void AudioBufferAddWithScale(const float* aInput,
float aScale,
float* aOutput,
uint32_t aSize)
{
#ifdef BUILD_ARM_NEON
if (mozilla::supports_neon()) {
AudioBufferAddWithScale_NEON(aInput, aScale, aOutput, aSize);
return;
}
#endif
#ifdef USE_SSE2
if (mozilla::supports_sse2()) {
if (aScale == 1.0f) {
while (aSize && (!IS_ALIGNED16(aInput) || !IS_ALIGNED16(aOutput))) {
*aOutput += *aInput;
++aOutput;
++aInput;
--aSize;
}
} else {
while (aSize && (!IS_ALIGNED16(aInput) || !IS_ALIGNED16(aOutput))) {
*aOutput += *aInput*aScale;
++aOutput;
++aInput;
--aSize;
}
}
// we need to round aSize down to the nearest multiple of 16
uint32_t alignedSize = aSize & ~0x0F;
if (alignedSize > 0) {
AudioBufferAddWithScale_SSE(aInput, aScale, aOutput, alignedSize);
// adjust parameters for use with scalar operations below
aInput += alignedSize;
aOutput += alignedSize;
aSize -= alignedSize;
}
}
#endif
if (aScale == 1.0f) {
for (uint32_t i = 0; i < aSize; ++i) {
aOutput[i] += aInput[i];
}
} else {
for (uint32_t i = 0; i < aSize; ++i) {
aOutput[i] += aInput[i]*aScale;
}
}
}
/**
* Converts the data in aSegment to a single chunk aBlock. aSegment must have
* duration WEBAUDIO_BLOCK_SIZE. aFallbackChannelCount is a superset of the
* channels in every chunk of aSegment. aBlock must be float format or null.
*/
static void ConvertSegmentToAudioBlock(AudioSegment* aSegment,
AudioBlock* aBlock,
int32_t aFallbackChannelCount)
{
NS_ASSERTION(aSegment->GetDuration() == WEBAUDIO_BLOCK_SIZE, "Bad segment duration");
{
AudioSegment::ChunkIterator ci(*aSegment);
NS_ASSERTION(!ci.IsEnded(), "Should be at least one chunk!");
if (ci->GetDuration() == WEBAUDIO_BLOCK_SIZE &&
(ci->IsNull() || ci->mBufferFormat == AUDIO_FORMAT_FLOAT32)) {
bool aligned = true;
for (size_t i = 0; i < ci->mChannelData.Length(); ++i) {
if (!IS_ALIGNED16(ci->mChannelData[i])) {
aligned = false;
break;
}
}
// Return this chunk directly to avoid copying data.
if (aligned) {
*aBlock = *ci;
return;
}
}
}
aBlock->AllocateChannels(aFallbackChannelCount);
uint32_t duration = 0;
for (AudioSegment::ChunkIterator ci(*aSegment); !ci.IsEnded(); ci.Next()) {
switch (ci->mBufferFormat) {
case AUDIO_FORMAT_S16: {
CopyChunkToBlock<int16_t>(*ci, aBlock, duration);
break;
}
case AUDIO_FORMAT_FLOAT32: {
CopyChunkToBlock<float>(*ci, aBlock, duration);
break;
}
case AUDIO_FORMAT_SILENCE: {
// The actual type of the sample does not matter here, but we still need
// to send some audio to the graph.
CopyChunkToBlock<float>(*ci, aBlock, duration);
break;
}
}
duration += ci->GetDuration();
}
}
/**
* 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(AudioBlock* aOutput,
uint32_t aChannels,
uint32_t* aOffsetWithinBlock,
StreamTime* aCurrentPosition,
uint32_t aBufferMax)
{
MOZ_ASSERT(*aCurrentPosition < mStop);
uint32_t availableInOutput =
std::min<StreamTime>(WEBAUDIO_BLOCK_SIZE - *aOffsetWithinBlock,
mStop - *aCurrentPosition);
if (mResampler) {
CopyFromInputBufferWithResampling(aOutput, aChannels,
aOffsetWithinBlock, availableInOutput,
aCurrentPosition, aBufferMax);
return;
}
if (aChannels == 0) {
aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
// There is no attempt here to limit advance so that mBufferPosition is
// limited to aBufferMax. The only observable affect of skipping the
// check would be in the precise timing of the ended event if the loop
// attribute is reset after playback has looped.
*aOffsetWithinBlock += availableInOutput;
*aCurrentPosition += availableInOutput;
// Rounding at the start and end of the period means that fractional
// increments essentially accumulate if outRate remains constant. If
// outRate is varying, then accumulation happens on average but not
// precisely.
TrackTicks start = *aCurrentPosition *
mBufferSampleRate / mResamplerOutRate;
TrackTicks end = (*aCurrentPosition + availableInOutput) *
mBufferSampleRate / mResamplerOutRate;
mBufferPosition += end - start;
return;
}
uint32_t numFrames = std::min(aBufferMax - mBufferPosition,
availableInOutput);
bool shouldBorrow = false;
if (numFrames == WEBAUDIO_BLOCK_SIZE &&
mBuffer.mBufferFormat == AUDIO_FORMAT_FLOAT32) {
shouldBorrow = true;
for (uint32_t i = 0; i < aChannels; ++i) {
if (!IS_ALIGNED16(mBuffer.ChannelData<float>()[i] + mBufferPosition)) {
shouldBorrow = false;
break;
}
}
}
MOZ_ASSERT(mBufferPosition < aBufferMax);
if (shouldBorrow) {
BorrowFromInputBuffer(aOutput, aChannels);
} else {
if (*aOffsetWithinBlock == 0) {
aOutput->AllocateChannels(aChannels);
}
if (mBuffer.mBufferFormat == AUDIO_FORMAT_FLOAT32) {
CopyFromInputBuffer<float>(aOutput, aChannels,
*aOffsetWithinBlock, numFrames);
} else {
MOZ_ASSERT(mBuffer.mBufferFormat == AUDIO_FORMAT_S16);
CopyFromInputBuffer<int16_t>(aOutput, aChannels,
*aOffsetWithinBlock, numFrames);
}
}
*aOffsetWithinBlock += numFrames;
*aCurrentPosition += numFrames;
mBufferPosition += numFrames;
}
