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)
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");
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;
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;
}
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,
gint16 *out_data;
int n, err;
- if (timestamp != -1) {
- dec->segment.last_stop = timestamp;
- dec->granulepos = -1;
- }
-
if (dec->state == NULL) {
GstCaps *caps;
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);
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));
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) {
}
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,
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));
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;
}
#include <gst/gsttagsetter.h>
#include <gst/tag/tag.h>
+#include <gst/base/gstbytewriter.h>
#include <gst/audio/audio.h>
#include "gstopusenc.h"
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)
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)
{
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,
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
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)
{
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)
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,
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;
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;
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)
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;
-}
projects / platform / upstream / gstreamer.git / commitdiff