST_OBJECT_UNLOCK (qtdemux); @@ -7323,14 +7323,15 @@ too_many_streams: } } -static void +static GstFlowReturn qtdemux_expose_streams (GstQTDemux * qtdemux) { gint i; + GstFlowReturn ret = GST_FLOW_OK; GST_DEBUG_OBJECT (qtdemux, "exposing streams"); - for (i = 0; i < qtdemux->n_streams; i++) { + for (i = 0; ret == GST_FLOW_OK && i < qtdemux->n_streams; i++) { QtDemuxStream *stream = qtdemux->streams[i]; guint32 sample_num = 0; guint samples = 20; @@ -7344,7 +7345,7 @@ qtdemux_expose_streams (GstQTDemux * qtdemux) /* need all moov samples first */ GST_OBJECT_LOCK (qtdemux); while (stream->n_samples == 0) - if (!qtdemux_add_fragmented_samples (qtdemux)) + if ((ret = qtdemux_add_fragmented_samples (qtdemux)) != GST_FLOW_OK) break; GST_OBJECT_UNLOCK (qtdemux); } else { @@ -7352,6 +7353,10 @@ qtdemux_expose_streams (GstQTDemux * qtdemux) qtdemux->moof_offset = 0; } + /* prepare braking */ + if (ret != GST_FLOW_ERROR) + ret = GST_FLOW_OK; + /* in pull mode, we should have parsed some sample info by now; * and quite some code will not handle no samples. * in push mode, we'll just have to deal with it */ @@ -7411,6 +7416,8 @@ qtdemux_expose_streams (GstQTDemux * qtdemux) gst_element_post_message (GST_ELEMENT_CAST (qtdemux), m); qtdemux->posted_redirect = TRUE; } + + return ret; } /* check if major or compatible brand is 3GP */ From c9adc3750b1ffb99d00e2f76a497b01b21c85c5f Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Thu, 10 Feb 2011 23:52:51 +0000 Subject: [PATCH 297/351] doap: update mailing list location --- gst-plugins-good.doap | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/gst-plugins-good.doap b/gst-plugins-good.doap index 5345f171f1..23dbfac28d 100644 --- a/gst-plugins-good.doap +++ b/gst-plugins-good.doap @@ -20,7 +20,7 @@ the plug-in code, LGPL or LGPL-compatible for the supporting library). - + C From 6f0a4fa85bd1983113050d397288ca4419e392fb Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Mon, 14 Feb 2011 00:38:45 +0000 Subject: [PATCH 298/351] qtdemux: refactor iTunes tag parsing a bit --- gst/qtdemux/qtdemux.c | 45 ++++++++++++++++++++++++++----------------- 1 file changed, 27 insertions(+), 18 deletions(-) diff --git a/gst/qtdemux/qtdemux.c b/gst/qtdemux/qtdemux.c index 96e320d775..80aa6e3701 100644 --- a/gst/qtdemux/qtdemux.c +++ b/gst/qtdemux/qtdemux.c @@ -8021,26 +8021,35 @@ qtdemux_tag_add_revdns (GstQTDemux * demux, const char *tag, datatype = QT_UINT32 (((gchar *) data->data) + 8) & 0xFFFFFF; if (strncmp (meanstr, "com.apple.iTunes", meansize - 12) == 0) { - if (strncmp (namestr, "replaygain_track_gain", namesize - 12) == 0) { - qtdemux_add_double_tag_from_str (demux, - GST_TAG_TRACK_GAIN, ((guint8 *) data->data) + 16, datasize - 16); + static const struct + { + const gchar name[24]; + const gchar tag[24]; + } tags[] = { + { + "replaygain_track_gain", GST_TAG_TRACK_GAIN}, { + "replaygain_track_peak", GST_TAG_TRACK_PEAK}, { + "replaygain_album_gain", GST_TAG_ALBUM_GAIN}, { + "replaygain_album_peak", GST_TAG_ALBUM_PEAK} + }; + int i; - } else if (strncmp (namestr, "replaygain_track_peak", namesize - 12) == 0) { - qtdemux_add_double_tag_from_str (demux, - GST_TAG_TRACK_PEAK, ((guint8 *) data->data) + 16, datasize - 16); - - } else if (strncmp (namestr, "replaygain_album_gain", namesize - 12) == 0) { - qtdemux_add_double_tag_from_str (demux, - GST_TAG_ALBUM_GAIN, ((guint8 *) data->data) + 16, datasize - 16); - - } else if (strncmp (namestr, "replaygain_album_peak", namesize - 12) == 0) { - qtdemux_add_double_tag_from_str (demux, - GST_TAG_ALBUM_PEAK, ((guint8 *) data->data) + 16, datasize - 16); - - } else { - goto unknown_tag; + for (i = 0; i < G_N_ELEMENTS (tags); ++i) { + if (!g_ascii_strncasecmp (tags[i].name, namestr, namesize - 12)) { + switch (gst_tag_get_type (tags[i].tag)) { + case G_TYPE_DOUBLE: + qtdemux_add_double_tag_from_str (demux, tags[i].tag, + ((guint8 *) data->data) + 16, datasize - 16); + break; + default: + /* not reached */ + break; + } + break; + } } - + if (i == G_N_ELEMENTS (tags)) + goto unknown_tag; } else { goto unknown_tag; } From aa7e638d8ede9315c82d03a6fa1d12cd09ab5f2f Mon Sep 17 00:00:00 2001 From: Heath Nielson Date: Mon, 14 Feb 2011 00:49:00 +0000 Subject: [PATCH 299/351] qtdemux: extract MusicBrainz tags Extract MusicBrainz tags added by MusicBrainz's Picard tagger application. These tags (esp. the album id) are helpful for rhythmbox et.al. to automatically downloads cover art. https://bugzilla.gnome.org/show_bug.cgi?id=642205 --- gst/qtdemux/qtdemux.c | 15 ++++++++++++--- 1 file changed, 12 insertions(+), 3 deletions(-) diff --git a/gst/qtdemux/qtdemux.c b/gst/qtdemux/qtdemux.c index 80aa6e3701..587eb613e7 100644 --- a/gst/qtdemux/qtdemux.c +++ b/gst/qtdemux/qtdemux.c @@ -466,6 +466,8 @@ gst_qtdemux_class_init (GstQTDemuxClass * klass) gstelement_class->set_index = GST_DEBUG_FUNCPTR (gst_qtdemux_set_index); gstelement_class->get_index = GST_DEBUG_FUNCPTR (gst_qtdemux_get_index); + + gst_tag_register_musicbrainz_tags (); } static void @@ -8023,14 +8025,18 @@ qtdemux_tag_add_revdns (GstQTDemux * demux, const char *tag, if (strncmp (meanstr, "com.apple.iTunes", meansize - 12) == 0) { static const struct { - const gchar name[24]; - const gchar tag[24]; + const gchar name[28]; + const gchar tag[28]; } tags[] = { { "replaygain_track_gain", GST_TAG_TRACK_GAIN}, { "replaygain_track_peak", GST_TAG_TRACK_PEAK}, { "replaygain_album_gain", GST_TAG_ALBUM_GAIN}, { - "replaygain_album_peak", GST_TAG_ALBUM_PEAK} + "replaygain_album_peak", GST_TAG_ALBUM_PEAK}, { + "MusicBrainz Track Id", GST_TAG_MUSICBRAINZ_TRACKID}, { + "MusicBrainz Artist Id", GST_TAG_MUSICBRAINZ_ARTISTID}, { + "MusicBrainz Album Id", GST_TAG_MUSICBRAINZ_ALBUMID}, { + "MusicBrainz Album Artist Id", GST_TAG_MUSICBRAINZ_ALBUMARTISTID} }; int i; @@ -8041,6 +8047,9 @@ qtdemux_tag_add_revdns (GstQTDemux * demux, const char *tag, qtdemux_add_double_tag_from_str (demux, tags[i].tag, ((guint8 *) data->data) + 16, datasize - 16); break; + case G_TYPE_STRING: + qtdemux_tag_add_str (demux, tags[i].tag, NULL, node); + break; default: /* not reached */ break; From 3cca27ced16fbcc2a93653b9db0aeffadbe275b1 Mon Sep 17 00:00:00 2001 From: Miguel Angel Cabrera Moya Date: Mon, 7 Feb 2011 23:32:53 +0100 Subject: [PATCH 300/351] rtspsrc: fix minor leaks when handling server requests. https://bugzilla.gnome.org/show_bug.cgi?id=640163 --- gst/rtsp/gstrtspsrc.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c index 89b11d90a0..980fd026f9 100644 --- a/gst/rtsp/gstrtspsrc.c +++ b/gst/rtsp/gstrtspsrc.c @@ -3374,9 +3374,10 @@ gst_rtspsrc_handle_request (GstRTSPSrc * src, GstRTSPConnection * conn, gst_rtsp_message_dump (&response); res = gst_rtspsrc_connection_send (src, conn, &response, NULL); - gst_rtsp_message_unset (&response); if (res < 0) goto send_error; + + gst_rtsp_message_unset (&response); } else if (res == GST_RTSP_EEOF) return res; @@ -3385,6 +3386,7 @@ gst_rtspsrc_handle_request (GstRTSPSrc * src, GstRTSPConnection * conn, /* ERRORS */ send_error: { + gst_rtsp_message_unset (&response); return res; } } From 663852d8539179cf7bee6bdaa181583fa6f8eb7a Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Mon, 14 Feb 2011 12:53:27 +0200 Subject: [PATCH 301/351] Automatic update of common submodule From f94d739 to 1de7f6a --- common | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/common b/common index f94d739915..1de7f6ab2d 160000 --- a/common +++ b/common @@ -1 +1 @@ -Subproject commit f94d73991563ea2dcae2218b4847e32998f06816 +Subproject commit 1de7f6ab2d4bc1af69f06079cf0f4e2cbbfdc178 From d203b1ef5040a180891f0ca5ee00c57d33865300 Mon Sep 17 00:00:00 2001 From: Tuukka Pasanen Date: Mon, 14 Feb 2011 15:21:29 +0200 Subject: [PATCH 302/351] taginject: resend tags when they are changed Allow setting new tags on the property while running and send them. Fixes #640249 --- gst/debugutils/gsttaginject.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/gst/debugutils/gsttaginject.c b/gst/debugutils/gsttaginject.c index 9197ad58a4..a5e78fe9aa 100644 --- a/gst/debugutils/gsttaginject.c +++ b/gst/debugutils/gsttaginject.c @@ -168,6 +168,9 @@ gst_tag_inject_set_property (GObject * object, guint prop_id, if (!(self->tags = gst_structure_from_string (structure, NULL))) { GST_WARNING ("unparsable taglist = '%s'", structure); } + + /* make sure that tags will be send */ + self->tags_sent = FALSE; g_free (structure); break; } From 4279aa6a686524f3798615fffe8259ef864a0b59 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Mon, 14 Feb 2011 16:46:46 +0100 Subject: [PATCH 303/351] theorapay: handle 0 sized packets Handle 0 sized packets (repeat frame) in the payloader and depayloader. Fixes #641827 --- gst/rtp/gstrtptheoradepay.c | 6 +++--- gst/rtp/gstrtptheorapay.c | 16 ++++++++++------ 2 files changed, 13 insertions(+), 9 deletions(-) diff --git a/gst/rtp/gstrtptheoradepay.c b/gst/rtp/gstrtptheoradepay.c index a21cc6c6df..c1a576bc5e 100644 --- a/gst/rtp/gstrtptheoradepay.c +++ b/gst/rtp/gstrtptheoradepay.c @@ -501,7 +501,7 @@ gst_rtp_theora_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf) to_free = payload; } - GST_DEBUG_OBJECT (depayload, "assemble done"); + GST_DEBUG_OBJECT (depayload, "assemble done, payload_len %d", payload_len); /* we not assembling anymore now */ rtptheoradepay->assembling = FALSE; @@ -524,7 +524,7 @@ gst_rtp_theora_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf) */ timestamp = gst_rtp_buffer_get_timestamp (buf); - while (payload_len > 2) { + while (payload_len >= 2) { guint16 length; length = GST_READ_UINT16_BE (payload); @@ -559,7 +559,7 @@ gst_rtp_theora_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf) memcpy (GST_BUFFER_DATA (outbuf), payload, length); } - if ((payload[0] & 0xC0) == 0x0) + if (payload_len > 0 && (payload[0] & 0xC0) == 0x0) rtptheoradepay->needs_keyframe = FALSE; payload += length; diff --git a/gst/rtp/gstrtptheorapay.c b/gst/rtp/gstrtptheorapay.c index 7782853508..0092fbc548 100644 --- a/gst/rtp/gstrtptheorapay.c +++ b/gst/rtp/gstrtptheorapay.c @@ -563,7 +563,7 @@ gst_rtp_theora_pay_payload_buffer (GstRtpTheoraPay * rtptheorapay, guint8 TDT, /* put buffer in packet, it either fits completely or needs to be fragmented * over multiple RTP packets. */ - while (size) { + do { plen = MIN (rtptheorapay->payload_left - 2, size); GST_DEBUG_OBJECT (rtptheorapay, "append %u bytes", plen); @@ -571,7 +571,8 @@ gst_rtp_theora_pay_payload_buffer (GstRtpTheoraPay * rtptheorapay, guint8 TDT, /* data is copied in the payload with a 2 byte length header */ ppos[0] = ((plen - not_in_length) >> 8) & 0xff; ppos[1] = ((plen - not_in_length) & 0xff); - memcpy (&ppos[2], data, plen); + if (plen) + memcpy (&ppos[2], data, plen); /* only first (only) configuration cuts length field */ /* NOTE: spec (if any) is not clear on this ... */ @@ -617,7 +618,7 @@ gst_rtp_theora_pay_payload_buffer (GstRtpTheoraPay * rtptheorapay, guint8 TDT, if (duration != GST_CLOCK_TIME_NONE) rtptheorapay->payload_duration += duration; } - } + } while (size); return ret; } @@ -644,11 +645,14 @@ gst_rtp_theora_pay_handle_buffer (GstBaseRTPPayload * basepayload, GST_DEBUG_OBJECT (rtptheorapay, "size %u, duration %" GST_TIME_FORMAT, size, GST_TIME_ARGS (duration)); - if (G_UNLIKELY (size < 1 || size > 0xffff)) + if (G_UNLIKELY (size > 0xffff)) goto wrong_size; /* find packet type */ - if (data[0] & 0x80) { + if (size == 0) { + TDT = 0; + keyframe = FALSE; + } else if (data[0] & 0x80) { /* header */ if (data[0] == 0x80) { /* identification, we need to parse this in order to get the clock rate. @@ -743,7 +747,7 @@ done: wrong_size: { GST_ELEMENT_WARNING (rtptheorapay, STREAM, DECODE, - ("Invalid packet size (1 < %d <= 0xffff)", size), (NULL)); + ("Invalid packet size (%d <= 0xffff)", size), (NULL)); gst_buffer_unref (buffer); return GST_FLOW_OK; } From d466f07def637e6f6be8e1fb5ff2120ed5096ff9 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Mon, 14 Feb 2011 17:49:54 +0100 Subject: [PATCH 304/351] matroskademux: avoid sorting NULL array of cluster positions --- gst/matroska/matroska-demux.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/gst/matroska/matroska-demux.c b/gst/matroska/matroska-demux.c index 76431193c2..52f14679ee 100644 --- a/gst/matroska/matroska-demux.c +++ b/gst/matroska/matroska-demux.c @@ -5482,7 +5482,8 @@ gst_matroska_demux_parse_contents (GstMatroskaDemux * demux, GstEbmlRead * ebml) DEBUG_ELEMENT_STOP (demux, ebml, "SeekHead", ret); /* Sort clusters by position for easier searching */ - g_array_sort (demux->clusters, (GCompareFunc) gst_matroska_cluster_compare); + if (demux->clusters) + g_array_sort (demux->clusters, (GCompareFunc) gst_matroska_cluster_compare); return ret; } From b2fb4edcb72474aed9be5846450deb4be6b33cd5 Mon Sep 17 00:00:00 2001 From: Andoni Morales Alastruey Date: Mon, 14 Feb 2011 20:20:08 +0100 Subject: [PATCH 305/351] souphttpsrc: Add URI query handler Fixes bug #642337. --- ext/soup/gstsouphttpsrc.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) diff --git a/ext/soup/gstsouphttpsrc.c b/ext/soup/gstsouphttpsrc.c index 4d77984564..819fdecdec 100644 --- a/ext/soup/gstsouphttpsrc.c +++ b/ext/soup/gstsouphttpsrc.c @@ -135,6 +135,7 @@ static gboolean gst_soup_http_src_get_size (GstBaseSrc * bsrc, guint64 * size); static gboolean gst_soup_http_src_is_seekable (GstBaseSrc * bsrc); static gboolean gst_soup_http_src_do_seek (GstBaseSrc * bsrc, GstSegment * segment); +static gboolean gst_soup_http_src_query (GstBaseSrc * bsrc, GstQuery * query); static gboolean gst_soup_http_src_unlock (GstBaseSrc * bsrc); static gboolean gst_soup_http_src_unlock_stop (GstBaseSrc * bsrc); static gboolean gst_soup_http_src_set_location (GstSoupHTTPSrc * src, @@ -307,6 +308,7 @@ gst_soup_http_src_class_init (GstSoupHTTPSrcClass * klass) gstbasesrc_class->is_seekable = GST_DEBUG_FUNCPTR (gst_soup_http_src_is_seekable); gstbasesrc_class->do_seek = GST_DEBUG_FUNCPTR (gst_soup_http_src_do_seek); + gstbasesrc_class->query = GST_DEBUG_FUNCPTR (gst_soup_http_src_query); gstpushsrc_class->create = GST_DEBUG_FUNCPTR (gst_soup_http_src_create); } @@ -1394,6 +1396,28 @@ gst_soup_http_src_do_seek (GstBaseSrc * bsrc, GstSegment * segment) return TRUE; } +static gboolean +gst_soup_http_src_query (GstBaseSrc * bsrc, GstQuery * query) +{ + GstSoupHTTPSrc *src = GST_SOUP_HTTP_SRC (bsrc); + gboolean ret; + + switch (GST_QUERY_TYPE (query)) { + case GST_QUERY_URI: + gst_query_set_uri (query, src->location); + ret = TRUE; + break; + default: + ret = FALSE; + break; + } + + if (!ret) + ret = GST_BASE_SRC_CLASS (parent_class)->query (bsrc, query); + + return ret; +} + static gboolean gst_soup_http_src_set_location (GstSoupHTTPSrc * src, const gchar * uri) { From cfeb84b82171bab4bb845d1d9d5171458d06a50e Mon Sep 17 00:00:00 2001 From: Andy Wingo Date: Tue, 15 Feb 2011 16:30:20 +0100 Subject: [PATCH 306/351] plug qtdemux refcount leaks * gst/qtdemux/qtdemux.c (gst_qtdemux_src_convert): Unref the qtdemux; we weren't doing so before. (gst_qtdemux_handle_src_event, gst_qtdemux_chain): Fix some error cases which would leak a ref to the qtdemux. --- gst/qtdemux/qtdemux.c | 19 ++++++++++++++----- 1 file changed, 14 insertions(+), 5 deletions(-) diff --git a/gst/qtdemux/qtdemux.c b/gst/qtdemux/qtdemux.c index 587eb613e7..ceafefadf7 100644 --- a/gst/qtdemux/qtdemux.c +++ b/gst/qtdemux/qtdemux.c @@ -580,8 +580,10 @@ gst_qtdemux_src_convert (GstPad * pad, GstFormat src_format, gint64 src_value, GstQTDemux *qtdemux = GST_QTDEMUX (gst_pad_get_parent (pad)); gint32 index; - if (stream->subtype != FOURCC_vide) - return FALSE; + if (stream->subtype != FOURCC_vide) { + res = FALSE; + goto done; + } switch (src_format) { case GST_FORMAT_TIME: @@ -630,6 +632,9 @@ gst_qtdemux_src_convert (GstPad * pad, GstFormat src_format, gint64 src_value, res = FALSE; } +done: + gst_object_unref (qtdemux); + return res; } #endif @@ -1492,6 +1497,7 @@ gst_qtdemux_handle_src_event (GstPad * pad, GstEvent * event) gst_object_unref (qtdemux); +done: return res; /* ERRORS */ @@ -1499,7 +1505,8 @@ index_failed: { GST_ERROR_OBJECT (qtdemux, "Index failed"); gst_event_unref (event); - return FALSE; + res = FALSE; + goto done; } } @@ -4102,8 +4109,10 @@ gst_qtdemux_chain (GstPad * sinkpad, GstBuffer * inbuf) if (demux->got_moov && demux->fragmented) { GST_DEBUG_OBJECT (demux, "Parsing [moof]"); if (!qtdemux_parse_moof (demux, data, demux->neededbytes, - demux->offset, NULL)) - return GST_FLOW_ERROR; + demux->offset, NULL)) { + ret = GST_FLOW_ERROR; + goto done; + } } else { GST_DEBUG_OBJECT (demux, "Discarding [moof]"); } From 5dac2e5fde3cd5ceaec4f24d4f5d651dce9c43ad Mon Sep 17 00:00:00 2001 From: Thiago Santos Date: Tue, 15 Feb 2011 15:59:32 -0300 Subject: [PATCH 307/351] v4l2: Add PJPG mapping Adds mapping of progressive jpeg format --- sys/v4l2/gstv4l2object.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sys/v4l2/gstv4l2object.c b/sys/v4l2/gstv4l2object.c index 5c9022c18b..6cf9036e2e 100644 --- a/sys/v4l2/gstv4l2object.c +++ b/sys/v4l2/gstv4l2object.c @@ -774,6 +774,7 @@ static const GstV4L2FormatDesc gst_v4l2_formats[] = { /* compressed formats */ {V4L2_PIX_FMT_MJPEG, TRUE}, {V4L2_PIX_FMT_JPEG, TRUE}, + {V4L2_PIX_FMT_PJPG, TRUE}, {V4L2_PIX_FMT_DV, TRUE}, {V4L2_PIX_FMT_MPEG, FALSE}, @@ -813,10 +814,13 @@ gst_v4l2_object_get_format_from_fourcc (GstV4l2Object * v4l2object, if (fmt->pixelformat == fourcc) return fmt; /* special case for jpeg */ - if ((fmt->pixelformat == V4L2_PIX_FMT_MJPEG && fourcc == V4L2_PIX_FMT_JPEG) - || (fmt->pixelformat == V4L2_PIX_FMT_JPEG - && fourcc == V4L2_PIX_FMT_MJPEG)) { - return fmt; + if (fmt->pixelformat == V4L2_PIX_FMT_MJPEG || + fmt->pixelformat == V4L2_PIX_FMT_JPEG || + fmt->pixelformat == V4L2_PIX_FMT_PJPG) { + if (fourcc == V4L2_PIX_FMT_JPEG + || fourcc == V4L2_PIX_FMT_MJPEG || fourcc == V4L2_PIX_FMT_PJPG) { + return fmt; + } } walk = g_slist_next (walk); } @@ -854,6 +858,7 @@ gst_v4l2_object_format_get_rank (const struct v4l2_fmtdesc *fmt) switch (fourcc) { case V4L2_PIX_FMT_MJPEG: + case V4L2_PIX_FMT_PJPG: rank = JPEG_BASE_RANK; break; case V4L2_PIX_FMT_JPEG: @@ -1072,6 +1077,7 @@ gst_v4l2_object_v4l2fourcc_to_structure (guint32 fourcc) switch (fourcc) { case V4L2_PIX_FMT_MJPEG: /* Motion-JPEG */ + case V4L2_PIX_FMT_PJPG: /* Progressive-JPEG */ case V4L2_PIX_FMT_JPEG: /* JFIF JPEG */ structure = gst_structure_new ("image/jpeg", NULL); break; From 5dea7172c571f611e31a2ef19247cee1d0825cde Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Sat, 3 Feb 2007 23:35:26 +0000 Subject: [PATCH 308/351] [MOVED FROM BAD 1/5] Fix up to use the newly ported (actually working) GstAudioFilter. Original commit message from CVS: * configure.ac: * gst/equalizer/Makefile.am: * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_base_init), (gst_iir_equalizer_class_init), (gst_iir_equalizer_init), (setup_filter), (gst_iir_equalizer_compute_frequencies), (gst_iir_equalizer_set_property), (gst_iir_equalizer_get_property), (gst_iir_equalizer_transform_ip), (gst_iir_equalizer_setup), (plugin_init): * gst/equalizer/gstiirequalizer.h: Fix up to use the newly ported (actually working) GstAudioFilter. Bump core/base requirements to CVS for this. * tests/icles/.cvsignore: * tests/icles/Makefile.am: * tests/icles/equalizer-test.c: (check_bus), (equalizer_set_band_value), (equalizer_set_all_band_values), (equalizer_set_band_value_and_wait), (equalizer_set_all_band_values_and_wait), (do_slider_fiddling), (main): Add brain-dead interactive test for equalizer. --- tests/icles/equalizer-test.c | 293 +++++++++++++++++++++++++++++++++++ 1 file changed, 293 insertions(+) create mode 100644 tests/icles/equalizer-test.c diff --git a/tests/icles/equalizer-test.c b/tests/icles/equalizer-test.c new file mode 100644 index 0000000000..c76039824a --- /dev/null +++ b/tests/icles/equalizer-test.c @@ -0,0 +1,293 @@ +/* GStreamer test for the equalizer element + * Copyright (C) 2007 Tim-Philipp Müller + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifdef HAVE_CONFIG_H +# include "config.h" +#endif + +#include + +#include +#include + +GST_DEBUG_CATEGORY_STATIC (equalizer_test_debug); +#define GST_CAT_DEFAULT equalizer_test_debug + +static GstBus *pipeline_bus; + +static gboolean +check_bus (GstClockTime max_wait_time) +{ + GstMessage *msg; + + msg = gst_bus_poll (pipeline_bus, GST_MESSAGE_ERROR | GST_MESSAGE_EOS, + max_wait_time); + + if (msg == NULL) + return FALSE; + + if (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_ERROR) { + GError *err = NULL; + gchar *debug = NULL; + + g_assert (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_ERROR); + gst_message_parse_error (msg, &err, &debug); + GST_ERROR ("ERROR: %s [%s]", err->message, debug); + g_print ("\n===========> ERROR: %s\n%s\n\n", err->message, debug); + g_error_free (err); + g_free (debug); + } + + if (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_EOS) { + g_print ("\n === EOS ===\n\n"); + } + + gst_message_unref (msg); + return TRUE; +} + +static void +equalizer_set_band_value (GstElement * eq, GValueArray * arr, guint band, + gdouble val) +{ + g_value_set_double (g_value_array_get_nth (arr, band), val); + g_object_set (eq, "band-values", arr, NULL); + g_print ("Band %2d: %.2f\n", band, val); +} + +static void +equalizer_set_all_band_values (GstElement * eq, GValueArray * arr, gdouble val) +{ + gint i; + + for (i = 0; i < arr->n_values; ++i) { + g_value_set_double (g_value_array_get_nth (arr, i), val); + } + g_object_set (eq, "band-values", arr, NULL); + g_print ("All bands: %.2f\n", val); +} + +static gboolean +equalizer_set_band_value_and_wait (GstElement * eq, GValueArray * arr, + guint band, gdouble val) +{ + equalizer_set_band_value (eq, arr, band, val); + return check_bus (100 * GST_MSECOND); +} + +static gboolean +equalizer_set_all_band_values_and_wait (GstElement * eq, GValueArray * arr, + gdouble val) +{ + equalizer_set_all_band_values (eq, arr, val); + return check_bus (100 * GST_MSECOND); +} + +static void +do_slider_fiddling (GstElement * playbin, GstElement * eq) +{ + GValueArray *arr; + gboolean stop; + guint num_bands, i; + gdouble d, step = 0.2; + + stop = FALSE; + + g_object_get (eq, "num-bands", &num_bands, NULL); + + g_print ("%u bands.\n", num_bands); + + arr = g_value_array_new (num_bands); + for (i = 0; i < num_bands; ++i) { + GValue val = { 0, }; + + g_value_init (&val, G_TYPE_DOUBLE); + g_value_set_double (&val, 0.0); + g_value_array_append (arr, &val); + } + + g_object_set (eq, "band-values", arr, NULL); + + while (!stop) { + for (i = 0; !stop && i < num_bands; ++i) { + d = 0.0; + while (!stop && d <= 1.0) { + stop = equalizer_set_band_value_and_wait (eq, arr, i, d); + d += step; + } + d = 1.0; + while (!stop && d >= -1.0) { + stop = equalizer_set_band_value_and_wait (eq, arr, i, d); + d -= step; + } + d = -1.0; + while (!stop && d <= 0.0) { + stop = equalizer_set_band_value_and_wait (eq, arr, i, d); + d += step; + } + } + + d = 0.0; + while (!stop && d <= 1.0) { + stop = equalizer_set_all_band_values_and_wait (eq, arr, d); + d += step; + } + d = 1.0; + while (!stop && d >= -1.0) { + stop = equalizer_set_all_band_values_and_wait (eq, arr, d); + d -= step; + } + d = -1.0; + while (!stop && d <= 0.0) { + stop = equalizer_set_all_band_values_and_wait (eq, arr, d); + d += step; + } + } + + g_value_array_free (arr); +} + +int +main (int argc, char **argv) +{ + gchar *opt_audiosink_str = NULL; + gchar **filenames = NULL; + const GOptionEntry test_goptions[] = { + {"audiosink", '\0', 0, G_OPTION_ARG_STRING, &opt_audiosink_str, + "audiosink to use (default: autoaudiosink)", NULL}, + {G_OPTION_REMAINING, 0, 0, G_OPTION_ARG_FILENAME_ARRAY, &filenames, NULL}, + {NULL, '\0', 0, 0, NULL, NULL, NULL} + }; + GOptionContext *ctx; + GError *opt_err = NULL; + + GstStateChangeReturn ret; + GstElement *playbin, *sink, *bin, *eq, *auconv; + GstPad *eq_sinkpad; + gchar *uri; + + if (!g_thread_supported ()) + g_thread_init (NULL); + + GST_DEBUG_CATEGORY_INIT (equalizer_test_debug, "equalizertest", 0, "eqtest"); + + /* command line option parsing */ + ctx = g_option_context_new ("FILENAME"); + g_option_context_add_group (ctx, gst_init_get_option_group ()); + g_option_context_add_main_entries (ctx, test_goptions, NULL); + + if (!g_option_context_parse (ctx, &argc, &argv, &opt_err)) { + g_error ("Error parsing command line options: %s", opt_err->message); + return -1; + } + + if (filenames == NULL || *filenames == NULL) { + g_printerr ("Please specify a file to play back\n"); + return -1; + } + + playbin = gst_element_factory_make ("playbin", "playbin"); + if (playbin == NULL) { + g_error ("Couldn't create 'playbin' element"); + return -1; + } + + if (opt_audiosink_str) { + g_print ("Trying audiosink '%s' ...", opt_audiosink_str); + sink = gst_element_factory_make (opt_audiosink_str, "sink"); + g_print ("%s\n", (sink) ? "ok" : "element couldn't be created"); + } else { + sink = NULL; + } + if (sink == NULL) { + g_print ("Trying audiosink '%s' ...", "autoaudiosink"); + sink = gst_element_factory_make ("autoaudiosink", "sink"); + g_print ("%s\n", (sink) ? "ok" : "element couldn't be created"); + } + if (sink == NULL) { + g_print ("Trying audiosink '%s' ...", "alsasink"); + sink = gst_element_factory_make ("alsasink", "sink"); + g_print ("%s\n", (sink) ? "ok" : "element couldn't be created"); + } + if (sink == NULL) { + g_print ("Trying audiosink '%s' ...", "osssink"); + sink = gst_element_factory_make ("osssink", "sink"); + g_print ("%s\n", (sink) ? "ok" : "element couldn't be created"); + } + + g_assert (sink != NULL); + + bin = gst_bin_new ("ausinkbin"); + g_assert (bin != NULL); + + eq = gst_element_factory_make ("equalizer", "equalizer"); + g_assert (eq != NULL); + + auconv = gst_element_factory_make ("audioconvert", "eqauconv"); + g_assert (auconv != NULL); + + gst_bin_add_many (GST_BIN (bin), eq, auconv, sink, NULL); + + if (!gst_element_link (eq, auconv)) + g_error ("Failed to link equalizer to audioconvert"); + + if (!gst_element_link (auconv, sink)) + g_error ("Failed to link audioconvert to audio sink"); + + + eq_sinkpad = gst_element_get_static_pad (eq, "sink"); + g_assert (eq_sinkpad != NULL); + + gst_element_add_pad (bin, gst_ghost_pad_new (NULL, eq_sinkpad)); + gst_object_unref (eq_sinkpad); + + g_object_set (playbin, "audio-sink", bin, NULL); + + /* won't work: uri = gst_uri_construct ("file", filenames[0]); */ + uri = g_strdup_printf ("file://%s", filenames[0]); + g_object_set (playbin, "uri", uri, NULL); + g_free (uri); + + pipeline_bus = GST_ELEMENT_BUS (playbin); + + ret = gst_element_set_state (playbin, GST_STATE_PLAYING); + if (ret == GST_STATE_CHANGE_FAILURE) { + g_printerr ("Failed to set playbin to PLAYING\n"); + check_bus (1 * GST_SECOND); + return -1; + } + + ret = gst_element_get_state (playbin, NULL, NULL, 5 * GST_SECOND); + if (ret == GST_STATE_CHANGE_ASYNC) { + g_printerr ("Failed to go to PLAYING in 5 seconds, bailing out\n"); + return -1; + } else if (ret != GST_STATE_CHANGE_SUCCESS) { + g_printerr ("State change to PLAYING failed\n"); + check_bus (1 * GST_SECOND); + return -1; + } + + g_print ("Playing ...\n"); + do_slider_fiddling (playbin, eq); + + gst_element_set_state (playbin, GST_STATE_NULL); + gst_object_unref (playbin); + + return 0; +} From 85f78281f303dae6320ea8ff718fc8b14ffd1e2d Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Wed, 14 Mar 2007 16:33:03 +0000 Subject: [PATCH 309/351] [MOVED FROM BAD 2/5] tests/icles/equalizer-test.c: Port the example to new equalizer api. Original commit message from CVS: * tests/icles/equalizer-test.c: (equalizer_set_band_value), (equalizer_set_all_band_values), (equalizer_set_band_value_and_wait), (equalizer_set_all_band_values_and_wait), (do_slider_fiddling), (main): Port the example to new equalizer api. --- tests/icles/equalizer-test.c | 63 ++++++++++++++++-------------------- 1 file changed, 27 insertions(+), 36 deletions(-) diff --git a/tests/icles/equalizer-test.c b/tests/icles/equalizer-test.c index c76039824a..8a70699395 100644 --- a/tests/icles/equalizer-test.c +++ b/tests/icles/equalizer-test.c @@ -62,47 +62,52 @@ check_bus (GstClockTime max_wait_time) return TRUE; } +// fix below + static void -equalizer_set_band_value (GstElement * eq, GValueArray * arr, guint band, - gdouble val) +equalizer_set_band_value (GstElement * eq, guint band, gdouble val) { - g_value_set_double (g_value_array_get_nth (arr, band), val); - g_object_set (eq, "band-values", arr, NULL); + GstObject *child; + + child = gst_child_proxy_get_child_by_index (GST_CHILD_PROXY (eq), band); + g_object_set (child, "gain", val, NULL); + gst_object_unref (child); g_print ("Band %2d: %.2f\n", band, val); } static void -equalizer_set_all_band_values (GstElement * eq, GValueArray * arr, gdouble val) +equalizer_set_all_band_values (GstElement * eq, guint num, gdouble val) { gint i; + GstObject *child; - for (i = 0; i < arr->n_values; ++i) { - g_value_set_double (g_value_array_get_nth (arr, i), val); + for (i = 0; i < num; i++) { + child = gst_child_proxy_get_child_by_index (GST_CHILD_PROXY (eq), i); + g_object_set (child, "gain", val, NULL); + gst_object_unref (child); } - g_object_set (eq, "band-values", arr, NULL); g_print ("All bands: %.2f\n", val); } +// fix above + static gboolean -equalizer_set_band_value_and_wait (GstElement * eq, GValueArray * arr, - guint band, gdouble val) +equalizer_set_band_value_and_wait (GstElement * eq, guint band, gdouble val) { - equalizer_set_band_value (eq, arr, band, val); + equalizer_set_band_value (eq, band, val); return check_bus (100 * GST_MSECOND); } static gboolean -equalizer_set_all_band_values_and_wait (GstElement * eq, GValueArray * arr, - gdouble val) +equalizer_set_all_band_values_and_wait (GstElement * eq, guint num, gdouble val) { - equalizer_set_all_band_values (eq, arr, val); + equalizer_set_all_band_values (eq, num, val); return check_bus (100 * GST_MSECOND); } static void do_slider_fiddling (GstElement * playbin, GstElement * eq) { - GValueArray *arr; gboolean stop; guint num_bands, i; gdouble d, step = 0.2; @@ -113,54 +118,41 @@ do_slider_fiddling (GstElement * playbin, GstElement * eq) g_print ("%u bands.\n", num_bands); - arr = g_value_array_new (num_bands); - for (i = 0; i < num_bands; ++i) { - GValue val = { 0, }; - - g_value_init (&val, G_TYPE_DOUBLE); - g_value_set_double (&val, 0.0); - g_value_array_append (arr, &val); - } - - g_object_set (eq, "band-values", arr, NULL); - while (!stop) { for (i = 0; !stop && i < num_bands; ++i) { d = 0.0; while (!stop && d <= 1.0) { - stop = equalizer_set_band_value_and_wait (eq, arr, i, d); + stop = equalizer_set_band_value_and_wait (eq, i, d); d += step; } d = 1.0; while (!stop && d >= -1.0) { - stop = equalizer_set_band_value_and_wait (eq, arr, i, d); + stop = equalizer_set_band_value_and_wait (eq, i, d); d -= step; } d = -1.0; while (!stop && d <= 0.0) { - stop = equalizer_set_band_value_and_wait (eq, arr, i, d); + stop = equalizer_set_band_value_and_wait (eq, i, d); d += step; } } d = 0.0; while (!stop && d <= 1.0) { - stop = equalizer_set_all_band_values_and_wait (eq, arr, d); + stop = equalizer_set_all_band_values_and_wait (eq, num_bands, d); d += step; } d = 1.0; while (!stop && d >= -1.0) { - stop = equalizer_set_all_band_values_and_wait (eq, arr, d); + stop = equalizer_set_all_band_values_and_wait (eq, num_bands, d); d -= step; } d = -1.0; while (!stop && d <= 0.0) { - stop = equalizer_set_all_band_values_and_wait (eq, arr, d); + stop = equalizer_set_all_band_values_and_wait (eq, num_bands, d); d += step; } } - - g_value_array_free (arr); } int @@ -236,7 +228,7 @@ main (int argc, char **argv) bin = gst_bin_new ("ausinkbin"); g_assert (bin != NULL); - eq = gst_element_factory_make ("equalizer", "equalizer"); + eq = gst_element_factory_make ("equalizer-nbands", "equalizer"); g_assert (eq != NULL); auconv = gst_element_factory_make ("audioconvert", "eqauconv"); @@ -250,7 +242,6 @@ main (int argc, char **argv) if (!gst_element_link (auconv, sink)) g_error ("Failed to link audioconvert to audio sink"); - eq_sinkpad = gst_element_get_static_pad (eq, "sink"); g_assert (eq_sinkpad != NULL); From 04689cdb126c95bad317d0f9e38d2bbc15614686 Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Mon, 21 May 2007 14:01:16 +0000 Subject: [PATCH 310/351] [MOVED FROM BAD 3/5] ChangeLog: ChangeLog surgery. gst/equalizer/gstiirequalizer.c (ARG_BAND_WIDTH, _do_init, ARG_GAIN, _GstIirEqualizerBa... Original commit message from CVS: * ChangeLog: ChangeLog surgery. * gst/equalizer/gstiirequalizer.c (ARG_BAND_WIDTH, _do_init, ARG_GAIN, _GstIirEqualizerBand, object, _GstIirEqualizerBandClass, parent_class, gst_iir_equalizer_band_set_property, gst_iir_equalizer_band_class_init, gst_iir_equalizer_band_get_type, gst_iir_equalizer_child_proxy_get_child_by_index, gst_iir_equalizer_child_proxy_get_children_count, gst_iir_equalizer_child_proxy_interface_init, setup_filter, gst_iir_equalizer_compute_frequencies, plugin_init): * tests/icles/equalizer-test.c: Add fixme and comment for example. --- tests/icles/equalizer-test.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/tests/icles/equalizer-test.c b/tests/icles/equalizer-test.c index 8a70699395..d45146a6d9 100644 --- a/tests/icles/equalizer-test.c +++ b/tests/icles/equalizer-test.c @@ -17,6 +17,11 @@ * Boston, MA 02111-1307, USA. */ +/* + * Will tests the equalizer by fading all bands in and out one by one and + * finaly all together. + */ + #ifdef HAVE_CONFIG_H # include "config.h" #endif From bece468db4b11edec4dfc70a17bc82a6e2cab698 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Sebastian=20Dr=C3=B6ge?= Date: Wed, 7 Nov 2007 15:36:59 +0000 Subject: [PATCH 311/351] [MOVED FROM BAD 4/5] tests/icles/equalizer-test.c: Fix gain ranges for the latest equalizer changes. Original commit message from CVS: * tests/icles/equalizer-test.c: (do_slider_fiddling): Fix gain ranges for the latest equalizer changes. --- tests/icles/equalizer-test.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/tests/icles/equalizer-test.c b/tests/icles/equalizer-test.c index d45146a6d9..d8fd8dda39 100644 --- a/tests/icles/equalizer-test.c +++ b/tests/icles/equalizer-test.c @@ -115,7 +115,7 @@ do_slider_fiddling (GstElement * playbin, GstElement * eq) { gboolean stop; guint num_bands, i; - gdouble d, step = 0.2; + gdouble d, step = 0.5; stop = FALSE; @@ -125,34 +125,34 @@ do_slider_fiddling (GstElement * playbin, GstElement * eq) while (!stop) { for (i = 0; !stop && i < num_bands; ++i) { - d = 0.0; - while (!stop && d <= 1.0) { + d = -24.0; + while (!stop && d <= 12.0) { stop = equalizer_set_band_value_and_wait (eq, i, d); d += step; } - d = 1.0; - while (!stop && d >= -1.0) { + d = 12.0; + while (!stop && d >= -24.0) { stop = equalizer_set_band_value_and_wait (eq, i, d); d -= step; } - d = -1.0; - while (!stop && d <= 0.0) { + d = -24.0; + while (!stop && d <= 12.0) { stop = equalizer_set_band_value_and_wait (eq, i, d); d += step; } } d = 0.0; - while (!stop && d <= 1.0) { + while (!stop && d <= 12.0) { stop = equalizer_set_all_band_values_and_wait (eq, num_bands, d); d += step; } - d = 1.0; - while (!stop && d >= -1.0) { + d = 12.0; + while (!stop && d >= -24.0) { stop = equalizer_set_all_band_values_and_wait (eq, num_bands, d); d -= step; } - d = -1.0; + d = -24.0; while (!stop && d <= 0.0) { stop = equalizer_set_all_band_values_and_wait (eq, num_bands, d); d += step; From dd77fddeea4bbc549528eb28415dd7c597169513 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Sebastian=20Dr=C3=B6ge?= Date: Wed, 16 Feb 2011 15:23:50 +0100 Subject: [PATCH 312/351] [MOVED FROM BAD 5/5] equalizer-test: Initialize debug category after gst_init() to fix segfault --- tests/icles/equalizer-test.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/tests/icles/equalizer-test.c b/tests/icles/equalizer-test.c index d8fd8dda39..e8126fad3a 100644 --- a/tests/icles/equalizer-test.c +++ b/tests/icles/equalizer-test.c @@ -182,8 +182,6 @@ main (int argc, char **argv) if (!g_thread_supported ()) g_thread_init (NULL); - GST_DEBUG_CATEGORY_INIT (equalizer_test_debug, "equalizertest", 0, "eqtest"); - /* command line option parsing */ ctx = g_option_context_new ("FILENAME"); g_option_context_add_group (ctx, gst_init_get_option_group ()); @@ -194,6 +192,8 @@ main (int argc, char **argv) return -1; } + GST_DEBUG_CATEGORY_INIT (equalizer_test_debug, "equalizertest", 0, "eqtest"); + if (filenames == NULL || *filenames == NULL) { g_printerr ("Please specify a file to play back\n"); return -1; From 4f78d8570771a0cf6526ccf9f372e7fca7aa2079 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Sebastian=20Dr=C3=B6ge?= Date: Wed, 16 Feb 2011 15:28:53 +0100 Subject: [PATCH 313/351] icles: Add equalizer-test to the build system --- tests/icles/Makefile.am | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/tests/icles/Makefile.am b/tests/icles/Makefile.am index 4405ecfb0f..5b1eb22b69 100644 --- a/tests/icles/Makefile.am +++ b/tests/icles/Makefile.am @@ -39,6 +39,10 @@ else X_TESTS = endif +equalizer_test_SOURCES = equalizer-test.c +equalizer_test_CFLAGS = $(GST_CFLAGS) +equalizer_test_LDADD = $(GST_LIBS) + videocrop_test_SOURCES = videocrop-test.c videocrop_test_CFLAGS = $(GST_CFLAGS) videocrop_test_LDADD = $(GST_LIBS) @@ -51,5 +55,5 @@ videocrop2_test_SOURCES = videocrop2-test.c videocrop2_test_CFLAGS = $(GST_CFLAGS) videocrop2_test_LDADD = $(GST_LIBS) -noinst_PROGRAMS = $(GTK_TESTS) $(OSS4_TESTS) $(V4L2_TESTS) $(X_TESTS) videocrop-test videobox-test videocrop2-test +noinst_PROGRAMS = $(GTK_TESTS) $(OSS4_TESTS) $(V4L2_TESTS) $(X_TESTS) equalizer-test videocrop-test videobox-test videocrop2-test From 1e034cb01ff097eb9fb8cd55f844489e26c0ccd1 Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Wed, 16 Feb 2011 17:06:51 +0200 Subject: [PATCH 314/351] avimux: rework _request_new_pad to handle explict req-pad-names Don't ignore explicit pad-names. Rearrange the code and the error handling a bit. Add a FIXME-0.11 for the bad pad-names. --- gst/avi/gstavimux.c | 62 +++++++++++++++++++++++++++++++-------------- 1 file changed, 43 insertions(+), 19 deletions(-) diff --git a/gst/avi/gstavimux.c b/gst/avi/gstavimux.c index 6a92a2b168..d0d2f0ecc6 100644 --- a/gst/avi/gstavimux.c +++ b/gst/avi/gstavimux.c @@ -41,7 +41,7 @@ * ! 'audio/x-raw-int,rate=44100,channels=2' ! queue ! mux. \ * avimux name=mux ! filesink location=test.avi * ]| This will create an .AVI file containing an uncompressed video stream - * with a test picture and an uncompressed audio stream containing a + * with a test picture and an uncompressed audio stream containing a * test sound. * |[ * gst-launch videotestsrc num-buffers=250 \ @@ -61,6 +61,7 @@ #endif #include "gst/gst-i18n-plugin.h" +#include #include #include @@ -964,6 +965,10 @@ gst_avi_mux_request_new_pad (GstElement * element, GstPad *newpad; GstAviPad *avipad; GstElementClass *klass; + gchar *name = NULL; + const gchar *pad_name = NULL; + GstPadSetCapsFunction setcapsfunc = NULL; + gint pad_id; g_return_val_if_fail (templ != NULL, NULL); @@ -978,22 +983,23 @@ gst_avi_mux_request_new_pad (GstElement * element, klass = GST_ELEMENT_GET_CLASS (element); - if (templ == gst_element_class_get_pad_template (klass, "audio_%d")) { - gchar *name; + /* FIXME-0.11: use %d instead of %02d for pad_names */ - /* setup pad */ - name = g_strdup_printf ("audio_%02d", avimux->audio_pads); - GST_DEBUG_OBJECT (avimux, "adding new pad: %s", name); - newpad = gst_pad_new_from_template (templ, name); - g_free (name); - gst_pad_set_setcaps_function (newpad, - GST_DEBUG_FUNCPTR (gst_avi_mux_audsink_set_caps)); + if (templ == gst_element_class_get_pad_template (klass, "audio_%d")) { + /* don't mix named and unnamed pads, if the pad already exists we fail when + * trying to add it */ + if (req_name != NULL && sscanf (req_name, "audio_%02d", &pad_id) == 1) { + pad_name = req_name; + } else { + name = g_strdup_printf ("audio_%02d", avimux->audio_pads++); + pad_name = name; + } + setcapsfunc = GST_DEBUG_FUNCPTR (gst_avi_mux_audsink_set_caps); /* init pad specific data */ avipad = g_malloc0 (sizeof (GstAviAudioPad)); avipad->is_video = FALSE; avipad->hdr.type = GST_MAKE_FOURCC ('a', 'u', 'd', 's'); - avimux->audio_pads++; /* audio goes last */ avimux->sinkpads = g_slist_append (avimux->sinkpads, avipad); } else if (templ == gst_element_class_get_pad_template (klass, "video_%d")) { @@ -1001,23 +1007,27 @@ gst_avi_mux_request_new_pad (GstElement * element, * some video info goes in a single avi header -and therefore mux struct- * so video restricted to one stream */ if (avimux->video_pads > 0) - return NULL; + goto too_many_video_pads; + /* setup pad */ - GST_DEBUG_OBJECT (avimux, "adding new pad: video_00"); - newpad = gst_pad_new_from_template (templ, "video_00"); - gst_pad_set_setcaps_function (newpad, - GST_DEBUG_FUNCPTR (gst_avi_mux_vidsink_set_caps)); - avipad = g_malloc0 (sizeof (GstAviVideoPad)); + pad_name = "video_00"; + avimux->video_pads++; + setcapsfunc = GST_DEBUG_FUNCPTR (gst_avi_mux_vidsink_set_caps); /* init pad specific data */ + avipad = g_malloc0 (sizeof (GstAviVideoPad)); avipad->is_video = TRUE; avipad->hdr.type = GST_MAKE_FOURCC ('v', 'i', 'd', 's'); - avimux->video_pads++; /* video goes first */ avimux->sinkpads = g_slist_prepend (avimux->sinkpads, avipad); } else goto wrong_template; + newpad = gst_pad_new_from_template (templ, pad_name); + gst_pad_set_setcaps_function (newpad, setcapsfunc); + + g_free (name); + avipad->collect = gst_collect_pads_add_pad (avimux->collect, newpad, sizeof (GstAviCollectData)); ((GstAviCollectData *) (avipad->collect))->avipad = avipad; @@ -1028,7 +1038,10 @@ gst_avi_mux_request_new_pad (GstElement * element, gst_pad_set_event_function (newpad, GST_DEBUG_FUNCPTR (gst_avi_mux_handle_event)); - gst_element_add_pad (element, newpad); + if (!gst_element_add_pad (element, newpad)) + goto pad_add_failed; + + GST_DEBUG_OBJECT (newpad, "Added new request pad"); return newpad; @@ -1048,6 +1061,17 @@ wrong_template: g_warning ("avimux: this is not our template!\n"); return NULL; } +too_many_video_pads: + { + GST_WARNING_OBJECT (avimux, "Can only have one video stream"); + return NULL; + } +pad_add_failed: + { + GST_WARNING_OBJECT (avimux, "Adding the new pad '%s' failed", pad_name); + gst_object_unref (newpad); + return NULL; + } } static void From 8d5be2e8a483626c529ecd191c4ece966243fe2f Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Wed, 16 Feb 2011 17:09:20 +0200 Subject: [PATCH 315/351] matroskamux: rework _request_new_pad to handle explict req-pad-names Don't ignore explicit pad-names. --- gst/matroska/matroska-mux.c | 56 ++++++++++++++++++++++++++++++------- 1 file changed, 46 insertions(+), 10 deletions(-) diff --git a/gst/matroska/matroska-mux.c b/gst/matroska/matroska-mux.c index 7cea073179..31fe40bfc3 100644 --- a/gst/matroska/matroska-mux.c +++ b/gst/matroska/matroska-mux.c @@ -46,6 +46,7 @@ #endif #include +#include #include #include @@ -89,7 +90,7 @@ static GstStaticPadTemplate src_templ = GST_STATIC_PAD_TEMPLATE ("src", "width = (int) [ 16, 4096 ], " \ "height = (int) [ 16, 4096 ] " -/* FIXME: +/* FIXME: * * require codec data, etc as needed */ @@ -571,7 +572,7 @@ gst_matroska_mux_reset (GstElement * element) * @pad: Pad which received the event. * @event: Received event. * - * handle events - copied from oggmux without understanding + * handle events - copied from oggmux without understanding * * Returns: #TRUE on success. */ @@ -1790,32 +1791,55 @@ gst_matroska_mux_subtitle_pad_setcaps (GstPad * pad, GstCaps * caps) */ static GstPad * gst_matroska_mux_request_new_pad (GstElement * element, - GstPadTemplate * templ, const gchar * pad_name) + GstPadTemplate * templ, const gchar * req_name) { GstElementClass *klass = GST_ELEMENT_GET_CLASS (element); GstMatroskaMux *mux = GST_MATROSKA_MUX (element); GstMatroskaPad *collect_pad; GstPad *newpad = NULL; gchar *name = NULL; + const gchar *pad_name = NULL; GstPadSetCapsFunction setcapsfunc = NULL; GstMatroskaTrackContext *context = NULL; + gint pad_id; if (templ == gst_element_class_get_pad_template (klass, "audio_%d")) { - name = g_strdup_printf ("audio_%d", mux->num_a_streams++); + /* don't mix named and unnamed pads, if the pad already exists we fail when + * trying to add it */ + if (req_name != NULL && sscanf (req_name, "audio_%d", &pad_id) == 1) { + pad_name = req_name; + } else { + name = g_strdup_printf ("audio_%d", mux->num_a_streams++); + pad_name = name; + } setcapsfunc = GST_DEBUG_FUNCPTR (gst_matroska_mux_audio_pad_setcaps); context = (GstMatroskaTrackContext *) g_new0 (GstMatroskaTrackAudioContext, 1); context->type = GST_MATROSKA_TRACK_TYPE_AUDIO; context->name = g_strdup ("Audio"); } else if (templ == gst_element_class_get_pad_template (klass, "video_%d")) { - name = g_strdup_printf ("video_%d", mux->num_v_streams++); + /* don't mix named and unnamed pads, if the pad already exists we fail when + * trying to add it */ + if (req_name != NULL && sscanf (req_name, "video_%d", &pad_id) == 1) { + pad_name = req_name; + } else { + name = g_strdup_printf ("video_%d", mux->num_v_streams++); + pad_name = name; + } setcapsfunc = GST_DEBUG_FUNCPTR (gst_matroska_mux_video_pad_setcaps); context = (GstMatroskaTrackContext *) g_new0 (GstMatroskaTrackVideoContext, 1); context->type = GST_MATROSKA_TRACK_TYPE_VIDEO; context->name = g_strdup ("Video"); } else if (templ == gst_element_class_get_pad_template (klass, "subtitle_%d")) { - name = g_strdup_printf ("subtitle_%d", mux->num_t_streams++); + /* don't mix named and unnamed pads, if the pad already exists we fail when + * trying to add it */ + if (req_name != NULL && sscanf (req_name, "subtitle_%d", &pad_id) == 1) { + pad_name = req_name; + } else { + name = g_strdup_printf ("subtitle_%d", mux->num_t_streams++); + pad_name = name; + } setcapsfunc = GST_DEBUG_FUNCPTR (gst_matroska_mux_subtitle_pad_setcaps); context = (GstMatroskaTrackContext *) g_new0 (GstMatroskaTrackSubtitleContext, 1); @@ -1826,7 +1850,7 @@ gst_matroska_mux_request_new_pad (GstElement * element, return NULL; } - newpad = gst_pad_new_from_template (templ, name); + newpad = gst_pad_new_from_template (templ, pad_name); g_free (name); collect_pad = (GstMatroskaPad *) gst_collect_pads_add_pad_full (mux->collect, newpad, @@ -1850,10 +1874,22 @@ gst_matroska_mux_request_new_pad (GstElement * element, gst_pad_set_setcaps_function (newpad, setcapsfunc); gst_pad_set_active (newpad, TRUE); - gst_element_add_pad (element, newpad); + if (!gst_element_add_pad (element, newpad)) + goto pad_add_failed; + mux->num_streams++; + GST_DEBUG_OBJECT (newpad, "Added new request pad"); + return newpad; + + /* ERROR cases */ +pad_add_failed: + { + GST_WARNING_OBJECT (mux, "Adding the new pad '%s' failed", pad_name); + gst_object_unref (newpad); + return NULL; + } } /** @@ -2350,7 +2386,7 @@ gst_matroska_mux_finish (GstMatroskaMux * mux) * @mux: #GstMatroskaMux * @popped: True if at least one buffer was popped from #GstCollectPads * - * Find a pad with the oldest data + * Find a pad with the oldest data * (data from this pad should be written first). * * Returns: Selected pad. @@ -2397,7 +2433,7 @@ gst_matroska_mux_best_pad (GstMatroskaMux * mux, gboolean * popped) * @flags: Buffer flags. * * Create a buffer containing buffer header. - * + * * Returns: New buffer. */ static GstBuffer * From 9852f1cee6ff5d8a72b8938dba67bdb2f4821cf9 Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Thu, 17 Feb 2011 12:44:01 +0200 Subject: [PATCH 316/351] qtdemux: formatting cleanup Trim trailing whitespaces and fix the formatting of double negation. --- gst/qtdemux/qtdemux.c | 28 ++++++++++++++-------------- 1 file changed, 14 insertions(+), 14 deletions(-) diff --git a/gst/qtdemux/qtdemux.c b/gst/qtdemux/qtdemux.c index ceafefadf7..df286a0964 100644 --- a/gst/qtdemux/qtdemux.c +++ b/gst/qtdemux/qtdemux.c @@ -155,14 +155,14 @@ struct _QtDemuxSample * .-----------------------------------------------------------. * track: | K.....K.........K........K.......K.......K...........K... | * '-----------------------------------------------------------' - * 0 1 2 3 4 + * 0 1 2 3 4 * .------------^ ^ .----------^ ^ * / .-------------' / .------------------' * / / .-----' / * .--------------. .--------------. * | segment 1 | | segment 2 | * '--------------' '--------------' - * + * * The challenge here is to cut out the right pieces of the track for each of * the playback segments. This fortunatly can easily be done with the SEGMENT * events of gstreamer. @@ -174,7 +174,7 @@ struct _QtDemuxSample * * We then proceed to push data from keyframe (a) to frame (b). The decoder * decodes but clips all before media_time 1. - * + * * After finishing a segment, we push out a new SEGMENT event with the clipping * boundaries of the new data. * @@ -1054,7 +1054,7 @@ gst_qtdemux_move_stream (GstQTDemux * qtdemux, QtDemuxStream * str, /* position changed, we have a discont */ str->sample_index = index; - /* Each time we move in the stream we store the position where we are + /* Each time we move in the stream we store the position where we are * starting from */ str->from_sample = index; str->discont = TRUE; @@ -2787,8 +2787,8 @@ beach: return ret; } -/* Seeks to the previous keyframe of the indexed stream and - * aligns other streams with respect to the keyframe timestamp +/* Seeks to the previous keyframe of the indexed stream and + * aligns other streams with respect to the keyframe timestamp * of indexed stream. Only called in case of Reverse Playback */ static GstFlowReturn @@ -2803,7 +2803,7 @@ gst_qtdemux_seek_to_previous_keyframe (GstQTDemux * qtdemux) guint64 seg_media_start_mov; /* segment media start time in mov format */ /* Now we choose an arbitrary stream, get the previous keyframe timestamp - * and finally align all the other streams on that timestamp with their + * and finally align all the other streams on that timestamp with their * respective keyframes */ for (n = 0; n < qtdemux->n_streams; n++) { QtDemuxStream *str = qtdemux->streams[n]; @@ -3011,7 +3011,7 @@ gst_qtdemux_activate_segment (GstQTDemux * qtdemux, QtDemuxStream * stream, /* qtdemux->segment.stop is in outside-time-realm, whereas * segment->media_stop is in track-time-realm. - * + * * In order to compare the two, we need to bring segment.stop * into the track-time-realm */ @@ -3328,7 +3328,7 @@ gst_qtdemux_sync_streams (GstQTDemux * demux) } /* UNEXPECTED and NOT_LINKED need to be combined. This means that we return: - * + * * GST_FLOW_NOT_LINKED: when all pads NOT_LINKED. * GST_FLOW_UNEXPECTED: when all pads UNEXPECTED or NOT_LINKED. */ @@ -3828,7 +3828,7 @@ pause: gint64 stop; /* FIXME: I am not sure this is the right fix. If the sinks are - * supposed to detect the segment is complete and accumulate + * supposed to detect the segment is complete and accumulate * automatically, it does not seem to work here. Need more work */ qtdemux->segment_running = TRUE; @@ -5224,7 +5224,7 @@ qtdemux_stbl_init (GstQTDemux * qtdemux, QtDemuxStream * stream, GNode * stbl) /* sync sample atom */ stream->stps_present = FALSE; if ((stream->stss_present = - ! !qtdemux_tree_get_child_by_type_full (stbl, FOURCC_stss, + !!qtdemux_tree_get_child_by_type_full (stbl, FOURCC_stss, &stream->stss) ? TRUE : FALSE) == TRUE) { /* copy atom data into a new buffer for later use */ stream->stss.data = g_memdup (stream->stss.data, stream->stss.size); @@ -5242,7 +5242,7 @@ qtdemux_stbl_init (GstQTDemux * qtdemux, QtDemuxStream * stream, GNode * stbl) /* partial sync sample atom */ if ((stream->stps_present = - ! !qtdemux_tree_get_child_by_type_full (stbl, FOURCC_stps, + !!qtdemux_tree_get_child_by_type_full (stbl, FOURCC_stps, &stream->stps) ? TRUE : FALSE) == TRUE) { /* copy atom data into a new buffer for later use */ stream->stps.data = g_memdup (stream->stps.data, stream->stps.size); @@ -5361,7 +5361,7 @@ qtdemux_stbl_init (GstQTDemux * qtdemux, QtDemuxStream * stream, GNode * stbl) /* composition time-to-sample */ if ((stream->ctts_present = - ! !qtdemux_tree_get_child_by_type_full (stbl, FOURCC_ctts, + !!qtdemux_tree_get_child_by_type_full (stbl, FOURCC_ctts, &stream->ctts) ? TRUE : FALSE) == TRUE) { /* copy atom data into a new buffer for later use */ stream->ctts.data = g_memdup (stream->ctts.data, stream->ctts.size); @@ -6068,7 +6068,7 @@ qtdemux_get_rtsp_uri_from_hndl (GstQTDemux * qtdemux, GNode * minf) GST_DEBUG_OBJECT (qtdemux, "Found hndl atom"); /* skip data reference handle bytes and the - * following pascal string and some extra 4 + * following pascal string and some extra 4 * bytes I have no idea what are */ if (!gst_byte_reader_skip (&dref, 4) || !gst_byte_reader_get_uint8 (&dref, &string_len) || From b38cf735a58ce0cd17ac2bd290825e1ad404ae28 Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Thu, 17 Feb 2011 12:46:14 +0200 Subject: [PATCH 317/351] qtdemux: mark delta-unit in the index We need to use the delta flag fro delta units and not none. Print more details to the debug log. --- gst/qtdemux/qtdemux.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/gst/qtdemux/qtdemux.c b/gst/qtdemux/qtdemux.c index df286a0964..18b4ff1470 100644 --- a/gst/qtdemux/qtdemux.c +++ b/gst/qtdemux/qtdemux.c @@ -1741,7 +1741,8 @@ gst_qtdemux_set_index (GstElement * element, GstIndex * index) /* object lock might be taken again */ if (index) gst_index_get_writer_id (index, GST_OBJECT (element), &demux->index_id); - GST_DEBUG_OBJECT (demux, "Set index %" GST_PTR_FORMAT, demux->element_index); + GST_DEBUG_OBJECT (demux, "Set index %" GST_PTR_FORMAT "for writer id %d", + demux->element_index, demux->index_id); } static GstIndex * @@ -3623,7 +3624,7 @@ gst_qtdemux_decorate_and_push_buffer (GstQTDemux * qtdemux, gst_index_add_association (qtdemux->element_index, qtdemux->index_id, keyframe ? GST_ASSOCIATION_FLAG_KEY_UNIT : - GST_ASSOCIATION_FLAG_NONE, GST_FORMAT_TIME, stream_time, + GST_ASSOCIATION_FLAG_DELTA_UNIT, GST_FORMAT_TIME, stream_time, GST_FORMAT_BYTES, byte_position, NULL); } } From 603c703cf258c67254634e715b0e766d405902b8 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Thu, 17 Feb 2011 11:58:42 +0000 Subject: [PATCH 318/351] .gitignore: ignore moved equalizer test binary --- tests/icles/.gitignore | 1 + 1 file changed, 1 insertion(+) diff --git a/tests/icles/.gitignore b/tests/icles/.gitignore index 9ba812160e..0b5704b32a 100644 --- a/tests/icles/.gitignore +++ b/tests/icles/.gitignore @@ -1,3 +1,4 @@ +equalizer-test gdkpixbufsink-test test-oss4 ximagesrc-test From ade128a002469e7a2fa8295d4472c3eeadd4f303 Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Thu, 17 Feb 2011 13:57:37 +0200 Subject: [PATCH 319/351] avidemux, flvdemux: mark delta-units in the index We need to use the 'delta' flag for delta units and not the 'none' flag. --- gst/avi/gstavidemux.c | 5 +++-- gst/flv/gstflvdemux.c | 5 +++-- 2 files changed, 6 insertions(+), 4 deletions(-) diff --git a/gst/avi/gstavidemux.c b/gst/avi/gstavidemux.c index a4a0525aa6..208f9d3a25 100644 --- a/gst/avi/gstavidemux.c +++ b/gst/avi/gstavidemux.c @@ -4485,8 +4485,9 @@ gst_avi_demux_add_assoc (GstAviDemux * avi, GstAviStream * stream, GST_LOG_OBJECT (avi, "adding association %" GST_TIME_FORMAT "-> %" G_GUINT64_FORMAT, GST_TIME_ARGS (timestamp), offset); gst_index_add_association (avi->element_index, avi->index_id, - keyframe ? GST_ASSOCIATION_FLAG_KEY_UNIT : GST_ASSOCIATION_FLAG_NONE, - GST_FORMAT_TIME, timestamp, GST_FORMAT_BYTES, offset, NULL); + keyframe ? GST_ASSOCIATION_FLAG_KEY_UNIT : + GST_ASSOCIATION_FLAG_DELTA_UNIT, GST_FORMAT_TIME, timestamp, + GST_FORMAT_BYTES, offset, NULL); /* well, current_total determines TIME and entry DEFAULT (frame #) ... */ gst_index_add_association (avi->element_index, stream->index_id, GST_ASSOCIATION_FLAG_NONE, diff --git a/gst/flv/gstflvdemux.c b/gst/flv/gstflvdemux.c index d976fedba2..86f9b14a13 100644 --- a/gst/flv/gstflvdemux.c +++ b/gst/flv/gstflvdemux.c @@ -129,8 +129,9 @@ gst_flv_demux_parse_and_add_index_entry (GstFlvDemux * demux, GstClockTime ts, associations[1].value = pos; gst_index_add_associationv (demux->index, demux->index_id, - (keyframe) ? GST_ASSOCIATION_FLAG_KEY_UNIT : GST_ASSOCIATION_FLAG_NONE, - 2, (const GstIndexAssociation *) &associations); + (keyframe) ? GST_ASSOCIATION_FLAG_KEY_UNIT : + GST_ASSOCIATION_FLAG_DELTA_UNIT, 2, + (const GstIndexAssociation *) &associations); if (pos > demux->index_max_pos) demux->index_max_pos = pos; From 99e4bdd8f200ff5dac0fc86a1a95c401699707a4 Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Thu, 17 Feb 2011 14:00:48 +0200 Subject: [PATCH 320/351] avidemux, flvdemux: formatting cleanup Trim trailing whitespaces and fix the formatting of double negation. --- gst/avi/gstavidemux.c | 4 ++-- gst/flv/gstflvdemux.c | 10 +++++----- 2 files changed, 7 insertions(+), 7 deletions(-) diff --git a/gst/avi/gstavidemux.c b/gst/avi/gstavidemux.c index 208f9d3a25..200d71b86e 100644 --- a/gst/avi/gstavidemux.c +++ b/gst/avi/gstavidemux.c @@ -3980,7 +3980,7 @@ gst_avi_demux_do_seek (GstAviDemux * avi, GstSegment * segment) GstAviStream *stream; seek_time = segment->last_stop; - keyframe = ! !(segment->flags & GST_SEEK_FLAG_KEY_UNIT); + keyframe = !!(segment->flags & GST_SEEK_FLAG_KEY_UNIT); GST_DEBUG_OBJECT (avi, "seek to: %" GST_TIME_FORMAT " keyframe seeking:%d", GST_TIME_ARGS (seek_time), keyframe); @@ -4246,7 +4246,7 @@ avi_demux_handle_seek_push (GstAviDemux * avi, GstPad * pad, GstEvent * event) gst_segment_set_seek (&seeksegment, rate, format, flags, cur_type, cur, stop_type, stop, &update); - keyframe = ! !(flags & GST_SEEK_FLAG_KEY_UNIT); + keyframe = !!(flags & GST_SEEK_FLAG_KEY_UNIT); cur = seeksegment.last_stop; GST_DEBUG_OBJECT (avi, diff --git a/gst/flv/gstflvdemux.c b/gst/flv/gstflvdemux.c index 86f9b14a13..87ebf0e869 100644 --- a/gst/flv/gstflvdemux.c +++ b/gst/flv/gstflvdemux.c @@ -519,7 +519,7 @@ gst_flv_demux_parse_tag_script (GstFlvDemux * demux, GstBuffer * buffer) return GST_FLOW_OK; } - /* The number of elements is just a hint, some files have + /* The number of elements is just a hint, some files have nb_elements == 0 and actually contain items. */ GST_DEBUG_OBJECT (demux, "there are approx. %d elements in the array", nb_elems); @@ -830,7 +830,7 @@ gst_flv_demux_parse_tag_audio (GstFlvDemux * demux, GstBuffer * buffer) gst_object_ref (demux->audio_pad)); /* We only emit no more pads when we have audio and video. Indeed we can - * not trust the FLV header to tell us if there will be only audio or + * not trust the FLV header to tell us if there will be only audio or * only video and we would just break discovery of some files */ if (demux->audio_pad && demux->video_pad) { GST_DEBUG_OBJECT (demux, "emitting no more pads"); @@ -1148,7 +1148,7 @@ gst_flv_demux_parse_tag_video (GstFlvDemux * demux, GstBuffer * buffer) goto beach; } - /* When we ve set pixel-aspect-ratio we use that boolean to detect a + /* When we ve set pixel-aspect-ratio we use that boolean to detect a * metadata tag that would come later and trigger a caps change */ demux->got_par = FALSE; @@ -1173,7 +1173,7 @@ gst_flv_demux_parse_tag_video (GstFlvDemux * demux, GstBuffer * buffer) gst_object_ref (demux->video_pad)); /* We only emit no more pads when we have audio and video. Indeed we can - * not trust the FLV header to tell us if there will be only audio or + * not trust the FLV header to tell us if there will be only audio or * only video and we would just break discovery of some files */ if (demux->audio_pad && demux->video_pad) { GST_DEBUG_OBJECT (demux, "emitting no more pads"); @@ -1193,7 +1193,7 @@ gst_flv_demux_parse_tag_video (GstFlvDemux * demux, GstBuffer * buffer) goto beach; } - /* When we ve set pixel-aspect-ratio we use that boolean to detect a + /* When we ve set pixel-aspect-ratio we use that boolean to detect a * metadata tag that would come later and trigger a caps change */ demux->got_par = FALSE; } From 80d820ff294ca8a9ee3ac19f880c20264b6934b3 Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Thu, 17 Feb 2011 17:56:29 +0200 Subject: [PATCH 321/351] avidemux: get the index writer id when the pad has a parent Otherwise the index writer has a weired name, as the pad has no parent yet. --- gst/avi/gstavidemux.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/gst/avi/gstavidemux.c b/gst/avi/gstavidemux.c index 200d71b86e..0de3edf0c5 100644 --- a/gst/avi/gstavidemux.c +++ b/gst/avi/gstavidemux.c @@ -1858,6 +1858,11 @@ gst_avi_demux_expose_streams (GstAviDemux * avi, gboolean force) GST_LOG_OBJECT (avi, "Added pad %s with caps %" GST_PTR_FORMAT, GST_PAD_NAME (stream->pad), GST_PAD_CAPS (stream->pad)); gst_element_add_pad ((GstElement *) avi, stream->pad); + + if (avi->element_index) + gst_index_get_writer_id (avi->element_index, + GST_OBJECT_CAST (stream->pad), &stream->index_id); + stream->exposed = TRUE; if (avi->main_stream == -1) avi->main_stream = i; @@ -2265,10 +2270,6 @@ gst_avi_demux_parse_stream (GstAviDemux * avi, GstBuffer * buf) GST_DEBUG_FUNCPTR (gst_avi_demux_src_convert)); #endif - if (avi->element_index) - gst_index_get_writer_id (avi->element_index, GST_OBJECT_CAST (stream->pad), - &stream->index_id); - stream->num = avi->num_streams; stream->start_entry = 0; From d8f29266e99c7b82e4968cace1da1d16b3db06b5 Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Thu, 17 Feb 2011 17:57:42 +0200 Subject: [PATCH 322/351] avidemux: also add the frame-type for the stream index --- gst/avi/gstavidemux.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/gst/avi/gstavidemux.c b/gst/avi/gstavidemux.c index 0de3edf0c5..321d216a8a 100644 --- a/gst/avi/gstavidemux.c +++ b/gst/avi/gstavidemux.c @@ -4491,9 +4491,10 @@ gst_avi_demux_add_assoc (GstAviDemux * avi, GstAviStream * stream, GST_FORMAT_BYTES, offset, NULL); /* well, current_total determines TIME and entry DEFAULT (frame #) ... */ gst_index_add_association (avi->element_index, stream->index_id, - GST_ASSOCIATION_FLAG_NONE, - GST_FORMAT_TIME, stream->current_total, GST_FORMAT_BYTES, offset, - GST_FORMAT_DEFAULT, stream->current_entry, NULL); + keyframe ? GST_ASSOCIATION_FLAG_KEY_UNIT : + GST_ASSOCIATION_FLAG_DELTA_UNIT, GST_FORMAT_TIME, stream->current_total, + GST_FORMAT_BYTES, offset, GST_FORMAT_DEFAULT, stream->current_entry, + NULL); } } From e54ba41ff7c415a348f20d5338addd40205be6e4 Mon Sep 17 00:00:00 2001 From: David Schleef Date: Fri, 17 Sep 2010 12:10:38 -0700 Subject: [PATCH 323/351] rtpvrawpay: Implement interlacing --- gst/rtp/gstrtpvrawpay.c | 461 ++++++++++++++++++++++------------------ gst/rtp/gstrtpvrawpay.h | 2 + 2 files changed, 253 insertions(+), 210 deletions(-) diff --git a/gst/rtp/gstrtpvrawpay.c b/gst/rtp/gstrtpvrawpay.c index 12b72e3eb5..c354639564 100644 --- a/gst/rtp/gstrtpvrawpay.c +++ b/gst/rtp/gstrtpvrawpay.c @@ -73,7 +73,7 @@ static GstStaticPadTemplate gst_rtp_vraw_pay_sink_template = "width = (int) [ 1, 32767 ], " "height = (int) [ 1, 32767 ]; " "video/x-raw-yuv, " - "format = (fourcc) { AYUV, UYVY, I420, Y41B }, " + "format = (fourcc) { AYUV, UYVY, I420, Y41B, UYVP }, " "width = (int) [ 1, 32767 ], " "height = (int) [ 1, 32767 ]; ") ); @@ -160,6 +160,8 @@ gst_rtp_vraw_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps) const gchar *depthstr, *samplingstr, *colorimetrystr; gchar *wstr, *hstr; gboolean interlaced; + const gchar *color_matrix; + gint depth; rtpvrawpay = GST_RTP_VRAW_PAY (payload); @@ -170,7 +172,6 @@ gst_rtp_vraw_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps) /* these values are the only thing we can do */ depthstr = "8"; - colorimetrystr = "SMPTE240M"; /* parse common width/height */ res = gst_structure_get_int (s, "width", &width); @@ -178,12 +179,19 @@ gst_rtp_vraw_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps) if (!res) goto missing_dimension; - /* fail on interlaced video for now */ if (!gst_structure_get_boolean (s, "interlaced", &interlaced)) interlaced = FALSE; - if (interlaced) - goto interlaced; + color_matrix = gst_structure_get_string (s, "color-matrix"); + colorimetrystr = "SMPTE240M"; + if (color_matrix) { + if (g_str_equal (color_matrix, "sdtv")) { + /* BT.601 implies a bit more than just color-matrix */ + colorimetrystr = "BT601-5"; + } else if (g_str_equal (color_matrix, "hdtv")) { + colorimetrystr = "BT709-2"; + } + } yp = up = vp = 0; xinc = yinc = 1; @@ -193,6 +201,7 @@ gst_rtp_vraw_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps) gboolean has_alpha; has_alpha = gst_structure_get_int (s, "alpha_mask", &amask); + depth = 8; if (!gst_structure_get_int (s, "red_mask", &rmask)) goto unknown_mask; @@ -233,6 +242,7 @@ gst_rtp_vraw_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps) samplingstr = "YCbCr-4:4:4"; pgroup = 3; ystride = width * 4; + depth = 8; break; case GST_MAKE_FOURCC ('U', 'Y', 'V', 'Y'): sampling = GST_VIDEO_FORMAT_UYVY; @@ -240,6 +250,7 @@ gst_rtp_vraw_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps) pgroup = 4; xinc = 2; ystride = GST_ROUND_UP_2 (width) * 2; + depth = 8; break; case GST_MAKE_FOURCC ('Y', '4', '1', 'B'): sampling = GST_VIDEO_FORMAT_Y41B; @@ -250,6 +261,7 @@ gst_rtp_vraw_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps) uvstride = GST_ROUND_UP_8 (width) / 4; up = ystride * height; vp = up + uvstride * height; + depth = 8; break; case GST_MAKE_FOURCC ('I', '4', '2', '0'): sampling = GST_VIDEO_FORMAT_I420; @@ -260,6 +272,16 @@ gst_rtp_vraw_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps) uvstride = GST_ROUND_UP_8 (width) / 2; up = ystride * GST_ROUND_UP_2 (height); vp = up + uvstride * GST_ROUND_UP_2 (height) / 2; + depth = 8; + break; + case GST_MAKE_FOURCC ('U', 'Y', 'V', 'P'): +#define GST_VIDEO_FORMAT_UYVP GST_VIDEO_FORMAT_UYVY /* FIXME */ + sampling = GST_VIDEO_FORMAT_UYVP; + samplingstr = "YCbCr-4:2:2"; + pgroup = 4; + xinc = 2; + ystride = GST_ROUND_UP_2 (width) * 2; + depth = 10; break; default: goto unknown_fourcc; @@ -267,6 +289,13 @@ gst_rtp_vraw_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps) } else goto unknown_format; + if (interlaced) { + yinc *= 2; + } + if (depth == 10) { + depthstr = "10"; + } + rtpvrawpay->width = width; rtpvrawpay->height = height; rtpvrawpay->sampling = sampling; @@ -278,6 +307,8 @@ gst_rtp_vraw_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps) rtpvrawpay->vp = vp; rtpvrawpay->ystride = ystride; rtpvrawpay->uvstride = uvstride; + rtpvrawpay->interlaced = interlaced; + rtpvrawpay->depth = depth; GST_DEBUG_OBJECT (payload, "width %d, height %d, sampling %d", width, height, sampling); @@ -289,10 +320,17 @@ gst_rtp_vraw_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps) hstr = g_strdup_printf ("%d", rtpvrawpay->height); gst_basertppayload_set_options (payload, "video", TRUE, "RAW", 90000); - res = gst_basertppayload_set_outcaps (payload, "sampling", G_TYPE_STRING, - samplingstr, "depth", G_TYPE_STRING, depthstr, "width", G_TYPE_STRING, - wstr, "height", G_TYPE_STRING, hstr, "colorimetry", G_TYPE_STRING, - colorimetrystr, NULL); + if (interlaced) { + res = gst_basertppayload_set_outcaps (payload, "sampling", G_TYPE_STRING, + samplingstr, "depth", G_TYPE_STRING, depthstr, "width", G_TYPE_STRING, + wstr, "height", G_TYPE_STRING, hstr, "colorimetry", G_TYPE_STRING, + colorimetrystr, "interlace", G_TYPE_STRING, "true", NULL); + } else { + res = gst_basertppayload_set_outcaps (payload, "sampling", G_TYPE_STRING, + samplingstr, "depth", G_TYPE_STRING, depthstr, "width", G_TYPE_STRING, + wstr, "height", G_TYPE_STRING, hstr, "colorimetry", G_TYPE_STRING, + colorimetrystr, NULL); + } g_free (wstr); g_free (hstr); @@ -314,11 +352,6 @@ unknown_fourcc: GST_ERROR_OBJECT (payload, "invalid or missing fourcc"); return FALSE; } -interlaced: - { - GST_ERROR_OBJECT (payload, "interlaced video not supported yet"); - return FALSE; - } missing_dimension: { GST_ERROR_OBJECT (payload, "missing width or height property"); @@ -337,6 +370,7 @@ gst_rtp_vraw_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buffer) guint size, pgroup; guint mtu; guint width, height; + gint field; rtpvrawpay = GST_RTP_VRAW_PAY (payload); @@ -361,220 +395,227 @@ gst_rtp_vraw_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buffer) height = rtpvrawpay->height; /* start with line 0, offset 0 */ - line = 0; - offset = 0; - /* write all lines */ - while (line < height) { - guint left; - GstBuffer *out; - guint8 *outdata, *headers; - gboolean next_line; - guint length, cont, pixels, fieldid; + for (field = 0; field < 1 + rtpvrawpay->interlaced; field++) { + line = field; + offset = 0; - /* get the max allowed payload length size, we try to fill the complete MTU */ - left = gst_rtp_buffer_calc_payload_len (mtu, 0, 0); - out = gst_rtp_buffer_new_allocate (left, 0, 0); + /* write all lines */ + while (line < height) { + guint left; + GstBuffer *out; + guint8 *outdata, *headers; + gboolean next_line; + guint length, cont, pixels; - GST_BUFFER_TIMESTAMP (out) = GST_BUFFER_TIMESTAMP (buffer); + /* get the max allowed payload length size, we try to fill the complete MTU */ + left = gst_rtp_buffer_calc_payload_len (mtu, 0, 0); + out = gst_rtp_buffer_new_allocate (left, 0, 0); - outdata = gst_rtp_buffer_get_payload (out); - - GST_LOG_OBJECT (rtpvrawpay, "created buffer of size %u for MTU %u", left, - mtu); - - /* - * 0 1 2 3 - * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 - * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - * | Extended Sequence Number | Length | - * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - * |F| Line No |C| Offset | - * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - * | Length |F| Line No | - * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - * |C| Offset | . - * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ . - * . . - * . Two (partial) lines of video data . - * . . - * +---------------------------------------------------------------+ - */ - - /* need 2 bytes for the extended sequence number */ - *outdata++ = 0; - *outdata++ = 0; - left -= 2; - - /* the headers start here */ - headers = outdata; - - /* while we can fit at least one header and one pixel */ - while (left > (6 + pgroup)) { - /* we need a 6 bytes header */ - left -= 6; - - /* get how may bytes we need for the remaining pixels */ - pixels = width - offset; - length = (pixels * pgroup) / rtpvrawpay->xinc; - - if (left >= length) { - /* pixels and header fit completely, we will write them and skip to the - * next line. */ - next_line = TRUE; + if (field == 0) { + GST_BUFFER_TIMESTAMP (out) = GST_BUFFER_TIMESTAMP (buffer); } else { - /* line does not fit completely, see how many pixels fit */ - pixels = (left / pgroup) * rtpvrawpay->xinc; + GST_BUFFER_TIMESTAMP (out) = GST_BUFFER_TIMESTAMP (buffer) + + GST_BUFFER_DURATION (buffer) / 2; + } + + outdata = gst_rtp_buffer_get_payload (out); + + GST_LOG_OBJECT (rtpvrawpay, "created buffer of size %u for MTU %u", left, + mtu); + + /* + * 0 1 2 3 + * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 + * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + * | Extended Sequence Number | Length | + * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + * |F| Line No |C| Offset | + * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + * | Length |F| Line No | + * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + * |C| Offset | . + * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ . + * . . + * . Two (partial) lines of video data . + * . . + * +---------------------------------------------------------------+ + */ + + /* need 2 bytes for the extended sequence number */ + *outdata++ = 0; + *outdata++ = 0; + left -= 2; + + /* the headers start here */ + headers = outdata; + + /* while we can fit at least one header and one pixel */ + while (left > (6 + pgroup)) { + /* we need a 6 bytes header */ + left -= 6; + + /* get how may bytes we need for the remaining pixels */ + pixels = width - offset; length = (pixels * pgroup) / rtpvrawpay->xinc; - next_line = FALSE; - } - GST_LOG_OBJECT (rtpvrawpay, "filling %u bytes in %u pixels", length, - pixels); - left -= length; - /* write length */ - *outdata++ = (length >> 8) & 0xff; - *outdata++ = length & 0xff; - - /* always 0 for now */ - fieldid = 0x00; - - /* write line no */ - *outdata++ = ((line >> 8) & 0x7f) | fieldid; - *outdata++ = line & 0xff; - - if (next_line) { - /* go to next line we do this here to make the check below easier */ - line += rtpvrawpay->yinc; - } - - /* calculate continuation marker */ - cont = (left > (6 + pgroup) && line < height) ? 0x80 : 0x00; - - /* write offset and continuation marker */ - *outdata++ = ((offset >> 8) & 0x7f) | cont; - *outdata++ = offset & 0xff; - - if (next_line) { - /* reset offset */ - offset = 0; - GST_LOG_OBJECT (rtpvrawpay, "go to next line %u", line); - } else { - offset += pixels; - GST_LOG_OBJECT (rtpvrawpay, "next offset %u", offset); - } - - if (!cont) - break; - } - GST_LOG_OBJECT (rtpvrawpay, "consumed %u bytes", - (guint) (outdata - headers)); - - /* second pass, read headers and write the data */ - while (TRUE) { - guint offs, lin; - - /* read length and cont */ - length = (headers[0] << 8) | headers[1]; - lin = ((headers[2] & 0x7f) << 8) | headers[3]; - offs = ((headers[4] & 0x7f) << 8) | headers[5]; - cont = headers[4] & 0x80; - pixels = length / pgroup; - headers += 6; - - GST_LOG_OBJECT (payload, "writing length %u, line %u, offset %u, cont %d", - length, lin, offs, cont); - - switch (rtpvrawpay->sampling) { - case GST_VIDEO_FORMAT_RGB: - case GST_VIDEO_FORMAT_RGBA: - case GST_VIDEO_FORMAT_BGR: - case GST_VIDEO_FORMAT_BGRA: - case GST_VIDEO_FORMAT_UYVY: - offs /= rtpvrawpay->xinc; - memcpy (outdata, yp + (lin * ystride) + (offs * pgroup), length); - outdata += length; - break; - case GST_VIDEO_FORMAT_AYUV: - { - gint i; - guint8 *datap; - - datap = yp + (lin * ystride) + (offs * 4); - - for (i = 0; i < pixels; i++) { - *outdata++ = datap[2]; - *outdata++ = datap[1]; - *outdata++ = datap[3]; - datap += 4; - } - break; + if (left >= length) { + /* pixels and header fit completely, we will write them and skip to the + * next line. */ + next_line = TRUE; + } else { + /* line does not fit completely, see how many pixels fit */ + pixels = (left / pgroup) * rtpvrawpay->xinc; + length = (pixels * pgroup) / rtpvrawpay->xinc; + next_line = FALSE; } - case GST_VIDEO_FORMAT_I420: - { - gint i; - guint uvoff; - guint8 *yd1p, *yd2p, *udp, *vdp; + GST_LOG_OBJECT (rtpvrawpay, "filling %u bytes in %u pixels", length, + pixels); + left -= length; - yd1p = yp + (lin * ystride) + (offs); - yd2p = yd1p + ystride; - uvoff = - (lin / rtpvrawpay->yinc * uvstride) + (offs / rtpvrawpay->xinc); - udp = up + uvoff; - vdp = vp + uvoff; + /* write length */ + *outdata++ = (length >> 8) & 0xff; + *outdata++ = length & 0xff; - for (i = 0; i < pixels; i++) { - *outdata++ = *yd1p++; - *outdata++ = *yd1p++; - *outdata++ = *yd2p++; - *outdata++ = *yd2p++; - *outdata++ = *udp++; - *outdata++ = *vdp++; - } - break; + /* write line no */ + *outdata++ = ((line >> 8) & 0x7f) | ((field << 7) & 0x80); + *outdata++ = line & 0xff; + + if (next_line) { + /* go to next line we do this here to make the check below easier */ + line += rtpvrawpay->yinc; } - case GST_VIDEO_FORMAT_Y41B: - { - gint i; - guint uvoff; - guint8 *ydp, *udp, *vdp; - ydp = yp + (lin * ystride) + offs; - uvoff = - (lin / rtpvrawpay->yinc * uvstride) + (offs / rtpvrawpay->xinc); - udp = up + uvoff; - vdp = vp + uvoff; + /* calculate continuation marker */ + cont = (left > (6 + pgroup) && line < height) ? 0x80 : 0x00; - for (i = 0; i < pixels; i++) { - *outdata++ = *udp++; - *outdata++ = *ydp++; - *outdata++ = *ydp++; - *outdata++ = *vdp++; - *outdata++ = *ydp++; - *outdata++ = *ydp++; - } - break; + /* write offset and continuation marker */ + *outdata++ = ((offset >> 8) & 0x7f) | cont; + *outdata++ = offset & 0xff; + + if (next_line) { + /* reset offset */ + offset = 0; + GST_LOG_OBJECT (rtpvrawpay, "go to next line %u", line); + } else { + offset += pixels; + GST_LOG_OBJECT (rtpvrawpay, "next offset %u", offset); } - default: - gst_buffer_unref (out); - goto unknown_sampling; + + if (!cont) + break; + } + GST_LOG_OBJECT (rtpvrawpay, "consumed %u bytes", + (guint) (outdata - headers)); + + /* second pass, read headers and write the data */ + while (TRUE) { + guint offs, lin; + + /* read length and cont */ + length = (headers[0] << 8) | headers[1]; + lin = ((headers[2] & 0x7f) << 8) | headers[3]; + offs = ((headers[4] & 0x7f) << 8) | headers[5]; + cont = headers[4] & 0x80; + pixels = length / pgroup; + headers += 6; + + GST_LOG_OBJECT (payload, + "writing length %u, line %u, offset %u, cont %d", length, lin, offs, + cont); + + switch (rtpvrawpay->sampling) { + case GST_VIDEO_FORMAT_RGB: + case GST_VIDEO_FORMAT_RGBA: + case GST_VIDEO_FORMAT_BGR: + case GST_VIDEO_FORMAT_BGRA: + case GST_VIDEO_FORMAT_UYVY: + offs /= rtpvrawpay->xinc; + memcpy (outdata, yp + (lin * ystride) + (offs * pgroup), length); + outdata += length; + break; + case GST_VIDEO_FORMAT_AYUV: + { + gint i; + guint8 *datap; + + datap = yp + (lin * ystride) + (offs * 4); + + for (i = 0; i < pixels; i++) { + *outdata++ = datap[2]; + *outdata++ = datap[1]; + *outdata++ = datap[3]; + datap += 4; + } + break; + } + case GST_VIDEO_FORMAT_I420: + { + gint i; + guint uvoff; + guint8 *yd1p, *yd2p, *udp, *vdp; + + yd1p = yp + (lin * ystride) + (offs); + yd2p = yd1p + ystride; + uvoff = + (lin / rtpvrawpay->yinc * uvstride) + (offs / rtpvrawpay->xinc); + udp = up + uvoff; + vdp = vp + uvoff; + + for (i = 0; i < pixels; i++) { + *outdata++ = *yd1p++; + *outdata++ = *yd1p++; + *outdata++ = *yd2p++; + *outdata++ = *yd2p++; + *outdata++ = *udp++; + *outdata++ = *vdp++; + } + break; + } + case GST_VIDEO_FORMAT_Y41B: + { + gint i; + guint uvoff; + guint8 *ydp, *udp, *vdp; + + ydp = yp + (lin * ystride) + offs; + uvoff = + (lin / rtpvrawpay->yinc * uvstride) + (offs / rtpvrawpay->xinc); + udp = up + uvoff; + vdp = vp + uvoff; + + for (i = 0; i < pixels; i++) { + *outdata++ = *udp++; + *outdata++ = *ydp++; + *outdata++ = *ydp++; + *outdata++ = *vdp++; + *outdata++ = *ydp++; + *outdata++ = *ydp++; + } + break; + } + default: + gst_buffer_unref (out); + goto unknown_sampling; + } + + if (!cont) + break; } - if (!cont) - break; + if (line >= height) { + GST_LOG_OBJECT (rtpvrawpay, "field/frame complete, set marker"); + gst_rtp_buffer_set_marker (out, TRUE); + } + if (left > 0) { + GST_LOG_OBJECT (rtpvrawpay, "we have %u bytes left", left); + GST_BUFFER_SIZE (out) -= left; + } + + /* push buffer */ + ret = gst_basertppayload_push (payload, out); } - if (line >= height) { - GST_LOG_OBJECT (rtpvrawpay, "frame complete, set marker"); - gst_rtp_buffer_set_marker (out, TRUE); - } - if (left > 0) { - GST_LOG_OBJECT (rtpvrawpay, "we have %u bytes left", left); - GST_BUFFER_SIZE (out) -= left; - } - - /* push buffer */ - ret = gst_basertppayload_push (payload, out); } gst_buffer_unref (buffer); diff --git a/gst/rtp/gstrtpvrawpay.h b/gst/rtp/gstrtpvrawpay.h index 44a7ee1b3d..0e6adf46dc 100644 --- a/gst/rtp/gstrtpvrawpay.h +++ b/gst/rtp/gstrtpvrawpay.h @@ -52,6 +52,8 @@ struct _GstRtpVRawPay guint yp, up, vp; gint ystride; gint uvstride; + gboolean interlaced; + gint depth; }; struct _GstRtpVRawPayClass From 1732baf7af3ce0ad79a6cff02f6fbe391da37eb3 Mon Sep 17 00:00:00 2001 From: David Schleef Date: Thu, 6 Jan 2011 15:44:24 -0800 Subject: [PATCH 324/351] debugutils: remove bitrotten negotiation element Wasn't enabled, didn't work, and planned features have been superceded by capsfilter and capsdebug. --- gst/debugutils/Makefile.am | 2 - gst/debugutils/negotiation.c | 282 ----------------------------------- 2 files changed, 284 deletions(-) delete mode 100644 gst/debugutils/negotiation.c diff --git a/gst/debugutils/Makefile.am b/gst/debugutils/Makefile.am index 1f423b65ea..d905413c1f 100644 --- a/gst/debugutils/Makefile.am +++ b/gst/debugutils/Makefile.am @@ -46,8 +46,6 @@ libgstdebug_la_SOURCES = \ cpureport.c \ testplugin.c -# negotiation.c - libgstdebug_la_CFLAGS = $(GST_CFLAGS) $(GST_BASE_CFLAGS) libgstdebug_la_LIBADD = $(GST_LIBS) $(GST_BASE_LIBS) libgstdebug_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) diff --git a/gst/debugutils/negotiation.c b/gst/debugutils/negotiation.c deleted file mode 100644 index 176653ff45..0000000000 --- a/gst/debugutils/negotiation.c +++ /dev/null @@ -1,282 +0,0 @@ -/* - * GStreamer - * Copyright (C) 1999-2001 Erik Walthinsen - * Copyright (C) 2002 David A. Schleef - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Library General Public - * License as published by the Free Software Foundation; either - * version 2 of the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Library General Public License for more details. - * - * You should have received a copy of the GNU Library General Public - * License along with this library; if not, write to the - * Free Software Foundation, Inc., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif -#include - -#include -#include -#include - - -#define GST_TYPE_NEGOTIATION \ - (gst_gst_negotiation_get_type()) -#define GST_NEGOTIATION(obj) \ - (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_NEGOTIATION,GstNegotiation)) -#define GST_NEGOTIATION_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_NEGOTIATION,GstNegotiation)) -#define GST_IS_NEGOTIATION(obj) \ - (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_NEGOTIATION)) -#define GST_IS_NEGOTIATION_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_NEGOTIATION)) - -typedef struct _GstNegotiation GstNegotiation; -typedef struct _GstNegotiationClass GstNegotiationClass; - -struct _GstNegotiation -{ - GstElement element; - - GstPad *sinkpad, *srcpad; - - GstCaps *caps; -}; - -struct _GstNegotiationClass -{ - GstElementClass parent_class; -}; - -GType gst_gst_negotiation_get_type (void); - -/* Filter signals and args */ -enum -{ - /* FILL ME */ - LAST_SIGNAL -}; - -enum -{ - ARG_0, - ARG_ALLOWED_CAPS -}; - -static GstStaticPadTemplate gst_negotiation_sink_factory = -GST_STATIC_PAD_TEMPLATE ("sink", - GST_PAD_SINK, - GST_PAD_ALWAYS, - GST_STATIC_CAPS_ANY); - -static GstStaticPadTemplate gst_negotiation_src_factory = -GST_STATIC_PAD_TEMPLATE ("src", - GST_PAD_SRC, - GST_PAD_ALWAYS, - GST_STATIC_CAPS_ANY); - -static void gst_negotiation_base_init (gpointer g_class); -static void gst_negotiation_class_init (GstNegotiationClass * klass); -static void gst_negotiation_init (GstNegotiation * filter); - -static GstCaps *gst_negotiation_getcaps (GstPad * pad); -static GstPadLinkReturn gst_negotiation_pad_link (GstPad * pad, - const GstCaps * caps); - -static void gst_negotiation_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec); -static void gst_negotiation_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec); - -static void gst_negotiation_update_caps (GstNegotiation * negotiation); -static void gst_negotiation_chain (GstPad * pad, GstData * _data); - -static GstElementClass *parent_class = NULL; - -GType -gst_gst_negotiation_get_type (void) -{ - static GType plugin_type = 0; - - if (!plugin_type) { - static const GTypeInfo plugin_info = { - sizeof (GstNegotiationClass), - gst_negotiation_base_init, - NULL, - (GClassInitFunc) gst_negotiation_class_init, - NULL, - NULL, - sizeof (GstNegotiation), - 0, - (GInstanceInitFunc) gst_negotiation_init, - }; - - plugin_type = g_type_register_static (GST_TYPE_ELEMENT, - "GstNegotiation", &plugin_info, 0); - } - return plugin_type; -} - -static void -gst_negotiation_base_init (gpointer g_class) -{ - GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); - - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&gst_negotiation_sink_factory)); - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&gst_negotiation_src_factory)); - gst_element_class_set_details_simple (element_class, "Negotiation", - "Testing", - "This element acts like identity, except that one can control how " - "negotiation works", "David A. Schleef "); -} - -static void -gst_negotiation_class_init (GstNegotiationClass * klass) -{ - GObjectClass *gobject_class; - GstElementClass *gstelement_class; - - gobject_class = (GObjectClass *) klass; - gstelement_class = (GstElementClass *) klass; - - parent_class = g_type_class_peek_parent (klass); - - gobject_class->set_property = gst_negotiation_set_property; - gobject_class->get_property = gst_negotiation_get_property; - - g_object_class_install_property (gobject_class, ARG_ALLOWED_CAPS, - g_param_spec_boxed ("allowed-caps", "Caps", - "The range of formats allowed by " "this element's peers", - GST_TYPE_CAPS, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); -} - -static void -gst_negotiation_init (GstNegotiation * filter) -{ - filter->sinkpad = - gst_pad_new_from_static_template (&gst_negotiation_sink_factory, "sink"); - gst_pad_set_getcaps_function (filter->sinkpad, gst_negotiation_getcaps); - gst_pad_set_link_function (filter->sinkpad, gst_negotiation_pad_link); - filter->srcpad = - gst_pad_new_from_static_template (&gst_negotiation_src_factory, "src"); - gst_pad_set_getcaps_function (filter->srcpad, gst_negotiation_getcaps); - gst_pad_set_link_function (filter->srcpad, gst_negotiation_pad_link); - - gst_element_add_pad (GST_ELEMENT (filter), filter->sinkpad); - gst_element_add_pad (GST_ELEMENT (filter), filter->srcpad); - gst_pad_set_chain_function (filter->sinkpad, gst_negotiation_chain); -} - -static GstCaps * -gst_negotiation_getcaps (GstPad * pad) -{ - GstNegotiation *negotiation = GST_NEGOTIATION (gst_pad_get_parent (pad)); - GstPad *otherpad; - GstCaps *caps; - - otherpad = (pad == negotiation->sinkpad) ? negotiation->srcpad : - negotiation->sinkpad; - - caps = gst_pad_get_allowed_caps (otherpad); - - GST_ERROR ("getcaps called on %" GST_PTR_FORMAT ", returning %" - GST_PTR_FORMAT, pad, caps); - - gst_negotiation_update_caps (negotiation); - gst_object_unref (negotiation); - - return caps; -} - -static GstPadLinkReturn -gst_negotiation_pad_link (GstPad * pad, const GstCaps * caps) -{ - GstNegotiation *negotiation = GST_NEGOTIATION (gst_pad_get_parent (pad)); - GstPad *otherpad; - GstPadLinkReturn ret; - - otherpad = (pad == negotiation->sinkpad) ? negotiation->srcpad : - negotiation->sinkpad; - - ret = gst_pad_try_set_caps (otherpad, caps); - - GST_ERROR ("pad_link called on %" GST_PTR_FORMAT " with caps %" - GST_PTR_FORMAT ", returning %d", pad, caps, ret); - gst_object_unref (negotiation); - - return ret; -} - -static void -gst_negotiation_update_caps (GstNegotiation * negotiation) -{ - GstCaps *srccaps; - GstCaps *sinkcaps; - GstCaps *icaps; - - srccaps = gst_pad_get_allowed_caps (negotiation->srcpad); - sinkcaps = gst_pad_get_allowed_caps (negotiation->sinkpad); - - icaps = gst_caps_intersect (srccaps, sinkcaps); - gst_caps_free (srccaps); - gst_caps_free (sinkcaps); - - gst_caps_replace (&negotiation->caps, icaps); - g_object_notify (G_OBJECT (negotiation), "allowed-caps"); - GST_DEBUG ("notify %" GST_PTR_FORMAT, icaps); -} - -static void -gst_negotiation_chain (GstPad * pad, GstData * _data) -{ - GstNegotiation *negotiation = GST_NEGOTIATION (gst_pad_get_parent (pad)); - - gst_pad_push (negotiation->srcpad, _data); - gst_object_unref (negotiation); -} - -static void -gst_negotiation_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec) -{ - GstNegotiation *filter; - - g_return_if_fail (GST_IS_NEGOTIATION (object)); - filter = GST_NEGOTIATION (object); - - switch (prop_id) { - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static void -gst_negotiation_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec) -{ - GstNegotiation *filter; - - g_return_if_fail (GST_IS_NEGOTIATION (object)); - filter = GST_NEGOTIATION (object); - - switch (prop_id) { - case ARG_ALLOWED_CAPS: - g_value_set_boxed (value, filter->caps); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} From 63bb9807a3e3cb16bbecb923bb1fab9ac804875a Mon Sep 17 00:00:00 2001 From: David Schleef Date: Sat, 22 Jan 2011 00:13:16 -0800 Subject: [PATCH 325/351] flvmux: don't set duration for live stream --- gst/flv/gstflvmux.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/gst/flv/gstflvmux.c b/gst/flv/gstflvmux.c index e731604711..68a0df4fc7 100644 --- a/gst/flv/gstflvmux.c +++ b/gst/flv/gstflvmux.c @@ -685,9 +685,11 @@ gst_flv_mux_create_metadata (GstFlvMux * mux) /* Some players expect the 'duration' to be always set. Fill it out later, after querying the pads or after getting EOS */ - tmp = gst_flv_mux_create_number_script_value ("duration", 0); - script_tag = gst_buffer_join (script_tag, tmp); - tags_written++; + if (!mux->streamable) { + tmp = gst_flv_mux_create_number_script_value ("duration", 86400); + script_tag = gst_buffer_join (script_tag, tmp); + tags_written++; + } /* Sometimes the information about the total file size is useful for the player. It will be filled later, after getting EOS */ From eab982ce6055d81c6b2b89d9bb2687a845a0b899 Mon Sep 17 00:00:00 2001 From: David Schleef Date: Thu, 17 Feb 2011 17:57:55 -0800 Subject: [PATCH 326/351] matroskademux: Earlier debug category initialization --- gst/matroska/matroska-demux.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/gst/matroska/matroska-demux.c b/gst/matroska/matroska-demux.c index 52f14679ee..401554b435 100644 --- a/gst/matroska/matroska-demux.c +++ b/gst/matroska/matroska-demux.c @@ -229,10 +229,6 @@ gst_matroska_demux_class_init (GstMatroskaDemuxClass * klass) GObjectClass *gobject_class = (GObjectClass *) klass; GstElementClass *gstelement_class = (GstElementClass *) klass; - /* parser helper separate debug */ - GST_DEBUG_CATEGORY_INIT (ebmlread_debug, "ebmlread", - 0, "EBML stream helper class"); - GST_DEBUG_CATEGORY_INIT (matroskademux_debug, "matroskademux", 0, "Matroska demuxer"); @@ -2693,8 +2689,8 @@ gst_matroska_demux_handle_seek_event (GstMatroskaDemux * demux, entry->pos + demux->ebml_segment_start); } - flush = ! !(flags & GST_SEEK_FLAG_FLUSH); - keyunit = ! !(flags & GST_SEEK_FLAG_KEY_UNIT); + flush = !!(flags & GST_SEEK_FLAG_FLUSH); + keyunit = !!(flags & GST_SEEK_FLAG_KEY_UNIT); if (flush) { GST_DEBUG_OBJECT (demux, "Starting flush"); @@ -7108,6 +7104,10 @@ gst_matroska_demux_plugin_init (GstPlugin * plugin) { gst_riff_init (); + /* parser helper separate debug */ + GST_DEBUG_CATEGORY_INIT (ebmlread_debug, "ebmlread", + 0, "EBML stream helper class"); + /* create an elementfactory for the matroska_demux element */ if (!gst_element_register (plugin, "matroskademux", GST_RANK_PRIMARY, GST_TYPE_MATROSKA_DEMUX)) From e017e148471aef48ef9cdc7640b92101c6d6464d Mon Sep 17 00:00:00 2001 From: David Schleef Date: Thu, 17 Feb 2011 17:59:25 -0800 Subject: [PATCH 327/351] matroskaparse: New element Copied from demux. Duplicates much code, also some dead code remaining. --- gst/matroska/Makefile.am | 2 + gst/matroska/matroska-parse.c | 5058 +++++++++++++++++++++++++++++++++ gst/matroska/matroska-parse.h | 146 + gst/matroska/matroska.c | 2 + 4 files changed, 5208 insertions(+) create mode 100644 gst/matroska/matroska-parse.c create mode 100644 gst/matroska/matroska-parse.h diff --git a/gst/matroska/Makefile.am b/gst/matroska/Makefile.am index 96777bd06a..5efbf4eb03 100644 --- a/gst/matroska/Makefile.am +++ b/gst/matroska/Makefile.am @@ -5,6 +5,7 @@ libgstmatroska_la_SOURCES = \ ebml-write.c \ matroska.c \ matroska-demux.c \ + matroska-parse.c \ matroska-ids.c \ matroska-mux.c \ webm-mux.c \ @@ -15,6 +16,7 @@ noinst_HEADERS = \ ebml-read.h \ ebml-write.h \ matroska-demux.h \ + matroska-parse.h \ matroska-ids.h \ matroska-mux.h \ webm-mux.h \ diff --git a/gst/matroska/matroska-parse.c b/gst/matroska/matroska-parse.c new file mode 100644 index 0000000000..88e0e152f9 --- /dev/null +++ b/gst/matroska/matroska-parse.c @@ -0,0 +1,5058 @@ +/* GStreamer Matroska muxer/demuxer + * (c) 2003 Ronald Bultje + * (c) 2006 Tim-Philipp Müller + * (c) 2008 Sebastian Dröge + * + * matroska-parse.c: matroska file/stream parser + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +/* TODO: check CRC32 if present + * TODO: there can be a segment after the first segment. Handle like + * chained oggs. Fixes #334082 + * TODO: Test samples: http://www.matroska.org/samples/matrix/index.html + * http://samples.mplayerhq.hu/Matroska/ + * TODO: check if parseing is done correct for all codecs according to spec + * TODO: seeking with incomplete or without CUE + */ + +/** + * SECTION:element-matroskaparse + * + * matroskaparse parsees a Matroska file into the different contained streams. + * + * + * Example launch line + * |[ + * gst-launch -v filesrc location=/path/to/mkv ! matroskaparse ! vorbisdec ! audioconvert ! audioresample ! autoaudiosink + * ]| This pipeline parsees a Matroska file and outputs the contained Vorbis audio. + * + */ + + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include +#include + +/* For AVI compatibility mode + and for fourcc stuff */ +#include +#include +#include + +#include + +#include + +#ifdef HAVE_ZLIB +#include +#endif + +#ifdef HAVE_BZ2 +#include +#endif + +#include + +#include "lzo.h" + +#include "matroska-parse.h" +#include "matroska-ids.h" + +GST_DEBUG_CATEGORY_STATIC (matroskaparse_debug); +#define GST_CAT_DEFAULT matroskaparse_debug + +#define DEBUG_ELEMENT_START(parse, ebml, element) \ + GST_DEBUG_OBJECT (parse, "Parsing " element " element at offset %" \ + G_GUINT64_FORMAT, gst_ebml_read_get_pos (ebml)) + +#define DEBUG_ELEMENT_STOP(parse, ebml, element, ret) \ + GST_DEBUG_OBJECT (parse, "Parsing " element " element " \ + " finished with '%s'", gst_flow_get_name (ret)) + +enum +{ + ARG_0, + ARG_METADATA, + ARG_STREAMINFO +}; + +static GstStaticPadTemplate sink_templ = GST_STATIC_PAD_TEMPLATE ("sink", + GST_PAD_SINK, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("video/x-matroska; video/webm") + ); + +static GstStaticPadTemplate src_templ = GST_STATIC_PAD_TEMPLATE ("src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("video/x-matroska; video/webm") + ); + +static GstFlowReturn gst_matroska_parse_parse_id (GstMatroskaParse * parse, + guint32 id, guint64 length, guint needed); + +/* element functions */ +//static void gst_matroska_parse_loop (GstPad * pad); + +static gboolean gst_matroska_parse_element_send_event (GstElement * element, + GstEvent * event); +static gboolean gst_matroska_parse_element_query (GstElement * element, + GstQuery * query); + +/* pad functions */ +static gboolean gst_matroska_parse_handle_seek_event (GstMatroskaParse * parse, + GstPad * pad, GstEvent * event); +static gboolean gst_matroska_parse_handle_src_event (GstPad * pad, + GstEvent * event); +static const GstQueryType *gst_matroska_parse_get_src_query_types (GstPad * + pad); +static gboolean gst_matroska_parse_handle_src_query (GstPad * pad, + GstQuery * query); + +static gboolean gst_matroska_parse_handle_sink_event (GstPad * pad, + GstEvent * event); +static GstFlowReturn gst_matroska_parse_chain (GstPad * pad, + GstBuffer * buffer); + +static GstStateChangeReturn +gst_matroska_parse_change_state (GstElement * element, + GstStateChange transition); +static void +gst_matroska_parse_set_index (GstElement * element, GstIndex * index); +static GstIndex *gst_matroska_parse_get_index (GstElement * element); + +/* stream methods */ +static void gst_matroska_parse_reset (GstElement * element); +static gboolean perform_seek_to_offset (GstMatroskaParse * parse, + guint64 offset); + +GType gst_matroska_parse_get_type (void); +GST_BOILERPLATE (GstMatroskaParse, gst_matroska_parse, GstElement, + GST_TYPE_ELEMENT); + +static void +gst_matroska_parse_base_init (gpointer klass) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (klass); + + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&src_templ)); + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&sink_templ)); + + gst_element_class_set_details_simple (element_class, "Matroska parser", + "Codec/Parser", + "Parses Matroska/WebM streams into video/audio/subtitles", + "GStreamer maintainers "); +} + +static void +gst_matroska_parse_finalize (GObject * object) +{ + GstMatroskaParse *parse = GST_MATROSKA_PARSE (object); + + if (parse->src) { + g_ptr_array_free (parse->src, TRUE); + parse->src = NULL; + } + + if (parse->global_tags) { + gst_tag_list_free (parse->global_tags); + parse->global_tags = NULL; + } + + g_object_unref (parse->adapter); + + G_OBJECT_CLASS (parent_class)->finalize (object); +} + +static void +gst_matroska_parse_class_init (GstMatroskaParseClass * klass) +{ + GObjectClass *gobject_class = (GObjectClass *) klass; + GstElementClass *gstelement_class = (GstElementClass *) klass; + + GST_DEBUG_CATEGORY_INIT (matroskaparse_debug, "matroskaparse", 0, + "Matroska parser"); + + gobject_class->finalize = gst_matroska_parse_finalize; + + gstelement_class->change_state = + GST_DEBUG_FUNCPTR (gst_matroska_parse_change_state); + gstelement_class->send_event = + GST_DEBUG_FUNCPTR (gst_matroska_parse_element_send_event); + gstelement_class->query = + GST_DEBUG_FUNCPTR (gst_matroska_parse_element_query); + + gstelement_class->set_index = + GST_DEBUG_FUNCPTR (gst_matroska_parse_set_index); + gstelement_class->get_index = + GST_DEBUG_FUNCPTR (gst_matroska_parse_get_index); +} + +static void +gst_matroska_parse_init (GstMatroskaParse * parse, + GstMatroskaParseClass * klass) +{ + parse->sinkpad = gst_pad_new_from_static_template (&sink_templ, "sink"); + gst_pad_set_chain_function (parse->sinkpad, + GST_DEBUG_FUNCPTR (gst_matroska_parse_chain)); + gst_pad_set_event_function (parse->sinkpad, + GST_DEBUG_FUNCPTR (gst_matroska_parse_handle_sink_event)); + gst_element_add_pad (GST_ELEMENT (parse), parse->sinkpad); + + parse->srcpad = gst_pad_new_from_static_template (&src_templ, "src"); + gst_pad_set_event_function (parse->srcpad, + GST_DEBUG_FUNCPTR (gst_matroska_parse_handle_src_event)); + gst_pad_set_query_type_function (parse->srcpad, + GST_DEBUG_FUNCPTR (gst_matroska_parse_get_src_query_types)); + gst_pad_set_query_function (parse->srcpad, + GST_DEBUG_FUNCPTR (gst_matroska_parse_handle_src_query)); + gst_pad_use_fixed_caps (parse->srcpad); + gst_pad_set_caps (parse->srcpad, gst_caps_new_simple ("video/x-matroska", + NULL)); + gst_element_add_pad (GST_ELEMENT (parse), parse->srcpad); + + /* initial stream no. */ + parse->src = NULL; + + parse->writing_app = NULL; + parse->muxing_app = NULL; + parse->index = NULL; + parse->global_tags = NULL; + + parse->adapter = gst_adapter_new (); + + /* finish off */ + gst_matroska_parse_reset (GST_ELEMENT (parse)); +} + +static void +gst_matroska_track_free (GstMatroskaTrackContext * track) +{ + g_free (track->codec_id); + g_free (track->codec_name); + g_free (track->name); + g_free (track->language); + g_free (track->codec_priv); + g_free (track->codec_state); + + if (track->encodings != NULL) { + int i; + + for (i = 0; i < track->encodings->len; ++i) { + GstMatroskaTrackEncoding *enc = &g_array_index (track->encodings, + GstMatroskaTrackEncoding, + i); + + g_free (enc->comp_settings); + } + g_array_free (track->encodings, TRUE); + } + + if (track->pending_tags) + gst_tag_list_free (track->pending_tags); + + if (track->index_table) + g_array_free (track->index_table, TRUE); + + g_free (track); +} + +static void +gst_matroska_parse_free_parsed_el (gpointer mem, gpointer user_data) +{ + g_slice_free (guint64, mem); +} + +static void +gst_matroska_parse_reset (GstElement * element) +{ + GstMatroskaParse *parse = GST_MATROSKA_PARSE (element); + guint i; + + GST_DEBUG_OBJECT (parse, "Resetting state"); + + /* reset input */ + parse->state = GST_MATROSKA_PARSE_STATE_START; + + /* clean up existing streams */ + if (parse->src) { + g_assert (parse->src->len == parse->num_streams); + for (i = 0; i < parse->src->len; i++) { + GstMatroskaTrackContext *context = g_ptr_array_index (parse->src, i); + + gst_caps_replace (&context->caps, NULL); + gst_matroska_track_free (context); + } + g_ptr_array_free (parse->src, TRUE); + } + parse->src = g_ptr_array_new (); + + parse->num_streams = 0; + parse->num_a_streams = 0; + parse->num_t_streams = 0; + parse->num_v_streams = 0; + + /* reset media info */ + g_free (parse->writing_app); + parse->writing_app = NULL; + g_free (parse->muxing_app); + parse->muxing_app = NULL; + + /* reset indexes */ + if (parse->index) { + g_array_free (parse->index, TRUE); + parse->index = NULL; + } + + /* reset timers */ + parse->clock = NULL; + parse->time_scale = 1000000; + parse->created = G_MININT64; + + parse->index_parsed = FALSE; + parse->tracks_parsed = FALSE; + parse->segmentinfo_parsed = FALSE; + parse->attachments_parsed = FALSE; + + g_list_foreach (parse->tags_parsed, + (GFunc) gst_matroska_parse_free_parsed_el, NULL); + g_list_free (parse->tags_parsed); + parse->tags_parsed = NULL; + + g_list_foreach (parse->seek_parsed, + (GFunc) gst_matroska_parse_free_parsed_el, NULL); + g_list_free (parse->seek_parsed); + parse->seek_parsed = NULL; + + gst_segment_init (&parse->segment, GST_FORMAT_TIME); + parse->last_stop_end = GST_CLOCK_TIME_NONE; + parse->seek_block = 0; + + parse->offset = 0; + parse->cluster_time = GST_CLOCK_TIME_NONE; + parse->cluster_offset = 0; + parse->next_cluster_offset = 0; + parse->index_offset = 0; + parse->seekable = FALSE; + parse->need_newsegment = FALSE; + parse->building_index = FALSE; + if (parse->seek_event) { + gst_event_unref (parse->seek_event); + parse->seek_event = NULL; + } + + parse->seek_index = NULL; + parse->seek_entry = 0; + + if (parse->close_segment) { + gst_event_unref (parse->close_segment); + parse->close_segment = NULL; + } + + if (parse->new_segment) { + gst_event_unref (parse->new_segment); + parse->new_segment = NULL; + } + + if (parse->element_index) { + gst_object_unref (parse->element_index); + parse->element_index = NULL; + } + parse->element_index_writer_id = -1; + + if (parse->global_tags) { + gst_tag_list_free (parse->global_tags); + } + parse->global_tags = gst_tag_list_new (); + + if (parse->cached_buffer) { + gst_buffer_unref (parse->cached_buffer); + parse->cached_buffer = NULL; + } +} + +/* + * Calls pull_range for (offset,size) without advancing our offset + */ +static GstFlowReturn +gst_matroska_parse_peek_bytes (GstMatroskaParse * parse, guint64 offset, + guint size, GstBuffer ** p_buf, guint8 ** bytes) +{ + GstFlowReturn ret; + + /* Caching here actually makes much less difference than one would expect. + * We do it mainly to avoid pulling buffers of 1 byte all the time */ + if (parse->cached_buffer) { + guint64 cache_offset = GST_BUFFER_OFFSET (parse->cached_buffer); + guint cache_size = GST_BUFFER_SIZE (parse->cached_buffer); + + if (cache_offset <= parse->offset && + (parse->offset + size) <= (cache_offset + cache_size)) { + if (p_buf) + *p_buf = gst_buffer_create_sub (parse->cached_buffer, + parse->offset - cache_offset, size); + if (bytes) + *bytes = GST_BUFFER_DATA (parse->cached_buffer) + parse->offset - + cache_offset; + return GST_FLOW_OK; + } + /* not enough data in the cache, free cache and get a new one */ + gst_buffer_unref (parse->cached_buffer); + parse->cached_buffer = NULL; + } + + /* refill the cache */ + ret = gst_pad_pull_range (parse->sinkpad, parse->offset, + MAX (size, 64 * 1024), &parse->cached_buffer); + if (ret != GST_FLOW_OK) { + parse->cached_buffer = NULL; + return ret; + } + + if (GST_BUFFER_SIZE (parse->cached_buffer) >= size) { + if (p_buf) + *p_buf = gst_buffer_create_sub (parse->cached_buffer, 0, size); + if (bytes) + *bytes = GST_BUFFER_DATA (parse->cached_buffer); + return GST_FLOW_OK; + } + + /* Not possible to get enough data, try a last time with + * requesting exactly the size we need */ + gst_buffer_unref (parse->cached_buffer); + parse->cached_buffer = NULL; + + ret = + gst_pad_pull_range (parse->sinkpad, parse->offset, size, + &parse->cached_buffer); + if (ret != GST_FLOW_OK) { + GST_DEBUG_OBJECT (parse, "pull_range returned %d", ret); + if (p_buf) + *p_buf = NULL; + if (bytes) + *bytes = NULL; + return ret; + } + + if (GST_BUFFER_SIZE (parse->cached_buffer) < size) { + GST_WARNING_OBJECT (parse, "Dropping short buffer at offset %" + G_GUINT64_FORMAT ": wanted %u bytes, got %u bytes", parse->offset, + size, GST_BUFFER_SIZE (parse->cached_buffer)); + + gst_buffer_unref (parse->cached_buffer); + parse->cached_buffer = NULL; + if (p_buf) + *p_buf = NULL; + if (bytes) + *bytes = NULL; + return GST_FLOW_UNEXPECTED; + } + + if (p_buf) + *p_buf = gst_buffer_create_sub (parse->cached_buffer, 0, size); + if (bytes) + *bytes = GST_BUFFER_DATA (parse->cached_buffer); + + return GST_FLOW_OK; +} + +static const guint8 * +gst_matroska_parse_peek_pull (GstMatroskaParse * parse, guint peek) +{ + guint8 *data = NULL; + + gst_matroska_parse_peek_bytes (parse, parse->offset, peek, NULL, &data); + return data; +} + +static GstFlowReturn +gst_matroska_parse_peek_id_length_pull (GstMatroskaParse * parse, guint32 * _id, + guint64 * _length, guint * _needed) +{ + return gst_ebml_peek_id_length (_id, _length, _needed, + (GstPeekData) gst_matroska_parse_peek_pull, (gpointer) parse, + GST_ELEMENT_CAST (parse), parse->offset); +} + +static gint64 +gst_matroska_parse_get_length (GstMatroskaParse * parse) +{ + GstFormat fmt = GST_FORMAT_BYTES; + gint64 end = -1; + + if (!gst_pad_query_peer_duration (parse->sinkpad, &fmt, &end) || + fmt != GST_FORMAT_BYTES || end < 0) + GST_DEBUG_OBJECT (parse, "no upstream length"); + + return end; +} + +static gint +gst_matroska_parse_stream_from_num (GstMatroskaParse * parse, guint track_num) +{ + guint n; + + g_assert (parse->src->len == parse->num_streams); + for (n = 0; n < parse->src->len; n++) { + GstMatroskaTrackContext *context = g_ptr_array_index (parse->src, n); + + if (context->num == track_num) { + return n; + } + } + + if (n == parse->num_streams) + GST_WARNING_OBJECT (parse, + "Failed to find corresponding pad for tracknum %d", track_num); + + return -1; +} + +static gint +gst_matroska_parse_encoding_cmp (GstMatroskaTrackEncoding * a, + GstMatroskaTrackEncoding * b) +{ + if (b->order > a->order) + return 1; + else if (b->order < a->order) + return -1; + else + return 0; +} + +static gboolean +gst_matroska_parse_encoding_order_unique (GArray * encodings, guint64 order) +{ + gint i; + + if (encodings == NULL || encodings->len == 0) + return TRUE; + + for (i = 0; i < encodings->len; i++) + if (g_array_index (encodings, GstMatroskaTrackEncoding, i).order == order) + return FALSE; + + return TRUE; +} + +static GstFlowReturn +gst_matroska_parse_read_track_encoding (GstMatroskaParse * parse, + GstEbmlRead * ebml, GstMatroskaTrackContext * context) +{ + GstMatroskaTrackEncoding enc = { 0, }; + GstFlowReturn ret; + guint32 id; + + DEBUG_ELEMENT_START (parse, ebml, "ContentEncoding"); + /* Set default values */ + enc.scope = 1; + /* All other default values are 0 */ + + if ((ret = gst_ebml_read_master (ebml, &id)) != GST_FLOW_OK) { + DEBUG_ELEMENT_STOP (parse, ebml, "ContentEncoding", ret); + return ret; + } + + while (ret == GST_FLOW_OK && gst_ebml_read_has_remaining (ebml, 1, TRUE)) { + if ((ret = gst_ebml_peek_id (ebml, &id)) != GST_FLOW_OK) + break; + + switch (id) { + case GST_MATROSKA_ID_CONTENTENCODINGORDER:{ + guint64 num; + + if ((ret = gst_ebml_read_uint (ebml, &id, &num)) != GST_FLOW_OK) + break; + + if (!gst_matroska_parse_encoding_order_unique (context->encodings, num)) { + GST_ERROR_OBJECT (parse, "ContentEncodingOrder %" G_GUINT64_FORMAT + "is not unique for track %d", num, context->num); + ret = GST_FLOW_ERROR; + break; + } + + GST_DEBUG_OBJECT (parse, "ContentEncodingOrder: %" G_GUINT64_FORMAT, + num); + enc.order = num; + break; + } + case GST_MATROSKA_ID_CONTENTENCODINGSCOPE:{ + guint64 num; + + if ((ret = gst_ebml_read_uint (ebml, &id, &num)) != GST_FLOW_OK) + break; + + if (num > 7 && num == 0) { + GST_ERROR_OBJECT (parse, "Invalid ContentEncodingScope %" + G_GUINT64_FORMAT, num); + ret = GST_FLOW_ERROR; + break; + } + + GST_DEBUG_OBJECT (parse, "ContentEncodingScope: %" G_GUINT64_FORMAT, + num); + enc.scope = num; + + break; + } + case GST_MATROSKA_ID_CONTENTENCODINGTYPE:{ + guint64 num; + + if ((ret = gst_ebml_read_uint (ebml, &id, &num)) != GST_FLOW_OK) + break; + + if (num > 1) { + GST_ERROR_OBJECT (parse, "Invalid ContentEncodingType %" + G_GUINT64_FORMAT, num); + ret = GST_FLOW_ERROR; + break; + } else if (num != 0) { + GST_ERROR_OBJECT (parse, "Encrypted tracks are not supported yet"); + ret = GST_FLOW_ERROR; + break; + } + GST_DEBUG_OBJECT (parse, "ContentEncodingType: %" G_GUINT64_FORMAT, + num); + enc.type = num; + break; + } + case GST_MATROSKA_ID_CONTENTCOMPRESSION:{ + + DEBUG_ELEMENT_START (parse, ebml, "ContentCompression"); + + if ((ret = gst_ebml_read_master (ebml, &id)) != GST_FLOW_OK) + break; + + while (ret == GST_FLOW_OK && + gst_ebml_read_has_remaining (ebml, 1, TRUE)) { + if ((ret = gst_ebml_peek_id (ebml, &id)) != GST_FLOW_OK) + break; + + switch (id) { + case GST_MATROSKA_ID_CONTENTCOMPALGO:{ + guint64 num; + + if ((ret = gst_ebml_read_uint (ebml, &id, &num)) != GST_FLOW_OK) { + break; + } + if (num > 3) { + GST_ERROR_OBJECT (parse, "Invalid ContentCompAlgo %" + G_GUINT64_FORMAT, num); + ret = GST_FLOW_ERROR; + break; + } + GST_DEBUG_OBJECT (parse, "ContentCompAlgo: %" G_GUINT64_FORMAT, + num); + enc.comp_algo = num; + + break; + } + case GST_MATROSKA_ID_CONTENTCOMPSETTINGS:{ + guint8 *data; + guint64 size; + + if ((ret = + gst_ebml_read_binary (ebml, &id, &data, + &size)) != GST_FLOW_OK) { + break; + } + enc.comp_settings = data; + enc.comp_settings_length = size; + GST_DEBUG_OBJECT (parse, + "ContentCompSettings of size %" G_GUINT64_FORMAT, size); + break; + } + default: + GST_WARNING_OBJECT (parse, + "Unknown ContentCompression subelement 0x%x - ignoring", id); + ret = gst_ebml_read_skip (ebml); + break; + } + } + DEBUG_ELEMENT_STOP (parse, ebml, "ContentCompression", ret); + break; + } + + case GST_MATROSKA_ID_CONTENTENCRYPTION: + GST_ERROR_OBJECT (parse, "Encrypted tracks not yet supported"); + gst_ebml_read_skip (ebml); + ret = GST_FLOW_ERROR; + break; + default: + GST_WARNING_OBJECT (parse, + "Unknown ContentEncoding subelement 0x%x - ignoring", id); + ret = gst_ebml_read_skip (ebml); + break; + } + } + + DEBUG_ELEMENT_STOP (parse, ebml, "ContentEncoding", ret); + if (ret != GST_FLOW_OK && ret != GST_FLOW_UNEXPECTED) + return ret; + + /* TODO: Check if the combination of values is valid */ + + g_array_append_val (context->encodings, enc); + + return ret; +} + +static gboolean +gst_matroska_decompress_data (GstMatroskaTrackEncoding * enc, + guint8 ** data_out, guint * size_out, + GstMatroskaTrackCompressionAlgorithm algo) +{ + guint8 *new_data = NULL; + guint new_size = 0; + guint8 *data = *data_out; + guint size = *size_out; + gboolean ret = TRUE; + + if (algo == GST_MATROSKA_TRACK_COMPRESSION_ALGORITHM_ZLIB) { +#ifdef HAVE_ZLIB + /* zlib encoded data */ + z_stream zstream; + guint orig_size; + int result; + + orig_size = size; + zstream.zalloc = (alloc_func) 0; + zstream.zfree = (free_func) 0; + zstream.opaque = (voidpf) 0; + if (inflateInit (&zstream) != Z_OK) { + GST_WARNING ("zlib initialization failed."); + ret = FALSE; + goto out; + } + zstream.next_in = (Bytef *) data; + zstream.avail_in = orig_size; + new_size = orig_size; + new_data = g_malloc (new_size); + zstream.avail_out = new_size; + zstream.next_out = (Bytef *) new_data; + + do { + result = inflate (&zstream, Z_NO_FLUSH); + if (result != Z_OK && result != Z_STREAM_END) { + GST_WARNING ("zlib decompression failed."); + g_free (new_data); + inflateEnd (&zstream); + break; + } + new_size += 4000; + new_data = g_realloc (new_data, new_size); + zstream.next_out = (Bytef *) (new_data + zstream.total_out); + zstream.avail_out += 4000; + } while (zstream.avail_in != 0 && result != Z_STREAM_END); + + if (result != Z_STREAM_END) { + ret = FALSE; + goto out; + } else { + new_size = zstream.total_out; + inflateEnd (&zstream); + } +#else + GST_WARNING ("zlib encoded tracks not supported."); + ret = FALSE; + goto out; +#endif + } else if (algo == GST_MATROSKA_TRACK_COMPRESSION_ALGORITHM_BZLIB) { +#ifdef HAVE_BZ2 + /* bzip2 encoded data */ + bz_stream bzstream; + guint orig_size; + int result; + + bzstream.bzalloc = NULL; + bzstream.bzfree = NULL; + bzstream.opaque = NULL; + orig_size = size; + + if (BZ2_bzDecompressInit (&bzstream, 0, 0) != BZ_OK) { + GST_WARNING ("bzip2 initialization failed."); + ret = FALSE; + goto out; + } + + bzstream.next_in = (char *) data; + bzstream.avail_in = orig_size; + new_size = orig_size; + new_data = g_malloc (new_size); + bzstream.avail_out = new_size; + bzstream.next_out = (char *) new_data; + + do { + result = BZ2_bzDecompress (&bzstream); + if (result != BZ_OK && result != BZ_STREAM_END) { + GST_WARNING ("bzip2 decompression failed."); + g_free (new_data); + BZ2_bzDecompressEnd (&bzstream); + break; + } + new_size += 4000; + new_data = g_realloc (new_data, new_size); + bzstream.next_out = (char *) (new_data + bzstream.total_out_lo32); + bzstream.avail_out += 4000; + } while (bzstream.avail_in != 0 && result != BZ_STREAM_END); + + if (result != BZ_STREAM_END) { + ret = FALSE; + goto out; + } else { + new_size = bzstream.total_out_lo32; + BZ2_bzDecompressEnd (&bzstream); + } +#else + GST_WARNING ("bzip2 encoded tracks not supported."); + ret = FALSE; + goto out; +#endif + } else if (algo == GST_MATROSKA_TRACK_COMPRESSION_ALGORITHM_LZO1X) { + /* lzo encoded data */ + int result; + int orig_size, out_size; + + orig_size = size; + out_size = size; + new_size = size; + new_data = g_malloc (new_size); + + do { + orig_size = size; + out_size = new_size; + + result = lzo1x_decode (new_data, &out_size, data, &orig_size); + + if (orig_size > 0) { + new_size += 4000; + new_data = g_realloc (new_data, new_size); + } + } while (orig_size > 0 && result == LZO_OUTPUT_FULL); + + new_size -= out_size; + + if (result != LZO_OUTPUT_FULL) { + GST_WARNING ("lzo decompression failed"); + g_free (new_data); + + ret = FALSE; + goto out; + } + + } else if (algo == GST_MATROSKA_TRACK_COMPRESSION_ALGORITHM_HEADERSTRIP) { + /* header stripped encoded data */ + if (enc->comp_settings_length > 0) { + new_data = g_malloc (size + enc->comp_settings_length); + new_size = size + enc->comp_settings_length; + + memcpy (new_data, enc->comp_settings, enc->comp_settings_length); + memcpy (new_data + enc->comp_settings_length, data, size); + } + } else { + GST_ERROR ("invalid compression algorithm %d", algo); + ret = FALSE; + } + +out: + + if (!ret) { + *data_out = NULL; + *size_out = 0; + } else { + *data_out = new_data; + *size_out = new_size; + } + + return ret; +} + +static gboolean +gst_matroska_decode_data (GArray * encodings, guint8 ** data_out, + guint * size_out, GstMatroskaTrackEncodingScope scope, gboolean free) +{ + guint8 *data; + guint size; + gboolean ret = TRUE; + gint i; + + g_return_val_if_fail (encodings != NULL, FALSE); + g_return_val_if_fail (data_out != NULL && *data_out != NULL, FALSE); + g_return_val_if_fail (size_out != NULL, FALSE); + + data = *data_out; + size = *size_out; + + for (i = 0; i < encodings->len; i++) { + GstMatroskaTrackEncoding *enc = + &g_array_index (encodings, GstMatroskaTrackEncoding, i); + guint8 *new_data = NULL; + guint new_size = 0; + + if ((enc->scope & scope) == 0) + continue; + + /* Encryption not supported yet */ + if (enc->type != 0) { + ret = FALSE; + break; + } + + new_data = data; + new_size = size; + + ret = + gst_matroska_decompress_data (enc, &new_data, &new_size, + enc->comp_algo); + + if (!ret) + break; + + if ((data == *data_out && free) || (data != *data_out)) + g_free (data); + + data = new_data; + size = new_size; + } + + if (!ret) { + if ((data == *data_out && free) || (data != *data_out)) + g_free (data); + + *data_out = NULL; + *size_out = 0; + } else { + *data_out = data; + *size_out = size; + } + + return ret; +} + +static GstFlowReturn +gst_matroska_decode_content_encodings (GArray * encodings) +{ + gint i; + + if (encodings == NULL) + return GST_FLOW_OK; + + for (i = 0; i < encodings->len; i++) { + GstMatroskaTrackEncoding *enc = + &g_array_index (encodings, GstMatroskaTrackEncoding, i); + GstMatroskaTrackEncoding *enc2; + guint8 *data = NULL; + guint size; + + if ((enc->scope & GST_MATROSKA_TRACK_ENCODING_SCOPE_NEXT_CONTENT_ENCODING) + == 0) + continue; + + /* Encryption not supported yet */ + if (enc->type != 0) + return GST_FLOW_ERROR; + + if (i + 1 >= encodings->len) + return GST_FLOW_ERROR; + + enc2 = &g_array_index (encodings, GstMatroskaTrackEncoding, i + 1); + + if (enc->comp_settings_length == 0) + continue; + + data = enc->comp_settings; + size = enc->comp_settings_length; + + if (!gst_matroska_decompress_data (enc, &data, &size, enc->comp_algo)) + return GST_FLOW_ERROR; + + g_free (enc->comp_settings); + + enc->comp_settings = data; + enc->comp_settings_length = size; + } + + return GST_FLOW_OK; +} + +static GstFlowReturn +gst_matroska_parse_read_track_encodings (GstMatroskaParse * parse, + GstEbmlRead * ebml, GstMatroskaTrackContext * context) +{ + GstFlowReturn ret; + guint32 id; + + DEBUG_ELEMENT_START (parse, ebml, "ContentEncodings"); + + if ((ret = gst_ebml_read_master (ebml, &id)) != GST_FLOW_OK) { + DEBUG_ELEMENT_STOP (parse, ebml, "ContentEncodings", ret); + return ret; + } + + context->encodings = + g_array_sized_new (FALSE, FALSE, sizeof (GstMatroskaTrackEncoding), 1); + + while (ret == GST_FLOW_OK && gst_ebml_read_has_remaining (ebml, 1, TRUE)) { + if ((ret = gst_ebml_peek_id (ebml, &id)) != GST_FLOW_OK) + break; + + switch (id) { + case GST_MATROSKA_ID_CONTENTENCODING: + ret = gst_matroska_parse_read_track_encoding (parse, ebml, context); + break; + default: + GST_WARNING_OBJECT (parse, + "Unknown ContentEncodings subelement 0x%x - ignoring", id); + ret = gst_ebml_read_skip (ebml); + break; + } + } + + DEBUG_ELEMENT_STOP (parse, ebml, "ContentEncodings", ret); + if (ret != GST_FLOW_OK && ret != GST_FLOW_UNEXPECTED) + return ret; + + /* Sort encodings according to their order */ + g_array_sort (context->encodings, + (GCompareFunc) gst_matroska_parse_encoding_cmp); + + return gst_matroska_decode_content_encodings (context->encodings); +} + +static gboolean +gst_matroska_parse_tracknumber_unique (GstMatroskaParse * parse, guint64 num) +{ + gint i; + + g_assert (parse->src->len == parse->num_streams); + for (i = 0; i < parse->src->len; i++) { + GstMatroskaTrackContext *context = g_ptr_array_index (parse->src, i); + + if (context->num == num) + return FALSE; + } + + return TRUE; +} + +static GstFlowReturn +gst_matroska_parse_add_stream (GstMatroskaParse * parse, GstEbmlRead * ebml) +{ + GstMatroskaTrackContext *context; + GstFlowReturn ret; + guint32 id; + + DEBUG_ELEMENT_START (parse, ebml, "TrackEntry"); + + /* start with the master */ + if ((ret = gst_ebml_read_master (ebml, &id)) != GST_FLOW_OK) { + DEBUG_ELEMENT_STOP (parse, ebml, "TrackEntry", ret); + return ret; + } + + /* allocate generic... if we know the type, we'll g_renew() + * with the precise type */ + context = g_new0 (GstMatroskaTrackContext, 1); + g_ptr_array_add (parse->src, context); + context->index = parse->num_streams; + context->index_writer_id = -1; + context->type = 0; /* no type yet */ + context->default_duration = 0; + context->pos = 0; + context->set_discont = TRUE; + context->timecodescale = 1.0; + context->flags = + GST_MATROSKA_TRACK_ENABLED | GST_MATROSKA_TRACK_DEFAULT | + GST_MATROSKA_TRACK_LACING; + context->last_flow = GST_FLOW_OK; + context->to_offset = G_MAXINT64; + parse->num_streams++; + g_assert (parse->src->len == parse->num_streams); + + GST_DEBUG_OBJECT (parse, "Stream number %d", context->index); + + /* try reading the trackentry headers */ + while (ret == GST_FLOW_OK && gst_ebml_read_has_remaining (ebml, 1, TRUE)) { + if ((ret = gst_ebml_peek_id (ebml, &id)) != GST_FLOW_OK) + break; + + switch (id) { + /* track number (unique stream ID) */ + case GST_MATROSKA_ID_TRACKNUMBER:{ + guint64 num; + + if ((ret = gst_ebml_read_uint (ebml, &id, &num)) != GST_FLOW_OK) + break; + + if (num == 0) { + GST_ERROR_OBJECT (parse, "Invalid TrackNumber 0"); + ret = GST_FLOW_ERROR; + break; + } else if (!gst_matroska_parse_tracknumber_unique (parse, num)) { + GST_ERROR_OBJECT (parse, "TrackNumber %" G_GUINT64_FORMAT + " is not unique", num); + ret = GST_FLOW_ERROR; + break; + } + + GST_DEBUG_OBJECT (parse, "TrackNumber: %" G_GUINT64_FORMAT, num); + context->num = num; + break; + } + /* track UID (unique identifier) */ + case GST_MATROSKA_ID_TRACKUID:{ + guint64 num; + + if ((ret = gst_ebml_read_uint (ebml, &id, &num)) != GST_FLOW_OK) + break; + + if (num == 0) { + GST_ERROR_OBJECT (parse, "Invalid TrackUID 0"); + ret = GST_FLOW_ERROR; + break; + } + + GST_DEBUG_OBJECT (parse, "TrackUID: %" G_GUINT64_FORMAT, num); + context->uid = num; + break; + } + + /* track type (video, audio, combined, subtitle, etc.) */ + case GST_MATROSKA_ID_TRACKTYPE:{ + guint64 track_type; + + if ((ret = gst_ebml_read_uint (ebml, &id, &track_type)) != GST_FLOW_OK) { + break; + } + + if (context->type != 0 && context->type != track_type) { + GST_WARNING_OBJECT (parse, + "More than one tracktype defined in a TrackEntry - skipping"); + break; + } else if (track_type < 1 || track_type > 254) { + GST_WARNING_OBJECT (parse, "Invalid TrackType %" G_GUINT64_FORMAT, + track_type); + break; + } + + GST_DEBUG_OBJECT (parse, "TrackType: %" G_GUINT64_FORMAT, track_type); + + /* ok, so we're actually going to reallocate this thing */ + switch (track_type) { + case GST_MATROSKA_TRACK_TYPE_VIDEO: + gst_matroska_track_init_video_context (&context); + break; + case GST_MATROSKA_TRACK_TYPE_AUDIO: + gst_matroska_track_init_audio_context (&context); + break; + case GST_MATROSKA_TRACK_TYPE_SUBTITLE: + gst_matroska_track_init_subtitle_context (&context); + break; + case GST_MATROSKA_TRACK_TYPE_COMPLEX: + case GST_MATROSKA_TRACK_TYPE_LOGO: + case GST_MATROSKA_TRACK_TYPE_BUTTONS: + case GST_MATROSKA_TRACK_TYPE_CONTROL: + default: + GST_WARNING_OBJECT (parse, + "Unknown or unsupported TrackType %" G_GUINT64_FORMAT, + track_type); + context->type = 0; + break; + } + g_ptr_array_index (parse->src, parse->num_streams - 1) = context; + break; + } + + /* tracktype specific stuff for video */ + case GST_MATROSKA_ID_TRACKVIDEO:{ + GstMatroskaTrackVideoContext *videocontext; + + DEBUG_ELEMENT_START (parse, ebml, "TrackVideo"); + + if (!gst_matroska_track_init_video_context (&context)) { + GST_WARNING_OBJECT (parse, + "TrackVideo element in non-video track - ignoring track"); + ret = GST_FLOW_ERROR; + break; + } else if ((ret = gst_ebml_read_master (ebml, &id)) != GST_FLOW_OK) { + break; + } + videocontext = (GstMatroskaTrackVideoContext *) context; + g_ptr_array_index (parse->src, parse->num_streams - 1) = context; + + while (ret == GST_FLOW_OK && + gst_ebml_read_has_remaining (ebml, 1, TRUE)) { + if ((ret = gst_ebml_peek_id (ebml, &id)) != GST_FLOW_OK) + break; + + switch (id) { + /* Should be one level up but some broken muxers write it here. */ + case GST_MATROSKA_ID_TRACKDEFAULTDURATION:{ + guint64 num; + + if ((ret = gst_ebml_read_uint (ebml, &id, &num)) != GST_FLOW_OK) + break; + + if (num == 0) { + GST_WARNING_OBJECT (parse, "Invalid TrackDefaultDuration 0"); + break; + } + + GST_DEBUG_OBJECT (parse, + "TrackDefaultDuration: %" G_GUINT64_FORMAT, num); + context->default_duration = num; + break; + } + + /* video framerate */ + /* NOTE: This one is here only for backward compatibility. + * Use _TRACKDEFAULDURATION one level up. */ + case GST_MATROSKA_ID_VIDEOFRAMERATE:{ + gdouble num; + + if ((ret = gst_ebml_read_float (ebml, &id, &num)) != GST_FLOW_OK) + break; + + if (num <= 0.0) { + GST_WARNING_OBJECT (parse, "Invalid TrackVideoFPS %lf", num); + break; + } + + GST_DEBUG_OBJECT (parse, "TrackVideoFrameRate: %lf", num); + if (context->default_duration == 0) + context->default_duration = + gst_gdouble_to_guint64 ((gdouble) GST_SECOND * (1.0 / num)); + videocontext->default_fps = num; + break; + } + + /* width of the size to display the video at */ + case GST_MATROSKA_ID_VIDEODISPLAYWIDTH:{ + guint64 num; + + if ((ret = gst_ebml_read_uint (ebml, &id, &num)) != GST_FLOW_OK) + break; + + if (num == 0) { + GST_WARNING_OBJECT (parse, "Invalid TrackVideoDisplayWidth 0"); + break; + } + + GST_DEBUG_OBJECT (parse, + "TrackVideoDisplayWidth: %" G_GUINT64_FORMAT, num); + videocontext->display_width = num; + break; + } + + /* height of the size to display the video at */ + case GST_MATROSKA_ID_VIDEODISPLAYHEIGHT:{ + guint64 num; + + if ((ret = gst_ebml_read_uint (ebml, &id, &num)) != GST_FLOW_OK) + break; + + if (num == 0) { + GST_WARNING_OBJECT (parse, "Invalid TrackVideoDisplayHeight 0"); + break; + } + + GST_DEBUG_OBJECT (parse, + "TrackVideoDisplayHeight: %" G_GUINT64_FORMAT, num); + videocontext->display_height = num; + break; + } + + /* width of the video in the file */ + case GST_MATROSKA_ID_VIDEOPIXELWIDTH:{ + guint64 num; + + if ((ret = gst_ebml_read_uint (ebml, &id, &num)) != GST_FLOW_OK) + break; + + if (num == 0) { + GST_WARNING_OBJECT (parse, "Invalid TrackVideoPixelWidth 0"); + break; + } + + GST_DEBUG_OBJECT (parse, + "TrackVideoPixelWidth: %" G_GUINT64_FORMAT, num); + videocontext->pixel_width = num; + break; + } + + /* height of the video in the file */ + case GST_MATROSKA_ID_VIDEOPIXELHEIGHT:{ + guint64 num; + + if ((ret = gst_ebml_read_uint (ebml, &id, &num)) != GST_FLOW_OK) + break; + + if (num == 0) { + GST_WARNING_OBJECT (parse, "Invalid TrackVideoPixelHeight 0"); + break; + } + + GST_DEBUG_OBJECT (parse, + "TrackVideoPixelHeight: %" G_GUINT64_FORMAT, num); + videocontext->pixel_height = num; + break; + } + + /* whether the video is interlaced */ + case GST_MATROSKA_ID_VIDEOFLAGINTERLACED:{ + guint64 num; + + if ((ret = gst_ebml_read_uint (ebml, &id, &num)) != GST_FLOW_OK) + break; + + if (num) + context->flags |= GST_MATROSKA_VIDEOTRACK_INTERLACED; + else + context->flags &= ~GST_MATROSKA_VIDEOTRACK_INTERLACED; + GST_DEBUG_OBJECT (parse, "TrackVideoInterlaced: %d", + (context->flags & GST_MATROSKA_VIDEOTRACK_INTERLACED) ? 1 : + 0); + break; + } + + /* aspect ratio behaviour */ + case GST_MATROSKA_ID_VIDEOASPECTRATIOTYPE:{ + guint64 num; + + if ((ret = gst_ebml_read_uint (ebml, &id, &num)) != GST_FLOW_OK) + break; + + if (num != GST_MATROSKA_ASPECT_RATIO_MODE_FREE && + num != GST_MATROSKA_ASPECT_RATIO_MODE_KEEP && + num != GST_MATROSKA_ASPECT_RATIO_MODE_FIXED) { + GST_WARNING_OBJECT (parse, + "Unknown TrackVideoAspectRatioType 0x%x", (guint) num); + break; + } + GST_DEBUG_OBJECT (parse, + "TrackVideoAspectRatioType: %" G_GUINT64_FORMAT, num); + videocontext->asr_mode = num; + break; + } + + /* colourspace (only matters for raw video) fourcc */ + case GST_MATROSKA_ID_VIDEOCOLOURSPACE:{ + guint8 *data; + guint64 datalen; + + if ((ret = + gst_ebml_read_binary (ebml, &id, &data, + &datalen)) != GST_FLOW_OK) + break; + + if (datalen != 4) { + g_free (data); + GST_WARNING_OBJECT (parse, + "Invalid TrackVideoColourSpace length %" G_GUINT64_FORMAT, + datalen); + break; + } + + memcpy (&videocontext->fourcc, data, 4); + GST_DEBUG_OBJECT (parse, + "TrackVideoColourSpace: %" GST_FOURCC_FORMAT, + GST_FOURCC_ARGS (videocontext->fourcc)); + g_free (data); + break; + } + + default: + GST_WARNING_OBJECT (parse, + "Unknown TrackVideo subelement 0x%x - ignoring", id); + /* fall through */ + case GST_MATROSKA_ID_VIDEOSTEREOMODE: + case GST_MATROSKA_ID_VIDEODISPLAYUNIT: + case GST_MATROSKA_ID_VIDEOPIXELCROPBOTTOM: + case GST_MATROSKA_ID_VIDEOPIXELCROPTOP: + case GST_MATROSKA_ID_VIDEOPIXELCROPLEFT: + case GST_MATROSKA_ID_VIDEOPIXELCROPRIGHT: + case GST_MATROSKA_ID_VIDEOGAMMAVALUE: + ret = gst_ebml_read_skip (ebml); + break; + } + } + + DEBUG_ELEMENT_STOP (parse, ebml, "TrackVideo", ret); + break; + } + + /* tracktype specific stuff for audio */ + case GST_MATROSKA_ID_TRACKAUDIO:{ + GstMatroskaTrackAudioContext *audiocontext; + + DEBUG_ELEMENT_START (parse, ebml, "TrackAudio"); + + if (!gst_matroska_track_init_audio_context (&context)) { + GST_WARNING_OBJECT (parse, + "TrackAudio element in non-audio track - ignoring track"); + ret = GST_FLOW_ERROR; + break; + } + + if ((ret = gst_ebml_read_master (ebml, &id)) != GST_FLOW_OK) + break; + + audiocontext = (GstMatroskaTrackAudioContext *) context; + g_ptr_array_index (parse->src, parse->num_streams - 1) = context; + + while (ret == GST_FLOW_OK && + gst_ebml_read_has_remaining (ebml, 1, TRUE)) { + if ((ret = gst_ebml_peek_id (ebml, &id)) != GST_FLOW_OK) + break; + + switch (id) { + /* samplerate */ + case GST_MATROSKA_ID_AUDIOSAMPLINGFREQ:{ + gdouble num; + + if ((ret = gst_ebml_read_float (ebml, &id, &num)) != GST_FLOW_OK) + break; + + + if (num <= 0.0) { + GST_WARNING_OBJECT (parse, + "Invalid TrackAudioSamplingFrequency %lf", num); + break; + } + + GST_DEBUG_OBJECT (parse, "TrackAudioSamplingFrequency: %lf", num); + audiocontext->samplerate = num; + break; + } + + /* bitdepth */ + case GST_MATROSKA_ID_AUDIOBITDEPTH:{ + guint64 num; + + if ((ret = gst_ebml_read_uint (ebml, &id, &num)) != GST_FLOW_OK) + break; + + if (num == 0) { + GST_WARNING_OBJECT (parse, "Invalid TrackAudioBitDepth 0"); + break; + } + + GST_DEBUG_OBJECT (parse, "TrackAudioBitDepth: %" G_GUINT64_FORMAT, + num); + audiocontext->bitdepth = num; + break; + } + + /* channels */ + case GST_MATROSKA_ID_AUDIOCHANNELS:{ + guint64 num; + + if ((ret = gst_ebml_read_uint (ebml, &id, &num)) != GST_FLOW_OK) + break; + + if (num == 0) { + GST_WARNING_OBJECT (parse, "Invalid TrackAudioChannels 0"); + break; + } + + GST_DEBUG_OBJECT (parse, "TrackAudioChannels: %" G_GUINT64_FORMAT, + num); + audiocontext->channels = num; + break; + } + + default: + GST_WARNING_OBJECT (parse, + "Unknown TrackAudio subelement 0x%x - ignoring", id); + /* fall through */ + case GST_MATROSKA_ID_AUDIOCHANNELPOSITIONS: + case GST_MATROSKA_ID_AUDIOOUTPUTSAMPLINGFREQ: + ret = gst_ebml_read_skip (ebml); + break; + } + } + + DEBUG_ELEMENT_STOP (parse, ebml, "TrackAudio", ret); + + break; + } + + /* codec identifier */ + case GST_MATROSKA_ID_CODECID:{ + gchar *text; + + if ((ret = gst_ebml_read_ascii (ebml, &id, &text)) != GST_FLOW_OK) + break; + + GST_DEBUG_OBJECT (parse, "CodecID: %s", GST_STR_NULL (text)); + context->codec_id = text; + break; + } + + /* codec private data */ + case GST_MATROSKA_ID_CODECPRIVATE:{ + guint8 *data; + guint64 size; + + if ((ret = + gst_ebml_read_binary (ebml, &id, &data, &size)) != GST_FLOW_OK) + break; + + context->codec_priv = data; + context->codec_priv_size = size; + + GST_DEBUG_OBJECT (parse, "CodecPrivate of size %" G_GUINT64_FORMAT, + size); + break; + } + + /* name of the codec */ + case GST_MATROSKA_ID_CODECNAME:{ + gchar *text; + + if ((ret = gst_ebml_read_utf8 (ebml, &id, &text)) != GST_FLOW_OK) + break; + + GST_DEBUG_OBJECT (parse, "CodecName: %s", GST_STR_NULL (text)); + context->codec_name = text; + break; + } + + /* name of this track */ + case GST_MATROSKA_ID_TRACKNAME:{ + gchar *text; + + if ((ret = gst_ebml_read_utf8 (ebml, &id, &text)) != GST_FLOW_OK) + break; + + context->name = text; + GST_DEBUG_OBJECT (parse, "TrackName: %s", GST_STR_NULL (text)); + break; + } + + /* language (matters for audio/subtitles, mostly) */ + case GST_MATROSKA_ID_TRACKLANGUAGE:{ + gchar *text; + + if ((ret = gst_ebml_read_utf8 (ebml, &id, &text)) != GST_FLOW_OK) + break; + + + context->language = text; + + /* fre-ca => fre */ + if (strlen (context->language) >= 4 && context->language[3] == '-') + context->language[3] = '\0'; + + GST_DEBUG_OBJECT (parse, "TrackLanguage: %s", + GST_STR_NULL (context->language)); + break; + } + + /* whether this is actually used */ + case GST_MATROSKA_ID_TRACKFLAGENABLED:{ + guint64 num; + + if ((ret = gst_ebml_read_uint (ebml, &id, &num)) != GST_FLOW_OK) + break; + + if (num) + context->flags |= GST_MATROSKA_TRACK_ENABLED; + else + context->flags &= ~GST_MATROSKA_TRACK_ENABLED; + + GST_DEBUG_OBJECT (parse, "TrackEnabled: %d", + (context->flags & GST_MATROSKA_TRACK_ENABLED) ? 1 : 0); + break; + } + + /* whether it's the default for this track type */ + case GST_MATROSKA_ID_TRACKFLAGDEFAULT:{ + guint64 num; + + if ((ret = gst_ebml_read_uint (ebml, &id, &num)) != GST_FLOW_OK) + break; + + if (num) + context->flags |= GST_MATROSKA_TRACK_DEFAULT; + else + context->flags &= ~GST_MATROSKA_TRACK_DEFAULT; + + GST_DEBUG_OBJECT (parse, "TrackDefault: %d", + (context->flags & GST_MATROSKA_TRACK_ENABLED) ? 1 : 0); + break; + } + + /* whether the track must be used during playback */ + case GST_MATROSKA_ID_TRACKFLAGFORCED:{ + guint64 num; + + if ((ret = gst_ebml_read_uint (ebml, &id, &num)) != GST_FLOW_OK) + break; + + if (num) + context->flags |= GST_MATROSKA_TRACK_FORCED; + else + context->flags &= ~GST_MATROSKA_TRACK_FORCED; + + GST_DEBUG_OBJECT (parse, "TrackForced: %d", + (context->flags & GST_MATROSKA_TRACK_ENABLED) ? 1 : 0); + break; + } + + /* lacing (like MPEG, where blocks don't end/start on frame + * boundaries) */ + case GST_MATROSKA_ID_TRACKFLAGLACING:{ + guint64 num; + + if ((ret = gst_ebml_read_uint (ebml, &id, &num)) != GST_FLOW_OK) + break; + + if (num) + context->flags |= GST_MATROSKA_TRACK_LACING; + else + context->flags &= ~GST_MATROSKA_TRACK_LACING; + + GST_DEBUG_OBJECT (parse, "TrackLacing: %d", + (context->flags & GST_MATROSKA_TRACK_ENABLED) ? 1 : 0); + break; + } + + /* default length (in time) of one data block in this track */ + case GST_MATROSKA_ID_TRACKDEFAULTDURATION:{ + guint64 num; + + if ((ret = gst_ebml_read_uint (ebml, &id, &num)) != GST_FLOW_OK) + break; + + + if (num == 0) { + GST_WARNING_OBJECT (parse, "Invalid TrackDefaultDuration 0"); + break; + } + + GST_DEBUG_OBJECT (parse, "TrackDefaultDuration: %" G_GUINT64_FORMAT, + num); + context->default_duration = num; + break; + } + + case GST_MATROSKA_ID_CONTENTENCODINGS:{ + ret = gst_matroska_parse_read_track_encodings (parse, ebml, context); + break; + } + + case GST_MATROSKA_ID_TRACKTIMECODESCALE:{ + gdouble num; + + if ((ret = gst_ebml_read_float (ebml, &id, &num)) != GST_FLOW_OK) + break; + + if (num <= 0.0) { + GST_WARNING_OBJECT (parse, "Invalid TrackTimeCodeScale %lf", num); + break; + } + + GST_DEBUG_OBJECT (parse, "TrackTimeCodeScale: %lf", num); + context->timecodescale = num; + break; + } + + default: + GST_WARNING ("Unknown TrackEntry subelement 0x%x - ignoring", id); + /* pass-through */ + + /* we ignore these because they're nothing useful (i.e. crap) + * or simply not implemented yet. */ + case GST_MATROSKA_ID_TRACKMINCACHE: + case GST_MATROSKA_ID_TRACKMAXCACHE: + case GST_MATROSKA_ID_MAXBLOCKADDITIONID: + case GST_MATROSKA_ID_TRACKATTACHMENTLINK: + case GST_MATROSKA_ID_TRACKOVERLAY: + case GST_MATROSKA_ID_TRACKTRANSLATE: + case GST_MATROSKA_ID_TRACKOFFSET: + case GST_MATROSKA_ID_CODECSETTINGS: + case GST_MATROSKA_ID_CODECINFOURL: + case GST_MATROSKA_ID_CODECDOWNLOADURL: + case GST_MATROSKA_ID_CODECDECODEALL: + ret = gst_ebml_read_skip (ebml); + break; + } + } + + DEBUG_ELEMENT_STOP (parse, ebml, "TrackEntry", ret); + + /* Decode codec private data if necessary */ + if (context->encodings && context->encodings->len > 0 && context->codec_priv + && context->codec_priv_size > 0) { + if (!gst_matroska_decode_data (context->encodings, + &context->codec_priv, &context->codec_priv_size, + GST_MATROSKA_TRACK_ENCODING_SCOPE_CODEC_DATA, TRUE)) { + GST_WARNING_OBJECT (parse, "Decoding codec private data failed"); + ret = GST_FLOW_ERROR; + } + } + + if (context->type == 0 || context->codec_id == NULL || (ret != GST_FLOW_OK + && ret != GST_FLOW_UNEXPECTED)) { + if (ret == GST_FLOW_OK || ret == GST_FLOW_UNEXPECTED) + GST_WARNING_OBJECT (ebml, "Unknown stream/codec in track entry header"); + + parse->num_streams--; + g_ptr_array_remove_index (parse->src, parse->num_streams); + g_assert (parse->src->len == parse->num_streams); + if (context) { + gst_matroska_track_free (context); + } + + return ret; + } + + if ((context->language == NULL || *context->language == '\0') && + (context->type == GST_MATROSKA_TRACK_TYPE_AUDIO || + context->type == GST_MATROSKA_TRACK_TYPE_SUBTITLE)) { + GST_LOG ("stream %d: language=eng (assuming default)", context->index); + context->language = g_strdup ("eng"); + } + + + /* tadaah! */ + return ret; +} + +static const GstQueryType * +gst_matroska_parse_get_src_query_types (GstPad * pad) +{ + static const GstQueryType query_types[] = { + GST_QUERY_POSITION, + GST_QUERY_DURATION, + GST_QUERY_SEEKING, + 0 + }; + + return query_types; +} + +static gboolean +gst_matroska_parse_query (GstMatroskaParse * parse, GstPad * pad, + GstQuery * query) +{ + gboolean res = FALSE; + GstMatroskaTrackContext *context = NULL; + + if (pad) { + context = gst_pad_get_element_private (pad); + } + + switch (GST_QUERY_TYPE (query)) { + case GST_QUERY_POSITION: + { + GstFormat format; + + gst_query_parse_position (query, &format, NULL); + + if (format == GST_FORMAT_TIME) { + GST_OBJECT_LOCK (parse); + if (context) + gst_query_set_position (query, GST_FORMAT_TIME, context->pos); + else + gst_query_set_position (query, GST_FORMAT_TIME, + parse->segment.last_stop); + GST_OBJECT_UNLOCK (parse); + } else if (format == GST_FORMAT_DEFAULT && context + && context->default_duration) { + GST_OBJECT_LOCK (parse); + gst_query_set_position (query, GST_FORMAT_DEFAULT, + context->pos / context->default_duration); + GST_OBJECT_UNLOCK (parse); + } else { + GST_DEBUG_OBJECT (parse, + "only position query in TIME and DEFAULT format is supported"); + } + + res = TRUE; + break; + } + case GST_QUERY_DURATION: + { + GstFormat format; + + gst_query_parse_duration (query, &format, NULL); + + if (format == GST_FORMAT_TIME) { + GST_OBJECT_LOCK (parse); + gst_query_set_duration (query, GST_FORMAT_TIME, + parse->segment.duration); + GST_OBJECT_UNLOCK (parse); + } else if (format == GST_FORMAT_DEFAULT && context + && context->default_duration) { + GST_OBJECT_LOCK (parse); + gst_query_set_duration (query, GST_FORMAT_DEFAULT, + parse->segment.duration / context->default_duration); + GST_OBJECT_UNLOCK (parse); + } else { + GST_DEBUG_OBJECT (parse, + "only duration query in TIME and DEFAULT format is supported"); + } + + res = TRUE; + break; + } + + case GST_QUERY_SEEKING: + { + GstFormat fmt; + + gst_query_parse_seeking (query, &fmt, NULL, NULL, NULL); + if (fmt == GST_FORMAT_TIME) { + gboolean seekable; + + /* assuming we'll be able to get an index ... */ + seekable = parse->seekable; + + gst_query_set_seeking (query, GST_FORMAT_TIME, seekable, + 0, parse->segment.duration); + res = TRUE; + } + break; + } + default: + res = gst_pad_query_default (pad, query); + break; + } + + return res; +} + +static gboolean +gst_matroska_parse_element_query (GstElement * element, GstQuery * query) +{ + return gst_matroska_parse_query (GST_MATROSKA_PARSE (element), NULL, query); +} + +static gboolean +gst_matroska_parse_handle_src_query (GstPad * pad, GstQuery * query) +{ + gboolean ret; + GstMatroskaParse *parse = GST_MATROSKA_PARSE (gst_pad_get_parent (pad)); + + ret = gst_matroska_parse_query (parse, pad, query); + + gst_object_unref (parse); + + return ret; +} + +static gint +gst_matroska_index_seek_find (GstMatroskaIndex * i1, GstClockTime * time, + gpointer user_data) +{ + if (i1->time < *time) + return -1; + else if (i1->time > *time) + return 1; + else + return 0; +} + +static GstMatroskaIndex * +gst_matroskaparse_do_index_seek (GstMatroskaParse * parse, + GstMatroskaTrackContext * track, gint64 seek_pos, GArray ** _index, + gint * _entry_index) +{ + GstMatroskaIndex *entry = NULL; + GArray *index; + + if (!parse->index || !parse->index->len) + return NULL; + + /* find entry just before or at the requested position */ + if (track && track->index_table) + index = track->index_table; + else + index = parse->index; + + entry = + gst_util_array_binary_search (index->data, index->len, + sizeof (GstMatroskaIndex), + (GCompareDataFunc) gst_matroska_index_seek_find, GST_SEARCH_MODE_BEFORE, + &seek_pos, NULL); + + if (entry == NULL) + entry = &g_array_index (index, GstMatroskaIndex, 0); + + if (_index) + *_index = index; + if (_entry_index) + *_entry_index = entry - (GstMatroskaIndex *) index->data; + + return entry; +} + +/* takes ownership of taglist */ +static void +gst_matroska_parse_found_global_tag (GstMatroskaParse * parse, + GstTagList * taglist) +{ + if (parse->global_tags) { + /* nothing sent yet, add to cache */ + gst_tag_list_insert (parse->global_tags, taglist, GST_TAG_MERGE_APPEND); + gst_tag_list_free (taglist); + } else { + /* hm, already sent, no need to cache and wait anymore */ + GST_DEBUG_OBJECT (parse, "Sending late global tags %" GST_PTR_FORMAT, + taglist); + gst_element_found_tags (GST_ELEMENT (parse), taglist); + } +} + +/* returns FALSE if there are no pads to deliver event to, + * otherwise TRUE (whatever the outcome of event sending), + * takes ownership of the passed event! */ +static gboolean +gst_matroska_parse_send_event (GstMatroskaParse * parse, GstEvent * event) +{ + gboolean ret = FALSE; + + g_return_val_if_fail (event != NULL, FALSE); + + GST_DEBUG_OBJECT (parse, "Sending event of type %s to all source pads", + GST_EVENT_TYPE_NAME (event)); + + gst_pad_push_event (parse->srcpad, event); + + return ret; +} + +static gboolean +gst_matroska_parse_element_send_event (GstElement * element, GstEvent * event) +{ + GstMatroskaParse *parse = GST_MATROSKA_PARSE (element); + gboolean res; + + g_return_val_if_fail (event != NULL, FALSE); + + if (GST_EVENT_TYPE (event) == GST_EVENT_SEEK) { + res = gst_matroska_parse_handle_seek_event (parse, NULL, event); + } else { + GST_WARNING_OBJECT (parse, "Unhandled event of type %s", + GST_EVENT_TYPE_NAME (event)); + res = FALSE; + } + gst_event_unref (event); + return res; +} + +/* determine track to seek in */ +static GstMatroskaTrackContext * +gst_matroska_parse_get_seek_track (GstMatroskaParse * parse, + GstMatroskaTrackContext * track) +{ + gint i; + + if (track && track->type == GST_MATROSKA_TRACK_TYPE_VIDEO) + return track; + + for (i = 0; i < parse->src->len; i++) { + GstMatroskaTrackContext *stream; + + stream = g_ptr_array_index (parse->src, i); + if (stream->type == GST_MATROSKA_TRACK_TYPE_VIDEO && stream->index_table) + track = stream; + } + + return track; +} + +static void +gst_matroska_parse_reset_streams (GstMatroskaParse * parse, GstClockTime time, + gboolean full) +{ + gint i; + + GST_DEBUG_OBJECT (parse, "resetting stream state"); + + g_assert (parse->src->len == parse->num_streams); + for (i = 0; i < parse->src->len; i++) { + GstMatroskaTrackContext *context = g_ptr_array_index (parse->src, i); + context->pos = time; + context->set_discont = TRUE; + context->eos = FALSE; + context->from_time = GST_CLOCK_TIME_NONE; + if (full) + context->last_flow = GST_FLOW_OK; + if (context->type == GST_MATROSKA_TRACK_TYPE_VIDEO) { + GstMatroskaTrackVideoContext *videocontext = + (GstMatroskaTrackVideoContext *) context; + /* parse object lock held by caller */ + videocontext->earliest_time = GST_CLOCK_TIME_NONE; + } + } +} + +/* searches for a cluster start from @pos, + * return GST_FLOW_OK and cluster position in @pos if found */ +static GstFlowReturn +gst_matroska_parse_search_cluster (GstMatroskaParse * parse, gint64 * pos) +{ + gint64 newpos = *pos; + gint64 orig_offset; + GstFlowReturn ret = GST_FLOW_OK; + const guint chunk = 64 * 1024; + GstBuffer *buf = NULL; + guint64 length; + guint32 id; + guint needed; + + orig_offset = parse->offset; + + /* read in at newpos and scan for ebml cluster id */ + while (1) { + GstByteReader reader; + gint cluster_pos; + + ret = gst_pad_pull_range (parse->sinkpad, newpos, chunk, &buf); + if (ret != GST_FLOW_OK) + break; + GST_DEBUG_OBJECT (parse, "read buffer size %d at offset %" G_GINT64_FORMAT, + GST_BUFFER_SIZE (buf), newpos); + gst_byte_reader_init_from_buffer (&reader, buf); + cluster_pos = 0; + resume: + cluster_pos = gst_byte_reader_masked_scan_uint32 (&reader, 0xffffffff, + GST_MATROSKA_ID_CLUSTER, cluster_pos, + GST_BUFFER_SIZE (buf) - cluster_pos); + if (cluster_pos >= 0) { + newpos += cluster_pos; + GST_DEBUG_OBJECT (parse, + "found cluster ebml id at offset %" G_GINT64_FORMAT, newpos); + /* extra checks whether we really sync'ed to a cluster: + * - either it is the first and only cluster + * - either there is a cluster after this one + * - either cluster length is undefined + */ + /* ok if first cluster (there may not a subsequent one) */ + if (newpos == parse->first_cluster_offset) { + GST_DEBUG_OBJECT (parse, "cluster is first cluster -> OK"); + break; + } + parse->offset = newpos; + ret = + gst_matroska_parse_peek_id_length_pull (parse, &id, &length, &needed); + if (ret != GST_FLOW_OK) + goto resume; + g_assert (id == GST_MATROSKA_ID_CLUSTER); + GST_DEBUG_OBJECT (parse, "cluster size %" G_GUINT64_FORMAT ", prefix %d", + length, needed); + /* ok if undefined length or first cluster */ + if (length == G_MAXUINT64) { + GST_DEBUG_OBJECT (parse, "cluster has undefined length -> OK"); + break; + } + /* skip cluster */ + parse->offset += length + needed; + ret = + gst_matroska_parse_peek_id_length_pull (parse, &id, &length, &needed); + if (ret != GST_FLOW_OK) + goto resume; + GST_DEBUG_OBJECT (parse, "next element is %scluster", + id == GST_MATROSKA_ID_CLUSTER ? "" : "not "); + if (id == GST_MATROSKA_ID_CLUSTER) + break; + /* not ok, resume */ + goto resume; + } else { + /* partial cluster id may have been in tail of buffer */ + newpos += MAX (GST_BUFFER_SIZE (buf), 4) - 3; + gst_buffer_unref (buf); + buf = NULL; + } + } + + if (buf) { + gst_buffer_unref (buf); + buf = NULL; + } + + parse->offset = orig_offset; + *pos = newpos; + return ret; +} + + +static gboolean +gst_matroska_parse_handle_seek_event (GstMatroskaParse * parse, + GstPad * pad, GstEvent * event) +{ + GstMatroskaIndex *entry = NULL; + GstSeekFlags flags; + GstSeekType cur_type, stop_type; + GstFormat format; + gdouble rate; + gint64 cur, stop; + GstMatroskaTrackContext *track = NULL; + GstSegment seeksegment = { 0, }; + gboolean update; + + if (pad) + track = gst_pad_get_element_private (pad); + + track = gst_matroska_parse_get_seek_track (parse, track); + + gst_event_parse_seek (event, &rate, &format, &flags, &cur_type, &cur, + &stop_type, &stop); + + /* we can only seek on time */ + if (format != GST_FORMAT_TIME) { + GST_DEBUG_OBJECT (parse, "Can only seek on TIME"); + return FALSE; + } + + /* copy segment, we need this because we still need the old + * segment when we close the current segment. */ + memcpy (&seeksegment, &parse->segment, sizeof (GstSegment)); + + if (event) { + GST_DEBUG_OBJECT (parse, "configuring seek"); + gst_segment_set_seek (&seeksegment, rate, format, flags, + cur_type, cur, stop_type, stop, &update); + } + + GST_DEBUG_OBJECT (parse, "New segment %" GST_SEGMENT_FORMAT, &seeksegment); + + /* check sanity before we start flushing and all that */ + GST_OBJECT_LOCK (parse); + if ((entry = gst_matroskaparse_do_index_seek (parse, track, + seeksegment.last_stop, &parse->seek_index, &parse->seek_entry)) == + NULL) { + /* pull mode without index can scan later on */ + GST_DEBUG_OBJECT (parse, "No matching seek entry in index"); + GST_OBJECT_UNLOCK (parse); + return FALSE; + } + GST_DEBUG_OBJECT (parse, "Seek position looks sane"); + GST_OBJECT_UNLOCK (parse); + + /* need to seek to cluster start to pick up cluster time */ + /* upstream takes care of flushing and all that + * ... and newsegment event handling takes care of the rest */ + return perform_seek_to_offset (parse, entry->pos + parse->ebml_segment_start); +} + +/* + * Handle whether we can perform the seek event or if we have to let the chain + * function handle seeks to build the seek indexes first. + */ +static gboolean +gst_matroska_parse_handle_seek_push (GstMatroskaParse * parse, GstPad * pad, + GstEvent * event) +{ + GstSeekFlags flags; + GstSeekType cur_type, stop_type; + GstFormat format; + gdouble rate; + gint64 cur, stop; + + gst_event_parse_seek (event, &rate, &format, &flags, &cur_type, &cur, + &stop_type, &stop); + + /* sanity checks */ + + /* we can only seek on time */ + if (format != GST_FORMAT_TIME) { + GST_DEBUG_OBJECT (parse, "Can only seek on TIME"); + return FALSE; + } + + if (stop_type != GST_SEEK_TYPE_NONE && stop != GST_CLOCK_TIME_NONE) { + GST_DEBUG_OBJECT (parse, "Seek end-time not supported in streaming mode"); + return FALSE; + } + + if (!(flags & GST_SEEK_FLAG_FLUSH)) { + GST_DEBUG_OBJECT (parse, + "Non-flushing seek not supported in streaming mode"); + return FALSE; + } + + if (flags & GST_SEEK_FLAG_SEGMENT) { + GST_DEBUG_OBJECT (parse, "Segment seek not supported in streaming mode"); + return FALSE; + } + + /* check for having parsed index already */ + if (!parse->index_parsed) { + gboolean building_index; + guint64 offset = 0; + + if (!parse->index_offset) { + GST_DEBUG_OBJECT (parse, "no index (location); no seek in push mode"); + return FALSE; + } + + GST_OBJECT_LOCK (parse); + /* handle the seek event in the chain function */ + parse->state = GST_MATROSKA_PARSE_STATE_SEEK; + /* no more seek can be issued until state reset to _DATA */ + + /* copy the event */ + if (parse->seek_event) + gst_event_unref (parse->seek_event); + parse->seek_event = gst_event_ref (event); + + /* set the building_index flag so that only one thread can setup the + * structures for index seeking. */ + building_index = parse->building_index; + if (!building_index) { + parse->building_index = TRUE; + offset = parse->index_offset; + } + GST_OBJECT_UNLOCK (parse); + + if (!building_index) { + /* seek to the first subindex or legacy index */ + GST_INFO_OBJECT (parse, "Seeking to Cues at %" G_GUINT64_FORMAT, offset); + return perform_seek_to_offset (parse, offset); + } + + /* well, we are handling it already */ + return TRUE; + } + + /* delegate to tweaked regular seek */ + return gst_matroska_parse_handle_seek_event (parse, pad, event); +} + +static gboolean +gst_matroska_parse_handle_src_event (GstPad * pad, GstEvent * event) +{ + GstMatroskaParse *parse = GST_MATROSKA_PARSE (gst_pad_get_parent (pad)); + gboolean res = TRUE; + + switch (GST_EVENT_TYPE (event)) { + case GST_EVENT_SEEK: + /* no seeking until we are (safely) ready */ + if (parse->state != GST_MATROSKA_PARSE_STATE_DATA) { + GST_DEBUG_OBJECT (parse, "not ready for seeking yet"); + return FALSE; + } + res = gst_matroska_parse_handle_seek_push (parse, pad, event); + gst_event_unref (event); + break; + + case GST_EVENT_QOS: + { + GstMatroskaTrackContext *context = gst_pad_get_element_private (pad); + if (context->type == GST_MATROSKA_TRACK_TYPE_VIDEO) { + GstMatroskaTrackVideoContext *videocontext = + (GstMatroskaTrackVideoContext *) context; + gdouble proportion; + GstClockTimeDiff diff; + GstClockTime timestamp; + + gst_event_parse_qos (event, &proportion, &diff, ×tamp); + + GST_OBJECT_LOCK (parse); + videocontext->earliest_time = timestamp + diff; + GST_OBJECT_UNLOCK (parse); + } + res = TRUE; + gst_event_unref (event); + break; + } + + /* events we don't need to handle */ + case GST_EVENT_NAVIGATION: + gst_event_unref (event); + res = FALSE; + break; + + case GST_EVENT_LATENCY: + default: + res = gst_pad_push_event (parse->sinkpad, event); + break; + } + + gst_object_unref (parse); + + return res; +} + + +/* skip unknown or alike element */ +static GstFlowReturn +gst_matroska_parse_parse_skip (GstMatroskaParse * parse, GstEbmlRead * ebml, + const gchar * parent_name, guint id) +{ + if (id == GST_EBML_ID_VOID) { + GST_DEBUG_OBJECT (parse, "Skipping EBML Void element"); + } else if (id == GST_EBML_ID_CRC32) { + GST_DEBUG_OBJECT (parse, "Skipping EBML CRC32 element"); + } else { + GST_WARNING_OBJECT (parse, + "Unknown %s subelement 0x%x - ignoring", parent_name, id); + } + + return gst_ebml_read_skip (ebml); +} + +static GstFlowReturn +gst_matroska_parse_parse_header (GstMatroskaParse * parse, GstEbmlRead * ebml) +{ + GstFlowReturn ret; + gchar *doctype; + guint version; + guint32 id; + + /* this function is the first to be called */ + + /* default init */ + doctype = NULL; + version = 1; + + ret = gst_ebml_peek_id (ebml, &id); + if (ret != GST_FLOW_OK) + return ret; + + GST_DEBUG_OBJECT (parse, "id: %08x", id); + + if (id != GST_EBML_ID_HEADER) { + GST_ERROR_OBJECT (parse, "Failed to read header"); + goto exit; + } + + ret = gst_ebml_read_master (ebml, &id); + if (ret != GST_FLOW_OK) + return ret; + + while (gst_ebml_read_has_remaining (ebml, 1, TRUE)) { + ret = gst_ebml_peek_id (ebml, &id); + if (ret != GST_FLOW_OK) + return ret; + + switch (id) { + /* is our read version uptodate? */ + case GST_EBML_ID_EBMLREADVERSION:{ + guint64 num; + + ret = gst_ebml_read_uint (ebml, &id, &num); + if (ret != GST_FLOW_OK) + return ret; + if (num != GST_EBML_VERSION) { + GST_ERROR_OBJECT (ebml, "Unsupported EBML version %" G_GUINT64_FORMAT, + num); + return GST_FLOW_ERROR; + } + + GST_DEBUG_OBJECT (ebml, "EbmlReadVersion: %" G_GUINT64_FORMAT, num); + break; + } + + /* we only handle 8 byte lengths at max */ + case GST_EBML_ID_EBMLMAXSIZELENGTH:{ + guint64 num; + + ret = gst_ebml_read_uint (ebml, &id, &num); + if (ret != GST_FLOW_OK) + return ret; + if (num > sizeof (guint64)) { + GST_ERROR_OBJECT (ebml, + "Unsupported EBML maximum size %" G_GUINT64_FORMAT, num); + return GST_FLOW_ERROR; + } + GST_DEBUG_OBJECT (ebml, "EbmlMaxSizeLength: %" G_GUINT64_FORMAT, num); + break; + } + + /* we handle 4 byte IDs at max */ + case GST_EBML_ID_EBMLMAXIDLENGTH:{ + guint64 num; + + ret = gst_ebml_read_uint (ebml, &id, &num); + if (ret != GST_FLOW_OK) + return ret; + if (num > sizeof (guint32)) { + GST_ERROR_OBJECT (ebml, + "Unsupported EBML maximum ID %" G_GUINT64_FORMAT, num); + return GST_FLOW_ERROR; + } + GST_DEBUG_OBJECT (ebml, "EbmlMaxIdLength: %" G_GUINT64_FORMAT, num); + break; + } + + case GST_EBML_ID_DOCTYPE:{ + gchar *text; + + ret = gst_ebml_read_ascii (ebml, &id, &text); + if (ret != GST_FLOW_OK) + return ret; + + GST_DEBUG_OBJECT (ebml, "EbmlDocType: %s", GST_STR_NULL (text)); + + if (doctype) + g_free (doctype); + doctype = text; + break; + } + + case GST_EBML_ID_DOCTYPEREADVERSION:{ + guint64 num; + + ret = gst_ebml_read_uint (ebml, &id, &num); + if (ret != GST_FLOW_OK) + return ret; + version = num; + GST_DEBUG_OBJECT (ebml, "EbmlReadVersion: %" G_GUINT64_FORMAT, num); + break; + } + + default: + ret = gst_matroska_parse_parse_skip (parse, ebml, "EBML header", id); + if (ret != GST_FLOW_OK) + return ret; + break; + + /* we ignore these two, as they don't tell us anything we care about */ + case GST_EBML_ID_EBMLVERSION: + case GST_EBML_ID_DOCTYPEVERSION: + ret = gst_ebml_read_skip (ebml); + if (ret != GST_FLOW_OK) + return ret; + break; + } + } + +exit: + + if ((doctype != NULL && !strcmp (doctype, GST_MATROSKA_DOCTYPE_MATROSKA)) || + (doctype != NULL && !strcmp (doctype, GST_MATROSKA_DOCTYPE_WEBM)) || + (doctype == NULL)) { + if (version <= 2) { + if (doctype) { + GST_INFO_OBJECT (parse, "Input is %s version %d", doctype, version); + } else { + GST_WARNING_OBJECT (parse, "Input is EBML without doctype, assuming " + "matroska (version %d)", version); + } + ret = GST_FLOW_OK; + } else { + GST_ELEMENT_ERROR (parse, STREAM, DEMUX, (NULL), + ("Parser version (2) is too old to read %s version %d", + GST_STR_NULL (doctype), version)); + ret = GST_FLOW_ERROR; + } + g_free (doctype); + } else { + GST_ELEMENT_ERROR (parse, STREAM, WRONG_TYPE, (NULL), + ("Input is not a matroska stream (doctype=%s)", doctype)); + ret = GST_FLOW_ERROR; + g_free (doctype); + } + + return ret; +} + +static GstFlowReturn +gst_matroska_parse_parse_tracks (GstMatroskaParse * parse, GstEbmlRead * ebml) +{ + GstFlowReturn ret = GST_FLOW_OK; + guint32 id; + + DEBUG_ELEMENT_START (parse, ebml, "Tracks"); + + if ((ret = gst_ebml_read_master (ebml, &id)) != GST_FLOW_OK) { + DEBUG_ELEMENT_STOP (parse, ebml, "Tracks", ret); + return ret; + } + + while (ret == GST_FLOW_OK && gst_ebml_read_has_remaining (ebml, 1, TRUE)) { + if ((ret = gst_ebml_peek_id (ebml, &id)) != GST_FLOW_OK) + break; + + switch (id) { + /* one track within the "all-tracks" header */ + case GST_MATROSKA_ID_TRACKENTRY: + ret = gst_matroska_parse_add_stream (parse, ebml); + break; + + default: + ret = gst_matroska_parse_parse_skip (parse, ebml, "Track", id); + break; + } + } + DEBUG_ELEMENT_STOP (parse, ebml, "Tracks", ret); + + parse->tracks_parsed = TRUE; + + return ret; +} + +static GstFlowReturn +gst_matroska_parse_parse_index_cuetrack (GstMatroskaParse * parse, + GstEbmlRead * ebml, guint * nentries) +{ + guint32 id; + GstFlowReturn ret; + GstMatroskaIndex idx; + + idx.pos = (guint64) - 1; + idx.track = 0; + idx.time = GST_CLOCK_TIME_NONE; + idx.block = 1; + + DEBUG_ELEMENT_START (parse, ebml, "CueTrackPositions"); + + if ((ret = gst_ebml_read_master (ebml, &id)) != GST_FLOW_OK) { + DEBUG_ELEMENT_STOP (parse, ebml, "CueTrackPositions", ret); + return ret; + } + + while (ret == GST_FLOW_OK && gst_ebml_read_has_remaining (ebml, 1, TRUE)) { + if ((ret = gst_ebml_peek_id (ebml, &id)) != GST_FLOW_OK) + break; + + switch (id) { + /* track number */ + case GST_MATROSKA_ID_CUETRACK: + { + guint64 num; + + if ((ret = gst_ebml_read_uint (ebml, &id, &num)) != GST_FLOW_OK) + break; + + if (num == 0) { + idx.track = 0; + GST_WARNING_OBJECT (parse, "Invalid CueTrack 0"); + break; + } + + GST_DEBUG_OBJECT (parse, "CueTrack: %" G_GUINT64_FORMAT, num); + idx.track = num; + break; + } + + /* position in file */ + case GST_MATROSKA_ID_CUECLUSTERPOSITION: + { + guint64 num; + + if ((ret = gst_ebml_read_uint (ebml, &id, &num)) != GST_FLOW_OK) + break; + + if (num > G_MAXINT64) { + GST_WARNING_OBJECT (parse, "CueClusterPosition %" G_GUINT64_FORMAT + " too large", num); + break; + } + + idx.pos = num; + break; + } + + /* number of block in the cluster */ + case GST_MATROSKA_ID_CUEBLOCKNUMBER: + { + guint64 num; + + if ((ret = gst_ebml_read_uint (ebml, &id, &num)) != GST_FLOW_OK) + break; + + if (num == 0) { + GST_WARNING_OBJECT (parse, "Invalid CueBlockNumber 0"); + break; + } + + GST_DEBUG_OBJECT (parse, "CueBlockNumber: %" G_GUINT64_FORMAT, num); + idx.block = num; + + /* mild sanity check, disregard strange cases ... */ + if (idx.block > G_MAXUINT16) { + GST_DEBUG_OBJECT (parse, "... looks suspicious, ignoring"); + idx.block = 1; + } + break; + } + + default: + ret = gst_matroska_parse_parse_skip (parse, ebml, "CueTrackPositions", + id); + break; + + case GST_MATROSKA_ID_CUECODECSTATE: + case GST_MATROSKA_ID_CUEREFERENCE: + ret = gst_ebml_read_skip (ebml); + break; + } + } + + DEBUG_ELEMENT_STOP (parse, ebml, "CueTrackPositions", ret); + + if ((ret == GST_FLOW_OK || ret == GST_FLOW_UNEXPECTED) + && idx.pos != (guint64) - 1 && idx.track > 0) { + g_array_append_val (parse->index, idx); + (*nentries)++; + } else if (ret == GST_FLOW_OK || ret == GST_FLOW_UNEXPECTED) { + GST_DEBUG_OBJECT (parse, "CueTrackPositions without valid content"); + } + + return ret; +} + +static GstFlowReturn +gst_matroska_parse_parse_index_pointentry (GstMatroskaParse * parse, + GstEbmlRead * ebml) +{ + guint32 id; + GstFlowReturn ret; + GstClockTime time = GST_CLOCK_TIME_NONE; + guint nentries = 0; + + DEBUG_ELEMENT_START (parse, ebml, "CuePoint"); + + if ((ret = gst_ebml_read_master (ebml, &id)) != GST_FLOW_OK) { + DEBUG_ELEMENT_STOP (parse, ebml, "CuePoint", ret); + return ret; + } + + while (ret == GST_FLOW_OK && gst_ebml_read_has_remaining (ebml, 1, TRUE)) { + if ((ret = gst_ebml_peek_id (ebml, &id)) != GST_FLOW_OK) + break; + + switch (id) { + /* one single index entry ('point') */ + case GST_MATROSKA_ID_CUETIME: + { + if ((ret = gst_ebml_read_uint (ebml, &id, &time)) != GST_FLOW_OK) + break; + + GST_DEBUG_OBJECT (parse, "CueTime: %" G_GUINT64_FORMAT, time); + time = time * parse->time_scale; + break; + } + + /* position in the file + track to which it belongs */ + case GST_MATROSKA_ID_CUETRACKPOSITIONS: + { + if ((ret = + gst_matroska_parse_parse_index_cuetrack (parse, ebml, + &nentries)) != GST_FLOW_OK) + break; + break; + } + + default: + ret = gst_matroska_parse_parse_skip (parse, ebml, "CuePoint", id); + break; + } + } + + DEBUG_ELEMENT_STOP (parse, ebml, "CuePoint", ret); + + if (nentries > 0) { + if (time == GST_CLOCK_TIME_NONE) { + GST_WARNING_OBJECT (parse, "CuePoint without valid time"); + g_array_remove_range (parse->index, parse->index->len - nentries, + nentries); + } else { + gint i; + + for (i = parse->index->len - nentries; i < parse->index->len; i++) { + GstMatroskaIndex *idx = + &g_array_index (parse->index, GstMatroskaIndex, i); + + idx->time = time; + GST_DEBUG_OBJECT (parse, "Index entry: pos=%" G_GUINT64_FORMAT + ", time=%" GST_TIME_FORMAT ", track=%u, block=%u", idx->pos, + GST_TIME_ARGS (idx->time), (guint) idx->track, (guint) idx->block); + } + } + } else { + GST_DEBUG_OBJECT (parse, "Empty CuePoint"); + } + + return ret; +} + +static gint +gst_matroska_index_compare (GstMatroskaIndex * i1, GstMatroskaIndex * i2) +{ + if (i1->time < i2->time) + return -1; + else if (i1->time > i2->time) + return 1; + else if (i1->block < i2->block) + return -1; + else if (i1->block > i2->block) + return 1; + else + return 0; +} + +static GstFlowReturn +gst_matroska_parse_parse_index (GstMatroskaParse * parse, GstEbmlRead * ebml) +{ + guint32 id; + GstFlowReturn ret = GST_FLOW_OK; + guint i; + + if (parse->index) + g_array_free (parse->index, TRUE); + parse->index = + g_array_sized_new (FALSE, FALSE, sizeof (GstMatroskaIndex), 128); + + DEBUG_ELEMENT_START (parse, ebml, "Cues"); + + if ((ret = gst_ebml_read_master (ebml, &id)) != GST_FLOW_OK) { + DEBUG_ELEMENT_STOP (parse, ebml, "Cues", ret); + return ret; + } + + while (ret == GST_FLOW_OK && gst_ebml_read_has_remaining (ebml, 1, TRUE)) { + if ((ret = gst_ebml_peek_id (ebml, &id)) != GST_FLOW_OK) + break; + + switch (id) { + /* one single index entry ('point') */ + case GST_MATROSKA_ID_POINTENTRY: + ret = gst_matroska_parse_parse_index_pointentry (parse, ebml); + break; + + default: + ret = gst_matroska_parse_parse_skip (parse, ebml, "Cues", id); + break; + } + } + DEBUG_ELEMENT_STOP (parse, ebml, "Cues", ret); + + /* Sort index by time, smallest time first, for easier searching */ + g_array_sort (parse->index, (GCompareFunc) gst_matroska_index_compare); + + /* Now sort the track specific index entries into their own arrays */ + for (i = 0; i < parse->index->len; i++) { + GstMatroskaIndex *idx = &g_array_index (parse->index, GstMatroskaIndex, i); + gint track_num; + GstMatroskaTrackContext *ctx; + + if (parse->element_index) { + gint writer_id; + + if (idx->track != 0 && + (track_num = + gst_matroska_parse_stream_from_num (parse, idx->track)) != -1) { + ctx = g_ptr_array_index (parse->src, track_num); + + if (ctx->index_writer_id == -1) + gst_index_get_writer_id (parse->element_index, GST_OBJECT (ctx->pad), + &ctx->index_writer_id); + writer_id = ctx->index_writer_id; + } else { + if (parse->element_index_writer_id == -1) + gst_index_get_writer_id (parse->element_index, GST_OBJECT (parse), + &parse->element_index_writer_id); + writer_id = parse->element_index_writer_id; + } + + GST_LOG_OBJECT (parse, "adding association %" GST_TIME_FORMAT "-> %" + G_GUINT64_FORMAT " for writer id %d", GST_TIME_ARGS (idx->time), + idx->pos, writer_id); + gst_index_add_association (parse->element_index, writer_id, + GST_ASSOCIATION_FLAG_KEY_UNIT, GST_FORMAT_TIME, idx->time, + GST_FORMAT_BYTES, idx->pos + parse->ebml_segment_start, NULL); + } + + if (idx->track == 0) + continue; + + track_num = gst_matroska_parse_stream_from_num (parse, idx->track); + if (track_num == -1) + continue; + + ctx = g_ptr_array_index (parse->src, track_num); + + if (ctx->index_table == NULL) + ctx->index_table = + g_array_sized_new (FALSE, FALSE, sizeof (GstMatroskaIndex), 128); + + g_array_append_vals (ctx->index_table, idx, 1); + } + + parse->index_parsed = TRUE; + + /* sanity check; empty index normalizes to no index */ + if (parse->index->len == 0) { + g_array_free (parse->index, TRUE); + parse->index = NULL; + } + + return ret; +} + +static GstFlowReturn +gst_matroska_parse_parse_info (GstMatroskaParse * parse, GstEbmlRead * ebml) +{ + GstFlowReturn ret = GST_FLOW_OK; + guint32 id; + + DEBUG_ELEMENT_START (parse, ebml, "SegmentInfo"); + + if ((ret = gst_ebml_read_master (ebml, &id)) != GST_FLOW_OK) { + DEBUG_ELEMENT_STOP (parse, ebml, "SegmentInfo", ret); + return ret; + } + + while (ret == GST_FLOW_OK && gst_ebml_read_has_remaining (ebml, 1, TRUE)) { + if ((ret = gst_ebml_peek_id (ebml, &id)) != GST_FLOW_OK) + break; + + switch (id) { + /* cluster timecode */ + case GST_MATROSKA_ID_TIMECODESCALE:{ + guint64 num; + + if ((ret = gst_ebml_read_uint (ebml, &id, &num)) != GST_FLOW_OK) + break; + + + GST_DEBUG_OBJECT (parse, "TimeCodeScale: %" G_GUINT64_FORMAT, num); + parse->time_scale = num; + break; + } + + case GST_MATROSKA_ID_DURATION:{ + gdouble num; + GstClockTime dur; + + if ((ret = gst_ebml_read_float (ebml, &id, &num)) != GST_FLOW_OK) + break; + + if (num <= 0.0) { + GST_WARNING_OBJECT (parse, "Invalid duration %lf", num); + break; + } + + GST_DEBUG_OBJECT (parse, "Duration: %lf", num); + + dur = gst_gdouble_to_guint64 (num * + gst_guint64_to_gdouble (parse->time_scale)); + if (GST_CLOCK_TIME_IS_VALID (dur) && dur <= G_MAXINT64) + gst_segment_set_duration (&parse->segment, GST_FORMAT_TIME, dur); + break; + } + + case GST_MATROSKA_ID_WRITINGAPP:{ + gchar *text; + + if ((ret = gst_ebml_read_utf8 (ebml, &id, &text)) != GST_FLOW_OK) + break; + + GST_DEBUG_OBJECT (parse, "WritingApp: %s", GST_STR_NULL (text)); + parse->writing_app = text; + break; + } + + case GST_MATROSKA_ID_MUXINGAPP:{ + gchar *text; + + if ((ret = gst_ebml_read_utf8 (ebml, &id, &text)) != GST_FLOW_OK) + break; + + GST_DEBUG_OBJECT (parse, "MuxingApp: %s", GST_STR_NULL (text)); + parse->muxing_app = text; + break; + } + + case GST_MATROSKA_ID_DATEUTC:{ + gint64 time; + + if ((ret = gst_ebml_read_date (ebml, &id, &time)) != GST_FLOW_OK) + break; + + GST_DEBUG_OBJECT (parse, "DateUTC: %" G_GINT64_FORMAT, time); + parse->created = time; + break; + } + + case GST_MATROSKA_ID_TITLE:{ + gchar *text; + GstTagList *taglist; + + if ((ret = gst_ebml_read_utf8 (ebml, &id, &text)) != GST_FLOW_OK) + break; + + GST_DEBUG_OBJECT (parse, "Title: %s", GST_STR_NULL (text)); + taglist = gst_tag_list_new (); + gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND, GST_TAG_TITLE, text, + NULL); + gst_matroska_parse_found_global_tag (parse, taglist); + g_free (text); + break; + } + + default: + ret = gst_matroska_parse_parse_skip (parse, ebml, "SegmentInfo", id); + break; + + /* fall through */ + case GST_MATROSKA_ID_SEGMENTUID: + case GST_MATROSKA_ID_SEGMENTFILENAME: + case GST_MATROSKA_ID_PREVUID: + case GST_MATROSKA_ID_PREVFILENAME: + case GST_MATROSKA_ID_NEXTUID: + case GST_MATROSKA_ID_NEXTFILENAME: + case GST_MATROSKA_ID_SEGMENTFAMILY: + case GST_MATROSKA_ID_CHAPTERTRANSLATE: + ret = gst_ebml_read_skip (ebml); + break; + } + } + + DEBUG_ELEMENT_STOP (parse, ebml, "SegmentInfo", ret); + + parse->segmentinfo_parsed = TRUE; + + return ret; +} + +static GstFlowReturn +gst_matroska_parse_parse_metadata_id_simple_tag (GstMatroskaParse * parse, + GstEbmlRead * ebml, GstTagList ** p_taglist) +{ + /* FIXME: check if there are more useful mappings */ + struct + { + const gchar *matroska_tagname; + const gchar *gstreamer_tagname; + } + tag_conv[] = { + { + GST_MATROSKA_TAG_ID_TITLE, GST_TAG_TITLE}, { + GST_MATROSKA_TAG_ID_AUTHOR, GST_TAG_ARTIST}, { + GST_MATROSKA_TAG_ID_ALBUM, GST_TAG_ALBUM}, { + GST_MATROSKA_TAG_ID_COMMENTS, GST_TAG_COMMENT}, { + GST_MATROSKA_TAG_ID_BITSPS, GST_TAG_BITRATE}, { + GST_MATROSKA_TAG_ID_BPS, GST_TAG_BITRATE}, { + GST_MATROSKA_TAG_ID_ENCODER, GST_TAG_ENCODER}, { + GST_MATROSKA_TAG_ID_DATE, GST_TAG_DATE}, { + GST_MATROSKA_TAG_ID_ISRC, GST_TAG_ISRC}, { + GST_MATROSKA_TAG_ID_COPYRIGHT, GST_TAG_COPYRIGHT}, { + GST_MATROSKA_TAG_ID_BPM, GST_TAG_BEATS_PER_MINUTE}, { + GST_MATROSKA_TAG_ID_TERMS_OF_USE, GST_TAG_LICENSE}, { + GST_MATROSKA_TAG_ID_COMPOSER, GST_TAG_COMPOSER}, { + GST_MATROSKA_TAG_ID_LEAD_PERFORMER, GST_TAG_PERFORMER}, { + GST_MATROSKA_TAG_ID_GENRE, GST_TAG_GENRE} + }; + GstFlowReturn ret; + guint32 id; + gchar *value = NULL; + gchar *tag = NULL; + + DEBUG_ELEMENT_START (parse, ebml, "SimpleTag"); + + if ((ret = gst_ebml_read_master (ebml, &id)) != GST_FLOW_OK) { + DEBUG_ELEMENT_STOP (parse, ebml, "SimpleTag", ret); + return ret; + } + + while (ret == GST_FLOW_OK && gst_ebml_read_has_remaining (ebml, 1, TRUE)) { + /* read all sub-entries */ + + if ((ret = gst_ebml_peek_id (ebml, &id)) != GST_FLOW_OK) + break; + + switch (id) { + case GST_MATROSKA_ID_TAGNAME: + g_free (tag); + tag = NULL; + ret = gst_ebml_read_ascii (ebml, &id, &tag); + GST_DEBUG_OBJECT (parse, "TagName: %s", GST_STR_NULL (tag)); + break; + + case GST_MATROSKA_ID_TAGSTRING: + g_free (value); + value = NULL; + ret = gst_ebml_read_utf8 (ebml, &id, &value); + GST_DEBUG_OBJECT (parse, "TagString: %s", GST_STR_NULL (value)); + break; + + default: + ret = gst_matroska_parse_parse_skip (parse, ebml, "SimpleTag", id); + break; + /* fall-through */ + + case GST_MATROSKA_ID_TAGLANGUAGE: + case GST_MATROSKA_ID_TAGDEFAULT: + case GST_MATROSKA_ID_TAGBINARY: + ret = gst_ebml_read_skip (ebml); + break; + } + } + + DEBUG_ELEMENT_STOP (parse, ebml, "SimpleTag", ret); + + if (tag && value) { + guint i; + + for (i = 0; i < G_N_ELEMENTS (tag_conv); i++) { + const gchar *tagname_gst = tag_conv[i].gstreamer_tagname; + + const gchar *tagname_mkv = tag_conv[i].matroska_tagname; + + if (strcmp (tagname_mkv, tag) == 0) { + GValue dest = { 0, }; + GType dest_type = gst_tag_get_type (tagname_gst); + + /* Ensure that any date string is complete */ + if (dest_type == GST_TYPE_DATE) { + guint year = 1901, month = 1, day = 1; + + /* Dates can be yyyy-MM-dd, yyyy-MM or yyyy, but we need + * the first type */ + if (sscanf (value, "%04u-%02u-%02u", &year, &month, &day) != 0) { + g_free (value); + value = g_strdup_printf ("%04u-%02u-%02u", year, month, day); + } + } + + g_value_init (&dest, dest_type); + if (gst_value_deserialize (&dest, value)) { + gst_tag_list_add_values (*p_taglist, GST_TAG_MERGE_APPEND, + tagname_gst, &dest, NULL); + } else { + GST_WARNING_OBJECT (parse, "Can't transform tag '%s' with " + "value '%s' to target type '%s'", tag, value, + g_type_name (dest_type)); + } + g_value_unset (&dest); + break; + } + } + } + + g_free (tag); + g_free (value); + + return ret; +} + +static GstFlowReturn +gst_matroska_parse_parse_metadata_id_tag (GstMatroskaParse * parse, + GstEbmlRead * ebml, GstTagList ** p_taglist) +{ + guint32 id; + GstFlowReturn ret; + + DEBUG_ELEMENT_START (parse, ebml, "Tag"); + + if ((ret = gst_ebml_read_master (ebml, &id)) != GST_FLOW_OK) { + DEBUG_ELEMENT_STOP (parse, ebml, "Tag", ret); + return ret; + } + + while (ret == GST_FLOW_OK && gst_ebml_read_has_remaining (ebml, 1, TRUE)) { + /* read all sub-entries */ + + if ((ret = gst_ebml_peek_id (ebml, &id)) != GST_FLOW_OK) + break; + + switch (id) { + case GST_MATROSKA_ID_SIMPLETAG: + ret = gst_matroska_parse_parse_metadata_id_simple_tag (parse, ebml, + p_taglist); + break; + + default: + ret = gst_matroska_parse_parse_skip (parse, ebml, "Tag", id); + break; + } + } + + DEBUG_ELEMENT_STOP (parse, ebml, "Tag", ret); + + return ret; +} + +static GstFlowReturn +gst_matroska_parse_parse_metadata (GstMatroskaParse * parse, GstEbmlRead * ebml) +{ + GstTagList *taglist; + GstFlowReturn ret = GST_FLOW_OK; + guint32 id; + GList *l; + guint64 curpos; + + curpos = gst_ebml_read_get_pos (ebml); + + /* Make sure we don't parse a tags element twice and + * post it's tags twice */ + curpos = gst_ebml_read_get_pos (ebml); + for (l = parse->tags_parsed; l; l = l->next) { + guint64 *pos = l->data; + + if (*pos == curpos) { + GST_DEBUG_OBJECT (parse, "Skipping already parsed Tags at offset %" + G_GUINT64_FORMAT, curpos); + return GST_FLOW_OK; + } + } + + parse->tags_parsed = + g_list_prepend (parse->tags_parsed, g_slice_new (guint64)); + *((guint64 *) parse->tags_parsed->data) = curpos; + /* fall-through */ + + if ((ret = gst_ebml_read_master (ebml, &id)) != GST_FLOW_OK) { + DEBUG_ELEMENT_STOP (parse, ebml, "Tags", ret); + return ret; + } + + taglist = gst_tag_list_new (); + + while (ret == GST_FLOW_OK && gst_ebml_read_has_remaining (ebml, 1, TRUE)) { + if ((ret = gst_ebml_peek_id (ebml, &id)) != GST_FLOW_OK) + break; + + switch (id) { + case GST_MATROSKA_ID_TAG: + ret = gst_matroska_parse_parse_metadata_id_tag (parse, ebml, &taglist); + break; + + default: + ret = gst_matroska_parse_parse_skip (parse, ebml, "Tags", id); + break; + /* FIXME: Use to limit the tags to specific pads */ + case GST_MATROSKA_ID_TARGETS: + ret = gst_ebml_read_skip (ebml); + break; + } + } + + DEBUG_ELEMENT_STOP (parse, ebml, "Tags", ret); + + gst_matroska_parse_found_global_tag (parse, taglist); + + return ret; +} + +static GstFlowReturn +gst_matroska_parse_parse_attached_file (GstMatroskaParse * parse, + GstEbmlRead * ebml, GstTagList * taglist) +{ + guint32 id; + GstFlowReturn ret; + gchar *description = NULL; + gchar *filename = NULL; + gchar *mimetype = NULL; + guint8 *data = NULL; + guint64 datalen = 0; + + DEBUG_ELEMENT_START (parse, ebml, "AttachedFile"); + + if ((ret = gst_ebml_read_master (ebml, &id)) != GST_FLOW_OK) { + DEBUG_ELEMENT_STOP (parse, ebml, "AttachedFile", ret); + return ret; + } + + while (ret == GST_FLOW_OK && gst_ebml_read_has_remaining (ebml, 1, TRUE)) { + /* read all sub-entries */ + + if ((ret = gst_ebml_peek_id (ebml, &id)) != GST_FLOW_OK) + break; + + switch (id) { + case GST_MATROSKA_ID_FILEDESCRIPTION: + if (description) { + GST_WARNING_OBJECT (parse, "FileDescription can only appear once"); + break; + } + + ret = gst_ebml_read_utf8 (ebml, &id, &description); + GST_DEBUG_OBJECT (parse, "FileDescription: %s", + GST_STR_NULL (description)); + break; + case GST_MATROSKA_ID_FILENAME: + if (filename) { + GST_WARNING_OBJECT (parse, "FileName can only appear once"); + break; + } + + ret = gst_ebml_read_utf8 (ebml, &id, &filename); + + GST_DEBUG_OBJECT (parse, "FileName: %s", GST_STR_NULL (filename)); + break; + case GST_MATROSKA_ID_FILEMIMETYPE: + if (mimetype) { + GST_WARNING_OBJECT (parse, "FileMimeType can only appear once"); + break; + } + + ret = gst_ebml_read_ascii (ebml, &id, &mimetype); + GST_DEBUG_OBJECT (parse, "FileMimeType: %s", GST_STR_NULL (mimetype)); + break; + case GST_MATROSKA_ID_FILEDATA: + if (data) { + GST_WARNING_OBJECT (parse, "FileData can only appear once"); + break; + } + + ret = gst_ebml_read_binary (ebml, &id, &data, &datalen); + GST_DEBUG_OBJECT (parse, "FileData of size %" G_GUINT64_FORMAT, + datalen); + break; + + default: + ret = gst_matroska_parse_parse_skip (parse, ebml, "AttachedFile", id); + break; + case GST_MATROSKA_ID_FILEUID: + ret = gst_ebml_read_skip (ebml); + break; + } + } + + DEBUG_ELEMENT_STOP (parse, ebml, "AttachedFile", ret); + + if (filename && mimetype && data && datalen > 0) { + GstTagImageType image_type = GST_TAG_IMAGE_TYPE_NONE; + GstBuffer *tagbuffer = NULL; + GstCaps *caps; + gchar *filename_lc = g_utf8_strdown (filename, -1); + + GST_DEBUG_OBJECT (parse, "Creating tag for attachment with filename '%s', " + "mimetype '%s', description '%s', size %" G_GUINT64_FORMAT, filename, + mimetype, GST_STR_NULL (description), datalen); + + /* TODO: better heuristics for different image types */ + if (strstr (filename_lc, "cover")) { + if (strstr (filename_lc, "back")) + image_type = GST_TAG_IMAGE_TYPE_BACK_COVER; + else + image_type = GST_TAG_IMAGE_TYPE_FRONT_COVER; + } else if (g_str_has_prefix (mimetype, "image/") || + g_str_has_suffix (filename_lc, "png") || + g_str_has_suffix (filename_lc, "jpg") || + g_str_has_suffix (filename_lc, "jpeg") || + g_str_has_suffix (filename_lc, "gif") || + g_str_has_suffix (filename_lc, "bmp")) { + image_type = GST_TAG_IMAGE_TYPE_UNDEFINED; + } + g_free (filename_lc); + + /* First try to create an image tag buffer from this */ + if (image_type != GST_TAG_IMAGE_TYPE_NONE) { + tagbuffer = + gst_tag_image_data_to_image_buffer (data, datalen, image_type); + + if (!tagbuffer) + image_type = GST_TAG_IMAGE_TYPE_NONE; + } + + /* if this failed create an attachment buffer */ + if (!tagbuffer) { + tagbuffer = gst_buffer_new_and_alloc (datalen); + + memcpy (GST_BUFFER_DATA (tagbuffer), data, datalen); + GST_BUFFER_SIZE (tagbuffer) = datalen; + + caps = gst_type_find_helper_for_buffer (NULL, tagbuffer, NULL); + if (caps == NULL) + caps = gst_caps_new_simple (mimetype, NULL); + gst_buffer_set_caps (tagbuffer, caps); + gst_caps_unref (caps); + } + + /* Set filename and description on the caps */ + caps = GST_BUFFER_CAPS (tagbuffer); + gst_caps_set_simple (caps, "filename", G_TYPE_STRING, filename, NULL); + if (description) + gst_caps_set_simple (caps, "description", G_TYPE_STRING, description, + NULL); + + GST_DEBUG_OBJECT (parse, + "Created attachment buffer with caps: %" GST_PTR_FORMAT, caps); + + /* and append to the tag list */ + if (image_type != GST_TAG_IMAGE_TYPE_NONE) + gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND, GST_TAG_IMAGE, tagbuffer, + NULL); + else + gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND, GST_TAG_ATTACHMENT, + tagbuffer, NULL); + } + + g_free (filename); + g_free (mimetype); + g_free (data); + g_free (description); + + return ret; +} + +static GstFlowReturn +gst_matroska_parse_parse_attachments (GstMatroskaParse * parse, + GstEbmlRead * ebml) +{ + guint32 id; + GstFlowReturn ret = GST_FLOW_OK; + GstTagList *taglist; + + DEBUG_ELEMENT_START (parse, ebml, "Attachments"); + + if ((ret = gst_ebml_read_master (ebml, &id)) != GST_FLOW_OK) { + DEBUG_ELEMENT_STOP (parse, ebml, "Attachments", ret); + return ret; + } + + taglist = gst_tag_list_new (); + + while (ret == GST_FLOW_OK && gst_ebml_read_has_remaining (ebml, 1, TRUE)) { + if ((ret = gst_ebml_peek_id (ebml, &id)) != GST_FLOW_OK) + break; + + switch (id) { + case GST_MATROSKA_ID_ATTACHEDFILE: + ret = gst_matroska_parse_parse_attached_file (parse, ebml, taglist); + break; + + default: + ret = gst_matroska_parse_parse_skip (parse, ebml, "Attachments", id); + break; + } + } + DEBUG_ELEMENT_STOP (parse, ebml, "Attachments", ret); + + if (gst_structure_n_fields (GST_STRUCTURE (taglist)) > 0) { + GST_DEBUG_OBJECT (parse, "Storing attachment tags"); + gst_matroska_parse_found_global_tag (parse, taglist); + } else { + GST_DEBUG_OBJECT (parse, "No valid attachments found"); + gst_tag_list_free (taglist); + } + + parse->attachments_parsed = TRUE; + + return ret; +} + +static GstFlowReturn +gst_matroska_parse_parse_chapters (GstMatroskaParse * parse, GstEbmlRead * ebml) +{ + guint32 id; + GstFlowReturn ret = GST_FLOW_OK; + + GST_WARNING_OBJECT (parse, "Parsing of chapters not implemented yet"); + + /* TODO: implement parsing of chapters */ + + DEBUG_ELEMENT_START (parse, ebml, "Chapters"); + + if ((ret = gst_ebml_read_master (ebml, &id)) != GST_FLOW_OK) { + DEBUG_ELEMENT_STOP (parse, ebml, "Chapters", ret); + return ret; + } + + while (ret == GST_FLOW_OK && gst_ebml_read_has_remaining (ebml, 1, TRUE)) { + if ((ret = gst_ebml_peek_id (ebml, &id)) != GST_FLOW_OK) + break; + + switch (id) { + default: + ret = gst_ebml_read_skip (ebml); + break; + } + } + + DEBUG_ELEMENT_STOP (parse, ebml, "Chapters", ret); + return ret; +} + +/* + * Read signed/unsigned "EBML" numbers. + * Return: number of bytes processed. + */ + +static gint +gst_matroska_ebmlnum_uint (guint8 * data, guint size, guint64 * num) +{ + gint len_mask = 0x80, read = 1, n = 1, num_ffs = 0; + guint64 total; + + if (size <= 0) { + return -1; + } + + total = data[0]; + while (read <= 8 && !(total & len_mask)) { + read++; + len_mask >>= 1; + } + if (read > 8) + return -1; + + if ((total &= (len_mask - 1)) == len_mask - 1) + num_ffs++; + if (size < read) + return -1; + while (n < read) { + if (data[n] == 0xff) + num_ffs++; + total = (total << 8) | data[n]; + n++; + } + + if (read == num_ffs && total != 0) + *num = G_MAXUINT64; + else + *num = total; + + return read; +} + +static gint +gst_matroska_ebmlnum_sint (guint8 * data, guint size, gint64 * num) +{ + guint64 unum; + gint res; + + /* read as unsigned number first */ + if ((res = gst_matroska_ebmlnum_uint (data, size, &unum)) < 0) + return -1; + + /* make signed */ + if (unum == G_MAXUINT64) + *num = G_MAXINT64; + else + *num = unum - ((1 << ((7 * res) - 1)) - 1); + + return res; +} + +static GstFlowReturn +gst_matroska_parse_parse_blockgroup_or_simpleblock (GstMatroskaParse * parse, + GstEbmlRead * ebml, guint64 cluster_time, guint64 cluster_offset, + gboolean is_simpleblock) +{ + GstMatroskaTrackContext *stream = NULL; + GstFlowReturn ret = GST_FLOW_OK; + gboolean readblock = FALSE; + guint32 id; + guint64 block_duration = 0; + GstBuffer *buf = NULL; + gint stream_num = -1, n, laces = 0; + guint size = 0; + gint *lace_size = NULL; + gint64 time = 0; + gint flags = 0; + gint64 referenceblock = 0; + gint64 offset; + + offset = gst_ebml_read_get_offset (ebml); + + while (ret == GST_FLOW_OK && gst_ebml_read_has_remaining (ebml, 1, TRUE)) { + if (!is_simpleblock) { + if ((ret = gst_ebml_peek_id (ebml, &id)) != GST_FLOW_OK) { + goto data_error; + } + } else { + id = GST_MATROSKA_ID_SIMPLEBLOCK; + } + + switch (id) { + /* one block inside the group. Note, block parsing is one + * of the harder things, so this code is a bit complicated. + * See http://www.matroska.org/ for documentation. */ + case GST_MATROSKA_ID_SIMPLEBLOCK: + case GST_MATROSKA_ID_BLOCK: + { + guint64 num; + guint8 *data; + + if (buf) { + gst_buffer_unref (buf); + buf = NULL; + } + if ((ret = gst_ebml_read_buffer (ebml, &id, &buf)) != GST_FLOW_OK) + break; + + data = GST_BUFFER_DATA (buf); + size = GST_BUFFER_SIZE (buf); + + /* first byte(s): blocknum */ + if ((n = gst_matroska_ebmlnum_uint (data, size, &num)) < 0) + goto data_error; + data += n; + size -= n; + + /* fetch stream from num */ + stream_num = gst_matroska_parse_stream_from_num (parse, num); + if (G_UNLIKELY (size < 3)) { + GST_WARNING_OBJECT (parse, "Invalid size %u", size); + /* non-fatal, try next block(group) */ + ret = GST_FLOW_OK; + goto done; + } else if (G_UNLIKELY (stream_num < 0 || + stream_num >= parse->num_streams)) { + /* let's not give up on a stray invalid track number */ + GST_WARNING_OBJECT (parse, + "Invalid stream %d for track number %" G_GUINT64_FORMAT + "; ignoring block", stream_num, num); + goto done; + } + + stream = g_ptr_array_index (parse->src, stream_num); + + /* time (relative to cluster time) */ + time = ((gint16) GST_READ_UINT16_BE (data)); + data += 2; + size -= 2; + flags = GST_READ_UINT8 (data); + data += 1; + size -= 1; + + GST_LOG_OBJECT (parse, "time %" G_GUINT64_FORMAT ", flags %d", time, + flags); + + switch ((flags & 0x06) >> 1) { + case 0x0: /* no lacing */ + laces = 1; + lace_size = g_new (gint, 1); + lace_size[0] = size; + break; + + case 0x1: /* xiph lacing */ + case 0x2: /* fixed-size lacing */ + case 0x3: /* EBML lacing */ + if (size == 0) + goto invalid_lacing; + laces = GST_READ_UINT8 (data) + 1; + data += 1; + size -= 1; + lace_size = g_new0 (gint, laces); + + switch ((flags & 0x06) >> 1) { + case 0x1: /* xiph lacing */ { + guint temp, total = 0; + + for (n = 0; ret == GST_FLOW_OK && n < laces - 1; n++) { + while (1) { + if (size == 0) + goto invalid_lacing; + temp = GST_READ_UINT8 (data); + lace_size[n] += temp; + data += 1; + size -= 1; + if (temp != 0xff) + break; + } + total += lace_size[n]; + } + lace_size[n] = size - total; + break; + } + + case 0x2: /* fixed-size lacing */ + for (n = 0; n < laces; n++) + lace_size[n] = size / laces; + break; + + case 0x3: /* EBML lacing */ { + guint total; + + if ((n = gst_matroska_ebmlnum_uint (data, size, &num)) < 0) + goto data_error; + data += n; + size -= n; + total = lace_size[0] = num; + for (n = 1; ret == GST_FLOW_OK && n < laces - 1; n++) { + gint64 snum; + gint r; + + if ((r = gst_matroska_ebmlnum_sint (data, size, &snum)) < 0) + goto data_error; + data += r; + size -= r; + lace_size[n] = lace_size[n - 1] + snum; + total += lace_size[n]; + } + if (n < laces) + lace_size[n] = size - total; + break; + } + } + break; + } + + if (ret != GST_FLOW_OK) + break; + + readblock = TRUE; + break; + } + + case GST_MATROSKA_ID_BLOCKDURATION:{ + ret = gst_ebml_read_uint (ebml, &id, &block_duration); + GST_DEBUG_OBJECT (parse, "BlockDuration: %" G_GUINT64_FORMAT, + block_duration); + break; + } + + case GST_MATROSKA_ID_REFERENCEBLOCK:{ + ret = gst_ebml_read_sint (ebml, &id, &referenceblock); + GST_DEBUG_OBJECT (parse, "ReferenceBlock: %" G_GINT64_FORMAT, + referenceblock); + break; + } + + case GST_MATROSKA_ID_CODECSTATE:{ + guint8 *data; + guint64 data_len = 0; + + if ((ret = + gst_ebml_read_binary (ebml, &id, &data, + &data_len)) != GST_FLOW_OK) + break; + + if (G_UNLIKELY (stream == NULL)) { + GST_WARNING_OBJECT (parse, + "Unexpected CodecState subelement - ignoring"); + break; + } + + g_free (stream->codec_state); + stream->codec_state = data; + stream->codec_state_size = data_len; + + break; + } + + default: + ret = gst_matroska_parse_parse_skip (parse, ebml, "BlockGroup", id); + break; + + case GST_MATROSKA_ID_BLOCKVIRTUAL: + case GST_MATROSKA_ID_BLOCKADDITIONS: + case GST_MATROSKA_ID_REFERENCEPRIORITY: + case GST_MATROSKA_ID_REFERENCEVIRTUAL: + case GST_MATROSKA_ID_SLICES: + GST_DEBUG_OBJECT (parse, + "Skipping BlockGroup subelement 0x%x - ignoring", id); + ret = gst_ebml_read_skip (ebml); + break; + } + + if (is_simpleblock) + break; + } + + /* reading a number or so could have failed */ + if (ret != GST_FLOW_OK) + goto data_error; + + if (ret == GST_FLOW_OK && readblock) { + guint64 duration = 0; + gint64 lace_time = 0; + gboolean delta_unit; + + stream = g_ptr_array_index (parse->src, stream_num); + + if (cluster_time != GST_CLOCK_TIME_NONE) { + /* FIXME: What to do with negative timestamps? Give timestamp 0 or -1? + * Drop unless the lace contains timestamp 0? */ + if (time < 0 && (-time) > cluster_time) { + lace_time = 0; + } else { + if (stream->timecodescale == 1.0) + lace_time = (cluster_time + time) * parse->time_scale; + else + lace_time = + gst_util_guint64_to_gdouble ((cluster_time + time) * + parse->time_scale) * stream->timecodescale; + } + } else { + lace_time = GST_CLOCK_TIME_NONE; + } + + if (lace_time != GST_CLOCK_TIME_NONE) { + parse->last_timestamp = lace_time; + } + /* need to refresh segment info ASAP */ + if (GST_CLOCK_TIME_IS_VALID (lace_time) && parse->need_newsegment) { + GST_DEBUG_OBJECT (parse, + "generating segment starting at %" GST_TIME_FORMAT, + GST_TIME_ARGS (lace_time)); + /* pretend we seeked here */ + gst_segment_set_seek (&parse->segment, parse->segment.rate, + GST_FORMAT_TIME, 0, GST_SEEK_TYPE_SET, lace_time, + GST_SEEK_TYPE_SET, GST_CLOCK_TIME_NONE, NULL); + /* now convey our segment notion downstream */ + gst_matroska_parse_send_event (parse, gst_event_new_new_segment (FALSE, + parse->segment.rate, parse->segment.format, parse->segment.start, + parse->segment.stop, parse->segment.start)); + parse->need_newsegment = FALSE; + } + + if (block_duration) { + if (stream->timecodescale == 1.0) + duration = gst_util_uint64_scale (block_duration, parse->time_scale, 1); + else + duration = + gst_util_gdouble_to_guint64 (gst_util_guint64_to_gdouble + (gst_util_uint64_scale (block_duration, parse->time_scale, + 1)) * stream->timecodescale); + } else if (stream->default_duration) { + duration = stream->default_duration * laces; + } + /* else duration is diff between timecode of this and next block */ + + /* For SimpleBlock, look at the keyframe bit in flags. Otherwise, + a ReferenceBlock implies that this is not a keyframe. In either + case, it only makes sense for video streams. */ + delta_unit = stream->type == GST_MATROSKA_TRACK_TYPE_VIDEO && + ((is_simpleblock && !(flags & 0x80)) || referenceblock); + + if (delta_unit && stream->set_discont) { + /* When doing seeks or such, we need to restart on key frames or + * decoders might choke. */ + GST_DEBUG_OBJECT (parse, "skipping delta unit"); + goto done; + } + + for (n = 0; n < laces; n++) { + if (G_UNLIKELY (lace_size[n] > size)) { + GST_WARNING_OBJECT (parse, "Invalid lace size"); + break; + } + + /* QoS for video track with an index. the assumption is that + index entries point to keyframes, but if that is not true we + will instad skip until the next keyframe. */ + if (GST_CLOCK_TIME_IS_VALID (lace_time) && + stream->type == GST_MATROSKA_TRACK_TYPE_VIDEO && + stream->index_table && parse->segment.rate > 0.0) { + GstMatroskaTrackVideoContext *videocontext = + (GstMatroskaTrackVideoContext *) stream; + GstClockTime earliest_time; + GstClockTime earliest_stream_time; + + GST_OBJECT_LOCK (parse); + earliest_time = videocontext->earliest_time; + GST_OBJECT_UNLOCK (parse); + earliest_stream_time = gst_segment_to_position (&parse->segment, + GST_FORMAT_TIME, earliest_time); + + if (GST_CLOCK_TIME_IS_VALID (lace_time) && + GST_CLOCK_TIME_IS_VALID (earliest_stream_time) && + lace_time <= earliest_stream_time) { + /* find index entry (keyframe) <= earliest_stream_time */ + GstMatroskaIndex *entry = + gst_util_array_binary_search (stream->index_table->data, + stream->index_table->len, sizeof (GstMatroskaIndex), + (GCompareDataFunc) gst_matroska_index_seek_find, + GST_SEARCH_MODE_BEFORE, &earliest_stream_time, NULL); + + /* if that entry (keyframe) is after the current the current + buffer, we can skip pushing (and thus decoding) all + buffers until that keyframe. */ + if (entry && GST_CLOCK_TIME_IS_VALID (entry->time) && + entry->time > lace_time) { + GST_LOG_OBJECT (parse, "Skipping lace before late keyframe"); + stream->set_discont = TRUE; + goto next_lace; + } + } + } +#if 0 + sub = gst_buffer_create_sub (buf, + GST_BUFFER_SIZE (buf) - size, lace_size[n]); + GST_DEBUG_OBJECT (parse, "created subbuffer %p", sub); + + if (delta_unit) + GST_BUFFER_FLAG_SET (sub, GST_BUFFER_FLAG_DELTA_UNIT); + else + GST_BUFFER_FLAG_UNSET (sub, GST_BUFFER_FLAG_DELTA_UNIT); + + if (stream->encodings != NULL && stream->encodings->len > 0) + sub = gst_matroska_decode_buffer (stream, sub); + + if (sub == NULL) { + GST_WARNING_OBJECT (parse, "Decoding buffer failed"); + goto next_lace; + } + + GST_BUFFER_TIMESTAMP (sub) = lace_time; + + if (GST_CLOCK_TIME_IS_VALID (lace_time)) { + GstClockTime last_stop_end; + + /* Check if this stream is after segment stop */ + if (GST_CLOCK_TIME_IS_VALID (parse->segment.stop) && + lace_time >= parse->segment.stop) { + GST_DEBUG_OBJECT (parse, + "Stream %d after segment stop %" GST_TIME_FORMAT, stream->index, + GST_TIME_ARGS (parse->segment.stop)); + gst_buffer_unref (sub); + goto eos; + } + if (offset >= stream->to_offset) { + GST_DEBUG_OBJECT (parse, "Stream %d after playback section", + stream->index); + gst_buffer_unref (sub); + goto eos; + } + + /* handle gaps, e.g. non-zero start-time, or an cue index entry + * that landed us with timestamps not quite intended */ + if (GST_CLOCK_TIME_IS_VALID (parse->segment.last_stop) && + parse->segment.rate > 0.0) { + GstClockTimeDiff diff; + + /* only send newsegments with increasing start times, + * otherwise if these go back and forth downstream (sinks) increase + * accumulated time and running_time */ + diff = GST_CLOCK_DIFF (parse->segment.last_stop, lace_time); + if (diff > 2 * GST_SECOND && lace_time > parse->segment.start && + (!GST_CLOCK_TIME_IS_VALID (parse->segment.stop) || + lace_time < parse->segment.stop)) { + GST_DEBUG_OBJECT (parse, + "Gap of %" G_GINT64_FORMAT " ns detected in" + "stream %d (%" GST_TIME_FORMAT " -> %" GST_TIME_FORMAT "). " + "Sending updated NEWSEGMENT events", diff, + stream->index, GST_TIME_ARGS (stream->pos), + GST_TIME_ARGS (lace_time)); + /* send newsegment events such that the gap is not accounted in + * accum time, hence running_time */ + /* close ahead of gap */ + gst_matroska_parse_send_event (parse, + gst_event_new_new_segment (TRUE, parse->segment.rate, + parse->segment.format, parse->segment.last_stop, + parse->segment.last_stop, parse->segment.last_stop)); + /* skip gap */ + gst_matroska_parse_send_event (parse, + gst_event_new_new_segment (FALSE, parse->segment.rate, + parse->segment.format, lace_time, parse->segment.stop, + lace_time)); + /* align segment view with downstream, + * prevents double-counting accum when closing segment */ + gst_segment_set_newsegment (&parse->segment, FALSE, + parse->segment.rate, parse->segment.format, lace_time, + parse->segment.stop, lace_time); + parse->segment.last_stop = lace_time; + } + } + + if (!GST_CLOCK_TIME_IS_VALID (parse->segment.last_stop) + || parse->segment.last_stop < lace_time) { + parse->segment.last_stop = lace_time; + } + + last_stop_end = lace_time; + if (duration) { + GST_BUFFER_DURATION (sub) = duration / laces; + last_stop_end += GST_BUFFER_DURATION (sub); + } + + if (!GST_CLOCK_TIME_IS_VALID (parse->last_stop_end) || + parse->last_stop_end < last_stop_end) + parse->last_stop_end = last_stop_end; + + if (parse->segment.duration == -1 || + parse->segment.duration < lace_time) { + gst_segment_set_duration (&parse->segment, GST_FORMAT_TIME, + last_stop_end); + gst_element_post_message (GST_ELEMENT_CAST (parse), + gst_message_new_duration (GST_OBJECT_CAST (parse), + GST_FORMAT_TIME, GST_CLOCK_TIME_NONE)); + } + } + + stream->pos = lace_time; + + gst_matroska_parse_sync_streams (parse); + + if (stream->set_discont) { + GST_DEBUG_OBJECT (parse, "marking DISCONT"); + GST_BUFFER_FLAG_SET (sub, GST_BUFFER_FLAG_DISCONT); + stream->set_discont = FALSE; + } + + /* reverse playback book-keeping */ + if (!GST_CLOCK_TIME_IS_VALID (stream->from_time)) + stream->from_time = lace_time; + if (stream->from_offset == -1) + stream->from_offset = offset; + + GST_DEBUG_OBJECT (parse, + "Pushing lace %d, data of size %d for stream %d, time=%" + GST_TIME_FORMAT " and duration=%" GST_TIME_FORMAT, n, + GST_BUFFER_SIZE (sub), stream_num, + GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (sub)), + GST_TIME_ARGS (GST_BUFFER_DURATION (sub))); + + if (parse->element_index) { + if (stream->index_writer_id == -1) + gst_index_get_writer_id (parse->element_index, + GST_OBJECT (stream->pad), &stream->index_writer_id); + + GST_LOG_OBJECT (parse, "adding association %" GST_TIME_FORMAT "-> %" + G_GUINT64_FORMAT " for writer id %d", + GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (sub)), cluster_offset, + stream->index_writer_id); + gst_index_add_association (parse->element_index, + stream->index_writer_id, GST_BUFFER_FLAG_IS_SET (sub, + GST_BUFFER_FLAG_DELTA_UNIT) ? 0 : GST_ASSOCIATION_FLAG_KEY_UNIT, + GST_FORMAT_TIME, GST_BUFFER_TIMESTAMP (sub), GST_FORMAT_BYTES, + cluster_offset, NULL); + } + + gst_buffer_set_caps (sub, GST_PAD_CAPS (parse->srcpad)); + + /* Postprocess the buffers depending on the codec used */ + if (stream->postprocess_frame) { + GST_LOG_OBJECT (parse, "running post process"); + ret = stream->postprocess_frame (GST_ELEMENT (parse), stream, &sub); + } + + ret = gst_pad_push (stream->pad, sub); + if (parse->segment.rate < 0) { + if (lace_time > parse->segment.stop && ret == GST_FLOW_UNEXPECTED) { + /* In reverse playback we can get a GST_FLOW_UNEXPECTED when + * we are at the end of the segment, so we just need to jump + * back to the previous section. */ + GST_DEBUG_OBJECT (parse, "downstream has reached end of segment"); + ret = GST_FLOW_OK; + } + } + /* combine flows */ + ret = gst_matroska_parse_combine_flows (parse, stream, ret); +#endif + + next_lace: + size -= lace_size[n]; + if (lace_time != GST_CLOCK_TIME_NONE && duration) + lace_time += duration / laces; + else + lace_time = GST_CLOCK_TIME_NONE; + } + } + +done: + if (buf) + gst_buffer_unref (buf); + g_free (lace_size); + + return ret; + + /* EXITS */ +invalid_lacing: + { + GST_ELEMENT_WARNING (parse, STREAM, DEMUX, (NULL), ("Invalid lacing size")); + /* non-fatal, try next block(group) */ + ret = GST_FLOW_OK; + goto done; + } +data_error: + { + GST_ELEMENT_WARNING (parse, STREAM, DEMUX, (NULL), ("Data error")); + /* non-fatal, try next block(group) */ + ret = GST_FLOW_OK; + goto done; + } +} + +/* return FALSE if block(group) should be skipped (due to a seek) */ +static inline gboolean +gst_matroska_parse_seek_block (GstMatroskaParse * parse) +{ + if (G_UNLIKELY (parse->seek_block)) { + if (!(--parse->seek_block)) { + return TRUE; + } else { + GST_LOG_OBJECT (parse, "should skip block due to seek"); + return FALSE; + } + } else { + return TRUE; + } +} + +static GstFlowReturn +gst_matroska_parse_parse_contents_seekentry (GstMatroskaParse * parse, + GstEbmlRead * ebml) +{ + GstFlowReturn ret; + guint64 seek_pos = (guint64) - 1; + guint32 seek_id = 0; + guint32 id; + + DEBUG_ELEMENT_START (parse, ebml, "Seek"); + + if ((ret = gst_ebml_read_master (ebml, &id)) != GST_FLOW_OK) { + DEBUG_ELEMENT_STOP (parse, ebml, "Seek", ret); + return ret; + } + + while (ret == GST_FLOW_OK && gst_ebml_read_has_remaining (ebml, 1, TRUE)) { + if ((ret = gst_ebml_peek_id (ebml, &id)) != GST_FLOW_OK) + break; + + switch (id) { + case GST_MATROSKA_ID_SEEKID: + { + guint64 t; + + if ((ret = gst_ebml_read_uint (ebml, &id, &t)) != GST_FLOW_OK) + break; + + GST_DEBUG_OBJECT (parse, "SeekID: %" G_GUINT64_FORMAT, t); + seek_id = t; + break; + } + + case GST_MATROSKA_ID_SEEKPOSITION: + { + guint64 t; + + if ((ret = gst_ebml_read_uint (ebml, &id, &t)) != GST_FLOW_OK) + break; + + if (t > G_MAXINT64) { + GST_WARNING_OBJECT (parse, + "Too large SeekPosition %" G_GUINT64_FORMAT, t); + break; + } + + GST_DEBUG_OBJECT (parse, "SeekPosition: %" G_GUINT64_FORMAT, t); + seek_pos = t; + break; + } + + default: + ret = gst_matroska_parse_parse_skip (parse, ebml, "SeekHead", id); + break; + } + } + + if (ret != GST_FLOW_OK && ret != GST_FLOW_UNEXPECTED) + return ret; + + if (!seek_id || seek_pos == (guint64) - 1) { + GST_WARNING_OBJECT (parse, "Incomplete seekhead entry (0x%x/%" + G_GUINT64_FORMAT ")", seek_id, seek_pos); + return GST_FLOW_OK; + } + + switch (seek_id) { + case GST_MATROSKA_ID_SEEKHEAD: + { + } + case GST_MATROSKA_ID_CUES: + case GST_MATROSKA_ID_TAGS: + case GST_MATROSKA_ID_TRACKS: + case GST_MATROSKA_ID_SEGMENTINFO: + case GST_MATROSKA_ID_ATTACHMENTS: + case GST_MATROSKA_ID_CHAPTERS: + { + guint64 before_pos, length; + + /* remember */ + length = gst_matroska_parse_get_length (parse); + before_pos = parse->offset; + + if (length == (guint64) - 1) { + GST_DEBUG_OBJECT (parse, "no upstream length, skipping SeakHead entry"); + break; + } + + /* check for validity */ + if (seek_pos + parse->ebml_segment_start + 12 >= length) { + GST_WARNING_OBJECT (parse, + "SeekHead reference lies outside file!" " (%" + G_GUINT64_FORMAT "+%" G_GUINT64_FORMAT "+12 >= %" + G_GUINT64_FORMAT ")", seek_pos, parse->ebml_segment_start, length); + break; + } + + /* only pick up index location when streaming */ + if (seek_id == GST_MATROSKA_ID_CUES) { + parse->index_offset = seek_pos + parse->ebml_segment_start; + GST_DEBUG_OBJECT (parse, "Cues located at offset %" G_GUINT64_FORMAT, + parse->index_offset); + } + break; + } + + default: + GST_DEBUG_OBJECT (parse, "Ignoring Seek entry for ID=0x%x", seek_id); + break; + } + DEBUG_ELEMENT_STOP (parse, ebml, "Seek", ret); + + return ret; +} + +static GstFlowReturn +gst_matroska_parse_parse_contents (GstMatroskaParse * parse, GstEbmlRead * ebml) +{ + GstFlowReturn ret = GST_FLOW_OK; + guint32 id; + + DEBUG_ELEMENT_START (parse, ebml, "SeekHead"); + + if ((ret = gst_ebml_read_master (ebml, &id)) != GST_FLOW_OK) { + DEBUG_ELEMENT_STOP (parse, ebml, "SeekHead", ret); + return ret; + } + + while (ret == GST_FLOW_OK && gst_ebml_read_has_remaining (ebml, 1, TRUE)) { + if ((ret = gst_ebml_peek_id (ebml, &id)) != GST_FLOW_OK) + break; + + switch (id) { + case GST_MATROSKA_ID_SEEKENTRY: + { + ret = gst_matroska_parse_parse_contents_seekentry (parse, ebml); + /* Ignore EOS and errors here */ + if (ret != GST_FLOW_OK) { + GST_DEBUG_OBJECT (parse, "Ignoring %s", gst_flow_get_name (ret)); + ret = GST_FLOW_OK; + } + break; + } + + default: + ret = gst_matroska_parse_parse_skip (parse, ebml, "SeekHead", id); + break; + } + } + + DEBUG_ELEMENT_STOP (parse, ebml, "SeekHead", ret); + + return ret; +} + +#define GST_FLOW_OVERFLOW GST_FLOW_CUSTOM_ERROR + +#define MAX_BLOCK_SIZE (15 * 1024 * 1024) + +static inline GstFlowReturn +gst_matroska_parse_check_read_size (GstMatroskaParse * parse, guint64 bytes) +{ + if (G_UNLIKELY (bytes > MAX_BLOCK_SIZE)) { + /* only a few blocks are expected/allowed to be large, + * and will be recursed into, whereas others will be read and must fit */ + /* fatal in streaming case, as we can't step over easily */ + GST_ELEMENT_ERROR (parse, STREAM, DEMUX, (NULL), + ("reading large block of size %" G_GUINT64_FORMAT " not supported; " + "file might be corrupt.", bytes)); + return GST_FLOW_ERROR; + } else { + return GST_FLOW_OK; + } +} + +/* returns TRUE if we truely are in error state, and should give up */ +static inline gboolean +gst_matroska_parse_check_parse_error (GstMatroskaParse * parse) +{ + gint64 pos; + + /* sigh, one last attempt above and beyond call of duty ...; + * search for cluster mark following current pos */ + pos = parse->offset; + GST_WARNING_OBJECT (parse, "parse error, looking for next cluster"); + if (gst_matroska_parse_search_cluster (parse, &pos) != GST_FLOW_OK) { + /* did not work, give up */ + return TRUE; + } else { + GST_DEBUG_OBJECT (parse, "... found at %" G_GUINT64_FORMAT, pos); + /* try that position */ + parse->offset = pos; + return FALSE; + } +} + +/* initializes @ebml with @bytes from input stream at current offset. + * Returns UNEXPECTED if insufficient available, + * ERROR if too much was attempted to read. */ +static inline GstFlowReturn +gst_matroska_parse_take (GstMatroskaParse * parse, guint64 bytes, + GstEbmlRead * ebml) +{ + GstBuffer *buffer = NULL; + GstFlowReturn ret = GST_FLOW_OK; + + GST_LOG_OBJECT (parse, "taking %" G_GUINT64_FORMAT " bytes for parsing", + bytes); + ret = gst_matroska_parse_check_read_size (parse, bytes); + if (G_UNLIKELY (ret != GST_FLOW_OK)) { + /* otherwise fatal */ + ret = GST_FLOW_ERROR; + goto exit; + } + if (gst_adapter_available (parse->adapter) >= bytes) + buffer = gst_adapter_take_buffer (parse->adapter, bytes); + else + ret = GST_FLOW_UNEXPECTED; + if (G_LIKELY (buffer)) { + gst_ebml_read_init (ebml, GST_ELEMENT_CAST (parse), buffer, parse->offset); + parse->offset += bytes; + } +exit: + return ret; +} + +static void +gst_matroska_parse_check_seekability (GstMatroskaParse * parse) +{ + GstQuery *query; + gboolean seekable = FALSE; + gint64 start = -1, stop = -1; + + query = gst_query_new_seeking (GST_FORMAT_BYTES); + if (!gst_pad_peer_query (parse->sinkpad, query)) { + GST_DEBUG_OBJECT (parse, "seeking query failed"); + goto done; + } + + gst_query_parse_seeking (query, NULL, &seekable, &start, &stop); + + /* try harder to query upstream size if we didn't get it the first time */ + if (seekable && stop == -1) { + GstFormat fmt = GST_FORMAT_BYTES; + + GST_DEBUG_OBJECT (parse, "doing duration query to fix up unset stop"); + gst_pad_query_peer_duration (parse->sinkpad, &fmt, &stop); + } + + /* if upstream doesn't know the size, it's likely that it's not seekable in + * practice even if it technically may be seekable */ + if (seekable && (start != 0 || stop <= start)) { + GST_DEBUG_OBJECT (parse, "seekable but unknown start/stop -> disable"); + seekable = FALSE; + } + +done: + GST_INFO_OBJECT (parse, "seekable: %d (%" G_GUINT64_FORMAT " - %" + G_GUINT64_FORMAT ")", seekable, start, stop); + parse->seekable = seekable; + + gst_query_unref (query); +} + +#if 0 +static GstFlowReturn +gst_matroska_parse_find_tracks (GstMatroskaParse * parse) +{ + guint32 id; + guint64 before_pos; + guint64 length; + guint needed; + GstFlowReturn ret = GST_FLOW_OK; + + GST_WARNING_OBJECT (parse, + "Found Cluster element before Tracks, searching Tracks"); + + /* remember */ + before_pos = parse->offset; + + /* Search Tracks element */ + while (TRUE) { + ret = gst_matroska_parse_peek_id_length_pull (parse, &id, &length, &needed); + if (ret != GST_FLOW_OK) + break; + + if (id != GST_MATROSKA_ID_TRACKS) { + /* we may be skipping large cluster here, so forego size check etc */ + /* ... but we can't skip undefined size; force error */ + if (length == G_MAXUINT64) { + ret = gst_matroska_parse_check_read_size (parse, length); + break; + } else { + parse->offset += needed; + parse->offset += length; + } + continue; + } + + /* will lead to track parsing ... */ + ret = gst_matroska_parse_parse_id (parse, id, length, needed); + break; + } + + /* seek back */ + parse->offset = before_pos; + + return ret; +} +#endif + +#define GST_READ_CHECK(stmt) \ +G_STMT_START { \ + if (G_UNLIKELY ((ret = (stmt)) != GST_FLOW_OK)) { \ + if (ret == GST_FLOW_OVERFLOW) { \ + ret = GST_FLOW_OK; \ + } \ + goto read_error; \ + } \ +} G_STMT_END + +static void +gst_matroska_parse_accumulate_streamheader (GstMatroskaParse * parse, + GstBuffer * buffer) +{ + if (parse->streamheader) { + GstBuffer *buf; + + buf = gst_buffer_span (parse->streamheader, 0, buffer, + GST_BUFFER_SIZE (parse->streamheader) + GST_BUFFER_SIZE (buffer)); + gst_buffer_unref (parse->streamheader); + parse->streamheader = buf; + } else { + parse->streamheader = gst_buffer_ref (buffer); + } + + GST_DEBUG ("%d", GST_BUFFER_SIZE (parse->streamheader)); +} + +static GstFlowReturn +gst_matroska_parse_output (GstMatroskaParse * parse, GstBuffer * buffer, + gboolean keyframe) +{ + GstFlowReturn ret = GST_FLOW_OK; + + if (!parse->pushed_headers) { + GstCaps *caps; + GstStructure *s; + GValue streamheader = { 0 }; + GValue bufval = { 0 }; + GstBuffer *buf; + + caps = gst_caps_new_simple ("video/x-matroska", NULL); + s = gst_caps_get_structure (caps, 0); + g_value_init (&streamheader, GST_TYPE_ARRAY); + g_value_init (&bufval, GST_TYPE_BUFFER); + GST_BUFFER_FLAG_SET (parse->streamheader, GST_BUFFER_FLAG_IN_CAPS); + gst_value_set_buffer (&bufval, parse->streamheader); + gst_value_array_append_value (&streamheader, &bufval); + g_value_unset (&bufval); + gst_structure_set_value (s, "streamheader", &streamheader); + g_value_unset (&streamheader); + //gst_caps_replace (parse->caps, caps); + gst_pad_set_caps (parse->srcpad, caps); + + buf = gst_buffer_make_metadata_writable (parse->streamheader); + gst_buffer_set_caps (buf, caps); + gst_caps_unref (caps); + + GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT); + GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS); + GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DELTA_UNIT); + + ret = gst_pad_push (parse->srcpad, buf); + + parse->pushed_headers = TRUE; + } + + if (!keyframe) { + GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DELTA_UNIT); + } else { + GST_BUFFER_FLAG_UNSET (buffer, GST_BUFFER_FLAG_DELTA_UNIT); + } + if (GST_BUFFER_TIMESTAMP (buffer) != GST_CLOCK_TIME_NONE) { + parse->last_timestamp = GST_BUFFER_TIMESTAMP (buffer); + } else { + GST_BUFFER_TIMESTAMP (buffer) = parse->last_timestamp; + } + gst_buffer_set_caps (buffer, GST_PAD_CAPS (parse->srcpad)); + ret = gst_pad_push (parse->srcpad, gst_buffer_ref (buffer)); + + return ret; +} + +static GstFlowReturn +gst_matroska_parse_parse_id (GstMatroskaParse * parse, guint32 id, + guint64 length, guint needed) +{ + GstEbmlRead ebml = { 0, }; + GstFlowReturn ret = GST_FLOW_OK; + guint64 read; + //GstBuffer *buffer; + + GST_DEBUG_OBJECT (parse, "Parsing Element id 0x%x, " + "size %" G_GUINT64_FORMAT ", prefix %d", id, length, needed); + +#if 0 + if (gst_adapter_available (parse->adapter) >= length + needed) { + buffer = gst_adapter_take_buffer (parse->adapter, length + needed); + gst_pad_push (parse->srcpad, buffer); + } else { + ret = GST_FLOW_UNEXPECTED; + } + //GST_READ_CHECK (gst_matroska_parse_take (parse, read, &ebml)); + + return ret; +#endif + + + + /* if we plan to read and parse this element, we need prefix (id + length) + * and the contents */ + /* mind about overflow wrap-around when dealing with undefined size */ + read = length; + if (G_LIKELY (length != G_MAXUINT64)) + read += needed; + + switch (parse->state) { + case GST_MATROSKA_PARSE_STATE_START: + switch (id) { + case GST_EBML_ID_HEADER: + GST_READ_CHECK (gst_matroska_parse_take (parse, read, &ebml)); + ret = gst_matroska_parse_parse_header (parse, &ebml); + if (ret != GST_FLOW_OK) + goto parse_failed; + parse->state = GST_MATROSKA_PARSE_STATE_SEGMENT; + gst_matroska_parse_check_seekability (parse); + gst_matroska_parse_accumulate_streamheader (parse, ebml.buf); + break; + default: + goto invalid_header; + break; + } + break; + case GST_MATROSKA_PARSE_STATE_SEGMENT: + switch (id) { + case GST_MATROSKA_ID_SEGMENT: + /* eat segment prefix */ + GST_READ_CHECK (gst_matroska_parse_take (parse, needed, &ebml)); + GST_DEBUG_OBJECT (parse, + "Found Segment start at offset %" G_GUINT64_FORMAT, + parse->offset); + /* seeks are from the beginning of the segment, + * after the segment ID/length */ + parse->ebml_segment_start = parse->offset; + parse->state = GST_MATROSKA_PARSE_STATE_HEADER; + gst_matroska_parse_accumulate_streamheader (parse, ebml.buf); + break; + default: + GST_WARNING_OBJECT (parse, + "Expected a Segment ID (0x%x), but received 0x%x!", + GST_MATROSKA_ID_SEGMENT, id); + GST_READ_CHECK (gst_matroska_parse_take (parse, needed, &ebml)); + gst_matroska_parse_accumulate_streamheader (parse, ebml.buf); + break; + } + break; + case GST_MATROSKA_PARSE_STATE_SCANNING: + if (id != GST_MATROSKA_ID_CLUSTER && + id != GST_MATROSKA_ID_CLUSTERTIMECODE) + goto skip; + /* fall-through */ + case GST_MATROSKA_PARSE_STATE_HEADER: + case GST_MATROSKA_PARSE_STATE_DATA: + case GST_MATROSKA_PARSE_STATE_SEEK: + switch (id) { + case GST_MATROSKA_ID_SEGMENTINFO: + GST_READ_CHECK (gst_matroska_parse_take (parse, read, &ebml)); + if (!parse->segmentinfo_parsed) { + ret = gst_matroska_parse_parse_info (parse, &ebml); + } + gst_matroska_parse_accumulate_streamheader (parse, ebml.buf); + break; + case GST_MATROSKA_ID_TRACKS: + GST_READ_CHECK (gst_matroska_parse_take (parse, read, &ebml)); + if (!parse->tracks_parsed) { + ret = gst_matroska_parse_parse_tracks (parse, &ebml); + } + gst_matroska_parse_accumulate_streamheader (parse, ebml.buf); + break; + case GST_MATROSKA_ID_CLUSTER: + if (G_UNLIKELY (!parse->tracks_parsed)) { + GST_DEBUG_OBJECT (parse, "Cluster before Track"); + goto not_streamable; + } + if (G_UNLIKELY (parse->state == GST_MATROSKA_PARSE_STATE_HEADER)) { + parse->state = GST_MATROSKA_PARSE_STATE_DATA; + parse->first_cluster_offset = parse->offset; + GST_DEBUG_OBJECT (parse, "signaling no more pads"); + } + parse->cluster_time = GST_CLOCK_TIME_NONE; + parse->cluster_offset = parse->offset; + if (G_UNLIKELY (!parse->seek_first && parse->seek_block)) { + GST_DEBUG_OBJECT (parse, "seek target block %" G_GUINT64_FORMAT + " not found in Cluster, trying next Cluster's first block instead", + parse->seek_block); + parse->seek_block = 0; + } + parse->seek_first = FALSE; + /* record next cluster for recovery */ + if (read != G_MAXUINT64) + parse->next_cluster_offset = parse->cluster_offset + read; + /* eat cluster prefix */ + GST_READ_CHECK (gst_matroska_parse_take (parse, needed, &ebml)); + ret = gst_matroska_parse_output (parse, ebml.buf, TRUE); + //gst_matroska_parse_accumulate_streamheader (parse, ebml.buf); + break; + case GST_MATROSKA_ID_CLUSTERTIMECODE: + { + guint64 num; + + GST_READ_CHECK (gst_matroska_parse_take (parse, read, &ebml)); + if ((ret = gst_ebml_read_uint (&ebml, &id, &num)) != GST_FLOW_OK) + goto parse_failed; + GST_DEBUG_OBJECT (parse, "ClusterTimeCode: %" G_GUINT64_FORMAT, num); + parse->cluster_time = num; + if (parse->element_index) { + if (parse->element_index_writer_id == -1) + gst_index_get_writer_id (parse->element_index, + GST_OBJECT (parse), &parse->element_index_writer_id); + GST_LOG_OBJECT (parse, "adding association %" GST_TIME_FORMAT "-> %" + G_GUINT64_FORMAT " for writer id %d", + GST_TIME_ARGS (parse->cluster_time), parse->cluster_offset, + parse->element_index_writer_id); + gst_index_add_association (parse->element_index, + parse->element_index_writer_id, GST_ASSOCIATION_FLAG_KEY_UNIT, + GST_FORMAT_TIME, parse->cluster_time, + GST_FORMAT_BYTES, parse->cluster_offset, NULL); + } + gst_matroska_parse_output (parse, ebml.buf, FALSE); + break; + } + case GST_MATROSKA_ID_BLOCKGROUP: + if (!gst_matroska_parse_seek_block (parse)) + goto skip; + GST_READ_CHECK (gst_matroska_parse_take (parse, read, &ebml)); + DEBUG_ELEMENT_START (parse, &ebml, "BlockGroup"); + if ((ret = gst_ebml_read_master (&ebml, &id)) == GST_FLOW_OK) { + ret = gst_matroska_parse_parse_blockgroup_or_simpleblock (parse, + &ebml, parse->cluster_time, parse->cluster_offset, FALSE); + } + DEBUG_ELEMENT_STOP (parse, &ebml, "BlockGroup", ret); + gst_matroska_parse_output (parse, ebml.buf, FALSE); + break; + case GST_MATROSKA_ID_SIMPLEBLOCK: + if (!gst_matroska_parse_seek_block (parse)) + goto skip; + GST_READ_CHECK (gst_matroska_parse_take (parse, read, &ebml)); + DEBUG_ELEMENT_START (parse, &ebml, "SimpleBlock"); + ret = gst_matroska_parse_parse_blockgroup_or_simpleblock (parse, + &ebml, parse->cluster_time, parse->cluster_offset, TRUE); + DEBUG_ELEMENT_STOP (parse, &ebml, "SimpleBlock", ret); + gst_matroska_parse_output (parse, ebml.buf, FALSE); + break; + case GST_MATROSKA_ID_ATTACHMENTS: + GST_READ_CHECK (gst_matroska_parse_take (parse, read, &ebml)); + if (!parse->attachments_parsed) { + ret = gst_matroska_parse_parse_attachments (parse, &ebml); + } + gst_matroska_parse_output (parse, ebml.buf, FALSE); + break; + case GST_MATROSKA_ID_TAGS: + GST_READ_CHECK (gst_matroska_parse_take (parse, read, &ebml)); + ret = gst_matroska_parse_parse_metadata (parse, &ebml); + gst_matroska_parse_output (parse, ebml.buf, FALSE); + break; + case GST_MATROSKA_ID_CHAPTERS: + GST_READ_CHECK (gst_matroska_parse_take (parse, read, &ebml)); + ret = gst_matroska_parse_parse_chapters (parse, &ebml); + gst_matroska_parse_output (parse, ebml.buf, FALSE); + break; + case GST_MATROSKA_ID_SEEKHEAD: + GST_READ_CHECK (gst_matroska_parse_take (parse, read, &ebml)); + ret = gst_matroska_parse_parse_contents (parse, &ebml); + gst_matroska_parse_output (parse, ebml.buf, FALSE); + break; + case GST_MATROSKA_ID_CUES: + GST_READ_CHECK (gst_matroska_parse_take (parse, read, &ebml)); + if (!parse->index_parsed) { + ret = gst_matroska_parse_parse_index (parse, &ebml); + /* only push based; delayed index building */ + if (ret == GST_FLOW_OK + && parse->state == GST_MATROSKA_PARSE_STATE_SEEK) { + GstEvent *event; + + GST_OBJECT_LOCK (parse); + event = parse->seek_event; + parse->seek_event = NULL; + GST_OBJECT_UNLOCK (parse); + + g_assert (event); + /* unlikely to fail, since we managed to seek to this point */ + if (!gst_matroska_parse_handle_seek_event (parse, NULL, event)) + goto seek_failed; + /* resume data handling, main thread clear to seek again */ + GST_OBJECT_LOCK (parse); + parse->state = GST_MATROSKA_PARSE_STATE_DATA; + GST_OBJECT_UNLOCK (parse); + } + } + gst_matroska_parse_output (parse, ebml.buf, FALSE); + break; + case GST_MATROSKA_ID_POSITION: + case GST_MATROSKA_ID_PREVSIZE: + case GST_MATROSKA_ID_ENCRYPTEDBLOCK: + case GST_MATROSKA_ID_SILENTTRACKS: + GST_DEBUG_OBJECT (parse, + "Skipping Cluster subelement 0x%x - ignoring", id); + /* fall-through */ + default: + skip: + GST_DEBUG_OBJECT (parse, "skipping Element 0x%x", id); + GST_READ_CHECK (gst_matroska_parse_take (parse, read, &ebml)); + gst_matroska_parse_output (parse, ebml.buf, FALSE); + break; + } + break; + } + + if (ret == GST_FLOW_PARSE) + goto parse_failed; + +exit: + gst_ebml_read_clear (&ebml); + return ret; + + /* ERRORS */ +read_error: + { + /* simply exit, maybe not enough data yet */ + /* no ebml to clear if read error */ + return ret; + } +parse_failed: + { + GST_ELEMENT_ERROR (parse, STREAM, DEMUX, (NULL), + ("Failed to parse Element 0x%x", id)); + ret = GST_FLOW_ERROR; + goto exit; + } +not_streamable: + { + GST_ELEMENT_ERROR (parse, STREAM, DEMUX, (NULL), + ("File layout does not permit streaming")); + ret = GST_FLOW_ERROR; + goto exit; + } +#if 0 +no_tracks: + { + GST_ELEMENT_ERROR (parse, STREAM, DEMUX, (NULL), + ("No Tracks element found")); + ret = GST_FLOW_ERROR; + goto exit; + } +#endif +invalid_header: + { + GST_ELEMENT_ERROR (parse, STREAM, DEMUX, (NULL), ("Invalid header")); + ret = GST_FLOW_ERROR; + goto exit; + } +seek_failed: + { + GST_ELEMENT_ERROR (parse, STREAM, DEMUX, (NULL), ("Failed to seek")); + ret = GST_FLOW_ERROR; + goto exit; + } +} + +#if 0 +static void +gst_matroska_parse_loop (GstPad * pad) +{ + GstMatroskaParse *parse = GST_MATROSKA_PARSE (GST_PAD_PARENT (pad)); + GstFlowReturn ret; + guint32 id; + guint64 length; + guint needed; + + /* If we have to close a segment, send a new segment to do this now */ + if (G_LIKELY (parse->state == GST_MATROSKA_PARSE_STATE_DATA)) { + if (G_UNLIKELY (parse->close_segment)) { + gst_matroska_parse_send_event (parse, parse->close_segment); + parse->close_segment = NULL; + } + if (G_UNLIKELY (parse->new_segment)) { + gst_matroska_parse_send_event (parse, parse->new_segment); + parse->new_segment = NULL; + } + } + + ret = gst_matroska_parse_peek_id_length_pull (parse, &id, &length, &needed); + if (ret == GST_FLOW_UNEXPECTED) + goto eos; + if (ret != GST_FLOW_OK) { + if (gst_matroska_parse_check_parse_error (parse)) + goto pause; + else + return; + } + + GST_LOG_OBJECT (parse, "Offset %" G_GUINT64_FORMAT ", Element id 0x%x, " + "size %" G_GUINT64_FORMAT ", needed %d", parse->offset, id, + length, needed); + + ret = gst_matroska_parse_parse_id (parse, id, length, needed); + if (ret == GST_FLOW_UNEXPECTED) + goto eos; + if (ret != GST_FLOW_OK) + goto pause; + + /* check if we're at the end of a configured segment */ + if (G_LIKELY (parse->src->len)) { + guint i; + + g_assert (parse->num_streams == parse->src->len); + for (i = 0; i < parse->src->len; i++) { + GstMatroskaTrackContext *context = g_ptr_array_index (parse->src, i); + GST_DEBUG_OBJECT (context->pad, "pos %" GST_TIME_FORMAT, + GST_TIME_ARGS (context->pos)); + if (context->eos == FALSE) + goto next; + } + + GST_INFO_OBJECT (parse, "All streams are EOS"); + ret = GST_FLOW_UNEXPECTED; + goto eos; + } + +next: + if (G_UNLIKELY (parse->offset == gst_matroska_parse_get_length (parse))) { + GST_LOG_OBJECT (parse, "Reached end of stream"); + ret = GST_FLOW_UNEXPECTED; + goto eos; + } + + return; + + /* ERRORS */ +eos: + { + if (parse->segment.rate < 0.0) { + ret = gst_matroska_parse_seek_to_previous_keyframe (parse); + if (ret == GST_FLOW_OK) + return; + } + /* fall-through */ + } +pause: + { + const gchar *reason = gst_flow_get_name (ret); + gboolean push_eos = FALSE; + + GST_LOG_OBJECT (parse, "pausing task, reason %s", reason); + parse->segment_running = FALSE; + gst_pad_pause_task (parse->sinkpad); + + if (ret == GST_FLOW_UNEXPECTED) { + /* perform EOS logic */ + + /* Close the segment, i.e. update segment stop with the duration + * if no stop was set */ + if (GST_CLOCK_TIME_IS_VALID (parse->last_stop_end) && + !GST_CLOCK_TIME_IS_VALID (parse->segment.stop)) { + GstEvent *event = + gst_event_new_new_segment_full (TRUE, parse->segment.rate, + parse->segment.applied_rate, parse->segment.format, + parse->segment.start, + MAX (parse->last_stop_end, parse->segment.start), + parse->segment.time); + gst_matroska_parse_send_event (parse, event); + } + + if (parse->segment.flags & GST_SEEK_FLAG_SEGMENT) { + gint64 stop; + + /* for segment playback we need to post when (in stream time) + * we stopped, this is either stop (when set) or the duration. */ + if ((stop = parse->segment.stop) == -1) + stop = parse->last_stop_end; + + GST_LOG_OBJECT (parse, "Sending segment done, at end of segment"); + gst_element_post_message (GST_ELEMENT (parse), + gst_message_new_segment_done (GST_OBJECT (parse), GST_FORMAT_TIME, + stop)); + } else { + push_eos = TRUE; + } + } else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_UNEXPECTED) { + /* for fatal errors we post an error message */ + GST_ELEMENT_ERROR (parse, STREAM, FAILED, (NULL), + ("stream stopped, reason %s", reason)); + push_eos = TRUE; + } + if (push_eos) { + /* send EOS, and prevent hanging if no streams yet */ + GST_LOG_OBJECT (parse, "Sending EOS, at end of stream"); + if (!gst_matroska_parse_send_event (parse, gst_event_new_eos ()) && + (ret == GST_FLOW_UNEXPECTED)) { + GST_ELEMENT_ERROR (parse, STREAM, DEMUX, + (NULL), ("got eos but no streams (yet)")); + } + } + return; + } +} +#endif + +/* + * Create and push a flushing seek event upstream + */ +static gboolean +perform_seek_to_offset (GstMatroskaParse * parse, guint64 offset) +{ + GstEvent *event; + gboolean res = 0; + + GST_DEBUG_OBJECT (parse, "Seeking to %" G_GUINT64_FORMAT, offset); + + event = + gst_event_new_seek (1.0, GST_FORMAT_BYTES, + GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE, GST_SEEK_TYPE_SET, offset, + GST_SEEK_TYPE_NONE, -1); + + res = gst_pad_push_event (parse->sinkpad, event); + + /* newsegment event will update offset */ + return res; +} + +static const guint8 * +gst_matroska_parse_peek_adapter (GstMatroskaParse * parse, guint peek) +{ + return gst_adapter_peek (parse->adapter, peek); +} + +static GstFlowReturn +gst_matroska_parse_peek_id_length_push (GstMatroskaParse * parse, guint32 * _id, + guint64 * _length, guint * _needed) +{ + return gst_ebml_peek_id_length (_id, _length, _needed, + (GstPeekData) gst_matroska_parse_peek_adapter, (gpointer) parse, + GST_ELEMENT_CAST (parse), parse->offset); +} + +static GstFlowReturn +gst_matroska_parse_chain (GstPad * pad, GstBuffer * buffer) +{ + GstMatroskaParse *parse = GST_MATROSKA_PARSE (GST_PAD_PARENT (pad)); + guint available; + GstFlowReturn ret = GST_FLOW_OK; + guint needed = 0; + guint32 id; + guint64 length; + + if (G_UNLIKELY (GST_BUFFER_IS_DISCONT (buffer))) { + GST_DEBUG_OBJECT (parse, "got DISCONT"); + gst_adapter_clear (parse->adapter); + GST_OBJECT_LOCK (parse); + gst_matroska_parse_reset_streams (parse, GST_CLOCK_TIME_NONE, FALSE); + GST_OBJECT_UNLOCK (parse); + } + + gst_adapter_push (parse->adapter, buffer); + buffer = NULL; + +next: + available = gst_adapter_available (parse->adapter); + + ret = gst_matroska_parse_peek_id_length_push (parse, &id, &length, &needed); + if (G_UNLIKELY (ret != GST_FLOW_OK && ret != GST_FLOW_UNEXPECTED)) + return ret; + + GST_LOG_OBJECT (parse, "Offset %" G_GUINT64_FORMAT ", Element id 0x%x, " + "size %" G_GUINT64_FORMAT ", needed %d, available %d", parse->offset, id, + length, needed, available); + + if (needed > available) + return GST_FLOW_OK; + + ret = gst_matroska_parse_parse_id (parse, id, length, needed); + if (ret == GST_FLOW_UNEXPECTED) { + /* need more data */ + return GST_FLOW_OK; + } else if (ret != GST_FLOW_OK) { + return ret; + } else + goto next; +} + +static gboolean +gst_matroska_parse_handle_sink_event (GstPad * pad, GstEvent * event) +{ + gboolean res = TRUE; + GstMatroskaParse *parse = GST_MATROSKA_PARSE (GST_PAD_PARENT (pad)); + + GST_DEBUG_OBJECT (parse, + "have event type %s: %p on sink pad", GST_EVENT_TYPE_NAME (event), event); + + switch (GST_EVENT_TYPE (event)) { + case GST_EVENT_NEWSEGMENT: + { + GstFormat format; + gdouble rate, arate; + gint64 start, stop, time = 0; + gboolean update; + GstSegment segment; + + /* some debug output */ + gst_segment_init (&segment, GST_FORMAT_UNDEFINED); + gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format, + &start, &stop, &time); + gst_segment_set_newsegment_full (&segment, update, rate, arate, format, + start, stop, time); + GST_DEBUG_OBJECT (parse, + "received format %d newsegment %" GST_SEGMENT_FORMAT, format, + &segment); + + if (parse->state < GST_MATROSKA_PARSE_STATE_DATA) { + GST_DEBUG_OBJECT (parse, "still starting"); + goto exit; + } + + /* we only expect a BYTE segment, e.g. following a seek */ + if (format != GST_FORMAT_BYTES) { + GST_DEBUG_OBJECT (parse, "unsupported segment format, ignoring"); + goto exit; + } + + GST_DEBUG_OBJECT (parse, "clearing segment state"); + /* clear current segment leftover */ + gst_adapter_clear (parse->adapter); + /* and some streaming setup */ + parse->offset = start; + /* do not know where we are; + * need to come across a cluster and generate newsegment */ + parse->segment.last_stop = GST_CLOCK_TIME_NONE; + parse->cluster_time = GST_CLOCK_TIME_NONE; + parse->cluster_offset = 0; + parse->need_newsegment = TRUE; + /* but keep some of the upstream segment */ + parse->segment.rate = rate; + exit: + /* chain will send initial newsegment after pads have been added, + * or otherwise come up with one */ + GST_DEBUG_OBJECT (parse, "eating event"); + gst_event_unref (event); + res = TRUE; + break; + } + case GST_EVENT_EOS: + { + if (parse->state != GST_MATROSKA_PARSE_STATE_DATA) { + gst_event_unref (event); + GST_ELEMENT_ERROR (parse, STREAM, DEMUX, + (NULL), ("got eos and didn't receive a complete header object")); + } else if (parse->num_streams == 0) { + GST_ELEMENT_ERROR (parse, STREAM, DEMUX, + (NULL), ("got eos but no streams (yet)")); + } else { + gst_matroska_parse_send_event (parse, event); + } + break; + } + case GST_EVENT_FLUSH_STOP: + { + gst_adapter_clear (parse->adapter); + GST_OBJECT_LOCK (parse); + gst_matroska_parse_reset_streams (parse, GST_CLOCK_TIME_NONE, TRUE); + GST_OBJECT_UNLOCK (parse); + parse->segment.last_stop = GST_CLOCK_TIME_NONE; + parse->cluster_time = GST_CLOCK_TIME_NONE; + parse->cluster_offset = 0; + /* fall-through */ + } + default: + res = gst_pad_event_default (pad, event); + break; + } + + return res; +} + +static void +gst_matroska_parse_set_index (GstElement * element, GstIndex * index) +{ + GstMatroskaParse *parse = GST_MATROSKA_PARSE (element); + + GST_OBJECT_LOCK (parse); + if (parse->element_index) + gst_object_unref (parse->element_index); + parse->element_index = index ? gst_object_ref (index) : NULL; + GST_OBJECT_UNLOCK (parse); + GST_DEBUG_OBJECT (parse, "Set index %" GST_PTR_FORMAT, parse->element_index); +} + +static GstIndex * +gst_matroska_parse_get_index (GstElement * element) +{ + GstIndex *result = NULL; + GstMatroskaParse *parse = GST_MATROSKA_PARSE (element); + + GST_OBJECT_LOCK (parse); + if (parse->element_index) + result = gst_object_ref (parse->element_index); + GST_OBJECT_UNLOCK (parse); + + GST_DEBUG_OBJECT (parse, "Returning index %" GST_PTR_FORMAT, result); + + return result; +} + +static GstStateChangeReturn +gst_matroska_parse_change_state (GstElement * element, + GstStateChange transition) +{ + GstMatroskaParse *parse = GST_MATROSKA_PARSE (element); + GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS; + + /* handle upwards state changes here */ + switch (transition) { + default: + break; + } + + ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); + + /* handle downwards state changes */ + switch (transition) { + case GST_STATE_CHANGE_PAUSED_TO_READY: + gst_matroska_parse_reset (GST_ELEMENT (parse)); + break; + default: + break; + } + + return ret; +} + +gboolean +gst_matroska_parse_plugin_init (GstPlugin * plugin) +{ + gst_riff_init (); + + /* create an elementfactory for the matroska_parse element */ + if (!gst_element_register (plugin, "matroskaparse", + GST_RANK_NONE, GST_TYPE_MATROSKA_PARSE)) + return FALSE; + + return TRUE; +} diff --git a/gst/matroska/matroska-parse.h b/gst/matroska/matroska-parse.h new file mode 100644 index 0000000000..595ead2e3c --- /dev/null +++ b/gst/matroska/matroska-parse.h @@ -0,0 +1,146 @@ +/* GStreamer Matroska muxer/demuxer + * (c) 2003 Ronald Bultje + * + * matroska-parse.h: matroska file/stream parseer definition + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifndef __GST_MATROSKA_PARSE_H__ +#define __GST_MATROSKA_PARSE_H__ + +#include +#include + +#include "ebml-read.h" +#include "matroska-ids.h" + +G_BEGIN_DECLS + +#define GST_TYPE_MATROSKA_PARSE \ + (gst_matroska_parse_get_type ()) +#define GST_MATROSKA_PARSE(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_MATROSKA_PARSE, GstMatroskaParse)) +#define GST_MATROSKA_PARSE_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_MATROSKA_PARSE, GstMatroskaParseClass)) +#define GST_IS_MATROSKA_PARSE(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_MATROSKA_PARSE)) +#define GST_IS_MATROSKA_PARSE_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_MATROSKA_PARSE)) + +typedef enum { + GST_MATROSKA_PARSE_STATE_START, + GST_MATROSKA_PARSE_STATE_SEGMENT, + GST_MATROSKA_PARSE_STATE_HEADER, + GST_MATROSKA_PARSE_STATE_DATA, + GST_MATROSKA_PARSE_STATE_SEEK, + GST_MATROSKA_PARSE_STATE_SCANNING +} GstMatroskaParseState; + +typedef struct _GstMatroskaParse { + GstElement parent; + + /* < private > */ + + GstIndex *element_index; + gint element_index_writer_id; + + /* pads */ + GstPad *sinkpad; + GstPad *srcpad; + GPtrArray *src; + GstClock *clock; + guint num_streams; + guint num_v_streams; + guint num_a_streams; + guint num_t_streams; + + GstBuffer *streamheader; + gboolean pushed_headers; + GstClockTime last_timestamp; + + /* metadata */ + gchar *muxing_app; + gchar *writing_app; + gint64 created; + + /* state */ + //gboolean streaming; + GstMatroskaParseState state; + guint level_up; + guint64 seek_block; + gboolean seek_first; + + /* did we parse cues/tracks/segmentinfo already? */ + gboolean index_parsed; + gboolean tracks_parsed; + gboolean segmentinfo_parsed; + gboolean attachments_parsed; + GList *tags_parsed; + GList *seek_parsed; + + /* start-of-segment */ + guint64 ebml_segment_start; + + /* a cue (index) table */ + GArray *index; + + /* timescale in the file */ + guint64 time_scale; + + /* keeping track of playback position */ + GstSegment segment; + gboolean segment_running; + GstClockTime last_stop_end; + + GstEvent *close_segment; + GstEvent *new_segment; + GstTagList *global_tags; + + /* pull mode caching */ + GstBuffer *cached_buffer; + + /* push and pull mode */ + guint64 offset; + /* some state saving */ + GstClockTime cluster_time; + guint64 cluster_offset; + guint64 first_cluster_offset; + guint64 next_cluster_offset; + + /* push based mode usual suspects */ + GstAdapter *adapter; + /* index stuff */ + gboolean seekable; + gboolean building_index; + guint64 index_offset; + GstEvent *seek_event; + gboolean need_newsegment; + + /* reverse playback */ + GArray *seek_index; + gint seek_entry; +} GstMatroskaParse; + +typedef struct _GstMatroskaParseClass { + GstElementClass parent; +} GstMatroskaParseClass; + +gboolean gst_matroska_parse_plugin_init (GstPlugin *plugin); + +G_END_DECLS + +#endif /* __GST_MATROSKA_PARSE_H__ */ diff --git a/gst/matroska/matroska.c b/gst/matroska/matroska.c index fd6c873b8b..57db7a31e0 100644 --- a/gst/matroska/matroska.c +++ b/gst/matroska/matroska.c @@ -24,6 +24,7 @@ #endif #include "matroska-demux.h" +#include "matroska-parse.h" #include "matroska-mux.h" #include "matroska-ids.h" #include "webm-mux.h" @@ -40,6 +41,7 @@ plugin_init (GstPlugin * plugin) gst_matroska_register_tags (); ret = gst_matroska_demux_plugin_init (plugin); + ret &= gst_matroska_parse_plugin_init (plugin); ret &= gst_element_register (plugin, "matroskamux", GST_RANK_PRIMARY, GST_TYPE_MATROSKA_MUX); ret &= gst_element_register (plugin, "webmmux", GST_RANK_PRIMARY, From 053f03c032d707e978b4d1232feef59e2da0b6d2 Mon Sep 17 00:00:00 2001 From: David Schleef Date: Tue, 15 Feb 2011 19:33:45 -0800 Subject: [PATCH 328/351] udpsink: warn when packet is too large --- gst/udp/gstmultiudpsink.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/gst/udp/gstmultiudpsink.c b/gst/udp/gstmultiudpsink.c index dd93e3f9d0..bcdb6cd0ba 100644 --- a/gst/udp/gstmultiudpsink.c +++ b/gst/udp/gstmultiudpsink.c @@ -44,6 +44,8 @@ GST_DEBUG_CATEGORY_STATIC (multiudpsink_debug); #define GST_CAT_DEFAULT (multiudpsink_debug) +#define UDP_MAX_SIZE 65507 + static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, @@ -496,6 +498,11 @@ gst_multiudpsink_render (GstBaseSink * bsink, GstBuffer * buffer) size = GST_BUFFER_SIZE (buffer); data = GST_BUFFER_DATA (buffer); + if (size > UDP_MAX_SIZE) { + GST_WARNING ("Attempting to send a UDP packet larger than maximum " + "size (%d > %d)", size, UDP_MAX_SIZE); + } + sink->bytes_to_serve += size; /* grab lock while iterating and sending to clients, this should be @@ -586,6 +593,11 @@ gst_multiudpsink_render_list (GstBaseSink * bsink, GstBufferList * list) msg.msg_iov = iov; while ((buf = gst_buffer_list_iterator_next (it))) { + if (GST_BUFFER_SIZE (buf) > UDP_MAX_SIZE) { + GST_WARNING ("Attempting to send a UDP packet larger than maximum " + "size (%d > %d)", GST_BUFFER_SIZE (buf), UDP_MAX_SIZE); + } + msg.msg_iov[msg.msg_iovlen].iov_len = GST_BUFFER_SIZE (buf); msg.msg_iov[msg.msg_iovlen].iov_base = GST_BUFFER_DATA (buf); msg.msg_iovlen++; From 367adf2b69541cc45077f4d54689495a5a84eff8 Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Fri, 18 Feb 2011 10:12:47 +0200 Subject: [PATCH 329/351] avidemux: stream->current_total is accumulated byte size and not time Use timestamp for the stream index as well. --- gst/avi/gstavidemux.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/gst/avi/gstavidemux.c b/gst/avi/gstavidemux.c index 321d216a8a..95131d3a41 100644 --- a/gst/avi/gstavidemux.c +++ b/gst/avi/gstavidemux.c @@ -4489,10 +4489,10 @@ gst_avi_demux_add_assoc (GstAviDemux * avi, GstAviStream * stream, keyframe ? GST_ASSOCIATION_FLAG_KEY_UNIT : GST_ASSOCIATION_FLAG_DELTA_UNIT, GST_FORMAT_TIME, timestamp, GST_FORMAT_BYTES, offset, NULL); - /* well, current_total determines TIME and entry DEFAULT (frame #) ... */ + /* current_entry is DEFAULT (frame #) */ gst_index_add_association (avi->element_index, stream->index_id, keyframe ? GST_ASSOCIATION_FLAG_KEY_UNIT : - GST_ASSOCIATION_FLAG_DELTA_UNIT, GST_FORMAT_TIME, stream->current_total, + GST_ASSOCIATION_FLAG_DELTA_UNIT, GST_FORMAT_TIME, timestamp, GST_FORMAT_BYTES, offset, GST_FORMAT_DEFAULT, stream->current_entry, NULL); } From 13d441315f6413fab690724cb361b0d7f8dc1ece Mon Sep 17 00:00:00 2001 From: David Schleef Date: Sat, 19 Feb 2011 15:48:22 -0800 Subject: [PATCH 330/351] pngdec: Handle 16-bit-per-channel images --- ext/libpng/gstpngdec.c | 25 +++++++++++++++---------- 1 file changed, 15 insertions(+), 10 deletions(-) diff --git a/ext/libpng/gstpngdec.c b/ext/libpng/gstpngdec.c index 47090e5efb..4e43dec393 100644 --- a/ext/libpng/gstpngdec.c +++ b/ext/libpng/gstpngdec.c @@ -82,11 +82,18 @@ gst_pngdec_get_type (void) return pngdec_type; } +/* FIXME remove this after -good depends on -base-0.10.33 */ +#ifdef GST_VIDEO_CAPS_ARGB_64 +#define CAPS GST_VIDEO_CAPS_RGBA ";" GST_VIDEO_CAPS_RGB ";" GST_VIDEO_CAPS_ARGB_64 +#else +#define CAPS GST_VIDEO_CAPS_RGBA ";" GST_VIDEO_CAPS_RGB +#endif + static GstStaticPadTemplate gst_pngdec_src_pad_template = - GST_STATIC_PAD_TEMPLATE ("src", +GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, - GST_STATIC_CAPS (GST_VIDEO_CAPS_RGBA ";" GST_VIDEO_CAPS_RGB) + GST_STATIC_CAPS (CAPS) ); static GstStaticPadTemplate gst_pngdec_sink_pad_template = @@ -370,12 +377,10 @@ gst_pngdec_caps_create_and_set (GstPngDec * pngdec) /* Get bits per channel */ bpc = png_get_bit_depth (pngdec->png, pngdec->info); - - /* We don't handle 16 bits per color, strip down to 8 */ - if (bpc == 16) { - GST_LOG_OBJECT (pngdec, - "this is a 16 bits per channel PNG image, strip down to 8 bits"); - png_set_strip_16 (pngdec->png); + if (bpc > 8) { + /* Add alpha channel if 16-bit depth */ + png_set_add_alpha (pngdec->png, 0xffff, PNG_FILLER_BEFORE); + png_set_swap (pngdec->png); } /* Get Color type */ @@ -428,11 +433,11 @@ gst_pngdec_caps_create_and_set (GstPngDec * pngdec) switch (pngdec->color_type) { case PNG_COLOR_TYPE_RGB: GST_LOG_OBJECT (pngdec, "we have no alpha channel, depth is 24 bits"); - pngdec->bpp = 24; + pngdec->bpp = 3 * bpc; break; case PNG_COLOR_TYPE_RGB_ALPHA: GST_LOG_OBJECT (pngdec, "we have an alpha channel, depth is 32 bits"); - pngdec->bpp = 32; + pngdec->bpp = 4 * bpc; break; default: GST_ELEMENT_ERROR (pngdec, STREAM, NOT_IMPLEMENTED, (NULL), From 8e6b876e76c94410db160afe5eb30f21452e419f Mon Sep 17 00:00:00 2001 From: David Schleef Date: Sun, 20 Feb 2011 23:39:41 -0800 Subject: [PATCH 331/351] Check that collectpads exists before removing pad The core now calls release pad from finalize, at which point the collectpads might have already been freed. --- ext/cairo/gsttextoverlay.c | 5 ++++- gst/avi/gstavimux.c | 5 ++++- gst/flv/gstflvmux.c | 5 ++++- gst/interleave/interleave.c | 4 +++- gst/matroska/matroska-mux.c | 5 ++++- gst/videomixer/videomixer.c | 5 ++++- 6 files changed, 23 insertions(+), 6 deletions(-) diff --git a/ext/cairo/gsttextoverlay.c b/ext/cairo/gsttextoverlay.c index 364b8bae94..90e81a5bd9 100644 --- a/ext/cairo/gsttextoverlay.c +++ b/ext/cairo/gsttextoverlay.c @@ -208,6 +208,7 @@ gst_text_overlay_finalize (GObject * object) gst_collect_pads_stop (overlay->collect); gst_object_unref (overlay->collect); + overlay->collect = NULL; g_free (overlay->text_fill_image); g_free (overlay->text_outline_image); @@ -631,7 +632,9 @@ gst_text_overlay_text_pad_unlinked (GstPad * pad) GST_DEBUG_OBJECT (overlay, "Text pad unlinked"); if (overlay->text_collect_data) { - gst_collect_pads_remove_pad (overlay->collect, overlay->text_sinkpad); + if (overlay->collect) { + gst_collect_pads_remove_pad (overlay->collect, overlay->text_sinkpad); + } overlay->text_collect_data = NULL; } diff --git a/gst/avi/gstavimux.c b/gst/avi/gstavimux.c index d0d2f0ecc6..1a8e5763ba 100644 --- a/gst/avi/gstavimux.c +++ b/gst/avi/gstavimux.c @@ -285,6 +285,7 @@ gst_avi_mux_finalize (GObject * object) mux->idx = NULL; gst_object_unref (mux->collect); + mux->collect = NULL; G_OBJECT_CLASS (parent_class)->finalize (object); } @@ -1089,7 +1090,9 @@ gst_avi_mux_release_pad (GstElement * element, GstPad * pad) * as it also represent number of streams present */ avipad->collect = NULL; GST_DEBUG_OBJECT (avimux, "removed pad '%s'", GST_PAD_NAME (pad)); - gst_collect_pads_remove_pad (avimux->collect, pad); + if (avimux->collect) { + gst_collect_pads_remove_pad (avimux->collect, pad); + } gst_element_remove_pad (element, pad); /* if not started yet, we can remove any sign this pad ever existed */ /* in this case _start will take care of the real pad count */ diff --git a/gst/flv/gstflvmux.c b/gst/flv/gstflvmux.c index 68a0df4fc7..9d69ace8a2 100644 --- a/gst/flv/gstflvmux.c +++ b/gst/flv/gstflvmux.c @@ -207,6 +207,7 @@ gst_flv_mux_finalize (GObject * object) GstFlvMux *mux = GST_FLV_MUX (object); gst_object_unref (mux->collect); + mux->collect = NULL; G_OBJECT_CLASS (parent_class)->finalize (object); } @@ -559,7 +560,9 @@ gst_flv_mux_release_pad (GstElement * element, GstPad * pad) if (cpad && cpad->video_codec_data) gst_buffer_unref (cpad->video_codec_data); - gst_collect_pads_remove_pad (mux->collect, pad); + if (mux->collect) { + gst_collect_pads_remove_pad (mux->collect, pad); + } gst_element_remove_pad (element, pad); } diff --git a/gst/interleave/interleave.c b/gst/interleave/interleave.c index 96a43c8d61..f6ff41af67 100644 --- a/gst/interleave/interleave.c +++ b/gst/interleave/interleave.c @@ -598,7 +598,9 @@ gst_interleave_release_pad (GstElement * element, GstPad * pad) GST_OBJECT_UNLOCK (self->collect); - gst_collect_pads_remove_pad (self->collect, pad); + if (self->collect) { + gst_collect_pads_remove_pad (self->collect, pad); + } gst_element_remove_pad (element, pad); } diff --git a/gst/matroska/matroska-mux.c b/gst/matroska/matroska-mux.c index 31fe40bfc3..6d2cc5dec4 100644 --- a/gst/matroska/matroska-mux.c +++ b/gst/matroska/matroska-mux.c @@ -385,6 +385,7 @@ gst_matroska_mux_finalize (GObject * object) GstMatroskaMux *mux = GST_MATROSKA_MUX (object); gst_object_unref (mux->collect); + mux->collect = NULL; gst_object_unref (mux->ebml_write); if (mux->writing_app) g_free (mux->writing_app); @@ -1929,7 +1930,9 @@ gst_matroska_mux_release_pad (GstElement * element, GstPad * pad) } } - gst_collect_pads_remove_pad (mux->collect, pad); + if (mux->collect) { + gst_collect_pads_remove_pad (mux->collect, pad); + } if (gst_element_remove_pad (element, pad)) mux->num_streams--; } diff --git a/gst/videomixer/videomixer.c b/gst/videomixer/videomixer.c index 1271d63216..925b06c6ac 100644 --- a/gst/videomixer/videomixer.c +++ b/gst/videomixer/videomixer.c @@ -777,6 +777,7 @@ gst_videomixer_finalize (GObject * object) GstVideoMixer *mix = GST_VIDEO_MIXER (object); gst_object_unref (mix->collect); + mix->collect = NULL; g_mutex_free (mix->state_lock); G_OBJECT_CLASS (parent_class)->finalize (object); @@ -1251,7 +1252,9 @@ gst_videomixer_release_pad (GstElement * element, GstPad * pad) mix->sinkpads = g_slist_remove (mix->sinkpads, pad); gst_videomixer_collect_free (mixpad->mixcol); - gst_collect_pads_remove_pad (mix->collect, pad); + if (mix->collect) { + gst_collect_pads_remove_pad (mix->collect, pad); + } gst_child_proxy_child_removed (GST_OBJECT (mix), GST_OBJECT (mixpad)); /* determine possibly new geometry and master */ gst_videomixer_set_master_geometry (mix); From 06972db7b6a55f951833327a659afc76bef6b5c8 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Mon, 21 Feb 2011 00:55:49 +0000 Subject: [PATCH 332/351] icydemux: fix tag list handling issues that might have caused crashes Fix slightly confused tag handling in some places: make it clear when we're taking ownership of a tag list and when not. For example, gst_icydemux_tag_found() was taking ownership when the source pad existed, but otherwise not (leak). Also, gst_event_parse_tag() does not return a newly-allocated taglist, but a tag list that belongs to the tag event, so don't give ownership of it away. While we're at it, some minor clean-ups: don't re-invent g_strndup() and simplify gst_icydemux_parse_and_send_tags() a bit, and don't leak the tag list in case no valid tags where found. https://bugzilla.gnome.org/show_bug.cgi?id=641330 --- gst/icydemux/gsticydemux.c | 29 ++++++++++++++++------------- 1 file changed, 16 insertions(+), 13 deletions(-) diff --git a/gst/icydemux/gsticydemux.c b/gst/icydemux/gsticydemux.c index 5965500560..25829598fa 100644 --- a/gst/icydemux/gsticydemux.c +++ b/gst/icydemux/gsticydemux.c @@ -301,6 +301,7 @@ gst_icydemux_unicodify (const gchar * str) return gst_tag_freeform_string_to_utf8 (str, -1, env_vars); } +/* takes ownership of tag list */ static gboolean gst_icydemux_tag_found (GstICYDemux * icydemux, GstTagList * tags) { @@ -309,10 +310,13 @@ gst_icydemux_tag_found (GstICYDemux * icydemux, GstTagList * tags) return gst_icydemux_send_tag_event (icydemux, tags); /* if we haven't a source pad yet, cache the tags */ - if (!icydemux->cached_tags) - icydemux->cached_tags = gst_tag_list_new (); - - gst_tag_list_insert (icydemux->cached_tags, tags, GST_TAG_MERGE_REPLACE_ALL); + if (!icydemux->cached_tags) { + icydemux->cached_tags = tags; + } else { + gst_tag_list_insert (icydemux->cached_tags, tags, + GST_TAG_MERGE_REPLACE_ALL); + gst_tag_list_free (tags); + } return TRUE; } @@ -320,12 +324,11 @@ gst_icydemux_tag_found (GstICYDemux * icydemux, GstTagList * tags) static void gst_icydemux_parse_and_send_tags (GstICYDemux * icydemux) { - GstTagList *tags = gst_tag_list_new (); + GstTagList *tags; const guint8 *data; int length, i; gchar *buffer; gchar **strings; - gboolean found_tag = FALSE; length = gst_adapter_available (icydemux->meta_adapter); @@ -333,10 +336,9 @@ gst_icydemux_parse_and_send_tags (GstICYDemux * icydemux) /* Now, copy this to a buffer where we can NULL-terminate it to make things * a bit easier, then do that parsing. */ - buffer = g_malloc (length + 1); - memcpy (buffer, data, length); - buffer[length] = 0; + buffer = g_strndup ((const gchar *) data, length); + tags = gst_tag_list_new (); strings = g_strsplit (buffer, "';", 0); for (i = 0; strings[i]; i++) { @@ -347,7 +349,6 @@ gst_icydemux_parse_and_send_tags (GstICYDemux * icydemux) gst_tag_list_add (tags, GST_TAG_MERGE_REPLACE, GST_TAG_TITLE, title, NULL); g_free (title); - found_tag = TRUE; } } else if (!g_ascii_strncasecmp (strings[i], "StreamUrl=", 10)) { char *url = gst_icydemux_unicodify (strings[i] + 11); @@ -356,7 +357,6 @@ gst_icydemux_parse_and_send_tags (GstICYDemux * icydemux) gst_tag_list_add (tags, GST_TAG_MERGE_REPLACE, GST_TAG_HOMEPAGE, url, NULL); g_free (url); - found_tag = TRUE; } } } @@ -365,8 +365,10 @@ gst_icydemux_parse_and_send_tags (GstICYDemux * icydemux) g_free (buffer); gst_adapter_clear (icydemux->meta_adapter); - if (found_tag) + if (!gst_tag_list_is_empty (tags)) gst_icydemux_tag_found (icydemux, tags); + else + gst_tag_list_free (tags); } static gboolean @@ -379,7 +381,7 @@ gst_icydemux_handle_event (GstPad * pad, GstEvent * event) GstTagList *tags; gst_event_parse_tag (event, &tags); - result = gst_icydemux_tag_found (icydemux, tags); + result = gst_icydemux_tag_found (icydemux, gst_tag_list_copy (tags)); gst_event_unref (event); return result; } @@ -617,6 +619,7 @@ gst_icydemux_change_state (GstElement * element, GstStateChange transition) return ret; } +/* takes ownership of tag list */ static gboolean gst_icydemux_send_tag_event (GstICYDemux * icydemux, GstTagList * tags) { From 6e1c70150212a718302c882dded00a41dee56eb3 Mon Sep 17 00:00:00 2001 From: Edward Hervey Date: Mon, 21 Feb 2011 12:09:07 +0100 Subject: [PATCH 333/351] Revert "Check that collectpads exists before removing pad" This reverts commit 8e6b876e76c94410db160afe5eb30f21452e419f. Depends on a core commit that was reverted --- ext/cairo/gsttextoverlay.c | 5 +---- gst/avi/gstavimux.c | 5 +---- gst/flv/gstflvmux.c | 5 +---- gst/interleave/interleave.c | 4 +--- gst/matroska/matroska-mux.c | 5 +---- gst/videomixer/videomixer.c | 5 +---- 6 files changed, 6 insertions(+), 23 deletions(-) diff --git a/ext/cairo/gsttextoverlay.c b/ext/cairo/gsttextoverlay.c index 90e81a5bd9..364b8bae94 100644 --- a/ext/cairo/gsttextoverlay.c +++ b/ext/cairo/gsttextoverlay.c @@ -208,7 +208,6 @@ gst_text_overlay_finalize (GObject * object) gst_collect_pads_stop (overlay->collect); gst_object_unref (overlay->collect); - overlay->collect = NULL; g_free (overlay->text_fill_image); g_free (overlay->text_outline_image); @@ -632,9 +631,7 @@ gst_text_overlay_text_pad_unlinked (GstPad * pad) GST_DEBUG_OBJECT (overlay, "Text pad unlinked"); if (overlay->text_collect_data) { - if (overlay->collect) { - gst_collect_pads_remove_pad (overlay->collect, overlay->text_sinkpad); - } + gst_collect_pads_remove_pad (overlay->collect, overlay->text_sinkpad); overlay->text_collect_data = NULL; } diff --git a/gst/avi/gstavimux.c b/gst/avi/gstavimux.c index 1a8e5763ba..d0d2f0ecc6 100644 --- a/gst/avi/gstavimux.c +++ b/gst/avi/gstavimux.c @@ -285,7 +285,6 @@ gst_avi_mux_finalize (GObject * object) mux->idx = NULL; gst_object_unref (mux->collect); - mux->collect = NULL; G_OBJECT_CLASS (parent_class)->finalize (object); } @@ -1090,9 +1089,7 @@ gst_avi_mux_release_pad (GstElement * element, GstPad * pad) * as it also represent number of streams present */ avipad->collect = NULL; GST_DEBUG_OBJECT (avimux, "removed pad '%s'", GST_PAD_NAME (pad)); - if (avimux->collect) { - gst_collect_pads_remove_pad (avimux->collect, pad); - } + gst_collect_pads_remove_pad (avimux->collect, pad); gst_element_remove_pad (element, pad); /* if not started yet, we can remove any sign this pad ever existed */ /* in this case _start will take care of the real pad count */ diff --git a/gst/flv/gstflvmux.c b/gst/flv/gstflvmux.c index 9d69ace8a2..68a0df4fc7 100644 --- a/gst/flv/gstflvmux.c +++ b/gst/flv/gstflvmux.c @@ -207,7 +207,6 @@ gst_flv_mux_finalize (GObject * object) GstFlvMux *mux = GST_FLV_MUX (object); gst_object_unref (mux->collect); - mux->collect = NULL; G_OBJECT_CLASS (parent_class)->finalize (object); } @@ -560,9 +559,7 @@ gst_flv_mux_release_pad (GstElement * element, GstPad * pad) if (cpad && cpad->video_codec_data) gst_buffer_unref (cpad->video_codec_data); - if (mux->collect) { - gst_collect_pads_remove_pad (mux->collect, pad); - } + gst_collect_pads_remove_pad (mux->collect, pad); gst_element_remove_pad (element, pad); } diff --git a/gst/interleave/interleave.c b/gst/interleave/interleave.c index f6ff41af67..96a43c8d61 100644 --- a/gst/interleave/interleave.c +++ b/gst/interleave/interleave.c @@ -598,9 +598,7 @@ gst_interleave_release_pad (GstElement * element, GstPad * pad) GST_OBJECT_UNLOCK (self->collect); - if (self->collect) { - gst_collect_pads_remove_pad (self->collect, pad); - } + gst_collect_pads_remove_pad (self->collect, pad); gst_element_remove_pad (element, pad); } diff --git a/gst/matroska/matroska-mux.c b/gst/matroska/matroska-mux.c index 6d2cc5dec4..31fe40bfc3 100644 --- a/gst/matroska/matroska-mux.c +++ b/gst/matroska/matroska-mux.c @@ -385,7 +385,6 @@ gst_matroska_mux_finalize (GObject * object) GstMatroskaMux *mux = GST_MATROSKA_MUX (object); gst_object_unref (mux->collect); - mux->collect = NULL; gst_object_unref (mux->ebml_write); if (mux->writing_app) g_free (mux->writing_app); @@ -1930,9 +1929,7 @@ gst_matroska_mux_release_pad (GstElement * element, GstPad * pad) } } - if (mux->collect) { - gst_collect_pads_remove_pad (mux->collect, pad); - } + gst_collect_pads_remove_pad (mux->collect, pad); if (gst_element_remove_pad (element, pad)) mux->num_streams--; } diff --git a/gst/videomixer/videomixer.c b/gst/videomixer/videomixer.c index 925b06c6ac..1271d63216 100644 --- a/gst/videomixer/videomixer.c +++ b/gst/videomixer/videomixer.c @@ -777,7 +777,6 @@ gst_videomixer_finalize (GObject * object) GstVideoMixer *mix = GST_VIDEO_MIXER (object); gst_object_unref (mix->collect); - mix->collect = NULL; g_mutex_free (mix->state_lock); G_OBJECT_CLASS (parent_class)->finalize (object); @@ -1252,9 +1251,7 @@ gst_videomixer_release_pad (GstElement * element, GstPad * pad) mix->sinkpads = g_slist_remove (mix->sinkpads, pad); gst_videomixer_collect_free (mixpad->mixcol); - if (mix->collect) { - gst_collect_pads_remove_pad (mix->collect, pad); - } + gst_collect_pads_remove_pad (mix->collect, pad); gst_child_proxy_child_removed (GST_OBJECT (mix), GST_OBJECT (mixpad)); /* determine possibly new geometry and master */ gst_videomixer_set_master_geometry (mix); From 45ea930a993c3cf0493621d54e7302b059bebb3a Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Mon, 21 Feb 2011 17:17:32 +0100 Subject: [PATCH 334/351] rtpbin: fix setting the SDES property Only the sdes veriable is protected with the object lock. Use the right object when setting the sdes property. --- gst/rtpmanager/gstrtpbin.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/gst/rtpmanager/gstrtpbin.c b/gst/rtpmanager/gstrtpbin.c index 5c61460049..a2f7dc423f 100644 --- a/gst/rtpmanager/gstrtpbin.c +++ b/gst/rtpmanager/gstrtpbin.c @@ -1767,11 +1767,13 @@ gst_rtp_bin_set_sdes_struct (GstRtpBin * bin, const GstStructure * sdes) if (bin->sdes) gst_structure_free (bin->sdes); bin->sdes = gst_structure_copy (sdes); + GST_OBJECT_UNLOCK (bin); /* store in all sessions */ - for (item = bin->sessions; item; item = g_slist_next (item)) - g_object_set (item->data, "sdes", sdes, NULL); - GST_OBJECT_UNLOCK (bin); + for (item = bin->sessions; item; item = g_slist_next (item)) { + GstRtpBinSession *session = item->data; + g_object_set (session->session, "sdes", sdes, NULL); + } GST_RTP_BIN_UNLOCK (bin); } From 32f757786b18640f0d87c30aeb26cec40d798e0c Mon Sep 17 00:00:00 2001 From: Robert Swain Date: Mon, 24 Jan 2011 12:18:39 +0100 Subject: [PATCH 335/351] deinterlace: fix assigned method_id when using fallback https://bugzilla.gnome.org/show_bug.cgi?id=642691 --- gst/deinterlace/gstdeinterlace.c | 1 + 1 file changed, 1 insertion(+) diff --git a/gst/deinterlace/gstdeinterlace.c b/gst/deinterlace/gstdeinterlace.c index 2f7d2ea0e6..98c8247937 100644 --- a/gst/deinterlace/gstdeinterlace.c +++ b/gst/deinterlace/gstdeinterlace.c @@ -286,6 +286,7 @@ gst_deinterlace_set_method (GstDeinterlace * self, GstDeinterlaceMethods method) self->height)) { GST_DEBUG_OBJECT (self, "Using method %d", i); method_type = tmp; + method = i; break; } } From 6b26017ae72507c75108285987bb3ed429f6524e Mon Sep 17 00:00:00 2001 From: Robert Swain Date: Mon, 24 Jan 2011 12:18:39 +0100 Subject: [PATCH 336/351] deinterlace: small clean-ups Improve debug output by printing the buffer pointer when popping a buffer and simplify code to use scanlines.bottom_field as appropriate. https://bugzilla.gnome.org/show_bug.cgi?id=642691 --- gst/deinterlace/gstdeinterlace.c | 4 ++-- gst/deinterlace/gstdeinterlacemethod.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/gst/deinterlace/gstdeinterlace.c b/gst/deinterlace/gstdeinterlace.c index 98c8247937..f10f1bc523 100644 --- a/gst/deinterlace/gstdeinterlace.c +++ b/gst/deinterlace/gstdeinterlace.c @@ -770,8 +770,8 @@ gst_deinterlace_pop_history (GstDeinterlace * self) self->history_count--; - GST_DEBUG_OBJECT (self, "Returning buffer: %" GST_TIME_FORMAT - " with duration %" GST_TIME_FORMAT " and size %u", + GST_DEBUG_OBJECT (self, "Returning buffer: %p %" GST_TIME_FORMAT + " with duration %" GST_TIME_FORMAT " and size %u", buffer, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)), GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)), GST_BUFFER_SIZE (buffer)); diff --git a/gst/deinterlace/gstdeinterlacemethod.c b/gst/deinterlace/gstdeinterlacemethod.c index 7228f01676..a5d1058e74 100644 --- a/gst/deinterlace/gstdeinterlacemethod.c +++ b/gst/deinterlace/gstdeinterlacemethod.c @@ -350,7 +350,7 @@ gst_deinterlace_simple_method_deinterlace_frame_packed (GstDeinterlaceMethod * memset (&scanlines, 0, sizeof (scanlines)); scanlines.bottom_field = (cur_field_flags == PICTURE_INTERLACED_BOTTOM); - if (!((i & 1) ^ (cur_field_flags == PICTURE_INTERLACED_BOTTOM))) { + if (!((i & 1) ^ scanlines.bottom_field)) { /* copying */ scanlines.tt0 = LINE2 (field0, (i - 2 >= 0) ? i - 2 : i); scanlines.m0 = LINE2 (field0, i); @@ -453,7 +453,7 @@ static void memset (&scanlines, 0, sizeof (scanlines)); scanlines.bottom_field = (cur_field_flags == PICTURE_INTERLACED_BOTTOM); - if (!((i & 1) ^ (cur_field_flags == PICTURE_INTERLACED_BOTTOM))) { + if (!((i & 1) ^ scanlines.bottom_field)) { /* copying */ scanlines.tt0 = LINE2 (field0, (i - 2 >= 0) ? i - 2 : i); scanlines.m0 = LINE2 (field0, i); From 0f88181f431ddf48092230bf84ca9e0bb9315c16 Mon Sep 17 00:00:00 2001 From: Blaise Gassend Date: Tue, 22 Feb 2011 13:29:26 +0100 Subject: [PATCH 337/351] rtpbin: handle NULL demux elements When using gstrtpbin with ignore-pt=true, the free_stream function tries to call gst_element_set_locked_state and gst_element_set_state on a stream->demux which is NULL. fixes #642412 --- gst/rtpmanager/gstrtpbin.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/gst/rtpmanager/gstrtpbin.c b/gst/rtpmanager/gstrtpbin.c index a2f7dc423f..e8d659ff0c 100644 --- a/gst/rtpmanager/gstrtpbin.c +++ b/gst/rtpmanager/gstrtpbin.c @@ -1336,10 +1336,12 @@ free_stream (GstRtpBinStream * stream) g_signal_handler_disconnect (stream->buffer, stream->buffer_ptreq_sig); g_signal_handler_disconnect (stream->buffer, stream->buffer_ntpstop_sig); - gst_element_set_locked_state (stream->demux, TRUE); + if (stream->demux) + gst_element_set_locked_state (stream->demux, TRUE); gst_element_set_locked_state (stream->buffer, TRUE); - gst_element_set_state (stream->demux, GST_STATE_NULL); + if (stream->demux) + gst_element_set_state (stream->demux, GST_STATE_NULL); gst_element_set_state (stream->buffer, GST_STATE_NULL); /* now remove this signal, we need this while going to NULL because it to From 768dbfaf920610456c3c398f8d54f84c51c8890c Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Tue, 22 Feb 2011 14:50:59 +0200 Subject: [PATCH 338/351] rtp-examples: fix ascii-art Some boxes where misaligned due to long "audiotetssrc" name. Trim trailing whitespace. --- tests/examples/rtp/client-H264-PCMA.sh | 20 +++---- tests/examples/rtp/client-PCMA.sh | 14 ++--- tests/examples/rtp/server-alsasrc-PCMA.sh | 26 ++++----- .../rtp/server-v4l2-H263p-alsasrc-AMR.sh | 2 +- .../rtp/server-v4l2-H264-alsasrc-PCMA.sh | 54 +++++++++---------- 5 files changed, 58 insertions(+), 58 deletions(-) diff --git a/tests/examples/rtp/client-H264-PCMA.sh b/tests/examples/rtp/client-H264-PCMA.sh index 9f007f1f9a..7b104ab2d9 100755 --- a/tests/examples/rtp/client-H264-PCMA.sh +++ b/tests/examples/rtp/client-H264-PCMA.sh @@ -1,6 +1,6 @@ #!/bin/sh # -# A simple RTP receiver +# A simple RTP receiver # # receives H264 encoded RTP video on port 5000, RTCP is received on port 5001. # the receiver RTCP reports are sent to port 5005 @@ -11,27 +11,27 @@ # RTP |udpsrc | | rtpbin | |h264depay| |h264dec| |xvimagesink| # port=5000 | src->recv_rtp recv_rtp->sink src->sink src->sink | # '-------' | | '---------' '-------' '-----------' -# | | +# | | # | | .-------. # | | |udpsink| RTCP # | send_rtcp->sink | port=5005 # .-------. | | '-------' sync=false # RTCP |udpsrc | | | async=false -# port=5001 | src->recv_rtcp | -# '-------' | | +# port=5001 | src->recv_rtcp | +# '-------' | | # | | -# .-------. | | .---------. .-------. .--------. +# .-------. | | .---------. .-------. .-------------. # RTP |udpsrc | | rtpbin | |pcmadepay| |alawdec| |autoaudiosink| -# port=5002 | src->recv_rtp recv_rtp->sink src->sink src->sink | -# '-------' | | '---------' '-------' '--------' -# | | +# port=5002 | src->recv_rtp recv_rtp->sink src->sink src->sink | +# '-------' | | '---------' '-------' '-------------' +# | | # | | .-------. # | | |udpsink| RTCP # | send_rtcp->sink | port=5007 # .-------. | | '-------' sync=false # RTCP |udpsrc | | | async=false -# port=5003 | src->recv_rtcp | -# '-------' '----------' +# port=5003 | src->recv_rtcp | +# '-------' '----------' # the destination machine to send RTCP to. This is the address of the sender and # is used to send back the RTCP reports of this receiver. If the data is sent diff --git a/tests/examples/rtp/client-PCMA.sh b/tests/examples/rtp/client-PCMA.sh index b2c92f7d9e..aad6502e43 100755 --- a/tests/examples/rtp/client-PCMA.sh +++ b/tests/examples/rtp/client-PCMA.sh @@ -1,22 +1,22 @@ #!/bin/sh # -# A simple RTP receiver +# A simple RTP receiver # # receives alaw encoded RTP audio on port 5002, RTCP is received on port 5003. # the receiver RTCP reports are sent to port 5007 # -# .-------. .----------. .---------. .-------. .--------. +# .-------. .----------. .---------. .-------. .-------------. # RTP |udpsrc | | rtpbin | |pcmadepay| |alawdec| |autoaudiosink| -# port=5002 | src->recv_rtp recv_rtp->sink src->sink src->sink | -# '-------' | | '---------' '-------' '--------' -# | | +# port=5002 | src->recv_rtp recv_rtp->sink src->sink src->sink | +# '-------' | | '---------' '-------' '-------------' +# | | # | | .-------. # | | |udpsink| RTCP # | send_rtcp->sink | port=5007 # .-------. | | '-------' sync=false # RTCP |udpsrc | | | async=false -# port=5003 | src->recv_rtcp | -# '-------' '----------' +# port=5003 | src->recv_rtcp | +# '-------' '----------' # the caps of the sender RTP stream. This is usually negotiated out of band with diff --git a/tests/examples/rtp/server-alsasrc-PCMA.sh b/tests/examples/rtp/server-alsasrc-PCMA.sh index c0fe0e7215..5340434ad7 100755 --- a/tests/examples/rtp/server-alsasrc-PCMA.sh +++ b/tests/examples/rtp/server-alsasrc-PCMA.sh @@ -1,22 +1,22 @@ #!/bin/sh # -# A simple RTP server +# A simple RTP server # sends the output of autoaudiosrc as alaw encoded RTP on port 5002, RTCP is sent on # port 5003. The destination is 127.0.0.1. # the receiver RTCP reports are received on port 5007 # -# .-------. .-------. .-------. .----------. .-------. -# |autoaudiosrc| |alawenc| |pcmapay| | rtpbin | |udpsink| RTP -# | src->sink src->sink src->send_rtp send_rtp->sink | port=5002 -# '-------' '-------' '-------' | | '-------' -# | | -# | | .-------. -# | | |udpsink| RTCP -# | send_rtcp->sink | port=5003 -# .-------. | | '-------' sync=false -# RTCP |udpsrc | | | async=false -# port=5007 | src->recv_rtcp | -# '-------' '----------' +# .--------. .-------. .-------. .----------. .-------. +# |audiosrc| |alawenc| |pcmapay| | rtpbin | |udpsink| RTP +# | src->sink src->sink src->send_rtp send_rtp->sink | port=5002 +# '--------' '-------' '-------' | | '-------' +# | | +# | | .-------. +# | | |udpsink| RTCP +# | send_rtcp->sink | port=5003 +# .-------. | | '-------' sync=false +# RTCP |udpsrc | | | async=false +# port=5007 | src->recv_rtcp | +# '-------' '----------' # change this to send the RTP data and RTCP to another host DEST=127.0.0.1 diff --git a/tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh b/tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh index 2a8dc5cf50..12439feaed 100755 --- a/tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh +++ b/tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh @@ -1,6 +1,6 @@ #!/bin/sh # -# A simple RTP server +# A simple RTP server # # change these to change the server sync. This causes the server to send the diff --git a/tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh b/tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh index 910b69e1a8..7ccb4f14b8 100755 --- a/tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh +++ b/tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh @@ -1,6 +1,6 @@ #!/bin/sh # -# A simple RTP server +# A simple RTP server # sends the output of v4l2src as h264 encoded RTP on port 5000, RTCP is sent on # port 5001. The destination is 127.0.0.1. # the video receiver RTCP reports are received on port 5005 @@ -8,31 +8,31 @@ # port 5003. The destination is 127.0.0.1. # the receiver RTCP reports are received on port 5007 # -# .-------. .-------. .-------. .----------. .-------. -# |v4lssrc| |h264enc| |h264pay| | rtpbin | |udpsink| RTP -# | src->sink src->sink src->send_rtp send_rtp->sink | port=5000 -# '-------' '-------' '-------' | | '-------' -# | | -# | | .-------. -# | | |udpsink| RTCP -# | send_rtcp->sink | port=5001 -# .-------. | | '-------' sync=false -# RTCP |udpsrc | | | async=false -# port=5005 | src->recv_rtcp | -# '-------' | | -# | | -# .-------. .-------. .-------. | | .-------. -# |autoaudiosrc| |alawenc| |pcmapay| | rtpbin | |udpsink| RTP -# | src->sink src->sink src->send_rtp send_rtp->sink | port=5002 -# '-------' '-------' '-------' | | '-------' -# | | -# | | .-------. -# | | |udpsink| RTCP -# | send_rtcp->sink | port=5003 -# .-------. | | '-------' sync=false -# RTCP |udpsrc | | | async=false -# port=5007 | src->recv_rtcp | -# '-------' '----------' +# .-------. .-------. .-------. .----------. .-------. +# |v4lssrc| |h264enc| |h264pay| | rtpbin | |udpsink| RTP +# | src->sink src->sink src->send_rtp send_rtp->sink | port=5000 +# '-------' '-------' '-------' | | '-------' +# | | +# | | .-------. +# | | |udpsink| RTCP +# | send_rtcp->sink | port=5001 +# .-------. | | '-------' sync=false +# RTCP |udpsrc | | | async=false +# port=5005 | src->recv_rtcp | +# '-------' | | +# | | +# .--------. .-------. .-------. | | .-------. +# |audiosrc| |alawenc| |pcmapay| | rtpbin | |udpsink| RTP +# | src->sink src->sink src->send_rtp send_rtp->sink | port=5002 +# '--------' '-------' '-------' | | '-------' +# | | +# | | .-------. +# | | |udpsink| RTCP +# | send_rtcp->sink | port=5003 +# .-------. | | '-------' sync=false +# RTCP |udpsrc | | | async=false +# port=5007 | src->recv_rtcp | +# '-------' '----------' # # ideally we should transport the properties on the RTP udpsink pads to the # receiver in order to transmit the SPS and PPS earlier. @@ -41,7 +41,7 @@ DEST=127.0.0.1 # tuning parameters to make the sender send the streams out of sync. Can be used -# ot test the client RTCP synchronisation. +# ot test the client RTCP synchronisation. #VOFFSET=900000000 VOFFSET=0 AOFFSET=0 From f73cc87239271e5b3b5f0cf05343e8ee6994985a Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Tue, 22 Feb 2011 14:52:18 +0200 Subject: [PATCH 339/351] rtp-examples: move capsfilter behind converters We need to have the capsfilter behin the converters to make the converters convert from the formats v4l2src can do to what we request with the capsfilter. --- tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh | 2 +- tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh b/tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh index 12439feaed..dd7fc8997a 100755 --- a/tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh +++ b/tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh @@ -14,7 +14,7 @@ VCAPS="video/x-raw-yuv,width=352,height=288,framerate=15/1" DEST=192.168.1.126 gst-launch -v gstrtpbin name=rtpbin \ - v4l2src ! $VCAPS ! videorate ! ffmpegcolorspace ! ffenc_h263p ! rtph263ppay ! rtpbin.send_rtp_sink_0 \ + v4l2src ! videorate ! ffmpegcolorspace ! $VCAPS ! ffenc_h263p ! rtph263ppay ! rtpbin.send_rtp_sink_0 \ rtpbin.send_rtp_src_0 ! queue ! udpsink host=$HOST port=5000 ts-offset=$AOFFSET \ rtpbin.send_rtcp_src_0 ! udpsink host=$HOST port=5001 sync=false async=false \ udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 \ diff --git a/tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh b/tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh index 7ccb4f14b8..8dc4d70900 100755 --- a/tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh +++ b/tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh @@ -50,7 +50,7 @@ AOFFSET=0 VELEM="v4l2src" #VELEM="videotestsrc is-live=1" VCAPS="video/x-raw-yuv,width=352,height=288,framerate=15/1" -VSOURCE="$VELEM ! $VCAPS ! queue ! videorate ! ffmpegcolorspace" +VSOURCE="$VELEM ! queue ! videorate ! ffmpegcolorspace ! $VCAPS" VENC="x264enc tune=zerolatency byte-stream=true bitrate=300 ! rtph264pay" VRTPSINK="udpsink port=5000 host=$DEST ts-offset=$VOFFSET name=vrtpsink" From d680842a7f33c156cb122dc56fbf8967be1dfc68 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Tue, 22 Feb 2011 14:00:11 +0100 Subject: [PATCH 340/351] avidemux: check for NULL audio stream format header when parsing stream --- gst/avi/gstavidemux.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/gst/avi/gstavidemux.c b/gst/avi/gstavidemux.c index 95131d3a41..0627710b6f 100644 --- a/gst/avi/gstavidemux.c +++ b/gst/avi/gstavidemux.c @@ -2020,10 +2020,12 @@ gst_avi_demux_parse_stream (GstAviDemux * avi, GstBuffer * buf) res = gst_riff_parse_strf_auds (element, sub, &stream->strf.auds, &stream->extradata); + sub = NULL; + if (!res) + break; stream->is_vbr = (stream->strh->samplesize == 0) && stream->strh->scale > 1 && stream->strf.auds->blockalign != 1; - sub = NULL; GST_DEBUG_OBJECT (element, "marking audio as VBR:%d, res %d", stream->is_vbr, res); /* we need these or we have no way to come up with timestamps */ From 940417edc32ac449b2e44110d2512ad8db4c4cd2 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Tue, 22 Feb 2011 14:01:27 +0100 Subject: [PATCH 341/351] avidemux: ensure sane parameters when parsing superindex --- gst/avi/gstavidemux.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/gst/avi/gstavidemux.c b/gst/avi/gstavidemux.c index 0627710b6f..011eb7d1c5 100644 --- a/gst/avi/gstavidemux.c +++ b/gst/avi/gstavidemux.c @@ -1270,6 +1270,11 @@ gst_avi_demux_parse_superindex (GstAviDemux * avi, GST_DEBUG_OBJECT (avi, "got %d indexes", num); + /* this can't work out well ... */ + if (num > G_MAXUINT32 >> 1 || bpe < 8) { + goto invalid_params; + } + indexes = g_new (guint64, num + 1); for (i = 0; i < num; i++) { if (size < 24 + bpe * (i + 1)) @@ -1293,6 +1298,14 @@ too_small: gst_buffer_unref (buf); return FALSE; } +invalid_params: + { + GST_ERROR_OBJECT (avi, "invalid index parameters (num = %d, bpe = %d)", + num, bpe); + if (buf) + gst_buffer_unref (buf); + return FALSE; + } } /* add an entry to the index of a stream. @num should be an estimate of the From e08bdf691d10d7f307b2de83d0dd01bf8b7f458d Mon Sep 17 00:00:00 2001 From: Edward Hervey Date: Tue, 22 Feb 2011 14:15:27 +0100 Subject: [PATCH 342/351] qtdemux: Fix unitialized variable. --- gst/qtdemux/qtdemux.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/gst/qtdemux/qtdemux.c b/gst/qtdemux/qtdemux.c index 18b4ff1470..2003a3b77b 100644 --- a/gst/qtdemux/qtdemux.c +++ b/gst/qtdemux/qtdemux.c @@ -703,7 +703,7 @@ gst_qtdemux_handle_src_query (GstPad * pad, GstQuery * query) } case GST_QUERY_CONVERT:{ GstFormat src_fmt, dest_fmt; - gint64 src_value, dest_value; + gint64 src_value, dest_value = 0; gst_query_parse_convert (query, &src_fmt, &src_value, &dest_fmt, NULL); From ff5e5a8f0daa1fdf89792d0726ea063bbd99db18 Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Tue, 22 Feb 2011 15:19:00 +0200 Subject: [PATCH 343/351] id3demux: return ID3TAGS_BROKEN_TAG for unsupported versions This prevents us for trying to work with a NULL taglist. --- gst/id3demux/id3tags.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/gst/id3demux/id3tags.c b/gst/id3demux/id3tags.c index 709cda87d3..d201d7c289 100644 --- a/gst/id3demux/id3tags.c +++ b/gst/id3demux/id3tags.c @@ -159,7 +159,7 @@ id3demux_read_id3v2_tag (GstBuffer * buffer, guint * id3v2_size, GST_WARNING ("ID3v2 tag is from revision 2.%d.%d, " "but decoder only supports 2.%d.%d. Ignoring as per spec.", version >> 8, version & 0xff, ID3V2_VERSION >> 8, ID3V2_VERSION & 0xff); - return ID3TAGS_READ_TAG; + return ID3TAGS_BROKEN_TAG; } GST_DEBUG ("ID3v2 header flags: %s %s %s %s", From 98501f6f4551662a10b1c0845d182e8abac9701a Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Tue, 22 Feb 2011 15:20:11 +0200 Subject: [PATCH 344/351] Revert "id3demux: ensure a taglist before adding the container tag" This reverts commit a86bab66893bb1a3323a756410573c117b8219ef. The issue is fixed with commit ff5e5a8f0daa1fdf89792d0726ea063bbd99db18 instead. --- gst/id3demux/gstid3demux.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/gst/id3demux/gstid3demux.c b/gst/id3demux/gstid3demux.c index e91524211f..93685041ab 100644 --- a/gst/id3demux/gstid3demux.c +++ b/gst/id3demux/gstid3demux.c @@ -25,7 +25,7 @@ * id3demux accepts data streams with either (or both) ID3v2 regions at the * start, or ID3v1 at the end. The mime type of the data between the tag blocks * is detected using typefind functions, and the appropriate output mime type - * set on outgoing buffers. + * set on outgoing buffers. * * The element is only able to read ID3v1 tags from a seekable stream, because * they are at the end of the stream. That is, when get_range mode is supported @@ -183,8 +183,6 @@ gst_id3demux_parse_tag (GstTagDemux * demux, GstBuffer * buffer, res = id3demux_read_id3v2_tag (buffer, tag_size, tags); if (G_LIKELY (res == ID3TAGS_READ_TAG)) { - if (*tags == NULL) - *tags = gst_tag_list_new (); gst_id3demux_add_container_format (*tags); return GST_TAG_DEMUX_RESULT_OK; } else { @@ -193,8 +191,9 @@ gst_id3demux_parse_tag (GstTagDemux * demux, GstBuffer * buffer, } else { *tags = gst_tag_list_new_from_id3v1 (GST_BUFFER_DATA (buffer)); - if (*tags == NULL) - *tags = gst_tag_list_new (); + if (G_UNLIKELY (*tags == NULL)) + return GST_TAG_DEMUX_RESULT_BROKEN_TAG; + gst_id3demux_add_container_format (*tags); *tag_size = ID3V1_TAG_SIZE; return GST_TAG_DEMUX_RESULT_OK; From 6402556157c8a1f9da6eae421b426293f6f5039c Mon Sep 17 00:00:00 2001 From: Robert Swain Date: Mon, 24 Jan 2011 12:48:18 +0100 Subject: [PATCH 345/351] deinterlace: Port greedyl to GstDeinterlaceSimpleMethod The main goal of this change is to reuse the complex but now neatly written scanline pointer calculation code from the simple methods. --- gst/deinterlace/tvtime/greedy.c | 274 ++++++++++---------------------- 1 file changed, 84 insertions(+), 190 deletions(-) diff --git a/gst/deinterlace/tvtime/greedy.c b/gst/deinterlace/tvtime/greedy.c index 383bc3409c..dcb495b7ee 100644 --- a/gst/deinterlace/tvtime/greedy.c +++ b/gst/deinterlace/tvtime/greedy.c @@ -49,20 +49,12 @@ GType gst_deinterlace_method_greedy_l_get_type (void); typedef struct { - GstDeinterlaceMethod parent; + GstDeinterlaceSimpleMethod parent; guint max_comb; } GstDeinterlaceMethodGreedyL; -typedef void (*GreedyLScanlineFunction) (GstDeinterlaceMethodGreedyL * self, - const guint8 * L2, const guint8 * L1, const guint8 * L3, const guint8 * L2P, - guint8 * Dest, gint width); - -typedef struct -{ - GstDeinterlaceMethodClass parent_class; - GreedyLScanlineFunction scanline; -} GstDeinterlaceMethodGreedyLClass; +typedef GstDeinterlaceSimpleMethodClass GstDeinterlaceMethodGreedyLClass; // This is a simple lightweight DeInterlace method that uses little CPU time // but gives very good results for low or intermedite motion. @@ -77,186 +69,61 @@ typedef struct // Blended Clip but this give too good results for the CPU to ignore here. static inline void -deinterlace_greedy_scanline_orc (GstDeinterlaceMethodGreedyL * self, - const guint8 * m0, const guint8 * t1, - const guint8 * b1, const guint8 * m2, guint8 * output, gint width) +deinterlace_greedy_interpolate_scanline_orc (GstDeinterlaceSimpleMethod * self, + guint8 * out, const GstDeinterlaceScanlineData * scanlines) { - deinterlace_line_greedy (output, m0, t1, b1, m2, self->max_comb, width); + /* FIXME - is this safe or just a hack? */ + guint max_comb = GST_DEINTERLACE_METHOD_GREEDY_L (self)->max_comb; + + deinterlace_line_greedy (out, scanlines->m3, scanlines->t2, scanlines->b2, + scanlines->m1, max_comb, self->parent.row_stride[0]); +} + +static inline void +deinterlace_greedy_interpolate_scanline_orc_planar_u (GstDeinterlaceSimpleMethod + * self, guint8 * out, const GstDeinterlaceScanlineData * scanlines) +{ + /* FIXME - is this safe or just a hack? */ + guint max_comb = GST_DEINTERLACE_METHOD_GREEDY_L (self)->max_comb; + + deinterlace_line_greedy (out, scanlines->m3, scanlines->t2, scanlines->b2, + scanlines->m1, max_comb, self->parent.row_stride[1]); +} + +static inline void +deinterlace_greedy_interpolate_scanline_orc_planar_v (GstDeinterlaceSimpleMethod + * self, guint8 * out, const GstDeinterlaceScanlineData * scanlines) +{ + /* FIXME - is this safe or just a hack? */ + guint max_comb = GST_DEINTERLACE_METHOD_GREEDY_L (self)->max_comb; + + deinterlace_line_greedy (out, scanlines->m3, scanlines->t2, scanlines->b2, + scanlines->m1, max_comb, self->parent.row_stride[2]); } static void -deinterlace_frame_di_greedy_packed (GstDeinterlaceMethod * method, - const GstDeinterlaceField * history, guint history_count, - GstBuffer * outbuf) +deinterlace_greedy_copy_scanline (GstDeinterlaceSimpleMethod * self, + guint8 * out, const GstDeinterlaceScanlineData * scanlines) { - GstDeinterlaceMethodGreedyL *self = GST_DEINTERLACE_METHOD_GREEDY_L (method); - GstDeinterlaceMethodGreedyLClass *klass = - GST_DEINTERLACE_METHOD_GREEDY_L_GET_CLASS (self); - gint InfoIsOdd = 0; - gint Line; - gint RowStride = method->row_stride[0]; - gint FieldHeight = method->frame_height / 2; - gint Pitch = method->row_stride[0] * 2; - const guint8 *L1; // ptr to Line1, of 3 - const guint8 *L2; // ptr to Line2, the weave line - const guint8 *L3; // ptr to Line3 - const guint8 *L2P; // ptr to prev Line2 - guint8 *Dest = GST_BUFFER_DATA (outbuf); - - // copy first even line no matter what, and the first odd line if we're - // processing an EVEN field. (note diff from other deint rtns.) - - if (history[history_count - 1].flags == PICTURE_INTERLACED_BOTTOM) { - InfoIsOdd = 1; - - L1 = GST_BUFFER_DATA (history[history_count - 2].buf); - if (history[history_count - 2].flags & PICTURE_INTERLACED_BOTTOM) - L1 += RowStride; - - L2 = GST_BUFFER_DATA (history[history_count - 1].buf); - if (history[history_count - 1].flags & PICTURE_INTERLACED_BOTTOM) - L2 += RowStride; - - L3 = L1 + Pitch; - L2P = GST_BUFFER_DATA (history[history_count - 3].buf); - if (history[history_count - 3].flags & PICTURE_INTERLACED_BOTTOM) - L2P += RowStride; - - // copy first even line - memcpy (Dest, L1, RowStride); - Dest += RowStride; - } else { - InfoIsOdd = 0; - L1 = GST_BUFFER_DATA (history[history_count - 2].buf); - if (history[history_count - 2].flags & PICTURE_INTERLACED_BOTTOM) - L1 += RowStride; - - L2 = GST_BUFFER_DATA (history[history_count - 1].buf) + Pitch; - if (history[history_count - 1].flags & PICTURE_INTERLACED_BOTTOM) - L2 += RowStride; - - L3 = L1 + Pitch; - L2P = GST_BUFFER_DATA (history[history_count - 3].buf) + Pitch; - if (history[history_count - 3].flags & PICTURE_INTERLACED_BOTTOM) - L2P += RowStride; - - // copy first even line - memcpy (Dest, L1, RowStride); - Dest += RowStride; - // then first odd line - memcpy (Dest, L1, RowStride); - Dest += RowStride; - } - - for (Line = 0; Line < (FieldHeight - 1); ++Line) { - klass->scanline (self, L2, L1, L3, L2P, Dest, RowStride); - Dest += RowStride; - memcpy (Dest, L3, RowStride); - Dest += RowStride; - - L1 += Pitch; - L2 += Pitch; - L3 += Pitch; - L2P += Pitch; - } - - if (InfoIsOdd) { - memcpy (Dest, L2, RowStride); - } + memcpy (out, scanlines->m2, self->parent.row_stride[0]); } static void -deinterlace_frame_di_greedy_planar_plane (GstDeinterlaceMethodGreedyL * self, - const guint8 * L1, const guint8 * L2, const guint8 * L3, const guint8 * L2P, - guint8 * Dest, gint RowStride, gint FieldHeight, gint Pitch, gint InfoIsOdd, - GreedyLScanlineFunction scanline) +deinterlace_greedy_copy_scanline_planar_u (GstDeinterlaceSimpleMethod * self, + guint8 * out, const GstDeinterlaceScanlineData * scanlines) { - gint Line; - - // copy first even line no matter what, and the first odd line if we're - // processing an EVEN field. (note diff from other deint rtns.) - - if (InfoIsOdd) { - // copy first even line - memcpy (Dest, L1, RowStride); - Dest += RowStride; - } else { - // copy first even line - memcpy (Dest, L1, RowStride); - Dest += RowStride; - // then first odd line - memcpy (Dest, L1, RowStride); - Dest += RowStride; - } - - for (Line = 0; Line < (FieldHeight - 1); ++Line) { - scanline (self, L2, L1, L3, L2P, Dest, RowStride); - Dest += RowStride; - memcpy (Dest, L3, RowStride); - Dest += RowStride; - - L1 += Pitch; - L2 += Pitch; - L3 += Pitch; - L2P += Pitch; - } - - if (InfoIsOdd) { - memcpy (Dest, L2, RowStride); - } + memcpy (out, scanlines->m2, self->parent.row_stride[1]); } static void -deinterlace_frame_di_greedy_planar (GstDeinterlaceMethod * method, - const GstDeinterlaceField * history, guint history_count, - GstBuffer * outbuf) +deinterlace_greedy_copy_scanline_planar_v (GstDeinterlaceSimpleMethod * self, + guint8 * out, const GstDeinterlaceScanlineData * scanlines) { - GstDeinterlaceMethodGreedyL *self = GST_DEINTERLACE_METHOD_GREEDY_L (method); - GstDeinterlaceMethodGreedyLClass *klass = - GST_DEINTERLACE_METHOD_GREEDY_L_GET_CLASS (self); - gint InfoIsOdd; - gint RowStride; - gint FieldHeight; - gint Pitch; - const guint8 *L1; // ptr to Line1, of 3 - const guint8 *L2; // ptr to Line2, the weave line - const guint8 *L3; // ptr to Line3 - const guint8 *L2P; // ptr to prev Line2 - guint8 *Dest; - gint i; - gint Offset; - GreedyLScanlineFunction scanline = klass->scanline; - - for (i = 0; i < 3; i++) { - Offset = method->offset[i]; - - InfoIsOdd = (history[history_count - 1].flags == PICTURE_INTERLACED_BOTTOM); - RowStride = method->row_stride[i]; - FieldHeight = method->height[i] / 2; - Pitch = method->row_stride[i] * 2; - - Dest = GST_BUFFER_DATA (outbuf) + Offset; - - L1 = GST_BUFFER_DATA (history[history_count - 2].buf) + Offset; - if (history[history_count - 2].flags & PICTURE_INTERLACED_BOTTOM) - L1 += RowStride; - - L2 = GST_BUFFER_DATA (history[history_count - 1].buf) + Offset; - if (history[history_count - 1].flags & PICTURE_INTERLACED_BOTTOM) - L2 += RowStride; - - L3 = L1 + Pitch; - L2P = GST_BUFFER_DATA (history[history_count - 3].buf) + Offset; - if (history[history_count - 3].flags & PICTURE_INTERLACED_BOTTOM) - L2P += RowStride; - - deinterlace_frame_di_greedy_planar_plane (self, L1, L2, L3, L2P, Dest, - RowStride, FieldHeight, Pitch, InfoIsOdd, scanline); - } + memcpy (out, scanlines->m2, self->parent.row_stride[2]); } G_DEFINE_TYPE (GstDeinterlaceMethodGreedyL, gst_deinterlace_method_greedy_l, - GST_TYPE_DEINTERLACE_METHOD); + GST_TYPE_DEINTERLACE_SIMPLE_METHOD); enum { @@ -299,6 +166,8 @@ gst_deinterlace_method_greedy_l_class_init (GstDeinterlaceMethodGreedyLClass * klass) { GstDeinterlaceMethodClass *dim_class = (GstDeinterlaceMethodClass *) klass; + GstDeinterlaceSimpleMethodClass *dism_class = + (GstDeinterlaceSimpleMethodClass *) klass; GObjectClass *gobject_class = (GObjectClass *) klass; gobject_class->set_property = gst_deinterlace_method_greedy_l_set_property; @@ -315,23 +184,48 @@ gst_deinterlace_method_greedy_l_class_init (GstDeinterlaceMethodGreedyLClass * dim_class->nick = "greedyl"; dim_class->latency = 1; - dim_class->deinterlace_frame_yuy2 = deinterlace_frame_di_greedy_packed; - dim_class->deinterlace_frame_yvyu = deinterlace_frame_di_greedy_packed; - dim_class->deinterlace_frame_uyvy = deinterlace_frame_di_greedy_packed; - dim_class->deinterlace_frame_y444 = deinterlace_frame_di_greedy_planar; - dim_class->deinterlace_frame_y42b = deinterlace_frame_di_greedy_planar; - dim_class->deinterlace_frame_i420 = deinterlace_frame_di_greedy_planar; - dim_class->deinterlace_frame_yv12 = deinterlace_frame_di_greedy_planar; - dim_class->deinterlace_frame_y41b = deinterlace_frame_di_greedy_planar; - dim_class->deinterlace_frame_ayuv = deinterlace_frame_di_greedy_planar; - dim_class->deinterlace_frame_argb = deinterlace_frame_di_greedy_packed; - dim_class->deinterlace_frame_rgba = deinterlace_frame_di_greedy_packed; - dim_class->deinterlace_frame_abgr = deinterlace_frame_di_greedy_packed; - dim_class->deinterlace_frame_bgra = deinterlace_frame_di_greedy_packed; - dim_class->deinterlace_frame_rgb = deinterlace_frame_di_greedy_packed; - dim_class->deinterlace_frame_bgr = deinterlace_frame_di_greedy_packed; + dism_class->interpolate_scanline_ayuv = + deinterlace_greedy_interpolate_scanline_orc; + dism_class->interpolate_scanline_yuy2 = + deinterlace_greedy_interpolate_scanline_orc; + dism_class->interpolate_scanline_yvyu = + deinterlace_greedy_interpolate_scanline_orc; + dism_class->interpolate_scanline_uyvy = + deinterlace_greedy_interpolate_scanline_orc; + dism_class->interpolate_scanline_argb = + deinterlace_greedy_interpolate_scanline_orc; + dism_class->interpolate_scanline_abgr = + deinterlace_greedy_interpolate_scanline_orc; + dism_class->interpolate_scanline_rgba = + deinterlace_greedy_interpolate_scanline_orc; + dism_class->interpolate_scanline_bgra = + deinterlace_greedy_interpolate_scanline_orc; + dism_class->interpolate_scanline_rgb = + deinterlace_greedy_interpolate_scanline_orc; + dism_class->interpolate_scanline_bgr = + deinterlace_greedy_interpolate_scanline_orc; + dism_class->interpolate_scanline_planar_y = + deinterlace_greedy_interpolate_scanline_orc; + dism_class->interpolate_scanline_planar_u = + deinterlace_greedy_interpolate_scanline_orc_planar_u; + dism_class->interpolate_scanline_planar_v = + deinterlace_greedy_interpolate_scanline_orc_planar_v; - klass->scanline = deinterlace_greedy_scanline_orc; + dism_class->copy_scanline_ayuv = deinterlace_greedy_copy_scanline; + dism_class->copy_scanline_yuy2 = deinterlace_greedy_copy_scanline; + dism_class->copy_scanline_yvyu = deinterlace_greedy_copy_scanline; + dism_class->copy_scanline_uyvy = deinterlace_greedy_copy_scanline; + dism_class->copy_scanline_argb = deinterlace_greedy_copy_scanline; + dism_class->copy_scanline_abgr = deinterlace_greedy_copy_scanline; + dism_class->copy_scanline_rgba = deinterlace_greedy_copy_scanline; + dism_class->copy_scanline_bgra = deinterlace_greedy_copy_scanline; + dism_class->copy_scanline_rgb = deinterlace_greedy_copy_scanline; + dism_class->copy_scanline_bgr = deinterlace_greedy_copy_scanline; + dism_class->copy_scanline_planar_y = deinterlace_greedy_copy_scanline; + dism_class->copy_scanline_planar_u = + deinterlace_greedy_copy_scanline_planar_u; + dism_class->copy_scanline_planar_v = + deinterlace_greedy_copy_scanline_planar_v; } static void From 6607cdcc08621c99ad96be4d7c57c92674d7c785 Mon Sep 17 00:00:00 2001 From: Robert Swain Date: Mon, 8 Nov 2010 14:25:59 +0100 Subject: [PATCH 346/351] deinterlace: Simplify setcaps The current code never uses upstream negotiation so the code can be significantly simplified. --- gst/deinterlace/gstdeinterlace.c | 81 ++++++++++++-------------------- gst/deinterlace/gstdeinterlace.h | 4 -- 2 files changed, 30 insertions(+), 55 deletions(-) diff --git a/gst/deinterlace/gstdeinterlace.c b/gst/deinterlace/gstdeinterlace.c index f10f1bc523..12b1dd6315 100644 --- a/gst/deinterlace/gstdeinterlace.c +++ b/gst/deinterlace/gstdeinterlace.c @@ -564,8 +564,6 @@ gst_deinterlace_init (GstDeinterlace * self, GstDeinterlaceClass * klass) GST_DEBUG_FUNCPTR (gst_deinterlace_src_query)); gst_pad_set_getcaps_function (self->srcpad, GST_DEBUG_FUNCPTR (gst_deinterlace_getcaps)); - gst_pad_set_setcaps_function (self->srcpad, - GST_DEBUG_FUNCPTR (gst_deinterlace_setcaps)); gst_element_add_pad (GST_ELEMENT (self), self->srcpad); self->mode = DEFAULT_MODE; @@ -645,14 +643,6 @@ gst_deinterlace_reset (GstDeinterlace * self) gst_segment_init (&self->segment, GST_FORMAT_UNDEFINED); - if (self->sink_caps) - gst_caps_unref (self->sink_caps); - self->sink_caps = NULL; - - if (self->src_caps) - gst_caps_unref (self->src_caps); - self->src_caps = NULL; - if (self->request_caps) gst_caps_unref (self->request_caps); self->request_caps = NULL; @@ -1002,13 +992,15 @@ gst_deinterlace_output_frame (GstDeinterlace * self, gboolean flushing) GST_DEBUG_OBJECT (self, "deinterlacing top field"); /* create new buffer */ - ret = gst_pad_alloc_buffer (self->srcpad, - GST_BUFFER_OFFSET_NONE, self->frame_size, self->src_caps, &outbuf); + ret = + gst_pad_alloc_buffer (self->srcpad, GST_BUFFER_OFFSET_NONE, + self->frame_size, GST_PAD_CAPS (self->srcpad), &outbuf); if (ret != GST_FLOW_OK) return ret; - if (self->src_caps != GST_BUFFER_CAPS (outbuf) && - !gst_caps_is_equal (self->src_caps, GST_BUFFER_CAPS (outbuf))) { + if (GST_PAD_CAPS (self->srcpad) != GST_BUFFER_CAPS (outbuf) + && !gst_caps_is_equal (GST_PAD_CAPS (self->srcpad), + GST_BUFFER_CAPS (outbuf))) { gst_caps_replace (&self->request_caps, GST_BUFFER_CAPS (outbuf)); GST_DEBUG_OBJECT (self, "Upstream wants new caps %" GST_PTR_FORMAT, self->request_caps); @@ -1019,7 +1011,7 @@ gst_deinterlace_output_frame (GstDeinterlace * self, gboolean flushing) if (!outbuf) return GST_FLOW_ERROR; - gst_buffer_set_caps (outbuf, self->src_caps); + gst_buffer_set_caps (outbuf, GST_PAD_CAPS (self->srcpad)); } g_return_val_if_fail (self->history_count - 1 - @@ -1080,13 +1072,15 @@ gst_deinterlace_output_frame (GstDeinterlace * self, gboolean flushing) GST_DEBUG_OBJECT (self, "deinterlacing bottom field"); /* create new buffer */ - ret = gst_pad_alloc_buffer (self->srcpad, - GST_BUFFER_OFFSET_NONE, self->frame_size, self->src_caps, &outbuf); + ret = + gst_pad_alloc_buffer (self->srcpad, GST_BUFFER_OFFSET_NONE, + self->frame_size, GST_PAD_CAPS (self->srcpad), &outbuf); if (ret != GST_FLOW_OK) return ret; - if (self->src_caps != GST_BUFFER_CAPS (outbuf) && - !gst_caps_is_equal (self->src_caps, GST_BUFFER_CAPS (outbuf))) { + if (GST_PAD_CAPS (self->srcpad) != GST_BUFFER_CAPS (outbuf) + && !gst_caps_is_equal (GST_PAD_CAPS (self->srcpad), + GST_BUFFER_CAPS (outbuf))) { gst_caps_replace (&self->request_caps, GST_BUFFER_CAPS (outbuf)); GST_DEBUG_OBJECT (self, "Upstream wants new caps %" GST_PTR_FORMAT, self->request_caps); @@ -1097,7 +1091,7 @@ gst_deinterlace_output_frame (GstDeinterlace * self, gboolean flushing) if (!outbuf) return GST_FLOW_ERROR; - gst_buffer_set_caps (outbuf, self->src_caps); + gst_buffer_set_caps (outbuf, GST_PAD_CAPS (self->srcpad)); } g_return_val_if_fail (self->history_count - 1 - @@ -1391,10 +1385,7 @@ gst_deinterlace_setcaps (GstPad * pad, GstCaps * caps) { gboolean res = TRUE; GstDeinterlace *self = GST_DEINTERLACE (gst_pad_get_parent (pad)); - GstPad *otherpad; - GstCaps *othercaps; - - otherpad = (pad == self->srcpad) ? self->sinkpad : self->srcpad; + GstCaps *srccaps; res = gst_video_format_parse_caps (caps, &self->format, &self->width, @@ -1410,57 +1401,45 @@ gst_deinterlace_setcaps (GstPad * pad, GstCaps * caps) if (!self->passthrough && self->fields == GST_DEINTERLACE_ALL) { gint fps_n = self->fps_n, fps_d = self->fps_d; - if (!gst_fraction_double (&fps_n, &fps_d, otherpad != self->srcpad)) + if (!gst_fraction_double (&fps_n, &fps_d, FALSE)) goto invalid_caps; - othercaps = gst_caps_copy (caps); + srccaps = gst_caps_copy (caps); - gst_caps_set_simple (othercaps, "framerate", GST_TYPE_FRACTION, fps_n, + gst_caps_set_simple (srccaps, "framerate", GST_TYPE_FRACTION, fps_n, fps_d, NULL); } else { - othercaps = gst_caps_ref (caps); + srccaps = gst_caps_ref (caps); } - if (otherpad == self->srcpad && self->mode != GST_DEINTERLACE_MODE_DISABLED) { - othercaps = gst_caps_make_writable (othercaps); - gst_caps_set_simple (othercaps, "interlaced", G_TYPE_BOOLEAN, FALSE, NULL); + if (self->mode != GST_DEINTERLACE_MODE_DISABLED) { + srccaps = gst_caps_make_writable (srccaps); + gst_caps_set_simple (srccaps, "interlaced", G_TYPE_BOOLEAN, FALSE, NULL); } gst_deinterlace_reset_history (self, FALSE); - if (!gst_pad_set_caps (otherpad, othercaps)) + if (!gst_pad_set_caps (self->srcpad, srccaps)) goto caps_not_accepted; self->frame_size = gst_video_format_get_size (self->format, self->width, self->height); if (G_LIKELY (self->fps_n != 0)) { - if (self->fields == GST_DEINTERLACE_ALL && otherpad == self->srcpad) - self->field_duration = - gst_util_uint64_scale (GST_SECOND, self->fps_d, self->fps_n); - else - self->field_duration = - gst_util_uint64_scale (GST_SECOND, self->fps_d, 2 * self->fps_n); + self->field_duration = + gst_util_uint64_scale (GST_SECOND, self->fps_d, 2 * self->fps_n); } else { self->field_duration = 0; } - if (pad == self->sinkpad) { - gst_caps_replace (&self->sink_caps, caps); - gst_caps_replace (&self->src_caps, othercaps); - } else { - gst_caps_replace (&self->src_caps, caps); - gst_caps_replace (&self->sink_caps, othercaps); - } - gst_deinterlace_set_method (self, self->method_id); gst_deinterlace_method_setup (self->method, self->format, self->width, self->height); - GST_DEBUG_OBJECT (pad, "Set caps: %" GST_PTR_FORMAT, caps); - GST_DEBUG_OBJECT (pad, "Other caps: %" GST_PTR_FORMAT, othercaps); + GST_DEBUG_OBJECT (pad, "Sink caps: %" GST_PTR_FORMAT, caps); + GST_DEBUG_OBJECT (pad, "Src caps: %" GST_PTR_FORMAT, srccaps); - gst_caps_unref (othercaps); + gst_caps_unref (srccaps); done: @@ -1474,8 +1453,8 @@ invalid_caps: caps_not_accepted: res = FALSE; - GST_ERROR_OBJECT (pad, "Caps not accepted: %" GST_PTR_FORMAT, othercaps); - gst_caps_unref (othercaps); + GST_ERROR_OBJECT (pad, "Caps not accepted: %" GST_PTR_FORMAT, srccaps); + gst_caps_unref (srccaps); goto done; } diff --git a/gst/deinterlace/gstdeinterlace.h b/gst/deinterlace/gstdeinterlace.h index 6641da2087..df996f8425 100644 --- a/gst/deinterlace/gstdeinterlace.h +++ b/gst/deinterlace/gstdeinterlace.h @@ -129,10 +129,6 @@ struct _GstDeinterlace gdouble proportion; GstClockTime earliest_time; - /* Upstream negotiation stuff */ - GstCaps *sink_caps; - GstCaps *src_caps; - GstCaps *request_caps; gboolean reconfigure; From d4eaff5e88c9b34a9e33ddaa9bb83859b2c37446 Mon Sep 17 00:00:00 2001 From: Carsten Kroll Date: Thu, 24 Feb 2011 13:56:04 +0100 Subject: [PATCH 347/351] dvdemux: First try if upstream handles TIME seeks before handling them here Fixes bug #642963. --- ext/dv/gstdvdemux.c | 31 ++++++++++++++++--------------- 1 file changed, 16 insertions(+), 15 deletions(-) diff --git a/ext/dv/gstdvdemux.c b/ext/dv/gstdvdemux.c index 5261d479f8..c1ecd9261c 100644 --- a/ext/dv/gstdvdemux.c +++ b/ext/dv/gstdvdemux.c @@ -895,17 +895,13 @@ gst_dvdemux_convert_segment (GstDVDemux * dvdemux, GstSegment * src, * * Convert the time seek to a bytes seek and send it * upstream - * - * FIXME, upstream might be able to perform time based - * seek too. - * * Does not take ownership of the event. */ static gboolean gst_dvdemux_handle_push_seek (GstDVDemux * dvdemux, GstPad * pad, GstEvent * event) { - gboolean res; + gboolean res = FALSE; gdouble rate; GstSeekFlags flags; GstFormat format; @@ -917,19 +913,24 @@ gst_dvdemux_handle_push_seek (GstDVDemux * dvdemux, GstPad * pad, gst_event_parse_seek (event, &rate, &format, &flags, &cur_type, &cur, &stop_type, &stop); - /* we convert the start/stop on the srcpad to the byte format - * on the sinkpad and forward the event */ - res = gst_dvdemux_convert_src_to_sink (dvdemux, pad, - format, cur, stop, GST_FORMAT_BYTES, &start_position, &end_position); - if (!res) - goto done; + /* First try if upstream can handle time based seeks */ + if (format == GST_FORMAT_TIME) + res = gst_pad_push_event (dvdemux->sinkpad, event); - /* now this is the updated seek event on bytes */ - newevent = gst_event_new_seek (rate, GST_FORMAT_BYTES, flags, - cur_type, start_position, stop_type, end_position); + if (!res) { + /* we convert the start/stop on the srcpad to the byte format + * on the sinkpad and forward the event */ + res = gst_dvdemux_convert_src_to_sink (dvdemux, pad, + format, cur, stop, GST_FORMAT_BYTES, &start_position, &end_position); + if (!res) + goto done; - res = gst_pad_push_event (dvdemux->sinkpad, newevent); + /* now this is the updated seek event on bytes */ + newevent = gst_event_new_seek (rate, GST_FORMAT_BYTES, flags, + cur_type, start_position, stop_type, end_position); + res = gst_pad_push_event (dvdemux->sinkpad, newevent); + } done: return res; } From ce0e34a0d5c745b87e7b07a79b471753947d18fd Mon Sep 17 00:00:00 2001 From: Carsten Kroll Date: Thu, 24 Feb 2011 14:00:37 +0100 Subject: [PATCH 348/351] deinterlace: Add support for NV12 colorspace Fixes bug #642961. --- gst/deinterlace/gstdeinterlace.c | 2 +- gst/deinterlace/gstdeinterlacemethod.c | 65 +++++++++++++++++++++++++- gst/deinterlace/gstdeinterlacemethod.h | 4 ++ gst/deinterlace/tvtime/linear.c | 1 + gst/deinterlace/tvtime/linearblend.c | 2 + gst/deinterlace/tvtime/scalerbob.c | 2 + gst/deinterlace/tvtime/vfir.c | 3 ++ gst/deinterlace/tvtime/weave.c | 2 + gst/deinterlace/tvtime/weavebff.c | 2 + gst/deinterlace/tvtime/weavetff.c | 2 + 10 files changed, 83 insertions(+), 2 deletions(-) diff --git a/gst/deinterlace/gstdeinterlace.c b/gst/deinterlace/gstdeinterlace.c index 12b1dd6315..7cc4121be5 100644 --- a/gst/deinterlace/gstdeinterlace.c +++ b/gst/deinterlace/gstdeinterlace.c @@ -161,7 +161,7 @@ gst_deinterlace_modes_get_type (void) } #define DEINTERLACE_CAPS \ - GST_VIDEO_CAPS_YUV ("{ AYUV, Y444, YUY2, YVYU, UYVY, Y42B, I420, YV12, Y41B }") ";" \ + GST_VIDEO_CAPS_YUV ("{ AYUV, Y444, YUY2, YVYU, UYVY, Y42B, I420, YV12, Y41B, NV12 }") ";" \ GST_VIDEO_CAPS_ARGB ";" GST_VIDEO_CAPS_ABGR ";" \ GST_VIDEO_CAPS_RGBA ";" GST_VIDEO_CAPS_BGRA ";" \ GST_VIDEO_CAPS_xRGB ";" GST_VIDEO_CAPS_xBGR ";" \ diff --git a/gst/deinterlace/gstdeinterlacemethod.c b/gst/deinterlace/gstdeinterlacemethod.c index a5d1058e74..7a74978f2b 100644 --- a/gst/deinterlace/gstdeinterlacemethod.c +++ b/gst/deinterlace/gstdeinterlacemethod.c @@ -69,6 +69,8 @@ gst_deinterlace_method_supported_impl (GstDeinterlaceMethodClass * klass, return (klass->deinterlace_frame_y41b != NULL); case GST_VIDEO_FORMAT_AYUV: return (klass->deinterlace_frame_ayuv != NULL); + case GST_VIDEO_FORMAT_NV12: + return (klass->deinterlace_frame_nv12 != NULL); case GST_VIDEO_FORMAT_ARGB: case GST_VIDEO_FORMAT_xRGB: return (klass->deinterlace_frame_argb != NULL); @@ -152,6 +154,9 @@ gst_deinterlace_method_setup_impl (GstDeinterlaceMethod * self, case GST_VIDEO_FORMAT_AYUV: self->deinterlace_frame = klass->deinterlace_frame_ayuv; break; + case GST_VIDEO_FORMAT_NV12: + self->deinterlace_frame = klass->deinterlace_frame_nv12; + break; case GST_VIDEO_FORMAT_ARGB: case GST_VIDEO_FORMAT_xRGB: self->deinterlace_frame = klass->deinterlace_frame_argb; @@ -268,6 +273,9 @@ gst_deinterlace_simple_method_supported (GstDeinterlaceMethodClass * mklass, case GST_VIDEO_FORMAT_AYUV: return (klass->interpolate_scanline_ayuv != NULL && klass->copy_scanline_ayuv != NULL); + case GST_VIDEO_FORMAT_NV12: + return (klass->interpolate_scanline_nv12 != NULL + && klass->copy_scanline_nv12 != NULL); case GST_VIDEO_FORMAT_I420: case GST_VIDEO_FORMAT_YV12: case GST_VIDEO_FORMAT_Y444: @@ -545,6 +553,52 @@ gst_deinterlace_simple_method_deinterlace_frame_planar (GstDeinterlaceMethod * } } +static void +gst_deinterlace_simple_method_deinterlace_frame_nv12 (GstDeinterlaceMethod * + method, const GstDeinterlaceField * history, guint history_count, + GstBuffer * outbuf) +{ + GstDeinterlaceSimpleMethod *self = GST_DEINTERLACE_SIMPLE_METHOD (method); + GstDeinterlaceMethodClass *dm_class = GST_DEINTERLACE_METHOD_GET_CLASS (self); + guint8 *out; + const guint8 *field0, *field1, *field2, *field3; + gint cur_field_idx = history_count - dm_class->fields_required; + guint cur_field_flags = history[cur_field_idx].flags; + gint i, offset; + + g_assert (self->interpolate_scanline_packed != NULL); + g_assert (self->copy_scanline_packed != NULL); + + for (i = 0; i < 2; i++) { + offset = self->parent.offset[i]; + + out = GST_BUFFER_DATA (outbuf) + offset; + + field0 = GST_BUFFER_DATA (history[cur_field_idx].buf) + offset; + + g_assert (dm_class->fields_required <= 4); + + field1 = NULL; + if (dm_class->fields_required >= 2) { + field1 = GST_BUFFER_DATA (history[cur_field_idx + 1].buf) + offset; + } + + field2 = NULL; + if (dm_class->fields_required >= 3) { + field2 = GST_BUFFER_DATA (history[cur_field_idx + 2].buf) + offset; + } + + field3 = NULL; + if (dm_class->fields_required >= 4) { + field3 = GST_BUFFER_DATA (history[cur_field_idx + 3].buf) + offset; + } + + gst_deinterlace_simple_method_deinterlace_frame_planar_plane (self, out, + field0, field1, field2, field3, cur_field_flags, i, + self->copy_scanline_packed, self->interpolate_scanline_packed); + } +} + static void gst_deinterlace_simple_method_setup (GstDeinterlaceMethod * method, GstVideoFormat format, gint width, gint height) @@ -615,6 +669,10 @@ gst_deinterlace_simple_method_setup (GstDeinterlaceMethod * method, self->interpolate_scanline_packed = klass->interpolate_scanline_bgr; self->copy_scanline_packed = klass->copy_scanline_bgr; break; + case GST_VIDEO_FORMAT_NV12: + self->interpolate_scanline_packed = klass->interpolate_scanline_nv12; + self->copy_scanline_packed = klass->copy_scanline_nv12; + break; case GST_VIDEO_FORMAT_I420: case GST_VIDEO_FORMAT_YV12: case GST_VIDEO_FORMAT_Y444: @@ -671,6 +729,8 @@ gst_deinterlace_simple_method_class_init (GstDeinterlaceSimpleMethodClass gst_deinterlace_simple_method_deinterlace_frame_planar; dm_class->deinterlace_frame_y41b = gst_deinterlace_simple_method_deinterlace_frame_planar; + dm_class->deinterlace_frame_nv12 = + gst_deinterlace_simple_method_deinterlace_frame_nv12; dm_class->fields_required = 2; dm_class->setup = gst_deinterlace_simple_method_setup; dm_class->supported = gst_deinterlace_simple_method_supported; @@ -691,6 +751,10 @@ gst_deinterlace_simple_method_class_init (GstDeinterlaceSimpleMethodClass gst_deinterlace_simple_method_interpolate_scanline_packed; klass->copy_scanline_uyvy = gst_deinterlace_simple_method_copy_scanline_packed; + klass->interpolate_scanline_nv12 = + gst_deinterlace_simple_method_interpolate_scanline_packed; + klass->copy_scanline_nv12 = + gst_deinterlace_simple_method_copy_scanline_packed; klass->interpolate_scanline_argb = gst_deinterlace_simple_method_interpolate_scanline_packed; @@ -709,7 +773,6 @@ gst_deinterlace_simple_method_class_init (GstDeinterlaceSimpleMethodClass gst_deinterlace_simple_method_interpolate_scanline_packed; klass->copy_scanline_bgra = gst_deinterlace_simple_method_copy_scanline_packed; - klass->interpolate_scanline_rgb = gst_deinterlace_simple_method_interpolate_scanline_packed; klass->copy_scanline_rgb = gst_deinterlace_simple_method_copy_scanline_packed; diff --git a/gst/deinterlace/gstdeinterlacemethod.h b/gst/deinterlace/gstdeinterlacemethod.h index d8d3343619..f8a047ed51 100644 --- a/gst/deinterlace/gstdeinterlacemethod.h +++ b/gst/deinterlace/gstdeinterlacemethod.h @@ -95,6 +95,7 @@ struct _GstDeinterlaceMethodClass { GstDeinterlaceMethodDeinterlaceFunction deinterlace_frame_y42b; GstDeinterlaceMethodDeinterlaceFunction deinterlace_frame_y41b; GstDeinterlaceMethodDeinterlaceFunction deinterlace_frame_ayuv; + GstDeinterlaceMethodDeinterlaceFunction deinterlace_frame_nv12; GstDeinterlaceMethodDeinterlaceFunction deinterlace_frame_argb; GstDeinterlaceMethodDeinterlaceFunction deinterlace_frame_abgr; GstDeinterlaceMethodDeinterlaceFunction deinterlace_frame_rgba; @@ -198,6 +199,9 @@ struct _GstDeinterlaceSimpleMethodClass { GstDeinterlaceSimpleMethodFunction copy_scanline_rgb; GstDeinterlaceSimpleMethodFunction interpolate_scanline_bgr; GstDeinterlaceSimpleMethodFunction copy_scanline_bgr; + GstDeinterlaceSimpleMethodFunction interpolate_scanline_nv12; + GstDeinterlaceSimpleMethodFunction copy_scanline_nv12; + /* Planar formats */ GstDeinterlaceSimpleMethodFunction copy_scanline_planar_y; diff --git a/gst/deinterlace/tvtime/linear.c b/gst/deinterlace/tvtime/linear.c index 97bb7e517f..ff4c2787f4 100644 --- a/gst/deinterlace/tvtime/linear.c +++ b/gst/deinterlace/tvtime/linear.c @@ -112,6 +112,7 @@ gst_deinterlace_method_linear_class_init (GstDeinterlaceMethodLinearClass * dism_class->interpolate_scanline_bgra = deinterlace_scanline_linear_packed_c; dism_class->interpolate_scanline_rgb = deinterlace_scanline_linear_packed_c; dism_class->interpolate_scanline_bgr = deinterlace_scanline_linear_packed_c; + dism_class->interpolate_scanline_nv12 = deinterlace_scanline_linear_packed_c; dism_class->interpolate_scanline_planar_y = deinterlace_scanline_linear_planar_y_c; dism_class->interpolate_scanline_planar_u = diff --git a/gst/deinterlace/tvtime/linearblend.c b/gst/deinterlace/tvtime/linearblend.c index 548ce7c9f0..1c2ccbc075 100644 --- a/gst/deinterlace/tvtime/linearblend.c +++ b/gst/deinterlace/tvtime/linearblend.c @@ -167,6 +167,8 @@ static void deinterlace_scanline_linear_blend_packed_c; dism_class->interpolate_scanline_bgr = deinterlace_scanline_linear_blend_packed_c; + dism_class->interpolate_scanline_nv12 = + deinterlace_scanline_linear_blend_packed_c; dism_class->interpolate_scanline_planar_y = deinterlace_scanline_linear_blend_planar_y_c; diff --git a/gst/deinterlace/tvtime/scalerbob.c b/gst/deinterlace/tvtime/scalerbob.c index 4612ef05dc..16f12bc002 100644 --- a/gst/deinterlace/tvtime/scalerbob.c +++ b/gst/deinterlace/tvtime/scalerbob.c @@ -90,6 +90,8 @@ gst_deinterlace_method_scaler_bob_class_init (GstDeinterlaceMethodScalerBobClass deinterlace_scanline_scaler_bob_packed; dism_class->interpolate_scanline_uyvy = deinterlace_scanline_scaler_bob_packed; + dism_class->interpolate_scanline_nv12 = + deinterlace_scanline_scaler_bob_packed; dism_class->interpolate_scanline_argb = deinterlace_scanline_scaler_bob_packed; dism_class->interpolate_scanline_abgr = diff --git a/gst/deinterlace/tvtime/vfir.c b/gst/deinterlace/tvtime/vfir.c index da32bf69fb..32c06c0bf6 100644 --- a/gst/deinterlace/tvtime/vfir.c +++ b/gst/deinterlace/tvtime/vfir.c @@ -259,6 +259,7 @@ gst_deinterlace_method_vfir_class_init (GstDeinterlaceMethodVFIRClass * klass) dism_class->interpolate_scanline_yuy2 = deinterlace_line_packed_mmx; dism_class->interpolate_scanline_yvyu = deinterlace_line_packed_mmx; dism_class->interpolate_scanline_uyvy = deinterlace_line_packed_mmx; + dism_class->interpolate_scanline_nv12 = deinterlace_line_packed_mmx; dism_class->interpolate_scanline_argb = deinterlace_line_packed_mmx; dism_class->interpolate_scanline_abgr = deinterlace_line_packed_mmx; dism_class->interpolate_scanline_rgba = deinterlace_line_packed_mmx; @@ -273,6 +274,7 @@ gst_deinterlace_method_vfir_class_init (GstDeinterlaceMethodVFIRClass * klass) dism_class->interpolate_scanline_yvyu = deinterlace_line_packed_c; dism_class->interpolate_scanline_uyvy = deinterlace_line_packed_c; dism_class->interpolate_scanline_ayuv = deinterlace_line_packed_c; + dism_class->interpolate_scanline_nv12 = deinterlace_line_packed_c; dism_class->interpolate_scanline_argb = deinterlace_line_packed_c; dism_class->interpolate_scanline_abgr = deinterlace_line_packed_c; dism_class->interpolate_scanline_rgba = deinterlace_line_packed_c; @@ -288,6 +290,7 @@ gst_deinterlace_method_vfir_class_init (GstDeinterlaceMethodVFIRClass * klass) dism_class->interpolate_scanline_yuy2 = deinterlace_line_packed_c; dism_class->interpolate_scanline_yvyu = deinterlace_line_packed_c; dism_class->interpolate_scanline_uyvy = deinterlace_line_packed_c; + dism_class->interpolate_scanline_nv12 = deinterlace_line_packed_c; dism_class->interpolate_scanline_argb = deinterlace_line_packed_c; dism_class->interpolate_scanline_abgr = deinterlace_line_packed_c; dism_class->interpolate_scanline_rgba = deinterlace_line_packed_c; diff --git a/gst/deinterlace/tvtime/weave.c b/gst/deinterlace/tvtime/weave.c index 0f9714235a..c44b624311 100644 --- a/gst/deinterlace/tvtime/weave.c +++ b/gst/deinterlace/tvtime/weave.c @@ -119,6 +119,7 @@ gst_deinterlace_method_weave_class_init (GstDeinterlaceMethodWeaveClass * klass) dism_class->interpolate_scanline_yuy2 = deinterlace_scanline_weave_packed; dism_class->interpolate_scanline_yvyu = deinterlace_scanline_weave_packed; dism_class->interpolate_scanline_uyvy = deinterlace_scanline_weave_packed; + dism_class->interpolate_scanline_nv12 = deinterlace_scanline_weave_packed; dism_class->interpolate_scanline_argb = deinterlace_scanline_weave_packed; dism_class->interpolate_scanline_abgr = deinterlace_scanline_weave_packed; dism_class->interpolate_scanline_rgba = deinterlace_scanline_weave_packed; @@ -136,6 +137,7 @@ gst_deinterlace_method_weave_class_init (GstDeinterlaceMethodWeaveClass * klass) dism_class->copy_scanline_yuy2 = copy_scanline_packed; dism_class->copy_scanline_yvyu = copy_scanline_packed; dism_class->copy_scanline_uyvy = copy_scanline_packed; + dism_class->copy_scanline_nv12 = copy_scanline_packed; dism_class->copy_scanline_argb = copy_scanline_packed; dism_class->copy_scanline_abgr = copy_scanline_packed; dism_class->copy_scanline_rgba = copy_scanline_packed; diff --git a/gst/deinterlace/tvtime/weavebff.c b/gst/deinterlace/tvtime/weavebff.c index 9c95ba816a..664d2c826f 100644 --- a/gst/deinterlace/tvtime/weavebff.c +++ b/gst/deinterlace/tvtime/weavebff.c @@ -140,6 +140,7 @@ gst_deinterlace_method_weave_bff_class_init (GstDeinterlaceMethodWeaveBFFClass * dism_class->interpolate_scanline_yuy2 = deinterlace_scanline_weave_packed; dism_class->interpolate_scanline_yvyu = deinterlace_scanline_weave_packed; dism_class->interpolate_scanline_uyvy = deinterlace_scanline_weave_packed; + dism_class->interpolate_scanline_nv12 = deinterlace_scanline_weave_packed; dism_class->interpolate_scanline_argb = deinterlace_scanline_weave_packed; dism_class->interpolate_scanline_abgr = deinterlace_scanline_weave_packed; dism_class->interpolate_scanline_rgba = deinterlace_scanline_weave_packed; @@ -157,6 +158,7 @@ gst_deinterlace_method_weave_bff_class_init (GstDeinterlaceMethodWeaveBFFClass * dism_class->copy_scanline_yuy2 = copy_scanline_packed; dism_class->copy_scanline_yvyu = copy_scanline_packed; dism_class->copy_scanline_uyvy = copy_scanline_packed; + dism_class->copy_scanline_nv12 = copy_scanline_packed; dism_class->copy_scanline_argb = copy_scanline_packed; dism_class->copy_scanline_abgr = copy_scanline_packed; dism_class->copy_scanline_rgba = copy_scanline_packed; diff --git a/gst/deinterlace/tvtime/weavetff.c b/gst/deinterlace/tvtime/weavetff.c index 2df0863856..cf0875394c 100644 --- a/gst/deinterlace/tvtime/weavetff.c +++ b/gst/deinterlace/tvtime/weavetff.c @@ -141,6 +141,7 @@ gst_deinterlace_method_weave_tff_class_init (GstDeinterlaceMethodWeaveTFFClass * dism_class->interpolate_scanline_yuy2 = deinterlace_scanline_weave_packed; dism_class->interpolate_scanline_yvyu = deinterlace_scanline_weave_packed; dism_class->interpolate_scanline_uyvy = deinterlace_scanline_weave_packed; + dism_class->interpolate_scanline_nv12 = deinterlace_scanline_weave_packed; dism_class->interpolate_scanline_argb = deinterlace_scanline_weave_packed; dism_class->interpolate_scanline_abgr = deinterlace_scanline_weave_packed; dism_class->interpolate_scanline_rgba = deinterlace_scanline_weave_packed; @@ -158,6 +159,7 @@ gst_deinterlace_method_weave_tff_class_init (GstDeinterlaceMethodWeaveTFFClass * dism_class->copy_scanline_yuy2 = copy_scanline_packed; dism_class->copy_scanline_yvyu = copy_scanline_packed; dism_class->copy_scanline_uyvy = copy_scanline_packed; + dism_class->copy_scanline_nv12 = copy_scanline_packed; dism_class->copy_scanline_argb = copy_scanline_packed; dism_class->copy_scanline_abgr = copy_scanline_packed; dism_class->copy_scanline_rgba = copy_scanline_packed; From 88cbcf5aa420e17c13ce887597fbc65c0a8018de Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Sebastian=20Dr=C3=B6ge?= Date: Thu, 24 Feb 2011 14:08:25 +0100 Subject: [PATCH 349/351] deinterlace: Add support for NV21 colorspace --- gst/deinterlace/gstdeinterlace.c | 2 +- gst/deinterlace/gstdeinterlacemethod.c | 14 ++++++++++++++ gst/deinterlace/gstdeinterlacemethod.h | 6 +++++- gst/deinterlace/tvtime/linear.c | 1 + gst/deinterlace/tvtime/linearblend.c | 2 ++ gst/deinterlace/tvtime/scalerbob.c | 2 ++ gst/deinterlace/tvtime/vfir.c | 3 +++ gst/deinterlace/tvtime/weave.c | 2 ++ gst/deinterlace/tvtime/weavebff.c | 2 ++ gst/deinterlace/tvtime/weavetff.c | 2 ++ 10 files changed, 34 insertions(+), 2 deletions(-) diff --git a/gst/deinterlace/gstdeinterlace.c b/gst/deinterlace/gstdeinterlace.c index 7cc4121be5..ce76f720c8 100644 --- a/gst/deinterlace/gstdeinterlace.c +++ b/gst/deinterlace/gstdeinterlace.c @@ -161,7 +161,7 @@ gst_deinterlace_modes_get_type (void) } #define DEINTERLACE_CAPS \ - GST_VIDEO_CAPS_YUV ("{ AYUV, Y444, YUY2, YVYU, UYVY, Y42B, I420, YV12, Y41B, NV12 }") ";" \ + GST_VIDEO_CAPS_YUV ("{ AYUV, Y444, YUY2, YVYU, UYVY, Y42B, I420, YV12, Y41B, NV12, NV21 }") ";" \ GST_VIDEO_CAPS_ARGB ";" GST_VIDEO_CAPS_ABGR ";" \ GST_VIDEO_CAPS_RGBA ";" GST_VIDEO_CAPS_BGRA ";" \ GST_VIDEO_CAPS_xRGB ";" GST_VIDEO_CAPS_xBGR ";" \ diff --git a/gst/deinterlace/gstdeinterlacemethod.c b/gst/deinterlace/gstdeinterlacemethod.c index 7a74978f2b..fe5996b52e 100644 --- a/gst/deinterlace/gstdeinterlacemethod.c +++ b/gst/deinterlace/gstdeinterlacemethod.c @@ -71,6 +71,8 @@ gst_deinterlace_method_supported_impl (GstDeinterlaceMethodClass * klass, return (klass->deinterlace_frame_ayuv != NULL); case GST_VIDEO_FORMAT_NV12: return (klass->deinterlace_frame_nv12 != NULL); + case GST_VIDEO_FORMAT_NV21: + return (klass->deinterlace_frame_nv21 != NULL); case GST_VIDEO_FORMAT_ARGB: case GST_VIDEO_FORMAT_xRGB: return (klass->deinterlace_frame_argb != NULL); @@ -157,6 +159,9 @@ gst_deinterlace_method_setup_impl (GstDeinterlaceMethod * self, case GST_VIDEO_FORMAT_NV12: self->deinterlace_frame = klass->deinterlace_frame_nv12; break; + case GST_VIDEO_FORMAT_NV21: + self->deinterlace_frame = klass->deinterlace_frame_nv21; + break; case GST_VIDEO_FORMAT_ARGB: case GST_VIDEO_FORMAT_xRGB: self->deinterlace_frame = klass->deinterlace_frame_argb; @@ -276,6 +281,9 @@ gst_deinterlace_simple_method_supported (GstDeinterlaceMethodClass * mklass, case GST_VIDEO_FORMAT_NV12: return (klass->interpolate_scanline_nv12 != NULL && klass->copy_scanline_nv12 != NULL); + case GST_VIDEO_FORMAT_NV21: + return (klass->interpolate_scanline_nv21 != NULL + && klass->copy_scanline_nv21 != NULL); case GST_VIDEO_FORMAT_I420: case GST_VIDEO_FORMAT_YV12: case GST_VIDEO_FORMAT_Y444: @@ -673,6 +681,10 @@ gst_deinterlace_simple_method_setup (GstDeinterlaceMethod * method, self->interpolate_scanline_packed = klass->interpolate_scanline_nv12; self->copy_scanline_packed = klass->copy_scanline_nv12; break; + case GST_VIDEO_FORMAT_NV21: + self->interpolate_scanline_packed = klass->interpolate_scanline_nv21; + self->copy_scanline_packed = klass->copy_scanline_nv21; + break; case GST_VIDEO_FORMAT_I420: case GST_VIDEO_FORMAT_YV12: case GST_VIDEO_FORMAT_Y444: @@ -731,6 +743,8 @@ gst_deinterlace_simple_method_class_init (GstDeinterlaceSimpleMethodClass gst_deinterlace_simple_method_deinterlace_frame_planar; dm_class->deinterlace_frame_nv12 = gst_deinterlace_simple_method_deinterlace_frame_nv12; + dm_class->deinterlace_frame_nv21 = + gst_deinterlace_simple_method_deinterlace_frame_nv12; dm_class->fields_required = 2; dm_class->setup = gst_deinterlace_simple_method_setup; dm_class->supported = gst_deinterlace_simple_method_supported; diff --git a/gst/deinterlace/gstdeinterlacemethod.h b/gst/deinterlace/gstdeinterlacemethod.h index f8a047ed51..11e2c35ac1 100644 --- a/gst/deinterlace/gstdeinterlacemethod.h +++ b/gst/deinterlace/gstdeinterlacemethod.h @@ -96,6 +96,7 @@ struct _GstDeinterlaceMethodClass { GstDeinterlaceMethodDeinterlaceFunction deinterlace_frame_y41b; GstDeinterlaceMethodDeinterlaceFunction deinterlace_frame_ayuv; GstDeinterlaceMethodDeinterlaceFunction deinterlace_frame_nv12; + GstDeinterlaceMethodDeinterlaceFunction deinterlace_frame_nv21; GstDeinterlaceMethodDeinterlaceFunction deinterlace_frame_argb; GstDeinterlaceMethodDeinterlaceFunction deinterlace_frame_abgr; GstDeinterlaceMethodDeinterlaceFunction deinterlace_frame_rgba; @@ -199,9 +200,12 @@ struct _GstDeinterlaceSimpleMethodClass { GstDeinterlaceSimpleMethodFunction copy_scanline_rgb; GstDeinterlaceSimpleMethodFunction interpolate_scanline_bgr; GstDeinterlaceSimpleMethodFunction copy_scanline_bgr; + + /* Semi-planar formats */ GstDeinterlaceSimpleMethodFunction interpolate_scanline_nv12; GstDeinterlaceSimpleMethodFunction copy_scanline_nv12; - + GstDeinterlaceSimpleMethodFunction interpolate_scanline_nv21; + GstDeinterlaceSimpleMethodFunction copy_scanline_nv21; /* Planar formats */ GstDeinterlaceSimpleMethodFunction copy_scanline_planar_y; diff --git a/gst/deinterlace/tvtime/linear.c b/gst/deinterlace/tvtime/linear.c index ff4c2787f4..05dac451de 100644 --- a/gst/deinterlace/tvtime/linear.c +++ b/gst/deinterlace/tvtime/linear.c @@ -113,6 +113,7 @@ gst_deinterlace_method_linear_class_init (GstDeinterlaceMethodLinearClass * dism_class->interpolate_scanline_rgb = deinterlace_scanline_linear_packed_c; dism_class->interpolate_scanline_bgr = deinterlace_scanline_linear_packed_c; dism_class->interpolate_scanline_nv12 = deinterlace_scanline_linear_packed_c; + dism_class->interpolate_scanline_nv21 = deinterlace_scanline_linear_packed_c; dism_class->interpolate_scanline_planar_y = deinterlace_scanline_linear_planar_y_c; dism_class->interpolate_scanline_planar_u = diff --git a/gst/deinterlace/tvtime/linearblend.c b/gst/deinterlace/tvtime/linearblend.c index 1c2ccbc075..a343d0fde2 100644 --- a/gst/deinterlace/tvtime/linearblend.c +++ b/gst/deinterlace/tvtime/linearblend.c @@ -169,6 +169,8 @@ static void deinterlace_scanline_linear_blend_packed_c; dism_class->interpolate_scanline_nv12 = deinterlace_scanline_linear_blend_packed_c; + dism_class->interpolate_scanline_nv21 = + deinterlace_scanline_linear_blend_packed_c; dism_class->interpolate_scanline_planar_y = deinterlace_scanline_linear_blend_planar_y_c; diff --git a/gst/deinterlace/tvtime/scalerbob.c b/gst/deinterlace/tvtime/scalerbob.c index 16f12bc002..bac4c62b71 100644 --- a/gst/deinterlace/tvtime/scalerbob.c +++ b/gst/deinterlace/tvtime/scalerbob.c @@ -92,6 +92,8 @@ gst_deinterlace_method_scaler_bob_class_init (GstDeinterlaceMethodScalerBobClass deinterlace_scanline_scaler_bob_packed; dism_class->interpolate_scanline_nv12 = deinterlace_scanline_scaler_bob_packed; + dism_class->interpolate_scanline_nv21 = + deinterlace_scanline_scaler_bob_packed; dism_class->interpolate_scanline_argb = deinterlace_scanline_scaler_bob_packed; dism_class->interpolate_scanline_abgr = diff --git a/gst/deinterlace/tvtime/vfir.c b/gst/deinterlace/tvtime/vfir.c index 32c06c0bf6..c589dd7444 100644 --- a/gst/deinterlace/tvtime/vfir.c +++ b/gst/deinterlace/tvtime/vfir.c @@ -260,6 +260,7 @@ gst_deinterlace_method_vfir_class_init (GstDeinterlaceMethodVFIRClass * klass) dism_class->interpolate_scanline_yvyu = deinterlace_line_packed_mmx; dism_class->interpolate_scanline_uyvy = deinterlace_line_packed_mmx; dism_class->interpolate_scanline_nv12 = deinterlace_line_packed_mmx; + dism_class->interpolate_scanline_nv21 = deinterlace_line_packed_mmx; dism_class->interpolate_scanline_argb = deinterlace_line_packed_mmx; dism_class->interpolate_scanline_abgr = deinterlace_line_packed_mmx; dism_class->interpolate_scanline_rgba = deinterlace_line_packed_mmx; @@ -275,6 +276,7 @@ gst_deinterlace_method_vfir_class_init (GstDeinterlaceMethodVFIRClass * klass) dism_class->interpolate_scanline_uyvy = deinterlace_line_packed_c; dism_class->interpolate_scanline_ayuv = deinterlace_line_packed_c; dism_class->interpolate_scanline_nv12 = deinterlace_line_packed_c; + dism_class->interpolate_scanline_nv21 = deinterlace_line_packed_c; dism_class->interpolate_scanline_argb = deinterlace_line_packed_c; dism_class->interpolate_scanline_abgr = deinterlace_line_packed_c; dism_class->interpolate_scanline_rgba = deinterlace_line_packed_c; @@ -291,6 +293,7 @@ gst_deinterlace_method_vfir_class_init (GstDeinterlaceMethodVFIRClass * klass) dism_class->interpolate_scanline_yvyu = deinterlace_line_packed_c; dism_class->interpolate_scanline_uyvy = deinterlace_line_packed_c; dism_class->interpolate_scanline_nv12 = deinterlace_line_packed_c; + dism_class->interpolate_scanline_nv21 = deinterlace_line_packed_c; dism_class->interpolate_scanline_argb = deinterlace_line_packed_c; dism_class->interpolate_scanline_abgr = deinterlace_line_packed_c; dism_class->interpolate_scanline_rgba = deinterlace_line_packed_c; diff --git a/gst/deinterlace/tvtime/weave.c b/gst/deinterlace/tvtime/weave.c index c44b624311..8d49979da2 100644 --- a/gst/deinterlace/tvtime/weave.c +++ b/gst/deinterlace/tvtime/weave.c @@ -120,6 +120,7 @@ gst_deinterlace_method_weave_class_init (GstDeinterlaceMethodWeaveClass * klass) dism_class->interpolate_scanline_yvyu = deinterlace_scanline_weave_packed; dism_class->interpolate_scanline_uyvy = deinterlace_scanline_weave_packed; dism_class->interpolate_scanline_nv12 = deinterlace_scanline_weave_packed; + dism_class->interpolate_scanline_nv21 = deinterlace_scanline_weave_packed; dism_class->interpolate_scanline_argb = deinterlace_scanline_weave_packed; dism_class->interpolate_scanline_abgr = deinterlace_scanline_weave_packed; dism_class->interpolate_scanline_rgba = deinterlace_scanline_weave_packed; @@ -138,6 +139,7 @@ gst_deinterlace_method_weave_class_init (GstDeinterlaceMethodWeaveClass * klass) dism_class->copy_scanline_yvyu = copy_scanline_packed; dism_class->copy_scanline_uyvy = copy_scanline_packed; dism_class->copy_scanline_nv12 = copy_scanline_packed; + dism_class->copy_scanline_nv21 = copy_scanline_packed; dism_class->copy_scanline_argb = copy_scanline_packed; dism_class->copy_scanline_abgr = copy_scanline_packed; dism_class->copy_scanline_rgba = copy_scanline_packed; diff --git a/gst/deinterlace/tvtime/weavebff.c b/gst/deinterlace/tvtime/weavebff.c index 664d2c826f..b96da1f417 100644 --- a/gst/deinterlace/tvtime/weavebff.c +++ b/gst/deinterlace/tvtime/weavebff.c @@ -141,6 +141,7 @@ gst_deinterlace_method_weave_bff_class_init (GstDeinterlaceMethodWeaveBFFClass * dism_class->interpolate_scanline_yvyu = deinterlace_scanline_weave_packed; dism_class->interpolate_scanline_uyvy = deinterlace_scanline_weave_packed; dism_class->interpolate_scanline_nv12 = deinterlace_scanline_weave_packed; + dism_class->interpolate_scanline_nv21 = deinterlace_scanline_weave_packed; dism_class->interpolate_scanline_argb = deinterlace_scanline_weave_packed; dism_class->interpolate_scanline_abgr = deinterlace_scanline_weave_packed; dism_class->interpolate_scanline_rgba = deinterlace_scanline_weave_packed; @@ -159,6 +160,7 @@ gst_deinterlace_method_weave_bff_class_init (GstDeinterlaceMethodWeaveBFFClass * dism_class->copy_scanline_yvyu = copy_scanline_packed; dism_class->copy_scanline_uyvy = copy_scanline_packed; dism_class->copy_scanline_nv12 = copy_scanline_packed; + dism_class->copy_scanline_nv21 = copy_scanline_packed; dism_class->copy_scanline_argb = copy_scanline_packed; dism_class->copy_scanline_abgr = copy_scanline_packed; dism_class->copy_scanline_rgba = copy_scanline_packed; diff --git a/gst/deinterlace/tvtime/weavetff.c b/gst/deinterlace/tvtime/weavetff.c index cf0875394c..b645b0b905 100644 --- a/gst/deinterlace/tvtime/weavetff.c +++ b/gst/deinterlace/tvtime/weavetff.c @@ -142,6 +142,7 @@ gst_deinterlace_method_weave_tff_class_init (GstDeinterlaceMethodWeaveTFFClass * dism_class->interpolate_scanline_yvyu = deinterlace_scanline_weave_packed; dism_class->interpolate_scanline_uyvy = deinterlace_scanline_weave_packed; dism_class->interpolate_scanline_nv12 = deinterlace_scanline_weave_packed; + dism_class->interpolate_scanline_nv21 = deinterlace_scanline_weave_packed; dism_class->interpolate_scanline_argb = deinterlace_scanline_weave_packed; dism_class->interpolate_scanline_abgr = deinterlace_scanline_weave_packed; dism_class->interpolate_scanline_rgba = deinterlace_scanline_weave_packed; @@ -160,6 +161,7 @@ gst_deinterlace_method_weave_tff_class_init (GstDeinterlaceMethodWeaveTFFClass * dism_class->copy_scanline_yvyu = copy_scanline_packed; dism_class->copy_scanline_uyvy = copy_scanline_packed; dism_class->copy_scanline_nv12 = copy_scanline_packed; + dism_class->copy_scanline_nv21 = copy_scanline_packed; dism_class->copy_scanline_argb = copy_scanline_packed; dism_class->copy_scanline_abgr = copy_scanline_packed; dism_class->copy_scanline_rgba = copy_scanline_packed; From 6565dd5fbc502877e315b27c25a39e43eb54847e Mon Sep 17 00:00:00 2001 From: David Schleef Date: Fri, 25 Feb 2011 19:35:51 -0800 Subject: [PATCH 350/351] qtdemux: Add support for 2Vuy and r210 --- gst/qtdemux/qtdemux.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/gst/qtdemux/qtdemux.c b/gst/qtdemux/qtdemux.c index 2003a3b77b..f48a261c25 100644 --- a/gst/qtdemux/qtdemux.c +++ b/gst/qtdemux/qtdemux.c @@ -8923,6 +8923,7 @@ qtdemux_video_caps (GstQTDemux * qtdemux, QtDemuxStream * stream, NULL); break; case GST_MAKE_FOURCC ('2', 'v', 'u', 'y'): + case GST_MAKE_FOURCC ('2', 'V', 'u', 'y'): _codec ("Raw packed YUV 4:2:2"); caps = gst_caps_new_simple ("video/x-raw-yuv", "format", GST_TYPE_FOURCC, GST_MAKE_FOURCC ('U', 'Y', 'V', 'Y'), @@ -8934,6 +8935,17 @@ qtdemux_video_caps (GstQTDemux * qtdemux, QtDemuxStream * stream, "format", GST_TYPE_FOURCC, GST_MAKE_FOURCC ('v', '2', '1', '0'), NULL); break; + case GST_MAKE_FOURCC ('r', '2', '1', '0'): + _codec ("Raw packed RGB 10-bit 4:4:4"); + caps = gst_caps_new_simple ("video/x-raw-rgb", + "endianness", G_TYPE_INT, G_BIG_ENDIAN, "depth", G_TYPE_INT, 30, + "bpp", G_TYPE_INT, 32, + "endianness", G_TYPE_INT, G_BIG_ENDIAN, + "red_mask", G_TYPE_INT, 0x3ff00000, + "green_mask", G_TYPE_INT, 0x000ffc00, + "blue_mask", G_TYPE_INT, 0x000003ff, NULL); + GST_ERROR ("caps %P", caps); + break; case GST_MAKE_FOURCC ('m', 'p', 'e', 'g'): case GST_MAKE_FOURCC ('m', 'p', 'g', '1'): _codec ("MPEG-1 video"); From caa971e605a61e2d31a3927dcf22149125e580ad Mon Sep 17 00:00:00 2001 From: David Schleef Date: Fri, 25 Feb 2011 20:12:35 -0800 Subject: [PATCH 351/351] qtdemux: remove accidental debug message in previous commit --- gst/qtdemux/qtdemux.c | 1 - 1 file changed, 1 deletion(-) diff --git a/gst/qtdemux/qtdemux.c b/gst/qtdemux/qtdemux.c index f48a261c25..22803745d5 100644 --- a/gst/qtdemux/qtdemux.c +++ b/gst/qtdemux/qtdemux.c @@ -8944,7 +8944,6 @@ qtdemux_video_caps (GstQTDemux * qtdemux, QtDemuxStream * stream, "red_mask", G_TYPE_INT, 0x3ff00000, "green_mask", G_TYPE_INT, 0x000ffc00, "blue_mask", G_TYPE_INT, 0x000003ff, NULL); - GST_ERROR ("caps %P", caps); break; case GST_MAKE_FOURCC ('m', 'p', 'e', 'g'): case GST_MAKE_FOURCC ('m', 'p', 'g', '1'):