static GstStateChangeReturn
opus_dec_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret;
GstOpusDec *dec = GST_OPUS_DEC (element);
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;
}
ret = parent_class->change_state (element, transition);
if (ret != GST_STATE_CHANGE_SUCCESS)
return ret;
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;
}
return ret;
}
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;
gst_opus_dec_reset (dec);
}
static gboolean
gst_opus_dec_stop (GstAudioDecoder * dec)
{
GstOpusDec *odec = GST_OPUS_DEC (dec);
gst_opus_dec_reset (odec);
return TRUE;
}
static void
gst_opus_dec_init (GstOpusDec * dec)
{
dec->use_inband_fec = FALSE;
dec->apply_gain = DEFAULT_APPLY_GAIN;
gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (dec), TRUE);
gst_opus_dec_reset (dec);
}
static void
gst_opus_dec_init (GstOpusDec * dec)
{
dec->sample_rate = 0;
dec->n_channels = 0;
dec->use_inband_fec = FALSE;
dec->apply_gain = DEFAULT_APPLY_GAIN;
gst_opus_dec_reset (dec);
}
static void
gst_opus_dec_init (GstOpusDec * dec)
{
dec->use_inband_fec = FALSE;
dec->apply_gain = DEFAULT_APPLY_GAIN;
gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (dec), TRUE);
gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
(dec), TRUE);
GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (dec));
gst_opus_dec_reset (dec);
}
static gboolean
gst_opus_dec_start (GstAudioDecoder * dec)
{
GstOpusDec *odec = GST_OPUS_DEC (dec);
gst_opus_dec_reset (odec);
/* we know about concealment */
gst_audio_decoder_set_plc_aware (dec, TRUE);
if (odec->use_inband_fec) {
gst_audio_decoder_set_latency (dec, 2 * GST_MSECOND + GST_MSECOND / 2,
120 * GST_MSECOND);
}
return TRUE;
}
static gboolean
gst_opus_dec_start (GstAudioDecoder * dec)
{
GstOpusDec *odec = GST_OPUS_DEC (dec);
gst_opus_dec_reset (odec);
/* we know about concealment */
gst_audio_decoder_set_plc_aware (dec, TRUE);
if (odec->use_inband_fec) {
/* opusdec outputs samples directly from an input buffer, except if
* FEC is on, in which case it buffers one buffer in case one buffer
* goes missing.
*/
gst_audio_decoder_set_latency (dec, 120 * GST_MSECOND, 120 * GST_MSECOND);
}
return TRUE;
}
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;
}
