void AsyncAudioDecoder::DecodingTask::notifyComplete()
{
if (audioBuffer() && successCallback())
successCallback()->handleEvent(audioBuffer());
else if (errorCallback())
errorCallback()->handleEvent(audioBuffer());
// Our ownership was given up in AsyncAudioDecoder::runLoop()
// Make sure to clean up here.
delete this;
}
void Connection::processBuffer() {
Buffer* buffer;
char* nextHeader;
char* nextBuffer;
while (!queue.isEmpty() ){
buffer=queue.dequeue();
nextHeader=buffer->getHeader();
nextBuffer=buffer->getBuffer();
//qDebug() << "processBuffer " << nextHeader[0];
// emit a signal to show what buffer we have
if(nextHeader[0]==SPECTRUM_BUFFER){
emit spectrumBuffer(nextHeader,nextBuffer);
}
else if(nextHeader[0]==AUDIO_BUFFER) {
emit audioBuffer(nextHeader,nextBuffer);
} else if(nextHeader[0]==BANDSCOPE_BUFFER) {
//qDebug() << "socketData: bandscope";
emit bandscopeBuffer(nextHeader,nextBuffer);
} else {
qDebug() << "Connection::socketData: invalid header: " << nextHeader[0];
}
}
}
void LAudioAppComponent::getNextAudioBlock( const AudioSourceChannelInfo& bufferToFill ) {
if(stopped
|| !audioOpen
|| !hasCallback("getNextAudioBlock")
|| (audioOpen && audioSourcePlayer.getCurrentSource()==nullptr) )
{
bufferToFill.clearActiveBufferRegion();
return;
}
if(hasCallback("getNextAudioBlock")) {
MessageManagerLock mml (Thread::getCurrentThread());
if (! mml.lockWasGained()) {
DBG("CAN'T GET LOCK");
return; // another thread is trying to kill us!
}
LAudioSampleBuffer audioBuffer(Ls, *bufferToFill.buffer);
callback("getNextAudioBlock", 0, {
bufferToFill.startSample,
bufferToFill.numSamples,
bufferToFill.buffer->getNumChannels(),
new LRefBase("AudioSampleBuffer", &audioBuffer)
});
if(volume>-1) {
if(volume) bufferToFill.buffer->applyGain(volume);
else bufferToFill.clearActiveBufferRegion();
}
}
}
void Connection::processBuffer() {
Buffer* buffer;
char* nextHeader;
char* nextBuffer;
// We only want to mute the audio (actually not for full duplex)
// the spectrum display should show the microphone waveform
/*
if(muted) { // If Rx muted, clear queue and don't process buffer - gvj
queue.clear();
}
*/
while (!queue.isEmpty()){
buffer=queue.dequeue();
nextHeader=buffer->getHeader();
nextBuffer=buffer->getBuffer();
// emit a signal to show what buffer we have
//qDebug() << "processBuffer " << nextHeader[0];
if(nextHeader[0]==SPECTRUM_BUFFER){
emit spectrumBuffer(nextHeader,nextBuffer);
}
else if(nextHeader[0]==AUDIO_BUFFER) {
// need to add a duplex state
if(!muted) emit audioBuffer(nextHeader,nextBuffer);
} else if(nextHeader[0]==BANDSCOPE_BUFFER) {
//qDebug() << "socketData: bandscope";
emit bandscopeBuffer(nextHeader,nextBuffer);
} else {
qDebug() << "Connection::socketData: invalid header: " << nextHeader[0];
queue.clear();
}
}
}
void KeyAnalysisThread::analyzeSingleKey()
{
DBG("Thread: " << this->getThreadName() << " analyzing single key of: " << fileToAnalyze->getUUID());
//skip already-analyzed files
if (fileToAnalyze->isFileAnalyzed() == "Yes")
{
DBG("Skipping analysis of " << fileToAnalyze->getFileName() << " : Already analyzed.");
return;
}
//DBG("File being analyzed: " << fileToAnalyze.getFileToBeAnalyzed().getFileName());
// r> audioFormatReader = audioFormatManager.createReaderFor(fileToAnalyze.getFileToBeAnalyzed());
size_t numSamples = audioFormatReader->lengthInSamples;
size_t numChannels = audioFormatReader->numChannels;
size_t sampleRate = audioFormatReader->sampleRate;
//size_t numFrames = numSamples / numChannels;
//DBG("Samples:" + String(numSamples) + " Channels:" + String(numChannels) + " Frames:" + String(numFrames));
audioData.addToSampleCount(numSamples*numChannels);
audioData.setChannels(numChannels);
audioData.setFrameRate(sampleRate);
AudioBuffer<float> audioBuffer(numChannels, numSamples);
// #TODO(Casey): Fix either audiobuffer or audioData to allow direct move of sample array (Performance)
// #TODO(Casey): integrate libsndfile as primary choice
//DBG("Reading wav to buffer");
audioFormatReader->read(&audioBuffer, 0, numSamples, 0, true, true);
//DBG("Copying buffer samples to audioData");
for (size_t i = 0; i < numSamples; ++i)
{
//DBG("On sample " + String(i) + " of " + String(numSamples));
for (size_t currChannel = 0; currChannel < numChannels; ++currChannel)
{
audioData.setSampleByFrame(i, currChannel, audioBuffer.getSample(currChannel, i));
}
//audioData.setSampleByFrame(i, 1, audioBuffer.getSample(1, i));
}
//DBG("Detecting key...");
static KeyFinder::KeyFinder keyFinder;
KeyFinder::key_t key = keyFinder.keyOfAudio(audioData);
//KeyFinder::key_t key = KeyFinder::key_t::C_MAJOR;
fileToAnalyze->setDetectedKey(key);
String foundKey(fileToAnalyze->getDetectedKeyAsString());
DBG("Detected key in analyzeAudioFiles: " + foundKey);
}
// ////////////////////////////////////////////////////////////////////
//
// ////////////////////////////////////////////////////////////////////
boost::shared_ptr<IAudioBuffer> DirectSound8Audio::VInitAudioBuffer(shared_ptr<SoundResHandle> soundResource)//const
{
boost::shared_ptr<IAudioBuffer> null;
const char *fileExtension = FindExtFromSoundType(soundResource->GetSoundType());
if(m_pDS == NULL) {
return (null);
}
switch(soundResource->GetSoundType()) {
case SOUND_TYPE_OGG:
case SOUND_TYPE_WAVE:
// We support WAVs and OGGs
break;
case SOUND_TYPE_MP3:
case SOUND_TYPE_MIDI: //If it's a midi file, then do nothin at this time... maybe we will support this in the future
GF_LOG_TRACE_ERR("DirectSound8Audio::VInitAudioBuffer()", "MP3s and MIDI are not supported");
return (null);
break;
default:
GF_LOG_TRACE_ERR("DirectSound8Audio::VInitAudioBuffer()", "Unknown sound file");
return (null);
break;
}//end switch
LPDIRECTSOUNDBUFFER sampleHandle;
// Create the direct sound buffer, and only request the flags needed
// since each requires some overhead and limits if the buffer can
// be hardware accelerated
DSBUFFERDESC dsbd;
ZeroMemory(&dsbd, sizeof(DSBUFFERDESC));
dsbd.dwSize = sizeof(DSBUFFERDESC);
dsbd.dwFlags = DSBCAPS_CTRLVOLUME;
dsbd.dwBufferBytes = soundResource->GetPCMBufferSize();
dsbd.guid3DAlgorithm = GUID_NULL;
dsbd.lpwfxFormat = const_cast<WAVEFORMATEX *>(soundResource->GetFormat());
HRESULT hr;
if(FAILED(hr = m_pDS->CreateSoundBuffer(&dsbd, &sampleHandle, NULL))) {
GF_LOG_TRACE_ERR("DirectSound8Audio::VInitAudioBuffer()", "");
return (null);
}
// Add handle to the list
boost::shared_ptr<IAudioBuffer> audioBuffer(GCC_NEW DirectSound8AudioBuffer(sampleHandle, soundResource));
if(audioBuffer) {
m_AllSamples.insert(m_AllSamples.begin(), audioBuffer);
}
return (audioBuffer);
}
status_t FMA2DPWriter::allocateBufferPool()
{
Mutex::Autolock lock(mFreeQLock);
for (int i = 0; i < BUFFER_POOL_SIZE; ++i) {
int *buffer = (int*)malloc(mBufferSize);
if(buffer){
audioBufferstruct audioBuffer(buffer,mBufferSize);
mFreeQ.push_back(audioBuffer);
}
else{
ALOGE("fatal:failed to alloate buffer pool");
return NO_INIT;
}
}
return OK;
}
