void cmdMusic(const std::string& args) {
// terrible hack for testing audio
// decode the entire file into a big buffer all at once
OggOpusFile* opusFile;
if (args.length() == 0) {
opusFile = op_open_memory(
static_cast<const uint8_t*>(EMBED_DATA(Who_Likes_to_Party_Kevin_MacLeod_incompetech_opus)),
EMBED_SIZE(Who_Likes_to_Party_Kevin_MacLeod_incompetech_opus),
NULL);
} else {
opusFile = op_open_file(args.c_str(), NULL);
}
if (!opusFile) {
narf::console->println("Failed to open music file " + args);
return;
}
auto newMusicSize = op_pcm_total(opusFile, -1) * 2; // stereo
auto newMusicSamples = new float[newMusicSize];
size_t decoded = 0;
while (decoded < newMusicSize) {
auto rc = op_read_float_stereo(opusFile, newMusicSamples + decoded, newMusicSize - decoded);
if (rc < 0) {
narf::console->println("opusfile decode failed");
decoded = 0;
break;
}
decoded += rc * 2; // return code is number of samples per channel, and we are decoding in stereo
}
if (decoded != newMusicSize) {
narf::console->println("opusfile decode returned wrong number of samples (got " + std::to_string(decoded) + ", expected " + std::to_string(newMusicSize) + ")");
delete[] newMusicSamples;
newMusicSamples = nullptr;
newMusicSize = 0;
}
SDL_LockMutex(musicMutex);
if (musicSamples) {
delete[] musicSamples;
}
musicSamples = newMusicSamples;
musicSize = newMusicSize;
musicCursor = 0;
SDL_UnlockMutex(musicMutex);
op_free(opusFile);
}
int CSound::DecodeOpus(int SampleID, const void *pData, unsigned DataSize)
{
if(SampleID == -1 || SampleID >= NUM_SAMPLES)
return -1;
CSample *pSample = &m_aSamples[SampleID];
OggOpusFile *OpusFile = op_open_memory((const unsigned char *) pData, DataSize, NULL);
if (OpusFile)
{
int NumChannels = op_channel_count(OpusFile, -1);
int NumSamples = op_pcm_total(OpusFile, -1); // per channel!
pSample->m_Channels = NumChannels;
if(pSample->m_Channels > 2)
{
dbg_msg("sound/opus", "file is not mono or stereo.");
return -1;
}
pSample->m_pData = (short *)mem_alloc(NumSamples * sizeof(short) * NumChannels, 1);
int Read;
int Pos = 0;
while (Pos < NumSamples)
{
Read = op_read(OpusFile, pSample->m_pData + Pos*NumChannels, NumSamples*NumChannels, NULL);
Pos += Read;
}
pSample->m_NumFrames = NumSamples; // ?
pSample->m_Rate = 48000;
pSample->m_LoopStart = -1;
pSample->m_LoopEnd = -1;
pSample->m_PausedAt = 0;
}
else
{
dbg_msg("sound/opus", "failed to decode sample");
return -1;
}
return SampleID;
}
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();
}
}
}
}
