// This function must be called before
// any other decoding functions
int nv_opus_init(int sampleRate, int channelCount, int streams,
int coupledStreams, const unsigned char *mapping) {
int err;
decoder = opus_multistream_decoder_create(
sampleRate,
channelCount,
streams,
coupledStreams,
mapping,
&err);
return err;
}
FOpusDecoderWrapper(uint16 SampleRate, uint8 NumChannels)
{
#if WITH_OPUS
check(NumChannels <= 8);
const UnrealChannelLayout& Layout = UnrealMappings[NumChannels-1];
#if USE_UE4_MEM_ALLOC
int32 DecSize = opus_multistream_decoder_get_size(Layout.NumStreams, Layout.NumCoupledStreams);
Decoder = (OpusMSDecoder*)FMemory::Malloc(DecSize);
DecError = opus_multistream_decoder_init(Decoder, SampleRate, NumChannels, Layout.NumStreams, Layout.NumCoupledStreams, Layout.Mapping);
#else
Decoder = opus_multistream_decoder_create(SampleRate, NumChannels, Layout.NumStreams, Layout.NumCoupledStreams, Layout.Mapping, &DecError);
#endif
#endif
}
void MoonlightInstance::AudDecInit(int audioConfiguration, POPUS_MULTISTREAM_CONFIGURATION opusConfig) {
int rc;
g_Instance->m_OpusDecoder = opus_multistream_decoder_create(opusConfig->sampleRate,
opusConfig->channelCount,
opusConfig->streams,
opusConfig->coupledStreams,
opusConfig->mapping,
&rc);
g_Instance->m_AudioPlayer = pp::Audio(g_Instance, pp::AudioConfig(g_Instance, PP_AUDIOSAMPLERATE_48000, FRAME_SIZE),
AudioPlayerSampleCallback, NULL);
// Start playback now
g_Instance->m_AudioPlayer.StartPlayback();
}
bool OpusState::Init(void)
{
if (!mActive)
return false;
int error;
NS_ASSERTION(mDecoder == nullptr, "leaking OpusDecoder");
mDecoder = opus_multistream_decoder_create(mParser->mRate,
mParser->mChannels,
mParser->mStreams,
mParser->mCoupledStreams,
mParser->mMappingTable,
&error);
mSkip = mParser->mPreSkip;
LOG(PR_LOG_DEBUG, ("Opus decoder init, to skip %d", mSkip));
return error == OPUS_OK;
}
/*****************************************************************************
* ProcessInitialHeader: processes the inital Opus header packet.
*****************************************************************************/
static int ProcessInitialHeader( decoder_t *p_dec, ogg_packet *p_oggpacket )
{
int err;
unsigned char new_stream_map[8];
decoder_sys_t *p_sys = p_dec->p_sys;
OpusHeader *p_header = &p_sys->header;
if( !opus_header_parse((unsigned char *)p_oggpacket->packet,p_oggpacket->bytes,p_header) )
{
msg_Err( p_dec, "cannot read Opus header" );
return VLC_EGENERIC;
}
msg_Dbg( p_dec, "Opus audio with %d channels", p_header->channels);
if((p_header->channels>2 && p_header->channel_mapping==0) ||
(p_header->channels>8 && p_header->channel_mapping==1) ||
p_header->channel_mapping>1)
{
msg_Err( p_dec, "Unsupported channel mapping" );
return VLC_EGENERIC;
}
/* Setup the format */
p_dec->fmt_out.audio.i_physical_channels =
p_dec->fmt_out.audio.i_original_channels =
pi_channels_maps[p_header->channels];
p_dec->fmt_out.audio.i_channels = p_header->channels;
p_dec->fmt_out.audio.i_rate = 48000;
if( p_header->channels>2 && p_header->channels<9 )
{
static const uint32_t *pi_ch[6] = { pi_3channels_in, pi_4channels_in,
pi_5channels_in, pi_6channels_in,
pi_7channels_in, pi_8channels_in };
uint8_t pi_chan_table[AOUT_CHAN_MAX];
aout_CheckChannelReorder( pi_ch[p_header->channels-3], NULL,
p_dec->fmt_out.audio.i_physical_channels,
pi_chan_table );
for(int i=0;i<p_header->channels;i++)
new_stream_map[pi_chan_table[i]]=p_header->stream_map[i];
}
/* Opus decoder init */
p_sys->p_st = opus_multistream_decoder_create( 48000, p_header->channels,
p_header->nb_streams, p_header->nb_coupled,
p_header->channels>2?new_stream_map:p_header->stream_map,
&err );
if( !p_sys->p_st || err!=OPUS_OK )
{
msg_Err( p_dec, "decoder initialization failed" );
return VLC_EGENERIC;
}
#ifdef OPUS_SET_GAIN
if( opus_multistream_decoder_ctl( p_sys->p_st,OPUS_SET_GAIN(p_header->gain) ) != OPUS_OK )
{
msg_Err( p_dec, "OPUS_SET_GAIN failed" );
opus_multistream_decoder_destroy( p_sys->p_st );
return VLC_EGENERIC;
}
#endif
date_Init( &p_sys->end_date, 48000, 1 );
return VLC_SUCCESS;
}
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 void omx_renderer_init(int audioConfiguration, POPUS_MULTISTREAM_CONFIGURATION opusConfig) {
int rc, error;
OMX_ERRORTYPE err;
unsigned char omxMapping[6];
char* componentName = "audio_render";
channelCount = opusConfig->channelCount;
/* The supplied mapping array has order: FL-FR-C-LFE-RL-RR
* OMX expects the order: FL-FR-LFE-C-RL-RR
* We need copy the mapping locally and swap the channels around.
*/
memcpy(omxMapping, opusConfig->mapping, sizeof(omxMapping));
if (opusConfig->channelCount > 2) {
omxMapping[2] = opusConfig->mapping[3];
omxMapping[3] = opusConfig->mapping[2];
}
decoder = opus_multistream_decoder_create(opusConfig->sampleRate,
opusConfig->channelCount,
opusConfig->streams,
opusConfig->coupledStreams,
omxMapping,
&rc);
handle = ilclient_init();
if (handle == NULL) {
fprintf(stderr, "IL client init failed\n");
exit(1);
}
if (ilclient_create_component(handle, &component, componentName, ILCLIENT_DISABLE_ALL_PORTS | ILCLIENT_ENABLE_INPUT_BUFFERS) != 0) {
fprintf(stderr, "Component create failed\n");
exit(1);
}
if (ilclient_change_component_state(component, OMX_StateIdle)!= 0) {
fprintf(stderr, "Couldn't change state to Idle\n");
exit(1);
}
// must be before we enable buffers
OMX_AUDIO_PARAM_PORTFORMATTYPE audioPortFormat;
memset(&audioPortFormat, 0, sizeof(OMX_AUDIO_PARAM_PORTFORMATTYPE));
audioPortFormat.nSize = sizeof(OMX_AUDIO_PARAM_PORTFORMATTYPE);
audioPortFormat.nVersion.nVersion = OMX_VERSION;
audioPortFormat.nPortIndex = 100;
OMX_GetParameter(ilclient_get_handle(component), OMX_IndexParamAudioPortFormat, &audioPortFormat);
audioPortFormat.eEncoding = OMX_AUDIO_CodingPCM;
OMX_SetParameter(ilclient_get_handle(component), OMX_IndexParamAudioPortFormat, &audioPortFormat);
OMX_AUDIO_PARAM_PCMMODETYPE sPCMMode;
memset(&sPCMMode, 0, sizeof(OMX_AUDIO_PARAM_PCMMODETYPE));
sPCMMode.nSize = sizeof(OMX_AUDIO_PARAM_PCMMODETYPE);
sPCMMode.nVersion.nVersion = OMX_VERSION;
sPCMMode.nPortIndex = 100;
sPCMMode.nChannels = channelCount;
sPCMMode.eNumData = OMX_NumericalDataSigned;
sPCMMode.eEndian = OMX_EndianLittle;
sPCMMode.nSamplingRate = opusConfig->sampleRate;
sPCMMode.bInterleaved = OMX_TRUE;
sPCMMode.nBitPerSample = 16;
sPCMMode.ePCMMode = OMX_AUDIO_PCMModeLinear;
switch(channelCount) {
case 1:
sPCMMode.eChannelMapping[0] = OMX_AUDIO_ChannelCF;
break;
case 8:
sPCMMode.eChannelMapping[7] = OMX_AUDIO_ChannelRS;
case 7:
sPCMMode.eChannelMapping[6] = OMX_AUDIO_ChannelLS;
case 6:
sPCMMode.eChannelMapping[5] = OMX_AUDIO_ChannelRR;
case 5:
sPCMMode.eChannelMapping[4] = OMX_AUDIO_ChannelLR;
case 4:
sPCMMode.eChannelMapping[3] = OMX_AUDIO_ChannelLFE;
case 3:
sPCMMode.eChannelMapping[2] = OMX_AUDIO_ChannelCF;
case 2:
sPCMMode.eChannelMapping[1] = OMX_AUDIO_ChannelRF;
sPCMMode.eChannelMapping[0] = OMX_AUDIO_ChannelLF;
break;
}
err = OMX_SetParameter(ilclient_get_handle(component), OMX_IndexParamAudioPcm, &sPCMMode);
if(err != OMX_ErrorNone){
fprintf(stderr, "PCM mode unsupported\n");
return;
}
OMX_CONFIG_BRCMAUDIODESTINATIONTYPE arDest;
if (audio_device == NULL)
audio_device = "hdmi";
if (audio_device && strlen(audio_device) < sizeof(arDest.sName)) {
memset(&arDest, 0, sizeof(OMX_CONFIG_BRCMAUDIODESTINATIONTYPE));
arDest.nSize = sizeof(OMX_CONFIG_BRCMAUDIODESTINATIONTYPE);
arDest.nVersion.nVersion = OMX_VERSION;
strcpy((char *)arDest.sName, audio_device);
err = OMX_SetParameter(ilclient_get_handle(component), OMX_IndexConfigBrcmAudioDestination, &arDest);
if (err != OMX_ErrorNone) {
fprintf(stderr, "Error on setting audio destination\nomx option must be set to hdmi or local\n");
exit(1);
}
}
// input port
ilclient_enable_port_buffers(component, 100, NULL, NULL, NULL);
ilclient_enable_port(component, 100);
err = ilclient_change_component_state(component, OMX_StateExecuting);
if (err < 0) {
fprintf(stderr, "Couldn't change state to Executing\n");
exit(1);
}
}
static av_cold int libopus_decode_init(AVCodecContext *avc)
{
struct libopus_context *opus = avc->priv_data;
int ret, channel_map = 0, gain_db = 0, nb_streams, nb_coupled;
uint8_t mapping_arr[8] = { 0, 1 }, *mapping;
avc->sample_rate = 48000;
avc->sample_fmt = avc->request_sample_fmt == AV_SAMPLE_FMT_FLT ?
AV_SAMPLE_FMT_FLT : AV_SAMPLE_FMT_S16;
avc->channel_layout = avc->channels > 8 ? 0 :
ff_vorbis_channel_layouts[avc->channels - 1];
if (avc->extradata_size >= OPUS_HEAD_SIZE) {
gain_db = sign_extend(AV_RL16(avc->extradata + 16), 16);
channel_map = AV_RL8 (avc->extradata + 18);
}
if (avc->extradata_size >= OPUS_HEAD_SIZE + 2 + avc->channels) {
nb_streams = avc->extradata[OPUS_HEAD_SIZE + 0];
nb_coupled = avc->extradata[OPUS_HEAD_SIZE + 1];
if (nb_streams + nb_coupled != avc->channels)
av_log(avc, AV_LOG_WARNING, "Inconsistent channel mapping.\n");
mapping = avc->extradata + OPUS_HEAD_SIZE + 2;
} else {
if (avc->channels > 2 || channel_map) {
av_log(avc, AV_LOG_ERROR,
"No channel mapping for %d channels.\n", avc->channels);
return AVERROR(EINVAL);
}
nb_streams = 1;
nb_coupled = avc->channels > 1;
mapping = mapping_arr;
}
if (avc->channels > 2 && avc->channels <= 8) {
const uint8_t *vorbis_offset = ff_vorbis_channel_layout_offsets[avc->channels - 1];
int ch;
/* Remap channels from vorbis order to libav order */
for (ch = 0; ch < avc->channels; ch++)
mapping_arr[ch] = mapping[vorbis_offset[ch]];
mapping = mapping_arr;
}
opus->dec = opus_multistream_decoder_create(avc->sample_rate, avc->channels,
nb_streams, nb_coupled,
mapping, &ret);
if (!opus->dec) {
av_log(avc, AV_LOG_ERROR, "Unable to create decoder: %s\n",
opus_strerror(ret));
return ff_opus_error_to_averror(ret);
}
ret = opus_multistream_decoder_ctl(opus->dec, OPUS_SET_GAIN(gain_db));
if (ret != OPUS_OK)
av_log(avc, AV_LOG_WARNING, "Failed to set gain: %s\n",
opus_strerror(ret));
avc->delay = 3840; /* Decoder delay (in samples) at 48kHz */
avcodec_get_frame_defaults(&opus->frame);
avc->coded_frame = &opus->frame;
return 0;
}
/*Process an Opus header and setup the opus decoder based on it.
It takes several pointers for header values which are needed
elsewhere in the code.*/
static OpusMSDecoder *process_header(ogg_packet *op, opus_int32 *rate,
int *mapping_family, int *channels, int *preskip, float *gain,
float manual_gain, int *streams, int wav_format, int quiet)
{
int err;
OpusMSDecoder *st;
OpusHeader header;
if (opus_header_parse(op->packet, op->bytes, &header)==0)
{
fprintf(stderr, "Cannot parse header\n");
return NULL;
}
*mapping_family = header.channel_mapping;
*channels = header.channels;
if(wav_format)adjust_wav_mapping(*mapping_family, *channels, header.stream_map);
if(!*rate)*rate=header.input_sample_rate;
/*If the rate is unspecified we decode to 48000*/
if(*rate==0)*rate=48000;
if(*rate<8000||*rate>192000){
fprintf(stderr,"Warning: Crazy input_rate %d, decoding to 48000 instead.\n",*rate);
*rate=48000;
}
*preskip = header.preskip;
st = opus_multistream_decoder_create(48000, header.channels, header.nb_streams, header.nb_coupled, header.stream_map, &err);
if(err != OPUS_OK){
fprintf(stderr, "Cannot create encoder: %s\n", opus_strerror(err));
return NULL;
}
if (!st)
{
fprintf (stderr, "Decoder initialization failed: %s\n", opus_strerror(err));
return NULL;
}
*streams=header.nb_streams;
if(header.gain!=0 || manual_gain!=0)
{
/*Gain API added in a newer libopus version, if we don't have it
we apply the gain ourselves. We also add in a user provided
manual gain at the same time.*/
int gainadj = (int)(manual_gain*256.)+header.gain;
#ifdef OPUS_SET_GAIN
err=opus_multistream_decoder_ctl(st,OPUS_SET_GAIN(gainadj));
if(err==OPUS_UNIMPLEMENTED)
{
#endif
*gain = pow(10., gainadj/5120.);
#ifdef OPUS_SET_GAIN
} else if (err!=OPUS_OK)
{
fprintf (stderr, "Error setting gain: %s\n", opus_strerror(err));
return NULL;
}
#endif
}
if (!quiet)
{
fprintf(stderr, "Decoding to %d Hz (%d channel%s)", *rate,
*channels, *channels>1?"s":"");
if(header.version!=1)fprintf(stderr, ", Header v%d",header.version);
fprintf(stderr, "\n");
if (header.gain!=0)fprintf(stderr,"Playback gain: %f dB\n", header.gain/256.);
if (manual_gain!=0)fprintf(stderr,"Manual gain: %f dB\n", manual_gain);
}
return st;
}
krad_opus_t *krad_opus_decoder_create (unsigned char *header_data, int header_length, float output_sample_rate) {
int c;
krad_opus_t *krad_opus = calloc (1, sizeof(krad_opus_t));
krad_opus->output_sample_rate = output_sample_rate;
krad_opus->opus_header = calloc (1, sizeof(OpusHeader));
if (opus_header_parse (header_data, header_length, krad_opus->opus_header) != 1) {
failfast ("krad_opus_decoder_create problem reading opus header");
}
// oops
//krad_opus->input_sample_rate = krad_opus->opus_header->input_sample_rate;
krad_opus->channels = krad_opus->opus_header->channels;
krad_opus->interleaved_samples = malloc(16 * 8192);
for (c = 0; c < krad_opus->channels; c++) {
krad_opus->ringbuf[c] = krad_ringbuffer_create (RINGBUFFER_SIZE);
krad_opus->resampled_ringbuf[c] = krad_ringbuffer_create (RINGBUFFER_SIZE);
krad_opus->samples[c] = malloc (16 * 8192);
krad_opus->read_samples[c] = malloc (16 * 8192);
krad_opus->resampled_samples[c] = malloc (16 * 8192);
krad_opus->src_resampler[c] = src_new (KRAD_OPUS_SRC_QUALITY, 1, &krad_opus->src_error[c]);
if (krad_opus->src_resampler[c] == NULL) {
failfast ("krad_opus_decoder_create src resampler error: %s", src_strerror (krad_opus->src_error[c]));
}
krad_opus->src_data[c].src_ratio = output_sample_rate / 48000;
printk ("krad_opus_decoder_create src resampler ratio is: %f", krad_opus->src_data[c].src_ratio);
}
krad_opus->streams = krad_opus->opus_header->nb_streams;
krad_opus->coupled_streams = krad_opus->opus_header->nb_coupled;
memcpy (krad_opus->mapping, krad_opus->opus_header->stream_map, 256);
printk ("krad_opus_decoder_create channels %d streams %d coupled %d",
krad_opus->channels,
krad_opus->streams,
krad_opus->coupled_streams
);
krad_opus->decoder = opus_multistream_decoder_create (48000,
krad_opus->opus_header->channels,
krad_opus->streams,
krad_opus->coupled_streams,
krad_opus->mapping,
&krad_opus->opus_decoder_error);
if (krad_opus->opus_decoder_error != OPUS_OK) {
failfast ("Cannot create decoder: %s", opus_strerror (krad_opus->opus_decoder_error));
}
return krad_opus;
}
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);
}
}
