static GstBuffer *
gst_rtp_L16_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
{
GstRtpL16Depay *rtpL16depay;
GstBuffer *outbuf;
gint payload_len;
gboolean marker;
GstRTPBuffer rtp = { NULL };
rtpL16depay = GST_RTP_L16_DEPAY (depayload);
gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
payload_len = gst_rtp_buffer_get_payload_len (&rtp);
if (payload_len <= 0)
goto empty_packet;
GST_DEBUG_OBJECT (rtpL16depay, "got payload of %d bytes", payload_len);
outbuf = gst_rtp_buffer_get_payload_buffer (&rtp);
marker = gst_rtp_buffer_get_marker (&rtp);
if (marker) {
/* mark talk spurt with DISCONT */
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
}
outbuf = gst_buffer_make_writable (outbuf);
if (rtpL16depay->order &&
!gst_audio_buffer_reorder_channels (outbuf,
rtpL16depay->info.finfo->format, rtpL16depay->info.channels,
rtpL16depay->info.position, rtpL16depay->order->pos)) {
goto reorder_failed;
}
gst_rtp_buffer_unmap (&rtp);
return outbuf;
/* ERRORS */
empty_packet:
{
GST_ELEMENT_WARNING (rtpL16depay, STREAM, DECODE,
("Empty Payload."), (NULL));
gst_rtp_buffer_unmap (&rtp);
return NULL;
}
reorder_failed:
{
GST_ELEMENT_ERROR (rtpL16depay, STREAM, DECODE,
("Channel reordering failed."), (NULL));
gst_rtp_buffer_unmap (&rtp);
return NULL;
}
}
static gboolean
gst_raw_audio_parse_process (GstRawBaseParse * raw_base_parse,
GstRawBaseParseConfig config, GstBuffer * in_data, gsize total_num_in_bytes,
gsize num_valid_in_bytes, GstBuffer ** processed_data)
{
GstRawAudioParse *raw_audio_parse = GST_RAW_AUDIO_PARSE (raw_base_parse);
GstRawAudioParseConfig *config_ptr =
gst_raw_audio_parse_get_config_ptr (raw_audio_parse, config);
if ((config_ptr->format == GST_RAW_AUDIO_PARSE_FORMAT_PCM)
&& config_ptr->needs_channel_reordering) {
/* Need to reorder samples, since they are in an invalid
* channel order. */
GstBuffer *outbuf;
GST_LOG_OBJECT (raw_audio_parse,
"using %" G_GSIZE_FORMAT " bytes out of the %" G_GSIZE_FORMAT
" bytes from the input buffer with reordering", num_valid_in_bytes,
total_num_in_bytes);
outbuf =
gst_buffer_copy_region (in_data,
GST_BUFFER_COPY_FLAGS | GST_BUFFER_COPY_TIMESTAMPS |
GST_BUFFER_COPY_META | GST_BUFFER_COPY_MEMORY, 0, num_valid_in_bytes);
gst_audio_buffer_reorder_channels (outbuf,
config_ptr->pcm_format,
config_ptr->num_channels,
config_ptr->channel_positions, config_ptr->reordered_channel_positions);
*processed_data = outbuf;
} else {
/* Nothing needs to be done with the sample data.
* Instruct the baseparse class to just take out_size bytes
* from the input buffer */
GST_LOG_OBJECT (raw_audio_parse,
"using %" G_GSIZE_FORMAT " bytes out of the %" G_GSIZE_FORMAT
" bytes from the input buffer without reordering", num_valid_in_bytes,
total_num_in_bytes);
*processed_data = NULL;
}
return TRUE;
}
static GstFlowReturn
gst_rtp_L16_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer)
{
GstRtpL16Pay *rtpL16pay;
rtpL16pay = GST_RTP_L16_PAY (basepayload);
buffer = gst_buffer_make_writable (buffer);
if (rtpL16pay->order &&
!gst_audio_buffer_reorder_channels (buffer, rtpL16pay->info.finfo->format,
rtpL16pay->info.channels, rtpL16pay->info.position,
rtpL16pay->order->pos)) {
return GST_FLOW_ERROR;
}
return GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->handle_buffer (basepayload,
buffer);
}
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 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 GstBuffer *
gst_rtp_L16_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
{
GstRtpL16Depay *rtpL16depay;
GstBuffer *outbuf;
gint payload_len;
gboolean marker;
GstAudioInfo *info;
rtpL16depay = GST_RTP_L16_DEPAY (depayload);
payload_len = gst_rtp_buffer_get_payload_len (rtp);
if (payload_len <= 0)
goto empty_packet;
GST_DEBUG_OBJECT (rtpL16depay, "got payload of %d bytes", payload_len);
outbuf = gst_rtp_buffer_get_payload_buffer (rtp);
marker = gst_rtp_buffer_get_marker (rtp);
if (marker) {
/* mark talk spurt with RESYNC */
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
}
outbuf = gst_buffer_make_writable (outbuf);
info = &rtpL16depay->info;
if (payload_len % ((info->finfo->width * info->channels) / 8) != 0)
goto wrong_payload_size;
if (rtpL16depay->order &&
!gst_audio_buffer_reorder_channels (outbuf,
info->finfo->format, info->channels,
info->position, rtpL16depay->order->pos)) {
goto reorder_failed;
}
gst_rtp_drop_meta (GST_ELEMENT_CAST (rtpL16depay), outbuf,
g_quark_from_static_string (GST_META_TAG_AUDIO_STR));
return outbuf;
/* ERRORS */
empty_packet:
{
GST_ELEMENT_WARNING (rtpL16depay, STREAM, DECODE,
("Empty Payload."), (NULL));
return NULL;
}
wrong_payload_size:
{
GST_ELEMENT_WARNING (rtpL16depay, STREAM, DECODE,
("Wrong Payload Size."), (NULL));
return NULL;
}
reorder_failed:
{
GST_ELEMENT_ERROR (rtpL16depay, STREAM, DECODE,
("Channel reordering failed."), (NULL));
return NULL;
}
}
static gint
gst_ffmpegauddec_audio_frame (GstFFMpegAudDec * ffmpegdec,
AVCodec * in_plugin, guint8 * data, guint size,
GstBuffer ** outbuf, GstFlowReturn * ret)
{
gint len = -1;
gint have_data = AVCODEC_MAX_AUDIO_FRAME_SIZE;
AVPacket packet;
AVFrame frame;
GST_DEBUG_OBJECT (ffmpegdec, "size: %d", size);
gst_avpacket_init (&packet, data, size);
memset (&frame, 0, sizeof (frame));
avcodec_get_frame_defaults (&frame);
len = avcodec_decode_audio4 (ffmpegdec->context, &frame, &have_data, &packet);
GST_DEBUG_OBJECT (ffmpegdec,
"Decode audio: len=%d, have_data=%d", len, have_data);
if (len >= 0 && have_data > 0) {
BufferInfo *buffer_info = frame.opaque;
gint nsamples, channels, byte_per_sample;
gsize output_size;
if (!gst_ffmpegauddec_negotiate (ffmpegdec, FALSE)) {
*outbuf = NULL;
*ret = GST_FLOW_NOT_NEGOTIATED;
len = -1;
goto beach;
}
channels = ffmpegdec->info.channels;
nsamples = frame.nb_samples;
byte_per_sample = ffmpegdec->info.finfo->width / 8;
/* frame.linesize[0] might contain padding, allocate only what's needed */
output_size = nsamples * byte_per_sample * channels;
GST_DEBUG_OBJECT (ffmpegdec, "Creating output buffer");
if (buffer_info) {
*outbuf = buffer_info->buffer;
gst_buffer_unmap (buffer_info->buffer, &buffer_info->map);
g_slice_free (BufferInfo, buffer_info);
frame.opaque = NULL;
} else if (av_sample_fmt_is_planar (ffmpegdec->context->sample_fmt)
&& channels > 1) {
gint i, j;
GstMapInfo minfo;
/* note: linesize[0] might contain padding, allocate only what's needed */
*outbuf =
gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER
(ffmpegdec), output_size);
gst_buffer_map (*outbuf, &minfo, GST_MAP_WRITE);
switch (ffmpegdec->info.finfo->width) {
case 8:{
guint8 *odata = minfo.data;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
odata[j] = ((const guint8 *) frame.extended_data[j])[i];
}
odata += channels;
}
break;
}
case 16:{
guint16 *odata = (guint16 *) minfo.data;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
odata[j] = ((const guint16 *) frame.extended_data[j])[i];
}
odata += channels;
}
break;
}
case 32:{
guint32 *odata = (guint32 *) minfo.data;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
odata[j] = ((const guint32 *) frame.extended_data[j])[i];
}
odata += channels;
}
break;
}
case 64:{
guint64 *odata = (guint64 *) minfo.data;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
odata[j] = ((const guint64 *) frame.extended_data[j])[i];
}
odata += channels;
}
break;
}
default:
g_assert_not_reached ();
break;
}
gst_buffer_unmap (*outbuf, &minfo);
} else {
*outbuf =
gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER
(ffmpegdec), output_size);
gst_buffer_fill (*outbuf, 0, frame.data[0], output_size);
}
GST_DEBUG_OBJECT (ffmpegdec, "Buffer created. Size: %d", have_data);
/* Reorder channels to the GStreamer channel order */
if (ffmpegdec->needs_reorder) {
*outbuf = gst_buffer_make_writable (*outbuf);
gst_audio_buffer_reorder_channels (*outbuf, ffmpegdec->info.finfo->format,
ffmpegdec->info.channels, ffmpegdec->ffmpeg_layout,
ffmpegdec->info.position);
}
} else {
*outbuf = NULL;
}
beach:
GST_DEBUG_OBJECT (ffmpegdec, "return flow %d, out %p, len %d",
*ret, *outbuf, len);
return len;
}
static gint
gst_ffmpegauddec_audio_frame (GstFFMpegAudDec * ffmpegdec,
AVCodec * in_plugin, guint8 * data, guint size, gint * have_data,
GstBuffer ** outbuf, GstFlowReturn * ret)
{
gint len = -1;
AVPacket packet;
GST_DEBUG_OBJECT (ffmpegdec, "size: %d", size);
gst_avpacket_init (&packet, data, size);
len =
avcodec_decode_audio4 (ffmpegdec->context, ffmpegdec->frame, have_data,
&packet);
GST_DEBUG_OBJECT (ffmpegdec,
"Decode audio: len=%d, have_data=%d", len, *have_data);
if (len >= 0 && *have_data) {
gint nsamples, channels, byte_per_sample;
gsize output_size;
if (!gst_ffmpegauddec_negotiate (ffmpegdec, ffmpegdec->context,
ffmpegdec->frame, FALSE)) {
*outbuf = NULL;
*ret = GST_FLOW_NOT_NEGOTIATED;
len = -1;
goto beach;
}
channels = ffmpegdec->info.channels;
nsamples = ffmpegdec->frame->nb_samples;
byte_per_sample = ffmpegdec->info.finfo->width / 8;
/* ffmpegdec->frame->linesize[0] might contain padding, allocate only what's needed */
output_size = nsamples * byte_per_sample * channels;
GST_DEBUG_OBJECT (ffmpegdec, "Creating output buffer");
if (av_sample_fmt_is_planar (ffmpegdec->context->sample_fmt)
&& channels > 1) {
gint i, j;
GstMapInfo minfo;
/* note: linesize[0] might contain padding, allocate only what's needed */
*outbuf =
gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER
(ffmpegdec), output_size);
gst_buffer_map (*outbuf, &minfo, GST_MAP_WRITE);
switch (ffmpegdec->info.finfo->width) {
case 8:{
guint8 *odata = minfo.data;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
odata[j] =
((const guint8 *) ffmpegdec->frame->extended_data[j])[i];
}
odata += channels;
}
break;
}
case 16:{
guint16 *odata = (guint16 *) minfo.data;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
odata[j] =
((const guint16 *) ffmpegdec->frame->extended_data[j])[i];
}
odata += channels;
}
break;
}
case 32:{
guint32 *odata = (guint32 *) minfo.data;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
odata[j] =
((const guint32 *) ffmpegdec->frame->extended_data[j])[i];
}
odata += channels;
}
break;
}
case 64:{
guint64 *odata = (guint64 *) minfo.data;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
odata[j] =
((const guint64 *) ffmpegdec->frame->extended_data[j])[i];
}
odata += channels;
}
break;
}
default:
g_assert_not_reached ();
break;
}
gst_buffer_unmap (*outbuf, &minfo);
} else {
*outbuf =
gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER
(ffmpegdec), output_size);
gst_buffer_fill (*outbuf, 0, ffmpegdec->frame->data[0], output_size);
}
GST_DEBUG_OBJECT (ffmpegdec, "Buffer created. Size: %" G_GSIZE_FORMAT,
output_size);
/* Reorder channels to the GStreamer channel order */
if (ffmpegdec->needs_reorder) {
*outbuf = gst_buffer_make_writable (*outbuf);
gst_audio_buffer_reorder_channels (*outbuf, ffmpegdec->info.finfo->format,
ffmpegdec->info.channels, ffmpegdec->ffmpeg_layout,
ffmpegdec->info.position);
}
/* Mark corrupted frames as corrupted */
if (ffmpegdec->frame->flags & AV_FRAME_FLAG_CORRUPT)
GST_BUFFER_FLAG_SET (*outbuf, GST_BUFFER_FLAG_CORRUPTED);
} else {
*outbuf = NULL;
}
beach:
av_frame_unref (ffmpegdec->frame);
GST_DEBUG_OBJECT (ffmpegdec, "return flow %d, out %p, len %d",
*ret, *outbuf, len);
return len;
}
static GstFlowReturn gst_imx_audio_uniaudio_dec_handle_frame(GstAudioDecoder *dec, GstBuffer *buffer)
{
GstMapInfo in_map;
GstBuffer *out_buffer;
gsize avail_out_size;
GstImxAudioUniaudioDec *imx_audio_uniaudio_dec = GST_IMX_AUDIO_UNIAUDIO_DEC(dec);
int32 dec_ret;
uint32 offset = 0;
uint8 *in_buf = NULL;
uint32 in_size = 0;
gboolean dec_loop = TRUE, flow_error = FALSE;
/* With some formats such as Vorbis, the first few buffers are actually redundant,
* since they contain codec data that was already specified in codec_data or
* streamheader caps earlier. If this is the case, skip these buffers. */
if (imx_audio_uniaudio_dec->skip_header_counter < imx_audio_uniaudio_dec->num_vorbis_headers)
{
GST_TRACE_OBJECT(dec, "skipping header buffer #%u", imx_audio_uniaudio_dec->skip_header_counter);
++imx_audio_uniaudio_dec->skip_header_counter;
return gst_audio_decoder_finish_frame(dec, NULL, 1);
}
if (buffer != NULL)
{
gst_buffer_map(buffer, &in_map, GST_MAP_READ);
in_buf = in_map.data;
in_size = in_map.size;
}
while (dec_loop)
{
GstBuffer *tmp_buf;
uint8 *out_buf = NULL;
uint32 out_size = 0;
if (buffer != NULL)
GST_TRACE_OBJECT(dec, "feeding %" G_GUINT32_FORMAT " bytes to the decoder", (guint32)in_size);
else
GST_TRACE_OBJECT(dec, "draining decoder");
dec_ret = imx_audio_uniaudio_dec->codec->decode_frame(
imx_audio_uniaudio_dec->handle,
in_buf, in_size,
&offset,
&out_buf, &out_size
);
GST_TRACE_OBJECT(dec, "decode_frame: return 0x%x offset %" G_GUINT32_FORMAT " out_size %" G_GUINT32_FORMAT, (unsigned int)dec_ret, (guint32)offset, (guint32)out_size);
if ((out_buf != NULL) && (out_size > 0))
{
tmp_buf = gst_audio_decoder_allocate_output_buffer(dec, out_size);
tmp_buf = gst_buffer_make_writable(tmp_buf);
gst_buffer_fill(tmp_buf, 0, out_buf, out_size);
gst_adapter_push(imx_audio_uniaudio_dec->out_adapter, tmp_buf);
}
if (out_buf != NULL)
{
gst_imx_audio_uniaudio_dec_free(out_buf);
}
if ((buffer != NULL) && (offset == in_map.size))
{
dec_loop = FALSE;
}
switch (dec_ret)
{
case ACODEC_SUCCESS:
break;
case ACODEC_END_OF_STREAM:
dec_loop = FALSE;
break;
case ACODEC_NOT_ENOUGH_DATA:
break;
case ACODEC_CAPIBILITY_CHANGE:
break;
default:
{
dec_loop = FALSE;
flow_error = TRUE;
GST_ELEMENT_ERROR(dec, STREAM, DECODE, ("could not decode"), ("error message: %s", imx_audio_uniaudio_dec->codec->get_last_error(imx_audio_uniaudio_dec->handle)));
}
}
}
if (buffer != NULL)
gst_buffer_unmap(buffer, &in_map);
if (flow_error)
return GST_FLOW_ERROR;
if (!(imx_audio_uniaudio_dec->has_audioinfo_set))
{
UniACodecParameter parameter;
GstAudioFormat pcm_fmt;
GstAudioInfo audio_info;
imx_audio_uniaudio_dec->codec->get_parameter(imx_audio_uniaudio_dec->handle, UNIA_OUTPUT_PCM_FORMAT, ¶meter);
if ((parameter.outputFormat.width == 0) || (parameter.outputFormat.depth == 0))
{
GST_DEBUG_OBJECT(imx_audio_uniaudio_dec, "no output format available yet");
return gst_audio_decoder_finish_frame(dec, NULL, 1);
}
GST_DEBUG_OBJECT(imx_audio_uniaudio_dec, "output sample width: %" G_GUINT32_FORMAT " depth: %" G_GUINT32_FORMAT, (guint32)(parameter.outputFormat.width), (guint32)(parameter.outputFormat.depth));
pcm_fmt = gst_audio_format_build_integer(TRUE, G_BYTE_ORDER, parameter.outputFormat.width, parameter.outputFormat.depth);
GST_DEBUG_OBJECT(imx_audio_uniaudio_dec, "setting output format to: %s %d Hz %d channels", gst_audio_format_to_string(pcm_fmt), (gint)(parameter.outputFormat.samplerate), (gint)(parameter.outputFormat.channels));
gst_imx_audio_uniaudio_dec_clear_channel_positions(imx_audio_uniaudio_dec);
gst_imx_audio_uniaudio_dec_fill_channel_positions(imx_audio_uniaudio_dec, parameter.outputFormat.layout, parameter.outputFormat.channels);
imx_audio_uniaudio_dec->pcm_format = pcm_fmt;
imx_audio_uniaudio_dec->num_channels = parameter.outputFormat.channels;
gst_audio_info_set_format(
&audio_info,
pcm_fmt,
parameter.outputFormat.samplerate,
parameter.outputFormat.channels,
imx_audio_uniaudio_dec->reordered_channel_positions
);
gst_audio_decoder_set_output_format(dec, &audio_info);
imx_audio_uniaudio_dec->has_audioinfo_set = TRUE;
}
avail_out_size = gst_adapter_available(imx_audio_uniaudio_dec->out_adapter);
if (avail_out_size > 0)
{
out_buffer = gst_adapter_take_buffer(imx_audio_uniaudio_dec->out_adapter, avail_out_size);
if (imx_audio_uniaudio_dec->original_channel_positions != imx_audio_uniaudio_dec->reordered_channel_positions)
{
gst_audio_buffer_reorder_channels(
out_buffer,
imx_audio_uniaudio_dec->pcm_format,
imx_audio_uniaudio_dec->num_channels,
imx_audio_uniaudio_dec->original_channel_positions,
imx_audio_uniaudio_dec->reordered_channel_positions
);
}
return gst_audio_decoder_finish_frame(dec, out_buffer, 1);
}
else
{
return gst_audio_decoder_finish_frame(dec, NULL, 1);
}
}
