void Reverb::initialize(const nsTArray<const float*>& impulseResponseBuffer,
size_t impulseResponseBufferLength,
size_t maxFFTSize, bool useBackgroundThreads)
{
m_impulseResponseLength = impulseResponseBufferLength;
// The reverb can handle a mono impulse response and still do stereo processing
size_t numResponseChannels = impulseResponseBuffer.Length();
m_convolvers.SetCapacity(numResponseChannels);
int convolverRenderPhase = 0;
for (size_t i = 0; i < numResponseChannels; ++i) {
const float* channel = impulseResponseBuffer[i];
size_t length = impulseResponseBufferLength;
nsAutoPtr<ReverbConvolver> convolver(new ReverbConvolver(channel, length, maxFFTSize, convolverRenderPhase, useBackgroundThreads));
m_convolvers.AppendElement(convolver.forget());
convolverRenderPhase += WEBAUDIO_BLOCK_SIZE;
}
// For "True" stereo processing we allocate a temporary buffer to avoid repeatedly allocating it in the process() method.
// It can be bad to allocate memory in a real-time thread.
if (numResponseChannels == 4) {
m_tempBuffer.AllocateChannels(2);
WriteZeroesToAudioBlock(&m_tempBuffer, 0, WEBAUDIO_BLOCK_SIZE);
}
}
/**
* Fill aOutput with as many zero frames as we can, and advance
* aOffsetWithinBlock and aCurrentPosition based on how many frames we write.
* This will never advance aOffsetWithinBlock past WEBAUDIO_BLOCK_SIZE or
* aCurrentPosition past aMaxPos. This function knows when it needs to
* allocate the output buffer, and also optimizes the case where it can avoid
* memory allocations.
*/
void FillWithZeroes(AudioChunk* aOutput,
uint32_t aChannels,
uint32_t* aOffsetWithinBlock,
TrackTicks* aCurrentPosition,
TrackTicks aMaxPos)
{
uint32_t numFrames = std::min(WEBAUDIO_BLOCK_SIZE - *aOffsetWithinBlock,
uint32_t(aMaxPos - *aCurrentPosition));
if (numFrames == WEBAUDIO_BLOCK_SIZE) {
aOutput->SetNull(numFrames);
} else {
if (aOutput->IsNull()) {
AllocateAudioBlock(aChannels, aOutput);
}
WriteZeroesToAudioBlock(aOutput, *aOffsetWithinBlock, numFrames);
}
*aOffsetWithinBlock += numFrames;
*aCurrentPosition += numFrames;
}
/**
* Fill aOutput with as many zero frames as we can, and advance
* aOffsetWithinBlock and aCurrentPosition based on how many frames we write.
* This will never advance aOffsetWithinBlock past WEBAUDIO_BLOCK_SIZE or
* aCurrentPosition past aMaxPos. This function knows when it needs to
* allocate the output buffer, and also optimizes the case where it can avoid
* memory allocations.
*/
void FillWithZeroes(AudioBlock* aOutput,
uint32_t aChannels,
uint32_t* aOffsetWithinBlock,
StreamTime* aCurrentPosition,
StreamTime aMaxPos)
{
MOZ_ASSERT(*aCurrentPosition < aMaxPos);
uint32_t numFrames =
std::min<StreamTime>(WEBAUDIO_BLOCK_SIZE - *aOffsetWithinBlock,
aMaxPos - *aCurrentPosition);
if (numFrames == WEBAUDIO_BLOCK_SIZE || !aChannels) {
aOutput->SetNull(numFrames);
} else {
if (*aOffsetWithinBlock == 0) {
aOutput->AllocateChannels(aChannels);
}
WriteZeroesToAudioBlock(aOutput, *aOffsetWithinBlock, numFrames);
}
*aOffsetWithinBlock += numFrames;
*aCurrentPosition += numFrames;
}
