int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, int channels) {
int error_l;
int error_r;
OpusDecInst* state;
// Create Opus decoder memory.
state = (OpusDecInst*) calloc(1, sizeof(OpusDecInst));
if (state == NULL) {
return -1;
}
// Create new memory for left and right channel, always at 48000 Hz.
state->decoder_left = opus_decoder_create(48000, channels, &error_l);
state->decoder_right = opus_decoder_create(48000, channels, &error_r);
if (error_l == OPUS_OK && error_r == OPUS_OK && state->decoder_left != NULL
&& state->decoder_right != NULL) {
// Creation of memory all ok.
state->channels = channels;
*inst = state;
return 0;
}
// If memory allocation was unsuccessful, free the entire state.
if (state->decoder_left) {
opus_decoder_destroy(state->decoder_left);
}
if (state->decoder_right) {
opus_decoder_destroy(state->decoder_right);
}
free(state);
state = NULL;
return -1;
}
static void audio_renderer_init() {
int rc;
decoder = opus_decoder_create(SAMPLE_RATE, CHANNEL_COUNT, &rc);
snd_pcm_hw_params_t *hw_params;
snd_pcm_sw_params_t *sw_params;
snd_pcm_uframes_t period_size = FRAME_SIZE * CHANNEL_COUNT * 2;
snd_pcm_uframes_t buffer_size = 12 * period_size;
unsigned int sampleRate = SAMPLE_RATE;
/* Open PCM device for playback. */
CHECK_RETURN(snd_pcm_open(&handle, audio_device, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK))
/* Set hardware parameters */
CHECK_RETURN(snd_pcm_hw_params_malloc(&hw_params));
CHECK_RETURN(snd_pcm_hw_params_any(handle, hw_params));
CHECK_RETURN(snd_pcm_hw_params_set_access(handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED));
CHECK_RETURN(snd_pcm_hw_params_set_format(handle, hw_params, SND_PCM_FORMAT_S16_LE));
CHECK_RETURN(snd_pcm_hw_params_set_rate_near(handle, hw_params, &sampleRate, NULL));
CHECK_RETURN(snd_pcm_hw_params_set_channels(handle, hw_params, CHANNEL_COUNT));
CHECK_RETURN(snd_pcm_hw_params_set_buffer_size_near(handle, hw_params, &buffer_size));
CHECK_RETURN(snd_pcm_hw_params_set_period_size_near(handle, hw_params, &period_size, NULL));
CHECK_RETURN(snd_pcm_hw_params(handle, hw_params));
snd_pcm_hw_params_free(hw_params);
/* Set software parameters */
CHECK_RETURN(snd_pcm_sw_params_malloc(&sw_params));
CHECK_RETURN(snd_pcm_sw_params_current(handle, sw_params));
CHECK_RETURN(snd_pcm_sw_params_set_start_threshold(handle, sw_params, buffer_size - period_size));
CHECK_RETURN(snd_pcm_sw_params_set_avail_min(handle, sw_params, period_size));
CHECK_RETURN(snd_pcm_sw_params(handle, sw_params));
snd_pcm_sw_params_free(sw_params);
CHECK_RETURN(snd_pcm_prepare(handle));
}
static int opus_decoder_construct(struct ast_trans_pvt *pvt, int sampling_rate)
{
struct opus_coder_pvt *opvt = pvt->pvt;
int error = 0;
if (!valid_sampling_rate(sampling_rate)) {
return -1;
}
opvt->sampling_rate = sampling_rate;
opvt->multiplier = 48000/sampling_rate;
opvt->fec = USE_FEC; /* FIXME: should be triggered by chan_sip */
opvt->opus = opus_decoder_create(sampling_rate, 1, &error);
if (error != OPUS_OK) {
ast_log(LOG_ERROR, "Error creating the Opus decoder: %s\n", opus_strerror(error));
return -1;
}
opvt->id = ast_atomic_fetchadd_int(&usage.decoder_id, 1) + 1;
ast_atomic_fetchadd_int(&usage.decoders, +1);
ast_debug(3, "Created decoder #%d (opus -> %d)\n", opvt->id, sampling_rate);
return 0;
}
static int tdav_codec_opus_open(tmedia_codec_t* self)
{
tdav_codec_opus_t* opus = (tdav_codec_opus_t*)self;
int opus_err;
if(!opus) {
TSK_DEBUG_ERROR("Invalid parameter");
return -1;
}
// Initialize the decoder
if(!opus->decoder.inst) {
TSK_DEBUG_INFO("[OPUS] Open decoder: rate=%d, channels=%d", (int)self->in.rate, (int)TMEDIA_CODEC_AUDIO(self)->in.channels);
if(!(opus->decoder.inst = opus_decoder_create((opus_int32)self->in.rate, (int)TMEDIA_CODEC_AUDIO(self)->in.channels, &opus_err)) || opus_err != OPUS_OK) {
TSK_DEBUG_ERROR("Failed to create Opus decoder(rate=%d, channels=%d) instance with error code=%d.", (int)self->in.rate, (int)TMEDIA_CODEC_AUDIO(self)->in.channels, opus_err);
return -2;
}
}
opus->decoder.last_seq = 0;
// Initialize the encoder
if(!opus->encoder.inst) {
TSK_DEBUG_INFO("[OPUS] Open encoder: rate=%d, channels=%d", (int)self->out.rate, (int)TMEDIA_CODEC_AUDIO(self)->out.channels);
if(!(opus->encoder.inst = opus_encoder_create((opus_int32)self->out.rate, (int)TMEDIA_CODEC_AUDIO(self)->out.channels, OPUS_APPLICATION_VOIP, &opus_err)) || opus_err != OPUS_OK) {
TSK_DEBUG_ERROR("Failed to create Opus decoder(rate=%d, channels=%d) instance with error code=%d.", (int)self->out.rate, (int)TMEDIA_CODEC_AUDIO(self)->out.channels, opus_err);
return -2;
}
}
#if TDAV_UNDER_MOBILE /* iOS, Android and WP8 */
opus_encoder_ctl(opus->encoder.inst, OPUS_SET_COMPLEXITY(3));
#endif
opus_encoder_ctl(opus->encoder.inst, OPUS_SET_SIGNAL(OPUS_SIGNAL_VOICE));
return 0;
}
void SoundOpusData::onRender()
{
if(_samples && samplesDecodingProgress != _frames)
{
Decoder* dec = 0;
foreach(e, decoders)
{
if(e->first == 0)
{
dec = e;
break;
}
}
if(!dec)
{
dec = decoders.push();
int err;
dec->decoder = opus_decoder_create(AUDIOLIB_FRAME_RATE, _channels, &err);
dec->first = 0;
dec->end = 0;
}
if(dec->end == samplesDecodingProgress)
{
uint packet = dec->end / SOUNDLIB_OPUS_PACKET_FRAMES;
opus_decode(dec->decoder, _opusData + SOUNDLIB_OPUS_PACKET_SIZE * _channels * packet, SOUNDLIB_OPUS_PACKET_SIZE * _channels, _samples + _channels * packet * SOUNDLIB_OPUS_PACKET_FRAMES, SOUNDLIB_OPUS_PACKET_FRAMES, 0);
dec->end += SOUNDLIB_OPUS_PACKET_FRAMES;
dec->unusedCount = 0;
}
samplesDecodingProgress = dec->end;
}
bool reconfigure_audio_decoder(ACSession *ac, int32_t sampling_rate, int8_t channels)
{
if (sampling_rate != ac->ld_sample_rate || channels != ac->ld_channel_count) {
if (current_time_monotonic() - ac->ldrts < 500) {
return false;
}
int status;
OpusDecoder *new_dec = opus_decoder_create(sampling_rate, channels, &status);
if (status != OPUS_OK) {
LOGGER_ERROR(ac->log, "Error while starting audio decoder(%d %d): %s", sampling_rate, channels, opus_strerror(status));
return false;
}
ac->ld_sample_rate = sampling_rate;
ac->ld_channel_count = channels;
ac->ldrts = current_time_monotonic();
opus_decoder_destroy(ac->decoder);
ac->decoder = new_dec;
LOGGER_DEBUG(ac->log, "Reconfigured audio decoder sr: %d cc: %d", sampling_rate, channels);
}
return true;
}
EncodeManager::EncodeManager()
{
_enc = opus_encoder_create(SAMPLE_RATE, 2, OPUS_APPLICATION_VOIP, &_err);
_dec = opus_decoder_create(SAMPLE_RATE, 2, &_err);
opus_encoder_ctl(this->_enc, OPUS_GET_BANDWIDTH(&this->_len));
this->_len = FRAMES_PER_BUFFER;
}
long opus_create(const char* format_parameters, const char** format_parameters_out,
amci_codec_fmt_info_t** format_description) {
opus_state_t* codec_inst;
int error;
unsigned int maxbandwidth = _OPUS_RATE;
int useinbandfec = _OPUS_INBAND_FEC_;
int stereo = 0;
if (format_parameters) {
DBG("OPUS params: >>%s<<.\n", format_parameters);
decode_format_parameters(format_parameters, &maxbandwidth, &useinbandfec, &stereo);
}
codec_inst = (opus_state_t*)malloc(sizeof(opus_state_t));
if (!codec_inst)
return -1;
DBG("OPUS: creating encoder with maxbandwidth=%u, stereo=%s, useinbandfec=%s\n",
maxbandwidth, stereo?"true":"false", useinbandfec?"true":"false");
codec_inst->opus_enc = opus_encoder_create(_OPUS_RATE,1,_OPUS_APPLICATION_,&error);
if (error) {
DBG("OPUS: error %d while creating encoder state.\n", error);
return -1;
}
opus_encoder_ctl(codec_inst->opus_enc, OPUS_SET_FORCE_CHANNELS(stereo ? 2:1));
unsigned int opus_set_bw = _OPUS_RATE;
if (maxbandwidth <= 8000) {
opus_set_bw = OPUS_BANDWIDTH_NARROWBAND;
} else if (maxbandwidth <= 12000) {
opus_set_bw = OPUS_BANDWIDTH_MEDIUMBAND;
} else if (maxbandwidth <= 16000) {
opus_set_bw = OPUS_BANDWIDTH_WIDEBAND;
} else if (maxbandwidth <= 24000) {
opus_set_bw = OPUS_BANDWIDTH_SUPERWIDEBAND;
} else {
opus_set_bw = OPUS_BANDWIDTH_FULLBAND;
}
opus_encoder_ctl(codec_inst->opus_enc, OPUS_SET_MAX_BANDWIDTH(opus_set_bw));
opus_encoder_ctl(codec_inst->opus_enc, OPUS_SET_PACKET_LOSS_PERC(_OPUS_PKT_LOSS_PCT_));
opus_encoder_ctl(codec_inst->opus_enc, OPUS_SET_COMPLEXITY(_OPUS_COMPLEXITY_));
opus_encoder_ctl(codec_inst->opus_enc, OPUS_SET_INBAND_FEC(useinbandfec ? 1:0));
opus_encoder_ctl(codec_inst->opus_enc, OPUS_SET_DTX(_OPUS_DTX_));
codec_inst->opus_dec = opus_decoder_create(_OPUS_RATE,1,&error);
if (error) {
DBG("OPUS: error %d while creating decoder state.\n", error);
opus_encoder_destroy(codec_inst->opus_enc);
return -1;
}
*format_description = opus_fmt_description;
return (long)codec_inst;
}
void Opus::setOptimalFormat(uint32_t sample_rate, uint8_t channels)
{
// Use a SR higher or equal to sample_rate.
// Typical case: 44.1kHz => 48kHz.
unsigned i = 0;
while (i < VALID_SAMPLING_RATE_NUM - 1 and VALID_SAMPLING_RATE[i] < sample_rate)
i++;
sample_rate = VALID_SAMPLING_RATE[i];
// Opus supports 1 or 2 channels.
channels = std::max(std::min(channels, (uint8_t) 2), (uint8_t) 1);
if (not (!encoder_ || !decoder_ || sample_rate != clockRateCur_ || channels != channelsCur_))
return;
clockRateCur_ = sample_rate;
channelsCur_ = channels;
int err;
if (encoder_)
opus_encoder_destroy(encoder_);
encoder_ = opus_encoder_create(sample_rate, channels, OPUS_APPLICATION_VOIP, &err);
if (err)
throw std::runtime_error("opus: could not create encoder");
if (decoder_)
opus_decoder_destroy(decoder_);
lastDecodedFrameSize_ = 0;
decoder_ = opus_decoder_create(sample_rate, channels, &err);
if (err)
throw std::runtime_error("opus: could not create decoder");
}
bool FVoiceDecoderOpus::Init(int32 InSampleRate, int32 InNumChannels)
{
UE_LOG(LogVoiceDecode, Display, TEXT("DecoderVersion: %s"), ANSI_TO_TCHAR(opus_get_version_string()));
SampleRate = InSampleRate;
NumChannels = InNumChannels;
FrameSize = SampleRate / 50;
int32 DecSize = 0;
int32 DecError = 0;
#if USE_UE4_MEM_ALLOC
DecSize = opus_decoder_get_size(NumChannels);
Decoder = (OpusDecoder*)FMemory::Malloc(DecSize);
DecError = opus_decoder_init(Decoder, SampleRate, NumChannels);
#else
Decoder = opus_decoder_create(SampleRate, NumChannels, &DecError);
#endif
if (DecError != OPUS_OK)
{
UE_LOG(LogVoiceDecode, Warning, TEXT("Failed to init Opus Decoder: %s"), ANSI_TO_TCHAR(opus_strerror(DecError)));
Destroy();
}
return DecError == OPUS_OK;
}
static int alloc(struct aucodec_st **stp, struct aucodec *ac,
struct aucodec_prm *encp, struct aucodec_prm *decp,
const char *fmtp)
{
struct aucodec_st *st;
const uint32_t srate = aucodec_srate(ac);
const uint8_t ch = aucodec_ch(ac);
uint32_t ptime = DEFAULT_PTIME;
int use_inbandfec;
int use_dtx;
int err = 0;
int opuserr;
(void)decp;
(void)fmtp;
st = mem_zalloc(sizeof(*st), destructor);
if (!st)
return ENOMEM;
if (encp && encp->ptime)
ptime = encp->ptime;
st->ac = mem_ref(ac);
st->frame_size = srate * ptime / 1000;
st->fsize = 2 * st->frame_size * ch;
/* Encoder */
st->enc = opus_encoder_create(srate, ch, opus.app, &opuserr);
if (!st->enc) {
err = ENOMEM;
goto out;
}
use_inbandfec = 1;
use_dtx = 1;
opus_encoder_ctl(st->enc, OPUS_SET_BITRATE(opus.bitrate));
opus_encoder_ctl(st->enc, OPUS_SET_BANDWIDTH(opus.bandwidth));
opus_encoder_ctl(st->enc, OPUS_SET_VBR(opus.vbr));
opus_encoder_ctl(st->enc, OPUS_SET_COMPLEXITY(opus.complex));
opus_encoder_ctl(st->enc, OPUS_SET_INBAND_FEC(use_inbandfec));
opus_encoder_ctl(st->enc, OPUS_SET_DTX(use_dtx));
/* Decoder */
st->dec = opus_decoder_create(srate, ch, &opuserr);
if (!st->dec) {
err = ENOMEM;
goto out;
}
out:
if (err)
mem_deref(st);
else
*stp = st;
return err;
}
int init_receive_audio(codec_state *cs)
{
int err = OPUS_OK;
cs->audio_decoder = opus_decoder_create(48000, 1, &err);
opus_decoder_init(cs->audio_decoder, 48000, 1);
printf("init audio decoder successful\n");
return 1;
}
// This function must be called before
// any other decoding functions
int nv_opus_init(void) {
int err;
decoder = opus_decoder_create(
nv_opus_get_sample_rate(),
nv_opus_get_channel_count(),
&err);
return err;
}
virtual bool Construct()
{
int error;
if ((m_decoder = opus_decoder_create(m_sampleRate, m_channels, &error)) != NULL)
return true;
PTRACE(1, MY_CODEC_LOG, "Decoder create error " << error << ' ' << opus_strerror(error));
return false;
}
EMSCRIPTEN_KEEPALIVE
Encoder* create_encoder_and_decoder(opus_int32 sample_rate, int channels, int application) {
Encoder* enc = malloc(sizeof(Encoder));
enc->encoder = opus_encoder_create( sample_rate, channels, application, &enc->encoder_error );
enc->decoder = opus_decoder_create( sample_rate, channels, &enc->decoder_error );
return enc;
}
static void ms_opus_dec_preprocess(MSFilter *f) {
int error;
OpusDecData *d = (OpusDecData *)f->data;
d->state = opus_decoder_create(d->samplerate, d->channels, &error);
if (error != OPUS_OK) {
ms_error("Opus decoder creation failed: %s", opus_strerror(error));
}
/* initialise the concealer context */
d->concealer = ms_concealer_context_new(UINT32_MAX);
}
AudioCodec::AudioCodec() {
this->FrameSize = 24000;
this->num_channels = 2;
this->enc = opus_encoder_create(this->FrameSize, this->num_channels,
OPUS_APPLICATION_VOIP, &this->error);
this->dec =
opus_decoder_create(this->FrameSize, this->num_channels, &this->error);
opus_int32 size;
opus_encoder_ctl(enc, OPUS_GET_BANDWIDTH(&size));
this->data_size = size;
}
JNIEXPORT jlong JNICALL
Java_org_jitsi_impl_neomedia_codec_audio_opus_Opus_decoder_1create
(JNIEnv *env, jclass clazz, jint Fs, jint channels)
{
int error;
OpusDecoder *decoder = opus_decoder_create(Fs, channels, &error);
if (OPUS_OK != error)
decoder = 0;
return (jlong) (intptr_t) decoder;
}
int init_audio_decoder(CodecState *cs, uint32_t audio_channels)
{
int rc;
cs->audio_decoder = opus_decoder_create(cs->audio_sample_rate, audio_channels, &rc );
if ( rc != OPUS_OK ) {
LOGGER_ERROR("Error while starting audio decoder: %s", opus_strerror(rc));
return -1;
}
return 0;
}
opus_decoder::opus_decoder()
{
int err;
_opus = opus_decoder_create(SAMPLE_RATE, CHANNELS, &err);
if(err != OPUS_OK)
{
LOG << "opus decoder create error: "; log_opus_error(err);
}
ENSURE(_opus);
}
//function sets up the encoder/decoder structs and buffers, returns required samples/frame
int setup_codecs(){
//create the encoder and decoder state structs
int error = 0;
if(!(enc_state = opus_encoder_create(SAMPLE_RATE,1,APP_MODE,&error)))
printf("Error Creating Encoder: %d\n",error);
if(!(dec_state = opus_decoder_create(SAMPLE_RATE,1,&error)))
printf("Error Creating Decoder: %d\n",error);
opus_encoder_ctl(enc_state, OPUS_SET_COMPLEXITY(QUALITY));
return FRAME_SIZE; //return the frame size (
}
static int init_audio_decoder(CSSession *cs)
{
int rc;
cs->audio_decoder = opus_decoder_create(cs->audio_decoder_sample_rate, cs->audio_decoder_channels, &rc );
if ( rc != OPUS_OK ) {
LOGGER_ERROR("Error while starting audio decoder: %s", opus_strerror(rc));
return -1;
}
return 0;
}
OpusDecoder *tc_opus_create_decoder(int sample_rate, int channels)
{
int err;
OpusDecoder * decoder = opus_decoder_create(sample_rate, channels, &err);
if (err<0) {
pa_log_error("failed to create decoder: %s", opus_strerror(err));
return NULL;
}
return decoder;
}
int init_audio_decoder(CodecState *cs, uint32_t audio_channels)
{
int rc;
cs->audio_decoder = opus_decoder_create(cs->audio_sample_rate, audio_channels, &rc );
if ( rc != OPUS_OK ) {
fprintf(stderr, "Error while starting audio decoder!\n");
return -1;
}
return 0;
}
WebRtc_Word16 WebRtcOpus_CreateDec(OPUS_decinst_t** inst, WebRtc_Word16 samplFreq){
*inst=(OPUS_decinst_t*)malloc(sizeof(OPUS_Dec_Inst_t));
if (*inst==NULL) {
return(-1);
}
int err;
((OPUS_Dec_Inst_t*)*inst)->dec = opus_decoder_create(16000, 1, &err);
if(err != OPUS_OK || ((OPUS_Dec_Inst_t*)*inst)->dec==NULL) {
return (-1);
}
return(0);
}
int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, int channels) {
OpusDecInst* state;
state = (OpusDecInst*) calloc(1, sizeof(OpusDecInst));
if (state) {
int error;
// Always create a 48000 Hz Opus decoder.
state->decoder = opus_decoder_create(48000, channels, &error);
if (error == OPUS_OK && state->decoder != NULL ) {
*inst = state;
return 0;
}
free(state);
state = NULL;
}
return -1;
}
void OpusDecode::start(int CHANNELS, int SAMPLE_RATE)
{
initialized = false;
/* Create a new decoder state. */
int err;
decoder = opus_decoder_create(SAMPLE_RATE, CHANNELS, &err);
if (err<0)
{
qDebug() << "failed to create decoder:" << opus_strerror(err);
return;
}
channels = CHANNELS;
initialized = true;
}
void OpusDecoderClass::startPrivate(int sample_rate, int channels, int frame_size, int max_frame_size)
{
if (m_initialized)
return;
/* Create a new decoder state. */
int err;
m_decoder = opus_decoder_create(sample_rate, channels, &err);
if (err < 0)
{
emit error(QString("Failed to create Opus decoder: %0").arg(opus_strerror(err)));
return;
}
m_channels = channels;
m_frame_size = frame_size;
m_max_frame_size = max_frame_size;
m_initialized = true;
}
JNIEXPORT jlong JNICALL Java_aopus_OpusLibrary_decoderCreate
(JNIEnv *env, jobject obj, jint clockRate, jint channels, jint packetTime)
{
Decoder *decoder = malloc(sizeof* decoder);
decoder->frameSizePerChannel = (clockRate * packetTime) / 1000;
decoder->maxBufferSize = clockRate * channels * packetTime * 2 / 1000; // 2 bytes per sample, 1000 ms per second
decoder->buffer = malloc(decoder->maxBufferSize);
decoder->channels = channels;
decoder->previousPacketInvalid = 0;
// initialize decoder
int error;
decoder->dec = opus_decoder_create(clockRate, channels, &error);
if (error != OPUS_OK)
{
__android_log_print(ANDROID_LOG_ERROR, "fm.libopus", "Could not initialize decoder. %d", error);
return 0;
}
return decoder;
}
static void sdl_renderer_init() {
int rc;
decoder = opus_decoder_create(SAMPLE_RATE, CHANNEL_COUNT, &rc);
SDL_InitSubSystem(SDL_INIT_AUDIO);
SDL_AudioSpec want, have;
SDL_zero(want);
want.freq = SAMPLE_RATE;
want.format = AUDIO_S16LSB;
want.channels = CHANNEL_COUNT;
want.samples = 4096;
dev = SDL_OpenAudioDevice(NULL, 0, &want, &have, SDL_AUDIO_ALLOW_FORMAT_CHANGE);
if (dev == 0) {
printf("Failed to open audio: %s\n", SDL_GetError());
} else {
if (have.format != want.format) // we let this one thing change.
printf("We didn't get requested audio format.\n");
SDL_PauseAudioDevice(dev, 0); // start audio playing.
}
}
