From f63f09483f22fd7e2672d3df58be8d8a69b4603c Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Fri, 7 Oct 2011 14:52:33 +0200 Subject: [PATCH] vorbisdec: port to audiodecoder --- ext/vorbis/Makefile.am | 3 +- ext/vorbis/gstvorbisdec.c | 899 ++++++-------------------------------- ext/vorbis/gstvorbisdec.h | 24 +- 3 files changed, 131 insertions(+), 795 deletions(-) diff --git a/ext/vorbis/Makefile.am b/ext/vorbis/Makefile.am index 59f57a2603..9678a31b9e 100644 --- a/ext/vorbis/Makefile.am +++ b/ext/vorbis/Makefile.am @@ -28,7 +28,8 @@ plugin_LTLIBRARIES += libgstivorbisdec.la libgstivorbisdec_la_SOURCES = gstivorbisdec.c \ gstvorbisdec.c gstvorbisdeclib.c gstvorbiscommon.c -libgstivorbisdec_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) \ +libgstivorbisdec_la_CFLAGS = -DGST_USE_UNSTABLE_API \ + $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) \ -DTREMOR $(IVORBIS_CFLAGS) libgstivorbisdec_la_LIBADD = \ $(top_builddir)/gst-libs/gst/tag/libgsttag-@GST_MAJORMINOR@.la \ diff --git a/ext/vorbis/gstvorbisdec.c b/ext/vorbis/gstvorbisdec.c index ee90661489..6e6601a29c 100644 --- a/ext/vorbis/gstvorbisdec.c +++ b/ext/vorbis/gstvorbisdec.c @@ -64,26 +64,16 @@ GST_STATIC_PAD_TEMPLATE ("sink", GST_STATIC_CAPS ("audio/x-vorbis") ); -GST_BOILERPLATE (GST_VORBIS_DEC_GLIB_TYPE_NAME, gst_vorbis_dec, GstElement, - GST_TYPE_ELEMENT); +GST_BOILERPLATE (GstVorbisDec, gst_vorbis_dec, GstAudioDecoder, + GST_TYPE_AUDIO_DECODER); static void vorbis_dec_finalize (GObject * object); -static gboolean vorbis_dec_sink_event (GstPad * pad, GstEvent * event); -static GstFlowReturn vorbis_dec_chain (GstPad * pad, GstBuffer * buffer); -static GstFlowReturn vorbis_dec_chain_forward (GstVorbisDec * vd, - gboolean discont, GstBuffer * buffer); -static GstFlowReturn vorbis_dec_chain_reverse (GstVorbisDec * vd, - gboolean discont, GstBuffer * buf); -static GstStateChangeReturn vorbis_dec_change_state (GstElement * element, - GstStateChange transition); -static gboolean vorbis_dec_src_event (GstPad * pad, GstEvent * event); -static gboolean vorbis_dec_src_query (GstPad * pad, GstQuery * query); -static gboolean vorbis_dec_convert (GstPad * pad, - GstFormat src_format, gint64 src_value, - GstFormat * dest_format, gint64 * dest_value); - -static gboolean vorbis_dec_sink_query (GstPad * pad, GstQuery * query); +static gboolean vorbis_dec_start (GstAudioDecoder * dec); +static gboolean vorbis_dec_stop (GstAudioDecoder * dec); +static GstFlowReturn vorbis_dec_handle_frame (GstAudioDecoder * dec, + GstBuffer * buffer); +static void vorbis_dec_flush (GstAudioDecoder * dec, gboolean hard); static void gst_vorbis_dec_base_init (gpointer g_class) @@ -107,55 +97,19 @@ static void gst_vorbis_dec_class_init (GstVorbisDecClass * klass) { GObjectClass *gobject_class = G_OBJECT_CLASS (klass); - GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass); + GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass); gobject_class->finalize = vorbis_dec_finalize; - gstelement_class->change_state = GST_DEBUG_FUNCPTR (vorbis_dec_change_state); -} - -static const GstQueryType * -vorbis_get_query_types (GstPad * pad) -{ - static const GstQueryType vorbis_dec_src_query_types[] = { - GST_QUERY_POSITION, - GST_QUERY_DURATION, - GST_QUERY_CONVERT, - 0 - }; - - return vorbis_dec_src_query_types; + base_class->start = GST_DEBUG_FUNCPTR (vorbis_dec_start); + base_class->stop = GST_DEBUG_FUNCPTR (vorbis_dec_stop); + base_class->handle_frame = GST_DEBUG_FUNCPTR (vorbis_dec_handle_frame); + base_class->flush = GST_DEBUG_FUNCPTR (vorbis_dec_flush); } static void gst_vorbis_dec_init (GstVorbisDec * dec, GstVorbisDecClass * g_class) { - dec->sinkpad = gst_pad_new_from_static_template (&vorbis_dec_sink_factory, - "sink"); - - gst_pad_set_event_function (dec->sinkpad, - GST_DEBUG_FUNCPTR (vorbis_dec_sink_event)); - gst_pad_set_chain_function (dec->sinkpad, - GST_DEBUG_FUNCPTR (vorbis_dec_chain)); - gst_pad_set_query_function (dec->sinkpad, - GST_DEBUG_FUNCPTR (vorbis_dec_sink_query)); - gst_element_add_pad (GST_ELEMENT (dec), dec->sinkpad); - - dec->srcpad = gst_pad_new_from_static_template (&vorbis_dec_src_factory, - "src"); - - gst_pad_set_event_function (dec->srcpad, - GST_DEBUG_FUNCPTR (vorbis_dec_src_event)); - gst_pad_set_query_type_function (dec->srcpad, - GST_DEBUG_FUNCPTR (vorbis_get_query_types)); - gst_pad_set_query_function (dec->srcpad, - GST_DEBUG_FUNCPTR (vorbis_dec_src_query)); - gst_pad_use_fixed_caps (dec->srcpad); - gst_element_add_pad (GST_ELEMENT (dec), dec->srcpad); - - dec->queued = NULL; - dec->pendingevents = NULL; - dec->taglist = NULL; } static void @@ -169,7 +123,6 @@ vorbis_dec_finalize (GObject * object) #ifndef USE_TREMOLO vorbis_block_clear (&vd->vb); #endif - vorbis_dsp_clear (&vd->vd); vorbis_comment_clear (&vd->vc); vorbis_info_clear (&vd->vi); @@ -180,232 +133,44 @@ vorbis_dec_finalize (GObject * object) static void gst_vorbis_dec_reset (GstVorbisDec * dec) { - dec->last_timestamp = GST_CLOCK_TIME_NONE; - dec->discont = TRUE; - dec->seqnum = gst_util_seqnum_next (); - gst_segment_init (&dec->segment, GST_FORMAT_TIME); - - g_list_foreach (dec->queued, (GFunc) gst_mini_object_unref, NULL); - g_list_free (dec->queued); - dec->queued = NULL; - g_list_foreach (dec->gather, (GFunc) gst_mini_object_unref, NULL); - g_list_free (dec->gather); - dec->gather = NULL; - g_list_foreach (dec->decode, (GFunc) gst_mini_object_unref, NULL); - g_list_free (dec->decode); - dec->decode = NULL; - g_list_foreach (dec->pendingevents, (GFunc) gst_mini_object_unref, NULL); - g_list_free (dec->pendingevents); - dec->pendingevents = NULL; - if (dec->taglist) gst_tag_list_free (dec->taglist); dec->taglist = NULL; } - static gboolean -vorbis_dec_convert (GstPad * pad, - GstFormat src_format, gint64 src_value, - GstFormat * dest_format, gint64 * dest_value) +vorbis_dec_start (GstAudioDecoder * dec) { - gboolean res = TRUE; - GstVorbisDec *dec; - guint64 scale = 1; + GstVorbisDec *vd = GST_VORBIS_DEC (dec); - if (src_format == *dest_format) { - *dest_value = src_value; - return TRUE; - } + GST_DEBUG_OBJECT (dec, "start"); + vorbis_info_init (&vd->vi); + vorbis_comment_init (&vd->vc); + vd->initialized = FALSE; + gst_vorbis_dec_reset (vd); - dec = GST_VORBIS_DEC (gst_pad_get_parent (pad)); - - if (!dec->initialized) - goto no_header; - - if (dec->sinkpad == pad && - (src_format == GST_FORMAT_BYTES || *dest_format == GST_FORMAT_BYTES)) - goto no_format; - - switch (src_format) { - case GST_FORMAT_TIME: - switch (*dest_format) { - case GST_FORMAT_BYTES: - scale = dec->width * dec->vi.channels; - case GST_FORMAT_DEFAULT: - *dest_value = - scale * gst_util_uint64_scale_int (src_value, dec->vi.rate, - GST_SECOND); - break; - default: - res = FALSE; - } - break; - case GST_FORMAT_DEFAULT: - switch (*dest_format) { - case GST_FORMAT_BYTES: - *dest_value = src_value * dec->width * dec->vi.channels; - break; - case GST_FORMAT_TIME: - *dest_value = - gst_util_uint64_scale_int (src_value, GST_SECOND, dec->vi.rate); - break; - default: - res = FALSE; - } - break; - case GST_FORMAT_BYTES: - switch (*dest_format) { - case GST_FORMAT_DEFAULT: - *dest_value = src_value / (dec->width * dec->vi.channels); - break; - case GST_FORMAT_TIME: - *dest_value = gst_util_uint64_scale_int (src_value, GST_SECOND, - dec->vi.rate * dec->width * dec->vi.channels); - break; - default: - res = FALSE; - } - break; - default: - res = FALSE; - } -done: - gst_object_unref (dec); - - return res; - - /* ERRORS */ -no_header: - { - GST_DEBUG_OBJECT (dec, "no header packets received"); - res = FALSE; - goto done; - } -no_format: - { - GST_DEBUG_OBJECT (dec, "formats unsupported"); - res = FALSE; - goto done; - } + return TRUE; } static gboolean -vorbis_dec_src_query (GstPad * pad, GstQuery * query) +vorbis_dec_stop (GstAudioDecoder * dec) { - GstVorbisDec *dec; - gboolean res = FALSE; + GstVorbisDec *vd = GST_VORBIS_DEC (dec); - dec = GST_VORBIS_DEC (gst_pad_get_parent (pad)); - if (G_UNLIKELY (dec == NULL)) - return FALSE; + GST_DEBUG_OBJECT (dec, "stop"); + vd->initialized = FALSE; +#ifndef USE_TREMOLO + vorbis_block_clear (&vd->vb); +#endif + vorbis_dsp_clear (&vd->vd); + vorbis_comment_clear (&vd->vc); + vorbis_info_clear (&vd->vi); + gst_vorbis_dec_reset (vd); - switch (GST_QUERY_TYPE (query)) { - case GST_QUERY_POSITION: - { - gint64 value; - GstFormat format; - gint64 time; - - gst_query_parse_position (query, &format, NULL); - - /* we start from the last seen time */ - time = dec->last_timestamp; - /* correct for the segment values */ - time = gst_segment_to_stream_time (&dec->segment, GST_FORMAT_TIME, time); - - GST_LOG_OBJECT (dec, - "query %p: our time: %" GST_TIME_FORMAT, query, GST_TIME_ARGS (time)); - - /* and convert to the final format */ - if (!(res = - vorbis_dec_convert (pad, GST_FORMAT_TIME, time, &format, &value))) - goto error; - - gst_query_set_position (query, format, value); - - GST_LOG_OBJECT (dec, - "query %p: we return %" G_GINT64_FORMAT " (format %u)", query, value, - format); - - break; - } - case GST_QUERY_DURATION: - { - res = gst_pad_peer_query (dec->sinkpad, query); - if (!res) - goto error; - - break; - } - case GST_QUERY_CONVERT: - { - GstFormat src_fmt, dest_fmt; - gint64 src_val, dest_val; - - gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val); - if (!(res = - vorbis_dec_convert (pad, src_fmt, src_val, &dest_fmt, &dest_val))) - goto error; - gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val); - break; - } - default: - res = gst_pad_query_default (pad, query); - break; - } -done: - gst_object_unref (dec); - - return res; - - /* ERRORS */ -error: - { - GST_WARNING_OBJECT (dec, "error handling query"); - goto done; - } -} - -static gboolean -vorbis_dec_sink_query (GstPad * pad, GstQuery * query) -{ - GstVorbisDec *dec; - gboolean res; - - dec = GST_VORBIS_DEC (gst_pad_get_parent (pad)); - - switch (GST_QUERY_TYPE (query)) { - case GST_QUERY_CONVERT: - { - GstFormat src_fmt, dest_fmt; - gint64 src_val, dest_val; - - gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val); - if (!(res = - vorbis_dec_convert (pad, src_fmt, src_val, &dest_fmt, &dest_val))) - goto error; - gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val); - break; - } - default: - res = gst_pad_query_default (pad, query); - break; - } - -done: - gst_object_unref (dec); - - return res; - - /* ERRORS */ -error: - { - GST_DEBUG_OBJECT (dec, "error converting value"); - goto done; - } + return TRUE; } +#if 0 static gboolean vorbis_dec_src_event (GstPad * pad, GstEvent * event) { @@ -413,10 +178,6 @@ vorbis_dec_src_event (GstPad * pad, GstEvent * event) GstVorbisDec *dec; dec = GST_VORBIS_DEC (gst_pad_get_parent (pad)); - if (G_UNLIKELY (dec == NULL)) { - gst_event_unref (event); - return FALSE; - } switch (GST_EVENT_TYPE (event)) { case GST_EVENT_SEEK: @@ -466,96 +227,7 @@ convert_error: goto done; } } - -static gboolean -vorbis_dec_sink_event (GstPad * pad, GstEvent * event) -{ - gboolean ret = FALSE; - GstVorbisDec *dec; - - dec = GST_VORBIS_DEC (gst_pad_get_parent (pad)); - - GST_LOG_OBJECT (dec, "handling event"); - switch (GST_EVENT_TYPE (event)) { - case GST_EVENT_EOS: - if (dec->segment.rate < 0.0) - vorbis_dec_chain_reverse (dec, TRUE, NULL); - ret = gst_pad_push_event (dec->srcpad, event); - break; - case GST_EVENT_FLUSH_START: - ret = gst_pad_push_event (dec->srcpad, event); - break; - case GST_EVENT_FLUSH_STOP: - /* here we must clean any state in the decoder */ -#ifdef HAVE_VORBIS_SYNTHESIS_RESTART - vorbis_synthesis_restart (&dec->vd); #endif - gst_vorbis_dec_reset (dec); - ret = gst_pad_push_event (dec->srcpad, event); - break; - case GST_EVENT_NEWSEGMENT: - { - GstFormat format; - gdouble rate, arate; - gint64 start, stop, time; - gboolean update; - - gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format, - &start, &stop, &time); - - /* we need time for now */ - if (format != GST_FORMAT_TIME) - goto newseg_wrong_format; - - GST_DEBUG_OBJECT (dec, - "newsegment: update %d, rate %g, arate %g, start %" GST_TIME_FORMAT - ", stop %" GST_TIME_FORMAT ", time %" GST_TIME_FORMAT, - update, rate, arate, GST_TIME_ARGS (start), GST_TIME_ARGS (stop), - GST_TIME_ARGS (time)); - - /* now configure the values */ - gst_segment_set_newsegment_full (&dec->segment, update, - rate, arate, format, start, stop, time); - dec->seqnum = gst_event_get_seqnum (event); - - if (dec->initialized) - /* and forward */ - ret = gst_pad_push_event (dec->srcpad, event); - else { - /* store it to send once we're initialized */ - dec->pendingevents = g_list_append (dec->pendingevents, event); - ret = TRUE; - } - break; - } - case GST_EVENT_TAG: - { - if (dec->initialized) - /* and forward */ - ret = gst_pad_push_event (dec->srcpad, event); - else { - /* store it to send once we're initialized */ - dec->pendingevents = g_list_append (dec->pendingevents, event); - ret = TRUE; - } - break; - } - default: - ret = gst_pad_push_event (dec->srcpad, event); - break; - } -done: - gst_object_unref (dec); - - return ret; - - /* ERRORS */ -newseg_wrong_format: - { - GST_DEBUG_OBJECT (dec, "received non TIME newsegment"); - goto done; - } -} static GstFlowReturn vorbis_handle_identification_packet (GstVorbisDec * vd) @@ -593,7 +265,7 @@ vorbis_handle_identification_packet (GstVorbisDec * vd) } /* negotiate width with downstream */ - caps = gst_pad_get_allowed_caps (vd->srcpad); + caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (vd)); if (caps) { if (!gst_caps_is_empty (caps)) { GstStructure *s; @@ -613,10 +285,11 @@ vorbis_handle_identification_packet (GstVorbisDec * vd) * for mono/stereo and avoid the depth switch in tremor case */ vd->copy_samples = get_copy_sample_func (vd->vi.channels, vd->width); - caps = gst_caps_copy (gst_pad_get_pad_template_caps (vd->srcpad)); - gst_caps_set_simple (caps, "rate", G_TYPE_INT, vd->vi.rate, - "channels", G_TYPE_INT, vd->vi.channels, - "width", G_TYPE_INT, width, NULL); + caps = + gst_caps_copy (gst_pad_get_pad_template_caps + (GST_AUDIO_DECODER_SRC_PAD (vd))); + gst_caps_set_simple (caps, "rate", G_TYPE_INT, vd->vi.rate, "channels", + G_TYPE_INT, vd->vi.channels, "width", G_TYPE_INT, width, NULL); if (pos) { gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos); @@ -626,7 +299,7 @@ vorbis_handle_identification_packet (GstVorbisDec * vd) g_free ((GstAudioChannelPosition *) pos); } - gst_pad_set_caps (vd->srcpad, caps); + gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (vd), caps); gst_caps_unref (caps); return GST_FLOW_OK; @@ -694,8 +367,8 @@ vorbis_handle_comment_packet (GstVorbisDec * vd, ogg_packet * packet) } if (vd->initialized) { - gst_element_found_tags_for_pad (GST_ELEMENT_CAST (vd), vd->srcpad, - vd->taglist); + gst_element_found_tags_for_pad (GST_ELEMENT_CAST (vd), + GST_AUDIO_DECODER_SRC_PAD (vd), vd->taglist); vd->taglist = NULL; } else { /* Only post them as messages for the time being. * @@ -710,7 +383,6 @@ vorbis_handle_comment_packet (GstVorbisDec * vd, ogg_packet * packet) static GstFlowReturn vorbis_handle_type_packet (GstVorbisDec * vd) { - GList *walk; gint res; g_assert (vd->initialized == FALSE); @@ -728,16 +400,10 @@ vorbis_handle_type_packet (GstVorbisDec * vd) vd->initialized = TRUE; - if (vd->pendingevents) { - for (walk = vd->pendingevents; walk; walk = g_list_next (walk)) - gst_pad_push_event (vd->srcpad, GST_EVENT_CAST (walk->data)); - g_list_free (vd->pendingevents); - vd->pendingevents = NULL; - } - if (vd->taglist) { /* The tags have already been sent on the bus as messages. */ - gst_pad_push_event (vd->srcpad, gst_event_new_tag (vd->taglist)); + gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (vd), + gst_event_new_tag (vd->taglist)); vd->taglist = NULL; } return GST_FLOW_OK; @@ -768,7 +434,7 @@ vorbis_handle_header_packet (GstVorbisDec * vd, ogg_packet * packet) /* Packetno = 0 if the first byte is exactly 0x01 */ packet->b_o_s = ((gst_ogg_packet_data (packet))[0] == 0x1) ? 1 : 0; -#ifdef USE_TREMOLO +#ifdef USE_TREMELO if ((ret = vorbis_dsp_headerin (&vd->vi, &vd->vc, packet))) #else if ((ret = vorbis_synthesis_headerin (&vd->vi, &vd->vc, packet))) @@ -791,6 +457,10 @@ vorbis_handle_header_packet (GstVorbisDec * vd, ogg_packet * packet) res = GST_FLOW_OK; break; } + + /* consumer header packet/frame */ + gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (vd), NULL, 1); + return res; /* ERRORS */ @@ -803,77 +473,72 @@ header_read_error: } static GstFlowReturn -vorbis_dec_push_forward (GstVorbisDec * dec, GstBuffer * buf) +vorbis_dec_handle_header_buffer (GstVorbisDec * vd, GstBuffer * buffer) { - GstFlowReturn result; + ogg_packet *packet; + ogg_packet_wrapper packet_wrapper; - /* clip */ - if (!(buf = gst_audio_buffer_clip (buf, &dec->segment, dec->vi.rate, - dec->vi.channels * dec->width))) { - GST_LOG_OBJECT (dec, "clipped buffer"); - return GST_FLOW_OK; - } + gst_ogg_packet_wrapper_from_buffer (&packet_wrapper, buffer); + packet = gst_ogg_packet_from_wrapper (&packet_wrapper); - if (dec->discont) { - GST_LOG_OBJECT (dec, "setting DISCONT"); - GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT); - dec->discont = FALSE; - } - - GST_DEBUG_OBJECT (dec, - "pushing time %" GST_TIME_FORMAT ", dur %" GST_TIME_FORMAT, - GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)), - GST_TIME_ARGS (GST_BUFFER_DURATION (buf))); - - result = gst_pad_push (dec->srcpad, buf); - - return result; + return vorbis_handle_header_packet (vd, packet); } +#define MIN_NUM_HEADERS 3 static GstFlowReturn -vorbis_dec_push_reverse (GstVorbisDec * dec, GstBuffer * buf) +vorbis_dec_handle_header_caps (GstVorbisDec * vd) { GstFlowReturn result = GST_FLOW_OK; + GstCaps *caps; + GstStructure *s = NULL; + const GValue *array = NULL; - dec->queued = g_list_prepend (dec->queued, buf); + caps = GST_PAD_CAPS (GST_AUDIO_DECODER_SINK_PAD (vd)); + if (caps) + s = gst_caps_get_structure (caps, 0); + if (s) + array = gst_structure_get_value (s, "streamheader"); - return result; -} + if (array && (gst_value_array_get_size (array) >= MIN_NUM_HEADERS)) { + const GValue *value = NULL; + GstBuffer *buf = NULL; + gint i = 0; -static void -vorbis_do_timestamps (GstVorbisDec * vd, GstBuffer * buf, gboolean reverse, - GstClockTime timestamp, GstClockTime duration) -{ - /* interpolate reverse */ - if (vd->last_timestamp != -1 && duration != -1 && reverse) - vd->last_timestamp -= duration; + while (result == GST_FLOW_OK) { + value = gst_value_array_get_value (array, i); + buf = gst_value_get_buffer (value); + if (!buf) + goto null_buffer; + result = vorbis_dec_handle_header_buffer (vd, buf); + i++; + } + } else + goto array_error; - /* take buffer timestamp, use interpolated timestamp otherwise */ - if (timestamp != -1) - vd->last_timestamp = timestamp; - else - timestamp = vd->last_timestamp; +done: + return (result != GST_FLOW_OK ? GST_FLOW_NOT_NEGOTIATED : GST_FLOW_OK); - /* interpolate forwards */ - if (vd->last_timestamp != -1 && duration != -1 && !reverse) - vd->last_timestamp += duration; - - GST_LOG_OBJECT (vd, - "keeping timestamp %" GST_TIME_FORMAT " ts %" GST_TIME_FORMAT " dur %" - GST_TIME_FORMAT, GST_TIME_ARGS (vd->last_timestamp), - GST_TIME_ARGS (timestamp), GST_TIME_ARGS (duration)); - - if (buf) { - GST_BUFFER_TIMESTAMP (buf) = timestamp; - GST_BUFFER_DURATION (buf) = duration; + /* ERRORS */ +array_error: + { + GST_WARNING_OBJECT (vd, "streamheader array not found"); + result = GST_FLOW_ERROR; + goto done; + } +null_buffer: + { + GST_WARNING_OBJECT (vd, "streamheader with null buffer received"); + result = GST_FLOW_ERROR; + goto done; } } + static GstFlowReturn vorbis_handle_data_packet (GstVorbisDec * vd, ogg_packet * packet, GstClockTime timestamp, GstClockTime duration) { -#ifdef USE_TREMOLO +#ifdef USE_TREMELO vorbis_sample_t *pcm; #else vorbis_sample_t **pcm; @@ -883,8 +548,11 @@ vorbis_handle_data_packet (GstVorbisDec * vd, ogg_packet * packet, GstFlowReturn result; gint size; - if (G_UNLIKELY (!vd->initialized)) - goto not_initialized; + if (G_UNLIKELY (!vd->initialized)) { + result = vorbis_dec_handle_header_caps (vd); + if (result != GST_FLOW_OK) + goto not_initialized; + } /* normal data packet */ /* FIXME, we can skip decoding if the packet is outside of the @@ -893,8 +561,8 @@ vorbis_handle_data_packet (GstVorbisDec * vd, ogg_packet * packet, * throw away too much. For now we decode everything and clip right * before pushing data. */ -#ifdef USE_TREMOLO - if (G_UNLIKELY (vorbis_dsp_synthesis (&vd->vd, packet, 1))) +#ifdef USE_TREMELO + if (G_UNLIKELY (vorbis_dsp_synthesis (&vd->vb, packet, 1))) goto could_not_read; #else if (G_UNLIKELY (vorbis_synthesis (&vd->vb, packet))) @@ -912,8 +580,8 @@ vorbis_handle_data_packet (GstVorbisDec * vd, ogg_packet * packet, if ((sample_count = vorbis_dsp_pcmout (&vd->vd, NULL, 0)) == 0) #else if ((sample_count = vorbis_synthesis_pcmout (&vd->vd, NULL)) == 0) -#endif goto done; +#endif size = sample_count * vd->vi.channels * vd->width; GST_LOG_OBJECT (vd, "%d samples ready for reading, size %d", sample_count, @@ -921,18 +589,19 @@ vorbis_handle_data_packet (GstVorbisDec * vd, ogg_packet * packet, /* alloc buffer for it */ result = - gst_pad_alloc_buffer_and_set_caps (vd->srcpad, GST_BUFFER_OFFSET_NONE, - size, GST_PAD_CAPS (vd->srcpad), &out); + gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_DECODER_SRC_PAD (vd), + GST_BUFFER_OFFSET_NONE, size, + GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (vd)), &out); if (G_UNLIKELY (result != GST_FLOW_OK)) goto done; /* get samples ready for reading now, should be sample_count */ #ifdef USE_TREMOLO pcm = GST_BUFFER_DATA (out); - if (G_UNLIKELY ((vorbis_dsp_pcmout (&vd->vd, pcm, - sample_count)) != sample_count)) + if (G_UNLIKELY (vorbis_dsp_pcmout (&vd->vd, pcm, sample_count) != + sample_count)) #else - if (G_UNLIKELY ((vorbis_synthesis_pcmout (&vd->vd, &pcm)) != sample_count)) + if (G_UNLIKELY (vorbis_synthesis_pcmout (&vd->vd, &pcm) != sample_count)) #endif goto wrong_samples; @@ -945,22 +614,10 @@ vorbis_handle_data_packet (GstVorbisDec * vd, ogg_packet * packet, GST_LOG_OBJECT (vd, "setting output size to %d", size); GST_BUFFER_SIZE (out) = size; - /* this should not overflow */ - if (duration == -1) - duration = sample_count * GST_SECOND / vd->vi.rate; - - vorbis_do_timestamps (vd, out, FALSE, timestamp, duration); - - if (vd->segment.rate >= 0.0) - result = vorbis_dec_push_forward (vd, out); - else - result = vorbis_dec_push_reverse (vd, out); - done: - if (out == NULL) { - /* no output, still keep track of timestamps */ - vorbis_do_timestamps (vd, NULL, FALSE, timestamp, duration); - } + /* whether or not data produced, consume one frame and advance time */ + result = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (vd), out, 1); + #ifdef USE_TREMOLO vorbis_dsp_read (&vd->vd, sample_count); #else @@ -974,7 +631,7 @@ not_initialized: { GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE, (NULL), ("no header sent yet")); - return GST_FLOW_ERROR; + return GST_FLOW_NOT_NEGOTIATED; } could_not_read: { @@ -998,82 +655,16 @@ wrong_samples: } static GstFlowReturn -vorbis_dec_handle_header_buffer (GstVorbisDec * vd, GstBuffer * buffer) -{ - ogg_packet *packet; - ogg_packet_wrapper packet_wrapper; - - gst_ogg_packet_wrapper_from_buffer (&packet_wrapper, buffer); - packet = gst_ogg_packet_from_wrapper (&packet_wrapper); - - return vorbis_handle_header_packet (vd, packet); -} - - -#define MIN_NUM_HEADERS 3 -static GstFlowReturn -vorbis_dec_handle_header_caps (GstVorbisDec * vd, GstBuffer * buffer) -{ - GstFlowReturn result = GST_FLOW_OK; - GstCaps *caps = GST_PAD_CAPS (vd->sinkpad); - GstStructure *s = gst_caps_get_structure (caps, 0); - const GValue *array = gst_structure_get_value (s, "streamheader"); - - if (array && (gst_value_array_get_size (array) >= MIN_NUM_HEADERS)) { - const GValue *value = NULL; - GstBuffer *buf = NULL; - - /* initial header */ - value = gst_value_array_get_value (array, 0); - buf = gst_value_get_buffer (value); - if (!buf) - goto null_buffer; - result = vorbis_dec_handle_header_buffer (vd, buf); - - /* comment header */ - if (result == GST_FLOW_OK) { - value = gst_value_array_get_value (array, 1); - buf = gst_value_get_buffer (value); - if (!buf) - goto null_buffer; - result = vorbis_dec_handle_header_buffer (vd, buf); - } - - /* bitstream codebook header */ - if (result == GST_FLOW_OK) { - value = gst_value_array_get_value (array, 2); - buf = gst_value_get_buffer (value); - if (!buf) - goto null_buffer; - result = vorbis_dec_handle_header_buffer (vd, buf); - } - } else - goto array_error; - -done: - return (result != GST_FLOW_OK ? GST_FLOW_NOT_NEGOTIATED : GST_FLOW_OK); - -array_error: - { - GST_WARNING_OBJECT (vd, "streamheader array not found"); - result = GST_FLOW_ERROR; - goto done; - } - -null_buffer: - { - GST_WARNING_OBJECT (vd, "streamheader with null buffer received"); - result = GST_FLOW_ERROR; - goto done; - } -} - -static GstFlowReturn -vorbis_dec_decode_buffer (GstVorbisDec * vd, GstBuffer * buffer) +vorbis_dec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer) { ogg_packet *packet; ogg_packet_wrapper packet_wrapper; GstFlowReturn result = GST_FLOW_OK; + GstVorbisDec *vd = GST_VORBIS_DEC (dec); + + /* no draining etc */ + if (G_UNLIKELY (!buffer)) + return GST_FLOW_OK; /* make ogg_packet out of the buffer */ gst_ogg_packet_wrapper_from_buffer (&packet_wrapper, buffer); @@ -1105,13 +696,6 @@ vorbis_dec_decode_buffer (GstVorbisDec * vd, GstBuffer * buffer) } else { GstClockTime timestamp, duration; - /* try to find header in caps so we can initialize the decoder */ - if (!vd->initialized) { - result = vorbis_dec_handle_header_caps (vd, buffer); - if (result != GST_FLOW_OK) - goto invalid_caps_header; - } - timestamp = GST_BUFFER_TIMESTAMP (buffer); duration = GST_BUFFER_DURATION (buffer); @@ -1135,252 +719,19 @@ empty_header: { GST_ELEMENT_ERROR (vd, STREAM, DECODE, (NULL), ("empty header received")); result = GST_FLOW_ERROR; - vd->discont = TRUE; - goto done; - } - -invalid_caps_header: - { - GST_ELEMENT_ERROR (vd, STREAM, DECODE, (NULL), - ("invalid streamheader in caps")); goto done; } } -/* - * Input: - * Buffer decoding order: 7 8 9 4 5 6 3 1 2 EOS - * Discont flag: D D D D - * - * - Each Discont marks a discont in the decoding order. - * - * for vorbis, each buffer is a keyframe when we have the previous - * buffer. This means that to decode buffer 7, we need buffer 6, which - * arrives out of order. - * - * we first gather buffers in the gather queue until we get a DISCONT. We - * prepend each incomming buffer so that they are in reversed order. - * - * gather queue: 9 8 7 - * decode queue: - * output queue: - * - * When a DISCONT is received (buffer 4), we move the gather queue to the - * decode queue. This is simply done be taking the head of the gather queue - * and prepending it to the decode queue. This yields: - * - * gather queue: - * decode queue: 7 8 9 - * output queue: - * - * Then we decode each buffer in the decode queue in order and put the output - * buffer in the output queue. The first buffer (7) will not produce any output - * because it needs the previous buffer (6) which did not arrive yet. This - * yields: - * - * gather queue: - * decode queue: 7 8 9 - * output queue: 9 8 - * - * Then we remove the consumed buffers from the decode queue. Buffer 7 is not - * completely consumed, we need to keep it around for when we receive buffer - * 6. This yields: - * - * gather queue: - * decode queue: 7 - * output queue: 9 8 - * - * Then we accumulate more buffers: - * - * gather queue: 6 5 4 - * decode queue: 7 - * output queue: - * - * prepending to the decode queue on DISCONT yields: - * - * gather queue: - * decode queue: 4 5 6 7 - * output queue: - * - * after decoding and keeping buffer 4: - * - * gather queue: - * decode queue: 4 - * output queue: 7 6 5 - * - * Etc.. - */ -static GstFlowReturn -vorbis_dec_flush_decode (GstVorbisDec * dec) +static void +vorbis_dec_flush (GstAudioDecoder * dec, gboolean hard) { - GstFlowReturn res = GST_FLOW_OK; - GList *walk; + GstVorbisDec *vd = GST_VORBIS_DEC (dec); - walk = dec->decode; - - GST_DEBUG_OBJECT (dec, "flushing buffers to decoder"); - - while (walk) { - GList *next; - GstBuffer *buf = GST_BUFFER_CAST (walk->data); - - GST_DEBUG_OBJECT (dec, "decoding buffer %p, ts %" GST_TIME_FORMAT, - buf, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf))); - - next = g_list_next (walk); - - /* decode buffer, prepend to output queue */ - res = vorbis_dec_decode_buffer (dec, buf); - - /* if we generated output, we can discard the buffer, else we - * keep it in the queue */ - if (dec->queued) { - GST_DEBUG_OBJECT (dec, "decoded buffer to %p", dec->queued->data); - dec->decode = g_list_delete_link (dec->decode, walk); - gst_buffer_unref (buf); - } else { - GST_DEBUG_OBJECT (dec, "buffer did not decode, keeping"); - } - walk = next; - } - while (dec->queued) { - GstBuffer *buf = GST_BUFFER_CAST (dec->queued->data); - GstClockTime timestamp, duration; - - timestamp = GST_BUFFER_TIMESTAMP (buf); - duration = GST_BUFFER_DURATION (buf); - - vorbis_do_timestamps (dec, buf, TRUE, timestamp, duration); - res = vorbis_dec_push_forward (dec, buf); - - dec->queued = g_list_delete_link (dec->queued, dec->queued); - } - return res; -} - -static GstFlowReturn -vorbis_dec_chain_reverse (GstVorbisDec * vd, gboolean discont, GstBuffer * buf) -{ - GstFlowReturn result = GST_FLOW_OK; - - /* if we have a discont, move buffers to the decode list */ - if (G_UNLIKELY (discont)) { - GST_DEBUG_OBJECT (vd, "received discont"); - while (vd->gather) { - GstBuffer *gbuf; - - gbuf = GST_BUFFER_CAST (vd->gather->data); - /* remove from the gather list */ - vd->gather = g_list_delete_link (vd->gather, vd->gather); - /* copy to decode queue */ - vd->decode = g_list_prepend (vd->decode, gbuf); - } - /* flush and decode the decode queue */ - result = vorbis_dec_flush_decode (vd); - } - - if (G_LIKELY (buf)) { - GST_DEBUG_OBJECT (vd, - "gathering buffer %p of size %u, time %" GST_TIME_FORMAT - ", dur %" GST_TIME_FORMAT, buf, GST_BUFFER_SIZE (buf), - GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)), - GST_TIME_ARGS (GST_BUFFER_DURATION (buf))); - - /* add buffer to gather queue */ - vd->gather = g_list_prepend (vd->gather, buf); - } - - return result; -} - -static GstFlowReturn -vorbis_dec_chain_forward (GstVorbisDec * vd, gboolean discont, - GstBuffer * buffer) -{ - GstFlowReturn result; - - result = vorbis_dec_decode_buffer (vd, buffer); - - gst_buffer_unref (buffer); - - return result; -} - -static GstFlowReturn -vorbis_dec_chain (GstPad * pad, GstBuffer * buffer) -{ - GstVorbisDec *vd; - GstFlowReturn result = GST_FLOW_OK; - gboolean discont; - - vd = GST_VORBIS_DEC (gst_pad_get_parent (pad)); - - discont = GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT); - - /* resync on DISCONT */ - if (G_UNLIKELY (discont)) { - GST_DEBUG_OBJECT (vd, "received DISCONT buffer"); - vd->last_timestamp = GST_CLOCK_TIME_NONE; #ifdef HAVE_VORBIS_SYNTHESIS_RESTART - vorbis_synthesis_restart (&vd->vd); -#endif - vd->discont = TRUE; - } - - if (vd->segment.rate >= 0.0) - result = vorbis_dec_chain_forward (vd, discont, buffer); - else - result = vorbis_dec_chain_reverse (vd, discont, buffer); - - gst_object_unref (vd); - - return result; -} - -static GstStateChangeReturn -vorbis_dec_change_state (GstElement * element, GstStateChange transition) -{ - GstVorbisDec *vd = GST_VORBIS_DEC (element); - GstStateChangeReturn res; - - switch (transition) { - case GST_STATE_CHANGE_NULL_TO_READY: - break; - case GST_STATE_CHANGE_READY_TO_PAUSED: - vorbis_info_init (&vd->vi); - vorbis_comment_init (&vd->vc); - vd->initialized = FALSE; - gst_vorbis_dec_reset (vd); - break; - case GST_STATE_CHANGE_PAUSED_TO_PLAYING: - break; - default: - break; - } - - res = parent_class->change_state (element, transition); - - switch (transition) { - case GST_STATE_CHANGE_PLAYING_TO_PAUSED: - break; - case GST_STATE_CHANGE_PAUSED_TO_READY: - GST_DEBUG_OBJECT (vd, "PAUSED -> READY, clearing vorbis structures"); - vd->initialized = FALSE; - -#ifndef USE_TREMOLO - vorbis_block_clear (&vd->vb); + vorbis_synthesis_restart (&vd->vd); #endif - vorbis_dsp_clear (&vd->vd); - vorbis_comment_clear (&vd->vc); - vorbis_info_clear (&vd->vi); - gst_vorbis_dec_reset (vd); - break; - case GST_STATE_CHANGE_READY_TO_NULL: - break; - default: - break; - } - - return res; + if (hard) + gst_vorbis_dec_reset (vd); } diff --git a/ext/vorbis/gstvorbisdec.h b/ext/vorbis/gstvorbisdec.h index 04e4677410..56f9ca406a 100644 --- a/ext/vorbis/gstvorbisdec.h +++ b/ext/vorbis/gstvorbisdec.h @@ -27,6 +27,7 @@ #endif #include +#include #include "gstvorbisdeclib.h" G_BEGIN_DECLS @@ -51,15 +52,11 @@ typedef struct _GstVorbisDecClass GstVorbisDecClass; * Opaque data structure. */ struct _GstVorbisDec { - GstElement element; - - GstPad *sinkpad; - GstPad *srcpad; + GstAudioDecoder element; vorbis_dsp_state vd; vorbis_info vi; vorbis_comment vc; - #ifndef USE_TREMOLO vorbis_block vb; #endif @@ -67,26 +64,13 @@ struct _GstVorbisDec { gboolean initialized; guint width; - /* list of buffers that need timestamps */ - GList *queued; - /* gather/decode queues for reverse playback */ - GList *gather; - GList *decode; - - GstSegment segment; - gboolean discont; - guint32 seqnum; - - GstClockTime last_timestamp; - - GList *pendingevents; GstTagList *taglist; - + CopySampleFunc copy_samples; }; struct _GstVorbisDecClass { - GstElementClass parent_class; + GstAudioDecoderClass parent_class; }; GType gst_vorbis_dec_get_type(void);