From f978a60f38ad536efb688df574cf2fadc4d96db2 Mon Sep 17 00:00:00 2001 From: Vincent Penquerc'h Date: Wed, 16 Nov 2011 16:56:43 +0000 Subject: [PATCH] opus: port to base audio encoder/decoder --- ext/opus/Makefile.am | 2 + ext/opus/gstopusdec.c | 817 +++++++++++---------------------------------- ext/opus/gstopusdec.h | 14 +- ext/opus/gstopusenc.c | 901 ++++++++++++++++---------------------------------- ext/opus/gstopusenc.h | 27 +- 5 files changed, 492 insertions(+), 1269 deletions(-) diff --git a/ext/opus/Makefile.am b/ext/opus/Makefile.am index aa50ba9..57e1692 100644 --- a/ext/opus/Makefile.am +++ b/ext/opus/Makefile.am @@ -2,10 +2,12 @@ plugin_LTLIBRARIES = libgstopus.la libgstopus_la_SOURCES = gstopus.c gstopusdec.c gstopusenc.c libgstopus_la_CFLAGS = \ + -DGST_USE_UNSTABLE_API \ $(GST_PLUGINS_BASE_CFLAGS) \ $(GST_CFLAGS) \ $(OPUS_CFLAGS) libgstopus_la_LIBADD = \ + -lgstaudio-$(GST_MAJORMINOR) \ $(GST_PLUGINS_BASE_LIBS) -lgsttag-$(GST_MAJORMINOR) \ $(GST_BASE_LIBS) \ $(GST_LIBS) \ diff --git a/ext/opus/gstopusdec.c b/ext/opus/gstopusdec.c index 35f501a..1abe320 100644 --- a/ext/opus/gstopusdec.c +++ b/ext/opus/gstopusdec.c @@ -68,31 +68,17 @@ GST_STATIC_PAD_TEMPLATE ("sink", GST_STATIC_CAPS ("audio/x-opus") ); -GST_BOILERPLATE (GstOpusDec, gst_opus_dec, GstElement, GST_TYPE_ELEMENT); - -static gboolean opus_dec_sink_event (GstPad * pad, GstEvent * event); -static GstFlowReturn opus_dec_chain (GstPad * pad, GstBuffer * buf); -static gboolean opus_dec_sink_setcaps (GstPad * pad, GstCaps * caps); -static GstStateChangeReturn opus_dec_change_state (GstElement * element, - GstStateChange transition); - -static gboolean opus_dec_src_event (GstPad * pad, GstEvent * event); -static gboolean opus_dec_src_query (GstPad * pad, GstQuery * query); -static gboolean opus_dec_sink_query (GstPad * pad, GstQuery * query); -static const GstQueryType *opus_get_src_query_types (GstPad * pad); -static const GstQueryType *opus_get_sink_query_types (GstPad * pad); -static gboolean opus_dec_convert (GstPad * pad, - GstFormat src_format, gint64 src_value, - GstFormat * dest_format, gint64 * dest_value); - -static GstFlowReturn opus_dec_chain_parse_data (GstOpusDec * dec, - GstBuffer * buf, GstClockTime timestamp, GstClockTime duration); -static GstFlowReturn opus_dec_chain_parse_header (GstOpusDec * dec, - GstBuffer * buf); -#if 0 -static GstFlowReturn opus_dec_chain_parse_comments (GstOpusDec * dec, +GST_BOILERPLATE (GstOpusDec, gst_opus_dec, GstAudioDecoder, + GST_TYPE_AUDIO_DECODER); + +static GstFlowReturn gst_opus_dec_parse_header (GstOpusDec * dec, GstBuffer * buf); -#endif +static gboolean gst_opus_dec_start (GstAudioDecoder * dec); +static gboolean gst_opus_dec_stop (GstAudioDecoder * dec); +static GstFlowReturn gst_opus_dec_handle_frame (GstAudioDecoder * dec, + GstBuffer * buffer); +static gboolean gst_opus_dec_set_format (GstAudioDecoder * bdec, + GstCaps * caps); static void gst_opus_dec_base_init (gpointer g_class) @@ -112,11 +98,16 @@ gst_opus_dec_base_init (gpointer g_class) static void gst_opus_dec_class_init (GstOpusDecClass * klass) { + GstAudioDecoderClass *adclass; GstElementClass *gstelement_class; + adclass = (GstAudioDecoderClass *) klass; gstelement_class = (GstElementClass *) klass; - gstelement_class->change_state = GST_DEBUG_FUNCPTR (opus_dec_change_state); + adclass->start = GST_DEBUG_FUNCPTR (gst_opus_dec_start); + adclass->stop = GST_DEBUG_FUNCPTR (gst_opus_dec_stop); + adclass->handle_frame = GST_DEBUG_FUNCPTR (gst_opus_dec_handle_frame); + adclass->set_format = GST_DEBUG_FUNCPTR (gst_opus_dec_set_format); GST_DEBUG_CATEGORY_INIT (opusdec_debug, "opusdec", 0, "opus decoding element"); @@ -125,8 +116,6 @@ gst_opus_dec_class_init (GstOpusDecClass * klass) static void gst_opus_dec_reset (GstOpusDec * dec) { - gst_segment_init (&dec->segment, GST_FORMAT_UNDEFINED); - dec->granulepos = -1; dec->packetno = 0; dec->frame_size = 0; dec->frame_samples = 960; @@ -135,50 +124,14 @@ gst_opus_dec_reset (GstOpusDec * dec) opus_decoder_destroy (dec->state); dec->state = NULL; } -#if 0 - if (dec->mode) { - opus_mode_destroy (dec->mode); - dec->mode = NULL; - } -#endif gst_buffer_replace (&dec->streamheader, NULL); gst_buffer_replace (&dec->vorbiscomment, NULL); - g_list_foreach (dec->extra_headers, (GFunc) gst_mini_object_unref, NULL); - g_list_free (dec->extra_headers); - dec->extra_headers = NULL; - -#if 0 - memset (&dec->header, 0, sizeof (dec->header)); -#endif } static void gst_opus_dec_init (GstOpusDec * dec, GstOpusDecClass * g_class) { - dec->sinkpad = - gst_pad_new_from_static_template (&opus_dec_sink_factory, "sink"); - gst_pad_set_chain_function (dec->sinkpad, GST_DEBUG_FUNCPTR (opus_dec_chain)); - gst_pad_set_event_function (dec->sinkpad, - GST_DEBUG_FUNCPTR (opus_dec_sink_event)); - gst_pad_set_query_type_function (dec->sinkpad, - GST_DEBUG_FUNCPTR (opus_get_sink_query_types)); - gst_pad_set_query_function (dec->sinkpad, - GST_DEBUG_FUNCPTR (opus_dec_sink_query)); - gst_pad_set_setcaps_function (dec->sinkpad, - GST_DEBUG_FUNCPTR (opus_dec_sink_setcaps)); - gst_element_add_pad (GST_ELEMENT (dec), dec->sinkpad); - - dec->srcpad = gst_pad_new_from_static_template (&opus_dec_src_factory, "src"); - gst_pad_use_fixed_caps (dec->srcpad); - gst_pad_set_event_function (dec->srcpad, - GST_DEBUG_FUNCPTR (opus_dec_src_event)); - gst_pad_set_query_type_function (dec->srcpad, - GST_DEBUG_FUNCPTR (opus_get_src_query_types)); - gst_pad_set_query_function (dec->srcpad, - GST_DEBUG_FUNCPTR (opus_dec_src_query)); - gst_element_add_pad (GST_ELEMENT (dec), dec->srcpad); - dec->sample_rate = 48000; dec->n_channels = 2; @@ -186,532 +139,39 @@ gst_opus_dec_init (GstOpusDec * dec, GstOpusDecClass * g_class) } static gboolean -opus_dec_sink_setcaps (GstPad * pad, GstCaps * caps) -{ - GstOpusDec *dec = GST_OPUS_DEC (gst_pad_get_parent (pad)); - gboolean ret = TRUE; - GstStructure *s; - const GValue *streamheader; - - GST_DEBUG_OBJECT (pad, "Setting sink caps to %" 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; - 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 = opus_dec_chain_parse_header (dec, buf); - if (res != GST_FLOW_OK) - goto done; - gst_buffer_replace (&dec->streamheader, buf); - } -#if 0 - vorbiscomment = gst_value_array_get_value (streamheader, 1); - if (vorbiscomment && G_VALUE_HOLDS (vorbiscomment, GST_TYPE_BUFFER)) { - buf = gst_value_get_buffer (vorbiscomment); - res = opus_dec_chain_parse_comments (dec, buf); - if (res != GST_FLOW_OK) - goto done; - gst_buffer_replace (&dec->vorbiscomment, buf); - } -#endif - - g_list_foreach (dec->extra_headers, (GFunc) gst_mini_object_unref, NULL); - g_list_free (dec->extra_headers); - dec->extra_headers = NULL; - - if (gst_value_array_get_size (streamheader) > 2) { - gint i, n; - - n = gst_value_array_get_size (streamheader); - for (i = 2; i < n; i++) { - header = gst_value_array_get_value (streamheader, i); - buf = gst_value_get_buffer (header); - dec->extra_headers = - g_list_prepend (dec->extra_headers, gst_buffer_ref (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"); - } - switch (dec->frame_size) { - case 2: - dec->frame_samples = dec->sample_rate / 400; - break; - case 5: - dec->frame_samples = dec->sample_rate / 200; - break; - case 10: - dec->frame_samples = dec->sample_rate / 100; - break; - case 20: - dec->frame_samples = dec->sample_rate / 50; - break; - case 40: - dec->frame_samples = dec->sample_rate / 25; - break; - case 60: - dec->frame_samples = 3 * dec->sample_rate / 50; - break; - default: - GST_WARNING_OBJECT (dec, "Unsupported frame size: %d", dec->frame_size); - break; - } - - dec->frame_duration = gst_util_uint64_scale_int (dec->frame_samples, - GST_SECOND, dec->sample_rate); - - 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)); - -done: - gst_object_unref (dec); - return ret; -} - -static gboolean -opus_dec_convert (GstPad * pad, - GstFormat src_format, gint64 src_value, - GstFormat * dest_format, gint64 * dest_value) -{ - gboolean res = TRUE; - GstOpusDec *dec; - guint64 scale = 1; - - dec = GST_OPUS_DEC (gst_pad_get_parent (pad)); - - if (dec->packetno < 1) { - res = FALSE; - goto cleanup; - } - - if (src_format == *dest_format) { - *dest_value = src_value; - res = TRUE; - goto cleanup; - } - - if (pad == dec->sinkpad && - (src_format == GST_FORMAT_BYTES || *dest_format == GST_FORMAT_BYTES)) { - res = FALSE; - goto cleanup; - } - - switch (src_format) { - case GST_FORMAT_TIME: - switch (*dest_format) { - case GST_FORMAT_BYTES: - scale = sizeof (gint16) * dec->n_channels; - case GST_FORMAT_DEFAULT: - *dest_value = - gst_util_uint64_scale_int (scale * src_value, - dec->sample_rate, GST_SECOND); - break; - default: - res = FALSE; - break; - } - break; - case GST_FORMAT_DEFAULT: - switch (*dest_format) { - case GST_FORMAT_BYTES: - *dest_value = src_value * sizeof (gint16) * dec->n_channels; - break; - case GST_FORMAT_TIME: - *dest_value = - gst_util_uint64_scale_int (src_value, GST_SECOND, - dec->sample_rate); - break; - default: - res = FALSE; - break; - } - break; - case GST_FORMAT_BYTES: - switch (*dest_format) { - case GST_FORMAT_DEFAULT: - *dest_value = src_value / (sizeof (gint16) * dec->n_channels); - break; - case GST_FORMAT_TIME: - *dest_value = gst_util_uint64_scale_int (src_value, GST_SECOND, - dec->sample_rate * sizeof (gint16) * dec->n_channels); - break; - default: - res = FALSE; - break; - } - break; - default: - res = FALSE; - break; - } - -cleanup: - gst_object_unref (dec); - return res; -} - -static const GstQueryType * -opus_get_sink_query_types (GstPad * pad) -{ - static const GstQueryType opus_dec_sink_query_types[] = { - GST_QUERY_CONVERT, - 0 - }; - - return opus_dec_sink_query_types; -} - -static gboolean -opus_dec_sink_query (GstPad * pad, GstQuery * query) +gst_opus_dec_start (GstAudioDecoder * dec) { - GstOpusDec *dec; - gboolean res; - - dec = GST_OPUS_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); - res = opus_dec_convert (pad, src_fmt, src_val, &dest_fmt, &dest_val); - if (res) { - gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val); - } - break; - } - default: - res = gst_pad_query_default (pad, query); - break; - } + GstOpusDec *odec = GST_OPUS_DEC (dec); - gst_object_unref (dec); - return res; -} + gst_opus_dec_reset (odec); -static const GstQueryType * -opus_get_src_query_types (GstPad * pad) -{ - static const GstQueryType opus_dec_src_query_types[] = { - GST_QUERY_POSITION, - GST_QUERY_DURATION, - 0 - }; + /* we know about concealment */ + gst_audio_decoder_set_plc_aware (dec, TRUE); - return opus_dec_src_query_types; + return TRUE; } static gboolean -opus_dec_src_query (GstPad * pad, GstQuery * query) +gst_opus_dec_stop (GstAudioDecoder * dec) { - GstOpusDec *dec; - gboolean res = FALSE; - - dec = GST_OPUS_DEC (gst_pad_get_parent (pad)); + GstOpusDec *odec = GST_OPUS_DEC (dec); - switch (GST_QUERY_TYPE (query)) { - case GST_QUERY_POSITION:{ - GstSegment segment; - GstFormat format; - gint64 cur; - - gst_query_parse_position (query, &format, NULL); - - GST_PAD_STREAM_LOCK (dec->sinkpad); - segment = dec->segment; - GST_PAD_STREAM_UNLOCK (dec->sinkpad); - - if (segment.format != GST_FORMAT_TIME) { - GST_DEBUG_OBJECT (dec, "segment not initialised yet"); - break; - } + gst_opus_dec_reset (odec); - if ((res = opus_dec_convert (dec->srcpad, GST_FORMAT_TIME, - segment.last_stop, &format, &cur))) { - gst_query_set_position (query, format, cur); - } - break; - } - case GST_QUERY_DURATION:{ - GstFormat format = GST_FORMAT_TIME; - gint64 dur; - - /* get duration from demuxer */ - if (!gst_pad_query_peer_duration (dec->sinkpad, &format, &dur)) - break; - - gst_query_parse_duration (query, &format, NULL); - - /* and convert it into the requested format */ - if ((res = opus_dec_convert (dec->srcpad, GST_FORMAT_TIME, - dur, &format, &dur))) { - gst_query_set_duration (query, format, dur); - } - break; - } - default: - res = gst_pad_query_default (pad, query); - break; - } - - gst_object_unref (dec); - return res; -} - -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_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); - - if (format != GST_FORMAT_TIME) - goto newseg_wrong_format; - - if (rate <= 0.0) - goto newseg_wrong_rate; - - if (update) { - /* time progressed without data, see if we can fill the gap with - * some concealment data */ - if (dec->segment.last_stop < start) { - GstClockTime duration; - - duration = start - dec->segment.last_stop; - opus_dec_chain_parse_data (dec, NULL, dec->segment.last_stop, - duration); - } - } - - /* now configure the values */ - gst_segment_set_newsegment_full (&dec->segment, update, - rate, arate, GST_FORMAT_TIME, start, stop, time); - - dec->granulepos = -1; - - GST_DEBUG_OBJECT (dec, "segment now: cur = %" GST_TIME_FORMAT " [%" - GST_TIME_FORMAT " - %" GST_TIME_FORMAT "]", - GST_TIME_ARGS (dec->segment.last_stop), - GST_TIME_ARGS (dec->segment.start), - GST_TIME_ARGS (dec->segment.stop)); - - ret = gst_pad_push_event (dec->srcpad, 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 TRUE; } static GstFlowReturn -opus_dec_chain_parse_header (GstOpusDec * dec, GstBuffer * buf) +gst_opus_dec_parse_header (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-int", - "rate", G_TYPE_INT, dec->sample_rate, - "channels", G_TYPE_INT, dec->n_channels, - "signed", G_TYPE_BOOLEAN, TRUE, - "endianness", G_TYPE_INT, G_BYTE_ORDER, - "width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, 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) +gst_opus_dec_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, @@ -724,11 +184,6 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf, gint16 *out_data; int n, err; - if (timestamp != -1) { - dec->segment.last_stop = timestamp; - dec->granulepos = -1; - } - if (dec->state == NULL) { GstCaps *caps; @@ -747,7 +202,7 @@ 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"); gst_caps_unref (caps); @@ -767,14 +222,15 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf, size = 0; } + GST_DEBUG ("frames %d", opus_packet_get_nb_frames (data, size)); GST_DEBUG ("bandwidth %d", opus_packet_get_bandwidth (data)); GST_DEBUG ("samples_per_frame %d", opus_packet_get_samples_per_frame (data, 48000)); GST_DEBUG ("channels %d", opus_packet_get_nb_channels (data)); - res = gst_pad_alloc_buffer_and_set_caps (dec->srcpad, + res = gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), GST_BUFFER_OFFSET_NONE, dec->frame_samples * dec->n_channels * 2, - GST_PAD_CAPS (dec->srcpad), &outbuf); + GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)), &outbuf); if (res != GST_FLOW_OK) { GST_DEBUG_OBJECT (dec, "buf alloc flow: %s", gst_flow_get_name (res)); @@ -783,7 +239,7 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf, out_data = (gint16 *) GST_BUFFER_DATA (outbuf); - GST_LOG_OBJECT (dec, "decoding frame"); + GST_LOG_OBJECT (dec, "decoding %d sample frame", dec->frame_samples); n = opus_decode (dec->state, data, size, out_data, dec->frame_samples, 0); if (n < 0) { @@ -792,8 +248,7 @@ 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, @@ -801,18 +256,12 @@ 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.last_stop += 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); if (res != GST_FLOW_OK) GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res)); @@ -824,63 +273,173 @@ creation_failed: 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; + + 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; - if (GST_BUFFER_IS_DISCONT (buf)) { - dec->discont = TRUE; + 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); + } } + 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"); + } + + 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_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)); - res = opus_dec_chain_parse_data (dec, buf, GST_BUFFER_TIMESTAMP (buf), - GST_BUFFER_DURATION (buf)); + caps = gst_caps_new_simple ("audio/x-raw-int", + "rate", G_TYPE_INT, dec->sample_rate, + "channels", G_TYPE_INT, dec->n_channels, + "signed", G_TYPE_BOOLEAN, TRUE, + "endianness", G_TYPE_INT, G_BYTE_ORDER, + "width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, NULL); + gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps); + gst_caps_unref (caps); -//done: - dec->packetno++; +done: + return ret; +} + +static gboolean +memcmp_buffers (GstBuffer * buf1, GstBuffer * buf2) +{ + gsize size1, size2; - gst_buffer_unref (buf); - gst_object_unref (dec); + size1 = GST_BUFFER_SIZE (buf1); + size2 = GST_BUFFER_SIZE (buf2); - return res; + if (size1 != size2) + return FALSE; + + return !memcmp (GST_BUFFER_DATA (buf1), GST_BUFFER_DATA (buf2), size1); } -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 = 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); - break; - case GST_STATE_CHANGE_READY_TO_NULL: - break; - default: - break; + /* 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_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_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; + } + } } - return ret; + dec->packetno++; + + return res; } diff --git a/ext/opus/gstopusdec.h b/ext/opus/gstopusdec.h index 886a907..8389580 100644 --- a/ext/opus/gstopusdec.h +++ b/ext/opus/gstopusdec.h @@ -22,6 +22,7 @@ #define __GST_OPUS_DEC_H__ #include +#include #include 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; @@ -54,20 +51,15 @@ 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; }; struct _GstOpusDecClass { - GstElementClass parent_class; + GstAudioDecoderClass parent_class; }; GType gst_opus_dec_get_type (void); diff --git a/ext/opus/gstopusenc.c b/ext/opus/gstopusenc.c index 8d40cdf..4be63cb 100644 --- a/ext/opus/gstopusenc.c +++ b/ext/opus/gstopusenc.c @@ -47,6 +47,7 @@ #include #include +#include #include #include "gstopusenc.h" @@ -125,18 +126,26 @@ enum static void gst_opus_enc_finalize (GObject * object); -static gboolean gst_opus_enc_sinkevent (GstPad * pad, GstEvent * event); -static GstFlowReturn gst_opus_enc_chain (GstPad * pad, GstBuffer * buf); +static gboolean gst_opus_enc_sink_event (GstAudioEncoder * benc, + GstEvent * event); static gboolean gst_opus_enc_setup (GstOpusEnc * enc); static void gst_opus_enc_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static void gst_opus_enc_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); -static GstStateChangeReturn gst_opus_enc_change_state (GstElement * element, - GstStateChange transition); -static GstFlowReturn gst_opus_enc_encode (GstOpusEnc * enc, gboolean flush); +static gboolean gst_opus_enc_start (GstAudioEncoder * benc); +static gboolean gst_opus_enc_stop (GstAudioEncoder * benc); +static gboolean gst_opus_enc_set_format (GstAudioEncoder * benc, + GstAudioInfo * info); +static GstFlowReturn gst_opus_enc_handle_frame (GstAudioEncoder * benc, + GstBuffer * buf); +static GstFlowReturn gst_opus_enc_pre_push (GstAudioEncoder * benc, + GstBuffer ** buffer); +static gint64 gst_opus_enc_get_latency (GstOpusEnc * enc); + +static GstFlowReturn gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buffer); static void gst_opus_enc_setup_interfaces (GType opusenc_type) @@ -156,8 +165,8 @@ gst_opus_enc_setup_interfaces (GType opusenc_type) GST_DEBUG_CATEGORY_INIT (opusenc_debug, "opusenc", 0, "Opus encoder"); } -GST_BOILERPLATE_FULL (GstOpusEnc, gst_opus_enc, GstElement, GST_TYPE_ELEMENT, - gst_opus_enc_setup_interfaces); +GST_BOILERPLATE_FULL (GstOpusEnc, gst_opus_enc, GstAudioEncoder, + GST_TYPE_AUDIO_ENCODER, gst_opus_enc_setup_interfaces); static void gst_opus_enc_base_init (gpointer g_class) @@ -179,13 +188,22 @@ gst_opus_enc_class_init (GstOpusEncClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; + GstAudioEncoderClass *base_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; + base_class = (GstAudioEncoderClass *) klass; gobject_class->set_property = gst_opus_enc_set_property; gobject_class->get_property = gst_opus_enc_get_property; + base_class->start = GST_DEBUG_FUNCPTR (gst_opus_enc_start); + base_class->stop = GST_DEBUG_FUNCPTR (gst_opus_enc_stop); + base_class->set_format = GST_DEBUG_FUNCPTR (gst_opus_enc_set_format); + base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_opus_enc_handle_frame); + base_class->pre_push = GST_DEBUG_FUNCPTR (gst_opus_enc_pre_push); + base_class->event = GST_DEBUG_FUNCPTR (gst_opus_enc_sink_event); + g_object_class_install_property (gobject_class, PROP_AUDIO, g_param_spec_boolean ("audio", "Audio or voice", "Audio or voice", DEFAULT_AUDIO, @@ -229,9 +247,6 @@ gst_opus_enc_class_init (GstOpusEncClass * klass) G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_opus_enc_finalize); - - gstelement_class->change_state = - GST_DEBUG_FUNCPTR (gst_opus_enc_change_state); } static void @@ -241,397 +256,164 @@ gst_opus_enc_finalize (GObject * object) enc = GST_OPUS_ENC (object); - g_object_unref (enc->adapter); - G_OBJECT_CLASS (parent_class)->finalize (object); } -static gboolean -gst_opus_enc_sink_setcaps (GstPad * pad, GstCaps * caps) -{ - GstOpusEnc *enc; - GstStructure *structure; - GstCaps *otherpadcaps; - - enc = GST_OPUS_ENC (GST_PAD_PARENT (pad)); - enc->setup = FALSE; - enc->frame_size = DEFAULT_FRAMESIZE; - otherpadcaps = gst_pad_get_allowed_caps (pad); - - structure = gst_caps_get_structure (caps, 0); - gst_structure_get_int (structure, "channels", &enc->n_channels); - gst_structure_get_int (structure, "rate", &enc->sample_rate); - - if (otherpadcaps) { - if (!gst_caps_is_empty (otherpadcaps)) { - GstStructure *ps = gst_caps_get_structure (otherpadcaps, 0); - gst_structure_get_int (ps, "frame-size", &enc->frame_size); - } - gst_caps_unref (otherpadcaps); - } - - GST_DEBUG_OBJECT (pad, "channels=%d rate=%d frame-size=%d", - enc->n_channels, enc->sample_rate, enc->frame_size); - switch (enc->frame_size) { - case 2: - enc->frame_samples = enc->sample_rate / 400; - break; - case 5: - enc->frame_samples = enc->sample_rate / 200; - break; - case 10: - enc->frame_samples = enc->sample_rate / 100; - break; - case 20: - enc->frame_samples = enc->sample_rate / 50; - break; - case 40: - enc->frame_samples = enc->sample_rate / 25; - break; - case 60: - enc->frame_samples = 3 * enc->sample_rate / 50; - break; - default: - GST_WARNING_OBJECT (enc, "Unsupported frame size: %d", enc->frame_size); - return FALSE; - break; - } - GST_DEBUG_OBJECT (pad, "frame_samples %d", enc->frame_samples); - - gst_opus_enc_setup (enc); - - return TRUE; -} - - -static GstCaps * -gst_opus_enc_sink_getcaps (GstPad * pad) +static void +gst_opus_enc_init (GstOpusEnc * enc, GstOpusEncClass * klass) { - GstCaps *caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad)); - GstCaps *peercaps = NULL; - GstOpusEnc *enc = GST_OPUS_ENC (gst_pad_get_parent_element (pad)); - - peercaps = gst_pad_peer_get_caps (enc->srcpad); - - if (peercaps) { - if (!gst_caps_is_empty (peercaps) && !gst_caps_is_any (peercaps)) { - GstStructure *ps = gst_caps_get_structure (peercaps, 0); - GstStructure *s = gst_caps_get_structure (caps, 0); - gint rate, channels; + GstAudioEncoder *benc = GST_AUDIO_ENCODER (enc); - if (gst_structure_get_int (ps, "rate", &rate)) { - gst_structure_fixate_field_nearest_int (s, "rate", rate); - } + GST_DEBUG_OBJECT (enc, "init"); - if (gst_structure_get_int (ps, "channels", &channels)) { - gst_structure_fixate_field_nearest_int (s, "channels", channels); - } - } - gst_caps_unref (peercaps); - } + enc->n_channels = -1; + enc->sample_rate = -1; + enc->frame_samples = 0; - gst_object_unref (enc); + enc->bitrate = DEFAULT_BITRATE; + enc->bandwidth = DEFAULT_BANDWIDTH; + enc->frame_size = DEFAULT_FRAMESIZE; + enc->cbr = DEFAULT_CBR; + enc->constrained_vbr = DEFAULT_CONSTRAINED_VBR; + enc->complexity = DEFAULT_COMPLEXITY; + enc->inband_fec = DEFAULT_INBAND_FEC; + enc->dtx = DEFAULT_DTX; + enc->packet_loss_percentage = DEFAULT_PACKET_LOSS_PERCENT; - return caps; + /* arrange granulepos marking (and required perfect ts) */ + gst_audio_encoder_set_mark_granule (benc, TRUE); + gst_audio_encoder_set_perfect_timestamp (benc, TRUE); } - static gboolean -gst_opus_enc_convert_src (GstPad * pad, GstFormat src_format, gint64 src_value, - GstFormat * dest_format, gint64 * dest_value) +gst_opus_enc_start (GstAudioEncoder * benc) { - gboolean res = TRUE; - GstOpusEnc *enc; - gint64 avg; + GstOpusEnc *enc = GST_OPUS_ENC (benc); - enc = GST_OPUS_ENC (GST_PAD_PARENT (pad)); - - if (enc->samples_in == 0 || enc->bytes_out == 0 || enc->sample_rate == 0) - return FALSE; - - avg = (enc->bytes_out * enc->sample_rate) / (enc->samples_in); - - switch (src_format) { - case GST_FORMAT_BYTES: - switch (*dest_format) { - case GST_FORMAT_TIME: - *dest_value = src_value * GST_SECOND / avg; - break; - default: - res = FALSE; - } - break; - case GST_FORMAT_TIME: - switch (*dest_format) { - case GST_FORMAT_BYTES: - *dest_value = src_value * avg / GST_SECOND; - break; - default: - res = FALSE; - } - break; - default: - res = FALSE; - } - return res; + GST_DEBUG_OBJECT (enc, "start"); + enc->tags = gst_tag_list_new (); + enc->header_sent = FALSE; + return TRUE; } static gboolean -gst_opus_enc_convert_sink (GstPad * pad, GstFormat src_format, - gint64 src_value, GstFormat * dest_format, gint64 * dest_value) +gst_opus_enc_stop (GstAudioEncoder * benc) { - gboolean res = TRUE; - guint scale = 1; - gint bytes_per_sample; - GstOpusEnc *enc; + GstOpusEnc *enc = GST_OPUS_ENC (benc); - enc = GST_OPUS_ENC (GST_PAD_PARENT (pad)); - - bytes_per_sample = enc->n_channels * 2; - - switch (src_format) { - case GST_FORMAT_BYTES: - switch (*dest_format) { - case GST_FORMAT_DEFAULT: - if (bytes_per_sample == 0) - return FALSE; - *dest_value = src_value / bytes_per_sample; - break; - case GST_FORMAT_TIME: - { - gint byterate = bytes_per_sample * enc->sample_rate; - - if (byterate == 0) - return FALSE; - *dest_value = src_value * GST_SECOND / byterate; - break; - } - default: - res = FALSE; - } - break; - case GST_FORMAT_DEFAULT: - switch (*dest_format) { - case GST_FORMAT_BYTES: - *dest_value = src_value * bytes_per_sample; - break; - case GST_FORMAT_TIME: - if (enc->sample_rate == 0) - return FALSE; - *dest_value = src_value * GST_SECOND / enc->sample_rate; - break; - default: - res = FALSE; - } - break; - case GST_FORMAT_TIME: - switch (*dest_format) { - case GST_FORMAT_BYTES: - scale = bytes_per_sample; - /* fallthrough */ - case GST_FORMAT_DEFAULT: - *dest_value = src_value * scale * enc->sample_rate / GST_SECOND; - break; - default: - res = FALSE; - } - break; - default: - res = FALSE; + GST_DEBUG_OBJECT (enc, "stop"); + enc->header_sent = FALSE; + if (enc->state) { + opus_encoder_destroy (enc->state); + enc->state = NULL; } - return res; + gst_tag_list_free (enc->tags); + enc->tags = NULL; + g_slist_foreach (enc->headers, (GFunc) gst_buffer_unref, NULL); + enc->headers = NULL; + + return TRUE; } static gint64 gst_opus_enc_get_latency (GstOpusEnc * enc) { - return gst_util_uint64_scale (enc->frame_samples, GST_SECOND, + gint64 latency = gst_util_uint64_scale (enc->frame_samples, GST_SECOND, enc->sample_rate); + GST_DEBUG_OBJECT (enc, "Latency: %" GST_TIME_FORMAT, GST_TIME_ARGS (latency)); + return latency; } -static const GstQueryType * -gst_opus_enc_get_query_types (GstPad * pad) +static gint +gst_opus_enc_get_frame_samples (GstOpusEnc * enc) { - static const GstQueryType gst_opus_enc_src_query_types[] = { - GST_QUERY_POSITION, - GST_QUERY_DURATION, - GST_QUERY_CONVERT, - GST_QUERY_LATENCY, - 0 - }; - - return gst_opus_enc_src_query_types; -} - -static gboolean -gst_opus_enc_src_query (GstPad * pad, GstQuery * query) -{ - gboolean res = TRUE; - GstOpusEnc *enc; - - enc = GST_OPUS_ENC (gst_pad_get_parent (pad)); - - switch (GST_QUERY_TYPE (query)) { - case GST_QUERY_POSITION: - { - GstFormat fmt, req_fmt; - gint64 pos, val; - - gst_query_parse_position (query, &req_fmt, NULL); - if ((res = gst_pad_query_peer_position (enc->sinkpad, &req_fmt, &val))) { - gst_query_set_position (query, req_fmt, val); - break; - } - - fmt = GST_FORMAT_TIME; - if (!(res = gst_pad_query_peer_position (enc->sinkpad, &fmt, &pos))) - break; - - if ((res = - gst_pad_query_peer_convert (enc->sinkpad, fmt, pos, &req_fmt, - &val))) - gst_query_set_position (query, req_fmt, val); - + gint frame_samples = 0; + switch (enc->frame_size) { + case 2: + frame_samples = enc->sample_rate / 400; break; - } - case GST_QUERY_DURATION: - { - GstFormat fmt, req_fmt; - gint64 dur, val; - - gst_query_parse_duration (query, &req_fmt, NULL); - if ((res = gst_pad_query_peer_duration (enc->sinkpad, &req_fmt, &val))) { - gst_query_set_duration (query, req_fmt, val); - break; - } - - fmt = GST_FORMAT_TIME; - if (!(res = gst_pad_query_peer_duration (enc->sinkpad, &fmt, &dur))) - break; - - if ((res = - gst_pad_query_peer_convert (enc->sinkpad, fmt, dur, &req_fmt, - &val))) { - gst_query_set_duration (query, req_fmt, val); - } + case 5: + frame_samples = enc->sample_rate / 200; 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 = gst_opus_enc_convert_src (pad, src_fmt, src_val, &dest_fmt, - &dest_val))) - goto error; - gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val); + case 10: + frame_samples = enc->sample_rate / 100; break; - } - case GST_QUERY_LATENCY: - { - gboolean live; - GstClockTime min_latency, max_latency; - gint64 latency; - - if ((res = gst_pad_peer_query (enc->sinkpad, query))) { - gst_query_parse_latency (query, &live, &min_latency, &max_latency); - - latency = gst_opus_enc_get_latency (enc); - - /* add our latency */ - min_latency += latency; - if (max_latency != -1) - max_latency += latency; - - gst_query_set_latency (query, live, min_latency, max_latency); - } + case 20: + frame_samples = enc->sample_rate / 50; + break; + case 40: + frame_samples = enc->sample_rate / 25; + break; + case 60: + frame_samples = 3 * enc->sample_rate / 50; break; - } default: - res = gst_pad_peer_query (pad, query); + GST_WARNING_OBJECT (enc, "Unsupported frame size: %d", enc->frame_size); + frame_samples = 0; break; } - -error: - - gst_object_unref (enc); - - return res; + return frame_samples; } static gboolean -gst_opus_enc_sink_query (GstPad * pad, GstQuery * query) +gst_opus_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info) { - gboolean res = TRUE; + GstOpusEnc *enc; - 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 = - gst_opus_enc_convert_sink (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; + enc = GST_OPUS_ENC (benc); + + enc->n_channels = GST_AUDIO_INFO_CHANNELS (info); + enc->sample_rate = GST_AUDIO_INFO_RATE (info); + GST_DEBUG_OBJECT (benc, "Setup with %d channels, %d Hz", enc->n_channels, + enc->sample_rate); + + /* handle reconfigure */ + if (enc->state) { + opus_encoder_destroy (enc->state); + enc->state = NULL; } + if (!gst_opus_enc_setup (enc)) + return FALSE; + + enc->frame_samples = gst_opus_enc_get_frame_samples (enc); -error: - return res; + /* feedback to base class */ + gst_audio_encoder_set_latency (benc, + gst_opus_enc_get_latency (enc), gst_opus_enc_get_latency (enc)); + gst_audio_encoder_set_frame_samples_min (benc, + enc->frame_samples * enc->n_channels * 2); + gst_audio_encoder_set_frame_samples_max (benc, + enc->frame_samples * enc->n_channels * 2); + gst_audio_encoder_set_frame_max (benc, 0); + + return TRUE; } -static void -gst_opus_enc_init (GstOpusEnc * enc, GstOpusEncClass * klass) +static GstBuffer * +gst_opus_enc_create_id_buffer (GstOpusEnc * enc) { - enc->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink"); - gst_element_add_pad (GST_ELEMENT (enc), enc->sinkpad); - gst_pad_set_event_function (enc->sinkpad, - GST_DEBUG_FUNCPTR (gst_opus_enc_sinkevent)); - gst_pad_set_chain_function (enc->sinkpad, - GST_DEBUG_FUNCPTR (gst_opus_enc_chain)); - gst_pad_set_setcaps_function (enc->sinkpad, - GST_DEBUG_FUNCPTR (gst_opus_enc_sink_setcaps)); - gst_pad_set_getcaps_function (enc->sinkpad, - GST_DEBUG_FUNCPTR (gst_opus_enc_sink_getcaps)); - gst_pad_set_query_function (enc->sinkpad, - GST_DEBUG_FUNCPTR (gst_opus_enc_sink_query)); - - enc->srcpad = gst_pad_new_from_static_template (&src_factory, "src"); - gst_pad_set_query_function (enc->srcpad, - GST_DEBUG_FUNCPTR (gst_opus_enc_src_query)); - gst_pad_set_query_type_function (enc->srcpad, - GST_DEBUG_FUNCPTR (gst_opus_enc_get_query_types)); - gst_element_add_pad (GST_ELEMENT (enc), enc->srcpad); + GstBuffer *buffer; + GstByteWriter bw; - enc->n_channels = -1; - enc->sample_rate = -1; - enc->frame_samples = 0; + gst_byte_writer_init (&bw); - enc->bitrate = DEFAULT_BITRATE; - enc->bandwidth = DEFAULT_BANDWIDTH; - enc->frame_size = DEFAULT_FRAMESIZE; - enc->cbr = DEFAULT_CBR; - enc->constrained_vbr = DEFAULT_CONSTRAINED_VBR; - enc->complexity = DEFAULT_COMPLEXITY; - enc->inband_fec = DEFAULT_INBAND_FEC; - enc->dtx = DEFAULT_DTX; - enc->packet_loss_percentage = DEFAULT_PACKET_LOSS_PERCENT; + /* See http://wiki.xiph.org/OggOpus */ + gst_byte_writer_put_string_utf8 (&bw, "OpusHead"); + gst_byte_writer_put_uint8 (&bw, 0); /* version number */ + gst_byte_writer_put_uint8 (&bw, enc->n_channels); + gst_byte_writer_put_uint16_le (&bw, 0); /* pre-skip *//* TODO: endianness ? */ + gst_byte_writer_put_uint32_le (&bw, enc->sample_rate); + gst_byte_writer_put_uint16_le (&bw, 0); /* output gain *//* TODO: endianness ? */ + gst_byte_writer_put_uint8 (&bw, 0); /* channel mapping *//* TODO: what is this ? */ - enc->setup = FALSE; - enc->header_sent = FALSE; + buffer = gst_byte_writer_reset_and_get_buffer (&bw); - enc->adapter = gst_adapter_new (); + GST_BUFFER_OFFSET (buffer) = 0; + GST_BUFFER_OFFSET_END (buffer) = 0; + + return buffer; } -#if 0 static GstBuffer * gst_opus_enc_create_metadata_buffer (GstOpusEnc * enc) { @@ -649,10 +431,11 @@ gst_opus_enc_create_metadata_buffer (GstOpusEnc * enc) empty_tags = gst_tag_list_new (); tags = empty_tags; } - comments = gst_tag_list_to_vorbiscomment_buffer (tags, NULL, - 0, "Encoded with GStreamer Opusenc"); + comments = + gst_tag_list_to_vorbiscomment_buffer (tags, (const guint8 *) "OpusTags", + 8, "Encoded with GStreamer Opusenc"); - GST_BUFFER_OFFSET (comments) = enc->bytes_out; + GST_BUFFER_OFFSET (comments) = 0; GST_BUFFER_OFFSET_END (comments) = 0; if (empty_tags) @@ -660,13 +443,14 @@ gst_opus_enc_create_metadata_buffer (GstOpusEnc * enc) return comments; } -#endif static gboolean gst_opus_enc_setup (GstOpusEnc * enc) { int error = OPUS_OK; + GST_DEBUG_OBJECT (enc, "setup"); + enc->setup = FALSE; enc->state = opus_encoder_create (enc->sample_rate, enc->n_channels, @@ -692,88 +476,20 @@ gst_opus_enc_setup (GstOpusEnc * enc) return TRUE; -#if 0 -mode_initialization_failed: - GST_ERROR_OBJECT (enc, "Mode initialization failed: %d", error); - return FALSE; -#endif - encoder_creation_failed: GST_ERROR_OBJECT (enc, "Encoder creation failed"); return FALSE; } - -/* push out the buffer and do internal bookkeeping */ -static GstFlowReturn -gst_opus_enc_push_buffer (GstOpusEnc * enc, GstBuffer * buffer) -{ - guint size; - - size = GST_BUFFER_SIZE (buffer); - - enc->bytes_out += size; - - GST_DEBUG_OBJECT (enc, "pushing output buffer of size %u", size); - - return gst_pad_push (enc->srcpad, buffer); -} - -#if 0 -static GstCaps * -gst_opus_enc_set_header_on_caps (GstCaps * caps, GstBuffer * buf1, - GstBuffer * buf2) -{ - GstStructure *structure = NULL; - GstBuffer *buf; - GValue array = { 0 }; - GValue value = { 0 }; - - caps = gst_caps_make_writable (caps); - structure = gst_caps_get_structure (caps, 0); - - g_assert (gst_buffer_is_metadata_writable (buf1)); - g_assert (gst_buffer_is_metadata_writable (buf2)); - - /* mark buffers */ - GST_BUFFER_FLAG_SET (buf1, GST_BUFFER_FLAG_IN_CAPS); - GST_BUFFER_FLAG_SET (buf2, GST_BUFFER_FLAG_IN_CAPS); - - /* put buffers in a fixed list */ - g_value_init (&array, GST_TYPE_ARRAY); - g_value_init (&value, GST_TYPE_BUFFER); - buf = gst_buffer_copy (buf1); - gst_value_set_buffer (&value, buf); - gst_buffer_unref (buf); - gst_value_array_append_value (&array, &value); - g_value_unset (&value); - g_value_init (&value, GST_TYPE_BUFFER); - buf = gst_buffer_copy (buf2); - gst_value_set_buffer (&value, buf); - gst_buffer_unref (buf); - gst_value_array_append_value (&array, &value); - gst_structure_set_value (structure, "streamheader", &array); - g_value_unset (&value); - g_value_unset (&array); - - return caps; -} -#endif - - static gboolean -gst_opus_enc_sinkevent (GstPad * pad, GstEvent * event) +gst_opus_enc_sink_event (GstAudioEncoder * benc, GstEvent * event) { - gboolean res = TRUE; GstOpusEnc *enc; - enc = GST_OPUS_ENC (gst_pad_get_parent (pad)); + enc = GST_OPUS_ENC (benc); + GST_DEBUG_OBJECT (enc, "sink event: %s", GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { - case GST_EVENT_EOS: - gst_opus_enc_encode (enc, TRUE); - res = gst_pad_event_default (pad, event); - break; case GST_EVENT_TAG: { GstTagList *list; @@ -782,62 +498,94 @@ gst_opus_enc_sinkevent (GstPad * pad, GstEvent * event) gst_event_parse_tag (event, &list); gst_tag_setter_merge_tags (setter, list, mode); - res = gst_pad_event_default (pad, event); break; } default: - res = gst_pad_event_default (pad, event); break; } - gst_object_unref (enc); - - return res; + return FALSE; } static GstFlowReturn -gst_opus_enc_encode (GstOpusEnc * enc, gboolean flush) +gst_opus_enc_pre_push (GstAudioEncoder * benc, GstBuffer ** buffer) { - GstFlowReturn ret = GST_FLOW_OK; - gint bytes = enc->frame_samples * 2 * enc->n_channels; - gint bytes_per_packet; + GstOpusEnc *enc; - bytes_per_packet = - (enc->bitrate * enc->frame_samples / enc->sample_rate + 4) / 8; + enc = GST_OPUS_ENC (benc); + + /* FIXME 0.11 ? get rid of this special ogg stuff and have it + * put and use 'codec data' in caps like anything else, + * with all the usual out-of-band advantage etc */ + if (G_UNLIKELY (enc->headers)) { + GSList *header = enc->headers; + + /* try to push all of these, if we lose one, might as well lose all */ + while (header) { + if (ret == GST_FLOW_OK) + ret = gst_pad_push (GST_AUDIO_ENCODER_SRC_PAD (enc), header->data); + else + gst_pad_push (GST_AUDIO_ENCODER_SRC_PAD (enc), header->data); + header = g_slist_next (header); + } - if (flush && gst_adapter_available (enc->adapter) % bytes != 0) { - guint diff = bytes - gst_adapter_available (enc->adapter) % bytes; - GstBuffer *buf = gst_buffer_new_and_alloc (diff); + g_slist_free (enc->headers); + enc->headers = NULL; + } - memset (GST_BUFFER_DATA (buf), 0, diff); - gst_adapter_push (enc->adapter, buf); + return ret; +} + +static GstFlowReturn +gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf) +{ + guint8 *bdata, *data, *mdata = NULL; + gsize bsize, size; + gsize bytes = enc->frame_samples * enc->n_channels * 2; + gsize bytes_per_packet = + (enc->bitrate * enc->frame_samples / enc->sample_rate + 4) / 8; + gint ret = GST_FLOW_OK; + + if (G_LIKELY (buf)) { + bdata = GST_BUFFER_DATA (buf); + bsize = GST_BUFFER_SIZE (buf); + if (G_UNLIKELY (bsize % bytes)) { + GST_DEBUG_OBJECT (enc, "draining; adding silence samples"); + + size = ((bsize / bytes) + 1) * bytes; + mdata = g_malloc0 (size); + memcpy (mdata, bdata, bsize); + bdata = NULL; + data = mdata; + } else { + data = bdata; + size = bsize; + } + } else { + GST_DEBUG_OBJECT (enc, "nothing to drain"); + goto done; } - while (gst_adapter_available (enc->adapter) >= bytes) { - gint16 *data; + while (size) { gint outsize; GstBuffer *outbuf; - ret = gst_pad_alloc_buffer_and_set_caps (enc->srcpad, - GST_BUFFER_OFFSET_NONE, bytes_per_packet, GST_PAD_CAPS (enc->srcpad), - &outbuf); + ret = gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc), + GST_BUFFER_OFFSET_NONE, bytes_per_packet, + GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (enc)), &outbuf); if (GST_FLOW_OK != ret) goto done; - data = (gint16 *) gst_adapter_take (enc->adapter, bytes); - enc->samples_in += enc->frame_samples; - - GST_DEBUG_OBJECT (enc, "encoding %d samples (%d bytes)", - enc->frame_samples, bytes); + GST_DEBUG_OBJECT (enc, "encoding %d samples (%d bytes) to %d bytes", + enc->frame_samples, bytes, bytes_per_packet); - outsize = opus_encode (enc->state, data, enc->frame_samples, + outsize = + opus_encode (enc->state, (const gint16 *) data, enc->frame_samples, GST_BUFFER_DATA (outbuf), bytes_per_packet); - g_free (data); - if (outsize < 0) { GST_ERROR_OBJECT (enc, "Encoding failed: %d", outsize); ret = GST_FLOW_ERROR; @@ -850,149 +598,132 @@ gst_opus_enc_encode (GstOpusEnc * enc, gboolean flush) goto done; } - GST_BUFFER_TIMESTAMP (outbuf) = enc->start_ts + - gst_util_uint64_scale_int (enc->frameno_out * enc->frame_samples, - GST_SECOND, enc->sample_rate); - GST_BUFFER_DURATION (outbuf) = - gst_util_uint64_scale_int (enc->frame_samples, GST_SECOND, - enc->sample_rate); - GST_BUFFER_OFFSET (outbuf) = - gst_util_uint64_scale_int (GST_BUFFER_OFFSET_END (outbuf), GST_SECOND, - enc->sample_rate); - - enc->frameno++; - enc->frameno_out++; - - ret = gst_opus_enc_push_buffer (enc, outbuf); + ret = + gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc), outbuf, + enc->frame_samples); if ((GST_FLOW_OK != ret) && (GST_FLOW_NOT_LINKED != ret)) goto done; + + data += bytes; + size -= bytes; } done: + if (mdata) + g_free (mdata); + return ret; } -static GstFlowReturn -gst_opus_enc_chain (GstPad * pad, GstBuffer * buf) +/* + * (really really) FIXME: move into core (dixit tpm) + */ +/** + * _gst_caps_set_buffer_array: + * @caps: a #GstCaps + * @field: field in caps to set + * @buf: header buffers + * + * Adds given buffers to an array of buffers set as the given @field + * on the given @caps. List of buffer arguments must be NULL-terminated. + * + * Returns: input caps with a streamheader field added, or NULL if some error + */ +static GstCaps * +_gst_caps_set_buffer_array (GstCaps * caps, const gchar * field, + GstBuffer * buf, ...) { - GstOpusEnc *enc; - GstFlowReturn ret = GST_FLOW_OK; - - enc = GST_OPUS_ENC (GST_PAD_PARENT (pad)); + GstStructure *structure = NULL; + va_list va; + GValue array = { 0 }; + GValue value = { 0 }; - if (!enc->setup) - goto not_setup; + g_return_val_if_fail (caps != NULL, NULL); + g_return_val_if_fail (gst_caps_is_fixed (caps), NULL); + g_return_val_if_fail (field != NULL, NULL); - if (!enc->header_sent) { - GstCaps *caps; + caps = gst_caps_make_writable (caps); + structure = gst_caps_get_structure (caps, 0); - caps = gst_pad_get_caps (enc->srcpad); - gst_caps_set_simple (caps, - "rate", G_TYPE_INT, enc->sample_rate, - "channels", G_TYPE_INT, enc->n_channels, - "frame-size", G_TYPE_INT, enc->frame_size, NULL); + g_value_init (&array, GST_TYPE_ARRAY); - /* negotiate with these caps */ - GST_DEBUG_OBJECT (enc, "here are the caps: %" GST_PTR_FORMAT, caps); - GST_LOG_OBJECT (enc, "rate=%d channels=%d frame-size=%d", - enc->sample_rate, enc->n_channels, enc->frame_size); - gst_pad_set_caps (enc->srcpad, caps); + va_start (va, buf); + /* put buffers in a fixed list */ + while (buf) { + g_assert (gst_buffer_is_writable (buf)); - enc->header_sent = TRUE; - } + /* mark buffer */ + GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS); - GST_DEBUG_OBJECT (enc, "received buffer of %u bytes", GST_BUFFER_SIZE (buf)); + g_value_init (&value, GST_TYPE_BUFFER); + buf = gst_buffer_copy (buf); + GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS); + gst_value_set_buffer (&value, buf); + gst_buffer_unref (buf); + gst_value_array_append_value (&array, &value); + g_value_unset (&value); - /* Save the timestamp of the first buffer. This will be later - * used as offset for all following buffers */ - if (enc->start_ts == GST_CLOCK_TIME_NONE) { - if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) { - enc->start_ts = GST_BUFFER_TIMESTAMP (buf); - } else { - enc->start_ts = 0; - } + buf = va_arg (va, GstBuffer *); } + gst_structure_set_value (structure, field, &array); + g_value_unset (&array); - /* 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->sample_rate) * enc->n_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; + return caps; +} - GST_BUFFER_TIMESTAMP (buf) += diff; - if (GST_BUFFER_DURATION_IS_VALID (buf)) - GST_BUFFER_DURATION (buf) -= diff; - } +static GstFlowReturn +gst_opus_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf) +{ + GstOpusEnc *enc; + GstFlowReturn ret = GST_FLOW_OK; - if (enc->next_ts != GST_CLOCK_TIME_NONE - && GST_BUFFER_TIMESTAMP_IS_VALID (buf)) { - guint64 max_diff = - gst_util_uint64_scale (enc->frame_size, GST_SECOND, enc->sample_rate); + enc = GST_OPUS_ENC (benc); + GST_DEBUG_OBJECT (enc, "handle_frame"); - if (GST_BUFFER_TIMESTAMP (buf) != enc->next_ts && - GST_BUFFER_TIMESTAMP (buf) - enc->next_ts > max_diff) { - GST_WARNING_OBJECT (enc, - "Discontinuity detected: %" G_GUINT64_FORMAT " > %" G_GUINT64_FORMAT, - GST_BUFFER_TIMESTAMP (buf) - enc->next_ts, max_diff); + if (!enc->header_sent) { + /* Opus streams in Ogg begin with two headers; the initial header (with + most of the codec setup parameters) which is mandated by the Ogg + bitstream spec. The second header holds any comment fields. */ + GstBuffer *buf1, *buf2; + GstCaps *caps; - gst_opus_enc_encode (enc, TRUE); + /* create header buffers */ + buf1 = gst_opus_enc_create_id_buffer (enc); + buf2 = gst_opus_enc_create_metadata_buffer (enc); - enc->frameno_out = 0; - enc->start_ts = GST_BUFFER_TIMESTAMP (buf); - } - } + /* mark and put on caps */ + caps = + gst_caps_new_simple ("audio/x-opus", "rate", G_TYPE_INT, + enc->sample_rate, "channels", G_TYPE_INT, enc->n_channels, "frame-size", + G_TYPE_INT, enc->frame_size, NULL); + caps = _gst_caps_set_buffer_array (caps, "streamheader", buf1, buf2, NULL); - 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; + /* negotiate with these caps */ + GST_DEBUG_OBJECT (enc, "here are the caps: %" GST_PTR_FORMAT, caps); - /* push buffer to adapter */ - gst_adapter_push (enc->adapter, buf); - buf = NULL; + gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc), caps); - ret = gst_opus_enc_encode (enc, FALSE); + /* push out buffers */ + /* store buffers for later pre_push sending */ + g_slist_foreach (enc->headers, (GFunc) gst_buffer_unref, NULL); + enc->headers = NULL; + GST_DEBUG_OBJECT (enc, "storing header buffers"); + enc->headers = g_slist_prepend (enc->headers, buf2); + enc->headers = g_slist_prepend (enc->headers, buf1); + enc->header_sent = TRUE; + } -done: + GST_DEBUG_OBJECT (enc, "received buffer %p of %u bytes", buf, + buf ? GST_BUFFER_SIZE (buf) : 0); - if (buf) - gst_buffer_unref (buf); + ret = gst_opus_enc_encode (enc, buf); return ret; - - /* ERRORS */ -not_setup: - { - GST_ELEMENT_ERROR (enc, CORE, NEGOTIATION, (NULL), - ("encoder not initialized (input is not audio?)")); - ret = GST_FLOW_NOT_NEGOTIATED; - goto done; - } - } - static void gst_opus_enc_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) @@ -1082,49 +813,3 @@ gst_opus_enc_set_property (GObject * object, guint prop_id, break; } } - -static GstStateChangeReturn -gst_opus_enc_change_state (GstElement * element, GstStateChange transition) -{ - GstOpusEnc *enc = GST_OPUS_ENC (element); - GstStateChangeReturn res; - - switch (transition) { - case GST_STATE_CHANGE_NULL_TO_READY: - break; - case GST_STATE_CHANGE_READY_TO_PAUSED: - enc->frameno = 0; - enc->samples_in = 0; - enc->frameno_out = 0; - enc->start_ts = GST_CLOCK_TIME_NONE; - enc->next_ts = GST_CLOCK_TIME_NONE; - break; - case GST_STATE_CHANGE_PAUSED_TO_PLAYING: - /* fall through */ - default: - break; - } - - res = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); - if (res == GST_STATE_CHANGE_FAILURE) - return res; - - switch (transition) { - case GST_STATE_CHANGE_PLAYING_TO_PAUSED: - break; - case GST_STATE_CHANGE_PAUSED_TO_READY: - enc->setup = FALSE; - enc->header_sent = FALSE; - if (enc->state) { - opus_encoder_destroy (enc->state); - enc->state = NULL; - } - break; - case GST_STATE_CHANGE_READY_TO_NULL: - gst_tag_setter_reset_tags (GST_TAG_SETTER (enc)); - default: - break; - } - - return res; -} diff --git a/ext/opus/gstopusenc.h b/ext/opus/gstopusenc.h index 5cb5459..fe7b94e 100644 --- a/ext/opus/gstopusenc.h +++ b/ext/opus/gstopusenc.h @@ -24,7 +24,7 @@ #include -#include +#include #include @@ -48,16 +48,9 @@ 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 +64,20 @@ 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; - - guint64 samples_in; - guint64 bytes_out; - guint64 frameno; - guint64 frameno_out; + GSList *headers; - 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); -- 2.7.4