unsigned SoundSourceOpus::read(volatile unsigned long size, const SAMPLE * destination) {
if (size % 2 != 0) {
qDebug() << "SoundSourceOpus got non-even size in read.";
size--;
}
// SAMPLE and opus_int16 are mostly same
// So just make pointer and hope for the best
opus_int16 *l_iDest = (opus_int16 *) destination;
unsigned int l_iNeeded = size;
unsigned int l_iReaded = 0;
unsigned int l_iRet=0;
// loop until requested number of samples has been retrieved
while (l_iNeeded > 0) {
// read samples into buffer
//ret = op_read_stereo(m_ptrOpusFile, l_iPcm, sizeof(l_iPcm)/sizeof(*l_iPcm));
l_iRet = op_read_stereo(m_ptrOpusFile, l_iDest, l_iNeeded);
if (l_iRet <= 0) {
// An error or EOF occured, break out and return what we have sofar.
break;
}
l_iNeeded -= l_iRet * 2;
l_iReaded += l_iRet * 2;
l_iDest += l_iRet * 2;
}
return l_iReaded;
}
int OpusDecoder::FillBuffer(uint8_t* buffer, int length) {
if (!oof)
return -1;
// op_read_stereo doesn't overwrite the buffer completely, must be cleared to prevent noise
memset(buffer, '\0', length);
// Use a 16bit buffer because op_read_stereo works on one
int length_16 = length / 2;
opus_int16* buffer_16 = reinterpret_cast<opus_int16*>(buffer);
int read = 0;
int to_read = length_16;
do {
read = op_read_stereo(oof, buffer_16 + (length_16 - to_read), to_read);
// stop decoding when error or end of file
if (read <= 0)
break;
// "read" contains number of samples per channel and the function filled 2 channels
to_read -= read * 2;
} while (to_read > 0);
if (read == 0)
finished = true;
if (read < 0) {
return -1;
}
// Return amount of read bytes in the 8 bit what the audio decoder expects
return (length_16 - to_read) * 2;
}
bool CSoundFile::ReadOpusSample(SAMPLEINDEX sample, FileReader &file)
{
file.Rewind();
#if defined(MPT_WITH_OPUSFILE)
int rate = 0;
int channels = 0;
std::vector<int16> raw_sample_data;
FileReader initial = file.GetChunk(65536); // 512 is recommended by libopusfile
if(op_test(NULL, initial.GetRawData<unsigned char>(), initial.GetLength()) != 0)
{
return false;
}
OggOpusFile *of = op_open_memory(file.GetRawData<unsigned char>(), file.GetLength(), NULL);
if(!of)
{
return false;
}
rate = 48000;
channels = op_channel_count(of, -1);
if(rate <= 0 || channels <= 0)
{
op_free(of);
of = NULL;
return false;
}
if(channels > 2 || op_link_count(of) != 1)
{
// We downmix multichannel to stereo as recommended by Opus specification in
// case we are not able to handle > 2 channels.
// We also decode chained files as stereo even if they start with a mono
// stream, which simplifies handling of link boundaries for us.
channels = 2;
}
std::vector<int16> decodeBuf(120 * 48000 / 1000); // 120ms (max Opus packet), 48kHz
bool eof = false;
while(!eof)
{
int framesRead = 0;
if(channels == 2)
{
framesRead = op_read_stereo(of, &(decodeBuf[0]), static_cast<int>(decodeBuf.size()));
} else if(channels == 1)
{
framesRead = op_read(of, &(decodeBuf[0]), static_cast<int>(decodeBuf.size()), NULL);
}
if(framesRead > 0)
{
raw_sample_data.insert(raw_sample_data.end(), decodeBuf.begin(), decodeBuf.begin() + (framesRead * channels));
} else if(framesRead == 0)
{
eof = true;
} else if(framesRead == OP_HOLE)
{
// continue
} else
{
// other errors are fatal, stop decoding
eof = true;
}
}
op_free(of);
of = NULL;
if(raw_sample_data.empty())
{
return false;
}
DestroySampleThreadsafe(sample);
strcpy(m_szNames[sample], "");
Samples[sample].Initialize();
Samples[sample].nC5Speed = rate;
Samples[sample].nLength = raw_sample_data.size() / channels;
Samples[sample].uFlags.set(CHN_16BIT);
Samples[sample].uFlags.set(CHN_STEREO, channels == 2);
Samples[sample].AllocateSample();
std::copy(raw_sample_data.begin(), raw_sample_data.end(), Samples[sample].pSample16);
Samples[sample].Convert(MOD_TYPE_IT, GetType());
Samples[sample].PrecomputeLoops(*this, false);
return Samples[sample].pSample != nullptr;
#else // !MPT_WITH_OPUSFILE
MPT_UNREFERENCED_PARAMETER(sample);
MPT_UNREFERENCED_PARAMETER(file);
return false;
#endif // MPT_WITH_OPUSFILE
}
void VoiceMessagesLoader::onLoad(AudioData *audio) {
bool started = false;
int32 audioindex = -1;
Loader *l = 0;
Loaders::iterator j = _loaders.end();
{
QMutexLocker lock(&voicemsgsMutex);
VoiceMessages *voice = audioVoice();
if (!voice) return;
for (int32 i = 0; i < AudioVoiceMsgSimultaneously; ++i) {
VoiceMessages::Msg &m(voice->_data[i]);
if (m.audio != audio || !m.loading) continue;
audioindex = i;
j = _loaders.find(audio);
if (j != _loaders.end() && (j.value()->fname != m.fname || j.value()->data.size() != m.data.size())) {
delete j.value();
_loaders.erase(j);
j = _loaders.end();
}
if (j == _loaders.end()) {
l = (j = _loaders.insert(audio, new Loader())).value();
l->fname = m.fname;
l->data = m.data;
int ret;
if (m.data.isEmpty()) {
l->file = op_open_file(m.fname.toUtf8().constData(), &ret);
} else {
l->file = op_open_memory((const unsigned char*)m.data.constData(), m.data.size(), &ret);
}
if (!l->file) {
LOG(("Audio Error: op_open_file failed for '%1', data size '%2', error code %3").arg(m.fname).arg(m.data.size()).arg(ret));
m.state = VoiceMessageStopped;
return loadError(j);
}
ogg_int64_t duration = op_pcm_total(l->file, -1);
if (duration < 0) {
LOG(("Audio Error: op_pcm_total failed to get full duration for '%1', data size '%2', error code %3").arg(m.fname).arg(m.data.size()).arg(duration));
m.state = VoiceMessageStopped;
return loadError(j);
}
m.duration = duration;
m.skipStart = 0;
m.skipEnd = duration;
m.position = 0;
m.started = 0;
started = true;
} else {
if (!m.skipEnd) continue;
l = j.value();
}
break;
}
}
if (j == _loaders.end()) {
LOG(("Audio Error: trying to load part of audio, that is not playing at the moment"));
emit error(audio);
return;
}
if (started) {
l->pcm_offset = op_pcm_tell(l->file);
l->pcm_print_offset = l->pcm_offset - AudioVoiceMsgFrequency;
}
bool finished = false;
DEBUG_LOG(("Audio Info: reading buffer for file '%1', data size '%2', current pcm_offset %3").arg(l->fname).arg(l->data.size()).arg(l->pcm_offset));
QByteArray result;
int64 samplesAdded = 0;
while (result.size() < AudioVoiceMsgBufferSize) {
opus_int16 pcm[AudioVoiceMsgFrequency * AudioVoiceMsgChannels];
int ret = op_read_stereo(l->file, pcm, sizeof(pcm) / sizeof(*pcm));
if (ret < 0) {
{
QMutexLocker lock(&voicemsgsMutex);
VoiceMessages *voice = audioVoice();
if (voice) {
VoiceMessages::Msg &m(voice->_data[audioindex]);
if (m.audio == audio) {
m.state = VoiceMessageStopped;
}
}
}
LOG(("Audio Error: op_read_stereo failed, error code %1").arg(ret));
return loadError(j);
}
int li = op_current_link(l->file);
if (li != l->prev_li) {
const OpusHead *head = op_head(l->file, li);
const OpusTags *tags = op_tags(l->file, li);
for (int32 ci = 0; ci < tags->comments; ++ci) {
const char *comment = tags->user_comments[ci];
if (opus_tagncompare("METADATA_BLOCK_PICTURE", 22, comment) == 0) {
OpusPictureTag pic;
int err = opus_picture_tag_parse(&pic, comment);
if (err >= 0) {
opus_picture_tag_clear(&pic);
}
}
}
if (!op_seekable(l->file)) {
l->pcm_offset = op_pcm_tell(l->file) - ret;
}
}
if (li != l->prev_li || l->pcm_offset >= l->pcm_print_offset + AudioVoiceMsgFrequency) {
l->pcm_print_offset = l->pcm_offset;
}
l->pcm_offset = op_pcm_tell(l->file);
if (!ret) {
DEBUG_LOG(("Audio Info: read completed"));
finished = true;
break;
}
result.append((const char*)pcm, sizeof(*pcm) * ret * AudioVoiceMsgChannels);
l->prev_li = li;
samplesAdded += ret;
{
QMutexLocker lock(&voicemsgsMutex);
VoiceMessages *voice = audioVoice();
if (!voice) return;
VoiceMessages::Msg &m(voice->_data[audioindex]);
if (m.audio != audio || !m.loading || m.fname != l->fname || m.data.size() != l->data.size()) {
LOG(("Audio Error: playing changed while loading"));
m.state = VoiceMessageStopped;
return loadError(j);
}
}
}
QMutexLocker lock(&voicemsgsMutex);
VoiceMessages *voice = audioVoice();
if (!voice) return;
VoiceMessages::Msg &m(voice->_data[audioindex]);
if (m.audio != audio || !m.loading || m.fname != l->fname || m.data.size() != l->data.size()) {
LOG(("Audio Error: playing changed while loading"));
m.state = VoiceMessageStopped;
return loadError(j);
}
if (started) {
if (m.source) {
alSourceStop(m.source);
for (int32 i = 0; i < 3; ++i) {
if (m.samplesCount[i]) {
alSourceUnqueueBuffers(m.source, 1, m.buffers + i);
m.samplesCount[i] = 0;
}
}
m.nextBuffer = 0;
}
}
if (samplesAdded) {
if (!m.source) {
alGenSources(1, &m.source);
alSourcef(m.source, AL_PITCH, 1.f);
alSourcef(m.source, AL_GAIN, 1.f);
alSource3f(m.source, AL_POSITION, 0, 0, 0);
alSource3f(m.source, AL_VELOCITY, 0, 0, 0);
alSourcei(m.source, AL_LOOPING, 0);
}
if (!m.buffers[m.nextBuffer]) alGenBuffers(3, m.buffers);
if (!_checkALError()) {
m.state = VoiceMessageStopped;
return loadError(j);
}
if (m.samplesCount[m.nextBuffer]) {
alSourceUnqueueBuffers(m.source, 1, m.buffers + m.nextBuffer);
m.skipStart += m.samplesCount[m.nextBuffer];
}
m.samplesCount[m.nextBuffer] = samplesAdded;
alBufferData(m.buffers[m.nextBuffer], AL_FORMAT_STEREO16, result.constData(), result.size(), AudioVoiceMsgFrequency);
alSourceQueueBuffers(m.source, 1, m.buffers + m.nextBuffer);
m.skipEnd -= samplesAdded;
m.nextBuffer = (m.nextBuffer + 1) % 3;
if (!_checkALError()) {
m.state = VoiceMessageStopped;
return loadError(j);
}
} else {
finished = true;
}
if (finished) {
m.skipEnd = 0;
m.duration = m.skipStart + m.samplesCount[0] + m.samplesCount[1] + m.samplesCount[2];
}
m.loading = false;
if (m.state == VoiceMessageResuming || m.state == VoiceMessagePlaying || m.state == VoiceMessageStarting) {
ALint state = AL_INITIAL;
alGetSourcei(m.source, AL_SOURCE_STATE, &state);
if (_checkALError()) {
if (state != AL_PLAYING) {
alSourcePlay(m.source);
emit needToCheck();
}
}
}
}
