static int libopus_decode(AVCodecContext *avc, void *frame,
int *got_frame_ptr, AVPacket *pkt)
{
struct libopus_context *opus = avc->priv_data;
int ret, nb_samples;
opus->frame.nb_samples = MAX_FRAME_SIZE;
ret = avc->get_buffer(avc, &opus->frame);
if (ret < 0) {
av_log(avc, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
if (avc->sample_fmt == AV_SAMPLE_FMT_S16)
nb_samples = opus_multistream_decode(opus->dec, pkt->data, pkt->size,
(opus_int16 *)opus->frame.data[0],
opus->frame.nb_samples, 0);
else
nb_samples = opus_multistream_decode_float(opus->dec, pkt->data, pkt->size,
(float *)opus->frame.data[0],
opus->frame.nb_samples, 0);
if (nb_samples < 0) {
av_log(avc, AV_LOG_ERROR, "Decoding error: %s\n",
opus_strerror(nb_samples));
return ff_opus_error_to_averror(nb_samples);
}
opus->frame.nb_samples = nb_samples;
*(AVFrame *)frame = opus->frame;
*got_frame_ptr = 1;
return pkt->size;
}
int32 Decode(const uint8* FrameData, uint16 FrameSize, int16* OutPCMData, int32 SampleSize)
{
#if WITH_OPUS
return opus_multistream_decode(Decoder, FrameData, FrameSize, OutPCMData, SampleSize, 0);
#else
return -1;
#endif
}
// packets must be decoded in order
// a packet loss must call this function with NULL indata and 0 inlen
// returns the number of decoded samples
int nv_opus_decode(unsigned char* indata, int inlen, short* outpcmdata, int framesize) {
int err;
// Decoding to 16-bit PCM with FEC off
// Maximum length assuming 48KHz sample rate
err = opus_multistream_decode(decoder, indata, inlen,
outpcmdata, framesize, 0);
return err;
}
static int libopus_decode(AVCodecContext *avc, void *data,
int *got_frame_ptr, AVPacket *pkt)
{
struct libopus_context *opus = avc->priv_data;
AVFrame *frame = data;
int ret, nb_samples;
frame->nb_samples = MAX_FRAME_SIZE;
ret = ff_get_buffer(avc, frame);
if (ret < 0) {
av_log(avc, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
if (avc->sample_fmt == AV_SAMPLE_FMT_S16)
nb_samples = opus_multistream_decode(opus->dec, pkt->data, pkt->size,
(opus_int16 *)frame->data[0],
frame->nb_samples, 0);
else
nb_samples = opus_multistream_decode_float(opus->dec, pkt->data, pkt->size,
(float *)frame->data[0],
frame->nb_samples, 0);
if (nb_samples < 0) {
av_log(avc, AV_LOG_ERROR, "Decoding error: %s\n",
opus_strerror(nb_samples));
return ff_opus_error_to_averror(nb_samples);
}
#ifndef OPUS_SET_GAIN
{
int i = avc->channels * nb_samples;
if (avc->sample_fmt == AV_SAMPLE_FMT_FLT) {
float *pcm = (float *)frame->data[0];
for (; i > 0; i--, pcm++)
*pcm = av_clipf(*pcm * opus->gain.d, -1, 1);
} else {
int16_t *pcm = (int16_t *)frame->data[0];
for (; i > 0; i--, pcm++)
*pcm = av_clip_int16(((int64_t)opus->gain.i * *pcm) >> 16);
}
}
#endif
frame->nb_samples = nb_samples;
*got_frame_ptr = 1;
return pkt->size;
}
static void omx_renderer_decode_and_play_sample(char* data, int length) {
int decodeLen = opus_multistream_decode(decoder, data, length, pcmBuffer, FRAME_SIZE, 0);
if (decodeLen > 0) {
buf = ilclient_get_input_buffer(component, 100, 1);
buf->nOffset = 0;
buf->nFlags = OMX_BUFFERFLAG_TIME_UNKNOWN;
int bufLength = decodeLen * sizeof(short) * channelCount;
memcpy(buf->pBuffer, pcmBuffer, bufLength);
buf->nFilledLen = bufLength;
int r = OMX_EmptyThisBuffer(ilclient_get_handle(component), buf);
if (r != OMX_ErrorNone) {
fprintf(stderr, "Empty buffer error\n");
}
} else {
printf("Opus error from decode: %d\n", decodeLen);
}
}
void MoonlightInstance::AudDecDecodeAndPlaySample(char* sampleData, int sampleLength) {
int decodeLen;
// Check if there is space for this sample in the buffer. Again, this can race
// but in the worst case, we'll not see the sample callback having consumed a sample.
if (((s_WriteIndex + 1) % CIRCULAR_BUFFER_SIZE) == s_ReadIndex) {
return;
}
decodeLen = opus_multistream_decode(g_Instance->m_OpusDecoder, (unsigned char *)sampleData, sampleLength,
s_CircularBuffer[s_WriteIndex], FRAME_SIZE, 0);
if (decodeLen > 0) {
// Use a full memory barrier to ensure the circular buffer is written before incrementing the index
__sync_synchronize();
// This can race with the reader in the sample callback, however this is a benign
// race since we'll either read the original value of s_WriteIndex (which is safe,
// we just won't consider this sample) or the new value of s_WriteIndex
s_WriteIndex = (s_WriteIndex + 1) % CIRCULAR_BUFFER_SIZE;
}
}
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;
}
void SoftOpus::onQueueFilled(OMX_U32 /* portIndex */) {
List<BufferInfo *> &inQueue = getPortQueue(0);
List<BufferInfo *> &outQueue = getPortQueue(1);
if (mOutputPortSettingsChange != NONE) {
return;
}
while (!mHaveEOS && !inQueue.empty() && !outQueue.empty()) {
BufferInfo *inInfo = *inQueue.begin();
OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
if (mInputBufferCount < 3) {
const uint8_t *data = inHeader->pBuffer + inHeader->nOffset;
size_t size = inHeader->nFilledLen;
if ((inHeader->nFlags & OMX_BUFFERFLAG_EOS) && size == 0) {
handleEOS();
return;
}
if (mInputBufferCount == 0) {
CHECK(mHeader == NULL);
mHeader = new OpusHeader();
memset(mHeader, 0, sizeof(*mHeader));
if (!ParseOpusHeader(data, size, mHeader)) {
ALOGV("Parsing Opus Header failed.");
notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
return;
}
uint8_t channel_mapping[kMaxChannels] = {0};
if (mHeader->channels <= kMaxChannelsWithDefaultLayout) {
memcpy(&channel_mapping,
kDefaultOpusChannelLayout,
kMaxChannelsWithDefaultLayout);
} else {
memcpy(&channel_mapping,
mHeader->stream_map,
mHeader->channels);
}
int status = OPUS_INVALID_STATE;
mDecoder = opus_multistream_decoder_create(kRate,
mHeader->channels,
mHeader->num_streams,
mHeader->num_coupled,
channel_mapping,
&status);
if (!mDecoder || status != OPUS_OK) {
ALOGV("opus_multistream_decoder_create failed status=%s",
opus_strerror(status));
notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
return;
}
status =
opus_multistream_decoder_ctl(mDecoder,
OPUS_SET_GAIN(mHeader->gain_db));
if (status != OPUS_OK) {
ALOGV("Failed to set OPUS header gain; status=%s",
opus_strerror(status));
notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
return;
}
} else if (mInputBufferCount == 1) {
mCodecDelay = ns_to_samples(
*(reinterpret_cast<int64_t*>(inHeader->pBuffer +
inHeader->nOffset)),
kRate);
mSamplesToDiscard = mCodecDelay;
} else {
mSeekPreRoll = ns_to_samples(
*(reinterpret_cast<int64_t*>(inHeader->pBuffer +
inHeader->nOffset)),
kRate);
notify(OMX_EventPortSettingsChanged, 1, 0, NULL);
mOutputPortSettingsChange = AWAITING_DISABLED;
}
if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
handleEOS();
return;
}
inQueue.erase(inQueue.begin());
inInfo->mOwnedByUs = false;
notifyEmptyBufferDone(inHeader);
++mInputBufferCount;
continue;
}
// Ignore CSD re-submissions.
if (mInputBufferCount >= 3 && (inHeader->nFlags & OMX_BUFFERFLAG_CODECCONFIG)) {
if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
handleEOS();
return;
}
inQueue.erase(inQueue.begin());
inInfo->mOwnedByUs = false;
notifyEmptyBufferDone(inHeader);
continue;
}
BufferInfo *outInfo = *outQueue.begin();
OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
if ((inHeader->nFlags & OMX_BUFFERFLAG_EOS) && inHeader->nFilledLen == 0) {
handleEOS();
return;
}
if (inHeader->nOffset == 0) {
mAnchorTimeUs = inHeader->nTimeStamp;
mNumFramesOutput = 0;
}
// When seeking to zero, |mCodecDelay| samples has to be discarded
// instead of |mSeekPreRoll| samples (as we would when seeking to any
// other timestamp).
if (inHeader->nTimeStamp == 0) {
mSamplesToDiscard = mCodecDelay;
}
const uint8_t *data = inHeader->pBuffer + inHeader->nOffset;
const uint32_t size = inHeader->nFilledLen;
size_t frameSize = kMaxOpusOutputPacketSizeSamples;
if (frameSize > outHeader->nAllocLen / sizeof(int16_t) / mHeader->channels) {
frameSize = outHeader->nAllocLen / sizeof(int16_t) / mHeader->channels;
android_errorWriteLog(0x534e4554, "27833616");
}
int numFrames = opus_multistream_decode(mDecoder,
data,
size,
(int16_t *)outHeader->pBuffer,
frameSize,
0);
if (numFrames < 0) {
ALOGE("opus_multistream_decode returned %d", numFrames);
notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
return;
}
outHeader->nOffset = 0;
if (mSamplesToDiscard > 0) {
if (mSamplesToDiscard > numFrames) {
mSamplesToDiscard -= numFrames;
numFrames = 0;
} else {
numFrames -= mSamplesToDiscard;
outHeader->nOffset = mSamplesToDiscard * sizeof(int16_t) *
mHeader->channels;
mSamplesToDiscard = 0;
}
}
outHeader->nFilledLen = numFrames * sizeof(int16_t) * mHeader->channels;
outHeader->nTimeStamp = mAnchorTimeUs +
(mNumFramesOutput * 1000000ll) /
kRate;
mNumFramesOutput += numFrames;
if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
outHeader->nFlags = OMX_BUFFERFLAG_EOS;
mHaveEOS = true;
} else {
outHeader->nFlags = 0;
}
inInfo->mOwnedByUs = false;
inQueue.erase(inQueue.begin());
notifyEmptyBufferDone(inHeader);
++mInputBufferCount;
outInfo->mOwnedByUs = false;
outQueue.erase(outQueue.begin());
notifyFillBufferDone(outHeader);
}
}
