void AudioNode::processIfNecessary(size_t framesToProcess)
{
ASSERT(context()->isAudioThread());
if (!isInitialized())
return;
// Ensure that we only process once per rendering quantum.
// This handles the "fanout" problem where an output is connected to multiple inputs.
// The first time we're called during this time slice we process, but after that we don't want to re-process,
// instead our output(s) will already have the results cached in their bus;
double currentTime = context()->currentTime();
if (m_lastProcessingTime != currentTime) {
m_lastProcessingTime = currentTime; // important to first update this time because of feedback loops in the rendering graph
pullInputs(framesToProcess);
bool silentInputs = inputsAreSilent();
if (!silentInputs)
m_lastNonSilentTime = (context()->currentSampleFrame() + framesToProcess) / static_cast<double>(m_sampleRate);
if (silentInputs && propagatesSilence())
silenceOutputs();
else {
process(framesToProcess);
unsilenceOutputs();
}
}
}
void OutputDeviceNodeXAudio::submitNextBuffer()
{
auto ctx = getContext();
if( ! ctx )
return;
lock_guard<mutex> lock( ctx->getMutex() );
// verify context still exists, since its destructor may have been holding the lock
ctx = getContext();
if( ! ctx )
return;
ctx->preProcess();
auto internalBuffer = getInternalBuffer();
internalBuffer->zero();
pullInputs( internalBuffer );
if( checkNotClipping() )
internalBuffer->zero();
if( getNumChannels() == 2 )
dsp::interleaveStereoBuffer( internalBuffer, &mBufferInterleaved );
HRESULT hr = mSourceVoice->SubmitSourceBuffer( &mXAudioBuffer );
CI_ASSERT( hr == S_OK );
ctx->postProcess();
}
void OutputDeviceNodeWebAudio::renderInputs( emscripten::val e )
{
auto outputBuffer = e["outputBuffer"];
auto inputBuffer = e["inputBuffer"];
auto ctx = getContext();
if( ! ctx )
{
CI_LOG_I( "can't get context" );
return;
}
// this technically doesn't do anything but leave it here for now.
lock_guard<mutex> lock( ctx->getMutex() );
ctx->preProcess();
auto internalBuffer = getInternalBuffer();
internalBuffer->zero();
pullInputs( internalBuffer );
if( checkNotClipping() )
{
internalBuffer->zero();
}
const size_t numFrames = internalBuffer->getNumFrames();
const size_t numChannels = internalBuffer->getNumChannels();
//dsp::interleave( internalBuffer->getData(), outputBuffer, numFrames, numChannels, numFrames );
ctx->postProcess();
// =========================================== //
// make sure that the ScriptNodeProcessor's bufferSize prop is set properly.
mImpl->setOutputBufferSize( numFrames );
// output array is what gets directed towards speakers
val output = outputBuffer.call<val>( "getChannelData", 0 );
// input is only going to be populated when using native WebAudio nodes.
val input = inputBuffer.call<val>( "getChannelData",0 );
// check first 100 values to see if input buffer is filled or empty(aka the value is 0). If empty, we won't bother processing,
bool mInputIsEmpty = true;
for( int a = 0; a < 100; ++a )
{
if( input[a].as<float>() != 0.0 )
{
mInputIsEmpty = false;
}
}
// get internal cinder data
float * data = static_cast<float*>( internalBuffer->getData() );
float * finalData;
float * idata;
if( !mInputIsEmpty )
{
// will hold input data
std::vector<float> _inputData;
// copy Float32Array(aka "input") into _inputData vector
ci::em::helpers::copyToVector( input,_inputData );
// convert to float* so we can pass information to dsp methods
idata = &_inputData[0];
// add input data to any cinder data
dsp::add(idata,data,finalData,numFrames);
}
else{
// if no input data just pass through
finalData = data;
}
// loop through and set all info from finalData pointer onto the output buffer.
for( int i = 0; i < numFrames; ++i )
{
output.set<float>( i,finalData[i] );
}
// release finalData pointer.
// releasing these causes things to crash :(
//free(finalData);
//free(idata);
}