mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-09-09 05:38:58 +00:00
opus: port to encoder/decoder base classes
This commit is contained in:
parent
357d7bdfed
commit
ac9c7bbfef
5 changed files with 548 additions and 1287 deletions
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@ -2,10 +2,12 @@ plugin_LTLIBRARIES = libgstopus.la
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libgstopus_la_SOURCES = gstopus.c gstopusdec.c gstopusenc.c
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libgstopus_la_CFLAGS = \
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-DGST_USE_UNSTABLE_API \
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$(GST_PLUGINS_BASE_CFLAGS) \
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$(GST_CFLAGS) \
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$(OPUS_CFLAGS)
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libgstopus_la_LIBADD = \
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-lgstaudio-$(GST_MAJORMINOR) \
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$(GST_PLUGINS_BASE_LIBS) -lgsttag-$(GST_MAJORMINOR) \
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$(GST_BASE_LIBS) \
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$(GST_LIBS) \
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@ -67,37 +67,31 @@ GST_STATIC_PAD_TEMPLATE ("sink",
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GST_STATIC_CAPS ("audio/x-opus")
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);
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G_DEFINE_TYPE (GstOpusDec, gst_opus_dec, GST_TYPE_ELEMENT);
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static gboolean opus_dec_sink_event (GstPad * pad, GstEvent * event);
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static GstFlowReturn opus_dec_chain (GstPad * pad, GstBuffer * buf);
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static gboolean opus_dec_sink_setcaps (GstPad * pad, GstCaps * caps);
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static GstStateChangeReturn opus_dec_change_state (GstElement * element,
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GstStateChange transition);
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static gboolean opus_dec_src_event (GstPad * pad, GstEvent * event);
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static gboolean opus_dec_src_query (GstPad * pad, GstQuery * query);
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static gboolean opus_dec_sink_query (GstPad * pad, GstQuery * query);
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static gboolean opus_dec_convert (GstPad * pad,
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GstFormat src_format, gint64 src_value,
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GstFormat * dest_format, gint64 * dest_value);
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G_DEFINE_TYPE (GstOpusDec, gst_opus_dec, GST_TYPE_AUDIO_DECODER);
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static gboolean gst_opus_dec_start (GstAudioDecoder * dec);
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static gboolean gst_opus_dec_stop (GstAudioDecoder * dec);
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static GstFlowReturn gst_opus_dec_handle_frame (GstAudioDecoder * dec,
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GstBuffer * buffer);
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static gboolean gst_opus_dec_set_format (GstAudioDecoder * bdec,
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GstCaps * caps);
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static GstFlowReturn opus_dec_chain_parse_data (GstOpusDec * dec,
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GstBuffer * buf, GstClockTime timestamp, GstClockTime duration);
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static GstFlowReturn opus_dec_chain_parse_header (GstOpusDec * dec,
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GstBuffer * buf);
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#if 0
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static GstFlowReturn opus_dec_chain_parse_comments (GstOpusDec * dec,
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GstBuffer * buf);
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#endif
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static void
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gst_opus_dec_class_init (GstOpusDecClass * klass)
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{
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GstAudioDecoderClass *adclass;
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GstElementClass *element_class;
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adclass = (GstAudioDecoderClass *) klass;
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element_class = (GstElementClass *) klass;
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adclass->start = GST_DEBUG_FUNCPTR (gst_opus_dec_start);
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adclass->stop = GST_DEBUG_FUNCPTR (gst_opus_dec_stop);
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adclass->handle_frame = GST_DEBUG_FUNCPTR (gst_opus_dec_handle_frame);
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adclass->set_format = GST_DEBUG_FUNCPTR (gst_opus_dec_set_format);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&opus_dec_src_factory));
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gst_element_class_add_pad_template (element_class,
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@ -107,8 +101,6 @@ gst_opus_dec_class_init (GstOpusDecClass * klass)
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"decode opus streams to audio",
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"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
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element_class->change_state = GST_DEBUG_FUNCPTR (opus_dec_change_state);
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GST_DEBUG_CATEGORY_INIT (opusdec_debug, "opusdec", 0,
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"opus decoding element");
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}
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@ -116,8 +108,6 @@ gst_opus_dec_class_init (GstOpusDecClass * klass)
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static void
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gst_opus_dec_reset (GstOpusDec * dec)
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{
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gst_segment_init (&dec->segment, GST_FORMAT_UNDEFINED);
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dec->granulepos = -1;
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dec->packetno = 0;
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dec->frame_size = 0;
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dec->frame_samples = 960;
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@ -126,50 +116,43 @@ gst_opus_dec_reset (GstOpusDec * dec)
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opus_decoder_destroy (dec->state);
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dec->state = NULL;
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}
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#if 0
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if (dec->mode) {
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opus_mode_destroy (dec->mode);
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dec->mode = NULL;
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}
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#endif
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gst_buffer_replace (&dec->streamheader, NULL);
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gst_buffer_replace (&dec->vorbiscomment, NULL);
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g_list_foreach (dec->extra_headers, (GFunc) gst_mini_object_unref, NULL);
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g_list_free (dec->extra_headers);
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dec->extra_headers = NULL;
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#if 0
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memset (&dec->header, 0, sizeof (dec->header));
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#endif
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}
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static void
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gst_opus_dec_init (GstOpusDec * dec)
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{
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dec->sinkpad =
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gst_pad_new_from_static_template (&opus_dec_sink_factory, "sink");
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gst_pad_set_chain_function (dec->sinkpad, GST_DEBUG_FUNCPTR (opus_dec_chain));
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gst_pad_set_event_function (dec->sinkpad,
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GST_DEBUG_FUNCPTR (opus_dec_sink_event));
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gst_pad_set_query_function (dec->sinkpad,
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GST_DEBUG_FUNCPTR (opus_dec_sink_query));
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gst_element_add_pad (GST_ELEMENT (dec), dec->sinkpad);
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dec->srcpad = gst_pad_new_from_static_template (&opus_dec_src_factory, "src");
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gst_pad_use_fixed_caps (dec->srcpad);
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gst_pad_set_event_function (dec->srcpad,
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GST_DEBUG_FUNCPTR (opus_dec_src_event));
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gst_pad_set_query_function (dec->srcpad,
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GST_DEBUG_FUNCPTR (opus_dec_src_query));
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gst_element_add_pad (GST_ELEMENT (dec), dec->srcpad);
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dec->sample_rate = 48000;
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dec->n_channels = 2;
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gst_opus_dec_reset (dec);
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}
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static gboolean
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gst_opus_dec_start (GstAudioDecoder * dec)
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{
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GstOpusDec *odec = GST_OPUS_DEC (dec);
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gst_opus_dec_reset (odec);
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/* we know about concealment */
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gst_audio_decoder_set_plc_aware (dec, TRUE);
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return TRUE;
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}
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static gboolean
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gst_opus_dec_stop (GstAudioDecoder * dec)
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{
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GstOpusDec *odec = GST_OPUS_DEC (dec);
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gst_opus_dec_reset (odec);
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return TRUE;
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}
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static GstFlowReturn
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gst_opus_dec_negotiate_pool (GstOpusDec * dec, GstCaps * caps, gsize bytes)
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{
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@ -181,7 +164,7 @@ gst_opus_dec_negotiate_pool (GstOpusDec * dec, GstCaps * caps, gsize bytes)
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/* find a pool for the negotiated caps now */
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query = gst_query_new_allocation (caps, TRUE);
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if (gst_pad_peer_query (dec->srcpad, query)) {
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if (gst_pad_peer_query (GST_AUDIO_DECODER_SRC_PAD (dec), query)) {
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GST_DEBUG_OBJECT (dec, "got downstream ALLOCATION hints");
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/* we got configuration from our peer, parse them */
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gst_query_parse_allocation_params (query, &size, &min, &max, &prefix,
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@ -216,517 +199,17 @@ gst_opus_dec_negotiate_pool (GstOpusDec * dec, GstCaps * caps, gsize bytes)
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return GST_FLOW_OK;
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}
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static gboolean
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opus_dec_sink_setcaps (GstPad * pad, GstCaps * caps)
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static GstFlowReturn
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gst_opus_dec_parse_header (GstOpusDec * dec, GstBuffer * buf)
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{
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GstOpusDec *dec = GST_OPUS_DEC (gst_pad_get_parent (pad));
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gboolean ret = TRUE;
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GstStructure *s;
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const GValue *streamheader;
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GST_DEBUG_OBJECT (pad, "Setting sink caps to %" GST_PTR_FORMAT, caps);
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s = gst_caps_get_structure (caps, 0);
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if ((streamheader = gst_structure_get_value (s, "streamheader")) &&
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G_VALUE_HOLDS (streamheader, GST_TYPE_ARRAY) &&
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gst_value_array_get_size (streamheader) >= 2) {
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const GValue *header;
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GstBuffer *buf;
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GstFlowReturn res = GST_FLOW_OK;
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header = gst_value_array_get_value (streamheader, 0);
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if (header && G_VALUE_HOLDS (header, GST_TYPE_BUFFER)) {
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buf = gst_value_get_buffer (header);
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res = opus_dec_chain_parse_header (dec, buf);
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if (res != GST_FLOW_OK)
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goto done;
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gst_buffer_replace (&dec->streamheader, buf);
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}
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#if 0
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vorbiscomment = gst_value_array_get_value (streamheader, 1);
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if (vorbiscomment && G_VALUE_HOLDS (vorbiscomment, GST_TYPE_BUFFER)) {
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buf = gst_value_get_buffer (vorbiscomment);
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res = opus_dec_chain_parse_comments (dec, buf);
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if (res != GST_FLOW_OK)
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goto done;
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gst_buffer_replace (&dec->vorbiscomment, buf);
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}
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#endif
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g_list_foreach (dec->extra_headers, (GFunc) gst_mini_object_unref, NULL);
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g_list_free (dec->extra_headers);
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dec->extra_headers = NULL;
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if (gst_value_array_get_size (streamheader) > 2) {
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gint i, n;
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n = gst_value_array_get_size (streamheader);
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for (i = 2; i < n; i++) {
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header = gst_value_array_get_value (streamheader, i);
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buf = gst_value_get_buffer (header);
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dec->extra_headers =
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g_list_prepend (dec->extra_headers, gst_buffer_ref (buf));
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}
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}
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}
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if (!gst_structure_get_int (s, "frame-size", &dec->frame_size)) {
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GST_WARNING_OBJECT (dec, "Frame size not included in caps");
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}
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if (!gst_structure_get_int (s, "channels", &dec->n_channels)) {
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GST_WARNING_OBJECT (dec, "Number of channels not included in caps");
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}
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if (!gst_structure_get_int (s, "rate", &dec->sample_rate)) {
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GST_WARNING_OBJECT (dec, "Sample rate not included in caps");
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}
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switch (dec->frame_size) {
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case 2:
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dec->frame_samples = dec->sample_rate / 400;
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break;
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case 5:
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dec->frame_samples = dec->sample_rate / 200;
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break;
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case 10:
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dec->frame_samples = dec->sample_rate / 100;
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break;
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case 20:
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dec->frame_samples = dec->sample_rate / 50;
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break;
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case 40:
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dec->frame_samples = dec->sample_rate / 25;
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break;
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case 60:
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dec->frame_samples = 3 * dec->sample_rate / 50;
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break;
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default:
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GST_WARNING_OBJECT (dec, "Unsupported frame size: %d", dec->frame_size);
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break;
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}
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dec->frame_duration = gst_util_uint64_scale_int (dec->frame_samples,
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GST_SECOND, dec->sample_rate);
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GST_INFO_OBJECT (dec,
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"Got frame size %d, %d channels, %d Hz, giving %d samples per frame, frame duration %"
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GST_TIME_FORMAT, dec->frame_size, dec->n_channels, dec->sample_rate,
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dec->frame_samples, GST_TIME_ARGS (dec->frame_duration));
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done:
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gst_object_unref (dec);
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return ret;
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}
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static gboolean
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opus_dec_convert (GstPad * pad,
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GstFormat src_format, gint64 src_value,
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GstFormat * dest_format, gint64 * dest_value)
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{
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gboolean res = TRUE;
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GstOpusDec *dec;
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guint64 scale = 1;
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dec = GST_OPUS_DEC (gst_pad_get_parent (pad));
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if (dec->packetno < 1) {
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res = FALSE;
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goto cleanup;
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}
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if (src_format == *dest_format) {
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*dest_value = src_value;
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res = TRUE;
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goto cleanup;
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}
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if (pad == dec->sinkpad &&
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(src_format == GST_FORMAT_BYTES || *dest_format == GST_FORMAT_BYTES)) {
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res = FALSE;
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goto cleanup;
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}
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switch (src_format) {
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case GST_FORMAT_TIME:
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switch (*dest_format) {
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case GST_FORMAT_BYTES:
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scale = sizeof (gint16) * dec->n_channels;
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case GST_FORMAT_DEFAULT:
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*dest_value =
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gst_util_uint64_scale_int (scale * src_value,
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dec->sample_rate, GST_SECOND);
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break;
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default:
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res = FALSE;
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break;
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}
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break;
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case GST_FORMAT_DEFAULT:
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switch (*dest_format) {
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case GST_FORMAT_BYTES:
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*dest_value = src_value * sizeof (gint16) * dec->n_channels;
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break;
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case GST_FORMAT_TIME:
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*dest_value =
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gst_util_uint64_scale_int (src_value, GST_SECOND,
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dec->sample_rate);
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break;
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default:
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res = FALSE;
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break;
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}
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break;
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case GST_FORMAT_BYTES:
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switch (*dest_format) {
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case GST_FORMAT_DEFAULT:
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*dest_value = src_value / (sizeof (gint16) * dec->n_channels);
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break;
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case GST_FORMAT_TIME:
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*dest_value = gst_util_uint64_scale_int (src_value, GST_SECOND,
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dec->sample_rate * sizeof (gint16) * dec->n_channels);
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break;
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default:
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res = FALSE;
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break;
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}
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break;
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default:
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res = FALSE;
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break;
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}
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cleanup:
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gst_object_unref (dec);
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return res;
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}
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static gboolean
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opus_dec_sink_query (GstPad * pad, GstQuery * query)
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{
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GstOpusDec *dec;
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gboolean res;
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dec = GST_OPUS_DEC (gst_pad_get_parent (pad));
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switch (GST_QUERY_TYPE (query)) {
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case GST_QUERY_CONVERT:
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{
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GstFormat src_fmt, dest_fmt;
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gint64 src_val, dest_val;
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gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
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res = opus_dec_convert (pad, src_fmt, src_val, &dest_fmt, &dest_val);
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if (res) {
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gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
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}
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break;
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}
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default:
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res = gst_pad_query_default (pad, query);
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break;
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}
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gst_object_unref (dec);
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return res;
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}
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static gboolean
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opus_dec_src_query (GstPad * pad, GstQuery * query)
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{
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GstOpusDec *dec;
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gboolean res = FALSE;
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dec = GST_OPUS_DEC (gst_pad_get_parent (pad));
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switch (GST_QUERY_TYPE (query)) {
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case GST_QUERY_POSITION:{
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GstSegment segment;
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GstFormat format;
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gint64 cur;
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gst_query_parse_position (query, &format, NULL);
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GST_PAD_STREAM_LOCK (dec->sinkpad);
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segment = dec->segment;
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GST_PAD_STREAM_UNLOCK (dec->sinkpad);
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if (segment.format != GST_FORMAT_TIME) {
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GST_DEBUG_OBJECT (dec, "segment not initialised yet");
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break;
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}
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if ((res = opus_dec_convert (dec->srcpad, GST_FORMAT_TIME,
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segment.position, &format, &cur))) {
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gst_query_set_position (query, format, cur);
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}
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break;
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}
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case GST_QUERY_DURATION:{
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GstFormat format = GST_FORMAT_TIME;
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gint64 dur;
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/* get duration from demuxer */
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if (!gst_pad_query_peer_duration (dec->sinkpad, format, &dur))
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break;
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gst_query_parse_duration (query, &format, NULL);
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/* and convert it into the requested format */
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if ((res = opus_dec_convert (dec->srcpad, GST_FORMAT_TIME,
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dur, &format, &dur))) {
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gst_query_set_duration (query, format, dur);
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}
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break;
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}
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default:
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res = gst_pad_query_default (pad, query);
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break;
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}
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gst_object_unref (dec);
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return res;
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}
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static gboolean
|
||||
opus_dec_src_event (GstPad * pad, GstEvent * event)
|
||||
{
|
||||
gboolean res = FALSE;
|
||||
GstOpusDec *dec = GST_OPUS_DEC (gst_pad_get_parent (pad));
|
||||
|
||||
GST_LOG_OBJECT (dec, "handling %s event", GST_EVENT_TYPE_NAME (event));
|
||||
|
||||
switch (GST_EVENT_TYPE (event)) {
|
||||
case GST_EVENT_SEEK:{
|
||||
GstFormat format, tformat;
|
||||
gdouble rate;
|
||||
GstEvent *real_seek;
|
||||
GstSeekFlags flags;
|
||||
GstSeekType cur_type, stop_type;
|
||||
gint64 cur, stop;
|
||||
gint64 tcur, tstop;
|
||||
|
||||
gst_event_parse_seek (event, &rate, &format, &flags, &cur_type, &cur,
|
||||
&stop_type, &stop);
|
||||
|
||||
/* we have to ask our peer to seek to time here as we know
|
||||
* nothing about how to generate a granulepos from the src
|
||||
* formats or anything.
|
||||
*
|
||||
* First bring the requested format to time
|
||||
*/
|
||||
tformat = GST_FORMAT_TIME;
|
||||
if (!(res = opus_dec_convert (pad, format, cur, &tformat, &tcur)))
|
||||
break;
|
||||
if (!(res = opus_dec_convert (pad, format, stop, &tformat, &tstop)))
|
||||
break;
|
||||
|
||||
/* then seek with time on the peer */
|
||||
real_seek = gst_event_new_seek (rate, GST_FORMAT_TIME,
|
||||
flags, cur_type, tcur, stop_type, tstop);
|
||||
|
||||
GST_LOG_OBJECT (dec, "seek to %" GST_TIME_FORMAT, GST_TIME_ARGS (tcur));
|
||||
|
||||
res = gst_pad_push_event (dec->sinkpad, real_seek);
|
||||
gst_event_unref (event);
|
||||
break;
|
||||
}
|
||||
default:
|
||||
res = gst_pad_event_default (pad, event);
|
||||
break;
|
||||
}
|
||||
|
||||
gst_object_unref (dec);
|
||||
return res;
|
||||
}
|
||||
|
||||
static gboolean
|
||||
opus_dec_sink_event (GstPad * pad, GstEvent * event)
|
||||
{
|
||||
GstOpusDec *dec;
|
||||
gboolean ret = FALSE;
|
||||
|
||||
dec = GST_OPUS_DEC (gst_pad_get_parent (pad));
|
||||
|
||||
GST_LOG_OBJECT (dec, "handling %s event", GST_EVENT_TYPE_NAME (event));
|
||||
|
||||
switch (GST_EVENT_TYPE (event)) {
|
||||
case GST_EVENT_SEGMENT:{
|
||||
const GstSegment *segment = NULL;
|
||||
|
||||
gst_event_parse_segment (event, &segment);
|
||||
|
||||
if (segment->format != GST_FORMAT_TIME)
|
||||
goto newseg_wrong_format;
|
||||
|
||||
if (segment->rate <= 0.0)
|
||||
goto newseg_wrong_rate;
|
||||
|
||||
#if 0
|
||||
/* TODO: update is gone */
|
||||
if (segment->update) {
|
||||
/* time progressed without data, see if we can fill the gap with
|
||||
* some concealment data */
|
||||
if (dec->segment.position < segment->start) {
|
||||
GstClockTime duration;
|
||||
|
||||
duration = segment->start - dec->segment.position;
|
||||
opus_dec_chain_parse_data (dec, NULL, dec->segment.position,
|
||||
duration);
|
||||
}
|
||||
}
|
||||
#endif
|
||||
|
||||
/* now configure the values */
|
||||
gst_segment_copy_into (segment, &dec->segment);
|
||||
|
||||
dec->granulepos = -1;
|
||||
|
||||
GST_DEBUG_OBJECT (dec, "segment now: cur = %" GST_TIME_FORMAT " [%"
|
||||
GST_TIME_FORMAT " - %" GST_TIME_FORMAT "]",
|
||||
GST_TIME_ARGS (dec->segment.position),
|
||||
GST_TIME_ARGS (dec->segment.start),
|
||||
GST_TIME_ARGS (dec->segment.stop));
|
||||
|
||||
ret = gst_pad_push_event (dec->srcpad, event);
|
||||
break;
|
||||
}
|
||||
|
||||
case GST_EVENT_CAPS:
|
||||
{
|
||||
GstCaps *caps;
|
||||
|
||||
gst_event_parse_caps (event, &caps);
|
||||
ret = opus_dec_sink_setcaps (pad, caps);
|
||||
gst_event_unref (event);
|
||||
break;
|
||||
}
|
||||
|
||||
default:
|
||||
ret = gst_pad_event_default (pad, event);
|
||||
break;
|
||||
}
|
||||
|
||||
gst_object_unref (dec);
|
||||
return ret;
|
||||
|
||||
/* ERRORS */
|
||||
newseg_wrong_format:
|
||||
{
|
||||
GST_DEBUG_OBJECT (dec, "received non TIME newsegment");
|
||||
gst_object_unref (dec);
|
||||
return FALSE;
|
||||
}
|
||||
newseg_wrong_rate:
|
||||
{
|
||||
GST_DEBUG_OBJECT (dec, "negative rates not supported yet");
|
||||
gst_object_unref (dec);
|
||||
return FALSE;
|
||||
}
|
||||
return GST_FLOW_OK;
|
||||
}
|
||||
|
||||
static GstFlowReturn
|
||||
opus_dec_chain_parse_header (GstOpusDec * dec, GstBuffer * buf)
|
||||
gst_opus_dec_parse_comments (GstOpusDec * dec, GstBuffer * buf)
|
||||
{
|
||||
GstCaps *caps;
|
||||
int err;
|
||||
|
||||
#if 0
|
||||
dec->samples_per_frame = opus_packet_get_samples_per_frame (
|
||||
(const unsigned char *) GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
|
||||
#endif
|
||||
|
||||
#if 0
|
||||
if (memcmp (dec->header.codec_id, "OPUS ", 8) != 0)
|
||||
goto invalid_header;
|
||||
#endif
|
||||
|
||||
dec->state = opus_decoder_create (dec->sample_rate, dec->n_channels, &err);
|
||||
if (!dec->state || err != OPUS_OK)
|
||||
goto init_failed;
|
||||
|
||||
dec->frame_duration = gst_util_uint64_scale_int (dec->frame_size,
|
||||
GST_SECOND, dec->sample_rate);
|
||||
|
||||
/* set caps */
|
||||
caps = gst_caps_new_simple ("audio/x-raw",
|
||||
"format", G_TYPE_STRING, "S16LE",
|
||||
"rate", G_TYPE_INT, dec->sample_rate,
|
||||
"channels", G_TYPE_INT, dec->n_channels, NULL);
|
||||
|
||||
GST_DEBUG_OBJECT (dec, "rate=%d channels=%d frame-size=%d",
|
||||
dec->sample_rate, dec->n_channels, dec->frame_size);
|
||||
|
||||
if (!gst_pad_set_caps (dec->srcpad, caps))
|
||||
goto nego_failed;
|
||||
|
||||
gst_caps_unref (caps);
|
||||
return GST_FLOW_OK;
|
||||
|
||||
/* ERRORS */
|
||||
#if 0
|
||||
invalid_header:
|
||||
{
|
||||
GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
|
||||
(NULL), ("Invalid header"));
|
||||
return GST_FLOW_ERROR;
|
||||
}
|
||||
mode_init_failed:
|
||||
{
|
||||
GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
|
||||
(NULL), ("Mode initialization failed: %d", error));
|
||||
return GST_FLOW_ERROR;
|
||||
}
|
||||
#endif
|
||||
init_failed:
|
||||
{
|
||||
GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
|
||||
(NULL), ("couldn't initialize decoder"));
|
||||
return GST_FLOW_ERROR;
|
||||
}
|
||||
nego_failed:
|
||||
{
|
||||
GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
|
||||
(NULL), ("couldn't negotiate format"));
|
||||
gst_caps_unref (caps);
|
||||
return GST_FLOW_NOT_NEGOTIATED;
|
||||
}
|
||||
}
|
||||
|
||||
#if 0
|
||||
static GstFlowReturn
|
||||
opus_dec_chain_parse_comments (GstOpusDec * dec, GstBuffer * buf)
|
||||
{
|
||||
GstTagList *list;
|
||||
gchar *encoder = NULL;
|
||||
|
||||
list = gst_tag_list_from_vorbiscomment_buffer (buf, NULL, 0, &encoder);
|
||||
|
||||
if (!list) {
|
||||
GST_WARNING_OBJECT (dec, "couldn't decode comments");
|
||||
list = gst_tag_list_new ();
|
||||
}
|
||||
|
||||
if (encoder) {
|
||||
gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
|
||||
GST_TAG_ENCODER, encoder, NULL);
|
||||
}
|
||||
|
||||
gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
|
||||
GST_TAG_AUDIO_CODEC, "Opus", NULL);
|
||||
|
||||
if (dec->header.bytes_per_packet > 0) {
|
||||
gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
|
||||
GST_TAG_BITRATE, (guint) dec->header.bytes_per_packet * 8, NULL);
|
||||
}
|
||||
|
||||
GST_INFO_OBJECT (dec, "tags: %" GST_PTR_FORMAT, list);
|
||||
|
||||
gst_element_found_tags_for_pad (GST_ELEMENT (dec), dec->srcpad, list);
|
||||
|
||||
g_free (encoder);
|
||||
g_free (ver);
|
||||
|
||||
return GST_FLOW_OK;
|
||||
}
|
||||
#endif
|
||||
|
||||
static GstFlowReturn
|
||||
opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf,
|
||||
|
@ -739,11 +222,6 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf,
|
|||
gint16 *out_data;
|
||||
int n, err;
|
||||
|
||||
if (timestamp != -1) {
|
||||
dec->segment.position = timestamp;
|
||||
dec->granulepos = -1;
|
||||
}
|
||||
|
||||
if (dec->state == NULL) {
|
||||
GstCaps *caps;
|
||||
|
||||
|
@ -760,7 +238,7 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf,
|
|||
GST_DEBUG_OBJECT (dec, "rate=%d channels=%d frame-size=%d",
|
||||
dec->sample_rate, dec->n_channels, dec->frame_size);
|
||||
|
||||
if (!gst_pad_set_caps (dec->srcpad, caps))
|
||||
if (!gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps))
|
||||
GST_ERROR ("nego failure");
|
||||
|
||||
/* negotiate a bufferpool */
|
||||
|
@ -792,8 +270,8 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf,
|
|||
48000));
|
||||
GST_DEBUG ("channels %d", opus_packet_get_nb_channels (data));
|
||||
|
||||
if (gst_pad_check_reconfigure (dec->srcpad)) {
|
||||
GstCaps *caps = gst_pad_get_current_caps (dec->srcpad);
|
||||
if (gst_pad_check_reconfigure (GST_AUDIO_DECODER_SRC_PAD (dec))) {
|
||||
GstCaps *caps = gst_pad_get_current_caps (GST_AUDIO_DECODER_SRC_PAD (dec));
|
||||
gst_opus_dec_negotiate_pool (dec, caps,
|
||||
dec->frame_samples * dec->n_channels * 2);
|
||||
gst_caps_unref (caps);
|
||||
|
@ -818,8 +296,7 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf,
|
|||
}
|
||||
|
||||
if (!GST_CLOCK_TIME_IS_VALID (timestamp)) {
|
||||
timestamp = gst_util_uint64_scale_int (dec->granulepos - dec->frame_size,
|
||||
GST_SECOND, dec->sample_rate);
|
||||
GST_WARNING_OBJECT (dec, "No timestamp in -> no timestamp out");
|
||||
}
|
||||
|
||||
GST_DEBUG_OBJECT (dec, "timestamp=%" GST_TIME_FORMAT,
|
||||
|
@ -827,18 +304,12 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf,
|
|||
|
||||
GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (buf);
|
||||
GST_BUFFER_DURATION (outbuf) = GST_BUFFER_DURATION (buf);
|
||||
if (dec->discont) {
|
||||
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
|
||||
dec->discont = 0;
|
||||
}
|
||||
|
||||
dec->segment.position += dec->frame_duration;
|
||||
|
||||
GST_LOG_OBJECT (dec, "pushing buffer with ts=%" GST_TIME_FORMAT ", dur=%"
|
||||
GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
|
||||
GST_TIME_ARGS (dec->frame_duration));
|
||||
|
||||
res = gst_pad_push (dec->srcpad, outbuf);
|
||||
res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), outbuf, 1);
|
||||
|
||||
gst_buffer_unmap (outbuf, out_data, out_size);
|
||||
|
||||
|
@ -856,70 +327,182 @@ no_bufferpool:
|
|||
return GST_FLOW_ERROR;
|
||||
}
|
||||
|
||||
static GstFlowReturn
|
||||
opus_dec_chain (GstPad * pad, GstBuffer * buf)
|
||||
static gint
|
||||
gst_opus_dec_get_frame_samples (GstOpusDec * dec)
|
||||
{
|
||||
GstFlowReturn res;
|
||||
GstOpusDec *dec;
|
||||
gint frame_samples = 0;
|
||||
switch (dec->frame_size) {
|
||||
case 2:
|
||||
frame_samples = dec->sample_rate / 400;
|
||||
break;
|
||||
case 5:
|
||||
frame_samples = dec->sample_rate / 200;
|
||||
break;
|
||||
case 10:
|
||||
frame_samples = dec->sample_rate / 100;
|
||||
break;
|
||||
case 20:
|
||||
frame_samples = dec->sample_rate / 50;
|
||||
break;
|
||||
case 40:
|
||||
frame_samples = dec->sample_rate / 25;
|
||||
break;
|
||||
case 60:
|
||||
frame_samples = 3 * dec->sample_rate / 50;
|
||||
break;
|
||||
default:
|
||||
GST_WARNING_OBJECT (dec, "Unsupported frame size: %d", dec->frame_size);
|
||||
frame_samples = 0;
|
||||
break;
|
||||
}
|
||||
return frame_samples;
|
||||
}
|
||||
|
||||
dec = GST_OPUS_DEC (gst_pad_get_parent (pad));
|
||||
GST_LOG_OBJECT (pad,
|
||||
"Got buffer ts %" GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT,
|
||||
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
|
||||
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
|
||||
static gboolean
|
||||
gst_opus_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
|
||||
{
|
||||
GstOpusDec *dec = GST_OPUS_DEC (bdec);
|
||||
gboolean ret = TRUE;
|
||||
GstStructure *s;
|
||||
const GValue *streamheader;
|
||||
|
||||
if (GST_BUFFER_IS_DISCONT (buf)) {
|
||||
dec->discont = TRUE;
|
||||
GST_DEBUG_OBJECT (dec, "set_format: %" GST_PTR_FORMAT, caps);
|
||||
|
||||
s = gst_caps_get_structure (caps, 0);
|
||||
if ((streamheader = gst_structure_get_value (s, "streamheader")) &&
|
||||
G_VALUE_HOLDS (streamheader, GST_TYPE_ARRAY) &&
|
||||
gst_value_array_get_size (streamheader) >= 2) {
|
||||
const GValue *header, *vorbiscomment;
|
||||
GstBuffer *buf;
|
||||
GstFlowReturn res = GST_FLOW_OK;
|
||||
|
||||
header = gst_value_array_get_value (streamheader, 0);
|
||||
if (header && G_VALUE_HOLDS (header, GST_TYPE_BUFFER)) {
|
||||
buf = gst_value_get_buffer (header);
|
||||
res = gst_opus_dec_parse_header (dec, buf);
|
||||
if (res != GST_FLOW_OK)
|
||||
goto done;
|
||||
gst_buffer_replace (&dec->streamheader, buf);
|
||||
}
|
||||
|
||||
vorbiscomment = gst_value_array_get_value (streamheader, 1);
|
||||
if (vorbiscomment && G_VALUE_HOLDS (vorbiscomment, GST_TYPE_BUFFER)) {
|
||||
buf = gst_value_get_buffer (vorbiscomment);
|
||||
res = gst_opus_dec_parse_comments (dec, buf);
|
||||
if (res != GST_FLOW_OK)
|
||||
goto done;
|
||||
gst_buffer_replace (&dec->vorbiscomment, buf);
|
||||
}
|
||||
}
|
||||
|
||||
res = opus_dec_chain_parse_data (dec, buf, GST_BUFFER_TIMESTAMP (buf),
|
||||
GST_BUFFER_DURATION (buf));
|
||||
if (!gst_structure_get_int (s, "frame-size", &dec->frame_size)) {
|
||||
GST_WARNING_OBJECT (dec, "Frame size not included in caps");
|
||||
}
|
||||
if (!gst_structure_get_int (s, "channels", &dec->n_channels)) {
|
||||
GST_WARNING_OBJECT (dec, "Number of channels not included in caps");
|
||||
}
|
||||
if (!gst_structure_get_int (s, "rate", &dec->sample_rate)) {
|
||||
GST_WARNING_OBJECT (dec, "Sample rate not included in caps");
|
||||
}
|
||||
|
||||
//done:
|
||||
dec->packetno++;
|
||||
dec->frame_samples = gst_opus_dec_get_frame_samples (dec);
|
||||
dec->frame_duration = gst_util_uint64_scale_int (dec->frame_samples,
|
||||
GST_SECOND, dec->sample_rate);
|
||||
|
||||
gst_buffer_unref (buf);
|
||||
gst_object_unref (dec);
|
||||
GST_INFO_OBJECT (dec,
|
||||
"Got frame size %d, %d channels, %d Hz, giving %d samples per frame, frame duration %"
|
||||
GST_TIME_FORMAT, dec->frame_size, dec->n_channels, dec->sample_rate,
|
||||
dec->frame_samples, GST_TIME_ARGS (dec->frame_duration));
|
||||
|
||||
caps = gst_caps_new_simple ("audio/x-raw",
|
||||
"format", G_TYPE_STRING, "S16LE",
|
||||
"rate", G_TYPE_INT, dec->sample_rate,
|
||||
"channels", G_TYPE_INT, dec->n_channels, NULL);
|
||||
gst_audio_decoder_set_outcaps (GST_AUDIO_DECODER (dec), caps);
|
||||
gst_caps_unref (caps);
|
||||
|
||||
done:
|
||||
return ret;
|
||||
}
|
||||
|
||||
static gboolean
|
||||
memcmp_buffers (GstBuffer * buf1, GstBuffer * buf2)
|
||||
{
|
||||
gsize size1, size2;
|
||||
gpointer data1;
|
||||
gboolean res;
|
||||
|
||||
size1 = gst_buffer_get_size (buf1);
|
||||
size2 = gst_buffer_get_size (buf2);
|
||||
|
||||
if (size1 != size2)
|
||||
return FALSE;
|
||||
|
||||
data1 = gst_buffer_map (buf1, NULL, NULL, GST_MAP_READ);
|
||||
res = gst_buffer_memcmp (buf2, 0, data1, size1) == 0;
|
||||
gst_buffer_unmap (buf1, data1, size1);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static GstStateChangeReturn
|
||||
opus_dec_change_state (GstElement * element, GstStateChange transition)
|
||||
|
||||
static GstFlowReturn
|
||||
gst_opus_dec_handle_frame (GstAudioDecoder * adec, GstBuffer * buf)
|
||||
{
|
||||
GstStateChangeReturn ret;
|
||||
GstOpusDec *dec = GST_OPUS_DEC (element);
|
||||
GstFlowReturn res;
|
||||
GstOpusDec *dec;
|
||||
|
||||
switch (transition) {
|
||||
case GST_STATE_CHANGE_NULL_TO_READY:
|
||||
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
||||
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
||||
default:
|
||||
break;
|
||||
}
|
||||
/* no fancy draining */
|
||||
if (G_UNLIKELY (!buf))
|
||||
return GST_FLOW_OK;
|
||||
|
||||
ret =
|
||||
GST_ELEMENT_CLASS (gst_opus_dec_parent_class)->change_state (element,
|
||||
transition);
|
||||
if (ret != GST_STATE_CHANGE_SUCCESS)
|
||||
return ret;
|
||||
dec = GST_OPUS_DEC (adec);
|
||||
GST_LOG_OBJECT (dec,
|
||||
"Got buffer ts %" GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT,
|
||||
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
|
||||
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
|
||||
|
||||
switch (transition) {
|
||||
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
||||
break;
|
||||
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
||||
gst_opus_dec_reset (dec);
|
||||
if (dec->pool) {
|
||||
gst_buffer_pool_set_active (dec->pool, FALSE);
|
||||
gst_object_unref (dec->pool);
|
||||
dec->pool = NULL;
|
||||
/* If we have the streamheader and vorbiscomment from the caps already
|
||||
* ignore them here */
|
||||
if (dec->streamheader && dec->vorbiscomment) {
|
||||
if (memcmp_buffers (dec->streamheader, buf)) {
|
||||
GST_DEBUG_OBJECT (dec, "found streamheader");
|
||||
gst_audio_decoder_finish_frame (adec, NULL, 1);
|
||||
res = GST_FLOW_OK;
|
||||
} else if (memcmp_buffers (dec->vorbiscomment, buf)) {
|
||||
GST_DEBUG_OBJECT (dec, "found vorbiscomments");
|
||||
gst_audio_decoder_finish_frame (adec, NULL, 1);
|
||||
res = GST_FLOW_OK;
|
||||
} else {
|
||||
res = opus_dec_chain_parse_data (dec, buf, GST_BUFFER_TIMESTAMP (buf),
|
||||
GST_BUFFER_DURATION (buf));
|
||||
}
|
||||
} else {
|
||||
/* Otherwise fall back to packet counting and assume that the
|
||||
* first two packets are the headers. */
|
||||
switch (dec->packetno) {
|
||||
case 0:
|
||||
GST_DEBUG_OBJECT (dec, "counted streamheader");
|
||||
res = GST_FLOW_OK;
|
||||
res = gst_opus_dec_parse_header (dec, buf);
|
||||
gst_audio_decoder_finish_frame (adec, NULL, 1);
|
||||
break;
|
||||
case 1:
|
||||
GST_DEBUG_OBJECT (dec, "counted vorbiscomments");
|
||||
res = GST_FLOW_OK;
|
||||
res = gst_opus_dec_parse_comments (dec, buf);
|
||||
gst_audio_decoder_finish_frame (adec, NULL, 1);
|
||||
break;
|
||||
default:
|
||||
{
|
||||
res = opus_dec_chain_parse_data (dec, buf, GST_BUFFER_TIMESTAMP (buf),
|
||||
GST_BUFFER_DURATION (buf));
|
||||
break;
|
||||
}
|
||||
break;
|
||||
case GST_STATE_CHANGE_READY_TO_NULL:
|
||||
break;
|
||||
default:
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
||||
return ret;
|
||||
dec->packetno++;
|
||||
|
||||
return res;
|
||||
}
|
||||
|
|
|
@ -22,6 +22,7 @@
|
|||
#define __GST_OPUS_DEC_H__
|
||||
|
||||
#include <gst/gst.h>
|
||||
#include <gst/audio/gstaudiodecoder.h>
|
||||
#include <opus/opus.h>
|
||||
|
||||
G_BEGIN_DECLS
|
||||
|
@ -41,11 +42,7 @@ typedef struct _GstOpusDec GstOpusDec;
|
|||
typedef struct _GstOpusDecClass GstOpusDecClass;
|
||||
|
||||
struct _GstOpusDec {
|
||||
GstElement element;
|
||||
|
||||
/* pads */
|
||||
GstPad *sinkpad;
|
||||
GstPad *srcpad;
|
||||
GstAudioDecoder element;
|
||||
|
||||
OpusDecoder *state;
|
||||
int frame_samples;
|
||||
|
@ -54,13 +51,8 @@ struct _GstOpusDec {
|
|||
GstClockTime frame_duration;
|
||||
guint64 packetno;
|
||||
|
||||
GstSegment segment; /* STREAM LOCK */
|
||||
gint64 granulepos; /* -1 = needs to be set from current time */
|
||||
gboolean discont;
|
||||
|
||||
GstBuffer *streamheader;
|
||||
GstBuffer *vorbiscomment;
|
||||
GList *extra_headers;
|
||||
|
||||
int sample_rate;
|
||||
int n_channels;
|
||||
|
@ -69,7 +61,7 @@ struct _GstOpusDec {
|
|||
};
|
||||
|
||||
struct _GstOpusDecClass {
|
||||
GstElementClass parent_class;
|
||||
GstAudioDecoderClass parent_class;
|
||||
};
|
||||
|
||||
GType gst_opus_dec_get_type (void);
|
||||
|
|
File diff suppressed because it is too large
Load diff
|
@ -24,7 +24,7 @@
|
|||
|
||||
|
||||
#include <gst/gst.h>
|
||||
#include <gst/base/gstadapter.h>
|
||||
#include <gst/audio/gstaudioencoder.h>
|
||||
|
||||
#include <opus/opus.h>
|
||||
|
||||
|
@ -48,16 +48,13 @@ typedef struct _GstOpusEnc GstOpusEnc;
|
|||
typedef struct _GstOpusEncClass GstOpusEncClass;
|
||||
|
||||
struct _GstOpusEnc {
|
||||
GstElement element;
|
||||
GstAudioEncoder element;
|
||||
|
||||
/* pads */
|
||||
GstPad *sinkpad;
|
||||
GstPad *srcpad;
|
||||
|
||||
//OpusHeader header;
|
||||
//OpusMode *mode;
|
||||
OpusEncoder *state;
|
||||
GstAdapter *adapter;
|
||||
|
||||
/* properties */
|
||||
gboolean audio_or_voip;
|
||||
|
@ -71,28 +68,19 @@ struct _GstOpusEnc {
|
|||
gboolean dtx;
|
||||
gint packet_loss_percentage;
|
||||
|
||||
int frame_samples;
|
||||
|
||||
gint frame_samples;
|
||||
gint n_channels;
|
||||
gint sample_rate;
|
||||
|
||||
gboolean setup;
|
||||
gboolean header_sent;
|
||||
gboolean eos;
|
||||
GSList *headers;
|
||||
|
||||
guint64 samples_in;
|
||||
guint64 bytes_out;
|
||||
|
||||
guint64 frameno;
|
||||
guint64 frameno_out;
|
||||
|
||||
GstClockTime start_ts;
|
||||
GstClockTime next_ts;
|
||||
guint64 granulepos_offset;
|
||||
GstTagList *tags;
|
||||
};
|
||||
|
||||
struct _GstOpusEncClass {
|
||||
GstElementClass parent_class;
|
||||
GstAudioEncoderClass parent_class;
|
||||
|
||||
/* signals */
|
||||
void (*frame_encoded) (GstElement *element);
|
||||
|
|
Loading…
Reference in a new issue