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;
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);
}
}
done:
return ret;
}
static GstFlowReturn
gst_opus_dec_handle_frame (GstAudioDecoder * adec, GstBuffer * buf)
{
GstFlowReturn res;
GstOpusDec *dec;
/* no fancy draining */
if (G_UNLIKELY (!buf))
return GST_FLOW_OK;
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)));
/* 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);
}
} else {
/* Otherwise fall back to packet counting and assume that the
* first two packets might be the headers, checking magic. */
switch (dec->packetno) {
case 0:
if (gst_opus_header_is_header (buf, "OpusHead", 8)) {
GST_DEBUG_OBJECT (dec, "found streamheader");
res = gst_opus_dec_parse_header (dec, buf);
gst_audio_decoder_finish_frame (adec, NULL, 1);
} else {
res = opus_dec_chain_parse_data (dec, buf);
}
break;
case 1:
if (gst_opus_header_is_header (buf, "OpusTags", 8)) {
GST_DEBUG_OBJECT (dec, "counted vorbiscomments");
res = gst_opus_dec_parse_comments (dec, buf);
gst_audio_decoder_finish_frame (adec, NULL, 1);
} else {
res = opus_dec_chain_parse_data (dec, buf);
}
break;
default:
{
res = opus_dec_chain_parse_data (dec, buf);
break;
}
}
}
dec->packetno++;
return res;
}
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;
GstCaps *old_caps;
GST_DEBUG_OBJECT (dec, "set_format: %" GST_PTR_FORMAT, caps);
if ((old_caps = gst_pad_get_current_caps (GST_AUDIO_DECODER_SINK_PAD (bdec)))) {
if (gst_caps_is_equal (caps, old_caps)) {
gst_caps_unref (old_caps);
GST_DEBUG_OBJECT (dec, "caps didn't change");
goto done;
}
GST_DEBUG_OBJECT (dec, "caps have changed, resetting decoder");
gst_opus_dec_reset (dec);
gst_caps_unref (old_caps);
}
s = gst_caps_get_structure (caps, 0);
if ((streamheader = gst_structure_get_value (s, "streamheader")) &&
G_VALUE_HOLDS (streamheader, GST_TYPE_ARRAY) &&
gst_value_array_get_size (streamheader) >= 2) {
const GValue *header, *vorbiscomment;
GstBuffer *buf;
GstFlowReturn res = GST_FLOW_OK;
header = gst_value_array_get_value (streamheader, 0);
if (header && G_VALUE_HOLDS (header, GST_TYPE_BUFFER)) {
buf = gst_value_get_buffer (header);
res = gst_opus_dec_parse_header (dec, buf);
if (res != GST_FLOW_OK)
goto done;
gst_buffer_replace (&dec->streamheader, buf);
}
vorbiscomment = gst_value_array_get_value (streamheader, 1);
if (vorbiscomment && G_VALUE_HOLDS (vorbiscomment, GST_TYPE_BUFFER)) {
buf = gst_value_get_buffer (vorbiscomment);
res = gst_opus_dec_parse_comments (dec, buf);
if (res != GST_FLOW_OK)
goto done;
gst_buffer_replace (&dec->vorbiscomment, buf);
}
} else {
/* defaults if not in the caps */
dec->n_channels = 2;
dec->sample_rate = 48000;
gst_structure_get_int (s, "channels", &dec->n_channels);
gst_structure_get_int (s, "rate", &dec->sample_rate);
/* default stereo mapping */
dec->channel_mapping_family = 0;
dec->channel_mapping[0] = 0;
dec->channel_mapping[1] = 1;
dec->n_streams = 1;
dec->n_stereo_streams = 1;
gst_opus_dec_negotiate (dec, NULL);
}
done:
return ret;
}
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;
GstCaps *old_caps;
GST_DEBUG_OBJECT (dec, "set_format: %" GST_PTR_FORMAT, caps);
if ((old_caps = gst_pad_get_current_caps (GST_AUDIO_DECODER_SINK_PAD (bdec)))) {
if (gst_caps_is_equal (caps, old_caps)) {
gst_caps_unref (old_caps);
GST_DEBUG_OBJECT (dec, "caps didn't change");
goto done;
}
GST_DEBUG_OBJECT (dec, "caps have changed, resetting decoder");
gst_opus_dec_reset (dec);
gst_caps_unref (old_caps);
}
s = gst_caps_get_structure (caps, 0);
if ((streamheader = gst_structure_get_value (s, "streamheader")) &&
G_VALUE_HOLDS (streamheader, GST_TYPE_ARRAY) &&
gst_value_array_get_size (streamheader) >= 2) {
const GValue *header, *vorbiscomment;
GstBuffer *buf;
GstFlowReturn res = GST_FLOW_OK;
header = gst_value_array_get_value (streamheader, 0);
if (header && G_VALUE_HOLDS (header, GST_TYPE_BUFFER)) {
buf = gst_value_get_buffer (header);
res = gst_opus_dec_parse_header (dec, buf);
if (res != GST_FLOW_OK) {
ret = FALSE;
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) {
ret = FALSE;
goto done;
}
gst_buffer_replace (&dec->vorbiscomment, buf);
}
} else {
const GstAudioChannelPosition *posn = NULL;
if (!gst_codec_utils_opus_parse_caps (caps, &dec->sample_rate,
&dec->n_channels, &dec->channel_mapping_family, &dec->n_streams,
&dec->n_stereo_streams, dec->channel_mapping)) {
ret = FALSE;
goto done;
}
if (dec->channel_mapping_family == 1 && dec->n_channels <= 8)
posn = gst_opus_channel_positions[dec->n_channels - 1];
if (!gst_opus_dec_negotiate (dec, posn))
return FALSE;
}
done:
return ret;
}
