static GstFlowReturn
opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buffer)
{
GstFlowReturn res = GST_FLOW_OK;
gsize size;
guint8 *data;
GstBuffer *outbuf;
gint16 *out_data;
int n, err;
int samples;
unsigned int packet_size;
GstBuffer *buf;
GstMapInfo map, omap;
if (dec->state == NULL) {
/* If we did not get any headers, default to 2 channels */
if (dec->n_channels == 0) {
GST_INFO_OBJECT (dec, "No header, assuming single stream");
dec->n_channels = 2;
dec->sample_rate = 48000;
/* 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);
}
GST_DEBUG_OBJECT (dec, "Creating decoder with %d channels, %d Hz",
dec->n_channels, dec->sample_rate);
#ifndef GST_DISABLE_GST_DEBUG
gst_opus_common_log_channel_mapping_table (GST_ELEMENT (dec), opusdec_debug,
"Mapping table", dec->n_channels, dec->channel_mapping);
#endif
GST_DEBUG_OBJECT (dec, "%d streams, %d stereo", dec->n_streams,
dec->n_stereo_streams);
dec->state =
opus_multistream_decoder_create (dec->sample_rate, dec->n_channels,
dec->n_streams, dec->n_stereo_streams, dec->channel_mapping, &err);
if (!dec->state || err != OPUS_OK)
goto creation_failed;
}
if (buffer) {
GST_DEBUG_OBJECT (dec, "Received buffer of size %" G_GSIZE_FORMAT,
gst_buffer_get_size (buffer));
} else {
GST_DEBUG_OBJECT (dec, "Received missing buffer");
}
/* if using in-band FEC, we introdude one extra frame's delay as we need
to potentially wait for next buffer to decode a missing buffer */
if (dec->use_inband_fec && !dec->primed) {
GST_DEBUG_OBJECT (dec, "First buffer received in FEC mode, early out");
gst_buffer_replace (&dec->last_buffer, buffer);
dec->primed = TRUE;
goto done;
}
/* That's the buffer we'll be sending to the opus decoder. */
buf = (dec->use_inband_fec
&& gst_buffer_get_size (dec->last_buffer) >
0) ? dec->last_buffer : buffer;
if (buf && gst_buffer_get_size (buf) > 0) {
gst_buffer_map (buf, &map, GST_MAP_READ);
data = map.data;
size = map.size;
GST_DEBUG_OBJECT (dec, "Using buffer of size %" G_GSIZE_FORMAT, size);
} else {
/* concealment data, pass NULL as the bits parameters */
GST_DEBUG_OBJECT (dec, "Using NULL buffer");
data = NULL;
size = 0;
}
/* use maximum size (120 ms) as the number of returned samples is
not constant over the stream. */
samples = 120 * dec->sample_rate / 1000;
packet_size = samples * dec->n_channels * 2;
outbuf =
gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (dec),
packet_size);
if (!outbuf) {
goto buffer_failed;
}
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
out_data = (gint16 *) omap.data;
if (dec->use_inband_fec) {
if (dec->last_buffer) {
/* normal delayed decode */
GST_LOG_OBJECT (dec, "FEC enabled, decoding last delayed buffer");
n = opus_multistream_decode (dec->state, data, size, out_data, samples,
0);
} else {
/* FEC reconstruction decode */
GST_LOG_OBJECT (dec, "FEC enabled, reconstructing last buffer");
n = opus_multistream_decode (dec->state, data, size, out_data, samples,
1);
}
} else {
/* normal decode */
GST_LOG_OBJECT (dec, "FEC disabled, decoding buffer");
n = opus_multistream_decode (dec->state, data, size, out_data, samples, 0);
}
gst_buffer_unmap (outbuf, &omap);
if (data != NULL)
gst_buffer_unmap (buf, &map);
if (n < 0) {
GST_ELEMENT_ERROR (dec, STREAM, DECODE, ("Decoding error: %d", n), (NULL));
gst_buffer_unref (outbuf);
return GST_FLOW_ERROR;
}
GST_DEBUG_OBJECT (dec, "decoded %d samples", n);
gst_buffer_set_size (outbuf, n * 2 * dec->n_channels);
/* Skip any samples that need skipping */
if (dec->pre_skip > 0) {
guint scaled_pre_skip = dec->pre_skip * dec->sample_rate / 48000;
guint skip = scaled_pre_skip > n ? n : scaled_pre_skip;
guint scaled_skip = skip * 48000 / dec->sample_rate;
gst_buffer_resize (outbuf, skip * 2 * dec->n_channels, -1);
dec->pre_skip -= scaled_skip;
GST_INFO_OBJECT (dec,
"Skipping %u samples (%u at 48000 Hz, %u left to skip)", skip,
scaled_skip, dec->pre_skip);
}
if (gst_buffer_get_size (outbuf) == 0) {
gst_buffer_unref (outbuf);
outbuf = NULL;
} else if (dec->opus_pos[0] != GST_AUDIO_CHANNEL_POSITION_INVALID) {
gst_audio_buffer_reorder_channels (outbuf, GST_AUDIO_FORMAT_S16,
dec->n_channels, dec->opus_pos, dec->info.position);
}
/* Apply gain */
/* Would be better off leaving this to a volume element, as this is
a naive conversion that does too many int/float conversions.
However, we don't have control over the pipeline...
So make it optional if the user program wants to use a volume,
but do it by default so the correct volume goes out by default */
if (dec->apply_gain && outbuf && dec->r128_gain) {
gsize rsize;
unsigned int i, nsamples;
double volume = dec->r128_gain_volume;
gint16 *samples;
gst_buffer_map (outbuf, &omap, GST_MAP_READWRITE);
samples = (gint16 *) omap.data;
rsize = omap.size;
GST_DEBUG_OBJECT (dec, "Applying gain: volume %f", volume);
nsamples = rsize / 2;
for (i = 0; i < nsamples; ++i) {
int sample = (int) (samples[i] * volume + 0.5);
samples[i] = sample < -32768 ? -32768 : sample > 32767 ? 32767 : sample;
}
gst_buffer_unmap (outbuf, &omap);
}
if (dec->use_inband_fec) {
gst_buffer_replace (&dec->last_buffer, buffer);
}
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));
done:
return res;
creation_failed:
GST_ERROR_OBJECT (dec, "Failed to create Opus decoder: %d", err);
return GST_FLOW_ERROR;
buffer_failed:
GST_ERROR_OBJECT (dec, "Failed to create %u byte buffer", packet_size);
return GST_FLOW_ERROR;
}
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 GstFlowReturn
gst_opus_dec_parse_header (GstOpusDec * dec, GstBuffer * buf)
{
const guint8 *data;
GstAudioChannelPosition pos[64];
const GstAudioChannelPosition *posn = NULL;
GstMapInfo map;
if (!gst_opus_header_is_id_header (buf)) {
GST_ERROR_OBJECT (dec, "Header is not an Opus ID header");
return GST_FLOW_ERROR;
}
gst_buffer_map (buf, &map, GST_MAP_READ);
data = map.data;
if (!(dec->n_channels == 0 || dec->n_channels == data[9])) {
gst_buffer_unmap (buf, &map);
GST_ERROR_OBJECT (dec, "Opus ID header has invalid channels");
return GST_FLOW_ERROR;
}
dec->n_channels = data[9];
dec->sample_rate = GST_READ_UINT32_LE (data + 12);
dec->pre_skip = GST_READ_UINT16_LE (data + 10);
dec->r128_gain = GST_READ_UINT16_LE (data + 16);
dec->r128_gain_volume = gst_opus_dec_get_r128_volume (dec->r128_gain);
GST_INFO_OBJECT (dec,
"Found pre-skip of %u samples, R128 gain %d (volume %f)",
dec->pre_skip, dec->r128_gain, dec->r128_gain_volume);
dec->channel_mapping_family = data[18];
if (dec->channel_mapping_family == 0) {
/* implicit mapping */
GST_INFO_OBJECT (dec, "Channel mapping family 0, implicit mapping");
dec->n_streams = dec->n_stereo_streams = 1;
dec->channel_mapping[0] = 0;
dec->channel_mapping[1] = 1;
} else {
dec->n_streams = data[19];
dec->n_stereo_streams = data[20];
memcpy (dec->channel_mapping, data + 21, dec->n_channels);
if (dec->channel_mapping_family == 1) {
GST_INFO_OBJECT (dec, "Channel mapping family 1, Vorbis mapping");
switch (dec->n_channels) {
case 1:
case 2:
/* nothing */
break;
case 3:
case 4:
case 5:
case 6:
case 7:
case 8:
posn = gst_opus_channel_positions[dec->n_channels - 1];
break;
default:{
gint i;
GST_ELEMENT_WARNING (GST_ELEMENT (dec), STREAM, DECODE,
(NULL), ("Using NONE channel layout for more than 8 channels"));
for (i = 0; i < dec->n_channels; i++)
pos[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
posn = pos;
}
}
} else {
GST_INFO_OBJECT (dec, "Channel mapping family %d",
dec->channel_mapping_family);
}
}
gst_opus_dec_negotiate (dec, posn);
gst_buffer_unmap (buf, &map);
return GST_FLOW_OK;
}
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;
}
static GstFlowReturn
opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buffer)
{
GstFlowReturn res = GST_FLOW_OK;
gsize size;
guint8 *data;
GstBuffer *outbuf, *bufd;
gint16 *out_data;
int n, err;
int samples;
unsigned int packet_size;
GstBuffer *buf;
GstMapInfo map, omap;
GstAudioClippingMeta *cmeta = NULL;
if (dec->state == NULL) {
/* If we did not get any headers, default to 2 channels */
if (dec->n_channels == 0) {
GST_INFO_OBJECT (dec, "No header, assuming single stream");
dec->n_channels = 2;
dec->sample_rate = 48000;
/* 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;
if (!gst_opus_dec_negotiate (dec, NULL))
return GST_FLOW_NOT_NEGOTIATED;
}
if (dec->n_channels == 2 && dec->n_streams == 1
&& dec->n_stereo_streams == 0) {
/* if we are automatically decoding 2 channels, but only have
a single encoded one, direct both channels to it */
dec->channel_mapping[1] = 0;
}
GST_DEBUG_OBJECT (dec, "Creating decoder with %d channels, %d Hz",
dec->n_channels, dec->sample_rate);
#ifndef GST_DISABLE_GST_DEBUG
gst_opus_common_log_channel_mapping_table (GST_ELEMENT (dec), opusdec_debug,
"Mapping table", dec->n_channels, dec->channel_mapping);
#endif
GST_DEBUG_OBJECT (dec, "%d streams, %d stereo", dec->n_streams,
dec->n_stereo_streams);
dec->state =
opus_multistream_decoder_create (dec->sample_rate, dec->n_channels,
dec->n_streams, dec->n_stereo_streams, dec->channel_mapping, &err);
if (!dec->state || err != OPUS_OK)
goto creation_failed;
}
if (buffer) {
GST_DEBUG_OBJECT (dec, "Received buffer of size %" G_GSIZE_FORMAT,
gst_buffer_get_size (buffer));
} else {
GST_DEBUG_OBJECT (dec, "Received missing buffer");
}
/* if using in-band FEC, we introdude one extra frame's delay as we need
to potentially wait for next buffer to decode a missing buffer */
if (dec->use_inband_fec && !dec->primed) {
GST_DEBUG_OBJECT (dec, "First buffer received in FEC mode, early out");
gst_buffer_replace (&dec->last_buffer, buffer);
dec->primed = TRUE;
goto done;
}
/* That's the buffer we'll be sending to the opus decoder. */
buf = (dec->use_inband_fec
&& gst_buffer_get_size (dec->last_buffer) >
0) ? dec->last_buffer : buffer;
/* That's the buffer we get duration from */
bufd = dec->use_inband_fec ? dec->last_buffer : buffer;
if (buf && gst_buffer_get_size (buf) > 0) {
gst_buffer_map (buf, &map, GST_MAP_READ);
data = map.data;
size = map.size;
GST_DEBUG_OBJECT (dec, "Using buffer of size %" G_GSIZE_FORMAT, size);
} else {
/* concealment data, pass NULL as the bits parameters */
GST_DEBUG_OBJECT (dec, "Using NULL buffer");
data = NULL;
size = 0;
}
if (gst_buffer_get_size (bufd) == 0) {
GstClockTime const opus_plc_alignment = 2500 * GST_USECOND;
GstClockTime aligned_missing_duration;
GstClockTime missing_duration = GST_BUFFER_DURATION (bufd);
if (!GST_CLOCK_TIME_IS_VALID (missing_duration) || missing_duration == 0) {
if (GST_CLOCK_TIME_IS_VALID (dec->last_known_buffer_duration)) {
missing_duration = dec->last_known_buffer_duration;
GST_WARNING_OBJECT (dec,
"Missing duration, using last duration %" GST_TIME_FORMAT,
GST_TIME_ARGS (missing_duration));
} else {
GST_WARNING_OBJECT (dec,
"Missing buffer, but unknown duration, and no previously known duration, assuming 20 ms");
missing_duration = 20 * GST_MSECOND;
}
}
GST_DEBUG_OBJECT (dec,
"missing buffer, doing PLC duration %" GST_TIME_FORMAT
" plus leftover %" GST_TIME_FORMAT, GST_TIME_ARGS (missing_duration),
GST_TIME_ARGS (dec->leftover_plc_duration));
/* add the leftover PLC duration to that of the buffer */
missing_duration += dec->leftover_plc_duration;
/* align the combined buffer and leftover PLC duration to multiples
* of 2.5ms, rounding to nearest, and store excess duration for later */
aligned_missing_duration =
((missing_duration +
opus_plc_alignment / 2) / opus_plc_alignment) * opus_plc_alignment;
dec->leftover_plc_duration = missing_duration - aligned_missing_duration;
/* Opus' PLC cannot operate with less than 2.5ms; skip PLC
* and accumulate the missing duration in the leftover_plc_duration
* for the next PLC attempt */
if (aligned_missing_duration < opus_plc_alignment) {
GST_DEBUG_OBJECT (dec,
"current duration %" GST_TIME_FORMAT
" of missing data not enough for PLC (minimum needed: %"
GST_TIME_FORMAT ") - skipping", GST_TIME_ARGS (missing_duration),
GST_TIME_ARGS (opus_plc_alignment));
goto done;
}
/* convert the duration (in nanoseconds) to sample count */
samples =
gst_util_uint64_scale_int (aligned_missing_duration, dec->sample_rate,
GST_SECOND);
GST_DEBUG_OBJECT (dec,
"calculated PLC frame length: %" GST_TIME_FORMAT
" num frame samples: %d new leftover: %" GST_TIME_FORMAT,
GST_TIME_ARGS (aligned_missing_duration), samples,
GST_TIME_ARGS (dec->leftover_plc_duration));
} else {
/* use maximum size (120 ms) as the number of returned samples is
not constant over the stream. */
samples = 120 * dec->sample_rate / 1000;
}
packet_size = samples * dec->n_channels * 2;
outbuf =
gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (dec),
packet_size);
if (!outbuf) {
goto buffer_failed;
}
if (size > 0)
dec->last_known_buffer_duration = packet_duration_opus (data, size);
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
out_data = (gint16 *) omap.data;
do {
if (dec->use_inband_fec) {
if (gst_buffer_get_size (dec->last_buffer) > 0) {
/* normal delayed decode */
GST_LOG_OBJECT (dec, "FEC enabled, decoding last delayed buffer");
n = opus_multistream_decode (dec->state, data, size, out_data, samples,
0);
} else {
/* FEC reconstruction decode */
GST_LOG_OBJECT (dec, "FEC enabled, reconstructing last buffer");
n = opus_multistream_decode (dec->state, data, size, out_data, samples,
1);
}
} else {
/* normal decode */
GST_LOG_OBJECT (dec, "FEC disabled, decoding buffer");
n = opus_multistream_decode (dec->state, data, size, out_data, samples,
0);
}
if (n == OPUS_BUFFER_TOO_SMALL) {
/* if too small, add 2.5 milliseconds and try again, up to the
* Opus max size of 120 milliseconds */
if (samples >= 120 * dec->sample_rate / 1000)
break;
samples += 25 * dec->sample_rate / 10000;
packet_size = samples * dec->n_channels * 2;
gst_buffer_unmap (outbuf, &omap);
gst_buffer_unref (outbuf);
outbuf =
gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (dec),
packet_size);
if (!outbuf) {
goto buffer_failed;
}
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
out_data = (gint16 *) omap.data;
}
} while (n == OPUS_BUFFER_TOO_SMALL);
gst_buffer_unmap (outbuf, &omap);
if (data != NULL)
gst_buffer_unmap (buf, &map);
if (n < 0) {
GstFlowReturn ret = GST_FLOW_ERROR;
gst_buffer_unref (outbuf);
GST_AUDIO_DECODER_ERROR (dec, 1, STREAM, DECODE, (NULL),
("Decoding error (%d): %s", n, opus_strerror (n)), ret);
return ret;
}
GST_DEBUG_OBJECT (dec, "decoded %d samples", n);
gst_buffer_set_size (outbuf, n * 2 * dec->n_channels);
GST_BUFFER_DURATION (outbuf) = samples * GST_SECOND / dec->sample_rate;
samples = n;
cmeta = gst_buffer_get_audio_clipping_meta (buf);
g_assert (!cmeta || cmeta->format == GST_FORMAT_DEFAULT);
/* Skip any samples that need skipping */
if (cmeta && cmeta->start) {
guint pre_skip = cmeta->start;
guint scaled_pre_skip = pre_skip * dec->sample_rate / 48000;
guint skip = scaled_pre_skip > n ? n : scaled_pre_skip;
guint scaled_skip = skip * 48000 / dec->sample_rate;
gst_buffer_resize (outbuf, skip * 2 * dec->n_channels, -1);
GST_INFO_OBJECT (dec,
"Skipping %u samples at the beginning (%u at 48000 Hz)",
skip, scaled_skip);
}
if (cmeta && cmeta->end) {
guint post_skip = cmeta->end;
guint scaled_post_skip = post_skip * dec->sample_rate / 48000;
guint skip = scaled_post_skip > n ? n : scaled_post_skip;
guint scaled_skip = skip * 48000 / dec->sample_rate;
guint outsize = gst_buffer_get_size (outbuf);
guint skip_bytes = skip * 2 * dec->n_channels;
if (outsize > skip_bytes)
outsize -= skip_bytes;
else
outsize = 0;
gst_buffer_resize (outbuf, 0, outsize);
GST_INFO_OBJECT (dec,
"Skipping %u samples at the end (%u at 48000 Hz)", skip, scaled_skip);
}
if (gst_buffer_get_size (outbuf) == 0) {
gst_buffer_unref (outbuf);
outbuf = NULL;
} else if (dec->opus_pos[0] != GST_AUDIO_CHANNEL_POSITION_INVALID) {
gst_audio_buffer_reorder_channels (outbuf, GST_AUDIO_FORMAT_S16,
dec->n_channels, dec->opus_pos, dec->info.position);
}
/* Apply gain */
/* Would be better off leaving this to a volume element, as this is
a naive conversion that does too many int/float conversions.
However, we don't have control over the pipeline...
So make it optional if the user program wants to use a volume,
but do it by default so the correct volume goes out by default */
if (dec->apply_gain && outbuf && dec->r128_gain) {
gsize rsize;
unsigned int i, nsamples;
double volume = dec->r128_gain_volume;
gint16 *samples;
gst_buffer_map (outbuf, &omap, GST_MAP_READWRITE);
samples = (gint16 *) omap.data;
rsize = omap.size;
GST_DEBUG_OBJECT (dec, "Applying gain: volume %f", volume);
nsamples = rsize / 2;
for (i = 0; i < nsamples; ++i) {
int sample = (int) (samples[i] * volume + 0.5);
samples[i] = sample < -32768 ? -32768 : sample > 32767 ? 32767 : sample;
}
gst_buffer_unmap (outbuf, &omap);
}
if (dec->use_inband_fec) {
gst_buffer_replace (&dec->last_buffer, buffer);
}
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));
done:
return res;
creation_failed:
GST_ELEMENT_ERROR (dec, LIBRARY, INIT, ("Failed to create Opus decoder"),
("Failed to create Opus decoder (%d): %s", err, opus_strerror (err)));
return GST_FLOW_ERROR;
buffer_failed:
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),
("Failed to create %u byte buffer", packet_size));
return GST_FLOW_ERROR;
}
static GstFlowReturn
gst_opus_dec_parse_header (GstOpusDec * dec, GstBuffer * buf)
{
GstAudioChannelPosition pos[64];
const GstAudioChannelPosition *posn = NULL;
if (!gst_opus_header_is_id_header (buf)) {
GST_ELEMENT_ERROR (dec, STREAM, FORMAT, (NULL),
("Header is not an Opus ID header"));
return GST_FLOW_ERROR;
}
if (!gst_codec_utils_opus_parse_header (buf,
&dec->sample_rate,
&dec->n_channels,
&dec->channel_mapping_family,
&dec->n_streams,
&dec->n_stereo_streams,
dec->channel_mapping, &dec->pre_skip, &dec->r128_gain)) {
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),
("Failed to parse Opus ID header"));
return GST_FLOW_ERROR;
}
dec->r128_gain_volume = gst_opus_dec_get_r128_volume (dec->r128_gain);
GST_INFO_OBJECT (dec,
"Found pre-skip of %u samples, R128 gain %d (volume %f)",
dec->pre_skip, dec->r128_gain, dec->r128_gain_volume);
if (dec->channel_mapping_family == 1) {
GST_INFO_OBJECT (dec, "Channel mapping family 1, Vorbis mapping");
switch (dec->n_channels) {
case 1:
case 2:
/* nothing */
break;
case 3:
case 4:
case 5:
case 6:
case 7:
case 8:
posn = gst_opus_channel_positions[dec->n_channels - 1];
break;
default:{
gint i;
GST_ELEMENT_WARNING (GST_ELEMENT (dec), STREAM, DECODE,
(NULL), ("Using NONE channel layout for more than 8 channels"));
for (i = 0; i < dec->n_channels; i++)
pos[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
posn = pos;
}
}
} else {
GST_INFO_OBJECT (dec, "Channel mapping family %d",
dec->channel_mapping_family);
}
if (!gst_opus_dec_negotiate (dec, posn))
return GST_FLOW_NOT_NEGOTIATED;
return GST_FLOW_OK;
}
static GstFlowReturn
gst_opus_dec_parse_header (GstOpusDec * dec, GstBuffer * buf)
{
const guint8 *data = GST_BUFFER_DATA (buf);
GstCaps *caps;
const GstAudioChannelPosition *pos = NULL;
if (!gst_opus_header_is_id_header (buf)) {
GST_ERROR_OBJECT (dec, "Header is not an Opus ID header");
return GST_FLOW_ERROR;
}
if (!(dec->n_channels == 0 || dec->n_channels == data[9])) {
GST_ERROR_OBJECT (dec, "Opus ID header has invalid channels");
return GST_FLOW_ERROR;
}
dec->n_channels = data[9];
dec->pre_skip = GST_READ_UINT16_LE (data + 10);
dec->r128_gain = GST_READ_UINT16_LE (data + 16);
dec->r128_gain_volume = gst_opus_dec_get_r128_volume (dec->r128_gain);
GST_INFO_OBJECT (dec,
"Found pre-skip of %u samples, R128 gain %d (volume %f)",
dec->pre_skip, dec->r128_gain, dec->r128_gain_volume);
dec->channel_mapping_family = data[18];
if (dec->channel_mapping_family == 0) {
/* implicit mapping */
GST_INFO_OBJECT (dec, "Channel mapping family 0, implicit mapping");
dec->n_streams = dec->n_stereo_streams = 1;
dec->channel_mapping[0] = 0;
dec->channel_mapping[1] = 1;
} else {
dec->n_streams = data[19];
dec->n_stereo_streams = data[20];
memcpy (dec->channel_mapping, data + 21, dec->n_channels);
if (dec->channel_mapping_family == 1) {
GST_INFO_OBJECT (dec, "Channel mapping family 1, Vorbis mapping");
switch (dec->n_channels) {
case 1:
case 2:
/* nothing */
break;
case 3:
case 4:
case 5:
case 6:
case 7:
case 8:
pos = gst_opus_channel_positions[dec->n_channels - 1];
break;
default:{
gint i;
GstAudioChannelPosition *posn =
g_new (GstAudioChannelPosition, dec->n_channels);
GST_ELEMENT_WARNING (GST_ELEMENT (dec), STREAM, DECODE,
(NULL), ("Using NONE channel layout for more than 8 channels"));
for (i = 0; i < dec->n_channels; i++)
posn[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
pos = posn;
}
}
} else {
GST_INFO_OBJECT (dec, "Channel mapping family %d",
dec->channel_mapping_family);
}
}
caps = gst_opus_dec_negotiate (dec);
if (pos) {
GST_DEBUG_OBJECT (dec, "Setting channel positions on caps");
gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
}
if (dec->n_channels > 8) {
g_free ((GstAudioChannelPosition *) pos);
}
GST_INFO_OBJECT (dec, "Setting src caps to %" GST_PTR_FORMAT, caps);
gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps);
gst_caps_unref (caps);
return GST_FLOW_OK;
}
