From 0d5b5d839a3fb92620bdc90579eab9192a74df3a Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Tue, 21 Feb 2012 18:42:31 +0100 Subject: [PATCH 01/55] mpegaudioparse: support parsing freeform bitrate stream --- gst/audioparsers/gstmpegaudioparse.c | 168 ++++++++++++++++++++++++--- gst/audioparsers/gstmpegaudioparse.h | 2 + 2 files changed, 156 insertions(+), 14 deletions(-) diff --git a/gst/audioparsers/gstmpegaudioparse.c b/gst/audioparsers/gstmpegaudioparse.c index efb97587ad..a8c68252c6 100644 --- a/gst/audioparsers/gstmpegaudioparse.c +++ b/gst/audioparsers/gstmpegaudioparse.c @@ -200,6 +200,7 @@ gst_mpeg_audio_parse_reset (GstMpegAudioParse * mp3parse) mp3parse->sent_codec_tag = FALSE; mp3parse->last_posted_crc = CRC_UNKNOWN; mp3parse->last_posted_channel_mode = MPEG_AUDIO_CHANNEL_MODE_UNKNOWN; + mp3parse->freerate = 0; mp3parse->hdr_bitrate = 0; @@ -307,14 +308,16 @@ mp3_type_frame_length_from_header (GstMpegAudioParse * mp3parse, guint32 header, bitrate = (header >> 12) & 0xF; bitrate = mp3types_bitrates[lsf][layer - 1][bitrate] * 1000; - /* The caller has ensured we have a valid header, so bitrate can't be - zero here. */ - g_assert (bitrate != 0); + if (!bitrate) { + GST_LOG_OBJECT (mp3parse, "using freeform bitrate"); + bitrate = mp3parse->freerate; + } samplerate = (header >> 10) & 0x3; samplerate = mp3types_freqs[lsf + mpg25][samplerate]; - padding = (header >> 9) & 0x1; + /* force 0 length if 0 bitrate */ + padding = (bitrate > 0) ? (header >> 9) & 0x1 : 0; mode = (header >> 6) & 0x3; channels = (mode == 3) ? 1 : 2; @@ -419,8 +422,7 @@ gst_mp3parse_validate_extended (GstMpegAudioParse * mp3parse, GstBuffer * buf, (guint) next_header & HDRMASK, bpf); *valid = FALSE; return TRUE; - } else if ((((next_header >> 12) & 0xf) == 0) || - (((next_header >> 12) & 0xf) == 0xf)) { + } else if (((next_header >> 12) & 0xf) == 0xf) { /* The essential parts were the same, but the bitrate held an invalid value - also reject */ GST_DEBUG_OBJECT (mp3parse, "next header invalid (bitrate)"); @@ -431,6 +433,13 @@ gst_mp3parse_validate_extended (GstMpegAudioParse * mp3parse, GstBuffer * buf, bpf = mp3_type_frame_length_from_header (mp3parse, next_header, NULL, NULL, NULL, NULL, NULL, NULL, NULL); + /* if no bitrate, and no freeform rate known, then fail */ + if (G_UNLIKELY (!bpf)) { + GST_DEBUG_OBJECT (mp3parse, "next header invalid (bitrate 0)"); + *valid = FALSE; + return TRUE; + } + offset += bpf; frames_found++; } @@ -461,11 +470,6 @@ gst_mpeg_audio_parse_head_check (GstMpegAudioParse * mp3parse, return FALSE; } /* if it's an invalid bitrate */ - if (((head >> 12) & 0xf) == 0x0) { - GST_WARNING_OBJECT (mp3parse, "invalid bitrate: 0x%lx." - "Free format files are not supported yet", (head >> 12) & 0xf); - return FALSE; - } if (((head >> 12) & 0xf) == 0xf) { GST_WARNING_OBJECT (mp3parse, "invalid bitrate: 0x%lx", (head >> 12) & 0xf); return FALSE; @@ -486,6 +490,115 @@ gst_mpeg_audio_parse_head_check (GstMpegAudioParse * mp3parse, return TRUE; } +/* Determines possible freeform frame rate/size by looking for next + * header with valid bitrate (0 or otherwise valid) (and sufficiently + * matching current header). + * + * Returns TRUE if we've found such one, and *rate then contains rate + * (or *rate contains 0 if decided no freeframe size could be determined). + * If not enough data, returns FALSE. + */ +static gboolean +gst_mp3parse_find_freerate (GstMpegAudioParse * mp3parse, GstBuffer * buf, + guint32 header, gboolean at_eos, gint * _rate) +{ + guint32 next_header; + const guint8 *data; + guint available; + int offset = 4; + gulong samplerate, rate, layer, padding; + gboolean valid; + gint lsf, mpg25; + + available = GST_BUFFER_SIZE (buf); + data = GST_BUFFER_DATA (buf); + + *_rate = 0; + + /* pick apart header again partially */ + if (header & (1 << 20)) { + lsf = (header & (1 << 19)) ? 0 : 1; + mpg25 = 0; + } else { + lsf = 1; + mpg25 = 1; + } + layer = 4 - ((header >> 17) & 0x3); + samplerate = (header >> 10) & 0x3; + samplerate = mp3types_freqs[lsf + mpg25][samplerate]; + padding = (header >> 9) & 0x1; + + for (; offset < available; ++offset) { + /* Check if we have enough data for all these frames, plus the next + frame header. */ + if (available < offset + 4) { + if (at_eos) { + /* Running out of data; failed to determine size */ + return TRUE; + } else { + return FALSE; + } + } + + valid = FALSE; + next_header = GST_READ_UINT32_BE (data + offset); + if ((next_header & 0xFFE00000) != 0xFFE00000) + goto next; + + GST_DEBUG_OBJECT (mp3parse, "At %d: header=%08X, header2=%08X", + offset, (unsigned int) header, (unsigned int) next_header); + + if ((next_header & HDRMASK) != (header & HDRMASK)) { + /* If any of the unmasked bits don't match, then it's not valid */ + GST_DEBUG_OBJECT (mp3parse, "next header doesn't match " + "(header=%08X (%08X), header2=%08X (%08X))", + (guint) header, (guint) header & HDRMASK, (guint) next_header, + (guint) next_header & HDRMASK); + goto next; + } else if (((next_header >> 12) & 0xf) == 0xf) { + /* The essential parts were the same, but the bitrate held an + invalid value - also reject */ + GST_DEBUG_OBJECT (mp3parse, "next header invalid (bitrate)"); + goto next; + } + + valid = TRUE; + + next: + /* almost accept as free frame */ + if (layer == 1) { + rate = samplerate * (offset - 4 * padding + 4) / 48000; + } else { + rate = samplerate * (offset - padding + 1) / (144 >> lsf) / 1000; + } + + if (valid) { + GST_LOG_OBJECT (mp3parse, "calculated rate %d", rate * 1000); + if (rate < 8 || (layer == 3 && rate > 640)) { + GST_DEBUG_OBJECT (mp3parse, "rate invalid"); + if (rate < 8) { + /* maybe some hope */ + continue; + } else { + GST_DEBUG_OBJECT (mp3parse, "aborting"); + /* give up */ + break; + } + } + *_rate = rate * 1000; + break; + } else { + /* avoid indefinite searching */ + if (rate > 1000) { + GST_DEBUG_OBJECT (mp3parse, "exceeded sanity rate; aborting"); + break; + } + } + } + + return TRUE; +} + static gboolean gst_mpeg_audio_parse_check_valid_frame (GstBaseParse * parse, GstBaseParseFrame * frame, guint * framesize, gint * skipsize) @@ -527,9 +640,14 @@ gst_mpeg_audio_parse_check_valid_frame (GstBaseParse * parse, GST_LOG_OBJECT (parse, "got frame"); + lost_sync = GST_BASE_PARSE_LOST_SYNC (parse); + draining = GST_BASE_PARSE_DRAINING (parse); + + if (G_UNLIKELY (lost_sync)) + mp3parse->freerate = 0; + bpf = mp3_type_frame_length_from_header (mp3parse, header, &version, &layer, &channels, &bitrate, &rate, &mode, &crc); - g_assert (bpf != 0); if (channels != mp3parse->channels || rate != mp3parse->rate || layer != mp3parse->layer || version != mp3parse->version) @@ -537,8 +655,30 @@ gst_mpeg_audio_parse_check_valid_frame (GstBaseParse * parse, else caps_change = FALSE; - lost_sync = GST_BASE_PARSE_LOST_SYNC (parse); - draining = GST_BASE_PARSE_DRAINING (parse); + /* maybe free format */ + if (bpf == 0) { + GST_LOG_OBJECT (mp3parse, "possibly free format"); + if (lost_sync || mp3parse->freerate == 0) { + GST_DEBUG_OBJECT (mp3parse, "finding free format rate"); + if (!gst_mp3parse_find_freerate (mp3parse, buf, header, draining, &valid)) { + /* not enough data */ + *framesize = G_MAXUINT; + *skipsize = 0; + return FALSE; + } else { + GST_DEBUG_OBJECT (parse, "determined freeform size %d", valid); + mp3parse->freerate = valid; + } + } + /* try again */ + bpf = mp3_type_frame_length_from_header (mp3parse, header, + &version, &layer, &channels, &bitrate, &rate, &mode, &crc); + if (!bpf) { + /* did not come up with valid freeform length, reject after all */ + *skipsize = 1; + return FALSE; + } + } if (!draining && (lost_sync || caps_change)) { if (!gst_mp3parse_validate_extended (mp3parse, buf, header, bpf, draining, diff --git a/gst/audioparsers/gstmpegaudioparse.h b/gst/audioparsers/gstmpegaudioparse.h index 7580001309..6b42673552 100644 --- a/gst/audioparsers/gstmpegaudioparse.h +++ b/gst/audioparsers/gstmpegaudioparse.h @@ -60,6 +60,8 @@ struct _GstMpegAudioParse { /* samples per frame */ gint spf; + gint freerate; + gboolean sent_codec_tag; guint last_posted_bitrate; gint last_posted_crc, last_crc; From a32e030faadeb538b05ecc7920d74db9efedf4d4 Mon Sep 17 00:00:00 2001 From: Vincent Untz Date: Tue, 21 Feb 2012 17:57:44 +0100 Subject: [PATCH 02/55] pulse: Fix a build warning when compiling with asserts disabled Return a value even if the code will never be reached, to make compilers happy. https://bugzilla.gnome.org/show_bug.cgi?id=670561 --- ext/pulse/pulseaudiosink.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/ext/pulse/pulseaudiosink.c b/ext/pulse/pulseaudiosink.c index cd9f7d914b..22b439b471 100644 --- a/ext/pulse/pulseaudiosink.c +++ b/ext/pulse/pulseaudiosink.c @@ -279,7 +279,9 @@ param_spec_copy (GParamSpec * spec) g_warning ("Unknown param type %ld for '%s'", (long) G_PARAM_SPEC_TYPE (spec), name); - g_assert_not_reached (); + + /* Make compiler happy */ + return NULL; } static void From 6e279032225f10670a8c7c031c28d86f2b7bbb0c Mon Sep 17 00:00:00 2001 From: David Schleef Date: Tue, 21 Feb 2012 17:16:32 -0800 Subject: [PATCH 03/55] autogen: avoid touching .po files during 'make' A simple workaround to deal with GNU gettext automake integration failing to deal with git. Fixes: #669207 --- autogen.sh | 3 +++ 1 file changed, 3 insertions(+) diff --git a/autogen.sh b/autogen.sh index 1fcf9f2827..645b2aeb38 100755 --- a/autogen.sh +++ b/autogen.sh @@ -29,6 +29,9 @@ then ln -s ../../common/hooks/pre-commit.hook .git/hooks/pre-commit fi +# GNU gettext automake support doesn't get along with git. +# https://bugzilla.gnome.org/show_bug.cgi?id=661128 +touch -t 200001010000 po/gst-plugins-good-0.10.pot CONFIGURE_DEF_OPT='--enable-maintainer-mode --enable-gtk-doc' From 38516ad367128d83f9e156529018adb4433cd328 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Wed, 22 Feb 2012 18:02:27 +0100 Subject: [PATCH 04/55] flacenc: remove post-port bogus _unref --- ext/flac/gstflacenc.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/ext/flac/gstflacenc.c b/ext/flac/gstflacenc.c index 6a7e1c0442..c2eb333cb2 100644 --- a/ext/flac/gstflacenc.c +++ b/ext/flac/gstflacenc.c @@ -814,7 +814,6 @@ gst_flac_enc_set_format (GstAudioEncoder * enc, GstAudioInfo * info) encoder_already_initialized: { g_warning ("flac already initialized -- fixme allow this"); - gst_object_unref (flacenc); return FALSE; } setting_src_caps_failed: @@ -822,14 +821,12 @@ setting_src_caps_failed: GST_DEBUG_OBJECT (flacenc, "Couldn't set caps on source pad: %" GST_PTR_FORMAT, caps); gst_caps_unref (caps); - gst_object_unref (flacenc); return FALSE; } failed_to_initialize: { GST_ELEMENT_ERROR (flacenc, LIBRARY, INIT, (NULL), ("could not initialize encoder (wrong parameters?)")); - gst_object_unref (flacenc); return FALSE; } } From 70ef85a1a24862d14b4645dde7bd5bfab346ba25 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Thu, 23 Feb 2012 11:07:35 +0000 Subject: [PATCH 05/55] Bump version after release --- configure.ac | 6 +- docs/plugins/gst-plugins-good-plugins.args | 68 +++++++++++++++++-- .../gst-plugins-good-plugins.hierarchy | 3 +- .../gst-plugins-good-plugins.interfaces | 1 + docs/plugins/inspect/plugin-1394.xml | 2 +- docs/plugins/inspect/plugin-aasink.xml | 2 +- docs/plugins/inspect/plugin-alaw.xml | 2 +- docs/plugins/inspect/plugin-alpha.xml | 2 +- docs/plugins/inspect/plugin-alphacolor.xml | 2 +- docs/plugins/inspect/plugin-annodex.xml | 2 +- docs/plugins/inspect/plugin-apetag.xml | 2 +- docs/plugins/inspect/plugin-audiofx.xml | 2 +- docs/plugins/inspect/plugin-audioparsers.xml | 4 +- docs/plugins/inspect/plugin-auparse.xml | 2 +- docs/plugins/inspect/plugin-autodetect.xml | 2 +- docs/plugins/inspect/plugin-avi.xml | 4 +- docs/plugins/inspect/plugin-cacasink.xml | 2 +- docs/plugins/inspect/plugin-cairo.xml | 2 +- docs/plugins/inspect/plugin-cutter.xml | 2 +- docs/plugins/inspect/plugin-debug.xml | 2 +- docs/plugins/inspect/plugin-deinterlace.xml | 2 +- docs/plugins/inspect/plugin-dv.xml | 2 +- docs/plugins/inspect/plugin-efence.xml | 2 +- docs/plugins/inspect/plugin-effectv.xml | 2 +- docs/plugins/inspect/plugin-equalizer.xml | 2 +- docs/plugins/inspect/plugin-esdsink.xml | 2 +- docs/plugins/inspect/plugin-flac.xml | 2 +- docs/plugins/inspect/plugin-flv.xml | 6 +- docs/plugins/inspect/plugin-flxdec.xml | 2 +- docs/plugins/inspect/plugin-gconfelements.xml | 2 +- docs/plugins/inspect/plugin-gdkpixbuf.xml | 2 +- docs/plugins/inspect/plugin-goom.xml | 2 +- docs/plugins/inspect/plugin-goom2k1.xml | 2 +- docs/plugins/inspect/plugin-gstrtpmanager.xml | 2 +- docs/plugins/inspect/plugin-halelements.xml | 2 +- docs/plugins/inspect/plugin-icydemux.xml | 2 +- docs/plugins/inspect/plugin-id3demux.xml | 2 +- docs/plugins/inspect/plugin-imagefreeze.xml | 2 +- docs/plugins/inspect/plugin-interleave.xml | 2 +- docs/plugins/inspect/plugin-isomp4.xml | 2 +- docs/plugins/inspect/plugin-jack.xml | 2 +- docs/plugins/inspect/plugin-jpeg.xml | 2 +- docs/plugins/inspect/plugin-level.xml | 2 +- docs/plugins/inspect/plugin-matroska.xml | 6 +- docs/plugins/inspect/plugin-mulaw.xml | 2 +- docs/plugins/inspect/plugin-multifile.xml | 2 +- docs/plugins/inspect/plugin-multipart.xml | 2 +- .../plugins/inspect/plugin-navigationtest.xml | 2 +- docs/plugins/inspect/plugin-oss4.xml | 2 +- docs/plugins/inspect/plugin-ossaudio.xml | 2 +- docs/plugins/inspect/plugin-png.xml | 2 +- docs/plugins/inspect/plugin-pulseaudio.xml | 2 +- docs/plugins/inspect/plugin-replaygain.xml | 2 +- docs/plugins/inspect/plugin-rtp.xml | 6 +- docs/plugins/inspect/plugin-rtsp.xml | 2 +- docs/plugins/inspect/plugin-shapewipe.xml | 2 +- docs/plugins/inspect/plugin-shout2send.xml | 2 +- docs/plugins/inspect/plugin-smpte.xml | 2 +- docs/plugins/inspect/plugin-soup.xml | 2 +- docs/plugins/inspect/plugin-spectrum.xml | 2 +- docs/plugins/inspect/plugin-speex.xml | 2 +- docs/plugins/inspect/plugin-taglib.xml | 2 +- docs/plugins/inspect/plugin-udp.xml | 2 +- docs/plugins/inspect/plugin-video4linux2.xml | 2 +- docs/plugins/inspect/plugin-videobox.xml | 2 +- docs/plugins/inspect/plugin-videocrop.xml | 2 +- docs/plugins/inspect/plugin-videofilter.xml | 2 +- docs/plugins/inspect/plugin-videomixer.xml | 2 +- docs/plugins/inspect/plugin-wavenc.xml | 2 +- docs/plugins/inspect/plugin-wavpack.xml | 2 +- docs/plugins/inspect/plugin-wavparse.xml | 2 +- docs/plugins/inspect/plugin-ximagesrc.xml | 2 +- docs/plugins/inspect/plugin-y4menc.xml | 2 +- win32/common/config.h | 31 +++------ 74 files changed, 157 insertions(+), 106 deletions(-) diff --git a/configure.ac b/configure.ac index 981094e5c5..77a8bc4b57 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, 0.10.30.1, +AC_INIT(GStreamer Good Plug-ins, 0.10.31.1, http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer, gst-plugins-good) @@ -54,8 +54,8 @@ AC_LIBTOOL_WIN32_DLL AM_PROG_LIBTOOL dnl *** required versions of GStreamer stuff *** -GST_REQ=0.10.35.1 -GSTPB_REQ=0.10.35.1 +GST_REQ=0.10.36 +GSTPB_REQ=0.10.36 dnl *** autotools stuff **** diff --git a/docs/plugins/gst-plugins-good-plugins.args b/docs/plugins/gst-plugins-good-plugins.args index 95f2db51b5..593b00bc2f 100644 --- a/docs/plugins/gst-plugins-good-plugins.args +++ b/docs/plugins/gst-plugins-good-plugins.args @@ -188,6 +188,16 @@ 0 + +GstMultiFileSink::max-file-size +guint64 + +rw +Maximum File Size +Maximum file size before starting a new file in max-size mode. +2147483648 + + GstMultiFileSrc::caps GstCaps* @@ -2484,7 +2494,7 @@ r Bytes served -Total number of bytes send to all clients. +Total number of bytes sent to all clients. 0 @@ -19935,7 +19945,7 @@ rw Client The PulseAudio client name to use. -"" +"lt-gst-plugins-good-plugins-scan" @@ -19995,7 +20005,7 @@ rw Client The PulseAudio client_name_to_use. -"" +"lt-gst-plugins-good-plugins-scan" @@ -21368,6 +21378,26 @@ NULL + +GstJackAudioSrc::client-name +gchar* + +rw +Client name +The client name of the Jack instance (NULL = default). +NULL + + + +GstJackAudioSrc::transport +GstJackTransport + +rw +Transport mode +Jack transport behaviour of the client. +No transport support + + GstJackAudioSink::client JackClient* @@ -21398,6 +21428,26 @@ NULL + +GstJackAudioSink::client-name +gchar* + +rw +Client name +The client name of the Jack instance (NULL = default). +NULL + + + +GstJackAudioSink::transport +GstJackTransport + +rw +Transport mode +Jack transport behaviour of the client. +No transport support + + GstFlacParse::check-frame-checksums gboolean @@ -22085,7 +22135,7 @@ rw Client The PulseAudio client name to use. -"" +"lt-gst-plugins-good-plugins-scan" @@ -22388,3 +22438,13 @@ "" + +GstWavParse::ignore-length +gboolean + +rw +Ignore length +Ignore length from the Wave header. +FALSE + + diff --git a/docs/plugins/gst-plugins-good-plugins.hierarchy b/docs/plugins/gst-plugins-good-plugins.hierarchy index 9f4fabee7d..350d02d420 100644 --- a/docs/plugins/gst-plugins-good-plugins.hierarchy +++ b/docs/plugins/gst-plugins-good-plugins.hierarchy @@ -244,7 +244,6 @@ GObject GstMatroskaMux GstWebMMux GstMatroskaParse - GstMonoscope GstMuLawDec GstMuLawEnc GstMultipartDemux @@ -270,7 +269,7 @@ GObject GstTagDemux GstApeDemux GstID3Demux - GstTagLibMux + GstTagMux GstApev2Mux GstId3v2Mux GstV4l2Radio diff --git a/docs/plugins/gst-plugins-good-plugins.interfaces b/docs/plugins/gst-plugins-good-plugins.interfaces index b356c99aa9..7345113c13 100644 --- a/docs/plugins/gst-plugins-good-plugins.interfaces +++ b/docs/plugins/gst-plugins-good-plugins.interfaces @@ -56,6 +56,7 @@ GstSpeexEnc GstTagSetter GstPreset GstSwitchSink GstChildProxy GstSwitchSrc GstChildProxy GstTagLibMux GstTagSetter +GstTagMux GstTagSetter GstUDPSink GstURIHandler GstUDPSrc GstURIHandler GstV4l2Radio GstURIHandler GstImplementsInterface GstTuner GstPropertyProbe diff --git a/docs/plugins/inspect/plugin-1394.xml b/docs/plugins/inspect/plugin-1394.xml index 56e1fec639..9a300f444c 100644 --- a/docs/plugins/inspect/plugin-1394.xml +++ b/docs/plugins/inspect/plugin-1394.xml @@ -3,7 +3,7 @@ Source for video data via IEEE1394 interface ../../ext/raw1394/.libs/libgst1394.so libgst1394.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-aasink.xml b/docs/plugins/inspect/plugin-aasink.xml index a8a18e2388..b15835ac95 100644 --- a/docs/plugins/inspect/plugin-aasink.xml +++ b/docs/plugins/inspect/plugin-aasink.xml @@ -3,7 +3,7 @@ ASCII Art video sink ../../ext/aalib/.libs/libgstaasink.so libgstaasink.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-alaw.xml b/docs/plugins/inspect/plugin-alaw.xml index 4f13d5bed4..cd4c42bc43 100644 --- a/docs/plugins/inspect/plugin-alaw.xml +++ b/docs/plugins/inspect/plugin-alaw.xml @@ -3,7 +3,7 @@ ALaw audio conversion routines ../../gst/law/.libs/libgstalaw.so libgstalaw.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-alpha.xml b/docs/plugins/inspect/plugin-alpha.xml index b935b0fd46..2b0b75f244 100644 --- a/docs/plugins/inspect/plugin-alpha.xml +++ b/docs/plugins/inspect/plugin-alpha.xml @@ -3,7 +3,7 @@ adds an alpha channel to video - constant or via chroma-keying ../../gst/alpha/.libs/libgstalpha.so libgstalpha.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-alphacolor.xml b/docs/plugins/inspect/plugin-alphacolor.xml index 84de6c7756..7009cb9b73 100644 --- a/docs/plugins/inspect/plugin-alphacolor.xml +++ b/docs/plugins/inspect/plugin-alphacolor.xml @@ -3,7 +3,7 @@ RGBA from/to AYUV colorspace conversion preserving the alpha channel ../../gst/alpha/.libs/libgstalphacolor.so libgstalphacolor.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-annodex.xml b/docs/plugins/inspect/plugin-annodex.xml index 9c086dbd0a..00ba705707 100644 --- a/docs/plugins/inspect/plugin-annodex.xml +++ b/docs/plugins/inspect/plugin-annodex.xml @@ -3,7 +3,7 @@ annodex stream manipulation (info about annodex: http://www.annodex.net) ../../ext/annodex/.libs/libgstannodex.so libgstannodex.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-apetag.xml b/docs/plugins/inspect/plugin-apetag.xml index 1be46d47bb..e5a5612fca 100644 --- a/docs/plugins/inspect/plugin-apetag.xml +++ b/docs/plugins/inspect/plugin-apetag.xml @@ -3,7 +3,7 @@ APEv1/2 tag reader ../../gst/apetag/.libs/libgstapetag.so libgstapetag.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-audiofx.xml b/docs/plugins/inspect/plugin-audiofx.xml index 577ae54a49..1ededdaf58 100644 --- a/docs/plugins/inspect/plugin-audiofx.xml +++ b/docs/plugins/inspect/plugin-audiofx.xml @@ -3,7 +3,7 @@ Audio effects plugin ../../gst/audiofx/.libs/libgstaudiofx.so libgstaudiofx.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-audioparsers.xml b/docs/plugins/inspect/plugin-audioparsers.xml index 09406505e2..c5735a9bf6 100644 --- a/docs/plugins/inspect/plugin-audioparsers.xml +++ b/docs/plugins/inspect/plugin-audioparsers.xml @@ -3,7 +3,7 @@ Parsers for various audio formats ../../gst/audioparsers/.libs/libgstaudioparsers.so libgstaudioparsers.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git @@ -26,7 +26,7 @@ src source always -
audio/mpeg, framed=(boolean)true, mpegversion=(int){ 2, 4 }, stream-format=(string){ raw, adts, adif }
+
audio/mpeg, framed=(boolean)true, mpegversion=(int){ 2, 4 }, stream-format=(string){ raw, adts, adif, loas }
diff --git a/docs/plugins/inspect/plugin-auparse.xml b/docs/plugins/inspect/plugin-auparse.xml index 52797c70bf..b36ed3eded 100644 --- a/docs/plugins/inspect/plugin-auparse.xml +++ b/docs/plugins/inspect/plugin-auparse.xml @@ -3,7 +3,7 @@ parses au streams ../../gst/auparse/.libs/libgstauparse.so libgstauparse.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-autodetect.xml b/docs/plugins/inspect/plugin-autodetect.xml index c91b864c2f..ed769cf6e5 100644 --- a/docs/plugins/inspect/plugin-autodetect.xml +++ b/docs/plugins/inspect/plugin-autodetect.xml @@ -3,7 +3,7 @@ Plugin contains auto-detection plugins for video/audio in- and outputs ../../gst/autodetect/.libs/libgstautodetect.so libgstautodetect.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-avi.xml b/docs/plugins/inspect/plugin-avi.xml index 0ec9e2e658..49af23ace3 100644 --- a/docs/plugins/inspect/plugin-avi.xml +++ b/docs/plugins/inspect/plugin-avi.xml @@ -3,7 +3,7 @@ AVI stream handling ../../gst/avi/.libs/libgstavi.so libgstavi.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git @@ -38,7 +38,7 @@ video_%02d source sometimes -
video/x-3ivx, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-asus, asusversion=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-asus, asusversion=(int)2, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-cirrus-logic-accupak, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-camstudio, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-compressed-yuv, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-raw-rgb, bpp=(int){ 8, 24, 32 }, depth=(int){ 8, 24 }, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-divx, divxversion=(int)3, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-divx, divxversion=(int)4, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-truemotion, trueversion=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-dv, systemstream=(boolean)false, dvversion=(int)25, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-dv, systemstream=(boolean)false, dvversion=(int)50, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-divx, divxversion=(int)5, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/mpeg, mpegversion=(int)4, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-flash-video, flvversion=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-vp6-flash, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-h263, variant=(string)itu, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-h263, variant=(string)lucent, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-h264, variant=(string)itu, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-huffyuv, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-intel-h263, variant=(string)intel, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-raw-yuv, format=(fourcc)I420, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-indeo, indeoversion=(int)3, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-indeo, indeoversion=(int)4, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-indeo, indeoversion=(int)5, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-h263, variant=(string)lead, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-h264, variant=(string)lead, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-h263, variant=(string)microsoft, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/mpeg, mpegversion=(int)4, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; image/jpeg, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-msmpeg, msmpegversion=(int)42, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-msmpeg, msmpegversion=(int)43, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/mpeg, systemstream=(boolean)false, mpegversion=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/mpeg, systemstream=(boolean)false, mpegversion=(int)2, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-msmpeg, msmpegversion=(int)41, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-mszh, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; image/png, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-rle, layout=(string)microsoft, depth=(int)[ 1, 64 ], framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-indeo, indeoversion=(int)2, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/sp5x, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-truemotion, trueversion=(int)2, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-camtasia, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-ultimotion, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-raw-yuv, format=(fourcc)UYVY, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-ati-vcr, vcrversion=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-ati-vcr, vcrversion=(int)2, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-h263, variant=(string)vdolive, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-h263, variant=(string)vivo, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-vmnc, version=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-vp3, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-h264, variant=(string)videosoft, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-wmv, wmvversion=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-wmv, wmvversion=(int)2, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-wmv, wmvversion=(int)3, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-xvid, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-xan, wcversion=(int)4, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-raw-yuv, format=(fourcc)YUY2, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-raw-yuv, format=(fourcc)YVU9, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-zlib, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-cinepak, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-h264, variant=(string)itu, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-msvideocodec, msvideoversion=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-h263, variant=(string)xirlink, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-dirac, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-ffv, ffvversion=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-kmvc, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-vp5, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-vp6, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-vp6-flash, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-vp7, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-vp8, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-mimic, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-apple-video, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-theora, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-fraps, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-aasc, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-raw-yuv, format=(fourcc)YV12, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-loco, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-zmbv, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-dv, systemstream=(boolean)true; video/x-avi-unknown
+
video/x-3ivx, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-asus, asusversion=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-asus, asusversion=(int)2, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-cirrus-logic-accupak, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-camstudio, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-compressed-yuv, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-raw-rgb, bpp=(int){ 8, 24, 32 }, depth=(int){ 8, 24 }, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-divx, divxversion=(int)3, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-divx, divxversion=(int)4, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-truemotion, trueversion=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-dv, systemstream=(boolean)false, dvversion=(int)25, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-dv, systemstream=(boolean)false, dvversion=(int)50, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-divx, divxversion=(int)5, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/mpeg, mpegversion=(int)4, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-flash-video, flvversion=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-vp6-flash, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-h263, variant=(string)itu, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-h263, variant=(string)lucent, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-h264, variant=(string)itu, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-huffyuv, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-intel-h263, variant=(string)intel, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-raw-yuv, format=(fourcc)I420, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-indeo, indeoversion=(int)3, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-indeo, indeoversion=(int)4, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-indeo, indeoversion=(int)5, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-h263, variant=(string)lead, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-h264, variant=(string)lead, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-h263, variant=(string)microsoft, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/mpeg, mpegversion=(int)4, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; image/jpeg, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-msmpeg, msmpegversion=(int)42, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-msmpeg, msmpegversion=(int)43, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/mpeg, systemstream=(boolean)false, mpegversion=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/mpeg, systemstream=(boolean)false, mpegversion=(int)2, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-msmpeg, msmpegversion=(int)41, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-mszh, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; image/png, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-rle, layout=(string)microsoft, depth=(int)[ 1, 64 ], framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-indeo, indeoversion=(int)2, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/sp5x, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-truemotion, trueversion=(int)2, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-camtasia, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-ultimotion, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-raw-yuv, format=(fourcc)UYVY, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-ati-vcr, vcrversion=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-ati-vcr, vcrversion=(int)2, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-h263, variant=(string)vdolive, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-h263, variant=(string)vivo, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-vmnc, version=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-vp3, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-h264, variant=(string)videosoft, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-wmv, wmvversion=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-wmv, wmvversion=(int)2, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-wmv, wmvversion=(int)3, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-xvid, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-xan, wcversion=(int)4, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-raw-yuv, format=(fourcc)YUY2, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-raw-yuv, format=(fourcc)YVU9, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-zlib, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-cinepak, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-h264, variant=(string)itu, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-msvideocodec, msvideoversion=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-h263, variant=(string)xirlink, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-dirac, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-ffv, ffvversion=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-kmvc, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-vp5, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-vp6, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-vp6-flash, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-vp7, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-vp8, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-mimic, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-apple-video, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-theora, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-fraps, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-aasc, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-raw-yuv, format=(fourcc)YV12, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-loco, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-zmbv, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-raw-yuv, format=(fourcc)v210, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-raw-rgb, endianness=(int)4321, depth=(int)30, bpp=(int)32, red_mask=(int)1072693248, green_mask=(int)1047552, blue_mask=(int)1023, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]; video/x-dv, systemstream=(boolean)true; video/x-avi-unknown
diff --git a/docs/plugins/inspect/plugin-cacasink.xml b/docs/plugins/inspect/plugin-cacasink.xml index 4205723bf7..eb94ba0717 100644 --- a/docs/plugins/inspect/plugin-cacasink.xml +++ b/docs/plugins/inspect/plugin-cacasink.xml @@ -3,7 +3,7 @@ Colored ASCII Art video sink ../../ext/libcaca/.libs/libgstcacasink.so libgstcacasink.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-cairo.xml b/docs/plugins/inspect/plugin-cairo.xml index 1691a118ee..7ef711a705 100644 --- a/docs/plugins/inspect/plugin-cairo.xml +++ b/docs/plugins/inspect/plugin-cairo.xml @@ -3,7 +3,7 @@ Cairo-based elements ../../ext/cairo/.libs/libgstcairo.so libgstcairo.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-cutter.xml b/docs/plugins/inspect/plugin-cutter.xml index 9168291bd1..6d10c0c4b9 100644 --- a/docs/plugins/inspect/plugin-cutter.xml +++ b/docs/plugins/inspect/plugin-cutter.xml @@ -3,7 +3,7 @@ Audio Cutter to split audio into non-silent bits ../../gst/cutter/.libs/libgstcutter.so libgstcutter.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-debug.xml b/docs/plugins/inspect/plugin-debug.xml index f8021cae5a..42690e65be 100644 --- a/docs/plugins/inspect/plugin-debug.xml +++ b/docs/plugins/inspect/plugin-debug.xml @@ -3,7 +3,7 @@ elements for testing and debugging ../../gst/debugutils/.libs/libgstdebug.so libgstdebug.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-deinterlace.xml b/docs/plugins/inspect/plugin-deinterlace.xml index 448356da98..ac76c14ac8 100644 --- a/docs/plugins/inspect/plugin-deinterlace.xml +++ b/docs/plugins/inspect/plugin-deinterlace.xml @@ -3,7 +3,7 @@ Deinterlacer ../../gst/deinterlace/.libs/libgstdeinterlace.so libgstdeinterlace.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-dv.xml b/docs/plugins/inspect/plugin-dv.xml index fc450764e8..530ed69354 100644 --- a/docs/plugins/inspect/plugin-dv.xml +++ b/docs/plugins/inspect/plugin-dv.xml @@ -3,7 +3,7 @@ DV demuxer and decoder based on libdv (libdv.sf.net) ../../ext/dv/.libs/libgstdv.so libgstdv.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-efence.xml b/docs/plugins/inspect/plugin-efence.xml index c6cef5238b..ecf3b31b01 100644 --- a/docs/plugins/inspect/plugin-efence.xml +++ b/docs/plugins/inspect/plugin-efence.xml @@ -3,7 +3,7 @@ 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 - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-effectv.xml b/docs/plugins/inspect/plugin-effectv.xml index f6e05ab241..4e83773289 100644 --- a/docs/plugins/inspect/plugin-effectv.xml +++ b/docs/plugins/inspect/plugin-effectv.xml @@ -3,7 +3,7 @@ effect plugins from the effectv project ../../gst/effectv/.libs/libgsteffectv.so libgsteffectv.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-equalizer.xml b/docs/plugins/inspect/plugin-equalizer.xml index dea4fb069c..2eab97721d 100644 --- a/docs/plugins/inspect/plugin-equalizer.xml +++ b/docs/plugins/inspect/plugin-equalizer.xml @@ -3,7 +3,7 @@ GStreamer audio equalizers ../../gst/equalizer/.libs/libgstequalizer.so libgstequalizer.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-esdsink.xml b/docs/plugins/inspect/plugin-esdsink.xml index 4508ffd705..52d7cc1108 100644 --- a/docs/plugins/inspect/plugin-esdsink.xml +++ b/docs/plugins/inspect/plugin-esdsink.xml @@ -3,7 +3,7 @@ ESD Element Plugins ../../ext/esd/.libs/libgstesd.so libgstesd.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-flac.xml b/docs/plugins/inspect/plugin-flac.xml index 26c69196c2..58278c14b8 100644 --- a/docs/plugins/inspect/plugin-flac.xml +++ b/docs/plugins/inspect/plugin-flac.xml @@ -3,7 +3,7 @@ The FLAC Lossless compressor Codec ../../ext/flac/.libs/libgstflac.so libgstflac.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-flv.xml b/docs/plugins/inspect/plugin-flv.xml index 487ef8fc1f..fe41fd28ae 100644 --- a/docs/plugins/inspect/plugin-flv.xml +++ b/docs/plugins/inspect/plugin-flv.xml @@ -3,7 +3,7 @@ FLV muxing and demuxing plugin ../../gst/flv/.libs/libgstflv.so libgstflv.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git @@ -26,7 +26,7 @@ audio source sometimes -
audio/x-adpcm, layout=(string)swf, channels=(int){ 1, 2 }, rate=(int){ 5512, 11025, 22050, 44100 }; audio/mpeg, mpegversion=(int)1, layer=(int)3, channels=(int){ 1, 2 }, rate=(int){ 5512, 8000, 11025, 22050, 44100 }, parsed=(boolean)true; audio/mpeg, mpegversion=(int)4, framed=(boolean)true; audio/x-nellymoser, channels=(int){ 1, 2 }, rate=(int){ 5512, 8000, 11025, 16000, 22050, 44100 }; audio/x-raw-int, endianness=(int)1234, channels=(int){ 1, 2 }, width=(int)8, depth=(int)8, rate=(int){ 5512, 11025, 22050, 44100 }, signed=(boolean)false; audio/x-raw-int, endianness=(int)1234, channels=(int){ 1, 2 }, width=(int)16, depth=(int)16, rate=(int){ 5512, 11025, 22050, 44100 }, signed=(boolean)true; audio/x-alaw, channels=(int){ 1, 2 }, rate=(int){ 5512, 11025, 22050, 44100 }; audio/x-mulaw, channels=(int){ 1, 2 }, rate=(int){ 5512, 11025, 22050, 44100 }; audio/x-speex, channels=(int){ 1, 2 }, rate=(int){ 5512, 11025, 22050, 44100 }
+
audio/x-adpcm, layout=(string)swf, channels=(int){ 1, 2 }, rate=(int){ 5512, 11025, 22050, 44100 }; audio/mpeg, mpegversion=(int)1, layer=(int)3, channels=(int){ 1, 2 }, rate=(int){ 5512, 8000, 11025, 22050, 44100 }, parsed=(boolean)true; audio/mpeg, mpegversion=(int)4, stream-format=(string)raw, framed=(boolean)true; audio/x-nellymoser, channels=(int){ 1, 2 }, rate=(int){ 5512, 8000, 11025, 16000, 22050, 44100 }; audio/x-raw-int, endianness=(int)1234, channels=(int){ 1, 2 }, width=(int)8, depth=(int)8, rate=(int){ 5512, 11025, 22050, 44100 }, signed=(boolean)false; audio/x-raw-int, endianness=(int)1234, channels=(int){ 1, 2 }, width=(int)16, depth=(int)16, rate=(int){ 5512, 11025, 22050, 44100 }, signed=(boolean)true; audio/x-alaw, channels=(int){ 1, 2 }, rate=(int){ 5512, 11025, 22050, 44100 }; audio/x-mulaw, channels=(int){ 1, 2 }, rate=(int){ 5512, 11025, 22050, 44100 }; audio/x-speex, channels=(int){ 1, 2 }, rate=(int){ 5512, 11025, 22050, 44100 }
video @@ -47,7 +47,7 @@ audio sink request -
audio/x-adpcm, layout=(string)swf, channels=(int){ 1, 2 }, rate=(int){ 5512, 11025, 22050, 44100 }; audio/mpeg, mpegversion=(int)1, layer=(int)3, channels=(int){ 1, 2 }, rate=(int){ 5512, 8000, 11025, 22050, 44100 }, parsed=(boolean)true; audio/mpeg, mpegversion=(int){ 2, 4 }, framed=(boolean)true; audio/x-nellymoser, channels=(int){ 1, 2 }, rate=(int){ 5512, 8000, 11025, 16000, 22050, 44100 }; audio/x-raw-int, endianness=(int)1234, channels=(int){ 1, 2 }, width=(int)8, depth=(int)8, rate=(int){ 5512, 11025, 22050, 44100 }, signed=(boolean)false; audio/x-raw-int, endianness=(int)1234, channels=(int){ 1, 2 }, width=(int)16, depth=(int)16, rate=(int){ 5512, 11025, 22050, 44100 }, signed=(boolean)true; audio/x-alaw, channels=(int){ 1, 2 }, rate=(int){ 5512, 11025, 22050, 44100 }; audio/x-mulaw, channels=(int){ 1, 2 }, rate=(int){ 5512, 11025, 22050, 44100 }; audio/x-speex, channels=(int){ 1, 2 }, rate=(int){ 5512, 11025, 22050, 44100 }
+
audio/x-adpcm, layout=(string)swf, channels=(int){ 1, 2 }, rate=(int){ 5512, 11025, 22050, 44100 }; audio/mpeg, mpegversion=(int)1, layer=(int)3, channels=(int){ 1, 2 }, rate=(int){ 5512, 8000, 11025, 22050, 44100 }, parsed=(boolean)true; audio/mpeg, mpegversion=(int)2, framed=(boolean)true; audio/mpeg, mpegversion=(int)4, stream-format=(string)raw, framed=(boolean)true; audio/x-nellymoser, channels=(int){ 1, 2 }, rate=(int){ 5512, 8000, 11025, 16000, 22050, 44100 }; audio/x-raw-int, endianness=(int)1234, channels=(int){ 1, 2 }, width=(int)8, depth=(int)8, rate=(int){ 5512, 11025, 22050, 44100 }, signed=(boolean)false; audio/x-raw-int, endianness=(int)1234, channels=(int){ 1, 2 }, width=(int)16, depth=(int)16, rate=(int){ 5512, 11025, 22050, 44100 }, signed=(boolean)true; audio/x-alaw, channels=(int){ 1, 2 }, rate=(int){ 5512, 11025, 22050, 44100 }; audio/x-mulaw, channels=(int){ 1, 2 }, rate=(int){ 5512, 11025, 22050, 44100 }; audio/x-speex, channels=(int){ 1, 2 }, rate=(int){ 5512, 11025, 22050, 44100 }
 video diff --git a/docs/plugins/inspect/plugin-flxdec.xml b/docs/plugins/inspect/plugin-flxdec.xml index e035f3dc85..6e33406ad3 100644 --- a/docs/plugins/inspect/plugin-flxdec.xml +++ b/docs/plugins/inspect/plugin-flxdec.xml @@ -3,7 +3,7 @@ FLC/FLI/FLX video decoder ../../gst/flx/.libs/libgstflxdec.so libgstflxdec.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-gconfelements.xml b/docs/plugins/inspect/plugin-gconfelements.xml index d7e6e2945f..66a8943401 100644 --- a/docs/plugins/inspect/plugin-gconfelements.xml +++ b/docs/plugins/inspect/plugin-gconfelements.xml @@ -3,7 +3,7 @@ elements wrapping the GStreamer/GConf audio/video output settings ../../ext/gconf/.libs/libgstgconfelements.so libgstgconfelements.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-gdkpixbuf.xml b/docs/plugins/inspect/plugin-gdkpixbuf.xml index 8b27462bf3..10b8c5db7f 100644 --- a/docs/plugins/inspect/plugin-gdkpixbuf.xml +++ b/docs/plugins/inspect/plugin-gdkpixbuf.xml @@ -3,7 +3,7 @@ GdkPixbuf-based image decoder, scaler and sink ../../ext/gdk_pixbuf/.libs/libgstgdkpixbuf.so libgstgdkpixbuf.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-goom.xml b/docs/plugins/inspect/plugin-goom.xml index 4cdc7c29d2..fa59308ba9 100644 --- a/docs/plugins/inspect/plugin-goom.xml +++ b/docs/plugins/inspect/plugin-goom.xml @@ -3,7 +3,7 @@ GOOM visualization filter ../../gst/goom/.libs/libgstgoom.so libgstgoom.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-goom2k1.xml b/docs/plugins/inspect/plugin-goom2k1.xml index f2500bab23..04befbe7c1 100644 --- a/docs/plugins/inspect/plugin-goom2k1.xml +++ b/docs/plugins/inspect/plugin-goom2k1.xml @@ -3,7 +3,7 @@ GOOM 2k1 visualization filter ../../gst/goom2k1/.libs/libgstgoom2k1.so libgstgoom2k1.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-gstrtpmanager.xml b/docs/plugins/inspect/plugin-gstrtpmanager.xml index b72b3e7047..c0e12d3f9d 100644 --- a/docs/plugins/inspect/plugin-gstrtpmanager.xml +++ b/docs/plugins/inspect/plugin-gstrtpmanager.xml @@ -3,7 +3,7 @@ RTP session management plugin library ../../gst/rtpmanager/.libs/libgstrtpmanager.so libgstrtpmanager.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-halelements.xml b/docs/plugins/inspect/plugin-halelements.xml index 689988a8d8..e0ce5dbe3e 100644 --- a/docs/plugins/inspect/plugin-halelements.xml +++ b/docs/plugins/inspect/plugin-halelements.xml @@ -3,7 +3,7 @@ elements wrapping the GStreamer/HAL audio input/output devices ../../ext/hal/.libs/libgsthalelements.so libgsthalelements.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-icydemux.xml b/docs/plugins/inspect/plugin-icydemux.xml index 1a2f808b35..f8fe0d373f 100644 --- a/docs/plugins/inspect/plugin-icydemux.xml +++ b/docs/plugins/inspect/plugin-icydemux.xml @@ -3,7 +3,7 @@ Demux ICY tags from a stream ../../gst/icydemux/.libs/libgsticydemux.so libgsticydemux.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-id3demux.xml b/docs/plugins/inspect/plugin-id3demux.xml index ceafebd36d..414ccc66e5 100644 --- a/docs/plugins/inspect/plugin-id3demux.xml +++ b/docs/plugins/inspect/plugin-id3demux.xml @@ -3,7 +3,7 @@ Demux ID3v1 and ID3v2 tags from a file ../../gst/id3demux/.libs/libgstid3demux.so libgstid3demux.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-imagefreeze.xml b/docs/plugins/inspect/plugin-imagefreeze.xml index 887076310c..2d5416051a 100644 --- a/docs/plugins/inspect/plugin-imagefreeze.xml +++ b/docs/plugins/inspect/plugin-imagefreeze.xml @@ -3,7 +3,7 @@ Still frame stream generator ../../gst/imagefreeze/.libs/libgstimagefreeze.so libgstimagefreeze.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-interleave.xml b/docs/plugins/inspect/plugin-interleave.xml index 9dd3d189c3..b1c80e5797 100644 --- a/docs/plugins/inspect/plugin-interleave.xml +++ b/docs/plugins/inspect/plugin-interleave.xml @@ -3,7 +3,7 @@ Audio interleaver/deinterleaver ../../gst/interleave/.libs/libgstinterleave.so libgstinterleave.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-isomp4.xml b/docs/plugins/inspect/plugin-isomp4.xml index 8df339885c..1cbde325dc 100644 --- a/docs/plugins/inspect/plugin-isomp4.xml +++ b/docs/plugins/inspect/plugin-isomp4.xml @@ -3,7 +3,7 @@ ISO base media file format support (mp4, 3gpp, qt, mj2) ../../gst/isomp4/.libs/libgstisomp4.so libgstisomp4.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-jack.xml b/docs/plugins/inspect/plugin-jack.xml index b31bc2584c..891333fe55 100644 --- a/docs/plugins/inspect/plugin-jack.xml +++ b/docs/plugins/inspect/plugin-jack.xml @@ -3,7 +3,7 @@ JACK audio elements ../../ext/jack/.libs/libgstjack.so libgstjack.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-jpeg.xml b/docs/plugins/inspect/plugin-jpeg.xml index bc579c0c6c..b2266b34ea 100644 --- a/docs/plugins/inspect/plugin-jpeg.xml +++ b/docs/plugins/inspect/plugin-jpeg.xml @@ -3,7 +3,7 @@ JPeg plugin library ../../ext/jpeg/.libs/libgstjpeg.so libgstjpeg.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-level.xml b/docs/plugins/inspect/plugin-level.xml index 7c7cb22596..3b3e700eb2 100644 --- a/docs/plugins/inspect/plugin-level.xml +++ b/docs/plugins/inspect/plugin-level.xml @@ -3,7 +3,7 @@ Audio level plugin ../../gst/level/.libs/libgstlevel.so libgstlevel.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-matroska.xml b/docs/plugins/inspect/plugin-matroska.xml index c5180bbe0f..aed890cce6 100644 --- a/docs/plugins/inspect/plugin-matroska.xml +++ b/docs/plugins/inspect/plugin-matroska.xml @@ -3,7 +3,7 @@ Matroska and WebM stream handling ../../gst/matroska/.libs/libgstmatroska.so libgstmatroska.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git @@ -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-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 ]; audio/x-alaw, channels=(int){ 1, 2 }, rate=(int)[ 8000, 192000 ]; audio/x-mulaw, channels=(int){ 1, 2 }, rate=(int)[ 8000, 192000 ]
+
audio/mpeg, mpegversion=(int)1, layer=(int)[ 1, 3 ], channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/mpeg, mpegversion=(int){ 2, 4 }, stream-format=(string)raw, 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 ]; audio/x-alaw, channels=(int){ 1, 2 }, rate=(int)[ 8000, 192000 ]; audio/x-mulaw, channels=(int){ 1, 2 }, rate=(int)[ 8000, 192000 ]
 subtitle_%d sink request -
subtitle/x-kate
+
subtitle/x-kate; text/plain; application/x-ssa; application/x-ass; application/x-usf; video/x-dvd-subpicture; application/x-subtitle-unknown
 video_%d diff --git a/docs/plugins/inspect/plugin-mulaw.xml b/docs/plugins/inspect/plugin-mulaw.xml index e805bdf63c..14d06f4ed8 100644 --- a/docs/plugins/inspect/plugin-mulaw.xml +++ b/docs/plugins/inspect/plugin-mulaw.xml @@ -3,7 +3,7 @@ MuLaw audio conversion routines ../../gst/law/.libs/libgstmulaw.so libgstmulaw.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-multifile.xml b/docs/plugins/inspect/plugin-multifile.xml index f14ec4bd7c..e46737073f 100644 --- a/docs/plugins/inspect/plugin-multifile.xml +++ b/docs/plugins/inspect/plugin-multifile.xml @@ -3,7 +3,7 @@ Reads/Writes buffers from/to sequentially named files ../../gst/multifile/.libs/libgstmultifile.so libgstmultifile.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-multipart.xml b/docs/plugins/inspect/plugin-multipart.xml index 0b9538a41c..03e3752c9e 100644 --- a/docs/plugins/inspect/plugin-multipart.xml +++ b/docs/plugins/inspect/plugin-multipart.xml @@ -3,7 +3,7 @@ multipart stream manipulation ../../gst/multipart/.libs/libgstmultipart.so libgstmultipart.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-navigationtest.xml b/docs/plugins/inspect/plugin-navigationtest.xml index 640982c9bf..7ee88d4c7b 100644 --- a/docs/plugins/inspect/plugin-navigationtest.xml +++ b/docs/plugins/inspect/plugin-navigationtest.xml @@ -3,7 +3,7 @@ Template for a video filter ../../gst/debugutils/.libs/libgstnavigationtest.so libgstnavigationtest.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-oss4.xml b/docs/plugins/inspect/plugin-oss4.xml index 533de3dbef..18a7c891d1 100644 --- a/docs/plugins/inspect/plugin-oss4.xml +++ b/docs/plugins/inspect/plugin-oss4.xml @@ -3,7 +3,7 @@ Open Sound System (OSS) version 4 support for GStreamer ../../sys/oss4/.libs/libgstoss4audio.so libgstoss4audio.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-ossaudio.xml b/docs/plugins/inspect/plugin-ossaudio.xml index 564ed53a17..693ededf5e 100644 --- a/docs/plugins/inspect/plugin-ossaudio.xml +++ b/docs/plugins/inspect/plugin-ossaudio.xml @@ -3,7 +3,7 @@ OSS (Open Sound System) support for GStreamer ../../sys/oss/.libs/libgstossaudio.so libgstossaudio.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-png.xml b/docs/plugins/inspect/plugin-png.xml index ec1e9d13f6..291d33e506 100644 --- a/docs/plugins/inspect/plugin-png.xml +++ b/docs/plugins/inspect/plugin-png.xml @@ -3,7 +3,7 @@ PNG plugin library ../../ext/libpng/.libs/libgstpng.so libgstpng.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-pulseaudio.xml b/docs/plugins/inspect/plugin-pulseaudio.xml index 7426de9275..628120d39e 100644 --- a/docs/plugins/inspect/plugin-pulseaudio.xml +++ b/docs/plugins/inspect/plugin-pulseaudio.xml @@ -3,7 +3,7 @@ PulseAudio plugin library ../../ext/pulse/.libs/libgstpulse.so libgstpulse.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-replaygain.xml b/docs/plugins/inspect/plugin-replaygain.xml index 419354bc99..1dd437b656 100644 --- a/docs/plugins/inspect/plugin-replaygain.xml +++ b/docs/plugins/inspect/plugin-replaygain.xml @@ -3,7 +3,7 @@ ReplayGain volume normalization ../../gst/replaygain/.libs/libgstreplaygain.so libgstreplaygain.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-rtp.xml b/docs/plugins/inspect/plugin-rtp.xml index bd6d7a9ba9..d90871aca5 100644 --- a/docs/plugins/inspect/plugin-rtp.xml +++ b/docs/plugins/inspect/plugin-rtp.xml @@ -3,7 +3,7 @@ Real-time protocol plugins ../../gst/rtp/.libs/libgstrtp.so libgstrtp.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git @@ -668,7 +668,7 @@ src source always -
video/x-h264
+
video/x-h264, stream-format=(string)avc, alignment=(string)au; video/x-h264, stream-format=(string)byte-stream, alignment=(string){ nal, au }
 @@ -683,7 +683,7 @@ sink sink always -
video/x-h264
+
video/x-h264, stream-format=(string)byte-stream, alignment=(string){ nal, au }; video/x-h264, stream-format=(string)avc, alignment=(string)au
src diff --git a/docs/plugins/inspect/plugin-rtsp.xml b/docs/plugins/inspect/plugin-rtsp.xml index ebe286380f..94785e59d6 100644 --- a/docs/plugins/inspect/plugin-rtsp.xml +++ b/docs/plugins/inspect/plugin-rtsp.xml @@ -3,7 +3,7 @@ transfer data via RTSP ../../gst/rtsp/.libs/libgstrtsp.so libgstrtsp.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-shapewipe.xml b/docs/plugins/inspect/plugin-shapewipe.xml index 4631cbcdab..193d70325e 100644 --- a/docs/plugins/inspect/plugin-shapewipe.xml +++ b/docs/plugins/inspect/plugin-shapewipe.xml @@ -3,7 +3,7 @@ Shape Wipe transition filter ../../gst/shapewipe/.libs/libgstshapewipe.so libgstshapewipe.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-shout2send.xml b/docs/plugins/inspect/plugin-shout2send.xml index 2f9d8f1dc6..9f8c7ec22f 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 - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good libshout2 diff --git a/docs/plugins/inspect/plugin-smpte.xml b/docs/plugins/inspect/plugin-smpte.xml index 514147e611..d9818ea777 100644 --- a/docs/plugins/inspect/plugin-smpte.xml +++ b/docs/plugins/inspect/plugin-smpte.xml @@ -3,7 +3,7 @@ Apply the standard SMPTE transitions on video images ../../gst/smpte/.libs/libgstsmpte.so libgstsmpte.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-soup.xml b/docs/plugins/inspect/plugin-soup.xml index da3d787dc8..5e6622dad2 100644 --- a/docs/plugins/inspect/plugin-soup.xml +++ b/docs/plugins/inspect/plugin-soup.xml @@ -3,7 +3,7 @@ libsoup HTTP client src/sink ../../ext/soup/.libs/libgstsouphttpsrc.so libgstsouphttpsrc.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-spectrum.xml b/docs/plugins/inspect/plugin-spectrum.xml index 5d98644b4f..7c66e4564f 100644 --- a/docs/plugins/inspect/plugin-spectrum.xml +++ b/docs/plugins/inspect/plugin-spectrum.xml @@ -3,7 +3,7 @@ Run an FFT on the audio signal, output spectrum data ../../gst/spectrum/.libs/libgstspectrum.so libgstspectrum.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-speex.xml b/docs/plugins/inspect/plugin-speex.xml index aee0ba804c..bea4630ec6 100644 --- a/docs/plugins/inspect/plugin-speex.xml +++ b/docs/plugins/inspect/plugin-speex.xml @@ -3,7 +3,7 @@ Speex plugin library ../../ext/speex/.libs/libgstspeex.so libgstspeex.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-taglib.xml b/docs/plugins/inspect/plugin-taglib.xml index f635b757da..7638b10dea 100644 --- a/docs/plugins/inspect/plugin-taglib.xml +++ b/docs/plugins/inspect/plugin-taglib.xml @@ -3,7 +3,7 @@ Tag writing plug-in based on taglib ../../ext/taglib/.libs/libgsttaglib.so libgsttaglib.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-udp.xml b/docs/plugins/inspect/plugin-udp.xml index e58e3c5cad..128697d8b3 100644 --- a/docs/plugins/inspect/plugin-udp.xml +++ b/docs/plugins/inspect/plugin-udp.xml @@ -3,7 +3,7 @@ transfer data via UDP ../../gst/udp/.libs/libgstudp.so libgstudp.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-video4linux2.xml b/docs/plugins/inspect/plugin-video4linux2.xml index 004de03a17..aca1e5fe0a 100644 --- a/docs/plugins/inspect/plugin-video4linux2.xml +++ b/docs/plugins/inspect/plugin-video4linux2.xml @@ -3,7 +3,7 @@ elements for Video 4 Linux ../../sys/v4l2/.libs/libgstvideo4linux2.so libgstvideo4linux2.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-videobox.xml b/docs/plugins/inspect/plugin-videobox.xml index 3a932372ae..2c87de01ae 100644 --- a/docs/plugins/inspect/plugin-videobox.xml +++ b/docs/plugins/inspect/plugin-videobox.xml @@ -3,7 +3,7 @@ resizes a video by adding borders or cropping ../../gst/videobox/.libs/libgstvideobox.so libgstvideobox.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-videocrop.xml b/docs/plugins/inspect/plugin-videocrop.xml index dd71aa646d..1e0be6bf0e 100644 --- a/docs/plugins/inspect/plugin-videocrop.xml +++ b/docs/plugins/inspect/plugin-videocrop.xml @@ -3,7 +3,7 @@ Crops video into a user-defined region ../../gst/videocrop/.libs/libgstvideocrop.so libgstvideocrop.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-videofilter.xml b/docs/plugins/inspect/plugin-videofilter.xml index 9ab132a71d..58483fb784 100644 --- a/docs/plugins/inspect/plugin-videofilter.xml +++ b/docs/plugins/inspect/plugin-videofilter.xml @@ -3,7 +3,7 @@ Video filters plugin ../../gst/videofilter/.libs/libgstvideofilter.so libgstvideofilter.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-videomixer.xml b/docs/plugins/inspect/plugin-videomixer.xml index aab551138f..3fda4715e7 100644 --- a/docs/plugins/inspect/plugin-videomixer.xml +++ b/docs/plugins/inspect/plugin-videomixer.xml @@ -3,7 +3,7 @@ Video mixer ../../gst/videomixer/.libs/libgstvideomixer.so libgstvideomixer.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-wavenc.xml b/docs/plugins/inspect/plugin-wavenc.xml index 890bbe16aa..5cf8ebc9f9 100644 --- a/docs/plugins/inspect/plugin-wavenc.xml +++ b/docs/plugins/inspect/plugin-wavenc.xml @@ -3,7 +3,7 @@ Encode raw audio into WAV ../../gst/wavenc/.libs/libgstwavenc.so libgstwavenc.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-wavpack.xml b/docs/plugins/inspect/plugin-wavpack.xml index d5b7cbddb6..8a489dc165 100644 --- a/docs/plugins/inspect/plugin-wavpack.xml +++ b/docs/plugins/inspect/plugin-wavpack.xml @@ -3,7 +3,7 @@ Wavpack lossless/lossy audio format handling ../../ext/wavpack/.libs/libgstwavpack.so libgstwavpack.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-wavparse.xml b/docs/plugins/inspect/plugin-wavparse.xml index 90b4780ed0..7f8ef480c7 100644 --- a/docs/plugins/inspect/plugin-wavparse.xml +++ b/docs/plugins/inspect/plugin-wavparse.xml @@ -3,7 +3,7 @@ Parse a .wav file into raw audio ../../gst/wavparse/.libs/libgstwavparse.so libgstwavparse.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-ximagesrc.xml b/docs/plugins/inspect/plugin-ximagesrc.xml index 41747a25d6..9ce8fd0037 100644 --- a/docs/plugins/inspect/plugin-ximagesrc.xml +++ b/docs/plugins/inspect/plugin-ximagesrc.xml @@ -3,7 +3,7 @@ X11 video input plugin using standard Xlib calls ../../sys/ximage/.libs/libgstximagesrc.so libgstximagesrc.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/docs/plugins/inspect/plugin-y4menc.xml b/docs/plugins/inspect/plugin-y4menc.xml index 5f04cd0547..b454642450 100644 --- a/docs/plugins/inspect/plugin-y4menc.xml +++ b/docs/plugins/inspect/plugin-y4menc.xml @@ -3,7 +3,7 @@ Encodes a YUV frame into the yuv4mpeg format (mjpegtools) ../../gst/y4m/.libs/libgsty4menc.so libgsty4menc.so - 0.10.30.1 + 0.10.31.1 LGPL gst-plugins-good GStreamer Good Plug-ins git diff --git a/win32/common/config.h b/win32/common/config.h index c8c9ac4915..3146889d23 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 "2011-05-18T13:36Z" +#define GST_PACKAGE_RELEASE_DATETIME "2012-02-23T10:45Z" /* struct v4l2_buffer missing */ #undef GST_V4L2_MISSING_BUFDECL @@ -277,24 +277,12 @@ /* Define to enable pulseaudio plug-in (used by pulseaudio). */ #undef HAVE_PULSE -/* defined if pulseaudio >= 0.9.11 is available */ -#undef HAVE_PULSE_0_9_11 - -/* defined if pulseaudio >= 0.9.12 is available */ -#undef HAVE_PULSE_0_9_12 - -/* defined if pulseaudio >= 0.9.13 is available */ -#undef HAVE_PULSE_0_9_13 - -/* defined if pulseaudio >= 0.9.15 is available */ -#undef HAVE_PULSE_0_9_15 - -/* defined if pulseaudio >= 0.9.16 is available */ -#undef HAVE_PULSE_0_9_16 - /* defined if pulseaudio >= 0.9.20 is available */ #undef HAVE_PULSE_0_9_20 +/* defined if pulseaudio >= 1.0 is available */ +#undef HAVE_PULSE_1_0 + /* Define if RDTSC is available */ #undef HAVE_RDTSC @@ -355,6 +343,9 @@ /* Define if valgrind should be used */ #undef HAVE_VALGRIND +/* Define to enable Win32 WaveForm (used by waveformsink). */ +#undef HAVE_WAVEFORM + /* Define to enable wavpack plug-in (used by wavpack). */ #undef HAVE_WAVPACK @@ -370,7 +361,7 @@ /* Define to enable X11 XVideo extensions. */ #undef HAVE_XVIDEO -/* Define to enable zlib support for id3demux/qtdemux/matroska. */ +/* Define to enable zlib support for qtdemux/matroska. */ #undef HAVE_ZLIB /* the host CPU */ @@ -396,7 +387,7 @@ #define PACKAGE_NAME "GStreamer Good Plug-ins" /* Define to the full name and version of this package. */ -#define PACKAGE_STRING "GStreamer Good Plug-ins 0.10.30.1" +#define PACKAGE_STRING "GStreamer Good Plug-ins 0.10.31.1" /* Define to the one symbol short name of this package. */ #define PACKAGE_TARNAME "gst-plugins-good" @@ -405,7 +396,7 @@ #undef PACKAGE_URL /* Define to the version of this package. */ -#define PACKAGE_VERSION "0.10.30.1" +#define PACKAGE_VERSION "0.10.31.1" /* directory where plugins are located */ #ifdef _DEBUG @@ -436,7 +427,7 @@ #undef STDC_HEADERS /* Version number of package */ -#define VERSION "0.10.30.1" +#define VERSION "0.10.31.1" /* old wavpack API */ #undef WAVPACK_OLD_API From f9207722ca8fd8dcc1e7215d8af85efe4debfdf4 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Thu, 23 Feb 2012 12:05:20 +0000 Subject: [PATCH 06/55] Update NEWS and RELEASE as well --- NEWS | 245 +++++++++++++++++++++++++++++++++++++++++++++- RELEASE | 299 ++++++++++++++++++++++++++++++++++++++++++++++++++++++-- 2 files changed, 535 insertions(+), 9 deletions(-) diff --git a/NEWS b/NEWS index 180d21b833..d4066eca9a 100644 --- a/NEWS +++ b/NEWS @@ -1,4 +1,246 @@ -This is GStreamer Good Plug-ins 0.10.30, "Adagio" +This is GStreamer Good Plug-ins 0.10.31, "Faster" + +Changes since 0.10.30: + + * audioparsers: propagate downstream caps constraints upstream + * ac3parse: add support for IEC 61937 alignment and conversion/switching between alignments + * ac3parse: let bsid 9 and 10 through + * auparse: implement seeking + * avidemux: fix wrong stride when inverting uncompressed video + * cairotextoverlay: add a "silent" property to skip rendering; forward new segment events + * deinterlace: add support for deinterlacing using buffer caps/flags (as set by e.g. fieldanalysis) + * deinterlace: new fieldanalysis-related properties: "locking" and "ignore-obscure" + * directsoundsink: fix negotiation/device setup: 16-bit audio is signed, 8-bit is unsigned + * effecttv: fix reverse negotiation; repair color modes in radioactv by taking rgb,bgr into account + * equalizer: also sync the parameters for the filter bands + * flacdec: better timestamp/offset handling; try upstream first for duration queries + * flacdec: send EOS when seeking after the end of file instead of failing + * flacenc: do not drop the first data buffer on the floor + * flacparse: detect when a file lies about fixed block size; ignore invalid minimum_blocksize + * flacparse: more accurate/better duration/timestamp handling + * flvdemux: better timestamp handling (negative cts, detect large pts gaps; fix discontinuity threshold check when timestamps go backwards) + * flvmux: properly determine final duration; metadata/header writing fixes + * gdkpixbufsink: fix inverted pixel-aspect-ratio info on pixbufs + * jack: add "client-name" property to jackaudiosink and jackaudiosrc + * jpegdec: add sof-marker to template caps, so we don't get plugged for lossless jpeg + * jpegdec: Implement upstream negotiation + * matroskademux: seeking fixes; better handling of non-finalized files + * matroskademux: better timestamp/duration handling, fix some stuttering A/V + * matroskademux: add "max-gap-time" property to make gap handling configurable + * matroskademux: UTF-8 subtitles may have markup + * matroskamux: do not use unoffical V_MJPEG codec id + * matroskamux: fix segment handling, so we actually use running time + * matroskamux: for streaming files, push tags first + * matroskamux: handle GstForceKeyUnit event + * multifile: new splitfilesrc element to read multiple files as if they were one single file + * multifilesrc: add "loop" property + * multifilesink: handle buffer lists, useful to keep groups of buffers (GOPs) in the same file + * multifilesink: add flag to cut after a force key unit event + * multifilesink: add "max-files" property + * multifilesink: add new 'max-size' mode and "max-file-size" property for switching to the next file based on size + * multifilesink: write stream-headers when switching to the next file in max-size mode + * multipartdemux: Add property to assume a single stream and emit no-more-pads + * multipartmux: Add \r\n to tail of pushed buffers + * navseek: toggle pause/play on space bar + * osxvideo: Fix leak of NSOpenGLPixelFormat object + * pcmadepay,pcmudepay: allow variable sample rate + * pngenc: increase arbitrary resolution limits + * pulse: Drop support for PA versions before 0.9.16 (1.x is recommended) + * pulse: new pulseaudiosink element to handle format changes (not autoplugged yet) + * pulsesink: add support for compressed audio format passthrough (S/PDIF, mp3-over-bluetooth) + * pulsesink: Allow writes in bigger chunks + * pulsesink: Use the extended stream API if available + * pulsesrc: add a "source-output-index" property; implement GstStreamVolume interface + * qtdemux: better fragmented support (avoid adjustment for keyframe seek; mark all audio track samples as keyframe) + * qtdemux: parse embedded ID32 tags; improve bitrate guessing/extraction + * qtdemux: push mode fixes, fix buffered streaming + * qtmux: add direct dirac mapping + * qtmux: calculate average bitrate for streams + * qtmux: fix ctts generation for streams that don't start at 0 timestamps + * qtmux: use GST_TAG_IMAGE for coverart too + * ismlmux: Use iso-fragmented as variant type (useful in connection with encodebin) + * rtph263ppay: implement getcaps following RFC 4629, picks the right annexes + * rtph263ppay: set H263-2000 if thats what the other side wants + * rtph264depay: complete merged AU on marker bit (thus reducing latency) + * rtph264depay: cope with FU-A E bit not being set (caused by buggy payloaders) + * rtph264depay: exclude NALu size from payload length on truncated packets + * rtph264pay: proxy downstream caps restrictions (converting profile-level-id from RTP caps into video/x-h264 style caps) + * rtph264pay: only set the marker bit on the last NALU of a multi-NALU access unit + * rtpjpegpay: add support for H.264 payload in MJPEG container + * rtpjpegpay: fix for "odd" resolutions not a multiple of DCTSIZE + * rtpmp4adepay: fix output buffer timestamps in case of multiple frames + * rtpmp4gdepay: improve bogus interleaved index compensating + * rtpmp4vpay: deprecated send-config property and replace by config-interval + * rtppcmapay/depay: static clock rates on static payloads, dynamic on dynamic + * rtpvrawpay,-depay: RGB video payloading/depayloading fixes + * rtpg722pay: Compensate for clockrate vs. samplerate difference + * rtpbin: allow configurable rtcp stream syncing interval + * rtpbin: new "rtcp-sync" property, alternative inter-stream syncing methods + * rtpjitterbuffer/rtpbin: relax dropping rtcp packets; misc other fixes + * rtpmanager: don't reveal the user's username, hostname or real name by default + * rtpsession: process received Full Intra Requests (FIR) + * rtpsession: add special mode to use FIR as repair as Google does + * rtpsession: send FIR requests in response to key unit requests with all-headers=TRUE + * rtpsession: always send application requested feedback in immediate mode + * rtpsession: put the PLI requests in each RTPSource + * rtpsession: wait longer to timeout SSRC collision + * rtspsrc: implement async network I/O + * rtspsrc: allow sending short RTSP requests to a server + * rtspsrc: configure rtcp interval if provided + * rtspsrc: open on play and pause when not done yet + * shout2send: send video/webm through libshout + * soup: new souphttpclientsink element + * udpsrc: drop dataless UDP packets + * v4l2: take care not to change the current format where appropriate + * v4l2src, v4l2sink: add "norm" property; default to a pixel-aspect-ratio of 1/1 + * v4l2src: do not ignore the highest frame interval or the largest resolution + * v4l2src: handle some feature query failures more gracefully + * videobox: avoid wrapping opaque to transparent + * wavenc: Allow setcaps to be called after a format was negotiated if it's compatible + * ximagesrc: add "xid" and "xname" properties to allow capturing a particular window + * ximagesrc: fallback to non-XShm mode if allocating the XShm image failed + * ximagesrc: clear flags on buffer reuse, so that flags like DISCONT aren't set accidentally + +Bugs fixed since 0.10.30: + + * 668320 : rtpmanager: RTCP receiver reports reveal full user name + * 652727 : multifilesrc: add ability to loop + * 657422 : [souphttpsrc] The souphttpsrc plugin doesn't work behind a proxy that requires authentication + * 432612 : [matroskamux] doesn't handle segments correctly + * 541215 : [avimux] Dirac muxing is broken and results in A/V sync issues + * 546932 : [ximagesrc] allow recording of specific window only + * 571400 : RTSP blocks in gst_element_set_state( GST_STATE_PAUSED ) and incorrect url + * 576524 : rtpbin, jitterbuffer: add mode to support for recording RTP streams + * 586450 : [cairotextoverlay] Forward upstream events to both sinkpads + * 595055 : [pulsesrc] Should implement GstStreamVolume interface + * 605834 : directsoundsink: 16-bit audio is always signed while 8-bit is always unsigned + * 610916 : rtspsrc dosen't work under windows + * 614803 : v4l2: add tv-norm property + * 616686 : multipartdemux: add " single-stream " property to emit no-more-pads earlier + * 616936 : [matroskademux] Incorrect display of subtitles with markup + * 619548 : qtdemux: Guess bitrate if only one stream's bitrate is unknown + * 619590 : [matroskademux] Doesn't protect segment and other fields from concurrent changes from different threads + * 620186 : qtdemux: Export max bitrate for AMR-NB/-WB streams + * 622412 : [rtpmp4vpay] remove send-config parameter; obsoleted by config-interval + * 624887 : pitivi playback hangs / errors while playing mov clips on clip change + * 630456 : [ximagesrc] Fallback to non-XShm mode if image allocation fails + * 631430 : [flvdemux] Cannot play .flv to the end + * 632504 : [rtspsrc] reduce or avoid (network) hang during shutdown + * 634093 : RTSP client asks for unicast from multicast only server + * 638300 : v4l2src: make this work more than once in a row + * 639217 : udpsrc: allow skip-first-bytes of full buffer size + * 640323 : [cairotextoverlay] forward new segment events from the sink to the source + * 643847 : deinterlace: Add support for deinterlacing using buffer caps/flags + * 644151 : [multifilesink] Add option to create a new file after each GstForceKeyUnit event + * 644154 : [matroskamux] Force a new cluster after each GstForceKeyUnit event + * 644512 : [auparse] Add seeking + * 647540 : autoaudiosink picks element to use by rank, but pulsesink/alsasink/jackaudiosink are all PRIMARY + * 648312 : [v4l2sink] Unconditionally accepts video/mpegts + * 648642 : rtpsession: Ensure ssrc collisions aren't timed out immediately + * 648937 : matroskademux: avoid building index when streamable + * 649067 : v4l2src: got unexpected frame size of 262254 instead of 614400 + * 649617 : [rtp] Deadlock and other fixes for rtpssrcdemux + * 649780 : flac: seek beyond end fails instead of EOSing immediately + * 649955 : flvmux: add support for mpegversion 2, which is also AAC + * 650258 : matroskademux/matroskaparse: gst_element_query_duration returns wrong value for Matroska files + * 650313 : ac3parse: Add support for iec61937 alignment + * 650503 : [dvdemux] Broken DURATION query handling + * 650555 : [aacparse] AAC profiles needed in caps + * 650691 : [flacparse] regression playing some flac files + * 650714 : [amrparse] skips first few frames (problem in checking sync) + * 650785 : [flacparse] duration query in DEFAULT format failing with flacparse in pipeline (regression) + * 650877 : matroska: refactor code common to matroskademux and matroskaparse + * 650912 : Rare leak in qtdemux + * 650916 : REGRESSION: ssrcdemux causing FLOW_NOT_LINKED + * 650937 : deinterlace: fix parameter type in trace + * 651059 : rtspsrc: uniform unknown message handling + * 651443 : multifilesink: add next-file=max-size mode and max-file-size property + * 652195 : matroskademux: seeking in non-finalized matroska files does not work correctly + * 652286 : matroskaparse: Gstreamer-CRITICAL when changing state from PAUSED to READY + * 652467 : matroska: missing < stdio.h > include for sscanf + * 653080 : matroskamux: make check for block_duration less sensitive + * 653091 : [dv1394src] Make the internal clock thread-safe + * 653327 : configure script for gst-plugins-good selects shout2 when it's not present + * 653559 : aacparse: too greedy minimum frame size + * 653709 : [ximagesrc] sets DISCONT on half the buffers + * 654175 : matroskademux: handle blocks with duration=0 + * 654379 : matroskamux: make default framerate optional per stream + * 654583 : Immediate RTCP in rtpsession + * 654585 : rtpmp4gdepay choppy sound + * 654744 : matroskademux: fix aspect ratio if header has only onle display variable set + * 654749 : goom: unbreak build on PPC on openbsd + * 654816 : [rtspsrc] rtspsrc doesn't get eos if it's wrapped into a bin + * 655530 : Logitech B990 HD Webcam yields poor video in MJPEG mode. + * 655570 : qtdemux: assertion error when playing Apple Trailers + * 655805 : Make the extended RTSP headers optional + * 655866 : jackaudiosink: Don't call g_alloca in jack_process_cb + * 655918 : qtdemux : qtdemux_add_fragmented_samples return error. + * 656104 : v4l2src fails to correctly configure the framerate + * 656606 : crash in gst_spectrum_reset_message_data() + * 656649 : flacparse: fix off by one in frame size check + * 656734 : [aacparse] Assumes 1024 samples per frame + * 657080 : aacparse: failing test due to two buffers being dropped for one sync loss + * 657179 : pulse: New pulseaudiosink element to handle format changes + * 657376 : rtspsrc regression + * 657830 : multiudpsink: make add/remove/clear/get-stats action signals + * 658178 : udpsrc: rough error reporting when using an invalid URI + * 658305 : [souphttpsrc] can’t seek during double speed playback + * 658419 : Add FIR support to rtpsession + * 658543 : [v4l2src] Use GST_RESOURCE_ERROR_BUSY if webcam is already used + * 658546 : ac3parse: RealAudio file with AC-3 audio no longer plays + * 659009 : [matroskademux] property for configuring gap handling + * 659065 : navseek: toggle pause/play on space bar + * 659153 : matroskademux: fix stuttering A/V + * 659237 : [gstrtpbin] clock is not unreffed after finish using it + * 659242 : [matroskademux] Unexpected EOS when seeking on paused matroska file + * 659798 : Segfault when you convert with audioconvert from audio file mkv to audio file avi + * 659808 : matroskademux: misc fixes + * 659837 : matroskamux: unable to mux audio/x-raw-int,rate=8000,channels=1,endianness=1234,width=16,depth=16,signed=true + * 659943 : [ac3parse] it does not correcly check for ac3/e-ac3 switch + * 660249 : won't play wav file: invalid WAV header (no fmt at start): ID32 + * 660275 : jpegdec doesn't implement upstream negotiation + * 660294 : goom2k1: Fix mingw compiler warnings + * 660448 : videomixer2: memory leak + * 660468 : speexenc: fix calculation of filler data size + * 660481 : v4l, ximagesrc: printf format warnings + * 660969 : qtmux memleak + * 661049 : matroskademux: support seek with start_type NONE + * 661215 : flacparse: fix last frame timestamp in fixed block size mode + * 661400 : rtpg722pay: G722 rtptime too fast + * 661477 : flvdemux: negative cts causes uint overflow, resulting in sinks waiting forever + * 661841 : [edgetv] video artifacts if videorate placed after edgetv + * 661874 : aacparse fails to forward caps to encoder + * 662856 : cairotextoverlay: add a 'silent' property to skip rendering + * 663186 : taginject is not gap aware + * 663334 : gst/flv/: add amfdefs.h to noinst_HEADERS + * 663580 : v4l2src negotiation failure with weird pixel-aspect-ratios + * 664548 : matroskaparse: memleak + * 664792 : Staircase effect in M-JPEG over RTP with unaligned image dimensions.. + * 664892 : [matroskademux] Doesn't set caps properly + * 665387 : v4l2src: fix stepwise enumeration ignoring the highest values + * 665412 : matroskamux: jpeg muxing regression + * 665502 : [flvdemux] broken a/v sync for some files + * 665666 : multifilesink: GstMultiFileSinkNext not documented + * 665872 : jackaudiosink, jackaudiosrc: add " client-name " property + * 665882 : gdkpixbufsink: " pixel-aspect-ratio " is the inverse of what it should be + * 665911 : Ability to specify ignore-length in wavparse + * 666361 : playbin2: regression: visualisations don't work with pulseaudiosink + * 666583 : matroskademux: too many bus messages in streamable mode + * 666602 : ac3parse: no valid frames found before end of stream (unexpected bsid=10) + * 666644 : udpsrc: infinite loop on dataless UDP packets + * 666688 : jpedec: peer_caps leak + * 666711 : rtspsrc: hostname lookup is not thread safe + * 667419 : matroskamux memleaks + * 667818 : osxvideo: Fix leak of NSOpenGLPixelFormat object + * 667820 : rtpptdemux: Plug potential pad leak. + * 667846 : rtph264depay: Exclude NALu size from payload length on truncated packets. + * 668648 : gst-plugins-good does not compile: cairo cannot find libgstvideo-0.10 + * 669455 : V4l2src can't open webcamstudio new vloopback + * 669590 : [shout2send] support webm streaming + * 670197 : v4l2src: webcam doesn't work due to fatal error when querying color balance attributes + * 650960 : flacparse makes decoded flac files start at sample offset 9215 + * 659947 : souphttpsink: rename to souphttpclientsink? + * 658659 : qtmux: Fix ctts entries for streams that don't start with timestamps from 0 Changes since 0.10.29: @@ -8,6 +250,7 @@ Changes since 0.10.29: Bugs fixed since 0.10.29: + * 650470 : rtp: some depayloaders break with malformed rtp packets * 652144 : udp: fix compiler warning on mingw-w64 diff --git a/RELEASE b/RELEASE index d36babf663..7de2eaba59 100644 --- a/RELEASE +++ b/RELEASE @@ -1,5 +1,5 @@ -Release notes for GStreamer Good Plug-ins 0.10.30 "Adagio" +Release notes for GStreamer Good Plug-ins 0.10.31 "Faster" @@ -52,14 +52,245 @@ contains a set of less supported plug-ins that haven't passed the Features of this release - * work around GLib atomic ops API change - * better handling of malformed buffers in RTP depayloders - * some minor compilation fixes + * audioparsers: propagate downstream caps constraints upstream + * ac3parse: add support for IEC 61937 alignment and conversion/switching between alignments + * ac3parse: let bsid 9 and 10 through + * auparse: implement seeking + * avidemux: fix wrong stride when inverting uncompressed video + * cairotextoverlay: add a "silent" property to skip rendering; forward new segment events + * deinterlace: add support for deinterlacing using buffer caps/flags (as set by e.g. fieldanalysis) + * deinterlace: new fieldanalysis-related properties: "locking" and "ignore-obscure" + * directsoundsink: fix negotiation/device setup: 16-bit audio is signed, 8-bit is unsigned + * effecttv: fix reverse negotiation; repair color modes in radioactv by taking rgb,bgr into account + * equalizer: also sync the parameters for the filter bands + * flacdec: better timestamp/offset handling; try upstream first for duration queries + * flacdec: send EOS when seeking after the end of file instead of failing + * flacenc: do not drop the first data buffer on the floor + * flacparse: detect when a file lies about fixed block size; ignore invalid minimum_blocksize + * flacparse: more accurate/better duration/timestamp handling + * flvdemux: better timestamp handling (negative cts, detect large pts gaps; fix discontinuity threshold check when timestamps go backwards) + * flvmux: properly determine final duration; metadata/header writing fixes + * gdkpixbufsink: fix inverted pixel-aspect-ratio info on pixbufs + * jack: add "client-name" property to jackaudiosink and jackaudiosrc + * jpegdec: add sof-marker to template caps, so we don't get plugged for lossless jpeg + * jpegdec: Implement upstream negotiation + * matroskademux: seeking fixes; better handling of non-finalized files + * matroskademux: better timestamp/duration handling, fix some stuttering A/V + * matroskademux: add "max-gap-time" property to make gap handling configurable + * matroskademux: UTF-8 subtitles may have markup + * matroskamux: do not use unoffical V_MJPEG codec id + * matroskamux: fix segment handling, so we actually use running time + * matroskamux: for streaming files, push tags first + * matroskamux: handle GstForceKeyUnit event + * multifile: new splitfilesrc element to read multiple files as if they were one single file + * multifilesrc: add "loop" property + * multifilesink: handle buffer lists, useful to keep groups of buffers (GOPs) in the same file + * multifilesink: add flag to cut after a force key unit event + * multifilesink: add "max-files" property + * multifilesink: add new 'max-size' mode and "max-file-size" property for switching to the next file based on size + * multifilesink: write stream-headers when switching to the next file in max-size mode + * multipartdemux: Add property to assume a single stream and emit no-more-pads + * multipartmux: Add \r\n to tail of pushed buffers + * navseek: toggle pause/play on space bar + * osxvideo: Fix leak of NSOpenGLPixelFormat object + * pcmadepay,pcmudepay: allow variable sample rate + * pngenc: increase arbitrary resolution limits + * pulse: Drop support for PA versions before 0.9.16 (1.x is recommended) + * pulse: new pulseaudiosink element to handle format changes (not autoplugged yet) + * pulsesink: add support for compressed audio format passthrough (S/PDIF, mp3-over-bluetooth) + * pulsesink: Allow writes in bigger chunks + * pulsesink: Use the extended stream API if available + * pulsesrc: add a "source-output-index" property; implement GstStreamVolume interface + * qtdemux: better fragmented support (avoid adjustment for keyframe seek; mark all audio track samples as keyframe) + * qtdemux: parse embedded ID32 tags; improve bitrate guessing/extraction + * qtdemux: push mode fixes, fix buffered streaming + * qtmux: add direct dirac mapping + * qtmux: calculate average bitrate for streams + * qtmux: fix ctts generation for streams that don't start at 0 timestamps + * qtmux: use GST_TAG_IMAGE for coverart too + * ismlmux: Use iso-fragmented as variant type (useful in connection with encodebin) + * rtph263ppay: implement getcaps following RFC 4629, picks the right annexes + * rtph263ppay: set H263-2000 if thats what the other side wants + * rtph264depay: complete merged AU on marker bit (thus reducing latency) + * rtph264depay: cope with FU-A E bit not being set (caused by buggy payloaders) + * rtph264depay: exclude NALu size from payload length on truncated packets + * rtph264pay: proxy downstream caps restrictions (converting profile-level-id from RTP caps into video/x-h264 style caps) + * rtph264pay: only set the marker bit on the last NALU of a multi-NALU access unit + * rtpjpegpay: add support for H.264 payload in MJPEG container + * rtpjpegpay: fix for "odd" resolutions not a multiple of DCTSIZE + * rtpmp4adepay: fix output buffer timestamps in case of multiple frames + * rtpmp4gdepay: improve bogus interleaved index compensating + * rtpmp4vpay: deprecated send-config property and replace by config-interval + * rtppcmapay/depay: static clock rates on static payloads, dynamic on dynamic + * rtpvrawpay,-depay: RGB video payloading/depayloading fixes + * rtpg722pay: Compensate for clockrate vs. samplerate difference + * rtpbin: allow configurable rtcp stream syncing interval + * rtpbin: new "rtcp-sync" property, alternative inter-stream syncing methods + * rtpjitterbuffer/rtpbin: relax dropping rtcp packets; misc other fixes + * rtpmanager: don't reveal the user's username, hostname or real name by default + * rtpsession: process received Full Intra Requests (FIR) + * rtpsession: add special mode to use FIR as repair as Google does + * rtpsession: send FIR requests in response to key unit requests with all-headers=TRUE + * rtpsession: always send application requested feedback in immediate mode + * rtpsession: put the PLI requests in each RTPSource + * rtpsession: wait longer to timeout SSRC collision + * rtspsrc: implement async network I/O + * rtspsrc: allow sending short RTSP requests to a server + * rtspsrc: configure rtcp interval if provided + * rtspsrc: open on play and pause when not done yet + * shout2send: send video/webm through libshout + * soup: new souphttpclientsink element + * udpsrc: drop dataless UDP packets + * v4l2: take care not to change the current format where appropriate + * v4l2src, v4l2sink: add "norm" property; default to a pixel-aspect-ratio of 1/1 + * v4l2src: do not ignore the highest frame interval or the largest resolution + * v4l2src: handle some feature query failures more gracefully + * videobox: avoid wrapping opaque to transparent + * wavenc: Allow setcaps to be called after a format was negotiated if it's compatible + * ximagesrc: add "xid" and "xname" properties to allow capturing a particular window + * ximagesrc: fallback to non-XShm mode if allocating the XShm image failed + * ximagesrc: clear flags on buffer reuse, so that flags like DISCONT aren't set accidentally Bugs fixed in this release - * 650470 : RTP: Some depayloaders break with malformed rtp packets - * 652144 : [udp] Fix compiler warning on mingw-w64 + * 668320 : rtpmanager: RTCP receiver reports reveal full user name + * 652727 : multifilesrc: add ability to loop + * 657422 : [souphttpsrc] The souphttpsrc plugin doesn't work behind a proxy that requires authentication + * 432612 : [matroskamux] doesn't handle segments correctly + * 541215 : [avimux] Dirac muxing is broken and results in A/V sync issues + * 546932 : [ximagesrc] allow recording of specific window only + * 571400 : RTSP blocks in gst_element_set_state( GST_STATE_PAUSED ) and incorrect url + * 576524 : rtpbin, jitterbuffer: add mode to support for recording RTP streams + * 586450 : [cairotextoverlay] Forward upstream events to both sinkpads + * 595055 : [pulsesrc] Should implement GstStreamVolume interface + * 605834 : directsoundsink: 16-bit audio is always signed while 8-bit is always unsigned + * 610916 : rtspsrc dosen't work under windows + * 614803 : v4l2: add tv-norm property + * 616686 : multipartdemux: add " single-stream " property to emit no-more-pads earlier + * 616936 : [matroskademux] Incorrect display of subtitles with markup + * 619548 : qtdemux: Guess bitrate if only one stream's bitrate is unknown + * 619590 : [matroskademux] Doesn't protect segment and other fields from concurrent changes from different threads + * 620186 : qtdemux: Export max bitrate for AMR-NB/-WB streams + * 622412 : [rtpmp4vpay] remove send-config parameter; obsoleted by config-interval + * 624887 : pitivi playback hangs / errors while playing mov clips on clip change + * 630456 : [ximagesrc] Fallback to non-XShm mode if image allocation fails + * 631430 : [flvdemux] Cannot play .flv to the end + * 632504 : [rtspsrc] reduce or avoid (network) hang during shutdown + * 634093 : RTSP client asks for unicast from multicast only server + * 638300 : v4l2src: make this work more than once in a row + * 639217 : udpsrc: allow skip-first-bytes of full buffer size + * 640323 : [cairotextoverlay] forward new segment events from the sink to the source + * 643847 : deinterlace: Add support for deinterlacing using buffer caps/flags + * 644151 : [multifilesink] Add option to create a new file after each GstForceKeyUnit event + * 644154 : [matroskamux] Force a new cluster after each GstForceKeyUnit event + * 644512 : [auparse] Add seeking + * 647540 : autoaudiosink picks element to use by rank, but pulsesink/alsasink/jackaudiosink are all PRIMARY + * 648312 : [v4l2sink] Unconditionally accepts video/mpegts + * 648642 : rtpsession: Ensure ssrc collisions aren't timed out immediately + * 648937 : matroskademux: avoid building index when streamable + * 649067 : v4l2src: got unexpected frame size of 262254 instead of 614400 + * 649617 : [rtp] Deadlock and other fixes for rtpssrcdemux + * 649780 : flac: seek beyond end fails instead of EOSing immediately + * 649955 : flvmux: add support for mpegversion 2, which is also AAC + * 650258 : matroskademux/matroskaparse: gst_element_query_duration returns wrong value for Matroska files + * 650313 : ac3parse: Add support for iec61937 alignment + * 650503 : [dvdemux] Broken DURATION query handling + * 650555 : [aacparse] AAC profiles needed in caps + * 650691 : [flacparse] regression playing some flac files + * 650714 : [amrparse] skips first few frames (problem in checking sync) + * 650785 : [flacparse] duration query in DEFAULT format failing with flacparse in pipeline (regression) + * 650877 : matroska: refactor code common to matroskademux and matroskaparse + * 650912 : Rare leak in qtdemux + * 650916 : REGRESSION: ssrcdemux causing FLOW_NOT_LINKED + * 650937 : deinterlace: fix parameter type in trace + * 651059 : rtspsrc: uniform unknown message handling + * 651443 : multifilesink: add next-file=max-size mode and max-file-size property + * 652195 : matroskademux: seeking in non-finalized matroska files does not work correctly + * 652286 : matroskaparse: Gstreamer-CRITICAL when changing state from PAUSED to READY + * 652467 : matroska: missing < stdio.h > include for sscanf + * 653080 : matroskamux: make check for block_duration less sensitive + * 653091 : [dv1394src] Make the internal clock thread-safe + * 653327 : configure script for gst-plugins-good selects shout2 when it's not present + * 653559 : aacparse: too greedy minimum frame size + * 653709 : [ximagesrc] sets DISCONT on half the buffers + * 654175 : matroskademux: handle blocks with duration=0 + * 654379 : matroskamux: make default framerate optional per stream + * 654583 : Immediate RTCP in rtpsession + * 654585 : rtpmp4gdepay choppy sound + * 654744 : matroskademux: fix aspect ratio if header has only onle display variable set + * 654749 : goom: unbreak build on PPC on openbsd + * 654816 : [rtspsrc] rtspsrc doesn't get eos if it's wrapped into a bin + * 655530 : Logitech B990 HD Webcam yields poor video in MJPEG mode. + * 655570 : qtdemux: assertion error when playing Apple Trailers + * 655805 : Make the extended RTSP headers optional + * 655866 : jackaudiosink: Don't call g_alloca in jack_process_cb + * 655918 : qtdemux : qtdemux_add_fragmented_samples return error. + * 656104 : v4l2src fails to correctly configure the framerate + * 656606 : crash in gst_spectrum_reset_message_data() + * 656649 : flacparse: fix off by one in frame size check + * 656734 : [aacparse] Assumes 1024 samples per frame + * 657080 : aacparse: failing test due to two buffers being dropped for one sync loss + * 657179 : pulse: New pulseaudiosink element to handle format changes + * 657376 : rtspsrc regression + * 657830 : multiudpsink: make add/remove/clear/get-stats action signals + * 658178 : udpsrc: rough error reporting when using an invalid URI + * 658305 : [souphttpsrc] can’t seek during double speed playback + * 658419 : Add FIR support to rtpsession + * 658543 : [v4l2src] Use GST_RESOURCE_ERROR_BUSY if webcam is already used + * 658546 : ac3parse: RealAudio file with AC-3 audio no longer plays + * 659009 : [matroskademux] property for configuring gap handling + * 659065 : navseek: toggle pause/play on space bar + * 659153 : matroskademux: fix stuttering A/V + * 659237 : [gstrtpbin] clock is not unreffed after finish using it + * 659242 : [matroskademux] Unexpected EOS when seeking on paused matroska file + * 659798 : Segfault when you convert with audioconvert from audio file mkv to audio file avi + * 659808 : matroskademux: misc fixes + * 659837 : matroskamux: unable to mux audio/x-raw-int,rate=8000,channels=1,endianness=1234,width=16,depth=16,signed=true + * 659943 : [ac3parse] it does not correcly check for ac3/e-ac3 switch + * 660249 : won't play wav file: invalid WAV header (no fmt at start): ID32 + * 660275 : jpegdec doesn't implement upstream negotiation + * 660294 : goom2k1: Fix mingw compiler warnings + * 660448 : videomixer2: memory leak + * 660468 : speexenc: fix calculation of filler data size + * 660481 : v4l, ximagesrc: printf format warnings + * 660969 : qtmux memleak + * 661049 : matroskademux: support seek with start_type NONE + * 661215 : flacparse: fix last frame timestamp in fixed block size mode + * 661400 : rtpg722pay: G722 rtptime too fast + * 661477 : flvdemux: negative cts causes uint overflow, resulting in sinks waiting forever + * 661841 : [edgetv] video artifacts if videorate placed after edgetv + * 661874 : aacparse fails to forward caps to encoder + * 662856 : cairotextoverlay: add a 'silent' property to skip rendering + * 663186 : taginject is not gap aware + * 663334 : gst/flv/: add amfdefs.h to noinst_HEADERS + * 663580 : v4l2src negotiation failure with weird pixel-aspect-ratios + * 664548 : matroskaparse: memleak + * 664792 : Staircase effect in M-JPEG over RTP with unaligned image dimensions.. + * 664892 : [matroskademux] Doesn't set caps properly + * 665387 : v4l2src: fix stepwise enumeration ignoring the highest values + * 665412 : matroskamux: jpeg muxing regression + * 665502 : [flvdemux] broken a/v sync for some files + * 665666 : multifilesink: GstMultiFileSinkNext not documented + * 665872 : jackaudiosink, jackaudiosrc: add " client-name " property + * 665882 : gdkpixbufsink: " pixel-aspect-ratio " is the inverse of what it should be + * 665911 : Ability to specify ignore-length in wavparse + * 666361 : playbin2: regression: visualisations don't work with pulseaudiosink + * 666583 : matroskademux: too many bus messages in streamable mode + * 666602 : ac3parse: no valid frames found before end of stream (unexpected bsid=10) + * 666644 : udpsrc: infinite loop on dataless UDP packets + * 666688 : jpedec: peer_caps leak + * 666711 : rtspsrc: hostname lookup is not thread safe + * 667419 : matroskamux memleaks + * 667818 : osxvideo: Fix leak of NSOpenGLPixelFormat object + * 667820 : rtpptdemux: Plug potential pad leak. + * 667846 : rtph264depay: Exclude NALu size from payload length on truncated packets. + * 668648 : gst-plugins-good does not compile: cairo cannot find libgstvideo-0.10 + * 669455 : V4l2src can't open webcamstudio new vloopback + * 669590 : [shout2send] support webm streaming + * 670197 : v4l2src: webcam doesn't work due to fatal error when querying color balance attributes + * 650960 : flacparse makes decoded flac files start at sample offset 9215 + * 659947 : souphttpsink: rename to souphttpclientsink? + * 658659 : qtmux: Fix ctts entries for streams that don't start with timestamps from 0 Download @@ -88,9 +319,61 @@ Applications Contributors to this release + * Alessandro Decina + * Alexey Fisher + * Andoni Morales Alastruey + * Antoine Jacoutot + * Arun Raghavan + * Branko Subasic + * Brian Li + * Chad + * David Henningsson * David Schleef - * Jose Antonio Santos Cadenas - * Pino Toscano + * David Svensson Fors + * Debarshi Ray + * Edward Hervey + * Gary Ching-Pang Lin + * Guillaume Desmottes + * Ha Nguyen + * Havard Graff + * Jan Schmidt + * Jayakrishnan M + * John Ogness + * Jonas Larsson + * Jonny Lamb + * Julien Isorce + * Konstantin Miller + * Lasse Laukkanen + * Marc Leeman + * Mark Nauwelaerts + * Mart Raudsepp + * Miguel Angel Cabrera Moya + * Monty Montgomery + * Nicola Murino + * Nicolas Baron + * Olivier Crête + * Pascal Buhler + * Peter Korsgaard + * Peter Seiderer + * Philip Jägenstedt + * Philippe Normand * Raimo Järvi + * Ralph Giles + * Raul Gutierrez Segales + * René Stadler + * Reynaldo H. Verdejo Pinochet + * Robert Krakora * Sebastian Dröge + * Sebastian Rasmussen + * Sjoerd Simons + * Stas Sergeev + * Stefan Kost + * Stefan Sauer + * Stig Sandnes + * Thiago Santos + * Tim-Philipp Müller + * Tristan Matthews + * Tuukka Pasanen + * Vincent Penquerc'h + * Wim Taymans   \ No newline at end of file From 0ab56771072ebbe5c53d97ce327c2488a4bd1380 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Fri, 24 Feb 2012 10:26:26 +0100 Subject: [PATCH 07/55] update for metadata change --- sys/v4l2/gstv4l2bufferpool.c | 3 +-- sys/ximage/ximageutil.c | 3 +-- 2 files changed, 2 insertions(+), 4 deletions(-) diff --git a/sys/v4l2/gstv4l2bufferpool.c b/sys/v4l2/gstv4l2bufferpool.c index 4e149dd764..7f4fcb645e 100644 --- a/sys/v4l2/gstv4l2bufferpool.c +++ b/sys/v4l2/gstv4l2bufferpool.c @@ -66,8 +66,7 @@ gst_v4l2_meta_get_info (void) meta_info = gst_meta_register ("GstV4l2Meta", "GstV4l2Meta", sizeof (GstV4l2Meta), (GstMetaInitFunction) NULL, - (GstMetaFreeFunction) NULL, (GstMetaCopyFunction) NULL, - (GstMetaTransformFunction) NULL); + (GstMetaFreeFunction) NULL, (GstMetaTransformFunction) NULL); } return meta_info; } diff --git a/sys/ximage/ximageutil.c b/sys/ximage/ximageutil.c index c221a58dc4..47d6d48570 100644 --- a/sys/ximage/ximageutil.c +++ b/sys/ximage/ximageutil.c @@ -32,8 +32,7 @@ gst_meta_ximage_get_info (void) meta_ximage_info = gst_meta_register ("GstMetaXImageSrc", "GstMetaXImageSrc", sizeof (GstMetaXImage), (GstMetaInitFunction) NULL, - (GstMetaFreeFunction) NULL, (GstMetaCopyFunction) NULL, - (GstMetaTransformFunction) NULL); + (GstMetaFreeFunction) NULL, (GstMetaTransformFunction) NULL); } return meta_ximage_info; } From 44828add73a09135afbd4109539ab5c25dcf85ba Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Fri, 24 Feb 2012 11:03:48 +0100 Subject: [PATCH 08/55] update for basetransform change --- gst/alpha/gstalpha.c | 2 +- gst/debugutils/gstcapssetter.c | 2 +- gst/videocrop/gstvideocrop.c | 2 +- gst/videofilter/gstvideoflip.c | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/gst/alpha/gstalpha.c b/gst/alpha/gstalpha.c index 4b53a80ce5..a0c3fab31c 100644 --- a/gst/alpha/gstalpha.c +++ b/gst/alpha/gstalpha.c @@ -393,7 +393,7 @@ gst_alpha_set_property (GObject * object, guint prop_id, } if (reconfigure) - gst_base_transform_reconfigure (GST_BASE_TRANSFORM_CAST (alpha)); + gst_base_transform_reconfigure_src (GST_BASE_TRANSFORM_CAST (alpha)); GST_ALPHA_UNLOCK (alpha); } diff --git a/gst/debugutils/gstcapssetter.c b/gst/debugutils/gstcapssetter.c index 977a502b40..ee065ff03b 100644 --- a/gst/debugutils/gstcapssetter.c +++ b/gst/debugutils/gstcapssetter.c @@ -294,7 +294,7 @@ gst_caps_setter_set_property (GObject * object, guint prop_id, } /* try to activate these new caps next time around */ - gst_base_transform_reconfigure (GST_BASE_TRANSFORM (filter)); + gst_base_transform_reconfigure_src (GST_BASE_TRANSFORM (filter)); break; } case PROP_JOIN: diff --git a/gst/videocrop/gstvideocrop.c b/gst/videocrop/gstvideocrop.c index 8654f90e16..34cec28fa6 100644 --- a/gst/videocrop/gstvideocrop.c +++ b/gst/videocrop/gstvideocrop.c @@ -711,7 +711,7 @@ gst_video_crop_set_property (GObject * object, guint prop_id, video_crop->crop_top); GST_OBJECT_UNLOCK (video_crop); - gst_base_transform_reconfigure (GST_BASE_TRANSFORM (video_crop)); + gst_base_transform_reconfigure_src (GST_BASE_TRANSFORM (video_crop)); g_mutex_unlock (&video_crop->lock); } diff --git a/gst/videofilter/gstvideoflip.c b/gst/videofilter/gstvideoflip.c index 9075a36e5c..35624b1c39 100644 --- a/gst/videofilter/gstvideoflip.c +++ b/gst/videofilter/gstvideoflip.c @@ -965,7 +965,7 @@ gst_video_flip_set_property (GObject * object, guint prop_id, gst_base_transform_set_passthrough (btrans, method == GST_VIDEO_FLIP_METHOD_IDENTITY); - gst_base_transform_reconfigure (btrans); + gst_base_transform_reconfigure_src (btrans); } else { GST_OBJECT_UNLOCK (videoflip); } From f4afccff5c8c8f4163544c40b6e671a5b98d54c3 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Sat, 25 Feb 2012 15:40:39 +0000 Subject: [PATCH 09/55] matroskamux: create streamable output if downstream is not seekable Ignore the "streamable" property setting and create streamable output if downstream is known not to be seekable (as queried via a SEEKABLE query). Fixes pipelines like webmmux ! appsink creating seemingly corrupted output if streamable has not been set to true. --- gst/matroska/matroska-mux.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) diff --git a/gst/matroska/matroska-mux.c b/gst/matroska/matroska-mux.c index d79c2224d1..fb86ef4983 100644 --- a/gst/matroska/matroska-mux.c +++ b/gst/matroska/matroska-mux.c @@ -2355,6 +2355,29 @@ gst_matroska_mux_start (GstMatroskaMux * mux) guint32 segment_uid[4]; GTimeVal time = { 0, 0 }; + /* if not streaming, check if downstream is seekable */ + if (!mux->streamable) { + gboolean seekable; + GstQuery *query; + + query = gst_query_new_seeking (GST_FORMAT_BYTES); + if (gst_pad_peer_query (mux->srcpad, query)) { + gst_query_parse_seeking (query, NULL, &seekable, NULL, NULL); + GST_INFO_OBJECT (mux, "downstream is %sseekable", seekable ? "" : "not "); + if (!seekable) { + mux->streamable = TRUE; + g_object_notify (G_OBJECT (mux), "streamable"); + GST_WARNING_OBJECT (mux, "downstream is not seekable, but " + "streamable=false. Will ignore that and create streamable output " + "instead"); + } + } else { + /* have to assume seeking is supported if query not handled downstream */ + /* FIXME 0.11: change to query not handled => seeking not supported */ + GST_WARNING_OBJECT (mux, "downstream did not handle seeking query"); + } + } + if (!strcmp (mux->doctype, GST_MATROSKA_DOCTYPE_WEBM)) { ebml->caps = gst_caps_new_simple ("video/webm", NULL); } else { From 6b0dd4758657630c6f043593a6b64ccb35b33ae0 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Sat, 25 Feb 2012 15:48:44 +0000 Subject: [PATCH 10/55] flvmux: create streamable output if downstream is not seekable Ignore the "streamable" property setting and create streamable output if downstream is known not to be seekable (as queried via a SEEKABLE query). Fixes pipelines like flvmux ! appsink possibly creating seemingly corrupted output if streamable has not been set to true. --- gst/flv/gstflvmux.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) diff --git a/gst/flv/gstflvmux.c b/gst/flv/gstflvmux.c index c427f529cc..f2b9688cd2 100644 --- a/gst/flv/gstflvmux.c +++ b/gst/flv/gstflvmux.c @@ -1106,6 +1106,29 @@ gst_flv_mux_write_header (GstFlvMux * mux) GSList *l; GstFlowReturn ret; + /* if not streaming, check if downstream is seekable */ + if (!mux->streamable) { + gboolean seekable; + GstQuery *query; + + query = gst_query_new_seeking (GST_FORMAT_BYTES); + if (gst_pad_peer_query (mux->srcpad, query)) { + gst_query_parse_seeking (query, NULL, &seekable, NULL, NULL); + GST_INFO_OBJECT (mux, "downstream is %sseekable", seekable ? "" : "not "); + if (!seekable) { + mux->streamable = TRUE; + g_object_notify (G_OBJECT (mux), "streamable"); + GST_WARNING_OBJECT (mux, "downstream is not seekable, but " + "streamable=false. Will ignore that and create streamable output " + "instead"); + } + } else { + /* have to assume seeking is supported if query not handled downstream */ + /* FIXME 0.11: change to query not handled => seeking not supported */ + GST_WARNING_OBJECT (mux, "downstream did not handle seeking query"); + } + } + header = gst_flv_mux_create_header (mux); metadata = gst_flv_mux_create_metadata (mux, TRUE); video_codec_data = NULL; From 676b371bdb2d80e9beee9c527b92c46214af4550 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Sat, 25 Feb 2012 15:55:15 +0000 Subject: [PATCH 11/55] qtmux: create streamable output if downstream is not seekable Ignore the "streamable" property setting and create streamable output if downstream is known not to be seekable (as queried via a SEEKABLE query). Fixes pipelines like qtmux ! appsink possibly creating seemingly corrupted output if streamable has not been set to true. --- gst/isomp4/gstqtmux.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) diff --git a/gst/isomp4/gstqtmux.c b/gst/isomp4/gstqtmux.c index 2ce98400b3..80fa5afbe4 100644 --- a/gst/isomp4/gstqtmux.c +++ b/gst/isomp4/gstqtmux.c @@ -1613,6 +1613,30 @@ gst_qt_mux_start_file (GstQTMux * qtmux) gst_pad_set_caps (qtmux->srcpad, caps); gst_caps_unref (caps); + /* if not streaming, check if downstream is seekable */ + if (!qtmux->streamable) { + gboolean seekable; + GstQuery *query; + + query = gst_query_new_seeking (GST_FORMAT_BYTES); + if (gst_pad_peer_query (qtmux->srcpad, query)) { + gst_query_parse_seeking (query, NULL, &seekable, NULL, NULL); + GST_INFO_OBJECT (qtmux, "downstream is %sseekable", + seekable ? "" : "not "); + if (!seekable) { + qtmux->streamable = TRUE; + g_object_notify (G_OBJECT (qtmux), "streamable"); + GST_WARNING_OBJECT (qtmux, "downstream is not seekable, but " + "streamable=false. Will ignore that and create streamable output " + "instead"); + } + } else { + /* have to assume seeking is supported if query not handled downstream */ + /* FIXME 0.11: change to query not handled => seeking not supported */ + GST_WARNING_OBJECT (qtmux, "downstream did not handle seeking query"); + } + } + /* let downstream know we think in BYTES and expect to do seeking later on */ gst_pad_push_event (qtmux->srcpad, gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_BYTES, 0, -1, 0)); From 6e96f4c201d2d11a84d16dd52bb083bf8683da24 Mon Sep 17 00:00:00 2001 From: Alessandro Decina Date: Sun, 26 Feb 2012 20:30:24 +0100 Subject: [PATCH 12/55] mpegaudioparse: fix compiler warning --- gst/audioparsers/gstmpegaudioparse.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/gst/audioparsers/gstmpegaudioparse.c b/gst/audioparsers/gstmpegaudioparse.c index a8c68252c6..0fd7cccb31 100644 --- a/gst/audioparsers/gstmpegaudioparse.c +++ b/gst/audioparsers/gstmpegaudioparse.c @@ -146,8 +146,7 @@ gst_mpeg_audio_parse_base_init (gpointer klass) { GstElementClass *element_class = GST_ELEMENT_CLASS (klass); - gst_element_class_add_static_pad_template (element_class, - &sink_template); + gst_element_class_add_static_pad_template (element_class, &sink_template); gst_element_class_add_static_pad_template (element_class, &src_template); gst_element_class_set_details_simple (element_class, "MPEG1 Audio Parser", @@ -573,7 +572,7 @@ gst_mp3parse_find_freerate (GstMpegAudioParse * mp3parse, GstBuffer * buf, } if (valid) { - GST_LOG_OBJECT (mp3parse, "calculated rate %d", rate * 1000); + GST_LOG_OBJECT (mp3parse, "calculated rate %lu", rate * 1000); if (rate < 8 || (layer == 3 && rate > 640)) { GST_DEBUG_OBJECT (mp3parse, "rate invalid"); if (rate < 8) { From 583342271f3eec9b2c882006736f44c9c75f0b9a Mon Sep 17 00:00:00 2001 From: Alessandro Decina Date: Sun, 26 Feb 2012 20:39:52 +0100 Subject: [PATCH 13/55] goom2k1: fix compiler warning --- gst/goom2k1/gstgoom.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/gst/goom2k1/gstgoom.c b/gst/goom2k1/gstgoom.c index 4421b17bb8..96c11e5dac 100644 --- a/gst/goom2k1/gstgoom.c +++ b/gst/goom2k1/gstgoom.c @@ -508,7 +508,7 @@ gst_goom_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer) } GST_DEBUG_OBJECT (goom, - "Input buffer has %d samples, time=%" G_GUINT64_FORMAT, + "Input buffer has %" G_GSIZE_FORMAT " samples, time=%" G_GUINT64_FORMAT, gst_buffer_get_size (buffer) / goom->bps, GST_BUFFER_TIMESTAMP (buffer)); /* Collect samples until we have enough for an output frame */ From 4ba15ca5d264671eef0da83e81588103e774fec2 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Mon, 27 Feb 2012 00:55:38 +0000 Subject: [PATCH 14/55] mpegaudioparse: fix up after merge --- gst/audioparsers/gstmpegaudioparse.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/gst/audioparsers/gstmpegaudioparse.c b/gst/audioparsers/gstmpegaudioparse.c index d087b2c51b..6c5614b4d5 100644 --- a/gst/audioparsers/gstmpegaudioparse.c +++ b/gst/audioparsers/gstmpegaudioparse.c @@ -497,7 +497,7 @@ gst_mpeg_audio_parse_head_check (GstMpegAudioParse * mp3parse, * If not enough data, returns FALSE. */ static gboolean -gst_mp3parse_find_freerate (GstMpegAudioParse * mp3parse, GstBuffer * buf, +gst_mp3parse_find_freerate (GstMpegAudioParse * mp3parse, GstMapInfo * map, guint32 header, gboolean at_eos, gint * _rate) { guint32 next_header; @@ -508,8 +508,8 @@ gst_mp3parse_find_freerate (GstMpegAudioParse * mp3parse, GstBuffer * buf, gboolean valid; gint lsf, mpg25; - available = GST_BUFFER_SIZE (buf); - data = GST_BUFFER_DATA (buf); + available = map->size; + data = map->data; *_rate = 0; @@ -665,9 +665,10 @@ gst_mpeg_audio_parse_handle_frame (GstBaseParse * parse, GST_LOG_OBJECT (mp3parse, "possibly free format"); if (lost_sync || mp3parse->freerate == 0) { GST_DEBUG_OBJECT (mp3parse, "finding free format rate"); - if (!gst_mp3parse_find_freerate (mp3parse, buf, header, draining, &valid)) { + if (!gst_mp3parse_find_freerate (mp3parse, &map, header, draining, + &valid)) { /* not enough data */ - *framesize = G_MAXUINT; + gst_base_parse_set_min_frame_size (parse, valid); *skipsize = 0; return FALSE; } else { From cc0511f5d80ef418c26d89ff3121f3375cc5ef69 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Mon, 27 Feb 2012 01:09:11 +0000 Subject: [PATCH 15/55] flvmux, matroskamux, qtmux: if in doubt about downstream seekability default to streaming=true If downstream didn't answer our SEEKING query and told us it's seekable, default to streaming=true. We couldn't do this in 0.10 for backwards compatibility reasons, but we can in 0.11. Play it safe. --- gst/flv/gstflvmux.c | 16 ++++++++-------- gst/isomp4/gstqtmux.c | 16 ++++++++-------- gst/matroska/matroska-mux.c | 16 ++++++++-------- 3 files changed, 24 insertions(+), 24 deletions(-) diff --git a/gst/flv/gstflvmux.c b/gst/flv/gstflvmux.c index 554cae9b50..8119e9db33 100644 --- a/gst/flv/gstflvmux.c +++ b/gst/flv/gstflvmux.c @@ -1153,17 +1153,17 @@ gst_flv_mux_write_header (GstFlvMux * mux) if (gst_pad_peer_query (mux->srcpad, query)) { gst_query_parse_seeking (query, NULL, &seekable, NULL, NULL); GST_INFO_OBJECT (mux, "downstream is %sseekable", seekable ? "" : "not "); - if (!seekable) { - mux->streamable = TRUE; - g_object_notify (G_OBJECT (mux), "streamable"); - GST_WARNING_OBJECT (mux, "downstream is not seekable, but " - "streamable=false. Will ignore that and create streamable output " - "instead"); - } } else { /* have to assume seeking is supported if query not handled downstream */ - /* FIXME 0.11: change to query not handled => seeking not supported */ GST_WARNING_OBJECT (mux, "downstream did not handle seeking query"); + seekable = FALSE; + } + if (!seekable) { + mux->streamable = TRUE; + g_object_notify (G_OBJECT (mux), "streamable"); + GST_WARNING_OBJECT (mux, "downstream is not seekable, but " + "streamable=false. Will ignore that and create streamable output " + "instead"); } } diff --git a/gst/isomp4/gstqtmux.c b/gst/isomp4/gstqtmux.c index e51f48c3bc..9e567637ec 100644 --- a/gst/isomp4/gstqtmux.c +++ b/gst/isomp4/gstqtmux.c @@ -1639,17 +1639,17 @@ gst_qt_mux_start_file (GstQTMux * qtmux) gst_query_parse_seeking (query, NULL, &seekable, NULL, NULL); GST_INFO_OBJECT (qtmux, "downstream is %sseekable", seekable ? "" : "not "); - if (!seekable) { - qtmux->streamable = TRUE; - g_object_notify (G_OBJECT (qtmux), "streamable"); - GST_WARNING_OBJECT (qtmux, "downstream is not seekable, but " - "streamable=false. Will ignore that and create streamable output " - "instead"); - } } else { /* have to assume seeking is supported if query not handled downstream */ - /* FIXME 0.11: change to query not handled => seeking not supported */ GST_WARNING_OBJECT (qtmux, "downstream did not handle seeking query"); + seekable = FALSE; + } + if (!seekable) { + qtmux->streamable = TRUE; + g_object_notify (G_OBJECT (qtmux), "streamable"); + GST_WARNING_OBJECT (qtmux, "downstream is not seekable, but " + "streamable=false. Will ignore that and create streamable output " + "instead"); } } diff --git a/gst/matroska/matroska-mux.c b/gst/matroska/matroska-mux.c index e7c42f348e..4b5c8f6afd 100644 --- a/gst/matroska/matroska-mux.c +++ b/gst/matroska/matroska-mux.c @@ -2349,17 +2349,17 @@ gst_matroska_mux_start (GstMatroskaMux * mux) if (gst_pad_peer_query (mux->srcpad, query)) { gst_query_parse_seeking (query, NULL, &seekable, NULL, NULL); GST_INFO_OBJECT (mux, "downstream is %sseekable", seekable ? "" : "not "); - if (!seekable) { - mux->streamable = TRUE; - g_object_notify (G_OBJECT (mux), "streamable"); - GST_WARNING_OBJECT (mux, "downstream is not seekable, but " - "streamable=false. Will ignore that and create streamable output " - "instead"); - } } else { /* have to assume seeking is supported if query not handled downstream */ - /* FIXME 0.11: change to query not handled => seeking not supported */ GST_WARNING_OBJECT (mux, "downstream did not handle seeking query"); + seekable = FALSE; + } + if (!seekable) { + mux->streamable = TRUE; + g_object_notify (G_OBJECT (mux), "streamable"); + GST_WARNING_OBJECT (mux, "downstream is not seekable, but " + "streamable=false. Will ignore that and create streamable output " + "instead"); } } From c17aa1565a4337888310bcc45a2722ae7b65748d Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Mon, 27 Feb 2012 09:14:04 +0100 Subject: [PATCH 16/55] v4l2: use public api instead of poking into the private structures of the base class --- 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 7f4fcb645e..10f29336ed 100644 --- a/sys/v4l2/gstv4l2bufferpool.c +++ b/sys/v4l2/gstv4l2bufferpool.c @@ -863,6 +863,7 @@ GstBufferPool * gst_v4l2_buffer_pool_new (GstV4l2Object * obj, GstCaps * caps) { GstV4l2BufferPool *pool; + GstStructure *s; gint fd; fd = v4l2_dup (obj->video_fd); @@ -873,8 +874,9 @@ gst_v4l2_buffer_pool_new (GstV4l2Object * obj, GstCaps * caps) pool->video_fd = fd; pool->obj = obj; - gst_buffer_pool_config_set (GST_BUFFER_POOL_CAST (pool)->config, caps, - obj->sizeimage, 2, 0, 0, 0); + s = gst_buffer_pool_get_config (GST_BUFFER_POOL_CAST (pool)); + gst_buffer_pool_config_set (s, caps, obj->sizeimage, 2, 0, 0, 0); + gst_buffer_pool_set_config (GST_BUFFER_POOL_CAST (pool), s); return GST_BUFFER_POOL (pool); From a51ce46d907d960c0dfc011cbdfb47eb9606fee5 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Mon, 27 Feb 2012 13:05:33 +0100 Subject: [PATCH 17/55] flacenc: fix event handling Fix dodgy segment event handling Chain up to parent event handler --- ext/flac/gstflacenc.c | 38 +++----------------------------------- 1 file changed, 3 insertions(+), 35 deletions(-) diff --git a/ext/flac/gstflacenc.c b/ext/flac/gstflacenc.c index 41d149296f..c63f7a8b6c 100644 --- a/ext/flac/gstflacenc.c +++ b/ext/flac/gstflacenc.c @@ -1101,43 +1101,9 @@ gst_flac_enc_sink_event (GstAudioEncoder * enc, GstEvent * event) GST_DEBUG ("Received %s event on sinkpad", GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { - case GST_EVENT_SEGMENT:{ - GstSegment seg; - gint64 start, stream_time; - - if (flacenc->offset == 0) { - gst_event_copy_segment (event, &seg); - start = seg.start; - stream_time = seg.time; - } else { - start = -1; - stream_time = -1; - } - - if (start > 0) { - if (flacenc->offset > 0) - GST_DEBUG ("Not handling mid-stream newsegment event"); - else - GST_DEBUG ("Not handling newsegment event with non-zero start"); - } else { - GstEvent *e; - - gst_segment_init (&seg, GST_FORMAT_BYTES); - e = gst_event_new_segment (&seg); - ret = gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (enc), e); - } - - if (stream_time > 0) { - GST_DEBUG ("Not handling non-zero stream time"); - } - - /* don't push it downstream, we'll generate our own via seek to 0 */ - gst_event_unref (event); - ret = TRUE; - break; - } case GST_EVENT_EOS: flacenc->eos = TRUE; + ret = GST_AUDIO_ENCODER_CLASS (parent_class)->event (enc, event); break; case GST_EVENT_TAG: if (flacenc->tags) { @@ -1147,8 +1113,10 @@ gst_flac_enc_sink_event (GstAudioEncoder * enc, GstEvent * event) } else { g_assert_not_reached (); } + ret = GST_AUDIO_ENCODER_CLASS (parent_class)->event (enc, event); break; default: + ret = GST_AUDIO_ENCODER_CLASS (parent_class)->event (enc, event); break; } From 36e6b25e73aee3eef370c122bd5f5ae71b7d4bd0 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Mon, 27 Feb 2012 13:09:31 +0100 Subject: [PATCH 18/55] speexenc: chain up to parent event handler --- ext/speex/gstspeexenc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/ext/speex/gstspeexenc.c b/ext/speex/gstspeexenc.c index 9023ec5e82..32e39fb6ce 100644 --- a/ext/speex/gstspeexenc.c +++ b/ext/speex/gstspeexenc.c @@ -530,7 +530,7 @@ gst_speex_enc_sink_event (GstAudioEncoder * benc, GstEvent * event) } /* we only peeked, let base class handle it */ - return FALSE; + return GST_AUDIO_ENCODER_CLASS (parent_class)->event (benc, event); } static GstFlowReturn From 9beda57c3a81eb920324654fdb50f5610720f999 Mon Sep 17 00:00:00 2001 From: Edward Hervey Date: Mon, 27 Feb 2012 14:47:25 +0100 Subject: [PATCH 19/55] Suppress deprecation warnings in selected files, for g_value_array_* mostly --- ext/annodex/gstannodex.c | 4 ++++ ext/annodex/gstcmmlparser.c | 4 ++++ ext/annodex/gstcmmltag.c | 4 ++++ ext/pulse/pulseprobe.c | 4 ++++ gst/audiofx/audiofirfilter.c | 4 ++++ gst/audiofx/audioiirfilter.c | 4 ++++ gst/interleave/interleave.c | 4 ++++ gst/rtpmanager/rtpsession.c | 4 ++++ gst/udp/gstdynudpsink.c | 4 ++++ gst/udp/gstmultiudpsink.c | 4 ++++ sys/oss4/oss4-audio.c | 4 ++++ sys/oss4/oss4-property-probe.c | 4 ++++ sys/v4l2/gstv4l2object.c | 4 ++++ tests/check/elements/audiofirfilter.c | 4 ++++ tests/check/elements/audioiirfilter.c | 5 +++++ tests/check/elements/cmmldec.c | 4 ++++ tests/check/elements/interleave.c | 4 ++++ tests/check/pipelines/wavenc.c | 4 ++++ tests/examples/audiofx/firfilter-example.c | 4 ++++ tests/examples/audiofx/iirfilter-example.c | 4 ++++ tests/examples/pulse/pulse.c | 4 ++++ tests/examples/rtp/server-alsasrc-PCMA.c | 4 ++++ tests/examples/v4l2/probe.c | 4 ++++ tests/icles/test-oss4.c | 4 ++++ 24 files changed, 97 insertions(+) diff --git a/ext/annodex/gstannodex.c b/ext/annodex/gstannodex.c index 05e10c8e44..72ea88d090 100644 --- a/ext/annodex/gstannodex.c +++ b/ext/annodex/gstannodex.c @@ -21,6 +21,10 @@ * Boston, MA 02111-1307, USA. */ +/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #ifdef HAVE_CONFIG_H #include "config.h" #endif diff --git a/ext/annodex/gstcmmlparser.c b/ext/annodex/gstcmmlparser.c index 0e2f7cd8e3..b76cf4a0bd 100644 --- a/ext/annodex/gstcmmlparser.c +++ b/ext/annodex/gstcmmlparser.c @@ -21,6 +21,10 @@ * Boston, MA 02111-1307, USA. */ +/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #include #include #include diff --git a/ext/annodex/gstcmmltag.c b/ext/annodex/gstcmmltag.c index 2cf5d5fef1..8b2a1893e0 100644 --- a/ext/annodex/gstcmmltag.c +++ b/ext/annodex/gstcmmltag.c @@ -21,6 +21,10 @@ * Boston, MA 02111-1307, USA. */ +/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #ifdef HAVE_CONFIG_H #include "config.h" #endif diff --git a/ext/pulse/pulseprobe.c b/ext/pulse/pulseprobe.c index 6ebbfb214f..8024b5f6dc 100644 --- a/ext/pulse/pulseprobe.c +++ b/ext/pulse/pulseprobe.c @@ -21,6 +21,10 @@ * USA. */ +/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #ifdef HAVE_CONFIG_H #include "config.h" #endif diff --git a/gst/audiofx/audiofirfilter.c b/gst/audiofx/audiofirfilter.c index bd5a51be4f..e693b4396a 100644 --- a/gst/audiofx/audiofirfilter.c +++ b/gst/audiofx/audiofirfilter.c @@ -45,6 +45,10 @@ * */ +/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #ifdef HAVE_CONFIG_H #include "config.h" #endif diff --git a/gst/audiofx/audioiirfilter.c b/gst/audiofx/audioiirfilter.c index ff5f9caec4..48ead7a300 100644 --- a/gst/audiofx/audioiirfilter.c +++ b/gst/audiofx/audioiirfilter.c @@ -41,6 +41,10 @@ * */ +/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #ifdef HAVE_CONFIG_H #include "config.h" #endif diff --git a/gst/interleave/interleave.c b/gst/interleave/interleave.c index 8815e1c049..020d576640 100644 --- a/gst/interleave/interleave.c +++ b/gst/interleave/interleave.c @@ -57,6 +57,10 @@ * */ +/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #ifdef HAVE_CONFIG_H # include "config.h" #endif diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c index 9f15bd698d..24e752b1c1 100644 --- a/gst/rtpmanager/rtpsession.c +++ b/gst/rtpmanager/rtpsession.c @@ -17,6 +17,10 @@ * Boston, MA 02111-1307, USA. */ +/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #include #include diff --git a/gst/udp/gstdynudpsink.c b/gst/udp/gstdynudpsink.c index 18c58ca109..a92cb6b6b1 100644 --- a/gst/udp/gstdynudpsink.c +++ b/gst/udp/gstdynudpsink.c @@ -19,6 +19,10 @@ * Boston, MA 02111-1307, USA. */ +/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #ifdef HAVE_CONFIG_H #include "config.h" #endif diff --git a/gst/udp/gstmultiudpsink.c b/gst/udp/gstmultiudpsink.c index b65ab8c268..2a4c41c2b9 100644 --- a/gst/udp/gstmultiudpsink.c +++ b/gst/udp/gstmultiudpsink.c @@ -27,6 +27,10 @@ * It can be combined with rtp payload encoders to implement RTP streaming. */ +/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #ifdef HAVE_CONFIG_H #include "config.h" #endif diff --git a/sys/oss4/oss4-audio.c b/sys/oss4/oss4-audio.c index 09f8495963..6317400dc5 100644 --- a/sys/oss4/oss4-audio.c +++ b/sys/oss4/oss4-audio.c @@ -17,6 +17,10 @@ * Boston, MA 02111-1307, USA. */ +/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #ifdef HAVE_CONFIG_H #include "config.h" #endif diff --git a/sys/oss4/oss4-property-probe.c b/sys/oss4/oss4-property-probe.c index a99410ea27..5674da50a4 100644 --- a/sys/oss4/oss4-property-probe.c +++ b/sys/oss4/oss4-property-probe.c @@ -17,6 +17,10 @@ * Boston, MA 02111-1307, USA. */ +/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #ifdef HAVE_CONFIG_H #include #endif diff --git a/sys/v4l2/gstv4l2object.c b/sys/v4l2/gstv4l2object.c index 81d1cb7823..6ff5ab4a47 100644 --- a/sys/v4l2/gstv4l2object.c +++ b/sys/v4l2/gstv4l2object.c @@ -18,6 +18,10 @@ * USA. */ +/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #ifdef HAVE_CONFIG_H #include #endif diff --git a/tests/check/elements/audiofirfilter.c b/tests/check/elements/audiofirfilter.c index c4eb723824..3741061bd8 100644 --- a/tests/check/elements/audiofirfilter.c +++ b/tests/check/elements/audiofirfilter.c @@ -18,6 +18,10 @@ * 02110-1301 USA */ +/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #include #include diff --git a/tests/check/elements/audioiirfilter.c b/tests/check/elements/audioiirfilter.c index d144792ba6..dd6403126f 100644 --- a/tests/check/elements/audioiirfilter.c +++ b/tests/check/elements/audioiirfilter.c @@ -18,6 +18,11 @@ * 02110-1301 USA */ +/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + + #include #include diff --git a/tests/check/elements/cmmldec.c b/tests/check/elements/cmmldec.c index a6b8559d5e..142a9042ec 100644 --- a/tests/check/elements/cmmldec.c +++ b/tests/check/elements/cmmldec.c @@ -21,6 +21,10 @@ * Boston, MA 02111-1307, USA. */ +/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #include #include diff --git a/tests/check/elements/interleave.c b/tests/check/elements/interleave.c index 1497a49c98..b6452270c8 100644 --- a/tests/check/elements/interleave.c +++ b/tests/check/elements/interleave.c @@ -18,6 +18,10 @@ * Boston, MA 02111-1307, USA. */ +/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #ifdef HAVE_CONFIG_H # include "config.h" #endif diff --git a/tests/check/pipelines/wavenc.c b/tests/check/pipelines/wavenc.c index c5344a75d0..4bf76a09a9 100644 --- a/tests/check/pipelines/wavenc.c +++ b/tests/check/pipelines/wavenc.c @@ -20,6 +20,10 @@ * Boston, MA 02111-1307, USA. */ +/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #include #include diff --git a/tests/examples/audiofx/firfilter-example.c b/tests/examples/audiofx/firfilter-example.c index b344e74e63..e2fa2dc25b 100644 --- a/tests/examples/audiofx/firfilter-example.c +++ b/tests/examples/audiofx/firfilter-example.c @@ -21,6 +21,10 @@ * by transforming the frequency response to the filter kernel. */ +/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #include #include diff --git a/tests/examples/audiofx/iirfilter-example.c b/tests/examples/audiofx/iirfilter-example.c index 7fac2ac92b..708bde1e1d 100644 --- a/tests/examples/audiofx/iirfilter-example.c +++ b/tests/examples/audiofx/iirfilter-example.c @@ -23,6 +23,10 @@ * of the IIR filter that is used. */ +/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #include #include diff --git a/tests/examples/pulse/pulse.c b/tests/examples/pulse/pulse.c index 54c1a872c3..a0bd3b314c 100644 --- a/tests/examples/pulse/pulse.c +++ b/tests/examples/pulse/pulse.c @@ -17,6 +17,10 @@ * Boston, MA 02111-1307, USA. */ +/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #include #include diff --git a/tests/examples/rtp/server-alsasrc-PCMA.c b/tests/examples/rtp/server-alsasrc-PCMA.c index d61ec30cb9..da1660330c 100644 --- a/tests/examples/rtp/server-alsasrc-PCMA.c +++ b/tests/examples/rtp/server-alsasrc-PCMA.c @@ -17,6 +17,10 @@ * Boston, MA 02111-1307, USA. */ +/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #include #include diff --git a/tests/examples/v4l2/probe.c b/tests/examples/v4l2/probe.c index 169288a489..3642bb64b5 100644 --- a/tests/examples/v4l2/probe.c +++ b/tests/examples/v4l2/probe.c @@ -17,6 +17,10 @@ * Boston, MA 02111-1307, USA. */ +/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #include #include #include diff --git a/tests/icles/test-oss4.c b/tests/icles/test-oss4.c index 233e891fd9..63aeca4e08 100644 --- a/tests/icles/test-oss4.c +++ b/tests/icles/test-oss4.c @@ -17,6 +17,10 @@ * Boston, MA 02111-1307, USA. */ +/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #ifdef HAVE_CONFIG_H #include #endif From b863df570f0fc6f1e868b40c7d97f19c003e93a5 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Mon, 27 Feb 2012 23:45:54 +0100 Subject: [PATCH 20/55] wavpackenc: port to audioencoder Also adjust unit test to slightly modified behaviour. --- ext/wavpack/gstwavpackenc.c | 478 +++++++++++++----------------- ext/wavpack/gstwavpackenc.h | 7 +- tests/check/elements/wavpackenc.c | 13 +- 3 files changed, 211 insertions(+), 287 deletions(-) diff --git a/ext/wavpack/gstwavpackenc.c b/ext/wavpack/gstwavpackenc.c index a22dd23a34..e2e19d4d38 100644 --- a/ext/wavpack/gstwavpackenc.c +++ b/ext/wavpack/gstwavpackenc.c @@ -55,12 +55,18 @@ #include "gstwavpackenc.h" #include "gstwavpackcommon.h" -static GstFlowReturn gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buffer); -static gboolean gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps); +static gboolean gst_wavpack_enc_start (GstAudioEncoder * enc); +static gboolean gst_wavpack_enc_stop (GstAudioEncoder * enc); +static gboolean gst_wavpack_enc_set_format (GstAudioEncoder * enc, + GstAudioInfo * info); +static GstFlowReturn gst_wavpack_enc_handle_frame (GstAudioEncoder * enc, + GstBuffer * in_buf); +static gboolean gst_wavpack_enc_sink_event (GstAudioEncoder * enc, + GstEvent * event); + static int gst_wavpack_enc_push_block (void *id, void *data, int32_t count); -static gboolean gst_wavpack_enc_sink_event (GstPad * pad, GstEvent * event); -static GstStateChangeReturn gst_wavpack_enc_change_state (GstElement * element, - GstStateChange transition); +static GstFlowReturn gst_wavpack_enc_drain (GstWavpackEnc * enc); + static void gst_wavpack_enc_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_wavpack_enc_get_property (GObject * object, guint prop_id, @@ -86,7 +92,7 @@ static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-int, " "width = (int) 32, " - "depth = (int) [ 1, 32], " + "depth = (int) { 24, 32 }, " "endianness = (int) BYTE_ORDER, " "channels = (int) [ 1, 8 ], " "rate = (int) [ 6000, 192000 ]," "signed = (boolean) TRUE") @@ -196,21 +202,8 @@ gst_wavpack_enc_joint_stereo_mode_get_type (void) return qtype; } -static void -_do_init (GType object_type) -{ - const GInterfaceInfo preset_interface_info = { - NULL, /* interface_init */ - NULL, /* interface_finalize */ - NULL /* interface_data */ - }; - - g_type_add_interface_static (object_type, GST_TYPE_PRESET, - &preset_interface_info); -} - -GST_BOILERPLATE_FULL (GstWavpackEnc, gst_wavpack_enc, GstElement, - GST_TYPE_ELEMENT, _do_init); +GST_BOILERPLATE (GstWavpackEnc, gst_wavpack_enc, GstAudioEncoder, + GST_TYPE_AUDIO_ENCODER); static void gst_wavpack_enc_base_init (gpointer klass) @@ -220,8 +213,7 @@ gst_wavpack_enc_base_init (gpointer klass) /* add pad templates */ gst_element_class_add_static_pad_template (element_class, &sink_factory); gst_element_class_add_static_pad_template (element_class, &src_factory); - gst_element_class_add_static_pad_template (element_class, - &wvcsrc_factory); + gst_element_class_add_static_pad_template (element_class, &wvcsrc_factory); /* set element details */ gst_element_class_set_details_simple (element_class, "Wavpack audio encoder", @@ -230,23 +222,24 @@ gst_wavpack_enc_base_init (gpointer klass) "Sebastian Dröge "); } - static void gst_wavpack_enc_class_init (GstWavpackEncClass * klass) { GObjectClass *gobject_class = (GObjectClass *) klass; - GstElementClass *gstelement_class = (GstElementClass *) klass; + GstAudioEncoderClass *base_class = (GstAudioEncoderClass *) (klass); parent_class = g_type_class_peek_parent (klass); - /* set state change handler */ - gstelement_class->change_state = - GST_DEBUG_FUNCPTR (gst_wavpack_enc_change_state); - /* set property handlers */ gobject_class->set_property = gst_wavpack_enc_set_property; gobject_class->get_property = gst_wavpack_enc_get_property; + base_class->start = GST_DEBUG_FUNCPTR (gst_wavpack_enc_start); + base_class->stop = GST_DEBUG_FUNCPTR (gst_wavpack_enc_stop); + base_class->set_format = GST_DEBUG_FUNCPTR (gst_wavpack_enc_set_format); + base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_wavpack_enc_handle_frame); + base_class->event = GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_event); + /* install all properties */ g_object_class_install_property (gobject_class, ARG_MODE, g_param_spec_enum ("mode", "Encoding mode", @@ -304,6 +297,9 @@ gst_wavpack_enc_reset (GstWavpackEnc * enc) g_checksum_free (enc->md5_context); enc->md5_context = NULL; } + if (enc->pending_segment) + gst_event_unref (enc->pending_segment); + enc->pending_segment = NULL; if (enc->pending_buffer) { gst_buffer_unref (enc->pending_buffer); @@ -330,18 +326,7 @@ gst_wavpack_enc_reset (GstWavpackEnc * enc) static void gst_wavpack_enc_init (GstWavpackEnc * enc, GstWavpackEncClass * gclass) { - enc->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink"); - gst_pad_set_setcaps_function (enc->sinkpad, - GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_set_caps)); - gst_pad_set_chain_function (enc->sinkpad, - GST_DEBUG_FUNCPTR (gst_wavpack_enc_chain)); - gst_pad_set_event_function (enc->sinkpad, - GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_event)); - gst_element_add_pad (GST_ELEMENT (enc), enc->sinkpad); - - /* setup src pad */ - enc->srcpad = gst_pad_new_from_static_template (&src_factory, "src"); - gst_element_add_pad (GST_ELEMENT (enc), enc->srcpad); + GstAudioEncoder *benc = GST_AUDIO_ENCODER (enc); /* initialize object attributes */ enc->wp_config = NULL; @@ -365,37 +350,51 @@ gst_wavpack_enc_init (GstWavpackEnc * enc, GstWavpackEncClass * gclass) enc->md5 = FALSE; enc->extra_processing = 0; enc->joint_stereo_mode = GST_WAVPACK_JS_MODE_AUTO; + + /* require perfect ts */ + gst_audio_encoder_set_perfect_timestamp (benc, TRUE); +} + + +static gboolean +gst_wavpack_enc_start (GstAudioEncoder * enc) +{ + GST_DEBUG_OBJECT (enc, "start"); + + return TRUE; } static gboolean -gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps) +gst_wavpack_enc_stop (GstAudioEncoder * enc) { - GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad)); - GstStructure *structure = gst_caps_get_structure (caps, 0); + GstWavpackEnc *wpenc = GST_WAVPACK_ENC (enc); + + GST_DEBUG_OBJECT (enc, "stop"); + gst_wavpack_enc_reset (wpenc); + + return TRUE; +} + +static gboolean +gst_wavpack_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info) +{ + GstWavpackEnc *enc = GST_WAVPACK_ENC (benc); GstAudioChannelPosition *pos; + GstCaps *caps; - if (!gst_structure_get_int (structure, "channels", &enc->channels) || - !gst_structure_get_int (structure, "rate", &enc->samplerate) || - !gst_structure_get_int (structure, "depth", &enc->depth)) { - GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL), - ("got invalid caps: %" GST_PTR_FORMAT, caps)); - gst_object_unref (enc); - return FALSE; - } + /* we may be configured again, but that change should have cleanup context */ + g_assert (enc->wp_context == NULL); + + enc->channels = GST_AUDIO_INFO_CHANNELS (info); + enc->depth = GST_AUDIO_INFO_DEPTH (info); + enc->samplerate = GST_AUDIO_INFO_RATE (info); + + pos = info->position; + g_assert (pos); - pos = gst_audio_get_channel_positions (structure); /* If one channel is NONE they'll be all undefined */ if (pos != NULL && pos[0] == GST_AUDIO_CHANNEL_POSITION_NONE) { - g_free (pos); - pos = NULL; - } - - if (pos == NULL) { - GST_ELEMENT_ERROR (enc, STREAM, FORMAT, (NULL), - ("input has no valid channel layout")); - - gst_object_unref (enc); - return FALSE; + goto invalid_channels; } enc->channel_mask = @@ -403,7 +402,6 @@ gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps) enc->need_channel_remap = gst_wavpack_set_channel_mapping (pos, enc->channels, enc->channel_mapping); - g_free (pos); /* set fixed src pad caps now that we know what we will get */ caps = gst_caps_new_simple ("audio/x-wavpack", @@ -414,18 +412,28 @@ gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps) if (!gst_wavpack_set_channel_layout (caps, enc->channel_mask)) GST_WARNING_OBJECT (enc, "setting channel layout failed"); - if (!gst_pad_set_caps (enc->srcpad, caps)) { - GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL), - ("setting caps failed: %" GST_PTR_FORMAT, caps)); - gst_caps_unref (caps); - gst_object_unref (enc); - return FALSE; - } - gst_pad_use_fixed_caps (enc->srcpad); + if (!gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc), caps)) + goto setting_src_caps_failed; gst_caps_unref (caps); - gst_object_unref (enc); + + /* no special feedback to base class; should provide all available samples */ + return TRUE; + + /* ERRORS */ +setting_src_caps_failed: + { + GST_DEBUG_OBJECT (enc, + "Couldn't set caps on source pad: %" GST_PTR_FORMAT, caps); + gst_caps_unref (caps); + return FALSE; + } +invalid_channels: + { + GST_DEBUG_OBJECT (enc, "input has invalid channel layout"); + return FALSE; + } } static void @@ -547,21 +555,14 @@ gst_wavpack_enc_push_block (void *id, void *data, int32_t count) GstBuffer *buffer; GstPad *pad; guchar *block = (guchar *) data; + gint samples = 0; - pad = (wid->correction) ? enc->wvcsrcpad : enc->srcpad; + pad = (wid->correction) ? enc->wvcsrcpad : GST_AUDIO_ENCODER_SRC_PAD (enc); flow = (wid->correction) ? &enc->wvcsrcpad_last_return : &enc-> srcpad_last_return; - *flow = gst_pad_alloc_buffer_and_set_caps (pad, GST_BUFFER_OFFSET_NONE, - count, GST_PAD_CAPS (pad), &buffer); - - if (*flow != GST_FLOW_OK) { - GST_WARNING_OBJECT (enc, "flow on %s:%s = %s", - GST_DEBUG_PAD_NAME (pad), gst_flow_get_name (*flow)); - return FALSE; - } - + buffer = gst_buffer_new_and_alloc (count); g_memmove (GST_BUFFER_DATA (buffer), block, count); if (count > sizeof (WavpackHeader) && memcmp (block, "wvpk", 4) == 0) { @@ -597,12 +598,14 @@ gst_wavpack_enc_push_block (void *id, void *data, int32_t count) enc->pending_buffer = NULL; enc->pending_offset = 0; - /* if it's the first wavpack block, send a NEW_SEGMENT event */ - if (wph.block_index == 0) { - gst_pad_push_event (pad, - gst_event_new_new_segment (FALSE, - 1.0, GST_FORMAT_TIME, 0, GST_BUFFER_OFFSET_NONE, 0)); + /* only send segment on correction pad, + * regular pad is handled normally by baseclass */ + if (wid->correction && enc->pending_segment) { + gst_pad_push_event (pad, enc->pending_segment); + enc->pending_segment = NULL; + } + if (wph.block_index == 0) { /* save header for later reference, so we can re-send it later on * EOS with fixed up values for total sample count etc. */ if (enc->first_block == NULL && !wid->correction) { @@ -612,29 +615,23 @@ gst_wavpack_enc_push_block (void *id, void *data, int32_t count) } } } - - /* set buffer timestamp, duration, offset, offset_end from - * the wavpack header */ - GST_BUFFER_TIMESTAMP (buffer) = enc->timestamp_offset + - gst_util_uint64_scale_int (GST_SECOND, wph.block_index, - enc->samplerate); - GST_BUFFER_DURATION (buffer) = - gst_util_uint64_scale_int (GST_SECOND, wph.block_samples, - enc->samplerate); - GST_BUFFER_OFFSET (buffer) = wph.block_index; - GST_BUFFER_OFFSET_END (buffer) = wph.block_index + wph.block_samples; + samples = wph.block_samples; } else { /* if it's something else set no timestamp and duration on the buffer */ GST_DEBUG_OBJECT (enc, "got %d bytes of unknown data", count); - - GST_BUFFER_TIMESTAMP (buffer) = GST_CLOCK_TIME_NONE; - GST_BUFFER_DURATION (buffer) = GST_CLOCK_TIME_NONE; } - /* push the buffer and forward errors */ - GST_DEBUG_OBJECT (enc, "pushing buffer with %d bytes", - GST_BUFFER_SIZE (buffer)); - *flow = gst_pad_push (pad, buffer); + if (wid->correction || wid->passthrough) { + /* push the buffer and forward errors */ + GST_DEBUG_OBJECT (enc, "pushing buffer with %d bytes", + GST_BUFFER_SIZE (buffer)); + *flow = gst_pad_push (pad, buffer); + } else { + GST_DEBUG_OBJECT (enc, "handing frame of %d bytes", + GST_BUFFER_SIZE (buffer)); + *flow = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc), buffer, + samples); + } if (*flow != GST_FLOW_OK) { GST_WARNING_OBJECT (enc, "flow on %s:%s = %s", @@ -664,18 +661,25 @@ gst_wavpack_enc_fix_channel_order (GstWavpackEnc * enc, gint32 * data, } static GstFlowReturn -gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buf) +gst_wavpack_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf) { - GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad)); - uint32_t sample_count = GST_BUFFER_SIZE (buf) / 4; + GstWavpackEnc *enc = GST_WAVPACK_ENC (benc); + uint32_t sample_count; GstFlowReturn ret; + /* base class ensures configuration */ + g_return_val_if_fail (enc->depth != 0, GST_FLOW_NOT_NEGOTIATED); + /* reset the last returns to GST_FLOW_OK. This is only set to something else * while WavpackPackSamples() or more specific gst_wavpack_enc_push_block() * so not valid anymore */ enc->srcpad_last_return = enc->wvcsrcpad_last_return = GST_FLOW_OK; - GST_DEBUG ("got %u raw samples", sample_count); + if (G_UNLIKELY (!buf)) + return gst_wavpack_enc_drain (enc); + + sample_count = GST_BUFFER_SIZE (buf) / 4; + GST_DEBUG_OBJECT (enc, "got %u raw samples", sample_count); /* check if we already have a valid WavpackContext, otherwise make one */ if (!enc->wp_context) { @@ -683,13 +687,8 @@ gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buf) enc->wp_context = WavpackOpenFileOutput (gst_wavpack_enc_push_block, &enc->wv_id, (enc->correction_mode > 0) ? &enc->wvc_id : NULL); - if (!enc->wp_context) { - GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL), - ("error creating Wavpack context")); - gst_object_unref (enc); - gst_buffer_unref (buf); - return GST_FLOW_ERROR; - } + if (!enc->wp_context) + goto context_failed; /* set the WavpackConfig according to our parameters */ gst_wavpack_enc_set_wp_config (enc); @@ -699,76 +698,12 @@ gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buf) if (!WavpackSetConfiguration (enc->wp_context, enc->wp_config, (uint32_t) (-1)) || !WavpackPackInit (enc->wp_context)) { - GST_ELEMENT_ERROR (enc, LIBRARY, SETTINGS, (NULL), - ("error setting up wavpack encoding context")); WavpackCloseFile (enc->wp_context); - gst_object_unref (enc); - gst_buffer_unref (buf); - return GST_FLOW_ERROR; + goto config_failed; } - GST_DEBUG ("setup of encoding context successfull"); + GST_DEBUG_OBJECT (enc, "setup of encoding context successfull"); } - /* Save the timestamp of the first buffer. This will be later - * used as offset for all following buffers */ - if (enc->timestamp_offset == GST_CLOCK_TIME_NONE) { - if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) { - enc->timestamp_offset = GST_BUFFER_TIMESTAMP (buf); - enc->next_ts = GST_BUFFER_TIMESTAMP (buf); - } else { - enc->timestamp_offset = 0; - enc->next_ts = 0; - } - } - - /* Check if we have a continous stream, if not drop some samples or the buffer or - * insert some silence samples */ - if (enc->next_ts != GST_CLOCK_TIME_NONE && - GST_BUFFER_TIMESTAMP (buf) < enc->next_ts) { - guint64 diff = enc->next_ts - GST_BUFFER_TIMESTAMP (buf); - guint64 diff_bytes; - - GST_WARNING_OBJECT (enc, "Buffer is older than previous " - "timestamp + duration (%" GST_TIME_FORMAT "< %" GST_TIME_FORMAT - "), cannot handle. Clipping buffer.", - GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)), - GST_TIME_ARGS (enc->next_ts)); - - diff_bytes = - GST_CLOCK_TIME_TO_FRAMES (diff, enc->samplerate) * enc->channels * 2; - if (diff_bytes >= GST_BUFFER_SIZE (buf)) { - gst_buffer_unref (buf); - return GST_FLOW_OK; - } - buf = gst_buffer_make_metadata_writable (buf); - GST_BUFFER_DATA (buf) += diff_bytes; - GST_BUFFER_SIZE (buf) -= diff_bytes; - - GST_BUFFER_TIMESTAMP (buf) += diff; - if (GST_BUFFER_DURATION_IS_VALID (buf)) - GST_BUFFER_DURATION (buf) -= diff; - } - - /* Allow a diff of at most 5 ms */ - if (enc->next_ts != GST_CLOCK_TIME_NONE - && GST_BUFFER_TIMESTAMP_IS_VALID (buf)) { - if (GST_BUFFER_TIMESTAMP (buf) != enc->next_ts && - GST_BUFFER_TIMESTAMP (buf) - enc->next_ts > 5 * GST_MSECOND) { - GST_WARNING_OBJECT (enc, - "Discontinuity detected: %" G_GUINT64_FORMAT " > %" G_GUINT64_FORMAT, - GST_BUFFER_TIMESTAMP (buf) - enc->next_ts, 5 * GST_MSECOND); - - WavpackFlushSamples (enc->wp_context); - enc->timestamp_offset += (GST_BUFFER_TIMESTAMP (buf) - enc->next_ts); - } - } - - if (GST_BUFFER_TIMESTAMP_IS_VALID (buf) - && GST_BUFFER_DURATION_IS_VALID (buf)) - enc->next_ts = GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf); - else - enc->next_ts = GST_CLOCK_TIME_NONE; - if (enc->need_channel_remap) { buf = gst_buffer_make_writable (buf); gst_wavpack_enc_fix_channel_order (enc, (gint32 *) GST_BUFFER_DATA (buf), @@ -785,7 +720,7 @@ gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buf) /* encode and handle return values from encoding */ if (WavpackPackSamples (enc->wp_context, (int32_t *) GST_BUFFER_DATA (buf), sample_count / enc->channels)) { - GST_DEBUG ("encoding samples successful"); + GST_DEBUG_OBJECT (enc, "encoding samples successful"); ret = GST_FLOW_OK; } else { if ((enc->srcpad_last_return == GST_FLOW_RESEND) || @@ -801,15 +736,35 @@ gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buf) (enc->wvcsrcpad_last_return == GST_FLOW_WRONG_STATE)) { ret = GST_FLOW_WRONG_STATE; } else { - GST_ELEMENT_ERROR (enc, LIBRARY, ENCODE, (NULL), - ("encoding samples failed")); - ret = GST_FLOW_ERROR; + goto encoding_failed; } } - gst_buffer_unref (buf); - gst_object_unref (enc); +exit: return ret; + + /* ERRORS */ +encoding_failed: + { + GST_ELEMENT_ERROR (enc, LIBRARY, ENCODE, (NULL), + ("encoding samples failed")); + ret = GST_FLOW_ERROR; + goto exit; + } +config_failed: + { + GST_ELEMENT_ERROR (enc, LIBRARY, SETTINGS, (NULL), + ("error setting up wavpack encoding context")); + ret = GST_FLOW_ERROR; + goto exit; + } +context_failed: + { + GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL), + ("error creating Wavpack context")); + ret = GST_FLOW_ERROR; + goto exit; + } } static void @@ -826,7 +781,7 @@ gst_wavpack_enc_rewrite_first_block (GstWavpackEnc * enc) WavpackUpdateNumSamples (enc->wp_context, enc->first_block); /* try to seek to the beginning of the output */ - ret = gst_pad_push_event (enc->srcpad, event); + ret = gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (enc), event); if (ret) { /* try to rewrite the first block */ GST_DEBUG_OBJECT (enc, "rewriting first block ..."); @@ -834,111 +789,84 @@ gst_wavpack_enc_rewrite_first_block (GstWavpackEnc * enc) ret = gst_wavpack_enc_push_block (&enc->wv_id, enc->first_block, enc->first_block_size); enc->wv_id.passthrough = FALSE; + g_free (enc->first_block); + enc->first_block = NULL; } else { GST_WARNING_OBJECT (enc, "rewriting of first block failed. " "Seeking to first block failed!"); } } -static gboolean -gst_wavpack_enc_sink_event (GstPad * pad, GstEvent * event) +static GstFlowReturn +gst_wavpack_enc_drain (GstWavpackEnc * enc) { - GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad)); - gboolean ret = TRUE; + if (!enc->wp_context) + return GST_FLOW_OK; - GST_DEBUG ("Received %s event on sinkpad", GST_EVENT_TYPE_NAME (event)); + GST_DEBUG_OBJECT (enc, "draining"); + + /* Encode all remaining samples and flush them to the src pads */ + WavpackFlushSamples (enc->wp_context); + + /* Drop all remaining data, this is no complete block otherwise + * it would've been pushed already */ + if (enc->pending_buffer) { + gst_buffer_unref (enc->pending_buffer); + enc->pending_buffer = NULL; + enc->pending_offset = 0; + } + + /* write the MD5 sum if we have to write one */ + if ((enc->md5) && (enc->md5_context)) { + guint8 md5_digest[16]; + gsize digest_len = sizeof (md5_digest); + + g_checksum_get_digest (enc->md5_context, md5_digest, &digest_len); + if (digest_len == sizeof (md5_digest)) { + WavpackStoreMD5Sum (enc->wp_context, md5_digest); + WavpackFlushSamples (enc->wp_context); + } else + GST_WARNING_OBJECT (enc, "Calculating MD5 digest failed"); + } + + /* Try to rewrite the first frame with the correct sample number */ + if (enc->first_block) + gst_wavpack_enc_rewrite_first_block (enc); + + /* close the context if not already happened */ + if (enc->wp_context) { + WavpackCloseFile (enc->wp_context); + enc->wp_context = NULL; + } + + return GST_FLOW_OK; +} + +static gboolean +gst_wavpack_enc_sink_event (GstAudioEncoder * benc, GstEvent * event) +{ + GstWavpackEnc *enc = GST_WAVPACK_ENC (benc); + + GST_DEBUG_OBJECT (enc, "Received %s event on sinkpad", + GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { - case GST_EVENT_EOS: - /* Encode all remaining samples and flush them to the src pads */ - WavpackFlushSamples (enc->wp_context); - - /* Drop all remaining data, this is no complete block otherwise - * it would've been pushed already */ - if (enc->pending_buffer) { - gst_buffer_unref (enc->pending_buffer); - enc->pending_buffer = NULL; - enc->pending_offset = 0; - } - - /* write the MD5 sum if we have to write one */ - if ((enc->md5) && (enc->md5_context)) { - guint8 md5_digest[16]; - gsize digest_len = sizeof (md5_digest); - - g_checksum_get_digest (enc->md5_context, md5_digest, &digest_len); - if (digest_len == sizeof (md5_digest)) - WavpackStoreMD5Sum (enc->wp_context, md5_digest); - else - GST_WARNING_OBJECT (enc, "Calculating MD5 digest failed"); - } - - /* Try to rewrite the first frame with the correct sample number */ - if (enc->first_block) - gst_wavpack_enc_rewrite_first_block (enc); - - /* close the context if not already happened */ - if (enc->wp_context) { - WavpackCloseFile (enc->wp_context); - enc->wp_context = NULL; - } - - ret = gst_pad_event_default (pad, event); - break; case GST_EVENT_NEWSEGMENT: if (enc->wp_context) { GST_WARNING_OBJECT (enc, "got NEWSEGMENT after encoding " "already started"); } - /* drop NEWSEGMENT events, we create our own when pushing - * the first buffer to the pads */ - gst_event_unref (event); - ret = TRUE; - break; - default: - ret = gst_pad_event_default (pad, event); - break; - } - - gst_object_unref (enc); - return ret; -} - -static GstStateChangeReturn -gst_wavpack_enc_change_state (GstElement * element, GstStateChange transition) -{ - GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS; - GstWavpackEnc *enc = GST_WAVPACK_ENC (element); - - switch (transition) { - case GST_STATE_CHANGE_NULL_TO_READY: - /* set the last returned GstFlowReturns of the two pads to GST_FLOW_OK - * as they're only set to something else in WavpackPackSamples() or more - * specific gst_wavpack_enc_push_block() and nothing happened there yet */ - enc->srcpad_last_return = enc->wvcsrcpad_last_return = GST_FLOW_OK; - break; - case GST_STATE_CHANGE_READY_TO_PAUSED: - break; - case GST_STATE_CHANGE_PAUSED_TO_PLAYING: - default: - break; - } - - ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); - - switch (transition) { - case GST_STATE_CHANGE_PLAYING_TO_PAUSED: - break; - case GST_STATE_CHANGE_PAUSED_TO_READY: - gst_wavpack_enc_reset (enc); - break; - case GST_STATE_CHANGE_READY_TO_NULL: + /* peek and hold NEWSEGMENT events for sending on correction pad */ + if (enc->pending_segment) + gst_event_unref (enc->pending_segment); + enc->pending_segment = gst_event_ref (event); break; default: break; } - return ret; + /* baseclass handles rest */ + return FALSE; } static void diff --git a/ext/wavpack/gstwavpackenc.h b/ext/wavpack/gstwavpackenc.h index d2df844e50..aab4296fbf 100644 --- a/ext/wavpack/gstwavpackenc.h +++ b/ext/wavpack/gstwavpackenc.h @@ -23,6 +23,7 @@ #define __GST_WAVPACK_ENC_H__ #include +#include #include @@ -50,10 +51,9 @@ typedef struct struct _GstWavpackEnc { - GstElement element; + GstAudioEncoder element; /*< private > */ - GstPad *sinkpad, *srcpad; GstPad *wvcsrcpad; GstFlowReturn srcpad_last_return; @@ -86,6 +86,7 @@ struct _GstWavpackEnc GstBuffer *pending_buffer; gint32 pending_offset; + GstEvent *pending_segment; GstClockTime timestamp_offset; GstClockTime next_ts; @@ -93,7 +94,7 @@ struct _GstWavpackEnc struct _GstWavpackEncClass { - GstElementClass parent; + GstAudioEncoderClass parent; }; GType gst_wavpack_enc_get_type (void); diff --git a/tests/check/elements/wavpackenc.c b/tests/check/elements/wavpackenc.c index 153668a9df..af852aaa26 100644 --- a/tests/check/elements/wavpackenc.c +++ b/tests/check/elements/wavpackenc.c @@ -32,14 +32,14 @@ static GstBus *bus; #define RAW_CAPS_STRING "audio/x-raw-int, " \ "width = (int) 32, " \ - "depth = (int) 16, " \ + "depth = (int) 32, " \ "channels = (int) 1, " \ "rate = (int) 44100, " \ "endianness = (int) BYTE_ORDER, " \ "signed = (boolean) true" #define WAVPACK_CAPS_STRING "audio/x-wavpack, " \ - "width = (int) 16, " \ + "width = (int) 32, " \ "channels = (int) 1, " \ "rate = (int) 44100, " \ "framed = (boolean) true" @@ -48,7 +48,7 @@ static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-wavpack, " - "width = (int) 16, " + "width = (int) 32, " "channels = (int) 1, " "rate = (int) 44100, " "framed = (boolean) true")); @@ -57,7 +57,7 @@ static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-int, " "width = (int) 32, " - "depth = (int) 16, " + "depth = (int) 32, " "channels = (int) 1, " "rate = (int) 44100, " "endianness = (int) BYTE_ORDER, " "signed = (boolean) true")); @@ -118,13 +118,10 @@ GST_START_TEST (test_encode_silence) gst_caps_unref (caps); GST_BUFFER_TIMESTAMP (inbuffer) = 0; ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1); - gst_buffer_ref (inbuffer); gst_element_set_bus (wavpackenc, bus); fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK); - ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1); - gst_buffer_unref (inbuffer); fail_if (gst_pad_push_event (mysrcpad, eos) != TRUE); @@ -134,9 +131,7 @@ GST_START_TEST (test_encode_silence) fail_if (outbuffer == NULL); fail_unless_equals_int (GST_BUFFER_TIMESTAMP (outbuffer), 0); - fail_unless_equals_int (GST_BUFFER_OFFSET (outbuffer), 0); fail_unless_equals_int (GST_BUFFER_DURATION (outbuffer), 5668934); - fail_unless_equals_int (GST_BUFFER_OFFSET_END (outbuffer), 250); fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), "wvpk", 4) == 0, "Failed to encode to valid Wavpack frames"); From 004377b0b526182c2c24f374d88c038d9b99800d Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Mon, 27 Feb 2012 23:46:03 +0100 Subject: [PATCH 21/55] wavpackdec: port to audiodecoder --- ext/wavpack/gstwavpackdec.c | 296 +++++++++++++----------------------- ext/wavpack/gstwavpackdec.h | 12 +- 2 files changed, 110 insertions(+), 198 deletions(-) diff --git a/ext/wavpack/gstwavpackdec.c b/ext/wavpack/gstwavpackdec.c index 4ba242d875..133b51b60a 100644 --- a/ext/wavpack/gstwavpackdec.c +++ b/ext/wavpack/gstwavpackdec.c @@ -55,8 +55,6 @@ #include "gstwavpackstreamreader.h" -#define WAVPACK_DEC_MAX_ERRORS 16 - GST_DEBUG_CATEGORY_STATIC (gst_wavpack_dec_debug); #define GST_CAT_DEFAULT gst_wavpack_dec_debug @@ -80,15 +78,18 @@ static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src", "endianness = (int) BYTE_ORDER, " "signed = (boolean) true") ); -static GstFlowReturn gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buffer); -static gboolean gst_wavpack_dec_sink_set_caps (GstPad * pad, GstCaps * caps); -static gboolean gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event); +static gboolean gst_wavpack_dec_start (GstAudioDecoder * dec); +static gboolean gst_wavpack_dec_stop (GstAudioDecoder * dec); +static gboolean gst_wavpack_dec_set_format (GstAudioDecoder * dec, + GstCaps * caps); +static GstFlowReturn gst_wavpack_dec_handle_frame (GstAudioDecoder * dec, + GstBuffer * buffer); + static void gst_wavpack_dec_finalize (GObject * object); -static GstStateChangeReturn gst_wavpack_dec_change_state (GstElement * element, - GstStateChange transition); static void gst_wavpack_dec_post_tags (GstWavpackDec * dec); -GST_BOILERPLATE (GstWavpackDec, gst_wavpack_dec, GstElement, GST_TYPE_ELEMENT); +GST_BOILERPLATE (GstWavpackDec, gst_wavpack_dec, GstAudioDecoder, + GST_TYPE_AUDIO_DECODER); static void gst_wavpack_dec_base_init (gpointer klass) @@ -108,11 +109,14 @@ static void gst_wavpack_dec_class_init (GstWavpackDecClass * klass) { GObjectClass *gobject_class = (GObjectClass *) klass; - GstElementClass *gstelement_class = (GstElementClass *) klass; + GstAudioDecoderClass *base_class = (GstAudioDecoderClass *) (klass); - gstelement_class->change_state = - GST_DEBUG_FUNCPTR (gst_wavpack_dec_change_state); gobject_class->finalize = gst_wavpack_dec_finalize; + + base_class->start = GST_DEBUG_FUNCPTR (gst_wavpack_dec_start); + base_class->stop = GST_DEBUG_FUNCPTR (gst_wavpack_dec_stop); + base_class->set_format = GST_DEBUG_FUNCPTR (gst_wavpack_dec_set_format); + base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_wavpack_dec_handle_frame); } static void @@ -121,33 +125,15 @@ gst_wavpack_dec_reset (GstWavpackDec * dec) dec->wv_id.buffer = NULL; dec->wv_id.position = dec->wv_id.length = 0; - dec->error_count = 0; - dec->channels = 0; dec->channel_mask = 0; dec->sample_rate = 0; dec->depth = 0; - - gst_segment_init (&dec->segment, GST_FORMAT_TIME); - dec->next_block_index = 0; } static void gst_wavpack_dec_init (GstWavpackDec * dec, GstWavpackDecClass * gklass) { - dec->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink"); - gst_pad_set_chain_function (dec->sinkpad, - GST_DEBUG_FUNCPTR (gst_wavpack_dec_chain)); - gst_pad_set_setcaps_function (dec->sinkpad, - GST_DEBUG_FUNCPTR (gst_wavpack_dec_sink_set_caps)); - gst_pad_set_event_function (dec->sinkpad, - GST_DEBUG_FUNCPTR (gst_wavpack_dec_sink_event)); - gst_element_add_pad (GST_ELEMENT (dec), dec->sinkpad); - - dec->srcpad = gst_pad_new_from_static_template (&src_factory, "src"); - gst_pad_use_fixed_caps (dec->srcpad); - gst_element_add_pad (GST_ELEMENT (dec), dec->srcpad); - dec->context = NULL; dec->stream_reader = gst_wavpack_stream_reader_new (); @@ -166,26 +152,73 @@ gst_wavpack_dec_finalize (GObject * object) } static gboolean -gst_wavpack_dec_sink_set_caps (GstPad * pad, GstCaps * caps) +gst_wavpack_dec_start (GstAudioDecoder * dec) { - GstWavpackDec *dec = GST_WAVPACK_DEC (gst_pad_get_parent (pad)); + GST_DEBUG_OBJECT (dec, "start"); + + /* never mind a few errors */ + gst_audio_decoder_set_max_errors (dec, 16); + /* don't bother us with flushing */ + gst_audio_decoder_set_drainable (dec, FALSE); + + return TRUE; +} + +static gboolean +gst_wavpack_dec_stop (GstAudioDecoder * dec) +{ + GstWavpackDec *wpdec = GST_WAVPACK_DEC (dec); + + GST_DEBUG_OBJECT (dec, "stop"); + + if (wpdec->context) { + WavpackCloseFile (wpdec->context); + wpdec->context = NULL; + } + + gst_wavpack_dec_reset (wpdec); + + return TRUE; +} + +static void +gst_wavpack_dec_negotiate (GstWavpackDec * dec) +{ + GstCaps *caps; + + caps = gst_caps_new_simple ("audio/x-raw-int", + "rate", G_TYPE_INT, dec->sample_rate, + "channels", G_TYPE_INT, dec->channels, + "depth", G_TYPE_INT, dec->depth, + "width", G_TYPE_INT, 32, + "endianness", G_TYPE_INT, G_BYTE_ORDER, + "signed", G_TYPE_BOOLEAN, TRUE, NULL); + + /* Only set the channel layout for more than two channels + * otherwise things break unfortunately */ + if (dec->channel_mask != 0 && dec->channels > 2) + if (!gst_wavpack_set_channel_layout (caps, dec->channel_mask)) + GST_WARNING_OBJECT (dec, "Failed to set channel layout"); + + GST_DEBUG_OBJECT (dec, "setting caps %" GST_PTR_FORMAT, caps); + + /* should always succeed */ + gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps); + gst_caps_unref (caps); +} + +static gboolean +gst_wavpack_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps) +{ + GstWavpackDec *dec = GST_WAVPACK_DEC (bdec); GstStructure *structure = gst_caps_get_structure (caps, 0); /* Check if we can set the caps here already */ if (gst_structure_get_int (structure, "channels", &dec->channels) && gst_structure_get_int (structure, "rate", &dec->sample_rate) && gst_structure_get_int (structure, "width", &dec->depth)) { - GstCaps *caps; GstAudioChannelPosition *pos; - caps = gst_caps_new_simple ("audio/x-raw-int", - "rate", G_TYPE_INT, dec->sample_rate, - "channels", G_TYPE_INT, dec->channels, - "depth", G_TYPE_INT, dec->depth, - "width", G_TYPE_INT, 32, - "endianness", G_TYPE_INT, G_BYTE_ORDER, - "signed", G_TYPE_BOOLEAN, TRUE, NULL); - /* If we already have the channel layout set from upstream * take this */ if (gst_structure_has_field (structure, "channel-positions")) { @@ -202,19 +235,13 @@ gst_wavpack_dec_sink_set_caps (GstPad * pad, GstCaps * caps) g_free (pos); } - GST_DEBUG_OBJECT (dec, "setting caps %" GST_PTR_FORMAT, caps); - - /* should always succeed */ - gst_pad_set_caps (dec->srcpad, caps); - gst_caps_unref (caps); + gst_wavpack_dec_negotiate (dec); /* send GST_TAG_AUDIO_CODEC and GST_TAG_BITRATE tags before something * is decoded or after the format has changed */ gst_wavpack_dec_post_tags (dec); } - gst_object_unref (dec); - return TRUE; } @@ -225,28 +252,26 @@ gst_wavpack_dec_post_tags (GstWavpackDec * dec) GstFormat format_time = GST_FORMAT_TIME, format_bytes = GST_FORMAT_BYTES; gint64 duration, size; - list = gst_tag_list_new (); - - gst_tag_list_add (list, GST_TAG_MERGE_REPLACE, - GST_TAG_AUDIO_CODEC, "Wavpack", NULL); - /* try to estimate the average bitrate */ - if (gst_pad_query_peer_duration (dec->sinkpad, &format_bytes, &size) && - gst_pad_query_peer_duration (dec->sinkpad, &format_time, &duration) && - size > 0 && duration > 0) { + if (gst_pad_query_peer_duration (GST_AUDIO_DECODER_SINK_PAD (dec), + &format_bytes, &size) && + gst_pad_query_peer_duration (GST_AUDIO_DECODER_SINK_PAD (dec), + &format_time, &duration) && size > 0 && duration > 0) { guint64 bitrate; + list = gst_tag_list_new (); + bitrate = gst_util_uint64_scale (size, 8 * GST_SECOND, duration); gst_tag_list_add (list, GST_TAG_MERGE_REPLACE, GST_TAG_BITRATE, (guint) bitrate, NULL); - } - gst_element_post_message (GST_ELEMENT (dec), - gst_message_new_tag (GST_OBJECT (dec), list)); + gst_element_post_message (GST_ELEMENT (dec), + gst_message_new_tag (GST_OBJECT (dec), list)); + } } static GstFlowReturn -gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf) +gst_wavpack_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf) { GstWavpackDec *dec; GstBuffer *outbuf = NULL; @@ -255,7 +280,9 @@ gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf) int32_t decoded, unpacked_size; gboolean format_changed; - dec = GST_WAVPACK_DEC (GST_PAD_PARENT (pad)); + dec = GST_WAVPACK_DEC (bdec); + + g_return_val_if_fail (buf != NULL, GST_FLOW_ERROR); /* check input, we only accept framed input with complete chunks */ if (GST_BUFFER_SIZE (buf) < sizeof (WavpackHeader)) @@ -283,21 +310,15 @@ gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf) dec->context = WavpackOpenFileInputEx (dec->stream_reader, &dec->wv_id, NULL, error_msg, OPEN_STREAMING, 0); + /* expect this to work */ if (!dec->context) { - GST_WARNING ("Couldn't decode buffer: %s", error_msg); - dec->error_count++; - if (dec->error_count <= WAVPACK_DEC_MAX_ERRORS) { - goto out; /* just return OK for now */ - } else { - goto decode_error; - } + GST_WARNING_OBJECT (dec, "Couldn't decode buffer: %s", error_msg); + goto context_failed; } } g_assert (dec->context != NULL); - dec->error_count = 0; - format_changed = (dec->sample_rate != WavpackGetSampleRate (dec->context)) || (dec->channels != WavpackGetNumChannels (dec->context)) || @@ -308,22 +329,13 @@ gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf) (dec->channel_mask != WavpackGetChannelMask (dec->context)); #endif - if (!GST_PAD_CAPS (dec->srcpad) || format_changed) { - GstCaps *caps; + if (!GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)) || format_changed) { gint channel_mask; dec->sample_rate = WavpackGetSampleRate (dec->context); dec->channels = WavpackGetNumChannels (dec->context); dec->depth = WavpackGetBitsPerSample (dec->context); - caps = gst_caps_new_simple ("audio/x-raw-int", - "rate", G_TYPE_INT, dec->sample_rate, - "channels", G_TYPE_INT, dec->channels, - "depth", G_TYPE_INT, dec->depth, - "width", G_TYPE_INT, 32, - "endianness", G_TYPE_INT, G_BYTE_ORDER, - "signed", G_TYPE_BOOLEAN, TRUE, NULL); - #ifdef WAVPACK_OLD_API channel_mask = dec->context->config.channel_mask; #else @@ -334,17 +346,7 @@ gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf) dec->channel_mask = channel_mask; - /* Only set the channel layout for more than two channels - * otherwise things break unfortunately */ - if (channel_mask != 0 && dec->channels > 2) - if (!gst_wavpack_set_channel_layout (caps, channel_mask)) - GST_WARNING_OBJECT (dec, "Failed to set channel layout"); - - GST_DEBUG_OBJECT (dec, "setting caps %" GST_PTR_FORMAT, caps); - - /* should always succeed */ - gst_pad_set_caps (dec->srcpad, caps); - gst_caps_unref (caps); + gst_wavpack_dec_negotiate (dec); /* send GST_TAG_AUDIO_CODEC and GST_TAG_BITRATE tags before something * is decoded or after the format has changed */ @@ -353,34 +355,20 @@ gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf) /* alloc output buffer */ unpacked_size = 4 * wph.block_samples * dec->channels; - ret = gst_pad_alloc_buffer (dec->srcpad, GST_BUFFER_OFFSET (buf), - unpacked_size, GST_PAD_CAPS (dec->srcpad), &outbuf); + ret = gst_pad_alloc_buffer (GST_AUDIO_DECODER_SRC_PAD (dec), + GST_BUFFER_OFFSET (buf), unpacked_size, + GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)), &outbuf); if (ret != GST_FLOW_OK) goto out; - gst_buffer_copy_metadata (outbuf, buf, GST_BUFFER_COPY_TIMESTAMPS); - - /* If we got a DISCONT buffer forward the flag. Nothing else - * has to be done as libwavpack doesn't store state between - * Wavpack blocks */ - if (GST_BUFFER_IS_DISCONT (buf) || dec->next_block_index != wph.block_index) - GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT); - - dec->next_block_index = wph.block_index + wph.block_samples; - /* decode */ decoded = WavpackUnpackSamples (dec->context, (int32_t *) GST_BUFFER_DATA (outbuf), wph.block_samples); if (decoded != wph.block_samples) goto decode_error; - if ((outbuf = gst_audio_buffer_clip (outbuf, &dec->segment, - dec->sample_rate, 4 * dec->channels))) { - GST_LOG_OBJECT (dec, "pushing buffer with time %" GST_TIME_FORMAT, - GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf))); - ret = gst_pad_push (dec->srcpad, outbuf); - } + ret = gst_audio_decoder_finish_frame (bdec, outbuf, 1); out: @@ -388,23 +376,25 @@ out: GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (ret)); } - gst_buffer_unref (buf); - return ret; /* ERRORS */ input_not_framed: { GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Expected framed input")); - gst_buffer_unref (buf); return GST_FLOW_ERROR; } invalid_header: { GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Invalid wavpack header")); - gst_buffer_unref (buf); return GST_FLOW_ERROR; } +context_failed: + { + GST_AUDIO_DECODER_ERROR (bdec, 1, LIBRARY, INIT, (NULL), + ("error creating Wavpack context"), ret); + goto out; + } decode_error: { const gchar *reason = "unknown"; @@ -418,88 +408,16 @@ decode_error: } else { reason = "couldn't create decoder context"; } - GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), - ("Failed to decode wavpack stream: %s", reason)); + GST_AUDIO_DECODER_ERROR (bdec, 1, STREAM, DECODE, (NULL), + ("decoding error: %s", reason), ret); if (outbuf) gst_buffer_unref (outbuf); - gst_buffer_unref (buf); - return GST_FLOW_ERROR; + if (ret == GST_FLOW_OK) + gst_audio_decoder_finish_frame (bdec, NULL, 1); + return ret; } } -static gboolean -gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event) -{ - GstWavpackDec *dec = GST_WAVPACK_DEC (gst_pad_get_parent (pad)); - - GST_LOG_OBJECT (dec, "Received %s event", GST_EVENT_TYPE_NAME (event)); - switch (GST_EVENT_TYPE (event)) { - case GST_EVENT_NEWSEGMENT:{ - GstFormat fmt; - gboolean is_update; - gint64 start, end, base; - gdouble rate; - - gst_event_parse_new_segment (event, &is_update, &rate, &fmt, &start, - &end, &base); - if (fmt == GST_FORMAT_TIME) { - GST_DEBUG ("Got NEWSEGMENT event in GST_FORMAT_TIME, passing on (%" - GST_TIME_FORMAT " - %" GST_TIME_FORMAT ")", GST_TIME_ARGS (start), - GST_TIME_ARGS (end)); - gst_segment_set_newsegment (&dec->segment, is_update, rate, fmt, - start, end, base); - } else { - gst_segment_init (&dec->segment, GST_FORMAT_TIME); - } - break; - } - default: - break; - } - - gst_object_unref (dec); - return gst_pad_event_default (pad, event); -} - -static GstStateChangeReturn -gst_wavpack_dec_change_state (GstElement * element, GstStateChange transition) -{ - GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS; - GstWavpackDec *dec = GST_WAVPACK_DEC (element); - - switch (transition) { - case GST_STATE_CHANGE_NULL_TO_READY: - break; - case GST_STATE_CHANGE_READY_TO_PAUSED: - break; - case GST_STATE_CHANGE_PAUSED_TO_PLAYING: - break; - default: - break; - } - - ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); - - switch (transition) { - case GST_STATE_CHANGE_PLAYING_TO_PAUSED: - break; - case GST_STATE_CHANGE_PAUSED_TO_READY: - if (dec->context) { - WavpackCloseFile (dec->context); - dec->context = NULL; - } - - gst_wavpack_dec_reset (dec); - break; - case GST_STATE_CHANGE_READY_TO_NULL: - break; - default: - break; - } - - return ret; -} - gboolean gst_wavpack_dec_plugin_init (GstPlugin * plugin) { diff --git a/ext/wavpack/gstwavpackdec.h b/ext/wavpack/gstwavpackdec.h index eb6e4c3bb8..4b8b057987 100644 --- a/ext/wavpack/gstwavpackdec.h +++ b/ext/wavpack/gstwavpackdec.h @@ -24,6 +24,7 @@ #define __GST_WAVPACK_DEC_H__ #include +#include #include @@ -45,31 +46,24 @@ typedef struct _GstWavpackDecClass GstWavpackDecClass; struct _GstWavpackDec { - GstElement element; + GstAudioDecoder element; /*< private > */ - GstPad *sinkpad; - GstPad *srcpad; WavpackContext *context; WavpackStreamReader *stream_reader; read_id wv_id; - GstSegment segment; /* used for clipping, TIME format */ - guint32 next_block_index; - gint sample_rate; gint depth; gint channels; gint channel_mask; - - gint error_count; }; struct _GstWavpackDecClass { - GstElementClass parent; + GstAudioDecoderClass parent; }; GType gst_wavpack_dec_get_type (void); From 0723928e8bcd49ceaa41145756922f00991c7c88 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Mon, 27 Feb 2012 23:46:08 +0100 Subject: [PATCH 22/55] wavpackdec: determine depth from bytes per sample ... rather than from bits per sample, since spec states values are already left justified w.r.t. bits per sample but not w.r.t. bytes per sample (and so the latter determines the normalization, or indicated depth). --- ext/wavpack/gstwavpackdec.c | 4 ++-- ext/wavpack/gstwavpackenc.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/ext/wavpack/gstwavpackdec.c b/ext/wavpack/gstwavpackdec.c index 133b51b60a..365544b179 100644 --- a/ext/wavpack/gstwavpackdec.c +++ b/ext/wavpack/gstwavpackdec.c @@ -322,7 +322,7 @@ gst_wavpack_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf) format_changed = (dec->sample_rate != WavpackGetSampleRate (dec->context)) || (dec->channels != WavpackGetNumChannels (dec->context)) || - (dec->depth != WavpackGetBitsPerSample (dec->context)) || + (dec->depth != WavpackGetBytesPerSample (dec->context) * 8) || #ifdef WAVPACK_OLD_API (dec->channel_mask != dec->context->config.channel_mask); #else @@ -334,7 +334,7 @@ gst_wavpack_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf) dec->sample_rate = WavpackGetSampleRate (dec->context); dec->channels = WavpackGetNumChannels (dec->context); - dec->depth = WavpackGetBitsPerSample (dec->context); + dec->depth = WavpackGetBytesPerSample (dec->context) * 8; #ifdef WAVPACK_OLD_API channel_mask = dec->context->config.channel_mask; diff --git a/ext/wavpack/gstwavpackenc.c b/ext/wavpack/gstwavpackenc.c index e2e19d4d38..8595798f47 100644 --- a/ext/wavpack/gstwavpackenc.c +++ b/ext/wavpack/gstwavpackenc.c @@ -103,7 +103,7 @@ static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-wavpack, " "width = (int) [ 1, 32 ], " - "channels = (int) [ 1, 2 ], " + "channels = (int) [ 1, 8 ], " "rate = (int) [ 6000, 192000 ], " "framed = (boolean) TRUE") ); From a5144cbba0c11547a377a8eb8f5383ccf12d6d84 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Mon, 27 Feb 2012 23:46:15 +0100 Subject: [PATCH 23/55] wavpackdec: adjust to audio format limitations ... which does not allow expressing arbitrary depth in a GstAudioFormat. Also adjust unit test to modified behaviour. --- ext/wavpack/gstwavpackdec.c | 70 +++++++++++++++++++++++++++---- ext/wavpack/gstwavpackdec.h | 1 + tests/check/elements/wavpackdec.c | 17 ++------ 3 files changed, 67 insertions(+), 21 deletions(-) diff --git a/ext/wavpack/gstwavpackdec.c b/ext/wavpack/gstwavpackdec.c index 365544b179..2be81494f8 100644 --- a/ext/wavpack/gstwavpackdec.c +++ b/ext/wavpack/gstwavpackdec.c @@ -71,11 +71,20 @@ static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-int, " - "width = (int) 32, " - "depth = (int) [ 1, 32 ], " + "width = (int) 8, depth = (int) 8, " "channels = (int) [ 1, 8 ], " "rate = (int) [ 6000, 192000 ], " - "endianness = (int) BYTE_ORDER, " "signed = (boolean) true") + "endianness = (int) BYTE_ORDER, " "signed = (boolean) true; " + "audio/x-raw-int, " + "width = (int) 16, depth = (int) 16, " + "channels = (int) [ 1, 8 ], " + "rate = (int) [ 6000, 192000 ], " + "endianness = (int) BYTE_ORDER, " "signed = (boolean) true; " + "audio/x-raw-int, " + "width = (int) 32, depth = (int) 32, " + "channels = (int) [ 1, 8 ], " + "rate = (int) [ 6000, 192000 ], " + "endianness = (int) BYTE_ORDER, " "signed = (boolean) true; ") ); static gboolean gst_wavpack_dec_start (GstAudioDecoder * dec); @@ -186,11 +195,17 @@ gst_wavpack_dec_negotiate (GstWavpackDec * dec) { GstCaps *caps; + /* arrange for 1, 2 or 4-byte width == depth output */ + if (dec->depth == 24) + dec->width = 32; + else + dec->width = dec->depth; + caps = gst_caps_new_simple ("audio/x-raw-int", "rate", G_TYPE_INT, dec->sample_rate, "channels", G_TYPE_INT, dec->channels, - "depth", G_TYPE_INT, dec->depth, - "width", G_TYPE_INT, 32, + "depth", G_TYPE_INT, dec->width, + "width", G_TYPE_INT, dec->width, "endianness", G_TYPE_INT, G_BYTE_ORDER, "signed", G_TYPE_BOOLEAN, TRUE, NULL); @@ -279,6 +294,9 @@ gst_wavpack_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf) WavpackHeader wph; int32_t decoded, unpacked_size; gboolean format_changed; + gint width, depth, i, max; + gint32 *dec_data = NULL; + guint8 *out_data; dec = GST_WAVPACK_DEC (bdec); @@ -354,7 +372,7 @@ gst_wavpack_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf) } /* alloc output buffer */ - unpacked_size = 4 * wph.block_samples * dec->channels; + unpacked_size = (dec->width / 8) * wph.block_samples * dec->channels; ret = gst_pad_alloc_buffer (GST_AUDIO_DECODER_SRC_PAD (dec), GST_BUFFER_OFFSET (buf), unpacked_size, GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)), &outbuf); @@ -362,12 +380,47 @@ gst_wavpack_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf) if (ret != GST_FLOW_OK) goto out; + dec_data = g_malloc (4 * wph.block_samples * dec->channels); + out_data = GST_BUFFER_DATA (outbuf); + /* decode */ - decoded = WavpackUnpackSamples (dec->context, - (int32_t *) GST_BUFFER_DATA (outbuf), wph.block_samples); + decoded = WavpackUnpackSamples (dec->context, dec_data, wph.block_samples); if (decoded != wph.block_samples) goto decode_error; + width = dec->width; + depth = dec->depth; + max = dec->channels * wph.block_samples; + if (width == 8) { + gint8 *outbuffer = (gint8 *) out_data; + + for (i = 0; i < max; i--) { + *outbuffer++ = (gint8) (dec_data[i]); + } + } else if (width == 16) { + gint16 *outbuffer = (gint16 *) out_data; + + for (i = 0; i < max; i++) { + *outbuffer++ = (gint8) (dec_data[i]); + } + } else if (dec->width == 32) { + gint32 *outbuffer = (gint32 *) out_data; + + if (width != depth) { + for (i = 0; i < max; i++) { + *outbuffer++ = (gint32) (dec_data[i] << (width - depth)); + } + } else { + for (i = 0; i < max; i++) { + *outbuffer++ = (gint32) dec_data[i]; + } + } + } else { + g_assert_not_reached (); + } + + g_free (dec_data); + ret = gst_audio_decoder_finish_frame (bdec, outbuf, 1); out: @@ -410,6 +463,7 @@ decode_error: } GST_AUDIO_DECODER_ERROR (bdec, 1, STREAM, DECODE, (NULL), ("decoding error: %s", reason), ret); + g_free (dec_data); if (outbuf) gst_buffer_unref (outbuf); if (ret == GST_FLOW_OK) diff --git a/ext/wavpack/gstwavpackdec.h b/ext/wavpack/gstwavpackdec.h index 4b8b057987..68e8e33858 100644 --- a/ext/wavpack/gstwavpackdec.h +++ b/ext/wavpack/gstwavpackdec.h @@ -57,6 +57,7 @@ struct _GstWavpackDec gint sample_rate; gint depth; + gint width; gint channels; gint channel_mask; }; diff --git a/tests/check/elements/wavpackdec.c b/tests/check/elements/wavpackdec.c index 0d17323336..b1498e1667 100644 --- a/tests/check/elements/wavpackdec.c +++ b/tests/check/elements/wavpackdec.c @@ -52,7 +52,7 @@ static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-int, " - "width = (int) 32, " + "width = (int) 16, " "depth = (int) 16, " "channels = (int) 1, " "rate = (int) 44100, " @@ -112,23 +112,20 @@ GST_START_TEST (test_decode_frame) memcpy (GST_BUFFER_DATA (inbuffer), test_frame, sizeof (test_frame)); ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1); GST_BUFFER_TIMESTAMP (inbuffer) = 0; - gst_buffer_ref (inbuffer); gst_element_set_bus (wavpackdec, bus); /* should decode the buffer without problems */ fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK); - ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1); - gst_buffer_unref (inbuffer); outbuffer = GST_BUFFER (buffers->data); fail_if (outbuffer == NULL); /* uncompressed data should be 102400 bytes */ - fail_unless_equals_int (GST_BUFFER_SIZE (outbuffer), 102400); + fail_unless_equals_int (GST_BUFFER_SIZE (outbuffer), 51200); - /* and all 102400 bytes must be 0, i.e. silence */ - for (i = 0; i < 102400; i++) + /* and all bytes must be 0, i.e. silence */ + for (i = 0; i < 51200; i++) fail_unless_equals_int (GST_BUFFER_DATA (outbuffer)[i], 0); ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1); @@ -166,14 +163,11 @@ GST_START_TEST (test_decode_frame_with_broken_header) GST_BUFFER_DATA (inbuffer)[2] = 'e'; ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1); GST_BUFFER_TIMESTAMP (inbuffer) = 0; - gst_buffer_ref (inbuffer); gst_element_set_bus (wavpackdec, bus); /* should fail gracefully */ fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_ERROR); - ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1); - gst_buffer_unref (inbuffer); fail_if ((message = gst_bus_pop (bus)) == NULL); fail_unless_message_error (message, STREAM, DECODE); @@ -204,14 +198,11 @@ GST_START_TEST (test_decode_frame_with_incomplete_frame) memcpy (GST_BUFFER_DATA (inbuffer), test_frame, sizeof (test_frame) - 2); ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1); GST_BUFFER_TIMESTAMP (inbuffer) = 0; - gst_buffer_ref (inbuffer); gst_element_set_bus (wavpackdec, bus); /* should fail gracefully */ fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_ERROR); - ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1); - gst_buffer_unref (inbuffer); fail_if ((message = gst_bus_pop (bus)) == NULL); fail_unless_message_error (message, STREAM, DECODE); From dda3219e4c052de45a987a41674e78e960b20c8b Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Tue, 28 Feb 2012 11:38:59 +0100 Subject: [PATCH 24/55] update for metadata tags --- sys/v4l2/gstv4l2bufferpool.c | 3 ++- sys/ximage/ximageutil.c | 3 ++- 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sys/v4l2/gstv4l2bufferpool.c b/sys/v4l2/gstv4l2bufferpool.c index 10f29336ed..6c77cec422 100644 --- a/sys/v4l2/gstv4l2bufferpool.c +++ b/sys/v4l2/gstv4l2bufferpool.c @@ -61,12 +61,13 @@ const GstMetaInfo * gst_v4l2_meta_get_info (void) { static const GstMetaInfo *meta_info = NULL; + static const gchar *tags[] = { "memory" }; if (meta_info == NULL) { meta_info = gst_meta_register ("GstV4l2Meta", "GstV4l2Meta", sizeof (GstV4l2Meta), (GstMetaInitFunction) NULL, - (GstMetaFreeFunction) NULL, (GstMetaTransformFunction) NULL); + (GstMetaFreeFunction) NULL, (GstMetaTransformFunction) NULL, tags); } return meta_info; } diff --git a/sys/ximage/ximageutil.c b/sys/ximage/ximageutil.c index 47d6d48570..84cc01a5ae 100644 --- a/sys/ximage/ximageutil.c +++ b/sys/ximage/ximageutil.c @@ -27,12 +27,13 @@ const GstMetaInfo * gst_meta_ximage_get_info (void) { static const GstMetaInfo *meta_ximage_info = NULL; + static const gchar *tags[] = { "memory", NULL }; if (meta_ximage_info == NULL) { meta_ximage_info = gst_meta_register ("GstMetaXImageSrc", "GstMetaXImageSrc", sizeof (GstMetaXImage), (GstMetaInitFunction) NULL, - (GstMetaFreeFunction) NULL, (GstMetaTransformFunction) NULL); + (GstMetaFreeFunction) NULL, (GstMetaTransformFunction) NULL, tags); } return meta_ximage_info; } From 50cd7c9ac6f2d64d8b43e03f2b248efa02dafe4c Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Tue, 28 Feb 2012 13:51:10 +0100 Subject: [PATCH 25/55] audioparsers: add baseparse based wavpackparse --- gst/audioparsers/Makefile.am | 4 +- gst/audioparsers/gstwavpackparse.c | 648 +++++++++++++++++++++++++++++ gst/audioparsers/gstwavpackparse.h | 134 ++++++ gst/audioparsers/plugin.c | 3 + 4 files changed, 787 insertions(+), 2 deletions(-) create mode 100644 gst/audioparsers/gstwavpackparse.c create mode 100644 gst/audioparsers/gstwavpackparse.h diff --git a/gst/audioparsers/Makefile.am b/gst/audioparsers/Makefile.am index 22bc81fa09..4d4d53ed21 100644 --- a/gst/audioparsers/Makefile.am +++ b/gst/audioparsers/Makefile.am @@ -3,7 +3,7 @@ plugin_LTLIBRARIES = libgstaudioparsers.la libgstaudioparsers_la_SOURCES = \ gstaacparse.c gstamrparse.c gstac3parse.c \ gstdcaparse.c gstflacparse.c gstmpegaudioparse.c \ - plugin.c + gstwavpackparse.c plugin.c libgstaudioparsers_la_CFLAGS = \ $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) @@ -15,4 +15,4 @@ libgstaudioparsers_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) libgstaudioparsers_la_LIBTOOLFLAGS = --tag=disable-static noinst_HEADERS = gstaacparse.h gstamrparse.h gstac3parse.h \ - gstdcaparse.h gstflacparse.h gstmpegaudioparse.h + gstdcaparse.h gstflacparse.h gstmpegaudioparse.h gstwavpackparse.h diff --git a/gst/audioparsers/gstwavpackparse.c b/gst/audioparsers/gstwavpackparse.c new file mode 100644 index 0000000000..61b0eb46b8 --- /dev/null +++ b/gst/audioparsers/gstwavpackparse.c @@ -0,0 +1,648 @@ +/* GStreamer Wavpack parser + * Copyright (C) 2012 Mark Nauwelaerts + * Copyright (C) 2012 Nokia Corporation. All rights reserved. + * Contact: Stefan Kost + * + * 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-wavpackparse + * @short_description: Wavpack parser + * @see_also: #GstAmrParse, #GstAACParse + * + * This is an Wavpack parser. + * + * + * Example launch line + * |[ + * gst-launch filesrc location=abc.wavpack ! wavpackparse ! wavpackdec ! audioresample ! audioconvert ! autoaudiosink + * ]| + * + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include + +#include "gstwavpackparse.h" + +#include +#include + +GST_DEBUG_CATEGORY_STATIC (wavpack_parse_debug); +#define GST_CAT_DEFAULT wavpack_parse_debug + +static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/x-wavpack, " + "width = (int) [ 1, 32 ], " + "channels = (int) [ 1, 8 ], " + "rate = (int) [ 6000, 192000 ], " "framed = (boolean) TRUE; " + "audio/x-wavpack-correction, " "framed = (boolean) TRUE") + ); + +static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", + GST_PAD_SINK, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/x-wavpack")); + +static void gst_wavpack_parse_finalize (GObject * object); + +static gboolean gst_wavpack_parse_start (GstBaseParse * parse); +static gboolean gst_wavpack_parse_stop (GstBaseParse * parse); +static gboolean gst_wavpack_parse_check_valid_frame (GstBaseParse * parse, + GstBaseParseFrame * frame, guint * size, gint * skipsize); +static GstFlowReturn gst_wavpack_parse_parse_frame (GstBaseParse * parse, + GstBaseParseFrame * frame); +static GstCaps *gst_wavpack_parse_get_sink_caps (GstBaseParse * parse); + +/* FIXME remove when all properly renamed */ +typedef GstWavpackParse GstWavpackParse2; +typedef GstWavpackParseClass GstWavpackParse2Class; + +GST_BOILERPLATE (GstWavpackParse2, gst_wavpack_parse, GstBaseParse, + GST_TYPE_BASE_PARSE); + +static void +gst_wavpack_parse_base_init (gpointer klass) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (klass); + + gst_element_class_add_static_pad_template (element_class, &sink_template); + gst_element_class_add_static_pad_template (element_class, &src_template); + + gst_element_class_set_details_simple (element_class, + "Wavpack audio stream parser", "Codec/Parser/Audio", + "Wavpack parser", "Mark Nauwelaerts "); +} + +static void +gst_wavpack_parse_class_init (GstWavpackParseClass * klass) +{ + GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass); + GObjectClass *object_class = G_OBJECT_CLASS (klass); + + GST_DEBUG_CATEGORY_INIT (wavpack_parse_debug, "wavpackparse", 0, + "Wavpack audio stream parser"); + + object_class->finalize = gst_wavpack_parse_finalize; + + parse_class->start = GST_DEBUG_FUNCPTR (gst_wavpack_parse_start); + parse_class->stop = GST_DEBUG_FUNCPTR (gst_wavpack_parse_stop); + parse_class->check_valid_frame = + GST_DEBUG_FUNCPTR (gst_wavpack_parse_check_valid_frame); + parse_class->parse_frame = GST_DEBUG_FUNCPTR (gst_wavpack_parse_parse_frame); + parse_class->get_sink_caps = + GST_DEBUG_FUNCPTR (gst_wavpack_parse_get_sink_caps); +} + +static void +gst_wavpack_parse_reset (GstWavpackParse * wvparse) +{ + wvparse->channels = -1; + wvparse->channel_mask = 0; + wvparse->sample_rate = -1; + wvparse->width = -1; + wvparse->total_samples = 0; +} + +static void +gst_wavpack_parse_init (GstWavpackParse * wvparse, GstWavpackParseClass * klass) +{ + gst_wavpack_parse_reset (wvparse); +} + +static void +gst_wavpack_parse_finalize (GObject * object) +{ + G_OBJECT_CLASS (parent_class)->finalize (object); +} + +static gboolean +gst_wavpack_parse_start (GstBaseParse * parse) +{ + GstWavpackParse *wvparse = GST_WAVPACK_PARSE (parse); + + GST_DEBUG_OBJECT (parse, "starting"); + + gst_wavpack_parse_reset (wvparse); + + /* need header at least */ + gst_base_parse_set_min_frame_size (GST_BASE_PARSE (wvparse), + sizeof (WavpackHeader)); + + /* inform baseclass we can come up with ts, based on counters in packets */ + gst_base_parse_set_has_timing_info (GST_BASE_PARSE_CAST (wvparse), TRUE); + gst_base_parse_set_syncable (GST_BASE_PARSE_CAST (wvparse), TRUE); + + return TRUE; +} + +static gboolean +gst_wavpack_parse_stop (GstBaseParse * parse) +{ + GST_DEBUG_OBJECT (parse, "stopping"); + + return TRUE; +} + +static gint +gst_wavpack_get_default_channel_mask (gint nchannels) +{ + gint channel_mask = 0; + + /* Set the default channel mask for the given number of channels. + * It's the same as for WAVE_FORMAT_EXTENDED: + * http://www.microsoft.com/whdc/device/audio/multichaud.mspx + */ + switch (nchannels) { + case 11: + channel_mask |= 0x00400; + channel_mask |= 0x00200; + case 9: + channel_mask |= 0x00100; + case 8: + channel_mask |= 0x00080; + channel_mask |= 0x00040; + case 6: + channel_mask |= 0x00020; + channel_mask |= 0x00010; + case 4: + channel_mask |= 0x00008; + case 3: + channel_mask |= 0x00004; + case 2: + channel_mask |= 0x00002; + channel_mask |= 0x00001; + break; + case 1: + /* For mono use front center */ + channel_mask |= 0x00004; + break; + } + + return channel_mask; +} + +static const struct +{ + const guint32 ms_mask; + const GstAudioChannelPosition gst_pos; +} layout_mapping[] = { + { + 0x00001, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT}, { + 0x00002, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}, { + 0x00004, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER}, { + 0x00008, GST_AUDIO_CHANNEL_POSITION_LFE}, { + 0x00010, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT}, { + 0x00020, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}, { + 0x00040, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER}, { + 0x00080, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER}, { + 0x00100, GST_AUDIO_CHANNEL_POSITION_REAR_CENTER}, { + 0x00200, GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT}, { + 0x00400, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT}, { + 0x00800, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_CENTER */ + { + 0x01000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_FRONT_LEFT */ + { + 0x02000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_FRONT_CENTER */ + { + 0x04000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_FRONT_RIGHT */ + { + 0x08000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_BACK_LEFT */ + { + 0x10000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_BACK_CENTER */ + { + 0x20000, GST_AUDIO_CHANNEL_POSITION_INVALID} /* TOP_BACK_RIGHT */ +}; + +#define MAX_CHANNEL_POSITIONS G_N_ELEMENTS (layout_mapping) + +static gboolean +gst_wavpack_set_channel_layout (GstCaps * caps, gint layout) +{ + GstAudioChannelPosition pos[MAX_CHANNEL_POSITIONS]; + GstStructure *s; + gint num_channels, i, p; + + s = gst_caps_get_structure (caps, 0); + if (!gst_structure_get_int (s, "channels", &num_channels)) + g_return_val_if_reached (FALSE); + + if (num_channels == 1 && layout == 0x00004) { + pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_MONO; + gst_audio_set_channel_positions (s, pos); + return TRUE; + } + + p = 0; + for (i = 0; i < MAX_CHANNEL_POSITIONS; ++i) { + if ((layout & layout_mapping[i].ms_mask) != 0) { + if (p >= num_channels) { + GST_WARNING ("More bits set in the channel layout map than there " + "are channels! Broken file"); + return FALSE; + } + if (layout_mapping[i].gst_pos == GST_AUDIO_CHANNEL_POSITION_INVALID) { + GST_WARNING ("Unsupported channel position (mask 0x%08x) in channel " + "layout map - ignoring those channels", layout_mapping[i].ms_mask); + /* what to do? just ignore it and let downstream deal with a channel + * layout that has INVALID positions in it for now ... */ + } + pos[p] = layout_mapping[i].gst_pos; + ++p; + } + } + + if (p != num_channels) { + GST_WARNING ("Only %d bits set in the channel layout map, but there are " + "supposed to be %d channels! Broken file", p, num_channels); + return FALSE; + } + + gst_audio_set_channel_positions (s, pos); + return TRUE; +} + +static const guint32 sample_rates[] = { + 6000, 8000, 9600, 11025, 12000, 16000, 22050, + 24000, 32000, 44100, 48000, 64000, 88200, 96000, 192000 +}; + +#define CHECK(call) { \ + if (!call) \ + goto read_failed; \ +} + +/* caller ensures properly sync'ed with enough data */ +static gboolean +gst_wavpack_parse_frame_metadata (GstWavpackParse * parse, GstBuffer * buf, + gint skip, WavpackHeader * wph, WavpackInfo * wpi) +{ + GstByteReader br; + gint i; + + g_return_val_if_fail (wph != NULL || wpi != NULL, FALSE); + g_return_val_if_fail (GST_BUFFER_SIZE (buf) >= skip + sizeof (WavpackHeader), + FALSE); + + gst_byte_reader_init (&br, GST_BUFFER_DATA (buf) + skip, wph->ckSize + 8); + /* skip past header */ + gst_byte_reader_skip_unchecked (&br, sizeof (WavpackHeader)); + + /* get some basics from header */ + i = (wph->flags >> 23) & 0xF; + if (!wpi->rate) + wpi->rate = (i < G_N_ELEMENTS (sample_rates)) ? sample_rates[i] : 44100; + wpi->width = ((wph->flags & 0x3) + 1) * 8; + if (!wpi->channels) + wpi->channels = (wph->flags & 0x4) ? 1 : 2; + if (!wpi->channel_mask) + wpi->channel_mask = 5 - wpi->channels; + + /* need to dig metadata blocks for some more */ + while (gst_byte_reader_get_remaining (&br)) { + gint size = 0; + guint16 size2 = 0; + guint8 c, id; + const guint8 *data; + GstByteReader mbr; + + CHECK (gst_byte_reader_get_uint8 (&br, &id)); + CHECK (gst_byte_reader_get_uint8 (&br, &c)); + if (id & ID_LARGE) + CHECK (gst_byte_reader_get_uint16_le (&br, &size2)); + size = size2; + size <<= 8; + size += c; + size <<= 1; + if (id & ID_ODD_SIZE) + size--; + + CHECK (gst_byte_reader_get_data (&br, size + (size & 1), &data)); + gst_byte_reader_init (&mbr, data, size); + + switch (id) { + case ID_WVC_BITSTREAM: + GST_LOG_OBJECT (parse, "correction bitstream"); + wpi->correction = TRUE; + break; + case ID_WV_BITSTREAM: + case ID_WVX_BITSTREAM: + break; + case ID_SAMPLE_RATE: + if (size == 3) { + CHECK (gst_byte_reader_get_uint24_le (&mbr, &wpi->rate)); + GST_LOG_OBJECT (parse, "updated with custom rate %d", wpi->rate); + } else { + GST_DEBUG_OBJECT (parse, "unexpected size for SAMPLE_RATE metadata"); + } + break; + case ID_CHANNEL_INFO: + { + guint16 channels; + guint32 mask = 0; + + if (size == 6) { + CHECK (gst_byte_reader_get_uint16_le (&mbr, &channels)); + channels = channels & 0xFFF; + CHECK (gst_byte_reader_get_uint24_le (&mbr, &mask)); + } else if (size) { + CHECK (gst_byte_reader_get_uint8 (&mbr, &c)); + channels = c; + while (gst_byte_reader_get_uint8 (&mbr, &c)) + mask |= (((guint32) c) << 8); + } else { + GST_DEBUG_OBJECT (parse, "unexpected size for CHANNEL_INFO metadata"); + break; + } + wpi->channels = channels; + wpi->channel_mask = mask; + break; + } + default: + GST_LOG_OBJECT (parse, "unparsed ID 0x%x", id); + break; + } + } + + return TRUE; + + /* ERRORS */ +read_failed: + { + GST_DEBUG_OBJECT (parse, "short read while parsing metadata"); + /* let's look the other way anyway */ + return TRUE; + } +} + +/* caller ensures properly sync'ed with enough data */ +static gboolean +gst_wavpack_parse_frame_header (GstWavpackParse * parse, GstBuffer * buf, + gint skip, WavpackHeader * _wph) +{ + GstByteReader br = GST_BYTE_READER_INIT_FROM_BUFFER (buf); + WavpackHeader wph; + + g_return_val_if_fail (GST_BUFFER_SIZE (buf) >= skip + sizeof (WavpackHeader), + FALSE); + + /* marker */ + gst_byte_reader_skip_unchecked (&br, skip + 4); + + /* read */ + gst_byte_reader_get_uint32_le (&br, &wph.ckSize); + gst_byte_reader_get_uint16_le (&br, &wph.version); + gst_byte_reader_get_uint8 (&br, &wph.track_no); + gst_byte_reader_get_uint8 (&br, &wph.index_no); + gst_byte_reader_get_uint32_le (&br, &wph.total_samples); + gst_byte_reader_get_uint32_le (&br, &wph.block_index); + gst_byte_reader_get_uint32_le (&br, &wph.block_samples); + gst_byte_reader_get_uint32_le (&br, &wph.flags); + gst_byte_reader_get_uint32_le (&br, &wph.crc); + + /* dump */ + GST_LOG_OBJECT (parse, "size %d", wph.ckSize); + GST_LOG_OBJECT (parse, "version 0x%x", wph.version); + GST_LOG_OBJECT (parse, "total samples %d", wph.total_samples); + GST_LOG_OBJECT (parse, "block index %d", wph.block_index); + GST_LOG_OBJECT (parse, "block samples %d", wph.block_samples); + GST_LOG_OBJECT (parse, "flags 0x%x", wph.flags); + GST_LOG_OBJECT (parse, "crc 0x%x", wph.flags); + + if (!parse->total_samples && wph.block_index == 0 && wph.total_samples != -1) { + GST_DEBUG_OBJECT (parse, "determined duration of %u samples", + wph.total_samples); + parse->total_samples = wph.total_samples; + } + + if (_wph) + *_wph = wph; + + return TRUE; +} + +static gboolean +gst_wavpack_parse_check_valid_frame (GstBaseParse * parse, + GstBaseParseFrame * frame, guint * framesize, gint * skipsize) +{ + GstWavpackParse *wvparse = GST_WAVPACK_PARSE (parse); + GstBuffer *buf = frame->buffer; + GstByteReader reader = GST_BYTE_READER_INIT_FROM_BUFFER (buf); + gint off; + gboolean lost_sync, draining, final; + guint frmsize = 0; + WavpackHeader wph; + WavpackInfo wpi = { 0, }; + + if (G_UNLIKELY (GST_BUFFER_SIZE (buf) < sizeof (WavpackHeader))) + return FALSE; + + /* scan for 'wvpk' marker */ + off = gst_byte_reader_masked_scan_uint32 (&reader, 0xffffffff, 0x7776706b, + 0, GST_BUFFER_SIZE (buf)); + + GST_LOG_OBJECT (parse, "possible sync at buffer offset %d", off); + + /* didn't find anything that looks like a sync word, skip */ + if (off < 0) { + *skipsize = GST_BUFFER_SIZE (buf) - 3; + goto skip; + } + + /* possible frame header, but not at offset 0? skip bytes before sync */ + if (off > 0) { + *skipsize = off; + goto skip; + } + + /* make sure the values in the frame header look sane */ + gst_wavpack_parse_frame_header (wvparse, buf, 0, &wph); + frmsize = wph.ckSize + 8; + + /* need the entire frame for parsing */ + if (gst_byte_reader_get_remaining (&reader) < frmsize) + goto more; + + /* got a frame, now we can dig for some more metadata */ + GST_LOG_OBJECT (parse, "got frame"); + gst_wavpack_parse_frame_metadata (wvparse, buf, 0, &wph, &wpi); + + lost_sync = GST_BASE_PARSE_LOST_SYNC (parse); + draining = GST_BASE_PARSE_DRAINING (parse); + + while (!(final = (wph.flags & FLAG_FINAL_BLOCK)) || (lost_sync && !draining)) { + guint32 word = 0; + + GST_LOG_OBJECT (wvparse, "checking next frame syncword; " + "lost_sync: %d, draining: %d, final: %d", lost_sync, draining, final); + + if (!gst_byte_reader_skip (&reader, wph.ckSize + 8) || + !gst_byte_reader_peek_uint32_be (&reader, &word)) { + GST_DEBUG_OBJECT (wvparse, "... but not sufficient data"); + frmsize += 4; + goto more; + } else { + if (word != 0x7776706b) { + GST_DEBUG_OBJECT (wvparse, "0x%x not OK", word); + *skipsize = off + 2; + goto skip; + } + /* need to parse each frame/block for metadata if several ones */ + if (!final) { + gint av; + + GST_LOG_OBJECT (wvparse, "checking frame at offset %d (0x%x)", + frmsize, frmsize); + av = gst_byte_reader_get_remaining (&reader); + if (av < sizeof (WavpackHeader)) { + frmsize += sizeof (WavpackHeader); + goto more; + } + gst_wavpack_parse_frame_header (wvparse, buf, frmsize, &wph); + off = frmsize; + frmsize += wph.ckSize + 8; + if (av < wph.ckSize + 8) + goto more; + gst_wavpack_parse_frame_metadata (wvparse, buf, off, &wph, &wpi); + /* could also check for matching block_index and block_samples ?? */ + } + } + + /* resynced if we make it here */ + lost_sync = FALSE; + } + + /* found frame (up to final), record gathered metadata */ + wvparse->wpi = wpi; + wvparse->wph = wph; + + *framesize = frmsize; + gst_base_parse_set_min_frame_size (parse, sizeof (WavpackHeader)); + + return TRUE; + +skip: + GST_LOG_OBJECT (wvparse, "skipping %d", *skipsize); + return FALSE; + +more: + GST_LOG_OBJECT (wvparse, "need at least %u", frmsize); + gst_base_parse_set_min_frame_size (parse, frmsize); + *skipsize = 0; + return FALSE; +} + +static GstFlowReturn +gst_wavpack_parse_parse_frame (GstBaseParse * parse, GstBaseParseFrame * frame) +{ + GstWavpackParse *wvparse = GST_WAVPACK_PARSE (parse); + GstBuffer *buf = frame->buffer; + guint rate, chans, width, mask; + + /* re-use previously parsed data */ + rate = wvparse->wpi.rate; + width = wvparse->wpi.width; + chans = wvparse->wpi.channels; + mask = wvparse->wpi.channel_mask; + + GST_LOG_OBJECT (parse, "rate: %u, width: %u, chans: %u", rate, width, chans); + + GST_BUFFER_TIMESTAMP (buf) = + gst_util_uint64_scale_int (wvparse->wph.block_index, GST_SECOND, rate); + GST_BUFFER_DURATION (buf) = + gst_util_uint64_scale_int (wvparse->wph.block_index + + wvparse->wph.block_samples, GST_SECOND, rate) - + GST_BUFFER_TIMESTAMP (buf); + + if (G_UNLIKELY (wvparse->sample_rate != rate || wvparse->channels != chans + || wvparse->width != width || wvparse->channel_mask != mask)) { + GstCaps *caps; + + if (wvparse->wpi.correction) { + caps = gst_caps_new_simple ("audio/x-wavpack-correction", + "framed", G_TYPE_BOOLEAN, TRUE, NULL); + } else { + caps = gst_caps_new_simple ("audio/x-wavpack", + "channels", G_TYPE_INT, chans, + "rate", G_TYPE_INT, rate, + "width", G_TYPE_INT, width, "framed", G_TYPE_BOOLEAN, TRUE, NULL); + + if (!mask) + mask = gst_wavpack_get_default_channel_mask (wvparse->channels); + if (mask != 0) { + if (!gst_wavpack_set_channel_layout (caps, mask)) { + GST_WARNING_OBJECT (wvparse, "Failed to set channel layout"); + } + } + } + + gst_buffer_set_caps (buf, caps); + gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps); + gst_caps_unref (caps); + + wvparse->sample_rate = rate; + wvparse->channels = chans; + wvparse->width = width; + wvparse->channel_mask = mask; + + if (wvparse->total_samples) { + GST_DEBUG_OBJECT (wvparse, "setting duration"); + gst_base_parse_set_duration (GST_BASE_PARSE (wvparse), + GST_FORMAT_TIME, gst_util_uint64_scale_int (wvparse->total_samples, + GST_SECOND, wvparse->sample_rate), 0); + } + } + + return GST_FLOW_OK; +} + +static GstCaps * +gst_wavpack_parse_get_sink_caps (GstBaseParse * parse) +{ + GstCaps *peercaps; + GstCaps *res; + + peercaps = gst_pad_get_allowed_caps (GST_BASE_PARSE_SRC_PAD (parse)); + if (peercaps) { + guint i, n; + + /* Remove the framed field */ + peercaps = gst_caps_make_writable (peercaps); + n = gst_caps_get_size (peercaps); + for (i = 0; i < n; i++) { + GstStructure *s = gst_caps_get_structure (peercaps, i); + + gst_structure_remove_field (s, "framed"); + } + + res = + gst_caps_intersect_full (peercaps, + gst_pad_get_pad_template_caps (GST_BASE_PARSE_SRC_PAD (parse)), + GST_CAPS_INTERSECT_FIRST); + gst_caps_unref (peercaps); + } else { + res = + gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD + (parse))); + } + + return res; +} diff --git a/gst/audioparsers/gstwavpackparse.h b/gst/audioparsers/gstwavpackparse.h new file mode 100644 index 0000000000..7fc246eead --- /dev/null +++ b/gst/audioparsers/gstwavpackparse.h @@ -0,0 +1,134 @@ +/* GStreamer Wavpack parser + * Copyright (C) 2012 Mark Nauwelaerts + * Copyright (C) 2012 Nokia Corporation. All rights reserved. + * Contact: Stefan Kost + * + * 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_WAVPACK_PARSE_H__ +#define __GST_WAVPACK_PARSE_H__ + +#include +#include + +G_BEGIN_DECLS + +#define GST_TYPE_WAVPACK_PARSE \ + (gst_wavpack_parse_get_type()) +#define GST_WAVPACK_PARSE(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj), GST_TYPE_WAVPACK_PARSE, GstWavpackParse)) +#define GST_WAVPACK_PARSE_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass), GST_TYPE_WAVPACK_PARSE, GstWavpackParseClass)) +#define GST_IS_WAVPACK_PARSE(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj), GST_TYPE_WAVPACK_PARSE)) +#define GST_IS_WAVPACK_PARSE_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_TYPE((klass), GST_TYPE_WAVPACK_PARSE)) + + +#define ID_UNIQUE 0x3f +#define ID_OPTIONAL_DATA 0x20 +#define ID_ODD_SIZE 0x40 +#define ID_LARGE 0x80 + +#define ID_DUMMY 0x0 +#define ID_ENCODER_INFO 0x1 +#define ID_DECORR_TERMS 0x2 +#define ID_DECORR_WEIGHTS 0x3 +#define ID_DECORR_SAMPLES 0x4 +#define ID_ENTROPY_VARS 0x5 +#define ID_HYBRID_PROFILE 0x6 +#define ID_SHAPING_WEIGHTS 0x7 +#define ID_FLOAT_INFO 0x8 +#define ID_INT32_INFO 0x9 +#define ID_WV_BITSTREAM 0xa +#define ID_WVC_BITSTREAM 0xb +#define ID_WVX_BITSTREAM 0xc +#define ID_CHANNEL_INFO 0xd + +#define ID_RIFF_HEADER (ID_OPTIONAL_DATA | 0x1) +#define ID_RIFF_TRAILER (ID_OPTIONAL_DATA | 0x2) +#define ID_REPLAY_GAIN (ID_OPTIONAL_DATA | 0x3) +#define ID_CUESHEET (ID_OPTIONAL_DATA | 0x4) +#define ID_CONFIG_BLOCK (ID_OPTIONAL_DATA | 0x5) +#define ID_MD5_CHECKSUM (ID_OPTIONAL_DATA | 0x6) +#define ID_SAMPLE_RATE (ID_OPTIONAL_DATA | 0x7) + +#define FLAG_FINAL_BLOCK (1 << 12) + +typedef struct { + char ckID [4]; /* "wvpk" */ + guint32 ckSize; /* size of entire block (minus 8, of course) */ + guint16 version; /* 0x402 to 0x410 are currently valid for decode */ + guchar track_no; /* track number (0 if not used, like now) */ + guchar index_no; /* track sub-index (0 if not used, like now) */ + guint32 total_samples; /* total samples for entire file, but this is + * only valid if block_index == 0 and a value of + * -1 indicates unknown length */ + guint32 block_index; /* index of first sample in block relative to + * beginning of file (normally this would start + * at 0 for the first block) */ + guint32 block_samples; /* number of samples in this block (0 = no audio) */ + guint32 flags; /* various flags for id and decoding */ + guint32 crc; /* crc for actual decoded data */ +} WavpackHeader; + +typedef struct { + gboolean correction; + guint rate; + guint width; + guint channels; + guint channel_mask; +} WavpackInfo; + +typedef struct _GstWavpackParse GstWavpackParse; +typedef struct _GstWavpackParseClass GstWavpackParseClass; + +/** + * GstWavpackParse: + * + * The opaque GstWavpackParse object + */ +struct _GstWavpackParse { + GstBaseParse baseparse; + + /*< private >*/ + gint sample_rate; + gint channels; + gint width; + gint channel_mask; + + guint total_samples; + + WavpackHeader wph; + WavpackInfo wpi; +}; + +/** + * GstWavpackParseClass: + * @parent_class: Element parent class. + * + * The opaque GstWavpackParseClass data structure. + */ +struct _GstWavpackParseClass { + GstBaseParseClass baseparse_class; +}; + +GType gst_wavpack_parse_get_type (void); + +G_END_DECLS + +#endif /* __GST_WAVPACK_PARSE_H__ */ diff --git a/gst/audioparsers/plugin.c b/gst/audioparsers/plugin.c index ae8332d3fc..16f98ff6c7 100644 --- a/gst/audioparsers/plugin.c +++ b/gst/audioparsers/plugin.c @@ -27,6 +27,7 @@ #include "gstdcaparse.h" #include "gstflacparse.h" #include "gstmpegaudioparse.h" +#include "gstwavpackparse.h" static gboolean plugin_init (GstPlugin * plugin) @@ -45,6 +46,8 @@ plugin_init (GstPlugin * plugin) GST_RANK_PRIMARY + 1, GST_TYPE_FLAC_PARSE); ret &= gst_element_register (plugin, "mpegaudioparse", GST_RANK_PRIMARY + 2, GST_TYPE_MPEG_AUDIO_PARSE); + ret &= gst_element_register (plugin, "wavpackparse2", + GST_RANK_SECONDARY, GST_TYPE_WAVPACK_PARSE); return ret; } From 9c789aa434d43b1983d5e6b58bbad744981a6a64 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Wed, 29 Feb 2012 17:26:01 +0100 Subject: [PATCH 26/55] update for metadata API changes --- sys/v4l2/gstv4l2bufferpool.c | 18 +++++++++++++++--- sys/v4l2/gstv4l2bufferpool.h | 3 ++- sys/v4l2/gstv4l2sink.c | 4 ++-- sys/v4l2/gstv4l2src.c | 2 +- sys/ximage/ximageutil.c | 18 +++++++++++++++--- sys/ximage/ximageutil.h | 3 ++- 6 files changed, 37 insertions(+), 11 deletions(-) diff --git a/sys/v4l2/gstv4l2bufferpool.c b/sys/v4l2/gstv4l2bufferpool.c index 6c77cec422..7a68926657 100644 --- a/sys/v4l2/gstv4l2bufferpool.c +++ b/sys/v4l2/gstv4l2bufferpool.c @@ -57,17 +57,29 @@ GST_DEBUG_CATEGORY_EXTERN (v4l2_debug); /* * GstV4l2Buffer: */ +GType +gst_v4l2_meta_api_get_type (void) +{ + static volatile GType type; + static const gchar *tags[] = { "memory", NULL }; + + if (g_once_init_enter (&type)) { + GType _type = gst_meta_api_type_register ("GstV4l2MetaAPI", tags); + g_once_init_leave (&type, _type); + } + return type; +} + const GstMetaInfo * gst_v4l2_meta_get_info (void) { static const GstMetaInfo *meta_info = NULL; - static const gchar *tags[] = { "memory" }; if (meta_info == NULL) { meta_info = - gst_meta_register ("GstV4l2Meta", "GstV4l2Meta", + gst_meta_register (gst_v4l2_meta_api_get_type (), "GstV4l2Meta", sizeof (GstV4l2Meta), (GstMetaInitFunction) NULL, - (GstMetaFreeFunction) NULL, (GstMetaTransformFunction) NULL, tags); + (GstMetaFreeFunction) NULL, (GstMetaTransformFunction) NULL); } return meta_info; } diff --git a/sys/v4l2/gstv4l2bufferpool.h b/sys/v4l2/gstv4l2bufferpool.h index 12d3ccc7df..5503628174 100644 --- a/sys/v4l2/gstv4l2bufferpool.h +++ b/sys/v4l2/gstv4l2bufferpool.h @@ -79,8 +79,9 @@ struct _GstV4l2Meta { struct v4l2_buffer vbuffer; }; +GType gst_v4l2_meta_api_get_type (void); const GstMetaInfo * gst_v4l2_meta_get_info (void); -#define GST_V4L2_META_GET(buf) ((GstV4l2Meta *)gst_buffer_get_meta(buf,gst_v4l2_meta_get_info())) +#define GST_V4L2_META_GET(buf) ((GstV4l2Meta *)gst_buffer_get_meta(buf,gst_v4l2_meta_api_get_type())) #define GST_V4L2_META_ADD(buf) ((GstV4l2Meta *)gst_buffer_add_meta(buf,gst_v4l2_meta_get_info(),NULL)) GType gst_v4l2_buffer_pool_get_type (void); diff --git a/sys/v4l2/gstv4l2sink.c b/sys/v4l2/gstv4l2sink.c index 5e1e2d0d28..bcb0bb4a5f 100644 --- a/sys/v4l2/gstv4l2sink.c +++ b/sys/v4l2/gstv4l2sink.c @@ -653,8 +653,8 @@ gst_v4l2sink_propose_allocation (GstBaseSink * bsink, GstQuery * query) gst_query_set_allocation_params (query, size, 2, 0, 0, 0, pool); /* we also support various metadata */ - gst_query_add_allocation_meta (query, GST_VIDEO_META_API); - gst_query_add_allocation_meta (query, GST_VIDEO_CROP_META_API); + gst_query_add_allocation_meta (query, GST_VIDEO_META_API_TYPE); + gst_query_add_allocation_meta (query, GST_VIDEO_CROP_META_API_TYPE); if (pool) gst_object_unref (pool); diff --git a/sys/v4l2/gstv4l2src.c b/sys/v4l2/gstv4l2src.c index 4e29931ba3..5722bdea84 100644 --- a/sys/v4l2/gstv4l2src.c +++ b/sys/v4l2/gstv4l2src.c @@ -571,7 +571,7 @@ gst_v4l2src_decide_allocation (GstBaseSrc * bsrc, GstQuery * query) alignment); /* if downstream supports video metadata, add this to the pool config */ - if (gst_query_has_allocation_meta (query, GST_VIDEO_META_API)) + if (gst_query_has_allocation_meta (query, GST_VIDEO_META_API_TYPE)) gst_buffer_pool_config_add_option (config, GST_BUFFER_POOL_OPTION_VIDEO_META); diff --git a/sys/ximage/ximageutil.c b/sys/ximage/ximageutil.c index 84cc01a5ae..907aa3aefd 100644 --- a/sys/ximage/ximageutil.c +++ b/sys/ximage/ximageutil.c @@ -23,17 +23,29 @@ #include "ximageutil.h" +GType +gst_meta_ximage_api_get_type (void) +{ + static volatile GType type; + static const gchar *tags[] = { "memory", NULL }; + + if (g_once_init_enter (&type)) { + GType _type = gst_meta_api_type_register ("GstMetaXImageSrcAPI", tags); + g_once_init_leave (&type, _type); + } + return type; +} + const GstMetaInfo * gst_meta_ximage_get_info (void) { static const GstMetaInfo *meta_ximage_info = NULL; - static const gchar *tags[] = { "memory", NULL }; if (meta_ximage_info == NULL) { meta_ximage_info = - gst_meta_register ("GstMetaXImageSrc", "GstMetaXImageSrc", + gst_meta_register (gst_meta_ximage_api_get_type (), "GstMetaXImageSrc", sizeof (GstMetaXImage), (GstMetaInitFunction) NULL, - (GstMetaFreeFunction) NULL, (GstMetaTransformFunction) NULL, tags); + (GstMetaFreeFunction) NULL, (GstMetaTransformFunction) NULL); } return meta_ximage_info; } diff --git a/sys/ximage/ximageutil.h b/sys/ximage/ximageutil.h index aa674de8fa..aa03e7a857 100644 --- a/sys/ximage/ximageutil.h +++ b/sys/ximage/ximageutil.h @@ -160,8 +160,9 @@ struct _GstMetaXImage { BufferReturnFunc return_func; }; +GType gst_meta_ximage_api_get_type (void); const GstMetaInfo * gst_meta_ximage_get_info (void); -#define GST_META_XIMAGE_GET(buf) ((GstMetaXImage *)gst_buffer_get_meta(buf,gst_meta_ximage_get_info())) +#define GST_META_XIMAGE_GET(buf) ((GstMetaXImage *)gst_buffer_get_meta(buf,gst_meta_ximage_api_get_type())) #define GST_META_XIMAGE_ADD(buf) ((GstMetaXImage *)gst_buffer_add_meta(buf,gst_meta_ximage_get_info(),NULL)) GstBuffer *gst_ximageutil_ximage_new (GstXContext *xcontext, From 606a88f197eebc53ff05c5a78f911154d23b778f Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Wed, 29 Feb 2012 22:24:46 +0100 Subject: [PATCH 27/55] gdkpixbufdec: port to 0.11 --- ext/gdk_pixbuf/gstgdkpixbuf.c | 223 ++++++++++++++++++++++++---------- ext/gdk_pixbuf/gstgdkpixbuf.h | 5 +- 2 files changed, 161 insertions(+), 67 deletions(-) diff --git a/ext/gdk_pixbuf/gstgdkpixbuf.c b/ext/gdk_pixbuf/gstgdkpixbuf.c index a65d2e8632..bac7df2032 100644 --- a/ext/gdk_pixbuf/gstgdkpixbuf.c +++ b/ext/gdk_pixbuf/gstgdkpixbuf.c @@ -67,7 +67,8 @@ static GstStaticPadTemplate gst_gdk_pixbuf_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, - GST_STATIC_CAPS (GST_VIDEO_CAPS_RGB "; " GST_VIDEO_CAPS_RGBA) + GST_STATIC_CAPS (GST_VIDEO_CAPS_MAKE ("RGB") "; " + GST_VIDEO_CAPS_MAKE ("RGBA")) ); static void gst_gdk_pixbuf_set_property (GObject * object, guint prop_id, @@ -77,23 +78,24 @@ static void gst_gdk_pixbuf_get_property (GObject * object, guint prop_id, static GstStateChangeReturn gst_gdk_pixbuf_change_state (GstElement * element, GstStateChange transition); -static GstFlowReturn gst_gdk_pixbuf_chain (GstPad * pad, GstBuffer * buffer); -static gboolean gst_gdk_pixbuf_sink_event (GstPad * pad, GstEvent * event); +static GstFlowReturn gst_gdk_pixbuf_chain (GstPad * pad, GstObject * parent, + GstBuffer * buffer); +static gboolean gst_gdk_pixbuf_sink_event (GstPad * pad, GstObject * parent, + GstEvent * event); #ifdef enable_typefind static void gst_gdk_pixbuf_type_find (GstTypeFind * tf, gpointer ignore); #endif -GST_BOILERPLATE (GstGdkPixbuf, gst_gdk_pixbuf, GstElement, GST_TYPE_ELEMENT); +#define gst_gdk_pixbuf_parent_class parent_class +G_DEFINE_TYPE (GstGdkPixbuf, gst_gdk_pixbuf, GST_TYPE_ELEMENT); static gboolean -gst_gdk_pixbuf_sink_setcaps (GstPad * pad, GstCaps * caps) +gst_gdk_pixbuf_sink_setcaps (GstGdkPixbuf * filter, GstCaps * caps) { - GstGdkPixbuf *filter; const GValue *framerate; GstStructure *s; - filter = GST_GDK_PIXBUF (GST_PAD_PARENT (pad)); s = gst_caps_get_structure (caps, 0); if ((framerate = gst_structure_get_value (s, "framerate")) != NULL) { @@ -112,7 +114,7 @@ gst_gdk_pixbuf_sink_setcaps (GstPad * pad, GstCaps * caps) } static GstCaps * -gst_gdk_pixbuf_get_capslist (void) +gst_gdk_pixbuf_get_capslist (GstCaps * filter) { GSList *slist; GSList *slist0; @@ -132,7 +134,7 @@ gst_gdk_pixbuf_get_capslist (void) mimetypes = gdk_pixbuf_format_get_mime_types (pixbuf_format); for (mimetype = mimetypes; *mimetype; mimetype++) { - gst_caps_append_structure (capslist, gst_structure_new (*mimetype, NULL)); + gst_caps_append_structure (capslist, gst_structure_new_empty (*mimetype)); } g_strfreev (mimetypes); } @@ -143,29 +145,43 @@ gst_gdk_pixbuf_get_capslist (void) gst_caps_unref (tmpl_caps); gst_caps_unref (capslist); + + if (filter && return_caps) { + GstCaps *temp; + + temp = gst_caps_intersect (return_caps, filter); + gst_caps_unref (return_caps); + return_caps = temp; + } + return return_caps; } -static GstCaps * -gst_gdk_pixbuf_sink_getcaps (GstPad * pad) +static gboolean +gst_gdk_pixbuf_sink_query (GstPad * pad, GstObject * parent, GstQuery * query) { - return gst_gdk_pixbuf_get_capslist (); + gboolean res; + + switch (GST_QUERY_TYPE (query)) { + case GST_QUERY_CAPS: + { + GstCaps *filter, *caps; + + gst_query_parse_caps (query, &filter); + caps = gst_gdk_pixbuf_get_capslist (filter); + gst_query_set_caps_result (query, caps); + gst_caps_unref (caps); + + res = TRUE; + break; + } + default: + res = gst_pad_query_default (pad, parent, query); + break; + } + return res; } -static void -gst_gdk_pixbuf_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_gdk_pixbuf_src_template)); - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&gst_gdk_pixbuf_sink_template)); - gst_element_class_set_details_simple (element_class, - "GdkPixbuf image decoder", "Codec/Decoder/Image", - "Decodes images in a video stream using GdkPixbuf", - "David A. Schleef , Renato Filho "); -} /* initialize the plugin's class */ static void @@ -187,17 +203,24 @@ gst_gdk_pixbuf_class_init (GstGdkPixbufClass * klass) gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_gdk_pixbuf_change_state); + + gst_element_class_add_pad_template (gstelement_class, + gst_static_pad_template_get (&gst_gdk_pixbuf_src_template)); + gst_element_class_add_pad_template (gstelement_class, + gst_static_pad_template_get (&gst_gdk_pixbuf_sink_template)); + gst_element_class_set_details_simple (gstelement_class, + "GdkPixbuf image decoder", "Codec/Decoder/Image", + "Decodes images in a video stream using GdkPixbuf", + "David A. Schleef , Renato Filho "); } static void -gst_gdk_pixbuf_init (GstGdkPixbuf * filter, GstGdkPixbufClass * klass) +gst_gdk_pixbuf_init (GstGdkPixbuf * filter) { filter->sinkpad = gst_pad_new_from_static_template (&gst_gdk_pixbuf_sink_template, "sink"); - gst_pad_set_setcaps_function (filter->sinkpad, - GST_DEBUG_FUNCPTR (gst_gdk_pixbuf_sink_setcaps)); - gst_pad_set_getcaps_function (filter->sinkpad, - GST_DEBUG_FUNCPTR (gst_gdk_pixbuf_sink_getcaps)); + gst_pad_set_query_function (filter->sinkpad, + GST_DEBUG_FUNCPTR (gst_gdk_pixbuf_sink_query)); gst_pad_set_chain_function (filter->sinkpad, GST_DEBUG_FUNCPTR (gst_gdk_pixbuf_chain)); gst_pad_set_event_function (filter->sinkpad, @@ -213,6 +236,55 @@ gst_gdk_pixbuf_init (GstGdkPixbuf * filter, GstGdkPixbufClass * klass) filter->pixbuf_loader = NULL; } +static gboolean +gst_gdk_pixbuf_setup_pool (GstGdkPixbuf * filter, GstVideoInfo * info) +{ + GstCaps *target; + GstQuery *query; + GstBufferPool *pool = NULL; + guint size, min, max, prefix, alignment; + + target = gst_pad_get_current_caps (filter->srcpad); + + /* try to get a bufferpool now */ + /* find a pool for the negotiated caps now */ + query = gst_query_new_allocation (target, TRUE); + + if (gst_pad_peer_query (filter->srcpad, query)) { + /* we got configuration from our peer, parse them */ + gst_query_parse_allocation_params (query, &size, &min, &max, &prefix, + &alignment, &pool); + } else { + size = info->size; + min = max = 0; + prefix = 0; + alignment = 0; + } + + if (pool == NULL) { + GstStructure *config; + + /* we did not get a pool, make one ourselves then */ + pool = gst_buffer_pool_new (); + + config = gst_buffer_pool_get_config (pool); + gst_buffer_pool_config_set (config, target, size, min, max, prefix, + alignment); + gst_buffer_pool_set_config (pool, config); + } + + if (filter->pool) + gst_object_unref (filter->pool); + filter->pool = pool; + + /* and activate */ + gst_buffer_pool_set_active (filter->pool, TRUE); + + gst_caps_unref (target); + + return TRUE; +} + static GstFlowReturn gst_gdk_pixbuf_flush (GstGdkPixbuf * filter) { @@ -225,44 +297,50 @@ gst_gdk_pixbuf_flush (GstGdkPixbuf * filter) GstFlowReturn ret; GstCaps *caps = NULL; gint n_channels; + GstVideoFrame frame; pixbuf = gdk_pixbuf_loader_get_pixbuf (filter->pixbuf_loader); if (pixbuf == NULL) goto no_pixbuf; - if (filter->image_size == 0) { + if (filter->width == 0) { + GstVideoInfo info; + GstVideoFormat fmt; + filter->width = gdk_pixbuf_get_width (pixbuf); filter->height = gdk_pixbuf_get_height (pixbuf); - filter->rowstride = gdk_pixbuf_get_rowstride (pixbuf); - filter->image_size = filter->rowstride * filter->height; + GST_DEBUG ("Set size to %dx%d", filter->width, filter->height); n_channels = gdk_pixbuf_get_n_channels (pixbuf); switch (n_channels) { case 3: - caps = gst_caps_from_string (GST_VIDEO_CAPS_RGB); + fmt = GST_VIDEO_FORMAT_RGB; break; case 4: - caps = gst_caps_from_string (GST_VIDEO_CAPS_RGBA); + fmt = GST_VIDEO_FORMAT_RGBA; break; default: goto channels_not_supported; } - gst_caps_set_simple (caps, - "width", G_TYPE_INT, filter->width, - "height", G_TYPE_INT, filter->height, - "framerate", GST_TYPE_FRACTION, filter->framerate_numerator, - filter->framerate_denominator, NULL); - GST_DEBUG ("Set size to %dx%d", filter->width, filter->height); + gst_video_info_init (&info); + gst_video_info_set_format (&info, fmt, filter->width, filter->height); + info.fps_n = filter->framerate_numerator; + info.fps_d = filter->framerate_denominator; + caps = gst_video_info_to_caps (&info); + + filter->channels = n_channels; + filter->rowstride = GST_VIDEO_INFO_COMP_STRIDE (&info, 0); + filter->info = info; + gst_pad_set_caps (filter->srcpad, caps); gst_caps_unref (caps); + + gst_gdk_pixbuf_setup_pool (filter, &info); } - ret = gst_pad_alloc_buffer_and_set_caps (filter->srcpad, - GST_BUFFER_OFFSET_NONE, - filter->image_size, GST_PAD_CAPS (filter->srcpad), &outbuf); - + ret = gst_buffer_pool_acquire_buffer (filter->pool, &outbuf, NULL); if (ret != GST_FLOW_OK) goto no_buffer; @@ -271,16 +349,19 @@ gst_gdk_pixbuf_flush (GstGdkPixbuf * filter) in_pix = gdk_pixbuf_get_pixels (pixbuf); in_rowstride = gdk_pixbuf_get_rowstride (pixbuf); - out_pix = GST_BUFFER_DATA (outbuf); - /* FIXME, last line might not have rowstride pixels */ + gst_video_frame_map (&frame, &filter->info, outbuf, GST_MAP_WRITE); + out_pix = GST_VIDEO_FRAME_COMP_DATA (&frame, 0); + for (y = 0; y < filter->height; y++) { - memcpy (out_pix, in_pix, filter->rowstride); + memcpy (out_pix, in_pix, filter->width * filter->channels); in_pix += in_rowstride; out_pix += filter->rowstride; } - GST_DEBUG ("pushing... %d bytes", GST_BUFFER_SIZE (outbuf)); + gst_video_frame_unmap (&frame); + + GST_DEBUG ("pushing... %d bytes", gst_buffer_get_size (outbuf)); ret = gst_pad_push (filter->srcpad, outbuf); if (ret != GST_FLOW_OK) @@ -308,15 +389,24 @@ no_buffer: } static gboolean -gst_gdk_pixbuf_sink_event (GstPad * pad, GstEvent * event) +gst_gdk_pixbuf_sink_event (GstPad * pad, GstObject * parent, GstEvent * event) { GstFlowReturn res = GST_FLOW_OK; gboolean ret = TRUE; GstGdkPixbuf *pixbuf; - pixbuf = GST_GDK_PIXBUF (gst_pad_get_parent (pad)); + pixbuf = GST_GDK_PIXBUF (parent); switch (GST_EVENT_TYPE (event)) { + case GST_EVENT_CAPS: + { + GstCaps *caps; + + gst_event_parse_caps (event, &caps); + res = gst_gdk_pixbuf_sink_setcaps (pixbuf, caps); + gst_event_unref (event); + break; + } case GST_EVENT_EOS: if (pixbuf->pixbuf_loader != NULL) { gdk_pixbuf_loader_close (pixbuf->pixbuf_loader, NULL); @@ -332,7 +422,7 @@ gst_gdk_pixbuf_sink_event (GstPad * pad, GstEvent * event) } } break; - case GST_EVENT_NEWSEGMENT: + case GST_EVENT_SEGMENT: case GST_EVENT_FLUSH_STOP: if (pixbuf->pixbuf_loader != NULL) { gdk_pixbuf_loader_close (pixbuf->pixbuf_loader, NULL); @@ -345,27 +435,24 @@ gst_gdk_pixbuf_sink_event (GstPad * pad, GstEvent * event) } if (res == GST_FLOW_OK) { - ret = gst_pad_event_default (pad, event); + ret = gst_pad_event_default (pad, parent, event); } else { ret = FALSE; } - gst_object_unref (pixbuf); - return ret; } static GstFlowReturn -gst_gdk_pixbuf_chain (GstPad * pad, GstBuffer * buf) +gst_gdk_pixbuf_chain (GstPad * pad, GstObject * parent, GstBuffer * buf) { GstGdkPixbuf *filter; GstFlowReturn ret = GST_FLOW_OK; GError *error = NULL; GstClockTime timestamp; - guint8 *data; - guint size; + GstMapInfo map; - filter = GST_GDK_PIXBUF (gst_pad_get_parent (pad)); + filter = GST_GDK_PIXBUF (parent); timestamp = GST_BUFFER_TIMESTAMP (buf); @@ -378,11 +465,11 @@ gst_gdk_pixbuf_chain (GstPad * pad, GstBuffer * buf) if (filter->pixbuf_loader == NULL) filter->pixbuf_loader = gdk_pixbuf_loader_new (); - data = GST_BUFFER_DATA (buf); - size = GST_BUFFER_SIZE (buf); + gst_buffer_map (buf, &map, GST_MAP_READ); - GST_LOG_OBJECT (filter, "Writing buffer size %d", size); - if (!gdk_pixbuf_loader_write (filter->pixbuf_loader, data, size, &error)) + GST_LOG_OBJECT (filter, "Writing buffer size %d", (gint) map.size); + if (!gdk_pixbuf_loader_write (filter->pixbuf_loader, map.data, map.size, + &error)) goto error; /* packetised mode? */ @@ -393,8 +480,8 @@ gst_gdk_pixbuf_chain (GstPad * pad, GstBuffer * buf) filter->pixbuf_loader = NULL; } + gst_buffer_unmap (buf, &map); gst_buffer_unref (buf); - gst_object_unref (filter); return ret; @@ -404,8 +491,8 @@ error: GST_ELEMENT_ERROR (filter, STREAM, DECODE, (NULL), ("gdk_pixbuf_loader_write error: %s", error->message)); g_error_free (error); + gst_buffer_unmap (buf, &map); gst_buffer_unref (buf); - gst_object_unref (filter); return GST_FLOW_ERROR; } } @@ -462,6 +549,10 @@ gst_gdk_pixbuf_change_state (GstElement * element, GstStateChange transition) case GST_STATE_CHANGE_PAUSED_TO_READY: dec->framerate_numerator = 0; dec->framerate_denominator = 0; + if (dec->pool) { + gst_buffer_pool_set_active (dec->pool, FALSE); + gst_object_replace ((GstObject **) & dec->pool, NULL); + } break; default: break; diff --git a/ext/gdk_pixbuf/gstgdkpixbuf.h b/ext/gdk_pixbuf/gstgdkpixbuf.h index a4c6ee557c..a64b7239cd 100644 --- a/ext/gdk_pixbuf/gstgdkpixbuf.h +++ b/ext/gdk_pixbuf/gstgdkpixbuf.h @@ -52,7 +52,10 @@ struct _GstGdkPixbuf int width; int height; int rowstride; - unsigned int image_size; + int channels; + + GstVideoInfo info; + GstBufferPool *pool; gint framerate_numerator; gint framerate_denominator; From 84881b5399c746c71eb9995b1ffc517d7e313a39 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Wed, 29 Feb 2012 22:25:23 +0100 Subject: [PATCH 28/55] gdkpixbufscale: port to 0.11 --- ext/gdk_pixbuf/pixbufscale.c | 187 ++++++++++++++--------------------- ext/gdk_pixbuf/pixbufscale.h | 6 +- 2 files changed, 79 insertions(+), 114 deletions(-) diff --git a/ext/gdk_pixbuf/pixbufscale.c b/ext/gdk_pixbuf/pixbufscale.c index bedd982524..35af64e6dd 100644 --- a/ext/gdk_pixbuf/pixbufscale.c +++ b/ext/gdk_pixbuf/pixbufscale.c @@ -57,14 +57,14 @@ static GstStaticPadTemplate gst_pixbufscale_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, - GST_STATIC_CAPS (GST_VIDEO_CAPS_RGB) + GST_STATIC_CAPS (GST_VIDEO_CAPS_MAKE ("RGB")) ); static GstStaticPadTemplate gst_pixbufscale_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, - GST_STATIC_CAPS (GST_VIDEO_CAPS_RGB) + GST_STATIC_CAPS (GST_VIDEO_CAPS_MAKE ("RGB")) ); #define GST_TYPE_PIXBUFSCALE_METHOD (gst_pixbufscale_method_get_type()) @@ -93,28 +93,32 @@ static void gst_pixbufscale_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static GstCaps *gst_pixbufscale_transform_caps (GstBaseTransform * trans, - GstPadDirection direction, GstCaps * caps); -static gboolean gst_pixbufscale_set_caps (GstBaseTransform * trans, - GstCaps * in, GstCaps * out); -static gboolean gst_pixbufscale_get_unit_size (GstBaseTransform * trans, - GstCaps * caps, guint * size); -static void gst_pixbufscale_fixate_caps (GstBaseTransform * base, + GstPadDirection direction, GstCaps * caps, GstCaps * filter); +static gboolean gst_pixbufscale_set_info (GstVideoFilter * filter, + GstCaps * in, GstVideoInfo * in_info, GstCaps * out, + GstVideoInfo * out_info); +static GstCaps *gst_pixbufscale_fixate_caps (GstBaseTransform * base, GstPadDirection direction, GstCaps * caps, GstCaps * othercaps); -static GstFlowReturn gst_pixbufscale_transform (GstBaseTransform * trans, - GstBuffer * in, GstBuffer * out); -static gboolean gst_pixbufscale_handle_src_event (GstPad * pad, +static GstFlowReturn gst_pixbufscale_transform_frame (GstVideoFilter * filter, + GstVideoFrame * in, GstVideoFrame * out); +static gboolean gst_pixbufscale_handle_src_event (GstBaseTransform * trans, GstEvent * event); - -static gboolean parse_caps (GstCaps * caps, gint * width, gint * height); - -GST_BOILERPLATE (GstPixbufScale, gst_pixbufscale, GstBaseTransform, - GST_TYPE_BASE_TRANSFORM); +#define gst_pixbufscale_parent_class parent_class +G_DEFINE_TYPE (GstPixbufScale, gst_pixbufscale, GST_TYPE_VIDEO_FILTER); static void -gst_pixbufscale_base_init (gpointer g_class) +gst_pixbufscale_class_init (GstPixbufScaleClass * klass) { - GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); + GObjectClass *gobject_class; + GstElementClass *element_class; + GstBaseTransformClass *trans_class; + GstVideoFilterClass *filter_class; + + gobject_class = (GObjectClass *) klass; + element_class = (GstElementClass *) klass; + trans_class = (GstBaseTransformClass *) klass; + filter_class = (GstVideoFilterClass *) klass; gst_element_class_set_details_simple (element_class, "GdkPixbuf image scaler", "Filter/Effect/Video", "Resizes video", @@ -126,16 +130,6 @@ gst_pixbufscale_base_init (gpointer g_class) gst_static_pad_template_get (&gst_pixbufscale_src_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_pixbufscale_sink_template)); -} - -static void -gst_pixbufscale_class_init (GstPixbufScaleClass * klass) -{ - GObjectClass *gobject_class; - GstBaseTransformClass *trans_class; - - gobject_class = (GObjectClass *) klass; - trans_class = (GstBaseTransformClass *) klass; gobject_class->set_property = gst_pixbufscale_set_property; gobject_class->get_property = gst_pixbufscale_get_property; @@ -148,26 +142,21 @@ gst_pixbufscale_class_init (GstPixbufScaleClass * klass) trans_class->transform_caps = GST_DEBUG_FUNCPTR (gst_pixbufscale_transform_caps); - trans_class->set_caps = GST_DEBUG_FUNCPTR (gst_pixbufscale_set_caps); - trans_class->get_unit_size = - GST_DEBUG_FUNCPTR (gst_pixbufscale_get_unit_size); - trans_class->transform = GST_DEBUG_FUNCPTR (gst_pixbufscale_transform); trans_class->fixate_caps = GST_DEBUG_FUNCPTR (gst_pixbufscale_fixate_caps); + trans_class->src_event = GST_DEBUG_FUNCPTR (gst_pixbufscale_handle_src_event); trans_class->passthrough_on_same_caps = TRUE; + filter_class->set_info = GST_DEBUG_FUNCPTR (gst_pixbufscale_set_info); + filter_class->transform_frame = + GST_DEBUG_FUNCPTR (gst_pixbufscale_transform_frame); + parent_class = g_type_class_peek_parent (klass); } static void -gst_pixbufscale_init (GstPixbufScale * pixbufscale, - GstPixbufScaleClass * kclass) +gst_pixbufscale_init (GstPixbufScale * pixbufscale) { - GstBaseTransform *trans; - GST_DEBUG_OBJECT (pixbufscale, "_init"); - trans = GST_BASE_TRANSFORM (pixbufscale); - - gst_pad_set_event_function (trans->srcpad, gst_pixbufscale_handle_src_event); pixbufscale->method = GST_PIXBUFSCALE_TILES; pixbufscale->gdk_method = GDK_INTERP_TILES; @@ -227,7 +216,7 @@ gst_pixbufscale_get_property (GObject * object, guint prop_id, GValue * value, static GstCaps * gst_pixbufscale_transform_caps (GstBaseTransform * trans, - GstPadDirection direction, GstCaps * caps) + GstPadDirection direction, GstCaps * caps, GstCaps * filter) { GstCaps *ret; int i; @@ -243,76 +232,46 @@ gst_pixbufscale_transform_caps (GstBaseTransform * trans, gst_structure_remove_field (structure, "pixel-aspect-ratio"); } + if (filter) { + GstCaps *intersection; + + intersection = + gst_caps_intersect_full (filter, ret, GST_CAPS_INTERSECT_FIRST); + gst_caps_unref (ret); + ret = intersection; + } + GST_DEBUG_OBJECT (trans, "returning caps: %" GST_PTR_FORMAT, ret); return ret; } static gboolean -parse_caps (GstCaps * caps, gint * width, gint * height) +gst_pixbufscale_set_info (GstVideoFilter * filter, GstCaps * in, + GstVideoInfo * in_info, GstCaps * out, GstVideoInfo * out_info) { - gboolean ret; - GstStructure *structure; + if (in_info->width == out_info->width && in_info->height == out_info->height) { + gst_base_transform_set_passthrough (GST_BASE_TRANSFORM (filter), TRUE); + } else { + gst_base_transform_set_passthrough (GST_BASE_TRANSFORM (filter), FALSE); + } - structure = gst_caps_get_structure (caps, 0); - ret = gst_structure_get_int (structure, "width", width); - ret &= gst_structure_get_int (structure, "height", height); - - return ret; -} - -static gboolean -gst_pixbufscale_set_caps (GstBaseTransform * trans, GstCaps * in, GstCaps * out) -{ - GstPixbufScale *pixbufscale; - gboolean ret; - - pixbufscale = GST_PIXBUFSCALE (trans); - ret = parse_caps (in, &pixbufscale->from_width, &pixbufscale->from_height); - ret &= parse_caps (out, &pixbufscale->to_width, &pixbufscale->to_height); - if (!ret) - goto done; - - pixbufscale->from_stride = GST_ROUND_UP_4 (pixbufscale->from_width * 3); - pixbufscale->from_buf_size = - pixbufscale->from_stride * pixbufscale->from_height; - - pixbufscale->to_stride = GST_ROUND_UP_4 (pixbufscale->to_width * 3); - pixbufscale->to_buf_size = pixbufscale->to_stride * pixbufscale->to_height; - - GST_DEBUG_OBJECT (pixbufscale, "from=%dx%d, size %d -> to=%dx%d, size %d", - pixbufscale->from_width, pixbufscale->from_height, - pixbufscale->from_buf_size, pixbufscale->to_width, pixbufscale->to_height, - pixbufscale->to_buf_size); - -done: - return ret; -} - - -static gboolean -gst_pixbufscale_get_unit_size (GstBaseTransform * trans, - GstCaps * caps, guint * size) -{ - gint width, height; - - g_assert (size); - - if (!parse_caps (caps, &width, &height)) - return FALSE; - - *size = GST_ROUND_UP_4 (width * 3) * height; + GST_DEBUG_OBJECT (filter, "from=%dx%d, size %" G_GSIZE_FORMAT + " -> to=%dx%d, size %" G_GSIZE_FORMAT, + in_info->width, in_info->height, in_info->size, + out_info->width, out_info->height, out_info->size); return TRUE; } -static void +static GstCaps * gst_pixbufscale_fixate_caps (GstBaseTransform * base, GstPadDirection direction, GstCaps * caps, GstCaps * othercaps) { GstStructure *ins, *outs; const GValue *from_par, *to_par; - g_return_if_fail (gst_caps_is_fixed (caps)); + othercaps = gst_caps_make_writable (othercaps); + gst_caps_truncate (othercaps); GST_DEBUG_OBJECT (base, "trying to fixate othercaps %" GST_PTR_FORMAT " based on caps %" GST_PTR_FORMAT, othercaps, caps); @@ -337,7 +296,7 @@ gst_pixbufscale_fixate_caps (GstBaseTransform * base, GstPadDirection direction, if (count == 2) { GST_DEBUG_OBJECT (base, "dimensions already set to %dx%d, not fixating", w, h); - return; + goto done; } gst_structure_get_int (ins, "width", &from_w); @@ -396,33 +355,39 @@ gst_pixbufscale_fixate_caps (GstBaseTransform * base, GstPadDirection direction, } } +done: GST_DEBUG_OBJECT (base, "fixated othercaps to %" GST_PTR_FORMAT, othercaps); + + return othercaps; } static GstFlowReturn -gst_pixbufscale_transform (GstBaseTransform * trans, - GstBuffer * in, GstBuffer * out) +gst_pixbufscale_transform_frame (GstVideoFilter * filter, + GstVideoFrame * in, GstVideoFrame * out) { GstPixbufScale *scale; GdkPixbuf *src_pixbuf, *dest_pixbuf; - scale = GST_PIXBUFSCALE (trans); + scale = GST_PIXBUFSCALE (filter); src_pixbuf = - gdk_pixbuf_new_from_data (GST_BUFFER_DATA (in), GDK_COLORSPACE_RGB, FALSE, - 8, scale->from_width, scale->from_height, - GST_RGB24_ROWSTRIDE (scale->from_width), NULL, NULL); + gdk_pixbuf_new_from_data (GST_VIDEO_FRAME_COMP_DATA (in, 0), + GDK_COLORSPACE_RGB, FALSE, 8, GST_VIDEO_FRAME_WIDTH (in), + GST_VIDEO_FRAME_HEIGHT (in), + GST_VIDEO_FRAME_COMP_STRIDE (in, 0), NULL, NULL); dest_pixbuf = - gdk_pixbuf_new_from_data (GST_BUFFER_DATA (out), GDK_COLORSPACE_RGB, - FALSE, 8, scale->to_width, scale->to_height, - GST_RGB24_ROWSTRIDE (scale->to_width), NULL, NULL); + gdk_pixbuf_new_from_data (GST_VIDEO_FRAME_COMP_DATA (out, 0), + GDK_COLORSPACE_RGB, FALSE, 8, GST_VIDEO_FRAME_WIDTH (out), + GST_VIDEO_FRAME_HEIGHT (out), + GST_VIDEO_FRAME_COMP_STRIDE (out, 0), NULL, NULL); gdk_pixbuf_scale (src_pixbuf, dest_pixbuf, 0, 0, - scale->to_width, - scale->to_height, 0, 0, - (double) scale->to_width / scale->from_width, - (double) scale->to_height / scale->from_height, scale->gdk_method); + GST_VIDEO_FRAME_WIDTH (out), + GST_VIDEO_FRAME_HEIGHT (out), 0, 0, + (double) GST_VIDEO_FRAME_WIDTH (out) / GST_VIDEO_FRAME_WIDTH (in), + (double) GST_VIDEO_FRAME_HEIGHT (out) / GST_VIDEO_FRAME_HEIGHT (in), + scale->gdk_method); g_object_unref (src_pixbuf); g_object_unref (dest_pixbuf); @@ -431,14 +396,14 @@ gst_pixbufscale_transform (GstBaseTransform * trans, } static gboolean -gst_pixbufscale_handle_src_event (GstPad * pad, GstEvent * event) +gst_pixbufscale_handle_src_event (GstBaseTransform * trans, GstEvent * event) { GstPixbufScale *pixbufscale; gboolean ret; double a; GstStructure *structure; - pixbufscale = GST_PIXBUFSCALE (gst_pad_get_parent (pad)); + pixbufscale = GST_PIXBUFSCALE (trans); GST_DEBUG_OBJECT (pixbufscale, "handling %s event", GST_EVENT_TYPE_NAME (event)); @@ -462,9 +427,7 @@ gst_pixbufscale_handle_src_event (GstPad * pad, GstEvent * event) break; } - ret = gst_pad_event_default (pad, event); - - gst_object_unref (pixbufscale); + ret = GST_BASE_TRANSFORM_CLASS (parent_class)->src_event (trans, event); return ret; } diff --git a/ext/gdk_pixbuf/pixbufscale.h b/ext/gdk_pixbuf/pixbufscale.h index 3e40477724..0b3afcb561 100644 --- a/ext/gdk_pixbuf/pixbufscale.h +++ b/ext/gdk_pixbuf/pixbufscale.h @@ -25,6 +25,8 @@ #include #include +#include + #include G_BEGIN_DECLS @@ -52,7 +54,7 @@ typedef struct _GstPixbufScale GstPixbufScale; typedef struct _GstPixbufScaleClass GstPixbufScaleClass; struct _GstPixbufScale { - GstBaseTransform element; + GstVideoFilter element; /* video state */ gint to_width; @@ -72,7 +74,7 @@ struct _GstPixbufScale { }; struct _GstPixbufScaleClass { - GstBaseTransformClass parent_class; + GstVideoFilterClass parent_class; }; GType gst_pixbufscale_get_type(void); From 06bf7481b139e4fc051cbdeb46bb0686e051cb67 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Wed, 29 Feb 2012 22:28:01 +0100 Subject: [PATCH 29/55] gdkpixbufsink: port to 0.11 --- configure.ac | 2 +- ext/gdk_pixbuf/gstgdkpixbufsink.c | 76 ++++++++++++++++--------------- ext/gdk_pixbuf/gstgdkpixbufsink.h | 4 +- 3 files changed, 43 insertions(+), 39 deletions(-) diff --git a/configure.ac b/configure.ac index fe60f9cade..25e97b60d3 100644 --- a/configure.ac +++ b/configure.ac @@ -314,7 +314,7 @@ dnl Make sure you have a space before and after all plugins GST_PLUGINS_NONPORTED="deinterlace interleave flx \ imagefreeze smpte \ videobox \ - cairo cairo_gobject dv1394 gdk_pixbuf \ + cairo cairo_gobject dv1394 \ oss oss4 \ wavpack \ osx_video osx_audio " diff --git a/ext/gdk_pixbuf/gstgdkpixbufsink.c b/ext/gdk_pixbuf/gstgdkpixbufsink.c index 9235386779..07af0de56a 100644 --- a/ext/gdk_pixbuf/gstgdkpixbufsink.c +++ b/ext/gdk_pixbuf/gstgdkpixbufsink.c @@ -110,8 +110,7 @@ enum }; -GST_BOILERPLATE (GstGdkPixbufSink, gst_gdk_pixbuf_sink, GstVideoSink, - GST_TYPE_VIDEO_SINK); +G_DEFINE_TYPE (GstGdkPixbufSink, gst_gdk_pixbuf_sink, GST_TYPE_VIDEO_SINK); static void gst_gdk_pixbuf_sink_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); @@ -129,18 +128,20 @@ static GstFlowReturn gst_gdk_pixbuf_sink_preroll (GstBaseSink * bsink, static GdkPixbuf *gst_gdk_pixbuf_sink_get_pixbuf_from_buffer (GstGdkPixbufSink * sink, GstBuffer * buf); -#define WxH ", width = (int) [ 16, 4096 ], height = (int) [ 16, 4096 ]" - static GstStaticPadTemplate pixbufsink_sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, - GST_STATIC_CAPS (GST_VIDEO_CAPS_RGB WxH ";" GST_VIDEO_CAPS_RGBA WxH)); + GST_STATIC_CAPS (GST_VIDEO_CAPS_MAKE ("RGB") ";" + GST_VIDEO_CAPS_MAKE ("RGBA")) + ); static void -gst_gdk_pixbuf_sink_base_init (gpointer g_class) +gst_gdk_pixbuf_sink_class_init (GstGdkPixbufSinkClass * klass) { - GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); + GstBaseSinkClass *basesink_class = GST_BASE_SINK_CLASS (klass); + GstElementClass *element_class = GST_ELEMENT_CLASS (klass); + GObjectClass *gobject_class = G_OBJECT_CLASS (klass); gst_element_class_set_details_simple (element_class, "GdkPixbuf sink", "Sink/Video", "Output images as GdkPixbuf objects in bus messages", @@ -148,13 +149,6 @@ gst_gdk_pixbuf_sink_base_init (gpointer g_class) gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&pixbufsink_sink_factory)); -} - -static void -gst_gdk_pixbuf_sink_class_init (GstGdkPixbufSinkClass * klass) -{ - GstBaseSinkClass *basesink_class = GST_BASE_SINK_CLASS (klass); - GObjectClass *gobject_class = G_OBJECT_CLASS (klass); gobject_class->set_property = gst_gdk_pixbuf_sink_set_property; gobject_class->get_property = gst_gdk_pixbuf_sink_get_property; @@ -190,8 +184,7 @@ gst_gdk_pixbuf_sink_class_init (GstGdkPixbufSinkClass * klass) } static void -gst_gdk_pixbuf_sink_init (GstGdkPixbufSink * sink, - GstGdkPixbufSinkClass * klass) +gst_gdk_pixbuf_sink_init (GstGdkPixbufSink * sink) { sink->par_n = 0; sink->par_d = 0; @@ -240,46 +233,51 @@ static gboolean gst_gdk_pixbuf_sink_set_caps (GstBaseSink * basesink, GstCaps * caps) { GstGdkPixbufSink *sink = GST_GDK_PIXBUF_SINK (basesink); + GstVideoInfo info; GstVideoFormat fmt; - gint w, h, par_n, par_d; + gint w, h, s, par_n, par_d; GST_LOG_OBJECT (sink, "caps: %" GST_PTR_FORMAT, caps); - if (!gst_video_format_parse_caps (caps, &fmt, &w, &h)) { + if (!gst_video_info_from_caps (&info, caps)) { GST_WARNING_OBJECT (sink, "parse_caps failed"); return FALSE; } - if (!gst_video_parse_caps_pixel_aspect_ratio (caps, &par_n, &par_d)) { - GST_LOG_OBJECT (sink, "no pixel aspect ratio"); - return FALSE; - } + fmt = GST_VIDEO_INFO_FORMAT (&info); + w = GST_VIDEO_INFO_WIDTH (&info); + h = GST_VIDEO_INFO_HEIGHT (&info); + s = GST_VIDEO_INFO_COMP_PSTRIDE (&info, 0); + par_n = GST_VIDEO_INFO_PAR_N (&info); + par_d = GST_VIDEO_INFO_PAR_N (&info); - g_assert ((fmt == GST_VIDEO_FORMAT_RGB && - gst_video_format_get_pixel_stride (fmt, 0) == 3) || - (fmt == GST_VIDEO_FORMAT_RGBA && - gst_video_format_get_pixel_stride (fmt, 0) == 4)); + g_assert ((fmt == GST_VIDEO_FORMAT_RGB && s == 3) || + (fmt == GST_VIDEO_FORMAT_RGBA && s == 4)); GST_VIDEO_SINK_WIDTH (sink) = w; GST_VIDEO_SINK_HEIGHT (sink) = h; - sink->rowstride = gst_video_format_get_row_stride (fmt, 0, w); - sink->has_alpha = (fmt == GST_VIDEO_FORMAT_RGBA); - sink->par_n = par_n; sink->par_d = par_d; + sink->has_alpha = GST_VIDEO_INFO_HAS_ALPHA (&info); + GST_INFO_OBJECT (sink, "format : %d", fmt); GST_INFO_OBJECT (sink, "width x height : %d x %d", w, h); GST_INFO_OBJECT (sink, "pixel-aspect-ratio : %d/%d", par_n, par_d); + sink->info = info; + return TRUE; } static void -gst_gdk_pixbuf_sink_pixbuf_destroy_notify (guchar * pixels, GstBuffer * buf) +gst_gdk_pixbuf_sink_pixbuf_destroy_notify (guchar * pixels, + GstVideoFrame * frame) { - gst_buffer_unref (buf); + gst_video_frame_unmap (frame); + gst_buffer_unref (frame->buffer); + g_slice_free (GstVideoFrame, frame); } static GdkPixbuf * @@ -287,24 +285,30 @@ gst_gdk_pixbuf_sink_get_pixbuf_from_buffer (GstGdkPixbufSink * sink, GstBuffer * buf) { GdkPixbuf *pix = NULL; + GstVideoFrame *frame; gint minsize, bytes_per_pixel; g_return_val_if_fail (GST_VIDEO_SINK_WIDTH (sink) > 0, NULL); g_return_val_if_fail (GST_VIDEO_SINK_HEIGHT (sink) > 0, NULL); + frame = g_slice_new0 (GstVideoFrame); + gst_video_frame_map (frame, &sink->info, buf, GST_MAP_READ); + bytes_per_pixel = (sink->has_alpha) ? 4 : 3; /* last row needn't have row padding */ - minsize = (sink->rowstride * (GST_VIDEO_SINK_HEIGHT (sink) - 1)) + + minsize = (GST_VIDEO_FRAME_COMP_STRIDE (frame, 0) * + (GST_VIDEO_SINK_HEIGHT (sink) - 1)) + (bytes_per_pixel * GST_VIDEO_SINK_WIDTH (sink)); - g_return_val_if_fail (GST_BUFFER_SIZE (buf) >= minsize, NULL); + g_return_val_if_fail (gst_buffer_get_size (buf) >= minsize, NULL); - pix = gdk_pixbuf_new_from_data (GST_BUFFER_DATA (buf), + gst_buffer_ref (buf); + pix = gdk_pixbuf_new_from_data (GST_VIDEO_FRAME_COMP_DATA (frame, 0), GDK_COLORSPACE_RGB, sink->has_alpha, 8, GST_VIDEO_SINK_WIDTH (sink), - GST_VIDEO_SINK_HEIGHT (sink), sink->rowstride, + GST_VIDEO_SINK_HEIGHT (sink), GST_VIDEO_FRAME_COMP_STRIDE (frame, 0), (GdkPixbufDestroyNotify) gst_gdk_pixbuf_sink_pixbuf_destroy_notify, - gst_buffer_ref (buf)); + frame); return pix; } diff --git a/ext/gdk_pixbuf/gstgdkpixbufsink.h b/ext/gdk_pixbuf/gstgdkpixbufsink.h index b0eff5e5a8..f42436eaf9 100644 --- a/ext/gdk_pixbuf/gstgdkpixbufsink.h +++ b/ext/gdk_pixbuf/gstgdkpixbufsink.h @@ -21,7 +21,7 @@ #define GST_GDK_PIXBUF_SINK_H #include - +#include #include #include @@ -47,9 +47,9 @@ struct _GstGdkPixbufSink /*< private >*/ /* current caps */ + GstVideoInfo info; gint width; gint height; - gint rowstride; gint par_n; gint par_d; gboolean has_alpha; From 3720d1cb0955a946d7a94d417362861c6f431eb5 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Wed, 29 Feb 2012 22:30:56 +0100 Subject: [PATCH 30/55] gdkpixbufscale: remove deprecated property --- ext/gdk_pixbuf/gstgdkpixbuf.c | 44 ----------------------------------- 1 file changed, 44 deletions(-) diff --git a/ext/gdk_pixbuf/gstgdkpixbuf.c b/ext/gdk_pixbuf/gstgdkpixbuf.c index bac7df2032..70ae78d514 100644 --- a/ext/gdk_pixbuf/gstgdkpixbuf.c +++ b/ext/gdk_pixbuf/gstgdkpixbuf.c @@ -36,7 +36,6 @@ GST_DEBUG_CATEGORY_STATIC (gst_gdk_pixbuf_debug); enum { ARG_0, - ARG_SILENT /* FIXME 0.11: remove */ }; static GstStaticPadTemplate gst_gdk_pixbuf_sink_template = @@ -71,11 +70,6 @@ static GstStaticPadTemplate gst_gdk_pixbuf_src_template = GST_VIDEO_CAPS_MAKE ("RGBA")) ); -static void gst_gdk_pixbuf_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec); -static void gst_gdk_pixbuf_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec); - static GstStateChangeReturn gst_gdk_pixbuf_change_state (GstElement * element, GstStateChange transition); static GstFlowReturn gst_gdk_pixbuf_chain (GstPad * pad, GstObject * parent, @@ -187,20 +181,10 @@ gst_gdk_pixbuf_sink_query (GstPad * pad, GstObject * parent, GstQuery * query) static void gst_gdk_pixbuf_class_init (GstGdkPixbufClass * klass) { - GObjectClass *gobject_class; GstElementClass *gstelement_class; - gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; - gobject_class->set_property = gst_gdk_pixbuf_set_property; - gobject_class->get_property = gst_gdk_pixbuf_get_property; - - g_object_class_install_property (gobject_class, ARG_SILENT, - g_param_spec_boolean ("silent", "Silent", - "Produce verbose output ? (deprecated)", FALSE, - G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); - gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_gdk_pixbuf_change_state); @@ -497,34 +481,6 @@ error: } } -static void -gst_gdk_pixbuf_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec) -{ - switch (prop_id) { - case ARG_SILENT: - /* filter->silent = g_value_get_boolean (value); */ - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static void -gst_gdk_pixbuf_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec) -{ - switch (prop_id) { - case ARG_SILENT: - /* g_value_set_boolean (value, filter->silent); */ - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - static GstStateChangeReturn gst_gdk_pixbuf_change_state (GstElement * element, GstStateChange transition) { From 9f4fb6feb9635695d3345f6f03319004c7210fe4 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Wed, 29 Feb 2012 22:31:21 +0100 Subject: [PATCH 31/55] gdkpixbufsink: remove deprecated property --- ext/gdk_pixbuf/gstgdkpixbufsink.c | 9 --------- 1 file changed, 9 deletions(-) diff --git a/ext/gdk_pixbuf/gstgdkpixbufsink.c b/ext/gdk_pixbuf/gstgdkpixbufsink.c index 07af0de56a..509fc7e978 100644 --- a/ext/gdk_pixbuf/gstgdkpixbufsink.c +++ b/ext/gdk_pixbuf/gstgdkpixbufsink.c @@ -103,7 +103,6 @@ enum { PROP_0, - PROP_SEND_MESSAGES, PROP_POST_MESSAGES, PROP_LAST_PIXBUF, PROP_LAST @@ -153,12 +152,6 @@ gst_gdk_pixbuf_sink_class_init (GstGdkPixbufSinkClass * klass) gobject_class->set_property = gst_gdk_pixbuf_sink_set_property; gobject_class->get_property = gst_gdk_pixbuf_sink_get_property; - /* FIXME 0.11, remove in favour of post-messages */ - g_object_class_install_property (gobject_class, PROP_SEND_MESSAGES, - g_param_spec_boolean ("send-messages", "Send Messages", - "Whether to post messages containing pixbufs on the bus " - " (deprecated, use post-messages)", - DEFAULT_SEND_MESSAGES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstGdkPixbuf:post-messages: * @@ -391,7 +384,6 @@ gst_gdk_pixbuf_sink_set_property (GObject * object, guint prop_id, sink = GST_GDK_PIXBUF_SINK (object); switch (prop_id) { - case PROP_SEND_MESSAGES: case PROP_POST_MESSAGES: GST_OBJECT_LOCK (sink); sink->post_messages = g_value_get_boolean (value); @@ -412,7 +404,6 @@ gst_gdk_pixbuf_sink_get_property (GObject * object, guint prop_id, sink = GST_GDK_PIXBUF_SINK (object); switch (prop_id) { - case PROP_SEND_MESSAGES: case PROP_POST_MESSAGES: GST_OBJECT_LOCK (sink); g_value_set_boolean (value, sink->post_messages); From 3b846d7c7dc8d9f0732251e5ed3ed754d6d033e7 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Thu, 1 Mar 2012 11:36:34 +0100 Subject: [PATCH 32/55] audioparsers: disable non-ported wavpackparse --- gst/audioparsers/Makefile.am | 2 +- gst/audioparsers/plugin.c | 2 ++ 2 files changed, 3 insertions(+), 1 deletion(-) diff --git a/gst/audioparsers/Makefile.am b/gst/audioparsers/Makefile.am index 4d4d53ed21..5ad9c72f3b 100644 --- a/gst/audioparsers/Makefile.am +++ b/gst/audioparsers/Makefile.am @@ -3,7 +3,7 @@ plugin_LTLIBRARIES = libgstaudioparsers.la libgstaudioparsers_la_SOURCES = \ gstaacparse.c gstamrparse.c gstac3parse.c \ gstdcaparse.c gstflacparse.c gstmpegaudioparse.c \ - gstwavpackparse.c plugin.c + plugin.c libgstaudioparsers_la_CFLAGS = \ $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) diff --git a/gst/audioparsers/plugin.c b/gst/audioparsers/plugin.c index 16f98ff6c7..d18ad01ed0 100644 --- a/gst/audioparsers/plugin.c +++ b/gst/audioparsers/plugin.c @@ -46,8 +46,10 @@ plugin_init (GstPlugin * plugin) GST_RANK_PRIMARY + 1, GST_TYPE_FLAC_PARSE); ret &= gst_element_register (plugin, "mpegaudioparse", GST_RANK_PRIMARY + 2, GST_TYPE_MPEG_AUDIO_PARSE); +#if 0 ret &= gst_element_register (plugin, "wavpackparse2", GST_RANK_SECONDARY, GST_TYPE_WAVPACK_PARSE); +#endif return ret; } From 431ecdf6b790746ab915fb9a1c24ed041e79fb0c Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Thu, 1 Mar 2012 17:15:57 +0100 Subject: [PATCH 33/55] aasink: fix stride --- ext/aalib/gstaasink.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/ext/aalib/gstaasink.c b/ext/aalib/gstaasink.c index 62a885b6e3..5679c8940d 100644 --- a/ext/aalib/gstaasink.c +++ b/ext/aalib/gstaasink.c @@ -273,7 +273,7 @@ gst_aasink_init (GstAASink * aasink) static void gst_aasink_scale (GstAASink * aasink, guchar * src, guchar * dest, - gint sw, gint sh, gint dw, gint dh) + gint sw, gint sh, gint ss, gint dw, gint dh) { gint ypos, yinc, y; gint xpos, xinc, x; @@ -287,7 +287,7 @@ gst_aasink_scale (GstAASink * aasink, guchar * src, guchar * dest, for (y = dh; y; y--) { while (ypos > 0x10000) { ypos -= 0x10000; - src += sw; + src += ss; } xpos = 0x10000; { @@ -334,6 +334,7 @@ gst_aasink_render (GstBaseSink * basesink, GstBuffer * buffer) aa_image (aasink->context), /* dest */ GST_VIDEO_INFO_WIDTH (&aasink->info), /* sw */ GST_VIDEO_INFO_HEIGHT (&aasink->info), /* sh */ + GST_VIDEO_FRAME_PLANE_STRIDE (&frame, 0), /* ss */ aa_imgwidth (aasink->context), /* dw */ aa_imgheight (aasink->context)); /* dh */ From 3299f3917966089dc8e3104f3ed7b1fb566bf21f Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Sebastian=20Dr=C3=B6ge?= Date: Fri, 2 Mar 2012 10:13:08 +0100 Subject: [PATCH 34/55] mixer/colorbalance: Update for API changes --- ext/pulse/pulsemixerctrl.h | 18 +++++++++++------- gst/videofilter/gstvideobalance.c | 8 +++++++- sys/v4l2/gstv4l2colorbalance.h | 9 +++++++-- 3 files changed, 25 insertions(+), 10 deletions(-) diff --git a/ext/pulse/pulsemixerctrl.h b/ext/pulse/pulsemixerctrl.h index 077b8841aa..dbb0f99aa1 100644 --- a/ext/pulse/pulsemixerctrl.h +++ b/ext/pulse/pulsemixerctrl.h @@ -92,7 +92,7 @@ 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) \ +#define GST_IMPLEMENT_PULSEMIXER_CTRL_METHODS(Type, interface_as_function) \ static const GList* \ interface_as_function ## _list_tracks (GstMixer * mixer) \ { \ @@ -155,19 +155,23 @@ interface_as_function ## _get_mixer_flags (GstMixer * 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); \ + return gst_pulsemixer_ctrl_get_mixer_flags (this->mixer); \ } \ -static void \ -interface_as_function ## _mixer_interface_init (GstMixerInterface * iface) \ +static GstMixerType \ +interface_as_function ## _get_mixer_type (GstMixer * mixer) \ +{ \ + return GST_MIXER_HARDWARE; \ +} \ +static void \ +interface_as_function ## _mixer_interface_init (GstMixerInterface * iface) \ { \ - GST_MIXER_TYPE (iface) = GST_MIXER_HARDWARE; \ - \ iface->list_tracks = interface_as_function ## _list_tracks; \ iface->set_volume = interface_as_function ## _set_volume; \ iface->get_volume = interface_as_function ## _get_volume; \ iface->set_mute = interface_as_function ## _set_mute; \ iface->set_record = interface_as_function ## _set_record; \ - iface->get_mixer_flags = interface_as_function ## _get_mixer_flags; \ + iface->get_mixer_flags = interface_as_function ## _get_mixer_flags; \ + iface->get_mixer_type = interface_as_function ## _get_mixer_type; \ } G_END_DECLS diff --git a/gst/videofilter/gstvideobalance.c b/gst/videofilter/gstvideobalance.c index 2716c8fe30..9d76c70331 100644 --- a/gst/videofilter/gstvideobalance.c +++ b/gst/videofilter/gstvideobalance.c @@ -637,13 +637,19 @@ gst_video_balance_colorbalance_get_value (GstColorBalance * balance, return value; } +static GstColorBalanceType +gst_video_balance_colorbalance_get_balance_type (GstColorBalance * balance) +{ + return GST_COLOR_BALANCE_SOFTWARE; +} + static void gst_video_balance_colorbalance_init (GstColorBalanceInterface * iface) { - GST_COLOR_BALANCE_TYPE (iface) = GST_COLOR_BALANCE_SOFTWARE; iface->list_channels = gst_video_balance_colorbalance_list_channels; iface->set_value = gst_video_balance_colorbalance_set_value; iface->get_value = gst_video_balance_colorbalance_get_value; + iface->get_balance_type = gst_video_balance_colorbalance_get_balance_type; } static GstColorBalanceChannel * diff --git a/sys/v4l2/gstv4l2colorbalance.h b/sys/v4l2/gstv4l2colorbalance.h index 4e6a75e369..7bf47e1fb4 100644 --- a/sys/v4l2/gstv4l2colorbalance.h +++ b/sys/v4l2/gstv4l2colorbalance.h @@ -90,15 +90,20 @@ interface_as_function ## _color_balance_get_value (GstColorBalance * balance, return gst_v4l2_color_balance_get_value(this->v4l2object, channel); \ } \ \ +static GstColorBalanceType \ +interface_as_function ## _color_balance_get_balance_type (GstColorBalance * balance) \ +{ \ + return GST_COLOR_BALANCE_HARDWARE; \ +} \ + \ static void \ interface_as_function ## _color_balance_interface_init (GstColorBalanceInterface * iface) \ { \ - GST_COLOR_BALANCE_TYPE (iface) = GST_COLOR_BALANCE_HARDWARE; \ - \ /* default virtual functions */ \ iface->list_channels = interface_as_function ## _color_balance_list_channels; \ iface->set_value = interface_as_function ## _color_balance_set_value; \ iface->get_value = interface_as_function ## _color_balance_get_value; \ + iface->get_balance_type = interface_as_function ## _color_balance_get_balance_type; \ } \ #endif /* __GST_V4L2_COLOR_BALANCE_H__ */ From 78bb66902b9ea01e35aa12295ed2d90e5d1d17f3 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Sebastian=20Dr=C3=B6ge?= Date: Fri, 2 Mar 2012 11:17:33 +0100 Subject: [PATCH 35/55] gst: Update for the gstmarshal.[ch] removal --- gst/audiofx/audiofirfilter.c | 2 +- gst/audiofx/audioiirfilter.c | 2 +- gst/flv/gstindex.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/gst/audiofx/audiofirfilter.c b/gst/audiofx/audiofirfilter.c index 8b2f723018..57fa360e1e 100644 --- a/gst/audiofx/audiofirfilter.c +++ b/gst/audiofx/audiofirfilter.c @@ -134,7 +134,7 @@ gst_audio_fir_filter_class_init (GstAudioFIRFilterClass * klass) gst_audio_fir_filter_signals[SIGNAL_RATE_CHANGED] = g_signal_new ("rate-changed", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstAudioFIRFilterClass, rate_changed), - NULL, NULL, gst_marshal_VOID__INT, G_TYPE_NONE, 1, G_TYPE_INT); + NULL, NULL, NULL, G_TYPE_NONE, 1, G_TYPE_INT); gst_element_class_set_details_simple (gstelement_class, "Audio FIR filter", "Filter/Effect/Audio", diff --git a/gst/audiofx/audioiirfilter.c b/gst/audiofx/audioiirfilter.c index 86a428ad83..adcca1de21 100644 --- a/gst/audiofx/audioiirfilter.c +++ b/gst/audiofx/audioiirfilter.c @@ -132,7 +132,7 @@ gst_audio_iir_filter_class_init (GstAudioIIRFilterClass * klass) gst_audio_iir_filter_signals[SIGNAL_RATE_CHANGED] = g_signal_new ("rate-changed", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstAudioIIRFilterClass, rate_changed), - NULL, NULL, gst_marshal_VOID__INT, G_TYPE_NONE, 1, G_TYPE_INT); + NULL, NULL, NULL, G_TYPE_NONE, 1, G_TYPE_INT); gst_element_class_set_details_simple (gstelement_class, "Audio IIR filter", "Filter/Effect/Audio", diff --git a/gst/flv/gstindex.c b/gst/flv/gstindex.c index 792d4b6f09..aff8dc124b 100644 --- a/gst/flv/gstindex.c +++ b/gst/flv/gstindex.c @@ -172,7 +172,7 @@ gst_index_class_init (GstIndexClass * klass) gst_index_signals[ENTRY_ADDED] = g_signal_new ("entry-added", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstIndexClass, entry_added), NULL, NULL, - gst_marshal_VOID__BOXED, G_TYPE_NONE, 1, GST_TYPE_INDEX_ENTRY); + NULL, G_TYPE_NONE, 1, GST_TYPE_INDEX_ENTRY); gobject_class->set_property = gst_index_set_property; gobject_class->get_property = gst_index_get_property; From 4cd9255f0a8a9e15d81561f00f02d275b5095f70 Mon Sep 17 00:00:00 2001 From: "Oleksij Rempel (Alexey Fisher)" Date: Thu, 1 Mar 2012 14:15:29 +0100 Subject: [PATCH 36/55] v4l2src: fix v4l2_munmap() for compressed formats Make sure we always call munmap() with the same size we called mmap() with before. Current v4l2src uses the same structure for VIDIOC_QUERYBUF, VIDIOC_QBUF and v4l2_munmap calls. The problem is that the video buffer size (length) may vary for compressed or emulated bufs. VIDIOC_QBUF will change it if we pass the pointer of a v4l2_buffer. This is why we should avoid using same variable for mmap and video buffers. https://bugzilla.gnome.org/show_bug.cgi?id=671126 --- sys/v4l2/gstv4l2bufferpool.c | 5 +++-- sys/v4l2/gstv4l2bufferpool.h | 3 +++ 2 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sys/v4l2/gstv4l2bufferpool.c b/sys/v4l2/gstv4l2bufferpool.c index b81c6a4118..51419ff984 100644 --- a/sys/v4l2/gstv4l2bufferpool.c +++ b/sys/v4l2/gstv4l2bufferpool.c @@ -106,9 +106,9 @@ gst_v4l2_buffer_finalize (GstV4l2Buffer * buffer) if (!resuscitated) { GST_LOG_OBJECT (pool->v4l2elem, "buffer %p (data %p, len %u) not recovered, unmapping", - buffer, GST_BUFFER_DATA (buffer), buffer->vbuffer.length); + buffer, GST_BUFFER_DATA (buffer), buffer->mmap_length); gst_mini_object_unref (GST_MINI_OBJECT (pool)); - v4l2_munmap ((void *) GST_BUFFER_DATA (buffer), buffer->vbuffer.length); + v4l2_munmap ((void *) GST_BUFFER_DATA (buffer), buffer->mmap_length); GST_MINI_OBJECT_CLASS (v4l2buffer_parent_class)->finalize (GST_MINI_OBJECT (buffer)); @@ -183,6 +183,7 @@ gst_v4l2_buffer_new (GstV4l2BufferPool * pool, guint index, GstCaps * caps) GST_LOG_OBJECT (pool->v4l2elem, " length: %u", ret->vbuffer.length); GST_LOG_OBJECT (pool->v4l2elem, " input: %u", ret->vbuffer.input); + ret->mmap_length = ret->vbuffer.length; data = (guint8 *) v4l2_mmap (0, ret->vbuffer.length, PROT_READ | PROT_WRITE, MAP_SHARED, pool->video_fd, ret->vbuffer.m.offset); diff --git a/sys/v4l2/gstv4l2bufferpool.h b/sys/v4l2/gstv4l2bufferpool.h index caad9ac2fc..36ea3233a6 100644 --- a/sys/v4l2/gstv4l2bufferpool.h +++ b/sys/v4l2/gstv4l2bufferpool.h @@ -70,6 +70,9 @@ struct _GstV4l2Buffer { GstBuffer buffer; struct v4l2_buffer vbuffer; + /* warning: the size of mmap buffer and + * the actual frame-buffer can be different. */ + size_t mmap_length; /* FIXME: have GstV4l2Src* instead, as this has GstV4l2BufferPool* */ /* FIXME: do we really want to fix this if GstV4l2Buffer/Pool is shared From 6f707fcbca2c60aed565247567bafdd168980161 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Sat, 3 Mar 2012 13:04:48 +0000 Subject: [PATCH 37/55] v4l2: clear DISCONT flag when recycling buffers into the buffer pool The base class may have set the DISCONT flag on the first buffer pushed out. We need to clear that when recycling buffers back into the buffer pool, otherwise we constantly push out buffers with the discont flag set, which might upset downstream elements, esp. for compressed formats like mpeg-ts. --- sys/v4l2/gstv4l2bufferpool.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sys/v4l2/gstv4l2bufferpool.c b/sys/v4l2/gstv4l2bufferpool.c index 51419ff984..a0b4c842aa 100644 --- a/sys/v4l2/gstv4l2bufferpool.c +++ b/sys/v4l2/gstv4l2bufferpool.c @@ -98,6 +98,7 @@ gst_v4l2_buffer_finalize (GstV4l2Buffer * buffer) GST_LOG_OBJECT (pool->v4l2elem, "reviving buffer %p, %d", buffer, index); gst_buffer_ref (GST_BUFFER (buffer)); GST_BUFFER_SIZE (buffer) = 0; + GST_BUFFER_FLAG_UNSET (buffer, GST_BUFFER_FLAG_DISCONT); pool->buffers[index] = buffer; } From 073a03ef5c308c04fcbb814dd923f122ab2453e7 Mon Sep 17 00:00:00 2001 From: Antoine Tremblay Date: Fri, 2 Mar 2012 11:49:02 -0500 Subject: [PATCH 38/55] avimux: support up to 6 channels of AC-3 https://bugzilla.gnome.org/show_bug.cgi?id=671220 --- gst/avi/gstavimux.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/gst/avi/gstavimux.c b/gst/avi/gstavimux.c index 5e3521df79..17b1d1eef4 100644 --- a/gst/avi/gstavimux.c +++ b/gst/avi/gstavimux.c @@ -180,7 +180,7 @@ static GstStaticPadTemplate audio_sink_factory = "rate = (int) [ 1000, 96000 ], " "channels = (int) [ 1, 2 ]; " #endif*/ "audio/x-ac3, " - "rate = (int) [ 1000, 96000 ], " "channels = (int) [ 1, 2 ]; " + "rate = (int) [ 1000, 96000 ], " "channels = (int) [ 1, 6 ]; " "audio/x-alaw, " "rate = (int) [ 1000, 48000 ], " "channels = (int) [ 1, 2 ]; " "audio/x-mulaw, " From 18e8aa48e6d4ccc185874739d2eb9151fbcbf02b Mon Sep 17 00:00:00 2001 From: Stefan Sauer Date: Sun, 4 Mar 2012 19:55:26 +0100 Subject: [PATCH 39/55] jack: change the transport-mode enum into flags One can use (or not use) master and slave mode independently. --- ext/jack/gstjack.c | 6 ++---- ext/jack/gstjack.h | 6 +++--- ext/jack/gstjackaudioclient.c | 4 ++-- ext/jack/gstjackaudiosink.c | 12 ++++++------ ext/jack/gstjackaudiosink.h | 2 +- ext/jack/gstjackaudiosrc.c | 12 ++++++------ ext/jack/gstjackaudiosrc.h | 4 ++-- 7 files changed, 22 insertions(+), 24 deletions(-) diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c index 019bef9f8d..680a12d5d9 100644 --- a/ext/jack/gstjack.c +++ b/ext/jack/gstjack.c @@ -52,16 +52,14 @@ gst_jack_transport_get_type (void) static volatile gsize type = 0; if (g_once_init_enter (&type)) { - static const GEnumValue enum_values[] = { - {GST_JACK_TRANSPORT_AUTONOMOUS, - "No transport support", "autonomous"}, + static const GFlagsValue flag_values[] = { {GST_JACK_TRANSPORT_MASTER, "Start and stop transport with state changes", "master"}, {GST_JACK_TRANSPORT_SLAVE, "Follow transport state changes", "slave"}, {0, NULL, NULL}, }; - GType tmp = g_enum_register_static ("GstJackTransport", enum_values); + GType tmp = g_flags_register_static ("GstJackTransport", flag_values); g_once_init_leave (&type, tmp); } return (GType) type; diff --git a/ext/jack/gstjack.h b/ext/jack/gstjack.h index d00c0e7f61..ff0712f583 100644 --- a/ext/jack/gstjack.h +++ b/ext/jack/gstjack.h @@ -53,9 +53,9 @@ typedef enum { * client behaviour regarding to the transport mechanism. */ typedef enum { - GST_JACK_TRANSPORT_AUTONOMOUS, - GST_JACK_TRANSPORT_MASTER, - GST_JACK_TRANSPORT_SLAVE + GST_JACK_TRANSPORT_AUTONOMOUS = 0, + GST_JACK_TRANSPORT_MASTER = (0 << 1), + GST_JACK_TRANSPORT_SLAVE = (1 << 1), } GstJackTransport; typedef jack_default_audio_sample_t sample_t; diff --git a/ext/jack/gstjackaudioclient.c b/ext/jack/gstjackaudioclient.c index ef544d168e..8680cda845 100644 --- a/ext/jack/gstjackaudioclient.c +++ b/ext/jack/gstjackaudioclient.c @@ -88,10 +88,10 @@ static gboolean jack_handle_transport_change (GstJackAudioClient * client, GstState state) { GstObject *obj = GST_OBJECT_PARENT (client->user_data); - GstJackTransport mode; + guint mode; g_object_get (obj, "transport", &mode, NULL); - if ((mode == GST_JACK_TRANSPORT_SLAVE) && (GST_STATE (obj) != state)) { + if ((mode & GST_JACK_TRANSPORT_SLAVE) && (GST_STATE (obj) != state)) { GST_INFO_OBJECT (obj, "requesting state change: %s", gst_element_state_get_name (state)); gst_element_post_message (GST_ELEMENT (obj), diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c index 1a7fc28ee8..527a8643d1 100644 --- a/ext/jack/gstjackaudiosink.c +++ b/ext/jack/gstjackaudiosink.c @@ -560,7 +560,7 @@ gst_jack_ring_buffer_start (GstRingBuffer * buf) GST_DEBUG_OBJECT (sink, "start"); - if (sink->transport == GST_JACK_TRANSPORT_MASTER) { + if (sink->transport & GST_JACK_TRANSPORT_MASTER) { jack_client_t *client; client = gst_jack_audio_client_get_client (sink->client); @@ -579,7 +579,7 @@ gst_jack_ring_buffer_pause (GstRingBuffer * buf) GST_DEBUG_OBJECT (sink, "pause"); - if (sink->transport == GST_JACK_TRANSPORT_MASTER) { + if (sink->transport & GST_JACK_TRANSPORT_MASTER) { jack_client_t *client; client = gst_jack_audio_client_get_client (sink->client); @@ -598,7 +598,7 @@ gst_jack_ring_buffer_stop (GstRingBuffer * buf) GST_DEBUG_OBJECT (sink, "stop"); - if (sink->transport == GST_JACK_TRANSPORT_MASTER) { + if (sink->transport & GST_JACK_TRANSPORT_MASTER) { jack_client_t *client; client = gst_jack_audio_client_get_client (sink->client); @@ -767,7 +767,7 @@ gst_jack_audio_sink_class_init (GstJackAudioSinkClass * klass) * Since: 0.10.31 */ g_object_class_install_property (gobject_class, PROP_TRANSPORT, - g_param_spec_enum ("transport", "Transport mode", + g_param_spec_flags ("transport", "Transport mode", "Jack transport behaviour of the client", GST_TYPE_JACK_TRANSPORT, DEFAULT_PROP_TRANSPORT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); @@ -840,7 +840,7 @@ gst_jack_audio_sink_set_property (GObject * object, guint prop_id, } break; case PROP_TRANSPORT: - sink->transport = g_value_get_enum (value); + sink->transport = g_value_get_flags (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); @@ -870,7 +870,7 @@ gst_jack_audio_sink_get_property (GObject * object, guint prop_id, g_value_set_boxed (value, sink->jclient); break; case PROP_TRANSPORT: - g_value_set_enum (value, sink->transport); + g_value_set_flags (value, sink->transport); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); diff --git a/ext/jack/gstjackaudiosink.h b/ext/jack/gstjackaudiosink.h index 6665f10d5a..fbb660d899 100644 --- a/ext/jack/gstjackaudiosink.h +++ b/ext/jack/gstjackaudiosink.h @@ -59,7 +59,7 @@ struct _GstJackAudioSink { gchar *server; jack_client_t *jclient; gchar *client_name; - GstJackTransport transport; + guint transport; /* our client */ GstJackAudioClient *client; diff --git a/ext/jack/gstjackaudiosrc.c b/ext/jack/gstjackaudiosrc.c index e2b2bc00a1..20638751ee 100644 --- a/ext/jack/gstjackaudiosrc.c +++ b/ext/jack/gstjackaudiosrc.c @@ -572,7 +572,7 @@ gst_jack_ring_buffer_start (GstRingBuffer * buf) GST_DEBUG_OBJECT (src, "start"); - if (src->transport == GST_JACK_TRANSPORT_MASTER) { + if (src->transport & GST_JACK_TRANSPORT_MASTER) { jack_client_t *client; client = gst_jack_audio_client_get_client (src->client); @@ -591,7 +591,7 @@ gst_jack_ring_buffer_pause (GstRingBuffer * buf) GST_DEBUG_OBJECT (src, "pause"); - if (src->transport == GST_JACK_TRANSPORT_MASTER) { + if (src->transport & GST_JACK_TRANSPORT_MASTER) { jack_client_t *client; client = gst_jack_audio_client_get_client (src->client); @@ -610,7 +610,7 @@ gst_jack_ring_buffer_stop (GstRingBuffer * buf) GST_DEBUG_OBJECT (src, "stop"); - if (src->transport == GST_JACK_TRANSPORT_MASTER) { + if (src->transport & GST_JACK_TRANSPORT_MASTER) { jack_client_t *client; client = gst_jack_audio_client_get_client (src->client); @@ -787,7 +787,7 @@ gst_jack_audio_src_class_init (GstJackAudioSrcClass * klass) * Since: 0.10.31 */ g_object_class_install_property (gobject_class, PROP_TRANSPORT, - g_param_spec_enum ("transport", "Transport mode", + g_param_spec_flags ("transport", "Transport mode", "Jack transport behaviour of the client", GST_TYPE_JACK_TRANSPORT, DEFAULT_PROP_TRANSPORT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); @@ -857,7 +857,7 @@ gst_jack_audio_src_set_property (GObject * object, guint prop_id, } break; case PROP_TRANSPORT: - src->transport = g_value_get_enum (value); + src->transport = g_value_get_flags (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); @@ -885,7 +885,7 @@ gst_jack_audio_src_get_property (GObject * object, guint prop_id, g_value_set_boxed (value, src->jclient); break; case PROP_TRANSPORT: - g_value_set_enum (value, src->transport); + g_value_set_flags (value, src->transport); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); diff --git a/ext/jack/gstjackaudiosrc.h b/ext/jack/gstjackaudiosrc.h index ae19ba523c..c63f40c05f 100644 --- a/ext/jack/gstjackaudiosrc.h +++ b/ext/jack/gstjackaudiosrc.h @@ -72,11 +72,11 @@ struct _GstJackAudioSrc GstCaps *caps; /* properties */ - GstJackConnect connect; + GstJackConnect connect; gchar *server; jack_client_t *jclient; gchar *client_name; - GstJackTransport transport; + guint transport; /* our client */ GstJackAudioClient *client; From 9212d79198418bee661427bf8d24a2681abdb80c Mon Sep 17 00:00:00 2001 From: Stefan Sauer Date: Mon, 5 Mar 2012 10:44:54 +0100 Subject: [PATCH 40/55] jack: fix obvious wrong definition for the master flag --- 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 ff0712f583..50033d9fef 100644 --- a/ext/jack/gstjack.h +++ b/ext/jack/gstjack.h @@ -54,7 +54,7 @@ typedef enum { */ typedef enum { GST_JACK_TRANSPORT_AUTONOMOUS = 0, - GST_JACK_TRANSPORT_MASTER = (0 << 1), + GST_JACK_TRANSPORT_MASTER = (1 << 0), GST_JACK_TRANSPORT_SLAVE = (1 << 1), } GstJackTransport; From 7ca56e24bd60c64cd22300cd2bd75d6c63b6a08f Mon Sep 17 00:00:00 2001 From: Stefan Sauer Date: Mon, 5 Mar 2012 10:45:46 +0100 Subject: [PATCH 41/55] wavpackparse: remove empty lines in varable declarations caused by old indent --- ext/wavpack/gstwavpackparse.c | 46 +++-------------------------------- 1 file changed, 3 insertions(+), 43 deletions(-) diff --git a/ext/wavpack/gstwavpackparse.c b/ext/wavpack/gstwavpackparse.c index e1ff785b91..e3625c52cc 100644 --- a/ext/wavpack/gstwavpackparse.c +++ b/ext/wavpack/gstwavpackparse.c @@ -121,8 +121,7 @@ gst_wavpack_parse_base_init (gpointer klass) GstElementClass *element_class = GST_ELEMENT_CLASS (klass); gst_element_class_add_static_pad_template (element_class, &src_factory); - gst_element_class_add_static_pad_template (element_class, - &wvc_src_factory); + gst_element_class_add_static_pad_template (element_class, &wvc_src_factory); gst_element_class_add_static_pad_template (element_class, &sink_factory); gst_element_class_set_details_simple (element_class, "Wavpack parser", @@ -144,7 +143,6 @@ static void gst_wavpack_parse_class_init (GstWavpackParseClass * klass) { GObjectClass *gobject_class; - GstElementClass *gstelement_class; gobject_class = (GObjectClass *) klass; @@ -168,7 +166,6 @@ gst_wavpack_parse_index_get_entry_from_sample (GstWavpackParse * wvparse, gint64 sample_offset) { gint i; - GSList *node; if (wvparse->entries == NULL) @@ -289,9 +286,7 @@ static gboolean gst_wavpack_parse_src_query (GstPad * pad, GstQuery * query) { GstWavpackParse *parse = GST_WAVPACK_PARSE (gst_pad_get_parent (pad)); - GstFormat format; - gboolean ret = FALSE; switch (GST_QUERY_TYPE (query)) { @@ -411,9 +406,7 @@ gst_wavpack_parse_scan_to_find_sample (GstWavpackParse * parse, gint64 sample, gint64 * byte_offset, gint64 * start_sample) { GstWavpackParseIndexEntry *entry; - GstFlowReturn ret; - gint64 off = 0; /* first, check if we have to scan at all */ @@ -441,7 +434,6 @@ gst_wavpack_parse_scan_to_find_sample (GstWavpackParse * parse, /* now scan forward until we find the chunk we're looking for or hit EOS */ do { WavpackHeader header; - GstBuffer *buf; buf = gst_wavpack_parse_pull_buffer (parse, off, sizeof (WavpackHeader), @@ -481,13 +473,9 @@ gst_wavpack_parse_send_newsegment (GstWavpackParse * wvparse, gboolean update) GstSegment *s = &wvparse->segment; gboolean ret; - gint64 stop_time = -1; - gint64 start_time = 0; - gint64 cur_pos_time; - gint64 diff; /* segment is in DEFAULT format, but we want to send a TIME newsegment */ @@ -524,31 +512,18 @@ gst_wavpack_parse_handle_seek_event (GstWavpackParse * wvparse, GstEvent * event) { GstSeekFlags seek_flags; - GstSeekType start_type; - GstSeekType stop_type; - GstSegment segment; - GstFormat format; - gboolean only_update; - gboolean flush, ret; - gdouble speed; - gint64 stop; - gint64 start; /* sample we want to seek to */ - gint64 byte_offset; /* byte offset the chunk we seek to starts at */ - gint64 chunk_start; /* first sample in chunk we seek to */ - guint rate; - gint64 last_stop; if (wvparse->adapter) { @@ -674,7 +649,6 @@ static gboolean gst_wavpack_parse_sink_event (GstPad * pad, GstEvent * event) { GstWavpackParse *parse; - gboolean ret = TRUE; parse = GST_WAVPACK_PARSE (gst_pad_get_parent (pad)); @@ -733,7 +707,6 @@ static gboolean gst_wavpack_parse_src_event (GstPad * pad, GstEvent * event) { GstWavpackParse *parse; - gboolean ret; parse = GST_WAVPACK_PARSE (gst_pad_get_parent (pad)); @@ -755,7 +728,6 @@ static void gst_wavpack_parse_init (GstWavpackParse * parse, GstWavpackParseClass * gclass) { GstElementClass *klass = GST_ELEMENT_GET_CLASS (parse); - GstPadTemplate *tmpl; tmpl = gst_element_class_get_pad_template (klass, "sink"); @@ -780,7 +752,6 @@ static gint64 gst_wavpack_parse_get_upstream_length (GstWavpackParse * parse) { gint64 length = -1; - GstFormat format = GST_FORMAT_BYTES; if (!gst_pad_query_peer_duration (parse->sinkpad, &format, &length)) { @@ -796,7 +767,6 @@ gst_wavpack_parse_pull_buffer (GstWavpackParse * wvparse, gint64 offset, guint size, GstFlowReturn * flow) { GstFlowReturn flow_ret; - GstBuffer *buf = NULL; if (offset + size > wvparse->upstream_length) { @@ -837,9 +807,7 @@ gst_wavpack_parse_create_src_pad (GstWavpackParse * wvparse, GstBuffer * buf, WavpackHeader * header) { GstWavpackMetadata meta; - GstCaps *caps = NULL; - guchar *bufptr; g_assert (wvparse->srcpad == NULL); @@ -954,8 +922,8 @@ gst_wavpack_parse_push_buffer (GstWavpackParse * wvparse, GstBuffer * buf, WavpackHeader * header) { GstFlowReturn ret; - wvparse->current_offset += header->ckSize + 8; + wvparse->current_offset += header->ckSize + 8; wvparse->segment.last_stop = header->block_index; if (wvparse->need_newsegment) { @@ -1021,8 +989,7 @@ gst_wavpack_parse_push_buffer (GstWavpackParse * wvparse, GstBuffer * buf, static guint8 * gst_wavpack_parse_find_marker (guint8 * buf, guint size) { - int i; - + gint i; guint8 *ret = NULL; if (G_UNLIKELY (size < 4)) @@ -1041,7 +1008,6 @@ static GstFlowReturn gst_wavpack_parse_resync_loop (GstWavpackParse * parse, WavpackHeader * header) { GstFlowReturn flow_ret = GST_FLOW_UNEXPECTED; - GstBuffer *buf = NULL; /* loop until we have a frame header or reach the end of the stream */ @@ -1108,7 +1074,6 @@ static void gst_wavpack_parse_loop (GstElement * element) { GstWavpackParse *parse = GST_WAVPACK_PARSE (element); - GstFlowReturn flow_ret; WavpackHeader header = { {0,}, 0, }; GstBuffer *buf = NULL; @@ -1183,7 +1148,6 @@ static gboolean gst_wavpack_parse_resync_adapter (GstAdapter * adapter) { const guint8 *buf, *marker; - guint avail = gst_adapter_available (adapter); if (avail < 4) @@ -1215,11 +1179,8 @@ static GstFlowReturn gst_wavpack_parse_chain (GstPad * pad, GstBuffer * buf) { GstWavpackParse *wvparse = GST_WAVPACK_PARSE (GST_PAD_PARENT (pad)); - GstFlowReturn ret = GST_FLOW_OK; - WavpackHeader wph; - const guint8 *tmp_buf; if (!wvparse->adapter) { @@ -1279,7 +1240,6 @@ static GstStateChangeReturn gst_wavpack_parse_change_state (GstElement * element, GstStateChange transition) { GstWavpackParse *wvparse = GST_WAVPACK_PARSE (element); - GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS; switch (transition) { From a4ed5daae64af8c2c13d6d5ad9e088ca795095d9 Mon Sep 17 00:00:00 2001 From: Stefan Sauer Date: Mon, 5 Mar 2012 10:51:33 +0100 Subject: [PATCH 42/55] wavpackparse: initialize header to silence older gcc versions --- gst/audioparsers/gstwavpackparse.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/gst/audioparsers/gstwavpackparse.c b/gst/audioparsers/gstwavpackparse.c index 61b0eb46b8..41e73a66f6 100644 --- a/gst/audioparsers/gstwavpackparse.c +++ b/gst/audioparsers/gstwavpackparse.c @@ -399,7 +399,7 @@ gst_wavpack_parse_frame_header (GstWavpackParse * parse, GstBuffer * buf, gint skip, WavpackHeader * _wph) { GstByteReader br = GST_BYTE_READER_INIT_FROM_BUFFER (buf); - WavpackHeader wph; + WavpackHeader wph = { {0,}, 0, }; g_return_val_if_fail (GST_BUFFER_SIZE (buf) >= skip + sizeof (WavpackHeader), FALSE); From 7dbb1551aef8dcd2da0d51e03706f8ebe81c3269 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Mon, 5 Mar 2012 10:46:51 +0100 Subject: [PATCH 43/55] wavpackenc: restore legacy buffer offset decorating somewhat ... at least sufficiently to aid in recognizing rewritten header buffer making unit test pass. --- ext/wavpack/gstwavpackenc.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/ext/wavpack/gstwavpackenc.c b/ext/wavpack/gstwavpackenc.c index 8595798f47..5094769af9 100644 --- a/ext/wavpack/gstwavpackenc.c +++ b/ext/wavpack/gstwavpackenc.c @@ -616,6 +616,13 @@ gst_wavpack_enc_push_block (void *id, void *data, int32_t count) } } samples = wph.block_samples; + + /* decorate buffer */ + /* NOTE: this will get overwritten by baseclass, but stay for those + * that are pushed directly + * FIXME: add setting to baseclass to avoid overwriting it ?? */ + GST_BUFFER_OFFSET (buffer) = wph.block_index; + GST_BUFFER_OFFSET_END (buffer) = wph.block_index + wph.block_samples; } else { /* if it's something else set no timestamp and duration on the buffer */ GST_DEBUG_OBJECT (enc, "got %d bytes of unknown data", count); From e43ff87364ad07517f193615a1457efc78802871 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Mon, 5 Mar 2012 10:47:44 +0100 Subject: [PATCH 44/55] wavpackdec: fix copying output data --- ext/wavpack/gstwavpackdec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/ext/wavpack/gstwavpackdec.c b/ext/wavpack/gstwavpackdec.c index 2be81494f8..e336e3287a 100644 --- a/ext/wavpack/gstwavpackdec.c +++ b/ext/wavpack/gstwavpackdec.c @@ -394,14 +394,14 @@ gst_wavpack_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf) if (width == 8) { gint8 *outbuffer = (gint8 *) out_data; - for (i = 0; i < max; i--) { + for (i = 0; i < max; i++) { *outbuffer++ = (gint8) (dec_data[i]); } } else if (width == 16) { gint16 *outbuffer = (gint16 *) out_data; for (i = 0; i < max; i++) { - *outbuffer++ = (gint8) (dec_data[i]); + *outbuffer++ = (gint16) (dec_data[i]); } } else if (dec->width == 32) { gint32 *outbuffer = (gint32 *) out_data; From dccbbe191dd7e3d5a845ba2ef99ce096bf99c2c5 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Mon, 5 Mar 2012 10:49:33 +0100 Subject: [PATCH 45/55] wavpackdec: allow some timestamp tolerance to arrange for perfect timestamping ... which also happens to make some more unit tests pass. --- ext/wavpack/gstwavpackdec.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/ext/wavpack/gstwavpackdec.c b/ext/wavpack/gstwavpackdec.c index e336e3287a..05caa0812c 100644 --- a/ext/wavpack/gstwavpackdec.c +++ b/ext/wavpack/gstwavpackdec.c @@ -169,6 +169,8 @@ gst_wavpack_dec_start (GstAudioDecoder * dec) gst_audio_decoder_set_max_errors (dec, 16); /* don't bother us with flushing */ gst_audio_decoder_set_drainable (dec, FALSE); + /* aim for some perfect timestamping */ + gst_audio_decoder_set_tolerance (dec, 10 * GST_MSECOND); return TRUE; } From 1dada15ccf45ffecb77cdbbac52178731958d144 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Mon, 5 Mar 2012 12:03:01 +0100 Subject: [PATCH 46/55] ximage: use new style caps --- sys/ximage/Makefile.am | 2 +- sys/ximage/gstximagesrc.c | 21 ++++++++++++--------- 2 files changed, 13 insertions(+), 10 deletions(-) diff --git a/sys/ximage/Makefile.am b/sys/ximage/Makefile.am index db3ab228df..553226e30a 100644 --- a/sys/ximage/Makefile.am +++ b/sys/ximage/Makefile.am @@ -7,7 +7,7 @@ libgstximagesrc_la_CFLAGS = \ $(GST_CFLAGS) \ $(X_CFLAGS) $(XFIXES_CFLAGS) $(XDAMAGE_CFLAGS) libgstximagesrc_la_LIBADD = \ - $(GST_PLUGINS_BASE_LIBS) \ + $(GST_PLUGINS_BASE_LIBS) -lgstvideo-$(GST_MAJORMINOR) \ $(GST_BASE_LIBS) \ $(X_LIBS) $(XSHM_LIBS) $(XFIXES_LIBS) $(XDAMAGE_LIBS) libgstximagesrc_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) diff --git a/sys/ximage/gstximagesrc.c b/sys/ximage/gstximagesrc.c index ef24e7158d..51b46d7c01 100644 --- a/sys/ximage/gstximagesrc.c +++ b/sys/ximage/gstximagesrc.c @@ -30,7 +30,7 @@ * * Example pipelines * |[ - * gst-launch ximagesrc ! video/x-raw-rgb,framerate=5/1 ! ffmpegcolorspace ! theoraenc ! oggmux ! filesink location=desktop.ogg + * gst-launch ximagesrc ! video/x-raw,framerate=5/1 ! ffmpegcolorspace ! theoraenc ! oggmux ! filesink location=desktop.ogg * ]| Encodes your X display to an Ogg theora video at 5 frames per second. * */ @@ -48,6 +48,7 @@ #include #include +#include #include "gst/glib-compat-private.h" @@ -56,7 +57,7 @@ GST_DEBUG_CATEGORY_STATIC (gst_debug_ximage_src); static GstStaticPadTemplate t = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, - GST_STATIC_CAPS ("video/x-raw-rgb, " + GST_STATIC_CAPS ("video/x-raw, " "framerate = (fraction) [ 0, MAX ], " "width = (int) [ 1, MAX ], " "height = (int) [ 1, MAX ], " "pixel-aspect-ratio = (fraction) [ 0, MAX ]")); @@ -1018,6 +1019,7 @@ gst_ximage_src_get_caps (GstBaseSrc * bs, GstCaps * filter) GstXImageSrc *s = GST_XIMAGE_SRC (bs); GstXContext *xcontext; gint width, height; + GstVideoFormat format; if ((!s->xcontext) && (!gst_ximage_src_open_display (s, s->display_name))) return @@ -1072,13 +1074,14 @@ gst_ximage_src_get_caps (GstBaseSrc * bs, GstCaps * filter) s->endy = height - 1; } GST_DEBUG ("width = %d, height=%d", width, height); - return gst_caps_new_simple ("video/x-raw-rgb", - "bpp", G_TYPE_INT, xcontext->bpp, - "depth", G_TYPE_INT, xcontext->depth, - "endianness", G_TYPE_INT, xcontext->endianness, - "red_mask", G_TYPE_INT, xcontext->r_mask_output, - "green_mask", G_TYPE_INT, xcontext->g_mask_output, - "blue_mask", G_TYPE_INT, xcontext->b_mask_output, + + format = + gst_video_format_from_masks (xcontext->depth, xcontext->bpp, + xcontext->endianness, xcontext->r_mask_output, xcontext->g_mask_output, + xcontext->b_mask_output, 0); + + return gst_caps_new_simple ("video/x-raw", + "format", G_TYPE_STRING, gst_video_format_to_string (format), "width", G_TYPE_INT, width, "height", G_TYPE_INT, height, "framerate", GST_TYPE_FRACTION_RANGE, 1, G_MAXINT, G_MAXINT, 1, From a83d2f70473d161dbb6c9ceff21dc214ec9db3a8 Mon Sep 17 00:00:00 2001 From: "Oleksij Rempel (Alexey Fisher)" Date: Mon, 5 Mar 2012 11:17:30 +0100 Subject: [PATCH 47/55] v4l2sink: don't use deprecated XKeycodeToKeysym https://bugzilla.gnome.org/show_bug.cgi?id=671299 Signed-off-by: Oleksij Rempel (Alexey Fisher) --- sys/v4l2/gstv4l2xoverlay.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sys/v4l2/gstv4l2xoverlay.c b/sys/v4l2/gstv4l2xoverlay.c index 60d0919067..f1c4d5196d 100644 --- a/sys/v4l2/gstv4l2xoverlay.c +++ b/sys/v4l2/gstv4l2xoverlay.c @@ -32,6 +32,8 @@ #include #include #include +/* for XkbKeycodeToKeysym */ +#include #include @@ -323,7 +325,7 @@ event_refresh (gpointer data) case KeyPress: case KeyRelease: g_mutex_lock (v4l2xv->mutex); - keysym = XKeycodeToKeysym (v4l2xv->dpy, e.xkey.keycode, 0); + keysym = XkbKeycodeToKeysym (v4l2xv->dpy, e.xkey.keycode, 0, 0); if (keysym != NoSymbol) { key_str = XKeysymToString (keysym); } else { From 2adc948d186a1162b111519ae3cee741c9602629 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Mon, 5 Mar 2012 12:43:17 +0100 Subject: [PATCH 48/55] x-raw-bayer -> x-bayer --- sys/v4l2/gstv4l2object.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sys/v4l2/gstv4l2object.c b/sys/v4l2/gstv4l2object.c index 43a03df735..6bcbf16b89 100644 --- a/sys/v4l2/gstv4l2object.c +++ b/sys/v4l2/gstv4l2object.c @@ -1311,7 +1311,7 @@ gst_v4l2_object_v4l2fourcc_to_structure (guint32 fourcc) break; #ifdef V4L2_PIX_FMT_SBGGR8 case V4L2_PIX_FMT_SBGGR8: - structure = gst_structure_new_empty ("video/x-raw-bayer"); + structure = gst_structure_new_empty ("video/x-bayer"); break; #endif #ifdef V4L2_PIX_FMT_SN9C10X @@ -1472,7 +1472,7 @@ gst_v4l2_object_get_caps_info (GstV4l2Object * v4l2object, GstCaps * caps, } else if (g_str_equal (mimetype, "image/jpeg")) { fourcc = V4L2_PIX_FMT_JPEG; #ifdef V4L2_PIX_FMT_SBGGR8 - } else if (g_str_equal (mimetype, "video/x-raw-bayer")) { + } else if (g_str_equal (mimetype, "video/x-bayer")) { fourcc = V4L2_PIX_FMT_SBGGR8; #endif #ifdef V4L2_PIX_FMT_SN9C10X From 690884bc57c661aefea22b8949a86675c8e3bae5 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Mon, 5 Mar 2012 12:15:44 +0100 Subject: [PATCH 49/55] audioparsers: port wavpackparse to 0.11 --- gst/audioparsers/Makefile.am | 2 +- gst/audioparsers/gstwavpackparse.c | 214 ++++++++++++++--------------- gst/audioparsers/gstwavpackparse.h | 3 - gst/audioparsers/plugin.c | 6 +- 4 files changed, 106 insertions(+), 119 deletions(-) diff --git a/gst/audioparsers/Makefile.am b/gst/audioparsers/Makefile.am index 5ad9c72f3b..4d4d53ed21 100644 --- a/gst/audioparsers/Makefile.am +++ b/gst/audioparsers/Makefile.am @@ -3,7 +3,7 @@ plugin_LTLIBRARIES = libgstaudioparsers.la libgstaudioparsers_la_SOURCES = \ gstaacparse.c gstamrparse.c gstac3parse.c \ gstdcaparse.c gstflacparse.c gstmpegaudioparse.c \ - plugin.c + gstwavpackparse.c plugin.c libgstaudioparsers_la_CFLAGS = \ $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) diff --git a/gst/audioparsers/gstwavpackparse.c b/gst/audioparsers/gstwavpackparse.c index 41e73a66f6..637198d74d 100644 --- a/gst/audioparsers/gstwavpackparse.c +++ b/gst/audioparsers/gstwavpackparse.c @@ -42,7 +42,7 @@ #include "gstwavpackparse.h" #include -#include +#include GST_DEBUG_CATEGORY_STATIC (wavpack_parse_debug); #define GST_CAT_DEFAULT wavpack_parse_debug @@ -51,7 +51,7 @@ static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-wavpack, " - "width = (int) [ 1, 32 ], " + "depth = (int) [ 1, 32 ], " "channels = (int) [ 1, 8 ], " "rate = (int) [ 6000, 192000 ], " "framed = (boolean) TRUE; " "audio/x-wavpack-correction, " "framed = (boolean) TRUE") @@ -66,36 +66,19 @@ static void gst_wavpack_parse_finalize (GObject * object); static gboolean gst_wavpack_parse_start (GstBaseParse * parse); static gboolean gst_wavpack_parse_stop (GstBaseParse * parse); -static gboolean gst_wavpack_parse_check_valid_frame (GstBaseParse * parse, - GstBaseParseFrame * frame, guint * size, gint * skipsize); -static GstFlowReturn gst_wavpack_parse_parse_frame (GstBaseParse * parse, - GstBaseParseFrame * frame); -static GstCaps *gst_wavpack_parse_get_sink_caps (GstBaseParse * parse); +static GstFlowReturn gst_wavpack_parse_handle_frame (GstBaseParse * parse, + GstBaseParseFrame * frame, gint * skipsize); +static GstCaps *gst_wavpack_parse_get_sink_caps (GstBaseParse * parse, + GstCaps * filter); -/* FIXME remove when all properly renamed */ -typedef GstWavpackParse GstWavpackParse2; -typedef GstWavpackParseClass GstWavpackParse2Class; - -GST_BOILERPLATE (GstWavpackParse2, gst_wavpack_parse, GstBaseParse, - GST_TYPE_BASE_PARSE); - -static void -gst_wavpack_parse_base_init (gpointer klass) -{ - GstElementClass *element_class = GST_ELEMENT_CLASS (klass); - - gst_element_class_add_static_pad_template (element_class, &sink_template); - gst_element_class_add_static_pad_template (element_class, &src_template); - - gst_element_class_set_details_simple (element_class, - "Wavpack audio stream parser", "Codec/Parser/Audio", - "Wavpack parser", "Mark Nauwelaerts "); -} +#define gst_wavpack_parse_parent_class parent_class +G_DEFINE_TYPE (GstWavpackParse, gst_wavpack_parse, GST_TYPE_BASE_PARSE); static void gst_wavpack_parse_class_init (GstWavpackParseClass * klass) { GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass); + GstElementClass *element_class = GST_ELEMENT_CLASS (klass); GObjectClass *object_class = G_OBJECT_CLASS (klass); GST_DEBUG_CATEGORY_INIT (wavpack_parse_debug, "wavpackparse", 0, @@ -105,11 +88,19 @@ gst_wavpack_parse_class_init (GstWavpackParseClass * klass) parse_class->start = GST_DEBUG_FUNCPTR (gst_wavpack_parse_start); parse_class->stop = GST_DEBUG_FUNCPTR (gst_wavpack_parse_stop); - parse_class->check_valid_frame = - GST_DEBUG_FUNCPTR (gst_wavpack_parse_check_valid_frame); - parse_class->parse_frame = GST_DEBUG_FUNCPTR (gst_wavpack_parse_parse_frame); + parse_class->handle_frame = + GST_DEBUG_FUNCPTR (gst_wavpack_parse_handle_frame); parse_class->get_sink_caps = GST_DEBUG_FUNCPTR (gst_wavpack_parse_get_sink_caps); + + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&sink_template)); + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&src_template)); + + gst_element_class_set_details_simple (element_class, + "Wavpack audio stream parser", "Codec/Parser/Audio", + "Wavpack parser", "Mark Nauwelaerts "); } static void @@ -123,7 +114,7 @@ gst_wavpack_parse_reset (GstWavpackParse * wvparse) } static void -gst_wavpack_parse_init (GstWavpackParse * wvparse, GstWavpackParseClass * klass) +gst_wavpack_parse_init (GstWavpackParse * wvparse) { gst_wavpack_parse_reset (wvparse); } @@ -209,7 +200,7 @@ static const struct 0x00001, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT}, { 0x00002, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}, { 0x00004, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER}, { - 0x00008, GST_AUDIO_CHANNEL_POSITION_LFE}, { + 0x00008, GST_AUDIO_CHANNEL_POSITION_LFE1}, { 0x00010, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT}, { 0x00020, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}, { 0x00040, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER}, { @@ -217,37 +208,25 @@ static const struct 0x00100, GST_AUDIO_CHANNEL_POSITION_REAR_CENTER}, { 0x00200, GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT}, { 0x00400, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT}, { - 0x00800, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_CENTER */ - { - 0x01000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_FRONT_LEFT */ - { - 0x02000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_FRONT_CENTER */ - { - 0x04000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_FRONT_RIGHT */ - { - 0x08000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_BACK_LEFT */ - { - 0x10000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_BACK_CENTER */ - { - 0x20000, GST_AUDIO_CHANNEL_POSITION_INVALID} /* TOP_BACK_RIGHT */ + 0x00800, GST_AUDIO_CHANNEL_POSITION_TOP_CENTER}, { + 0x01000, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_LEFT}, { + 0x02000, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_CENTER}, { + 0x04000, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_RIGHT}, { + 0x08000, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_LEFT}, { + 0x10000, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_CENTER}, { + 0x20000, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_RIGHT} }; #define MAX_CHANNEL_POSITIONS G_N_ELEMENTS (layout_mapping) static gboolean -gst_wavpack_set_channel_layout (GstCaps * caps, gint layout) +gst_wavpack_get_channel_positions (gint num_channels, gint layout, + GstAudioChannelPosition * pos) { - GstAudioChannelPosition pos[MAX_CHANNEL_POSITIONS]; - GstStructure *s; - gint num_channels, i, p; - - s = gst_caps_get_structure (caps, 0); - if (!gst_structure_get_int (s, "channels", &num_channels)) - g_return_val_if_reached (FALSE); + gint i, p; if (num_channels == 1 && layout == 0x00004) { - pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_MONO; - gst_audio_set_channel_positions (s, pos); + pos[0] = GST_AUDIO_CHANNEL_POSITION_MONO; return TRUE; } @@ -276,7 +255,6 @@ gst_wavpack_set_channel_layout (GstCaps * caps, gint layout) return FALSE; } - gst_audio_set_channel_positions (s, pos); return TRUE; } @@ -297,12 +275,15 @@ gst_wavpack_parse_frame_metadata (GstWavpackParse * parse, GstBuffer * buf, { GstByteReader br; gint i; + GstMapInfo map; g_return_val_if_fail (wph != NULL || wpi != NULL, FALSE); - g_return_val_if_fail (GST_BUFFER_SIZE (buf) >= skip + sizeof (WavpackHeader), - FALSE); + g_return_val_if_fail (gst_buffer_get_size (buf) >= + skip + sizeof (WavpackHeader), FALSE); - gst_byte_reader_init (&br, GST_BUFFER_DATA (buf) + skip, wph->ckSize + 8); + gst_buffer_map (buf, &map, GST_MAP_READ); + + gst_byte_reader_init (&br, map.data + skip, wph->ckSize + 8); /* skip past header */ gst_byte_reader_skip_unchecked (&br, sizeof (WavpackHeader)); @@ -382,11 +363,14 @@ gst_wavpack_parse_frame_metadata (GstWavpackParse * parse, GstBuffer * buf, } } + gst_buffer_unmap (buf, &map); + return TRUE; /* ERRORS */ read_failed: { + gst_buffer_unmap (buf, &map); GST_DEBUG_OBJECT (parse, "short read while parsing metadata"); /* let's look the other way anyway */ return TRUE; @@ -398,11 +382,15 @@ static gboolean gst_wavpack_parse_frame_header (GstWavpackParse * parse, GstBuffer * buf, gint skip, WavpackHeader * _wph) { - GstByteReader br = GST_BYTE_READER_INIT_FROM_BUFFER (buf); - WavpackHeader wph = { {0,}, 0, }; + GstByteReader br; + WavpackHeader wph; + GstMapInfo map; - g_return_val_if_fail (GST_BUFFER_SIZE (buf) >= skip + sizeof (WavpackHeader), - FALSE); + g_return_val_if_fail (gst_buffer_get_size (buf) >= + skip + sizeof (WavpackHeader), FALSE); + + gst_buffer_map (buf, &map, GST_MAP_READ); + gst_byte_reader_init (&br, map.data, map.size); /* marker */ gst_byte_reader_skip_unchecked (&br, skip + 4); @@ -436,34 +424,41 @@ gst_wavpack_parse_frame_header (GstWavpackParse * parse, GstBuffer * buf, if (_wph) *_wph = wph; + gst_buffer_unmap (buf, &map); + return TRUE; } -static gboolean -gst_wavpack_parse_check_valid_frame (GstBaseParse * parse, - GstBaseParseFrame * frame, guint * framesize, gint * skipsize) +static GstFlowReturn +gst_wavpack_parse_handle_frame (GstBaseParse * parse, + GstBaseParseFrame * frame, gint * skipsize) { GstWavpackParse *wvparse = GST_WAVPACK_PARSE (parse); GstBuffer *buf = frame->buffer; - GstByteReader reader = GST_BYTE_READER_INIT_FROM_BUFFER (buf); + GstByteReader reader; gint off; + guint rate, chans, width, mask; gboolean lost_sync, draining, final; guint frmsize = 0; WavpackHeader wph; WavpackInfo wpi = { 0, }; + GstMapInfo map; - if (G_UNLIKELY (GST_BUFFER_SIZE (buf) < sizeof (WavpackHeader))) + if (G_UNLIKELY (gst_buffer_get_size (buf) < sizeof (WavpackHeader))) return FALSE; + gst_buffer_map (buf, &map, GST_MAP_READ); + gst_byte_reader_init (&reader, map.data, map.size); + /* scan for 'wvpk' marker */ off = gst_byte_reader_masked_scan_uint32 (&reader, 0xffffffff, 0x7776706b, - 0, GST_BUFFER_SIZE (buf)); + 0, map.size); GST_LOG_OBJECT (parse, "possible sync at buffer offset %d", off); /* didn't find anything that looks like a sync word, skip */ if (off < 0) { - *skipsize = GST_BUFFER_SIZE (buf) - 3; + *skipsize = map.size - 3; goto skip; } @@ -530,71 +525,50 @@ gst_wavpack_parse_check_valid_frame (GstBaseParse * parse, lost_sync = FALSE; } - /* found frame (up to final), record gathered metadata */ - wvparse->wpi = wpi; - wvparse->wph = wph; - - *framesize = frmsize; - gst_base_parse_set_min_frame_size (parse, sizeof (WavpackHeader)); - - return TRUE; - -skip: - GST_LOG_OBJECT (wvparse, "skipping %d", *skipsize); - return FALSE; - -more: - GST_LOG_OBJECT (wvparse, "need at least %u", frmsize); - gst_base_parse_set_min_frame_size (parse, frmsize); - *skipsize = 0; - return FALSE; -} - -static GstFlowReturn -gst_wavpack_parse_parse_frame (GstBaseParse * parse, GstBaseParseFrame * frame) -{ - GstWavpackParse *wvparse = GST_WAVPACK_PARSE (parse); - GstBuffer *buf = frame->buffer; - guint rate, chans, width, mask; - - /* re-use previously parsed data */ - rate = wvparse->wpi.rate; - width = wvparse->wpi.width; - chans = wvparse->wpi.channels; - mask = wvparse->wpi.channel_mask; + rate = wpi.rate; + width = wpi.width; + chans = wpi.channels; + mask = wpi.channel_mask; GST_LOG_OBJECT (parse, "rate: %u, width: %u, chans: %u", rate, width, chans); GST_BUFFER_TIMESTAMP (buf) = - gst_util_uint64_scale_int (wvparse->wph.block_index, GST_SECOND, rate); + gst_util_uint64_scale_int (wph.block_index, GST_SECOND, rate); GST_BUFFER_DURATION (buf) = - gst_util_uint64_scale_int (wvparse->wph.block_index + - wvparse->wph.block_samples, GST_SECOND, rate) - - GST_BUFFER_TIMESTAMP (buf); + gst_util_uint64_scale_int (wph.block_index + wph.block_samples, + GST_SECOND, rate) - GST_BUFFER_TIMESTAMP (buf); if (G_UNLIKELY (wvparse->sample_rate != rate || wvparse->channels != chans || wvparse->width != width || wvparse->channel_mask != mask)) { GstCaps *caps; - if (wvparse->wpi.correction) { + if (wpi.correction) { caps = gst_caps_new_simple ("audio/x-wavpack-correction", "framed", G_TYPE_BOOLEAN, TRUE, NULL); } else { caps = gst_caps_new_simple ("audio/x-wavpack", "channels", G_TYPE_INT, chans, "rate", G_TYPE_INT, rate, - "width", G_TYPE_INT, width, "framed", G_TYPE_BOOLEAN, TRUE, NULL); + "depth", G_TYPE_INT, width, "framed", G_TYPE_BOOLEAN, TRUE, NULL); if (!mask) mask = gst_wavpack_get_default_channel_mask (wvparse->channels); if (mask != 0) { - if (!gst_wavpack_set_channel_layout (caps, mask)) { - GST_WARNING_OBJECT (wvparse, "Failed to set channel layout"); + GstAudioChannelPosition pos[64] = + { GST_AUDIO_CHANNEL_POSITION_INVALID, }; + guint64 gmask; + + if (!gst_wavpack_get_channel_positions (chans, mask, pos)) { + GST_WARNING_OBJECT (wvparse, "Failed to determine channel layout"); + } else { + gst_audio_channel_positions_to_mask (pos, chans, &gmask); + if (gmask) + gst_caps_set_simple (caps, + "channel-mask", GST_TYPE_BITMASK, gmask, NULL); } } } - gst_buffer_set_caps (buf, caps); gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps); gst_caps_unref (caps); @@ -611,15 +585,33 @@ gst_wavpack_parse_parse_frame (GstBaseParse * parse, GstBaseParseFrame * frame) } } + /* return to normal size */ + gst_base_parse_set_min_frame_size (parse, sizeof (WavpackHeader)); + gst_buffer_unmap (buf, &map); + + return gst_base_parse_finish_frame (parse, frame, frmsize); + +skip: + gst_buffer_unmap (buf, &map); + GST_LOG_OBJECT (wvparse, "skipping %d", *skipsize); + return GST_FLOW_OK; + +more: + gst_buffer_unmap (buf, &map); + GST_LOG_OBJECT (wvparse, "need at least %u", frmsize); + gst_base_parse_set_min_frame_size (parse, frmsize); + *skipsize = 0; return GST_FLOW_OK; } static GstCaps * -gst_wavpack_parse_get_sink_caps (GstBaseParse * parse) +gst_wavpack_parse_get_sink_caps (GstBaseParse * parse, GstCaps * filter) { GstCaps *peercaps; GstCaps *res; + /* FIXME: handle filter caps */ + peercaps = gst_pad_get_allowed_caps (GST_BASE_PARSE_SRC_PAD (parse)); if (peercaps) { guint i, n; diff --git a/gst/audioparsers/gstwavpackparse.h b/gst/audioparsers/gstwavpackparse.h index 7fc246eead..72ad340a15 100644 --- a/gst/audioparsers/gstwavpackparse.h +++ b/gst/audioparsers/gstwavpackparse.h @@ -112,9 +112,6 @@ struct _GstWavpackParse { gint channel_mask; guint total_samples; - - WavpackHeader wph; - WavpackInfo wpi; }; /** diff --git a/gst/audioparsers/plugin.c b/gst/audioparsers/plugin.c index d18ad01ed0..fe64a8a419 100644 --- a/gst/audioparsers/plugin.c +++ b/gst/audioparsers/plugin.c @@ -46,10 +46,8 @@ plugin_init (GstPlugin * plugin) GST_RANK_PRIMARY + 1, GST_TYPE_FLAC_PARSE); ret &= gst_element_register (plugin, "mpegaudioparse", GST_RANK_PRIMARY + 2, GST_TYPE_MPEG_AUDIO_PARSE); -#if 0 - ret &= gst_element_register (plugin, "wavpackparse2", - GST_RANK_SECONDARY, GST_TYPE_WAVPACK_PARSE); -#endif + ret &= gst_element_register (plugin, "wavpackparse", + GST_RANK_PRIMARY + 1, GST_TYPE_WAVPACK_PARSE); return ret; } From 8721817f6df7cf42a055f42ee65ee05659f89f1b Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Mon, 5 Mar 2012 13:34:36 +0100 Subject: [PATCH 50/55] wavpack: remove legacy wavpackparse --- docs/plugins/Makefile.am | 1 - ext/wavpack/Makefile.am | 2 - ext/wavpack/gstwavpack.c | 4 +- ext/wavpack/gstwavpackparse.c | 1302 --------------------------------- ext/wavpack/gstwavpackparse.h | 97 --- 5 files changed, 1 insertion(+), 1405 deletions(-) delete mode 100644 ext/wavpack/gstwavpackparse.c delete mode 100644 ext/wavpack/gstwavpackparse.h diff --git a/docs/plugins/Makefile.am b/docs/plugins/Makefile.am index 51dbc1259b..c39680140a 100644 --- a/docs/plugins/Makefile.am +++ b/docs/plugins/Makefile.am @@ -84,7 +84,6 @@ EXTRA_HFILES = \ $(top_srcdir)/ext/speex/gstspeexdec.h \ $(top_srcdir)/ext/wavpack/gstwavpackdec.h \ $(top_srcdir)/ext/wavpack/gstwavpackenc.h \ - $(top_srcdir)/ext/wavpack/gstwavpackparse.h \ $(top_srcdir)/gst/alpha/gstalpha.h \ $(top_srcdir)/gst/alpha/gstalphacolor.h \ $(top_srcdir)/gst/apetag/gstapedemux.h \ diff --git a/ext/wavpack/Makefile.am b/ext/wavpack/Makefile.am index 70d2431468..7c748da163 100644 --- a/ext/wavpack/Makefile.am +++ b/ext/wavpack/Makefile.am @@ -3,7 +3,6 @@ plugin_LTLIBRARIES = libgstwavpack.la libgstwavpack_la_SOURCES = \ gstwavpack.c \ gstwavpackcommon.c \ - gstwavpackparse.c \ gstwavpackdec.c \ gstwavpackenc.c \ gstwavpackstreamreader.c @@ -16,7 +15,6 @@ libgstwavpack_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) libgstwavpack_la_LIBTOOLFLAGS = --tag=disable-static noinst_HEADERS = \ - gstwavpackparse.h \ gstwavpackdec.h \ gstwavpackenc.h \ gstwavpackcommon.h \ diff --git a/ext/wavpack/gstwavpack.c b/ext/wavpack/gstwavpack.c index b01f443d29..6f765f7f97 100644 --- a/ext/wavpack/gstwavpack.c +++ b/ext/wavpack/gstwavpack.c @@ -25,7 +25,6 @@ #include -#include "gstwavpackparse.h" #include "gstwavpackdec.h" #include "gstwavpackenc.h" @@ -44,8 +43,7 @@ plugin_init (GstPlugin * plugin) bind_textdomain_codeset (GETTEXT_PACKAGE, "UTF-8"); #endif - return (gst_wavpack_parse_plugin_init (plugin) - && gst_wavpack_dec_plugin_init (plugin) + return (gst_wavpack_dec_plugin_init (plugin) && gst_wavpack_enc_plugin_init (plugin)); } diff --git a/ext/wavpack/gstwavpackparse.c b/ext/wavpack/gstwavpackparse.c deleted file mode 100644 index 0af0a90ea5..0000000000 --- a/ext/wavpack/gstwavpackparse.c +++ /dev/null @@ -1,1302 +0,0 @@ -/* GStreamer wavpack plugin - * Copyright (c) 2005 Arwed v. Merkatz - * Copyright (c) 2006 Tim-Philipp Müller - * Copyright (c) 2006 Sebastian Dröge - * - * gstwavpackparse.c: wavpack file 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. - */ - -/** - * SECTION:element-wavpackparse - * - * WavpackParse takes raw, unframed Wavpack streams and splits them into - * single Wavpack chunks with information like bit depth and the position - * in the stream. - * Wavpack is an open-source - * audio codec that features both lossless and lossy encoding. - * - * - * Example launch line - * |[ - * gst-launch filesrc location=test.wv ! wavpackparse ! wavpackdec ! fakesink - * ]| This pipeline decodes the Wavpack file test.wv into raw audio buffers. - * - */ - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif - -#include -#include - -#include -#include - -#include -#include "gstwavpackparse.h" -#include "gstwavpackstreamreader.h" -#include "gstwavpackcommon.h" - -GST_DEBUG_CATEGORY_STATIC (gst_wavpack_parse_debug); -#define GST_CAT_DEFAULT gst_wavpack_parse_debug - -static inline GstWavpackParseIndexEntry * -gst_wavpack_parse_index_entry_new (void) -{ - return g_slice_new (GstWavpackParseIndexEntry); -} - -static inline void -gst_wavpack_parse_index_entry_free (GstWavpackParseIndexEntry * entry) -{ - g_slice_free (GstWavpackParseIndexEntry, entry); -} - -static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", - GST_PAD_SINK, - GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/x-wavpack, " - "framed = (boolean) false; " - "audio/x-wavpack-correction, " "framed = (boolean) false") - ); - -static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src", - GST_PAD_SRC, - GST_PAD_SOMETIMES, - GST_STATIC_CAPS ("audio/x-wavpack, " - "width = (int) [ 1, 32 ], " - "channels = (int) [ 1, 8 ], " - "rate = (int) [ 6000, 192000 ], " "framed = (boolean) true") - ); - -static GstStaticPadTemplate wvc_src_factory = GST_STATIC_PAD_TEMPLATE ("wvcsrc", - GST_PAD_SRC, - GST_PAD_SOMETIMES, - GST_STATIC_CAPS ("audio/x-wavpack-correction, " "framed = (boolean) true") - ); - -static gboolean gst_wavpack_parse_sink_activate (GstPad * sinkpad); - -static gboolean -gst_wavpack_parse_sink_activate_pull (GstPad * sinkpad, gboolean active); - -static void gst_wavpack_parse_loop (GstElement * element); - -static GstStateChangeReturn gst_wavpack_parse_change_state (GstElement * - element, GstStateChange transition); -static void gst_wavpack_parse_reset (GstWavpackParse * parse); - -static gint64 gst_wavpack_parse_get_upstream_length (GstWavpackParse * wvparse); - -static GstBuffer *gst_wavpack_parse_pull_buffer (GstWavpackParse * wvparse, - gint64 offset, guint size, GstFlowReturn * flow); -static GstFlowReturn gst_wavpack_parse_chain (GstPad * pad, GstBuffer * buf); - -GST_BOILERPLATE (GstWavpackParse, gst_wavpack_parse, GstElement, - GST_TYPE_ELEMENT); - -static void -gst_wavpack_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_factory)); - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&wvc_src_factory)); - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&sink_factory)); - - gst_element_class_set_details_simple (element_class, "Wavpack parser", - "Codec/Demuxer/Audio", - "Parses Wavpack files", - "Arwed v. Merkatz , " - "Sebastian Dröge "); -} - -static void -gst_wavpack_parse_finalize (GObject * object) -{ - gst_wavpack_parse_reset (GST_WAVPACK_PARSE (object)); - - G_OBJECT_CLASS (parent_class)->finalize (object); -} - -static void -gst_wavpack_parse_class_init (GstWavpackParseClass * klass) -{ - GObjectClass *gobject_class; - GstElementClass *gstelement_class; - - gobject_class = (GObjectClass *) klass; - gstelement_class = (GstElementClass *) klass; - - gobject_class->finalize = gst_wavpack_parse_finalize; - gstelement_class->change_state = - GST_DEBUG_FUNCPTR (gst_wavpack_parse_change_state); -} - -static GstWavpackParseIndexEntry * -gst_wavpack_parse_index_get_last_entry (GstWavpackParse * wvparse) -{ - g_assert (wvparse->entries != NULL); - - return wvparse->entries->data; -} - -static GstWavpackParseIndexEntry * -gst_wavpack_parse_index_get_entry_from_sample (GstWavpackParse * wvparse, - gint64 sample_offset) -{ - gint i; - GSList *node; - - if (wvparse->entries == NULL) - return NULL; - - for (node = wvparse->entries, i = 0; node; node = node->next, i++) { - GstWavpackParseIndexEntry *entry; - - entry = node->data; - - GST_LOG_OBJECT (wvparse, "Index entry %03u: sample %" G_GINT64_FORMAT " @" - " byte %" G_GINT64_FORMAT, i, entry->sample_offset, entry->byte_offset); - - if (entry->sample_offset <= sample_offset && - sample_offset < entry->sample_offset_end) { - GST_LOG_OBJECT (wvparse, "found match"); - return entry; - } - - /* as the list is sorted and we first look at the latest entry - * we can abort searching for an entry if the sample we want is - * after the latest one */ - if (sample_offset >= entry->sample_offset_end) - break; - } - GST_LOG_OBJECT (wvparse, "no match in index"); - return NULL; -} - -static void -gst_wavpack_parse_index_append_entry (GstWavpackParse * wvparse, - gint64 byte_offset, gint64 sample_offset, gint64 num_samples) -{ - GstWavpackParseIndexEntry *entry; - - /* do we have this one already? */ - if (wvparse->entries) { - entry = gst_wavpack_parse_index_get_last_entry (wvparse); - if (entry->byte_offset >= byte_offset - || entry->sample_offset >= sample_offset) - return; - } - - GST_LOG_OBJECT (wvparse, "Adding index entry %8" G_GINT64_FORMAT " - %" - GST_TIME_FORMAT " @ offset 0x%08" G_GINT64_MODIFIER "x", sample_offset, - GST_TIME_ARGS (gst_util_uint64_scale_int (sample_offset, - GST_SECOND, wvparse->samplerate)), byte_offset); - - entry = gst_wavpack_parse_index_entry_new (); - entry->byte_offset = byte_offset; - entry->sample_offset = sample_offset; - entry->sample_offset_end = sample_offset + num_samples; - wvparse->entries = g_slist_prepend (wvparse->entries, entry); -} - -static void -gst_wavpack_parse_reset (GstWavpackParse * parse) -{ - parse->total_samples = G_GINT64_CONSTANT (-1); - parse->samplerate = 0; - parse->channels = 0; - - gst_segment_init (&parse->segment, GST_FORMAT_UNDEFINED); - parse->next_block_index = 0; - - parse->current_offset = 0; - parse->need_newsegment = TRUE; - parse->discont = TRUE; - parse->upstream_length = -1; - - if (parse->entries) { - g_slist_foreach (parse->entries, (GFunc) gst_wavpack_parse_index_entry_free, - NULL); - g_slist_free (parse->entries); - parse->entries = NULL; - } - - if (parse->adapter) { - gst_adapter_clear (parse->adapter); - g_object_unref (parse->adapter); - parse->adapter = NULL; - } - - if (parse->srcpad != NULL) { - gboolean res; - - GST_DEBUG_OBJECT (parse, "Removing src pad"); - res = gst_element_remove_pad (GST_ELEMENT (parse), parse->srcpad); - g_return_if_fail (res != FALSE); - gst_object_unref (parse->srcpad); - parse->srcpad = NULL; - } - - g_list_foreach (parse->queued_events, (GFunc) gst_mini_object_unref, NULL); - g_list_free (parse->queued_events); - parse->queued_events = NULL; - - if (parse->pending_buffer) - gst_buffer_unref (parse->pending_buffer); - - parse->pending_buffer = NULL; -} - -static const GstQueryType * -gst_wavpack_parse_get_src_query_types (GstPad * pad) -{ - static const GstQueryType types[] = { - GST_QUERY_POSITION, - GST_QUERY_DURATION, - GST_QUERY_SEEKING, - 0 - }; - - return types; -} - -static gboolean -gst_wavpack_parse_src_query (GstPad * pad, GstQuery * query) -{ - GstWavpackParse *parse = GST_WAVPACK_PARSE (gst_pad_get_parent (pad)); - GstFormat format; - gboolean ret = FALSE; - - switch (GST_QUERY_TYPE (query)) { - case GST_QUERY_POSITION:{ - gint64 cur; - - guint rate; - - GST_OBJECT_LOCK (parse); - cur = parse->segment.last_stop; - rate = parse->samplerate; - GST_OBJECT_UNLOCK (parse); - - if (rate == 0) { - GST_DEBUG_OBJECT (parse, "haven't read header yet"); - break; - } - - gst_query_parse_position (query, &format, NULL); - - switch (format) { - case GST_FORMAT_TIME: - cur = gst_util_uint64_scale_int (cur, GST_SECOND, rate); - gst_query_set_position (query, GST_FORMAT_TIME, cur); - ret = TRUE; - break; - case GST_FORMAT_DEFAULT: - gst_query_set_position (query, GST_FORMAT_DEFAULT, cur); - ret = TRUE; - break; - default: - GST_DEBUG_OBJECT (parse, "cannot handle position query in " - "%s format. Forwarding upstream.", gst_format_get_name (format)); - ret = gst_pad_query_default (pad, query); - break; - } - break; - } - case GST_QUERY_DURATION:{ - gint64 len; - - guint rate; - - GST_OBJECT_LOCK (parse); - rate = parse->samplerate; - len = parse->total_samples; - GST_OBJECT_UNLOCK (parse); - - if (rate == 0) { - GST_DEBUG_OBJECT (parse, "haven't read header yet"); - break; - } - - gst_query_parse_duration (query, &format, NULL); - - switch (format) { - case GST_FORMAT_TIME: - if (len != G_GINT64_CONSTANT (-1)) - len = gst_util_uint64_scale_int (len, GST_SECOND, rate); - gst_query_set_duration (query, GST_FORMAT_TIME, len); - ret = TRUE; - break; - case GST_FORMAT_DEFAULT: - gst_query_set_duration (query, GST_FORMAT_DEFAULT, len); - ret = TRUE; - break; - default: - GST_DEBUG_OBJECT (parse, "cannot handle duration query in " - "%s format. Forwarding upstream.", gst_format_get_name (format)); - ret = gst_pad_query_default (pad, query); - break; - } - break; - } - case GST_QUERY_SEEKING:{ - gst_query_parse_seeking (query, &format, NULL, NULL, NULL); - if (format == GST_FORMAT_TIME || format == GST_FORMAT_DEFAULT) { - gboolean seekable; - - gint64 duration = -1; - - /* only fails if we didn't read the headers yet and can't say - * anything about our seeking capabilities */ - if (!gst_pad_query_duration (pad, &format, &duration)) - break; - - /* can't seek in streaming mode yet */ - GST_OBJECT_LOCK (parse); - seekable = (parse->adapter == NULL); - GST_OBJECT_UNLOCK (parse); - - gst_query_set_seeking (query, format, seekable, 0, duration); - ret = TRUE; - } - break; - } - default:{ - ret = gst_pad_query_default (pad, query); - break; - } - } - - gst_object_unref (parse); - return ret; - -} - -/* returns TRUE on success, with byte_offset set to the offset of the - * wavpack chunk containing the sample requested. start_sample will be - * set to the first sample in the chunk starting at byte_offset. - * Scanning from the last known header offset to the wanted position - * when seeking forward isn't very clever, but seems fast enough in - * practice and has the nice side effect of populating our index - * table */ -static gboolean -gst_wavpack_parse_scan_to_find_sample (GstWavpackParse * parse, - gint64 sample, gint64 * byte_offset, gint64 * start_sample) -{ - GstWavpackParseIndexEntry *entry; - GstFlowReturn ret; - gint64 off = 0; - - /* first, check if we have to scan at all */ - entry = gst_wavpack_parse_index_get_entry_from_sample (parse, sample); - if (entry) { - *byte_offset = entry->byte_offset; - *start_sample = entry->sample_offset; - GST_LOG_OBJECT (parse, "Found index entry: sample %" G_GINT64_FORMAT - " @ offset %" G_GINT64_FORMAT, entry->sample_offset, - entry->byte_offset); - return TRUE; - } - - GST_LOG_OBJECT (parse, "No matching entry in index, scanning file ..."); - - /* if we have an index, we can start scanning from the last known offset - * in there, after all we know our wanted sample is not in the index */ - if (parse->entries) { - GstWavpackParseIndexEntry *entry; - - entry = gst_wavpack_parse_index_get_last_entry (parse); - off = entry->byte_offset; - } - - /* now scan forward until we find the chunk we're looking for or hit EOS */ - do { - WavpackHeader header; - GstBuffer *buf; - - buf = gst_wavpack_parse_pull_buffer (parse, off, sizeof (WavpackHeader), - &ret); - - if (buf == NULL) - break; - - gst_wavpack_read_header (&header, GST_BUFFER_DATA (buf)); - gst_buffer_unref (buf); - - if (header.flags & INITIAL_BLOCK) - gst_wavpack_parse_index_append_entry (parse, off, header.block_index, - header.block_samples); - else - continue; - - if (header.block_index <= sample && - sample < (header.block_index + header.block_samples)) { - *byte_offset = off; - *start_sample = header.block_index; - return TRUE; - } - - off += header.ckSize + 8; - } while (1); - - GST_DEBUG_OBJECT (parse, "scan failed: %s (off=0x%08" G_GINT64_MODIFIER "x)", - gst_flow_get_name (ret), off); - - return FALSE; -} - -static gboolean -gst_wavpack_parse_send_newsegment (GstWavpackParse * wvparse, gboolean update) -{ - GstSegment *s = &wvparse->segment; - - gboolean ret; - gint64 stop_time = -1; - gint64 start_time = 0; - gint64 cur_pos_time; - gint64 diff; - - /* segment is in DEFAULT format, but we want to send a TIME newsegment */ - start_time = gst_util_uint64_scale_int (s->start, GST_SECOND, - wvparse->samplerate); - - if (s->stop != -1) { - stop_time = gst_util_uint64_scale_int (s->stop, GST_SECOND, - wvparse->samplerate); - } - - GST_DEBUG_OBJECT (wvparse, "sending newsegment from %" GST_TIME_FORMAT - " to %" GST_TIME_FORMAT, GST_TIME_ARGS (start_time), - GST_TIME_ARGS (stop_time)); - - /* after a seek, s->last_stop will point to a chunk boundary, ie. from - * which sample we will start sending data again, while s->start will - * point to the sample we actually want to seek to and want to start - * playing right after the seek. Adjust clock-time for the difference - * so we start playing from start_time */ - cur_pos_time = gst_util_uint64_scale_int (s->last_stop, GST_SECOND, - wvparse->samplerate); - diff = start_time - cur_pos_time; - - ret = gst_pad_push_event (wvparse->srcpad, - gst_event_new_new_segment (update, s->rate, GST_FORMAT_TIME, - start_time, stop_time, start_time - diff)); - - return ret; -} - -static gboolean -gst_wavpack_parse_handle_seek_event (GstWavpackParse * wvparse, - GstEvent * event) -{ - GstSeekFlags seek_flags; - GstSeekType start_type; - GstSeekType stop_type; - GstSegment segment; - GstFormat format; - gboolean only_update; - gboolean flush, ret; - gdouble speed; - gint64 stop; - gint64 start; /* sample we want to seek to */ - gint64 byte_offset; /* byte offset the chunk we seek to starts at */ - gint64 chunk_start; /* first sample in chunk we seek to */ - guint rate; - gint64 last_stop; - - if (wvparse->adapter) { - GST_DEBUG_OBJECT (wvparse, "seeking in streaming mode not implemented yet"); - return FALSE; - } - - gst_event_parse_seek (event, &speed, &format, &seek_flags, &start_type, - &start, &stop_type, &stop); - - if (format != GST_FORMAT_DEFAULT && format != GST_FORMAT_TIME) { - GST_DEBUG ("seeking is only supported in TIME or DEFAULT format"); - return FALSE; - } - - if (speed < 0.0) { - GST_DEBUG ("only forward playback supported, rate %f not allowed", speed); - return FALSE; - } - - GST_OBJECT_LOCK (wvparse); - - rate = wvparse->samplerate; - if (rate == 0) { - GST_OBJECT_UNLOCK (wvparse); - GST_DEBUG ("haven't read header yet"); - return FALSE; - } - - /* figure out the last position we need to play. If it's configured (stop != - * -1), use that, else we play until the total duration of the file */ - if (stop == -1) - stop = wvparse->segment.duration; - - /* convert from time to samples if necessary */ - if (format == GST_FORMAT_TIME) { - if (start_type != GST_SEEK_TYPE_NONE) - start = gst_util_uint64_scale_int (start, rate, GST_SECOND); - if (stop_type != GST_SEEK_TYPE_NONE) - stop = gst_util_uint64_scale_int (stop, rate, GST_SECOND); - } - - if (start < 0) { - GST_OBJECT_UNLOCK (wvparse); - GST_DEBUG_OBJECT (wvparse, "Invalid start sample %" G_GINT64_FORMAT, start); - return FALSE; - } - - flush = ((seek_flags & GST_SEEK_FLAG_FLUSH) != 0); - - /* operate on segment copy until we know the seek worked */ - segment = wvparse->segment; - - gst_segment_set_seek (&segment, speed, GST_FORMAT_DEFAULT, - seek_flags, start_type, start, stop_type, stop, &only_update); - -#if 0 - if (only_update) { - wvparse->segment = segment; - gst_wavpack_parse_send_newsegment (wvparse, TRUE); - goto done; - } -#endif - - gst_pad_push_event (wvparse->sinkpad, gst_event_new_flush_start ()); - - if (flush) { - gst_pad_push_event (wvparse->srcpad, gst_event_new_flush_start ()); - } else { - gst_pad_pause_task (wvparse->sinkpad); - } - - GST_PAD_STREAM_LOCK (wvparse->sinkpad); - - /* Save current position */ - last_stop = wvparse->segment.last_stop; - - gst_pad_push_event (wvparse->sinkpad, gst_event_new_flush_stop ()); - - if (flush) { - gst_pad_push_event (wvparse->srcpad, gst_event_new_flush_stop ()); - } - - GST_DEBUG_OBJECT (wvparse, "Performing seek to %" GST_TIME_FORMAT " sample %" - G_GINT64_FORMAT, GST_TIME_ARGS (segment.start * GST_SECOND / rate), - start); - - ret = gst_wavpack_parse_scan_to_find_sample (wvparse, segment.start, - &byte_offset, &chunk_start); - - if (ret) { - GST_DEBUG_OBJECT (wvparse, "new offset: %" G_GINT64_FORMAT, byte_offset); - wvparse->current_offset = byte_offset; - /* we want to send a newsegment event with the actual seek position - * as start, even though our first buffer might start before the - * configured segment. We leave it up to the decoder or sink to crop - * the output buffers accordingly */ - wvparse->segment = segment; - wvparse->segment.last_stop = chunk_start; - wvparse->need_newsegment = TRUE; - wvparse->discont = (last_stop != chunk_start) ? TRUE : FALSE; - - /* if we're doing a segment seek, post a SEGMENT_START message */ - if (wvparse->segment.flags & GST_SEEK_FLAG_SEGMENT) { - gst_element_post_message (GST_ELEMENT_CAST (wvparse), - gst_message_new_segment_start (GST_OBJECT_CAST (wvparse), - wvparse->segment.format, wvparse->segment.last_stop)); - } - } else { - GST_DEBUG_OBJECT (wvparse, "seek failed: don't know where to seek to"); - } - - GST_PAD_STREAM_UNLOCK (wvparse->sinkpad); - GST_OBJECT_UNLOCK (wvparse); - - gst_pad_start_task (wvparse->sinkpad, - (GstTaskFunction) gst_wavpack_parse_loop, wvparse); - - return ret; -} - -static gboolean -gst_wavpack_parse_sink_event (GstPad * pad, GstEvent * event) -{ - GstWavpackParse *parse; - gboolean ret = TRUE; - - parse = GST_WAVPACK_PARSE (gst_pad_get_parent (pad)); - - switch (GST_EVENT_TYPE (event)) { - case GST_EVENT_FLUSH_STOP:{ - if (parse->adapter) { - gst_adapter_clear (parse->adapter); - } - if (parse->pending_buffer) { - gst_buffer_unref (parse->pending_buffer); - parse->pending_buffer = NULL; - parse->pending_offset = 0; - } - ret = gst_pad_push_event (parse->srcpad, event); - break; - } - case GST_EVENT_NEWSEGMENT:{ - parse->need_newsegment = TRUE; - gst_event_unref (event); - ret = TRUE; - break; - } - case GST_EVENT_EOS:{ - if (parse->adapter) { - /* remove all bytes that are left in the adapter after EOS. They can't - * be a complete Wavpack block and we can't do anything with them */ - gst_adapter_clear (parse->adapter); - } - if (parse->pending_buffer) { - gst_buffer_unref (parse->pending_buffer); - parse->pending_buffer = NULL; - parse->pending_offset = 0; - } - ret = gst_pad_push_event (parse->srcpad, event); - break; - } - default:{ - /* stream lock is recursive, should be fine for all events */ - GST_PAD_STREAM_LOCK (pad); - if (parse->srcpad == NULL) { - parse->queued_events = g_list_append (parse->queued_events, event); - } else { - ret = gst_pad_push_event (parse->srcpad, event); - } - GST_PAD_STREAM_UNLOCK (pad); - } - } - - - gst_object_unref (parse); - return ret; -} - -static gboolean -gst_wavpack_parse_src_event (GstPad * pad, GstEvent * event) -{ - GstWavpackParse *parse; - gboolean ret; - - parse = GST_WAVPACK_PARSE (gst_pad_get_parent (pad)); - - switch (GST_EVENT_TYPE (event)) { - case GST_EVENT_SEEK: - ret = gst_wavpack_parse_handle_seek_event (parse, event); - break; - default: - ret = gst_pad_event_default (pad, event); - break; - } - - gst_object_unref (parse); - return ret; -} - -static void -gst_wavpack_parse_init (GstWavpackParse * parse, GstWavpackParseClass * gclass) -{ - GstElementClass *klass = GST_ELEMENT_GET_CLASS (parse); - GstPadTemplate *tmpl; - - tmpl = gst_element_class_get_pad_template (klass, "sink"); - parse->sinkpad = gst_pad_new_from_template (tmpl, "sink"); - - gst_pad_set_activate_function (parse->sinkpad, - GST_DEBUG_FUNCPTR (gst_wavpack_parse_sink_activate)); - gst_pad_set_activatepull_function (parse->sinkpad, - GST_DEBUG_FUNCPTR (gst_wavpack_parse_sink_activate_pull)); - gst_pad_set_event_function (parse->sinkpad, - GST_DEBUG_FUNCPTR (gst_wavpack_parse_sink_event)); - gst_pad_set_chain_function (parse->sinkpad, - GST_DEBUG_FUNCPTR (gst_wavpack_parse_chain)); - - gst_element_add_pad (GST_ELEMENT (parse), parse->sinkpad); - - parse->srcpad = NULL; - gst_wavpack_parse_reset (parse); -} - -static gint64 -gst_wavpack_parse_get_upstream_length (GstWavpackParse * parse) -{ - gint64 length = -1; - GstFormat format = GST_FORMAT_BYTES; - - if (!gst_pad_query_peer_duration (parse->sinkpad, &format, &length)) { - length = -1; - } else { - GST_DEBUG ("upstream length: %" G_GINT64_FORMAT, length); - } - return length; -} - -static GstBuffer * -gst_wavpack_parse_pull_buffer (GstWavpackParse * wvparse, gint64 offset, - guint size, GstFlowReturn * flow) -{ - GstFlowReturn flow_ret; - GstBuffer *buf = NULL; - - if (offset + size > wvparse->upstream_length) { - wvparse->upstream_length = gst_wavpack_parse_get_upstream_length (wvparse); - if (offset + size > wvparse->upstream_length) { - GST_DEBUG_OBJECT (wvparse, "EOS: %" G_GINT64_FORMAT " + %u > %" - G_GINT64_FORMAT, offset, size, wvparse->upstream_length); - flow_ret = GST_FLOW_EOS; - goto done; - } - } - - flow_ret = gst_pad_pull_range (wvparse->sinkpad, offset, size, &buf); - - if (flow_ret != GST_FLOW_OK) { - GST_DEBUG_OBJECT (wvparse, "pull_range (%" G_GINT64_FORMAT ", %u) " - "failed, flow: %s", offset, size, gst_flow_get_name (flow_ret)); - buf = NULL; - goto done; - } - - if (GST_BUFFER_SIZE (buf) < size) { - GST_DEBUG_OBJECT (wvparse, "Short read at offset %" G_GINT64_FORMAT - ", got only %u of %u bytes", offset, GST_BUFFER_SIZE (buf), size); - gst_buffer_unref (buf); - buf = NULL; - flow_ret = GST_FLOW_EOS; - } - -done: - if (flow) - *flow = flow_ret; - return buf; -} - -static gboolean -gst_wavpack_parse_create_src_pad (GstWavpackParse * wvparse, GstBuffer * buf, - WavpackHeader * header) -{ - GstWavpackMetadata meta; - GstCaps *caps = NULL; - guchar *bufptr; - - g_assert (wvparse->srcpad == NULL); - - bufptr = GST_BUFFER_DATA (buf) + sizeof (WavpackHeader); - - while (gst_wavpack_read_metadata (&meta, GST_BUFFER_DATA (buf), &bufptr)) { - switch (meta.id) { - case ID_WVC_BITSTREAM:{ - caps = gst_caps_new_simple ("audio/x-wavpack-correction", - "framed", G_TYPE_BOOLEAN, TRUE, NULL); - wvparse->srcpad = - gst_pad_new_from_template (gst_element_class_get_pad_template - (GST_ELEMENT_GET_CLASS (wvparse), "wvcsrc"), "wvcsrc"); - break; - } - case ID_WV_BITSTREAM: - case ID_WVX_BITSTREAM:{ - WavpackStreamReader *stream_reader = gst_wavpack_stream_reader_new (); - - WavpackContext *wpc; - - gchar error_msg[80]; - - read_id rid; - - gint channel_mask; - - rid.buffer = GST_BUFFER_DATA (buf); - rid.length = GST_BUFFER_SIZE (buf); - rid.position = 0; - - wpc = - WavpackOpenFileInputEx (stream_reader, &rid, NULL, error_msg, 0, 0); - - if (!wpc) - return FALSE; - - wvparse->samplerate = WavpackGetSampleRate (wpc); - wvparse->channels = WavpackGetNumChannels (wpc); - wvparse->total_samples = - (header->total_samples == - 0xffffffff) ? G_GINT64_CONSTANT (-1) : header->total_samples; - - caps = gst_caps_new_simple ("audio/x-wavpack", - "width", G_TYPE_INT, WavpackGetBitsPerSample (wpc), - "channels", G_TYPE_INT, wvparse->channels, - "rate", G_TYPE_INT, wvparse->samplerate, - "framed", G_TYPE_BOOLEAN, TRUE, NULL); -#ifdef WAVPACK_OLD_API - channel_mask = wpc->config.channel_mask; -#else - channel_mask = WavpackGetChannelMask (wpc); -#endif - if (channel_mask == 0) - channel_mask = - gst_wavpack_get_default_channel_mask (wvparse->channels); - - if (channel_mask != 0) { - if (!gst_wavpack_set_channel_layout (caps, channel_mask)) { - GST_WARNING_OBJECT (wvparse, "Failed to set channel layout"); - gst_caps_unref (caps); - caps = NULL; - WavpackCloseFile (wpc); - g_free (stream_reader); - break; - } - } - - wvparse->srcpad = - gst_pad_new_from_template (gst_element_class_get_pad_template - (GST_ELEMENT_GET_CLASS (wvparse), "src"), "src"); - WavpackCloseFile (wpc); - g_free (stream_reader); - break; - } - default:{ - GST_LOG_OBJECT (wvparse, "unhandled ID: 0x%02x", meta.id); - break; - } - } - if (caps != NULL) - break; - } - - if (caps == NULL || wvparse->srcpad == NULL) - return FALSE; - - GST_DEBUG_OBJECT (wvparse, "Added src pad with caps %" GST_PTR_FORMAT, caps); - - gst_pad_set_query_function (wvparse->srcpad, - GST_DEBUG_FUNCPTR (gst_wavpack_parse_src_query)); - gst_pad_set_query_type_function (wvparse->srcpad, - GST_DEBUG_FUNCPTR (gst_wavpack_parse_get_src_query_types)); - gst_pad_set_event_function (wvparse->srcpad, - GST_DEBUG_FUNCPTR (gst_wavpack_parse_src_event)); - - gst_pad_set_caps (wvparse->srcpad, caps); - gst_caps_unref (caps); - gst_pad_use_fixed_caps (wvparse->srcpad); - - gst_object_ref (wvparse->srcpad); - gst_pad_set_active (wvparse->srcpad, TRUE); - gst_element_add_pad (GST_ELEMENT (wvparse), wvparse->srcpad); - gst_element_no_more_pads (GST_ELEMENT (wvparse)); - - return TRUE; -} - -static GstFlowReturn -gst_wavpack_parse_push_buffer (GstWavpackParse * wvparse, GstBuffer * buf, - WavpackHeader * header) -{ - GstFlowReturn ret; - - wvparse->current_offset += header->ckSize + 8; - wvparse->segment.last_stop = header->block_index; - - if (wvparse->need_newsegment) { - if (gst_wavpack_parse_send_newsegment (wvparse, FALSE)) - wvparse->need_newsegment = FALSE; - } - - /* send any queued events */ - if (wvparse->queued_events) { - GList *l; - - for (l = wvparse->queued_events; l != NULL; l = l->next) { - gst_pad_push_event (wvparse->srcpad, GST_EVENT (l->data)); - } - g_list_free (wvparse->queued_events); - wvparse->queued_events = NULL; - } - - if (wvparse->pending_buffer == NULL) { - wvparse->pending_buffer = buf; - wvparse->pending_offset = header->block_index; - } else if (wvparse->pending_offset == header->block_index) { - wvparse->pending_buffer = gst_buffer_join (wvparse->pending_buffer, buf); - } else { - GST_ERROR ("Got incomplete block, dropping"); - gst_buffer_unref (wvparse->pending_buffer); - wvparse->pending_buffer = buf; - wvparse->pending_offset = header->block_index; - } - - if (!(header->flags & FINAL_BLOCK)) - return GST_FLOW_OK; - - buf = wvparse->pending_buffer; - wvparse->pending_buffer = NULL; - - GST_BUFFER_TIMESTAMP (buf) = gst_util_uint64_scale_int (header->block_index, - GST_SECOND, wvparse->samplerate); - GST_BUFFER_DURATION (buf) = gst_util_uint64_scale_int (header->block_samples, - GST_SECOND, wvparse->samplerate); - GST_BUFFER_OFFSET (buf) = header->block_index; - GST_BUFFER_OFFSET_END (buf) = header->block_index + header->block_samples; - - if (wvparse->discont || wvparse->next_block_index != header->block_index) { - GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT); - wvparse->discont = FALSE; - } - - wvparse->next_block_index = header->block_index + header->block_samples; - - gst_buffer_set_caps (buf, GST_PAD_CAPS (wvparse->srcpad)); - - GST_LOG_OBJECT (wvparse, "Pushing buffer with time %" GST_TIME_FORMAT, - GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf))); - - ret = gst_pad_push (wvparse->srcpad, buf); - - wvparse->segment.last_stop = wvparse->next_block_index; - - return ret; -} - -static guint8 * -gst_wavpack_parse_find_marker (guint8 * buf, guint size) -{ - gint i; - guint8 *ret = NULL; - - if (G_UNLIKELY (size < 4)) - return NULL; - - for (i = 0; i < size - 4; i++) { - if (memcmp (buf + i, "wvpk", 4) == 0) { - ret = buf + i; - break; - } - } - return ret; -} - -static GstFlowReturn -gst_wavpack_parse_resync_loop (GstWavpackParse * parse, WavpackHeader * header) -{ - GstFlowReturn flow_ret = GST_FLOW_EOS; - GstBuffer *buf = NULL; - - /* loop until we have a frame header or reach the end of the stream */ - while (1) { - guint8 *data, *marker; - - guint len, size; - - if (buf) { - gst_buffer_unref (buf); - buf = NULL; - } - - if (parse->upstream_length == 0 || - parse->upstream_length <= parse->current_offset) { - parse->upstream_length = gst_wavpack_parse_get_upstream_length (parse); - if (parse->upstream_length == 0 || - parse->upstream_length <= parse->current_offset) { - break; - } - } - - len = MIN (parse->upstream_length - parse->current_offset, 2048); - - GST_LOG_OBJECT (parse, "offset: %" G_GINT64_FORMAT, parse->current_offset); - - buf = gst_wavpack_parse_pull_buffer (parse, parse->current_offset, - len, &flow_ret); - - /* whatever the problem is, there's nothing more for us to do for now */ - if (flow_ret != GST_FLOW_OK) - break; - - data = GST_BUFFER_DATA (buf); - size = GST_BUFFER_SIZE (buf); - - /* not enough data for a header? */ - if (size < sizeof (WavpackHeader)) - break; - - /* got a header right where we are at now? */ - if (gst_wavpack_read_header (header, data)) - break; - - /* nope, let's see if we can find one */ - marker = gst_wavpack_parse_find_marker (data + 1, size - 1); - - if (marker) { - parse->current_offset += marker - data; - /* do one more loop iteration to make sure we pull enough - * data for a full header, we'll bail out then */ - } else { - parse->current_offset += len - 4; - } - } - - if (buf) - gst_buffer_unref (buf); - - return flow_ret; -} - -static void -gst_wavpack_parse_loop (GstElement * element) -{ - GstWavpackParse *parse = GST_WAVPACK_PARSE (element); - GstFlowReturn flow_ret; - WavpackHeader header = { {0,}, 0, }; - GstBuffer *buf = NULL; - - flow_ret = gst_wavpack_parse_resync_loop (parse, &header); - - if (flow_ret != GST_FLOW_OK) - goto pause; - - GST_LOG_OBJECT (parse, "Read header at offset %" G_GINT64_FORMAT - ": chunk size = %u+8", parse->current_offset, header.ckSize); - - buf = gst_wavpack_parse_pull_buffer (parse, parse->current_offset, - header.ckSize + 8, &flow_ret); - - if (flow_ret != GST_FLOW_OK) - goto pause; - - if (parse->srcpad == NULL) { - if (!gst_wavpack_parse_create_src_pad (parse, buf, &header)) { - GST_ERROR_OBJECT (parse, "Failed to create src pad"); - flow_ret = GST_FLOW_ERROR; - goto pause; - } - } - if (header.flags & INITIAL_BLOCK) - gst_wavpack_parse_index_append_entry (parse, parse->current_offset, - header.block_index, header.block_samples); - - flow_ret = gst_wavpack_parse_push_buffer (parse, buf, &header); - if (flow_ret != GST_FLOW_OK) - goto pause; - - return; - -pause: - { - const gchar *reason = gst_flow_get_name (flow_ret); - - GST_LOG_OBJECT (parse, "pausing task, reason %s", reason); - gst_pad_pause_task (parse->sinkpad); - - if (flow_ret == GST_FLOW_EOS && parse->srcpad) { - if (parse->segment.flags & GST_SEEK_FLAG_SEGMENT) { - GstClockTime stop; - - GST_LOG_OBJECT (parse, "Sending segment done"); - - if ((stop = parse->segment.stop) == -1) - stop = parse->segment.duration; - - gst_element_post_message (GST_ELEMENT_CAST (parse), - gst_message_new_segment_done (GST_OBJECT_CAST (parse), - parse->segment.format, stop)); - } else { - GST_LOG_OBJECT (parse, "Sending EOS, at end of stream"); - gst_pad_push_event (parse->srcpad, gst_event_new_eos ()); - } - } else if (flow_ret == GST_FLOW_NOT_LINKED || flow_ret < GST_FLOW_EOS) { - GST_ELEMENT_ERROR (parse, STREAM, FAILED, - (_("Internal data stream error.")), ("stream stopped, reason %s", - reason)); - if (parse->srcpad) - gst_pad_push_event (parse->srcpad, gst_event_new_eos ()); - } - return; - } -} - -static gboolean -gst_wavpack_parse_resync_adapter (GstAdapter * adapter) -{ - const guint8 *buf, *marker; - guint avail = gst_adapter_available (adapter); - - if (avail < 4) - return FALSE; - - /* if the marker is at the beginning don't do the expensive search */ - buf = gst_adapter_peek (adapter, 4); - if (memcmp (buf, "wvpk", 4) == 0) - return TRUE; - - if (avail == 4) - return FALSE; - - /* search for the marker in the complete content of the adapter */ - buf = gst_adapter_peek (adapter, avail); - if (buf && (marker = gst_wavpack_parse_find_marker ((guint8 *) buf, avail))) { - gst_adapter_flush (adapter, marker - buf); - return TRUE; - } - - /* flush everything except the last 4 bytes. they could contain - * the start of a new marker */ - gst_adapter_flush (adapter, avail - 4); - - return FALSE; -} - -static GstFlowReturn -gst_wavpack_parse_chain (GstPad * pad, GstBuffer * buf) -{ - GstWavpackParse *wvparse = GST_WAVPACK_PARSE (GST_PAD_PARENT (pad)); - GstFlowReturn ret = GST_FLOW_OK; - WavpackHeader wph; - const guint8 *tmp_buf; - - if (!wvparse->adapter) { - wvparse->adapter = gst_adapter_new (); - } - - if (GST_BUFFER_IS_DISCONT (buf)) { - gst_adapter_clear (wvparse->adapter); - wvparse->discont = TRUE; - } - - gst_adapter_push (wvparse->adapter, buf); - - if (gst_adapter_available (wvparse->adapter) < sizeof (WavpackHeader)) - return ret; - - if (!gst_wavpack_parse_resync_adapter (wvparse->adapter)) - return ret; - - tmp_buf = gst_adapter_peek (wvparse->adapter, sizeof (WavpackHeader)); - gst_wavpack_read_header (&wph, (guint8 *) tmp_buf); - - while (gst_adapter_available (wvparse->adapter) >= wph.ckSize + 4 * 1 + 4) { - GstBuffer *outbuf = - gst_adapter_take_buffer (wvparse->adapter, wph.ckSize + 4 * 1 + 4); - - if (!outbuf) - return GST_FLOW_ERROR; - - if (wvparse->srcpad == NULL) { - if (!gst_wavpack_parse_create_src_pad (wvparse, outbuf, &wph)) { - GST_ERROR_OBJECT (wvparse, "Failed to create src pad"); - ret = GST_FLOW_ERROR; - break; - } - } - - ret = gst_wavpack_parse_push_buffer (wvparse, outbuf, &wph); - - if (ret != GST_FLOW_OK) - break; - - if (gst_adapter_available (wvparse->adapter) >= sizeof (WavpackHeader)) { - tmp_buf = gst_adapter_peek (wvparse->adapter, sizeof (WavpackHeader)); - - if (!gst_wavpack_parse_resync_adapter (wvparse->adapter)) - break; - - gst_wavpack_read_header (&wph, (guint8 *) tmp_buf); - } - } - - return ret; -} - -static GstStateChangeReturn -gst_wavpack_parse_change_state (GstElement * element, GstStateChange transition) -{ - GstWavpackParse *wvparse = GST_WAVPACK_PARSE (element); - GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS; - - switch (transition) { - case GST_STATE_CHANGE_READY_TO_PAUSED: - gst_segment_init (&wvparse->segment, GST_FORMAT_DEFAULT); - wvparse->segment.last_stop = 0; - default: - break; - } - - if (GST_ELEMENT_CLASS (parent_class)->change_state) - ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); - - switch (transition) { - case GST_STATE_CHANGE_PAUSED_TO_READY: - gst_wavpack_parse_reset (wvparse); - break; - default: - break; - } - - return ret; -} - -static gboolean -gst_wavpack_parse_sink_activate (GstPad * sinkpad) -{ - if (gst_pad_check_pull_range (sinkpad)) { - return gst_pad_activate_pull (sinkpad, TRUE); - } else { - return gst_pad_activate_push (sinkpad, TRUE); - } -} - -static gboolean -gst_wavpack_parse_sink_activate_pull (GstPad * sinkpad, gboolean active) -{ - gboolean result; - - if (active) { - result = gst_pad_start_task (sinkpad, - (GstTaskFunction) gst_wavpack_parse_loop, GST_PAD_PARENT (sinkpad)); - } else { - result = gst_pad_stop_task (sinkpad); - } - - return result; -} - -gboolean -gst_wavpack_parse_plugin_init (GstPlugin * plugin) -{ - if (!gst_element_register (plugin, "wavpackparse", - GST_RANK_PRIMARY, GST_TYPE_WAVPACK_PARSE)) { - return FALSE; - } - - GST_DEBUG_CATEGORY_INIT (gst_wavpack_parse_debug, "wavpack_parse", 0, - "Wavpack file parser"); - - return TRUE; -} diff --git a/ext/wavpack/gstwavpackparse.h b/ext/wavpack/gstwavpackparse.h deleted file mode 100644 index 60504a7ca6..0000000000 --- a/ext/wavpack/gstwavpackparse.h +++ /dev/null @@ -1,97 +0,0 @@ -/* GStreamer wavpack plugin - * (c) 2005 Arwed v. Merkatz - * - * gstwavpackparse.h: wavpack file 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. - */ - -#ifndef __GST_WAVPACK_PARSE_H__ -#define __GST_WAVPACK_PARSE_H__ - -#include -#include - -G_BEGIN_DECLS -#define GST_TYPE_WAVPACK_PARSE \ - (gst_wavpack_parse_get_type()) -#define GST_WAVPACK_PARSE(obj) \ - (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WAVPACK_PARSE,GstWavpackParse)) -#define GST_WAVPACK_PARSE_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_WAVPACK_PARSE,GstWavpackParseClass)) -#define GST_IS_WAVPACK_PARSE(obj) \ - (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WAVPACK_PARSE)) -#define GST_IS_WAVPACK_PARSE_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_WAVPACK_PARSE)) -typedef struct _GstWavpackParse GstWavpackParse; -typedef struct _GstWavpackParseClass GstWavpackParseClass; -typedef struct _GstWavpackParseIndexEntry GstWavpackParseIndexEntry; - -struct _GstWavpackParseIndexEntry -{ - gint64 byte_offset; /* byte offset of this chunk */ - gint64 sample_offset; /* first sample in this chunk */ - gint64 sample_offset_end; /* first sample in next chunk */ -}; - -struct _GstWavpackParse -{ - GstElement element; - - /*< private > */ - GstPad *sinkpad; - GstPad *srcpad; - - guint samplerate; - guint channels; - gint64 total_samples; - - gboolean need_newsegment; - gboolean discont; - - gint64 current_offset; /* byte offset on sink pad */ - gint64 upstream_length; /* length of file in bytes */ - - GstSegment segment; /* the currently configured segment, in - * samples/audio frames (DEFAULT format) */ - - GstBuffer *pending_buffer; - gint32 pending_offset; - guint32 next_block_index; - - GstAdapter *adapter; /* when operating chain-based, otherwise NULL */ - - /* List of GstWavpackParseIndexEntry structs, mapping known - * sample offsets to byte offsets. Is kept increasing without - * gaps (ie. append only and consecutive entries must always - * map to consecutive chunks in the file). */ - GSList *entries; - - /* Queued events (e.g. tag events we receive before we create the src pad) */ - GList *queued_events; /* STREAM_LOCK */ -}; - -struct _GstWavpackParseClass -{ - GstElementClass parent; -}; - -GType gst_wavpack_parse_get_type (void); - -gboolean gst_wavpack_parse_plugin_init (GstPlugin * plugin); - -G_END_DECLS -#endif /* __GST_WAVPACK_PARSE_H__ */ From e0c7004667e8d6c98f8f793c52aa0c79404c2df0 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Mon, 5 Mar 2012 12:17:39 +0100 Subject: [PATCH 51/55] wavpackenc: port to 0.11 --- ext/wavpack/gstwavpackcommon.c | 40 +++++-------- ext/wavpack/gstwavpackcommon.h | 4 +- ext/wavpack/gstwavpackenc.c | 100 +++++++++++++++++---------------- 3 files changed, 66 insertions(+), 78 deletions(-) diff --git a/ext/wavpack/gstwavpackcommon.c b/ext/wavpack/gstwavpackcommon.c index 252b64cc23..c38447079e 100644 --- a/ext/wavpack/gstwavpackcommon.c +++ b/ext/wavpack/gstwavpackcommon.c @@ -29,7 +29,6 @@ #include #include -#include GST_DEBUG_CATEGORY_EXTERN (wavpack_debug); #define GST_CAT_DEFAULT wavpack_debug @@ -147,7 +146,7 @@ static const struct 0x00001, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT}, { 0x00002, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}, { 0x00004, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER}, { - 0x00008, GST_AUDIO_CHANNEL_POSITION_LFE}, { + 0x00008, GST_AUDIO_CHANNEL_POSITION_LFE1}, { 0x00010, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT}, { 0x00020, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}, { 0x00040, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER}, { @@ -155,37 +154,25 @@ static const struct 0x00100, GST_AUDIO_CHANNEL_POSITION_REAR_CENTER}, { 0x00200, GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT}, { 0x00400, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT}, { - 0x00800, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_CENTER */ - { - 0x01000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_FRONT_LEFT */ - { - 0x02000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_FRONT_CENTER */ - { - 0x04000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_FRONT_RIGHT */ - { - 0x08000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_BACK_LEFT */ - { - 0x10000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_BACK_CENTER */ - { - 0x20000, GST_AUDIO_CHANNEL_POSITION_INVALID} /* TOP_BACK_RIGHT */ + 0x00800, GST_AUDIO_CHANNEL_POSITION_TOP_CENTER}, { + 0x01000, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_LEFT}, { + 0x02000, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_CENTER}, { + 0x04000, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_RIGHT}, { + 0x08000, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_LEFT}, { + 0x10000, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_CENTER}, { + 0x20000, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_RIGHT} }; #define MAX_CHANNEL_POSITIONS G_N_ELEMENTS (layout_mapping) gboolean -gst_wavpack_set_channel_layout (GstCaps * caps, gint layout) +gst_wavpack_get_channel_positions (gint num_channels, gint layout, + GstAudioChannelPosition * pos) { - GstAudioChannelPosition pos[MAX_CHANNEL_POSITIONS]; - GstStructure *s; - gint num_channels, i, p; - - s = gst_caps_get_structure (caps, 0); - if (!gst_structure_get_int (s, "channels", &num_channels)) - g_return_val_if_reached (FALSE); + gint i, p; if (num_channels == 1 && layout == 0x00004) { - pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_MONO; - gst_audio_set_channel_positions (s, pos); + pos[0] = GST_AUDIO_CHANNEL_POSITION_MONO; return TRUE; } @@ -214,7 +201,6 @@ gst_wavpack_set_channel_layout (GstCaps * caps, gint layout) return FALSE; } - gst_audio_set_channel_positions (s, pos); return TRUE; } @@ -242,7 +228,7 @@ gst_wavpack_get_channel_mask_from_positions (GstAudioChannelPosition * pos, gint channel_mask = 0; gint i, j; - if (nchannels == 1 && pos[0] == GST_AUDIO_CHANNEL_POSITION_FRONT_MONO) { + if (nchannels == 1 && pos[0] == GST_AUDIO_CHANNEL_POSITION_MONO) { channel_mask = 0x00000004; return channel_mask; } diff --git a/ext/wavpack/gstwavpackcommon.h b/ext/wavpack/gstwavpackcommon.h index 6a9e51609d..4f05683549 100644 --- a/ext/wavpack/gstwavpackcommon.h +++ b/ext/wavpack/gstwavpackcommon.h @@ -24,7 +24,7 @@ #define __GST_WAVPACK_COMMON_H__ #include -#include +#include #include typedef struct @@ -67,7 +67,7 @@ gboolean gst_wavpack_read_header (WavpackHeader * header, guint8 * buf); gboolean gst_wavpack_read_metadata (GstWavpackMetadata * meta, guint8 * header_data, guint8 ** p_data); gint gst_wavpack_get_default_channel_mask (gint nchannels); -gboolean gst_wavpack_set_channel_layout (GstCaps * caps, gint layout); +gboolean gst_wavpack_get_channel_positions (gint nchannels, gint layout, GstAudioChannelPosition *pos); GstAudioChannelPosition *gst_wavpack_get_default_channel_positions (gint nchannels); gint gst_wavpack_get_channel_mask_from_positions (GstAudioChannelPosition *pos, gint nchannels); gboolean gst_wavpack_set_channel_mapping (GstAudioChannelPosition *pos, gint nchannels, gint8 *channel_mapping); diff --git a/ext/wavpack/gstwavpackenc.c b/ext/wavpack/gstwavpackenc.c index 33c4254c34..a2f05351ca 100644 --- a/ext/wavpack/gstwavpackenc.c +++ b/ext/wavpack/gstwavpackenc.c @@ -31,7 +31,7 @@ * |[ * gst-launch audiotestsrc num-buffers=500 ! audioconvert ! wavpackenc ! filesink location=sinewave.wv * ]| This pipeline encodes audio from audiotestsrc into a Wavpack file. The audioconvert element is needed - * as the Wavpack encoder only accepts input with 32 bit width (and every depth between 1 and 32 bits). + * as the Wavpack encoder only accepts input with 32 bit width. * |[ * gst-launch cdda://1 ! audioconvert ! wavpackenc ! filesink location=track1.wv * ]| This pipeline encodes audio from an audio CD into a Wavpack file using @@ -90,19 +90,17 @@ GST_DEBUG_CATEGORY_STATIC (gst_wavpack_enc_debug); static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/x-raw-int, " - "width = (int) 32, " - "depth = (int) { 24, 32 }, " - "endianness = (int) BYTE_ORDER, " - "channels = (int) [ 1, 8 ], " - "rate = (int) [ 6000, 192000 ]," "signed = (boolean) TRUE") + GST_STATIC_CAPS ("audio/x-raw, " + "format = (string) " GST_AUDIO_NE (S32) ", " + "layout = (string) interleaved, " + "channels = (int) [ 1, 8 ], " "rate = (int) [ 6000, 192000 ]") ); static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-wavpack, " - "width = (int) [ 1, 32 ], " + "depth = (int) [ 1, 32 ], " "channels = (int) [ 1, 8 ], " "rate = (int) [ 6000, 192000 ], " "framed = (boolean) TRUE") ); @@ -202,13 +200,15 @@ gst_wavpack_enc_joint_stereo_mode_get_type (void) return qtype; } -GST_BOILERPLATE (GstWavpackEnc, gst_wavpack_enc, GstAudioEncoder, - GST_TYPE_AUDIO_ENCODER); +#define gst_wavpack_enc_parent_class parent_class +G_DEFINE_TYPE (GstWavpackEnc, gst_wavpack_enc, GST_TYPE_AUDIO_ENCODER); static void -gst_wavpack_enc_base_init (gpointer klass) +gst_wavpack_enc_class_init (GstWavpackEncClass * klass) { - GstElementClass *element_class = GST_ELEMENT_CLASS (klass); + GObjectClass *gobject_class = (GObjectClass *) klass; + GstElementClass *element_class = (GstElementClass *) (klass); + GstAudioEncoderClass *base_class = (GstAudioEncoderClass *) (klass); /* add pad templates */ gst_element_class_add_pad_template (element_class, @@ -223,15 +223,6 @@ gst_wavpack_enc_base_init (gpointer klass) "Codec/Encoder/Audio", "Encodes audio with the Wavpack lossless/lossy audio codec", "Sebastian Dröge "); -} - -static void -gst_wavpack_enc_class_init (GstWavpackEncClass * klass) -{ - GObjectClass *gobject_class = (GObjectClass *) klass; - GstAudioEncoderClass *base_class = (GstAudioEncoderClass *) (klass); - - parent_class = g_type_class_peek_parent (klass); /* set property handlers */ gobject_class->set_property = gst_wavpack_enc_set_property; @@ -327,7 +318,7 @@ gst_wavpack_enc_reset (GstWavpackEnc * enc) } static void -gst_wavpack_enc_init (GstWavpackEnc * enc, GstWavpackEncClass * gclass) +gst_wavpack_enc_init (GstWavpackEnc * enc) { GstAudioEncoder *benc = GST_AUDIO_ENCODER (enc); @@ -383,7 +374,9 @@ gst_wavpack_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info) { GstWavpackEnc *enc = GST_WAVPACK_ENC (benc); GstAudioChannelPosition *pos; + GstAudioChannelPosition opos[64] = { GST_AUDIO_CHANNEL_POSITION_INVALID, }; GstCaps *caps; + guint64 mask = 0; /* we may be configured again, but that change should have cleanup context */ g_assert (enc->wp_context == NULL); @@ -406,14 +399,17 @@ gst_wavpack_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info) gst_wavpack_set_channel_mapping (pos, enc->channels, enc->channel_mapping); + /* wavpack caps hold gst mask, not wavpack mask */ + gst_audio_channel_positions_to_mask (opos, enc->channels, &mask); + /* set fixed src pad caps now that we know what we will get */ caps = gst_caps_new_simple ("audio/x-wavpack", "channels", G_TYPE_INT, enc->channels, "rate", G_TYPE_INT, enc->samplerate, - "width", G_TYPE_INT, enc->depth, "framed", G_TYPE_BOOLEAN, TRUE, NULL); + "depth", G_TYPE_INT, enc->depth, "framed", G_TYPE_BOOLEAN, TRUE, NULL); - if (!gst_wavpack_set_channel_layout (caps, enc->channel_mask)) - GST_WARNING_OBJECT (enc, "setting channel layout failed"); + if (mask) + gst_caps_set_simple (caps, "channel-mask", GST_TYPE_BITMASK, mask, NULL); if (!gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc), caps)) goto setting_src_caps_failed; @@ -562,11 +558,11 @@ gst_wavpack_enc_push_block (void *id, void *data, int32_t count) pad = (wid->correction) ? enc->wvcsrcpad : GST_AUDIO_ENCODER_SRC_PAD (enc); flow = - (wid->correction) ? &enc->wvcsrcpad_last_return : &enc-> - srcpad_last_return; + (wid->correction) ? &enc-> + wvcsrcpad_last_return : &enc->srcpad_last_return; buffer = gst_buffer_new_and_alloc (count); - g_memmove (GST_BUFFER_DATA (buffer), block, count); + gst_buffer_fill (buffer, 0, data, count); if (count > sizeof (WavpackHeader) && memcmp (block, "wvpk", 4) == 0) { /* if it's a Wavpack block set buffer timestamp and duration, etc */ @@ -612,9 +608,12 @@ gst_wavpack_enc_push_block (void *id, void *data, int32_t count) /* save header for later reference, so we can re-send it later on * EOS with fixed up values for total sample count etc. */ if (enc->first_block == NULL && !wid->correction) { - enc->first_block = - g_memdup (GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer)); - enc->first_block_size = GST_BUFFER_SIZE (buffer); + GstMapInfo map; + + gst_buffer_map (buffer, &map, GST_MAP_READ); + enc->first_block = g_memdup (map.data, map.size); + enc->first_block_size = map.size; + gst_buffer_unmap (buffer, &map); } } } @@ -634,11 +633,11 @@ gst_wavpack_enc_push_block (void *id, void *data, int32_t count) if (wid->correction || wid->passthrough) { /* push the buffer and forward errors */ GST_DEBUG_OBJECT (enc, "pushing buffer with %d bytes", - GST_BUFFER_SIZE (buffer)); + gst_buffer_get_size (buffer)); *flow = gst_pad_push (pad, buffer); } else { GST_DEBUG_OBJECT (enc, "handing frame of %d bytes", - GST_BUFFER_SIZE (buffer)); + gst_buffer_get_size (buffer)); *flow = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc), buffer, samples); } @@ -676,6 +675,7 @@ gst_wavpack_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf) GstWavpackEnc *enc = GST_WAVPACK_ENC (benc); uint32_t sample_count; GstFlowReturn ret; + GstMapInfo map; /* base class ensures configuration */ g_return_val_if_fail (enc->depth != 0, GST_FLOW_NOT_NEGOTIATED); @@ -688,7 +688,7 @@ gst_wavpack_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf) if (G_UNLIKELY (!buf)) return gst_wavpack_enc_drain (enc); - sample_count = GST_BUFFER_SIZE (buf) / 4; + sample_count = gst_buffer_get_size (buf) / 4; GST_DEBUG_OBJECT (enc, "got %u raw samples", sample_count); /* check if we already have a valid WavpackContext, otherwise make one */ @@ -716,27 +716,28 @@ gst_wavpack_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf) if (enc->need_channel_remap) { buf = gst_buffer_make_writable (buf); - gst_wavpack_enc_fix_channel_order (enc, (gint32 *) GST_BUFFER_DATA (buf), - sample_count); + gst_buffer_map (buf, &map, GST_MAP_WRITE); + gst_wavpack_enc_fix_channel_order (enc, (gint32 *) map.data, sample_count); + gst_buffer_unmap (buf, &map); } + gst_buffer_map (buf, &map, GST_MAP_READ); + /* if we want to append the MD5 sum to the stream update it here * with the current raw samples */ if (enc->md5) { - g_checksum_update (enc->md5_context, GST_BUFFER_DATA (buf), - GST_BUFFER_SIZE (buf)); + g_checksum_update (enc->md5_context, map.data, map.size); } /* encode and handle return values from encoding */ - if (WavpackPackSamples (enc->wp_context, (int32_t *) GST_BUFFER_DATA (buf), + if (WavpackPackSamples (enc->wp_context, (int32_t *) map.data, sample_count / enc->channels)) { GST_DEBUG_OBJECT (enc, "encoding samples successful"); + gst_buffer_unmap (buf, &map); ret = GST_FLOW_OK; } else { - if ((enc->srcpad_last_return == GST_FLOW_RESEND) || - (enc->wvcsrcpad_last_return == GST_FLOW_RESEND)) { - ret = GST_FLOW_RESEND; - } else if ((enc->srcpad_last_return == GST_FLOW_OK) || + gst_buffer_unmap (buf, &map); + if ((enc->srcpad_last_return == GST_FLOW_OK) || (enc->wvcsrcpad_last_return == GST_FLOW_OK)) { ret = GST_FLOW_OK; } else if ((enc->srcpad_last_return == GST_FLOW_NOT_LINKED) && @@ -780,8 +781,7 @@ context_failed: static void gst_wavpack_enc_rewrite_first_block (GstWavpackEnc * enc) { - GstEvent *event = gst_event_new_new_segment (TRUE, 1.0, GST_FORMAT_BYTES, - 0, GST_BUFFER_OFFSET_NONE, 0); + GstSegment segment; gboolean ret; g_return_if_fail (enc); @@ -791,7 +791,9 @@ gst_wavpack_enc_rewrite_first_block (GstWavpackEnc * enc) WavpackUpdateNumSamples (enc->wp_context, enc->first_block); /* try to seek to the beginning of the output */ - ret = gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (enc), event); + gst_segment_init (&segment, GST_FORMAT_BYTES); + ret = gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (enc), + gst_event_new_segment (&segment)); if (ret) { /* try to rewrite the first block */ GST_DEBUG_OBJECT (enc, "rewriting first block ..."); @@ -861,7 +863,7 @@ gst_wavpack_enc_sink_event (GstAudioEncoder * benc, GstEvent * event) GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { - case GST_EVENT_NEWSEGMENT: + case GST_EVENT_SEGMENT: if (enc->wp_context) { GST_WARNING_OBJECT (enc, "got NEWSEGMENT after encoding " "already started"); @@ -876,7 +878,7 @@ gst_wavpack_enc_sink_event (GstAudioEncoder * benc, GstEvent * event) } /* baseclass handles rest */ - return FALSE; + return GST_AUDIO_ENCODER_CLASS (parent_class)->event (benc, event); } static void @@ -980,7 +982,7 @@ gst_wavpack_enc_plugin_init (GstPlugin * plugin) GST_RANK_NONE, GST_TYPE_WAVPACK_ENC)) return FALSE; - GST_DEBUG_CATEGORY_INIT (gst_wavpack_enc_debug, "wavpack_enc", 0, + GST_DEBUG_CATEGORY_INIT (gst_wavpack_enc_debug, "wavpackenc", 0, "Wavpack encoder"); return TRUE; From 2f7d8f55a4d56c7ec9e40204094a38c3e28c743b Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Mon, 5 Mar 2012 13:36:39 +0100 Subject: [PATCH 52/55] wavpackdec: port to 0.11 --- configure.ac | 1 - ext/wavpack/gstwavpackdec.c | 219 +++++++++++++++++------------------- ext/wavpack/gstwavpackdec.h | 3 + 3 files changed, 109 insertions(+), 114 deletions(-) diff --git a/configure.ac b/configure.ac index 25e97b60d3..9669204e9c 100644 --- a/configure.ac +++ b/configure.ac @@ -316,7 +316,6 @@ GST_PLUGINS_NONPORTED="deinterlace interleave flx \ videobox \ cairo cairo_gobject dv1394 \ oss oss4 \ - wavpack \ osx_video osx_audio " AC_SUBST(GST_PLUGINS_NONPORTED) diff --git a/ext/wavpack/gstwavpackdec.c b/ext/wavpack/gstwavpackdec.c index 8e24a31824..5673b7d604 100644 --- a/ext/wavpack/gstwavpackdec.c +++ b/ext/wavpack/gstwavpackdec.c @@ -44,7 +44,6 @@ #include #include -#include #include #include @@ -62,7 +61,7 @@ static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-wavpack, " - "width = (int) [ 1, 32 ], " + "depth = (int) [ 1, 32 ], " "channels = (int) [ 1, 8 ], " "rate = (int) [ 6000, 192000 ], " "framed = (boolean) true") ); @@ -70,21 +69,20 @@ static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/x-raw-int, " - "width = (int) 8, depth = (int) 8, " + GST_STATIC_CAPS ("audio/x-raw, " + "format = (string) S8, " + "layout = (string) interleaved, " "channels = (int) [ 1, 8 ], " - "rate = (int) [ 6000, 192000 ], " - "endianness = (int) BYTE_ORDER, " "signed = (boolean) true; " - "audio/x-raw-int, " - "width = (int) 16, depth = (int) 16, " + "rate = (int) [ 6000, 192000 ]; " + "audio/x-raw, " + "format = (string) " GST_AUDIO_NE (S16) ", " + "layout = (string) interleaved, " "channels = (int) [ 1, 8 ], " - "rate = (int) [ 6000, 192000 ], " - "endianness = (int) BYTE_ORDER, " "signed = (boolean) true; " - "audio/x-raw-int, " - "width = (int) 32, depth = (int) 32, " - "channels = (int) [ 1, 8 ], " - "rate = (int) [ 6000, 192000 ], " - "endianness = (int) BYTE_ORDER, " "signed = (boolean) true; ") + "rate = (int) [ 6000, 192000 ]; " + "audio/x-raw, " + "format = (string) " GST_AUDIO_NE (S32) ", " + "layout = (string) interleaved, " + "channels = (int) [ 1, 8 ], " "rate = (int) [ 6000, 192000 ]") ); static gboolean gst_wavpack_dec_start (GstAudioDecoder * dec); @@ -97,13 +95,15 @@ static GstFlowReturn gst_wavpack_dec_handle_frame (GstAudioDecoder * dec, static void gst_wavpack_dec_finalize (GObject * object); static void gst_wavpack_dec_post_tags (GstWavpackDec * dec); -GST_BOILERPLATE (GstWavpackDec, gst_wavpack_dec, GstAudioDecoder, - GST_TYPE_AUDIO_DECODER); +#define gst_wavpack_dec_parent_class parent_class +G_DEFINE_TYPE (GstWavpackDec, gst_wavpack_dec, GST_TYPE_AUDIO_DECODER); static void -gst_wavpack_dec_base_init (gpointer klass) +gst_wavpack_dec_class_init (GstWavpackDecClass * klass) { - GstElementClass *element_class = GST_ELEMENT_CLASS (klass); + GObjectClass *gobject_class = (GObjectClass *) klass; + GstElementClass *element_class = (GstElementClass *) (klass); + GstAudioDecoderClass *base_class = (GstAudioDecoderClass *) (klass); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&src_factory)); @@ -114,13 +114,6 @@ gst_wavpack_dec_base_init (gpointer klass) "Decodes Wavpack audio data", "Arwed v. Merkatz , " "Sebastian Dröge "); -} - -static void -gst_wavpack_dec_class_init (GstWavpackDecClass * klass) -{ - GObjectClass *gobject_class = (GObjectClass *) klass; - GstAudioDecoderClass *base_class = (GstAudioDecoderClass *) (klass); gobject_class->finalize = gst_wavpack_dec_finalize; @@ -143,7 +136,7 @@ gst_wavpack_dec_reset (GstWavpackDec * dec) } static void -gst_wavpack_dec_init (GstWavpackDec * dec, GstWavpackDecClass * gklass) +gst_wavpack_dec_init (GstWavpackDec * dec) { dec->context = NULL; dec->stream_reader = gst_wavpack_stream_reader_new (); @@ -197,69 +190,51 @@ gst_wavpack_dec_stop (GstAudioDecoder * dec) static void gst_wavpack_dec_negotiate (GstWavpackDec * dec) { - GstCaps *caps; + GstAudioInfo info; + GstAudioFormat fmt; + GstAudioChannelPosition pos[64] = { GST_AUDIO_CHANNEL_POSITION_INVALID, }; /* arrange for 1, 2 or 4-byte width == depth output */ - if (dec->depth == 24) - dec->width = 32; - else - dec->width = dec->depth; + dec->width = dec->depth; + switch (dec->depth) { + case 8: + fmt = GST_AUDIO_FORMAT_S8; + break; + case 16: + fmt = _GST_AUDIO_FORMAT_NE (S16); + break; + case 24: + case 32: + fmt = _GST_AUDIO_FORMAT_NE (S32); + dec->width = 32; + break; + default: + g_assert_not_reached (); + break; + } - caps = gst_caps_new_simple ("audio/x-raw-int", - "rate", G_TYPE_INT, dec->sample_rate, - "channels", G_TYPE_INT, dec->channels, - "depth", G_TYPE_INT, dec->width, - "width", G_TYPE_INT, dec->width, - "endianness", G_TYPE_INT, G_BYTE_ORDER, - "signed", G_TYPE_BOOLEAN, TRUE, NULL); + g_assert (dec->channel_mask != 0); - /* Only set the channel layout for more than two channels - * otherwise things break unfortunately */ - if (dec->channel_mask != 0 && dec->channels > 2) - if (!gst_wavpack_set_channel_layout (caps, dec->channel_mask)) - GST_WARNING_OBJECT (dec, "Failed to set channel layout"); + if (!gst_wavpack_get_channel_positions (dec->channels, + dec->channel_mask, pos)) + GST_WARNING_OBJECT (dec, "Failed to set channel layout"); - GST_DEBUG_OBJECT (dec, "setting caps %" GST_PTR_FORMAT, caps); + gst_audio_info_init (&info); + gst_audio_info_set_format (&info, fmt, dec->sample_rate, dec->channels, pos); + + gst_audio_channel_positions_to_valid_order (info.position, info.channels); + gst_audio_get_channel_reorder_map (info.channels, + info.position, pos, dec->channel_reorder_map); /* should always succeed */ - gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps); - gst_caps_unref (caps); + gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (dec), &info); } static gboolean gst_wavpack_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps) { - GstWavpackDec *dec = GST_WAVPACK_DEC (bdec); - GstStructure *structure = gst_caps_get_structure (caps, 0); - - /* Check if we can set the caps here already */ - if (gst_structure_get_int (structure, "channels", &dec->channels) && - gst_structure_get_int (structure, "rate", &dec->sample_rate) && - gst_structure_get_int (structure, "width", &dec->depth)) { - GstAudioChannelPosition *pos; - - /* If we already have the channel layout set from upstream - * take this */ - if (gst_structure_has_field (structure, "channel-positions")) { - pos = gst_audio_get_channel_positions (structure); - if (pos != NULL && dec->channels > 2) { - GstStructure *new_str = gst_caps_get_structure (caps, 0); - - gst_audio_set_channel_positions (new_str, pos); - dec->channel_mask = - gst_wavpack_get_channel_mask_from_positions (pos, dec->channels); - } - - if (pos != NULL) - g_free (pos); - } - - gst_wavpack_dec_negotiate (dec); - - /* send GST_TAG_AUDIO_CODEC and GST_TAG_BITRATE tags before something - * is decoded or after the format has changed */ - gst_wavpack_dec_post_tags (dec); - } + /* pretty much nothing to do here, + * we'll parse it all from the stream and setup then */ return TRUE; } @@ -272,13 +247,13 @@ gst_wavpack_dec_post_tags (GstWavpackDec * dec) gint64 duration, size; /* try to estimate the average bitrate */ - if (gst_pad_query_peer_duration (GST_AUDIO_DECODER_SINK_PAD (dec), - &format_bytes, &size) && - gst_pad_query_peer_duration (GST_AUDIO_DECODER_SINK_PAD (dec), - &format_time, &duration) && size > 0 && duration > 0) { + if (gst_pad_peer_query_duration (GST_AUDIO_DECODER_SINK_PAD (dec), + format_bytes, &size) && + gst_pad_peer_query_duration (GST_AUDIO_DECODER_SINK_PAD (dec), + format_time, &duration) && size > 0 && duration > 0) { guint64 bitrate; - list = gst_tag_list_new (); + list = gst_tag_list_new_empty (); bitrate = gst_util_uint64_scale (size, 8 * GST_SECOND, duration); gst_tag_list_add (list, GST_TAG_MERGE_REPLACE, GST_TAG_BITRATE, @@ -298,29 +273,32 @@ gst_wavpack_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf) WavpackHeader wph; int32_t decoded, unpacked_size; gboolean format_changed; - gint width, depth, i, max; + gint width, depth, i, j, max; gint32 *dec_data = NULL; guint8 *out_data; + GstMapInfo map, omap; dec = GST_WAVPACK_DEC (bdec); g_return_val_if_fail (buf != NULL, GST_FLOW_ERROR); + gst_buffer_map (buf, &map, GST_MAP_READ); + /* check input, we only accept framed input with complete chunks */ - if (GST_BUFFER_SIZE (buf) < sizeof (WavpackHeader)) + if (map.size < sizeof (WavpackHeader)) goto input_not_framed; - if (!gst_wavpack_read_header (&wph, GST_BUFFER_DATA (buf))) + if (!gst_wavpack_read_header (&wph, map.data)) goto invalid_header; - if (GST_BUFFER_SIZE (buf) < wph.ckSize + 4 * 1 + 4) + if (map.size < wph.ckSize + 4 * 1 + 4) goto input_not_framed; if (!(wph.flags & INITIAL_BLOCK)) goto input_not_framed; - dec->wv_id.buffer = GST_BUFFER_DATA (buf); - dec->wv_id.length = GST_BUFFER_SIZE (buf); + dec->wv_id.buffer = map.data; + dec->wv_id.length = map.size; dec->wv_id.position = 0; /* create a new wavpack context if there is none yet but if there @@ -351,7 +329,8 @@ gst_wavpack_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf) (dec->channel_mask != WavpackGetChannelMask (dec->context)); #endif - if (!GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)) || format_changed) { + if (!gst_pad_has_current_caps (GST_AUDIO_DECODER_SRC_PAD (dec)) || + format_changed) { gint channel_mask; dec->sample_rate = WavpackGetSampleRate (dec->context); @@ -376,58 +355,72 @@ gst_wavpack_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf) } /* alloc output buffer */ - unpacked_size = (dec->width / 8) * wph.block_samples * dec->channels; - ret = gst_pad_alloc_buffer (GST_AUDIO_DECODER_SRC_PAD (dec), - GST_BUFFER_OFFSET (buf), unpacked_size, - GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)), &outbuf); - - if (ret != GST_FLOW_OK) - goto out; - dec_data = g_malloc (4 * wph.block_samples * dec->channels); - out_data = GST_BUFFER_DATA (outbuf); /* decode */ decoded = WavpackUnpackSamples (dec->context, dec_data, wph.block_samples); if (decoded != wph.block_samples) goto decode_error; + unpacked_size = (dec->width / 8) * wph.block_samples * dec->channels; + outbuf = gst_buffer_new_and_alloc (unpacked_size); + + /* legacy; pass along offset, whatever that might entail */ + GST_BUFFER_OFFSET (outbuf) = GST_BUFFER_OFFSET (buf); + + gst_buffer_map (outbuf, &omap, GST_MAP_WRITE); + out_data = omap.data; + width = dec->width; depth = dec->depth; max = dec->channels * wph.block_samples; if (width == 8) { gint8 *outbuffer = (gint8 *) out_data; + gint *reorder_map = dec->channel_reorder_map; - for (i = 0; i < max; i++) { - *outbuffer++ = (gint8) (dec_data[i]); + for (i = 0; i < max; i += dec->channels) { + for (j = 0; j < dec->channels; j++) + *outbuffer++ = (gint8) (dec_data[i + reorder_map[j]]); } } else if (width == 16) { gint16 *outbuffer = (gint16 *) out_data; + gint *reorder_map = dec->channel_reorder_map; - for (i = 0; i < max; i++) { - *outbuffer++ = (gint16) (dec_data[i]); + for (i = 0; i < max; i += dec->channels) { + for (j = 0; j < dec->channels; j++) + *outbuffer++ = (gint16) (dec_data[i + reorder_map[j]]); } } else if (dec->width == 32) { gint32 *outbuffer = (gint32 *) out_data; + gint *reorder_map = dec->channel_reorder_map; if (width != depth) { - for (i = 0; i < max; i++) { - *outbuffer++ = (gint32) (dec_data[i] << (width - depth)); + for (i = 0; i < max; i += dec->channels) { + for (j = 0; j < dec->channels; j++) + *outbuffer++ = + (gint32) (dec_data[i + reorder_map[j]] << (width - depth)); } } else { - for (i = 0; i < max; i++) { - *outbuffer++ = (gint32) dec_data[i]; + for (i = 0; i < max; i += dec->channels) { + for (j = 0; j < dec->channels; j++) + *outbuffer++ = (gint32) (dec_data[i + reorder_map[j]]); } } } else { g_assert_not_reached (); } + gst_buffer_unmap (outbuf, &omap); + gst_buffer_unmap (buf, &map); + buf = NULL; + g_free (dec_data); ret = gst_audio_decoder_finish_frame (bdec, outbuf, 1); out: + if (buf) + gst_buffer_unmap (buf, &map); if (G_UNLIKELY (ret != GST_FLOW_OK)) { GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (ret)); @@ -439,12 +432,14 @@ out: input_not_framed: { GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Expected framed input")); - return GST_FLOW_ERROR; + ret = GST_FLOW_ERROR; + goto out; } invalid_header: { GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Invalid wavpack header")); - return GST_FLOW_ERROR; + ret = GST_FLOW_ERROR; + goto out; } context_failed: { @@ -468,11 +463,9 @@ decode_error: GST_AUDIO_DECODER_ERROR (bdec, 1, STREAM, DECODE, (NULL), ("decoding error: %s", reason), ret); g_free (dec_data); - if (outbuf) - gst_buffer_unref (outbuf); if (ret == GST_FLOW_OK) gst_audio_decoder_finish_frame (bdec, NULL, 1); - return ret; + goto out; } } @@ -482,7 +475,7 @@ gst_wavpack_dec_plugin_init (GstPlugin * plugin) if (!gst_element_register (plugin, "wavpackdec", GST_RANK_PRIMARY, GST_TYPE_WAVPACK_DEC)) return FALSE; - GST_DEBUG_CATEGORY_INIT (gst_wavpack_dec_debug, "wavpack_dec", 0, + GST_DEBUG_CATEGORY_INIT (gst_wavpack_dec_debug, "wavpackdec", 0, "Wavpack decoder"); return TRUE; } diff --git a/ext/wavpack/gstwavpackdec.h b/ext/wavpack/gstwavpackdec.h index 68e8e33858..006428ceac 100644 --- a/ext/wavpack/gstwavpackdec.h +++ b/ext/wavpack/gstwavpackdec.h @@ -60,6 +60,9 @@ struct _GstWavpackDec gint width; gint channels; gint channel_mask; + + gint channel_reorder_map[64]; + }; struct _GstWavpackDecClass From e637cfb212b8bdfb7b698fb4e1ca7d60b5fdb508 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Mon, 5 Mar 2012 12:20:07 +0100 Subject: [PATCH 53/55] tests: port wavpack tests to 0.11 --- tests/check/Makefile.am | 2 +- tests/check/elements/wavpackdec.c | 39 +++++---- tests/check/elements/wavpackenc.c | 37 +++++---- tests/check/elements/wavpackparse.c | 123 ++++++++-------------------- tests/check/pipelines/wavpack.c | 8 +- 5 files changed, 86 insertions(+), 123 deletions(-) diff --git a/tests/check/Makefile.am b/tests/check/Makefile.am index ec2bbb4997..8bf8f2b254 100644 --- a/tests/check/Makefile.am +++ b/tests/check/Makefile.am @@ -83,7 +83,6 @@ endif if USE_WAVPACK check_wavpack = \ - elements/wavpackparse \ elements/wavpackdec \ elements/wavpackenc \ pipelines/wavpack @@ -150,6 +149,7 @@ check_PROGRAMS = \ elements/udpsrc \ elements/videocrop \ elements/videofilter \ + elements/wavpackparse \ elements/y4menc \ pipelines/simple-launch-lines \ pipelines/effectv \ diff --git a/tests/check/elements/wavpackdec.c b/tests/check/elements/wavpackdec.c index b1498e1667..01ae67d202 100644 --- a/tests/check/elements/wavpackdec.c +++ b/tests/check/elements/wavpackdec.c @@ -29,6 +29,12 @@ * get_peer, and then remove references in every test function */ static GstPad *mysrcpad, *mysinkpad; +#if G_BYTE_ORDER == G_BIG_ENDIAN +#define AUDIO_FORMAT "S16BE" +#else +#define AUDIO_FORMAT "S16LE" +#endif + guint8 test_frame[] = { 0x77, 0x76, 0x70, 0x6B, /* "wvpk" */ 0x2E, 0x00, 0x00, 0x00, /* ckSize */ @@ -51,13 +57,12 @@ guint8 test_frame[] = { static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/x-raw-int, " - "width = (int) 16, " - "depth = (int) 16, " - "channels = (int) 1, " - "rate = (int) 44100, " - "endianness = (int) BYTE_ORDER, " "signed = (boolean) true") + GST_STATIC_CAPS ("audio/x-raw, " + "format = (string) " AUDIO_FORMAT ", " + "layout = (string) interleaved, " + "channels = (int) 1, " "rate = (int) 44100") ); + static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, @@ -73,8 +78,8 @@ setup_wavpackdec (void) GST_DEBUG ("setup_wavpackdec"); wavpackdec = gst_check_setup_element ("wavpackdec"); - mysrcpad = gst_check_setup_src_pad (wavpackdec, &srctemplate, NULL); - mysinkpad = gst_check_setup_sink_pad (wavpackdec, &sinktemplate, NULL); + mysrcpad = gst_check_setup_src_pad (wavpackdec, &srctemplate); + mysinkpad = gst_check_setup_sink_pad (wavpackdec, &sinktemplate); gst_pad_set_active (mysrcpad, TRUE); gst_pad_set_active (mysinkpad, TRUE); @@ -100,6 +105,7 @@ GST_START_TEST (test_decode_frame) GstBuffer *inbuffer, *outbuffer; GstBus *bus; int i; + GstMapInfo map; wavpackdec = setup_wavpackdec (); @@ -109,7 +115,7 @@ GST_START_TEST (test_decode_frame) bus = gst_bus_new (); inbuffer = gst_buffer_new_and_alloc (sizeof (test_frame)); - memcpy (GST_BUFFER_DATA (inbuffer), test_frame, sizeof (test_frame)); + gst_buffer_fill (inbuffer, 0, test_frame, sizeof (test_frame)); ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1); GST_BUFFER_TIMESTAMP (inbuffer) = 0; @@ -121,12 +127,17 @@ GST_START_TEST (test_decode_frame) outbuffer = GST_BUFFER (buffers->data); fail_if (outbuffer == NULL); + + gst_buffer_map (outbuffer, &map, GST_MAP_READ); + /* uncompressed data should be 102400 bytes */ - fail_unless_equals_int (GST_BUFFER_SIZE (outbuffer), 51200); + fail_unless_equals_int (map.size, 51200); /* and all bytes must be 0, i.e. silence */ for (i = 0; i < 51200; i++) - fail_unless_equals_int (GST_BUFFER_DATA (outbuffer)[i], 0); + fail_unless_equals_int (map.data[i], 0); + + gst_buffer_unmap (outbuffer, &map); ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1); gst_buffer_unref (outbuffer); @@ -158,9 +169,9 @@ GST_START_TEST (test_decode_frame_with_broken_header) bus = gst_bus_new (); inbuffer = gst_buffer_new_and_alloc (sizeof (test_frame)); - memcpy (GST_BUFFER_DATA (inbuffer), test_frame, sizeof (test_frame)); + gst_buffer_fill (inbuffer, 0, test_frame, sizeof (test_frame)); /* break header */ - GST_BUFFER_DATA (inbuffer)[2] = 'e'; + gst_buffer_memset (inbuffer, 2, 'e', 1); ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1); GST_BUFFER_TIMESTAMP (inbuffer) = 0; @@ -195,7 +206,7 @@ GST_START_TEST (test_decode_frame_with_incomplete_frame) bus = gst_bus_new (); inbuffer = gst_buffer_new_and_alloc (sizeof (test_frame) - 2); - memcpy (GST_BUFFER_DATA (inbuffer), test_frame, sizeof (test_frame) - 2); + gst_buffer_fill (inbuffer, 0, test_frame, sizeof (test_frame) - 2); ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1); GST_BUFFER_TIMESTAMP (inbuffer) = 0; diff --git a/tests/check/elements/wavpackenc.c b/tests/check/elements/wavpackenc.c index af852aaa26..72f6cd11c7 100644 --- a/tests/check/elements/wavpackenc.c +++ b/tests/check/elements/wavpackenc.c @@ -30,13 +30,17 @@ static GstPad *mysrcpad, *mysinkpad; static GstBus *bus; -#define RAW_CAPS_STRING "audio/x-raw-int, " \ - "width = (int) 32, " \ - "depth = (int) 32, " \ +#if G_BYTE_ORDER == G_BIG_ENDIAN +#define AUDIO_FORMAT "S32BE" +#else +#define AUDIO_FORMAT "S32LE" +#endif + +#define RAW_CAPS_STRING "audio/x-raw, " \ + "format = (string) " AUDIO_FORMAT ", " \ + "layout = (string) interleaved, " \ "channels = (int) 1, " \ - "rate = (int) 44100, " \ - "endianness = (int) BYTE_ORDER, " \ - "signed = (boolean) true" + "rate = (int) 44100" #define WAVPACK_CAPS_STRING "audio/x-wavpack, " \ "width = (int) 32, " \ @@ -55,12 +59,10 @@ static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink", static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/x-raw-int, " - "width = (int) 32, " - "depth = (int) 32, " - "channels = (int) 1, " - "rate = (int) 44100, " - "endianness = (int) BYTE_ORDER, " "signed = (boolean) true")); + GST_STATIC_CAPS ("audio/x-raw, " + "format = (string) " AUDIO_FORMAT ", " + "layout = (string) interleaved, " + "channels = (int) 1, " "rate = (int) 44100")); static GstElement * setup_wavpackenc (void) @@ -69,8 +71,8 @@ setup_wavpackenc (void) GST_DEBUG ("setup_wavpackenc"); wavpackenc = gst_check_setup_element ("wavpackenc"); - mysrcpad = gst_check_setup_src_pad (wavpackenc, &srctemplate, NULL); - mysinkpad = gst_check_setup_sink_pad (wavpackenc, &sinktemplate, NULL); + mysrcpad = gst_check_setup_src_pad (wavpackenc, &srctemplate); + mysinkpad = gst_check_setup_sink_pad (wavpackenc, &sinktemplate); gst_pad_set_active (mysrcpad, TRUE); gst_pad_set_active (mysinkpad, TRUE); @@ -111,10 +113,9 @@ GST_START_TEST (test_encode_silence) wavpackenc = setup_wavpackenc (); inbuffer = gst_buffer_new_and_alloc (1000); - for (i = 0; i < 1000; i++) - GST_BUFFER_DATA (inbuffer)[i] = 0; + gst_buffer_memset (inbuffer, 0, 0, 1000); caps = gst_caps_from_string (RAW_CAPS_STRING); - gst_buffer_set_caps (inbuffer, caps); + fail_unless (gst_pad_set_caps (mysrcpad, caps)); gst_caps_unref (caps); GST_BUFFER_TIMESTAMP (inbuffer) = 0; ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1); @@ -133,7 +134,7 @@ GST_START_TEST (test_encode_silence) fail_unless_equals_int (GST_BUFFER_TIMESTAMP (outbuffer), 0); fail_unless_equals_int (GST_BUFFER_DURATION (outbuffer), 5668934); - fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), "wvpk", 4) == 0, + fail_unless (gst_buffer_memcmp (outbuffer, 0, "wvpk", 4) == 0, "Failed to encode to valid Wavpack frames"); /* free all buffers */ diff --git a/tests/check/elements/wavpackparse.c b/tests/check/elements/wavpackparse.c index 5083903204..ca69670998 100644 --- a/tests/check/elements/wavpackparse.c +++ b/tests/check/elements/wavpackparse.c @@ -28,8 +28,6 @@ * src and sink pads we create; otherwise we always have to do get_pad, * get_peer, and then remove references in every test function */ static GstPad *mysrcpad, *mysinkpad; -static GstBus *bus; -static GstElement *wavpackparse; /* Wavpack file with 2 frames of silence */ guint8 test_file[] = { @@ -69,99 +67,53 @@ static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-wavpack")); -static void -wavpackparse_found_pad (GstElement * src, GstPad * pad, gpointer data) -{ - GstPad *srcpad; - - mysinkpad = gst_pad_new_from_static_template (&sinktemplate, "sink"); - fail_if (mysinkpad == NULL, "Couldn't create sinkpad"); - srcpad = gst_element_get_static_pad (wavpackparse, "src"); - fail_if (srcpad == NULL, "Failed to get srcpad from wavpackparse"); - gst_pad_set_chain_function (mysinkpad, gst_check_chain_func); - fail_unless (gst_pad_link (srcpad, mysinkpad) == GST_PAD_LINK_OK, - "Failed to link pads"); - gst_pad_set_active (mysinkpad, TRUE); - gst_object_unref (srcpad); -} - -static void +static GstElement * setup_wavpackparse (void) { - GstPad *sinkpad; + GstElement *wavpackparse; GST_DEBUG ("setup_wavpackparse"); - wavpackparse = gst_element_factory_make ("wavpackparse", "wavpackparse"); - fail_if (wavpackparse == NULL, "Could not create wavpackparse"); + wavpackparse = gst_check_setup_element ("wavpackparse"); + mysrcpad = gst_check_setup_src_pad (wavpackparse, &srctemplate); + mysinkpad = gst_check_setup_sink_pad (wavpackparse, &sinktemplate); + gst_pad_set_active (mysrcpad, TRUE); + gst_pad_set_active (mysinkpad, TRUE); - mysrcpad = gst_pad_new_from_static_template (&srctemplate, "src"); - fail_if (mysrcpad == NULL, "Could not create srcpad"); - - sinkpad = gst_element_get_static_pad (wavpackparse, "sink"); - fail_if (sinkpad == NULL, "Failed to get sinkpad from wavpackparse"); - fail_unless (gst_pad_link (mysrcpad, sinkpad) == GST_PAD_LINK_OK, - "Failed to link pads"); - gst_object_unref (sinkpad); - - g_signal_connect (wavpackparse, "pad-added", - G_CALLBACK (wavpackparse_found_pad), NULL); - - bus = gst_bus_new (); - gst_element_set_bus (wavpackparse, bus); - - fail_unless (gst_element_set_state (wavpackparse, - GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, - "could not set to playing"); + return wavpackparse; } static void -cleanup_wavpackparse (void) +cleanup_wavpackparse (GstElement * wavpackparse) { - GstPad *sinkpad, *srcpad; - GST_DEBUG ("cleanup_wavpackparse"); + gst_element_set_state (wavpackparse, GST_STATE_NULL); - gst_bus_set_flushing (bus, TRUE); - gst_element_set_bus (wavpackparse, NULL); - gst_object_unref (GST_OBJECT (bus)); - - sinkpad = gst_element_get_static_pad (wavpackparse, "sink"); - fail_if (sinkpad == NULL, "Failed to get sinkpad from wavpackparse"); - fail_unless (gst_pad_unlink (mysrcpad, sinkpad), "Failed to unlink pads"); - gst_pad_set_caps (mysrcpad, NULL); - gst_object_unref (sinkpad); - gst_object_unref (mysrcpad); - - srcpad = gst_element_get_static_pad (wavpackparse, "src"); - fail_if (srcpad == NULL, "Failed to get srcpad from wavpackparse"); - fail_unless (gst_pad_unlink (srcpad, mysinkpad), "Failed to unlink pads"); - gst_pad_set_caps (mysinkpad, NULL); - gst_object_unref (srcpad); - gst_object_unref (mysinkpad); - - fail_unless (gst_element_set_state (wavpackparse, GST_STATE_NULL) == - GST_STATE_CHANGE_SUCCESS, "could not set to null"); - - gst_object_unref (wavpackparse); + gst_pad_set_active (mysrcpad, FALSE); + gst_pad_set_active (mysinkpad, FALSE); + gst_check_teardown_src_pad (wavpackparse); + gst_check_teardown_sink_pad (wavpackparse); + gst_check_teardown_element (wavpackparse); } GST_START_TEST (test_parsing_valid_frames) { + GstElement *wavpackparse; GstBuffer *inbuffer, *outbuffer; int i, num_buffers; - GstFormat format = GST_FORMAT_DEFAULT; + GstFormat format = GST_FORMAT_TIME; gint64 pos; - setup_wavpackparse (); + wavpackparse = setup_wavpackparse (); + fail_unless (gst_element_set_state (wavpackparse, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, + "could not set to playing"); inbuffer = gst_buffer_new_and_alloc (sizeof (test_file)); - memcpy (GST_BUFFER_DATA (inbuffer), test_file, sizeof (test_file)); - gst_buffer_ref (inbuffer); + gst_buffer_fill (inbuffer, 0, test_file, sizeof (test_file)); /* should decode the buffer without problems */ fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK); - gst_buffer_unref (inbuffer); num_buffers = g_list_length (buffers); /* should get 2 buffers, each one complete wavpack frame */ @@ -171,21 +123,17 @@ GST_START_TEST (test_parsing_valid_frames) outbuffer = GST_BUFFER (buffers->data); fail_if (outbuffer == NULL); - fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), "wvpk", 4) == 0, + fail_unless (gst_buffer_memcmp (outbuffer, 0, "wvpk", 4) == 0, "Buffer contains no Wavpack frame"); fail_unless_equals_int (GST_BUFFER_DURATION (outbuffer), 580498866); switch (i) { case 0:{ fail_unless_equals_int (GST_BUFFER_TIMESTAMP (outbuffer), 0); - fail_unless_equals_int (GST_BUFFER_OFFSET (outbuffer), 0); - fail_unless_equals_int (GST_BUFFER_OFFSET_END (outbuffer), 25600); break; } case 1:{ fail_unless_equals_int (GST_BUFFER_TIMESTAMP (outbuffer), 580498866); - fail_unless_equals_int (GST_BUFFER_OFFSET (outbuffer), 25600); - fail_unless_equals_int (GST_BUFFER_OFFSET_END (outbuffer), 51200); break; } } @@ -196,35 +144,37 @@ GST_START_TEST (test_parsing_valid_frames) outbuffer = NULL; } - fail_unless (gst_element_query_position (wavpackparse, &format, &pos), + fail_unless (gst_element_query_position (wavpackparse, format, &pos), "Position query failed"); - fail_unless_equals_int (pos, 51200); - fail_unless (gst_element_query_duration (wavpackparse, &format, NULL), + fail_unless_equals_int64 (pos, 580498866 * 2); + fail_unless (gst_element_query_duration (wavpackparse, format, NULL), "Duration query failed"); g_list_free (buffers); buffers = NULL; - cleanup_wavpackparse (); + cleanup_wavpackparse (wavpackparse); } GST_END_TEST; GST_START_TEST (test_parsing_invalid_first_header) { + GstElement *wavpackparse; GstBuffer *inbuffer, *outbuffer; int i, num_buffers; - setup_wavpackparse (); + wavpackparse = setup_wavpackparse (); + fail_unless (gst_element_set_state (wavpackparse, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, + "could not set to playing"); inbuffer = gst_buffer_new_and_alloc (sizeof (test_file)); - memcpy (GST_BUFFER_DATA (inbuffer), test_file, sizeof (test_file)); - GST_BUFFER_DATA (inbuffer)[0] = 'k'; - gst_buffer_ref (inbuffer); + gst_buffer_fill (inbuffer, 0, test_file, sizeof (test_file)); + gst_buffer_memset (inbuffer, 0, 'k', 1); /* should decode the buffer without problems */ fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK); - gst_buffer_unref (inbuffer); num_buffers = g_list_length (buffers); @@ -235,14 +185,13 @@ GST_START_TEST (test_parsing_invalid_first_header) outbuffer = GST_BUFFER (buffers->data); fail_if (outbuffer == NULL); - fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), "wvpk", 4) == 0, + fail_unless (gst_buffer_memcmp (outbuffer, 0, "wvpk", 4) == 0, "Buffer contains no Wavpack frame"); fail_unless_equals_int (GST_BUFFER_DURATION (outbuffer), 580498866); switch (i) { case 0:{ fail_unless_equals_int (GST_BUFFER_TIMESTAMP (outbuffer), 580498866); - fail_unless_equals_int (GST_BUFFER_OFFSET (outbuffer), 25600); break; } } @@ -256,7 +205,7 @@ GST_START_TEST (test_parsing_invalid_first_header) g_list_free (buffers); buffers = NULL; - cleanup_wavpackparse (); + cleanup_wavpackparse (wavpackparse); } GST_END_TEST; diff --git a/tests/check/pipelines/wavpack.c b/tests/check/pipelines/wavpack.c index 6a166b2795..7fcba8740a 100644 --- a/tests/check/pipelines/wavpack.c +++ b/tests/check/pipelines/wavpack.c @@ -96,6 +96,7 @@ GST_START_TEST (test_encode_decode) GMainLoop *loop; GstBuffer *in, *out; guint bus_watch = 0; + GstMapInfo map; srcadapter = gst_adapter_new (); fail_unless (srcadapter != NULL); @@ -170,9 +171,10 @@ GST_START_TEST (test_encode_decode) gst_adapter_available (sinkadapter)); fail_unless (out != NULL); - fail_unless_equals_int (GST_BUFFER_SIZE (in), GST_BUFFER_SIZE (out)); - fail_unless (memcmp (GST_BUFFER_DATA (in), GST_BUFFER_DATA (out), - GST_BUFFER_SIZE (in)) == 0); + fail_unless_equals_int (gst_buffer_get_size (in), gst_buffer_get_size (out)); + gst_buffer_map (out, &map, GST_MAP_READ); + fail_unless (gst_buffer_memcmp (in, 0, map.data, map.size) == 0); + gst_buffer_unmap (out, &map); gst_buffer_unref (in); gst_buffer_unref (out); From 4d8002e9c40d2d21fa344992e7264a781b3c864c Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Mon, 5 Mar 2012 13:31:44 +0100 Subject: [PATCH 54/55] gdkpixbuf: fix event handling --- ext/gdk_pixbuf/gstgdkpixbuf.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/ext/gdk_pixbuf/gstgdkpixbuf.c b/ext/gdk_pixbuf/gstgdkpixbuf.c index 70ae78d514..d04b11a171 100644 --- a/ext/gdk_pixbuf/gstgdkpixbuf.c +++ b/ext/gdk_pixbuf/gstgdkpixbuf.c @@ -376,7 +376,7 @@ static gboolean gst_gdk_pixbuf_sink_event (GstPad * pad, GstObject * parent, GstEvent * event) { GstFlowReturn res = GST_FLOW_OK; - gboolean ret = TRUE; + gboolean ret = TRUE, forward = TRUE; GstGdkPixbuf *pixbuf; pixbuf = GST_GDK_PIXBUF (parent); @@ -387,8 +387,8 @@ gst_gdk_pixbuf_sink_event (GstPad * pad, GstObject * parent, GstEvent * event) GstCaps *caps; gst_event_parse_caps (event, &caps); - res = gst_gdk_pixbuf_sink_setcaps (pixbuf, caps); - gst_event_unref (event); + ret = gst_gdk_pixbuf_sink_setcaps (pixbuf, caps); + forward = FALSE; break; } case GST_EVENT_EOS: @@ -403,6 +403,8 @@ gst_gdk_pixbuf_sink_event (GstPad * pad, GstObject * parent, GstEvent * event) if (res != GST_FLOW_OK && res != GST_FLOW_FLUSHING) { GST_ELEMENT_ERROR (pixbuf, STREAM, FAILED, (NULL), ("Flow: %s", gst_flow_get_name (res))); + forward = FALSE; + ret = FALSE; } } break; @@ -417,13 +419,11 @@ gst_gdk_pixbuf_sink_event (GstPad * pad, GstObject * parent, GstEvent * event) default: break; } - - if (res == GST_FLOW_OK) { + if (forward) { ret = gst_pad_event_default (pad, parent, event); } else { - ret = FALSE; + gst_event_unref (event); } - return ret; } From 1f0cf56c8b7211fbfa12f2b36cf1de26b72988b6 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Mon, 5 Mar 2012 15:29:56 +0100 Subject: [PATCH 55/55] gdk: cleanups and fix rowstride Fix the output rowstride, we need to take the stride of the output video frame. Since we are also dealing with planes, take the plane data and stride. Don't store the same info twice in different variables. --- ext/gdk_pixbuf/gstgdkpixbuf.c | 51 ++++++++++++++++++----------------- ext/gdk_pixbuf/gstgdkpixbuf.h | 8 +----- 2 files changed, 27 insertions(+), 32 deletions(-) diff --git a/ext/gdk_pixbuf/gstgdkpixbuf.c b/ext/gdk_pixbuf/gstgdkpixbuf.c index d04b11a171..638acdd5ad 100644 --- a/ext/gdk_pixbuf/gstgdkpixbuf.c +++ b/ext/gdk_pixbuf/gstgdkpixbuf.c @@ -93,14 +93,13 @@ gst_gdk_pixbuf_sink_setcaps (GstGdkPixbuf * filter, GstCaps * caps) s = gst_caps_get_structure (caps, 0); if ((framerate = gst_structure_get_value (s, "framerate")) != NULL) { - filter->framerate_numerator = gst_value_get_fraction_numerator (framerate); - filter->framerate_denominator = - gst_value_get_fraction_denominator (framerate); + filter->in_fps_n = gst_value_get_fraction_numerator (framerate); + filter->in_fps_d = gst_value_get_fraction_denominator (framerate); GST_DEBUG_OBJECT (filter, "got framerate of %d/%d fps => packetized mode", - filter->framerate_numerator, filter->framerate_denominator); + filter->in_fps_n, filter->in_fps_d); } else { - filter->framerate_numerator = 0; - filter->framerate_denominator = 1; + filter->in_fps_n = 0; + filter->in_fps_d = 1; GST_DEBUG_OBJECT (filter, "no framerate, assuming single image"); } @@ -277,9 +276,10 @@ gst_gdk_pixbuf_flush (GstGdkPixbuf * filter) int y; guint8 *out_pix; guint8 *in_pix; - int in_rowstride; + int in_rowstride, out_rowstride; GstFlowReturn ret; GstCaps *caps = NULL; + gint width, height; gint n_channels; GstVideoFrame frame; @@ -287,13 +287,14 @@ gst_gdk_pixbuf_flush (GstGdkPixbuf * filter) if (pixbuf == NULL) goto no_pixbuf; - if (filter->width == 0) { + width = gdk_pixbuf_get_width (pixbuf); + height = gdk_pixbuf_get_height (pixbuf); + + if (GST_VIDEO_INFO_FORMAT (&filter->info) == GST_VIDEO_FORMAT_UNKNOWN) { GstVideoInfo info; GstVideoFormat fmt; - filter->width = gdk_pixbuf_get_width (pixbuf); - filter->height = gdk_pixbuf_get_height (pixbuf); - GST_DEBUG ("Set size to %dx%d", filter->width, filter->height); + GST_DEBUG ("Set size to %dx%d", width, height); n_channels = gdk_pixbuf_get_n_channels (pixbuf); switch (n_channels) { @@ -309,13 +310,11 @@ gst_gdk_pixbuf_flush (GstGdkPixbuf * filter) gst_video_info_init (&info); - gst_video_info_set_format (&info, fmt, filter->width, filter->height); - info.fps_n = filter->framerate_numerator; - info.fps_d = filter->framerate_denominator; + gst_video_info_set_format (&info, fmt, width, height); + info.fps_n = filter->in_fps_n; + info.fps_d = filter->in_fps_d; caps = gst_video_info_to_caps (&info); - filter->channels = n_channels; - filter->rowstride = GST_VIDEO_INFO_COMP_STRIDE (&info, 0); filter->info = info; gst_pad_set_caps (filter->srcpad, caps); @@ -335,12 +334,13 @@ gst_gdk_pixbuf_flush (GstGdkPixbuf * filter) in_rowstride = gdk_pixbuf_get_rowstride (pixbuf); gst_video_frame_map (&frame, &filter->info, outbuf, GST_MAP_WRITE); - out_pix = GST_VIDEO_FRAME_COMP_DATA (&frame, 0); + out_pix = GST_VIDEO_FRAME_PLANE_DATA (&frame, 0); + out_rowstride = GST_VIDEO_FRAME_PLANE_STRIDE (&frame, 0); - for (y = 0; y < filter->height; y++) { - memcpy (out_pix, in_pix, filter->width * filter->channels); + for (y = 0; y < height; y++) { + memcpy (out_pix, in_pix, width * GST_VIDEO_FRAME_COMP_PSTRIDE (&frame, 0)); in_pix += in_rowstride; - out_pix += filter->rowstride; + out_pix += out_rowstride; } gst_video_frame_unmap (&frame); @@ -457,7 +457,7 @@ gst_gdk_pixbuf_chain (GstPad * pad, GstObject * parent, GstBuffer * buf) goto error; /* packetised mode? */ - if (filter->framerate_numerator != 0) { + if (filter->in_fps_n != 0) { gdk_pixbuf_loader_close (filter->pixbuf_loader, NULL); ret = gst_gdk_pixbuf_flush (filter); g_object_unref (filter->pixbuf_loader); @@ -490,8 +490,9 @@ gst_gdk_pixbuf_change_state (GstElement * element, GstStateChange transition) switch (transition) { case GST_STATE_CHANGE_READY_TO_PAUSED: /* default to single image mode, setcaps function might not be called */ - dec->framerate_numerator = 0; - dec->framerate_denominator = 1; + dec->in_fps_n = 0; + dec->in_fps_d = 1; + gst_video_info_init (&dec->info); break; default: break; @@ -503,8 +504,8 @@ gst_gdk_pixbuf_change_state (GstElement * element, GstStateChange transition) switch (transition) { case GST_STATE_CHANGE_PAUSED_TO_READY: - dec->framerate_numerator = 0; - dec->framerate_denominator = 0; + dec->in_fps_n = 0; + dec->in_fps_d = 0; if (dec->pool) { gst_buffer_pool_set_active (dec->pool, FALSE); gst_object_replace ((GstObject **) & dec->pool, NULL); diff --git a/ext/gdk_pixbuf/gstgdkpixbuf.h b/ext/gdk_pixbuf/gstgdkpixbuf.h index a64b7239cd..11ddca3087 100644 --- a/ext/gdk_pixbuf/gstgdkpixbuf.h +++ b/ext/gdk_pixbuf/gstgdkpixbuf.h @@ -49,16 +49,10 @@ struct _GstGdkPixbuf GstClockTime last_timestamp; GdkPixbufLoader *pixbuf_loader; - int width; - int height; - int rowstride; - int channels; + gint in_fps_n, in_fps_d; GstVideoInfo info; GstBufferPool *pool; - - gint framerate_numerator; - gint framerate_denominator; }; struct _GstGdkPixbufClass