void UpdateSampleRateIfNeeded(AudioNodeStream* aStream, uint32_t aChannels)
{
if (mPlaybackRateTimeline.HasSimpleValue()) {
mPlaybackRate = mPlaybackRateTimeline.GetValue();
} else {
mPlaybackRate = mPlaybackRateTimeline.GetValueAtTime(aStream->GetCurrentPosition());
}
// Make sure the playback rate and the doppler shift are something
// our resampler can work with.
if (ComputeFinalOutSampleRate(aStream->SampleRate()) == 0) {
mPlaybackRate = 1.0;
mDopplerShift = 1.0;
}
uint32_t currentOutSampleRate, currentInSampleRate;
if (ShouldResample(aStream->SampleRate())) {
SpeexResamplerState* resampler = Resampler(aStream, aChannels);
speex_resampler_get_rate(resampler, ¤tInSampleRate, ¤tOutSampleRate);
uint32_t finalSampleRate = ComputeFinalOutSampleRate(aStream->SampleRate());
if (currentOutSampleRate != finalSampleRate) {
speex_resampler_set_rate(resampler, currentInSampleRate, finalSampleRate);
}
}
}
/**
* Copy as many frames as possible from the source buffer to aOutput, and
* advance aOffsetWithinBlock and aCurrentPosition based on how many frames
* we copy. This will never advance aOffsetWithinBlock past
* WEBAUDIO_BLOCK_SIZE, or aCurrentPosition past mStop. It takes data from
* the buffer at aBufferOffset, and never takes more data than aBufferMax.
* This function knows when it needs to allocate the output buffer, and also
* optimizes the case where it can avoid memory allocations.
*/
void CopyFromBuffer(AudioNodeStream* aStream,
AudioChunk* aOutput,
uint32_t aChannels,
uint32_t* aOffsetWithinBlock,
TrackTicks* aCurrentPosition,
uint32_t aBufferOffset,
uint32_t aBufferMax)
{
MOZ_ASSERT(*aCurrentPosition < mStop);
uint32_t numFrames =
std::min<TrackTicks>(std::min(WEBAUDIO_BLOCK_SIZE - *aOffsetWithinBlock,
aBufferMax - aBufferOffset),
mStop - *aCurrentPosition);
if (numFrames == WEBAUDIO_BLOCK_SIZE && !ShouldResample(aStream->SampleRate())) {
BorrowFromInputBuffer(aOutput, aChannels, aBufferOffset);
*aOffsetWithinBlock += numFrames;
*aCurrentPosition += numFrames;
mPosition += numFrames;
} else {
if (aOutput->IsNull()) {
MOZ_ASSERT(*aOffsetWithinBlock == 0);
AllocateAudioBlock(aChannels, aOutput);
}
if (!ShouldResample(aStream->SampleRate())) {
CopyFromInputBuffer(aOutput, aChannels, aBufferOffset, *aOffsetWithinBlock, numFrames);
*aOffsetWithinBlock += numFrames;
*aCurrentPosition += numFrames;
mPosition += numFrames;
} else {
uint32_t framesRead, framesWritten, availableInInputBuffer;
availableInInputBuffer = aBufferMax - aBufferOffset;
CopyFromInputBufferWithResampling(aStream, aOutput, aChannels, aBufferOffset, *aOffsetWithinBlock, availableInInputBuffer, framesRead, framesWritten);
*aOffsetWithinBlock += framesWritten;
*aCurrentPosition += framesRead;
mPosition += framesRead;
}
}
}
/**
* Copy as many frames as possible from the source buffer to aOutput, and
* advance aOffsetWithinBlock and aCurrentPosition based on how many frames
* we write. This will never advance aOffsetWithinBlock past
* WEBAUDIO_BLOCK_SIZE, or aCurrentPosition past mStop. It takes data from
* the buffer at aBufferOffset, and never takes more data than aBufferMax.
* This function knows when it needs to allocate the output buffer, and also
* optimizes the case where it can avoid memory allocations.
*/
void CopyFromBuffer(AudioNodeStream* aStream,
AudioChunk* aOutput,
uint32_t aChannels,
uint32_t* aOffsetWithinBlock,
TrackTicks* aCurrentPosition,
int32_t aBufferMax)
{
MOZ_ASSERT(*aCurrentPosition < mStop);
uint32_t numFrames =
std::min(std::min<TrackTicks>(WEBAUDIO_BLOCK_SIZE - *aOffsetWithinBlock,
aBufferMax - mBufferPosition),
mStop - *aCurrentPosition);
if (numFrames == WEBAUDIO_BLOCK_SIZE && !ShouldResample(aStream->SampleRate())) {
MOZ_ASSERT(mBufferPosition < aBufferMax);
BorrowFromInputBuffer(aOutput, aChannels);
*aOffsetWithinBlock += numFrames;
*aCurrentPosition += numFrames;
mBufferPosition += numFrames;
} else {
if (*aOffsetWithinBlock == 0) {
AllocateAudioBlock(aChannels, aOutput);
}
if (!ShouldResample(aStream->SampleRate())) {
MOZ_ASSERT(mBufferPosition < aBufferMax);
CopyFromInputBuffer(aOutput, aChannels, *aOffsetWithinBlock, numFrames);
*aOffsetWithinBlock += numFrames;
*aCurrentPosition += numFrames;
mBufferPosition += numFrames;
} else {
uint32_t framesWritten;
CopyFromInputBufferWithResampling(aStream, aOutput, aChannels, *aOffsetWithinBlock, framesWritten, aBufferMax);
*aOffsetWithinBlock += framesWritten;
*aCurrentPosition += framesWritten;
}
}
}
