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b=966636 rename resampling copy variables consistently and test remaining input better r=padenot

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--HG--
extra : rebase_source : 7d2c34865aeb21d67207e0d4e7b2fd3e8956d54c
This commit is contained in:
Karl Tomlinson 2014-02-05 14:42:18 +13:00
parent 75840f794d
commit 3ecf8b2128
1 changed files with 16 additions and 16 deletions
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32
content/media/webaudio/AudioBufferSourceNode.cpp
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@ -185,32 +185,33 @@ public:
void CopyFromInputBufferWithResampling(AudioNodeStream* aStream,
AudioChunk* aOutput,
uint32_t aChannels,
uintptr_t aSourceOffset,
uintptr_t aBufferOffset,
uint32_t aAvailableInInputBuffer,
uint32_t& aFramesWritten) {
uint32_t aOffsetWithinBlock,
uint32_t& aFramesWritten,
uint32_t aBufferOffset,
uint32_t aBufferMax) {
// TODO: adjust for mStop (see bug 913854 comment 9).
uint32_t availableInOutputBuffer = WEBAUDIO_BLOCK_SIZE - aBufferOffset;
uint32_t availableInOutputBuffer = WEBAUDIO_BLOCK_SIZE - aOffsetWithinBlock;
SpeexResamplerState* resampler = Resampler(aStream, aChannels);
MOZ_ASSERT(aChannels > 0);
if (aAvailableInInputBuffer) {
if (aBufferOffset < aBufferMax) {
uint32_t availableInInputBuffer = aBufferMax - aBufferOffset;
// Limit the number of input samples copied and possibly
// format-converted for resampling by estimating how many will be used.
// This may be a little small when filling the resampler with initial
// data, but we'll get called again and it will work out.
uint32_t num, den;
speex_resampler_get_ratio(resampler, &num, &den);
uint32_t inputLimit = std::min(aAvailableInInputBuffer,
uint32_t inputLimit = std::min(availableInInputBuffer,
availableInOutputBuffer * den / num + 10);
for (uint32_t i = 0; true; ) {
uint32_t inSamples = inputLimit;
const float* inputData = mBuffer->GetData(i) + aSourceOffset;
const float* inputData = mBuffer->GetData(i) + aBufferOffset;
uint32_t outSamples = availableInOutputBuffer;
float* outputData =
static_cast<float*>(const_cast<void*>(aOutput->mChannelData[i])) +
aBufferOffset;
aOffsetWithinBlock;
WebAudioUtils::SpeexResamplerProcess(resampler, i,
inputData, &inSamples,
@ -219,7 +220,7 @@ public:
mPosition += inSamples;
MOZ_ASSERT(mPosition <= mDuration || mLoop);
aFramesWritten = outSamples;
if (inSamples == aAvailableInInputBuffer && !mLoop) {
if (inSamples == availableInInputBuffer && !mLoop) {
// If the available output space were unbounded then the input
// latency would always be the correct amount of extra input to
// provide in order to advance the output position to align with
@ -242,7 +243,7 @@ public:
uint32_t outSamples = availableInOutputBuffer;
float* outputData =
static_cast<float*>(const_cast<void*>(aOutput->mChannelData[i])) +
aBufferOffset;
aOffsetWithinBlock;
// AudioDataValue* for aIn selects the function that does not try to
// copy and format-convert input data.
@ -312,6 +313,7 @@ public:
aBufferMax - aBufferOffset),
mStop - *aCurrentPosition);
if (numFrames == WEBAUDIO_BLOCK_SIZE && !ShouldResample(aStream->SampleRate())) {
MOZ_ASSERT(aBufferOffset < aBufferMax);
BorrowFromInputBuffer(aOutput, aChannels, aBufferOffset);
*aOffsetWithinBlock += numFrames;
*aCurrentPosition += numFrames;
@ -321,16 +323,14 @@ public:
AllocateAudioBlock(aChannels, aOutput);
}
if (!ShouldResample(aStream->SampleRate())) {
MOZ_ASSERT(aBufferOffset < aBufferMax);
CopyFromInputBuffer(aOutput, aChannels, aBufferOffset, *aOffsetWithinBlock, numFrames);
*aOffsetWithinBlock += numFrames;
*aCurrentPosition += numFrames;
mPosition += numFrames;
} else {
uint32_t framesWritten, availableInInputBuffer;
availableInInputBuffer = aBufferMax - aBufferOffset;
CopyFromInputBufferWithResampling(aStream, aOutput, aChannels, aBufferOffset, *aOffsetWithinBlock, availableInInputBuffer, framesWritten);
uint32_t framesWritten;
CopyFromInputBufferWithResampling(aStream, aOutput, aChannels, *aOffsetWithinBlock, framesWritten, aBufferOffset, aBufferMax);
*aOffsetWithinBlock += framesWritten;
*aCurrentPosition += framesWritten;
}