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b=967972 avoid overflow in buffer position when looping r=padenot

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As a side-effect, the buffer playback will now continue from the same buffer
sample number (even when looping), where possible, in the following situations:

* Changing the loop start or end parameters.

* Changing the buffer to one of a different sample rate.

--HG--
extra : transplant_source : 2%11%7Dm%92pu%60%E3G%D4%D4%DB0%2B%20%9A%1Bd%FA
This commit is contained in:
Karl Tomlinson 2014-02-11 11:19:26 +13:00
parent 4dee321de8
commit e64c4d709d
1 changed files with 25 additions and 28 deletions
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53
content/media/webaudio/AudioBufferSourceNode.cpp
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@ -153,14 +153,13 @@ public:
// Borrow a full buffer of size WEBAUDIO_BLOCK_SIZE from the source buffer
// at offset aSourceOffset. This avoids copying memory.
void BorrowFromInputBuffer(AudioChunk* aOutput,
uint32_t aChannels,
uintptr_t aSourceOffset)
uint32_t aChannels)
{
aOutput->mDuration = WEBAUDIO_BLOCK_SIZE;
aOutput->mBuffer = mBuffer;
aOutput->mChannelData.SetLength(aChannels);
for (uint32_t i = 0; i < aChannels; ++i) {
aOutput->mChannelData[i] = mBuffer->GetData(i) + aSourceOffset;
aOutput->mChannelData[i] = mBuffer->GetData(i) + mBufferPosition;
}
aOutput->mVolume = 1.0f;
aOutput->mBufferFormat = AUDIO_FORMAT_FLOAT32;
@ -170,13 +169,12 @@ public:
// and put it at offset aBufferOffset in the destination buffer.
void CopyFromInputBuffer(AudioChunk* aOutput,
uint32_t aChannels,
uintptr_t aSourceOffset,
uintptr_t aBufferOffset,
uintptr_t aOffsetWithinBlock,
uint32_t aNumberOfFrames) {
for (uint32_t i = 0; i < aChannels; ++i) {
float* baseChannelData = static_cast<float*>(const_cast<void*>(aOutput->mChannelData[i]));
memcpy(baseChannelData + aBufferOffset,
mBuffer->GetData(i) + aSourceOffset,
memcpy(baseChannelData + aOffsetWithinBlock,
mBuffer->GetData(i) + mBufferPosition,
aNumberOfFrames * sizeof(float));
}
}
@ -191,15 +189,14 @@ public:
uint32_t aChannels,
uint32_t aOffsetWithinBlock,
uint32_t& aFramesWritten,
uint32_t aBufferOffset,
uint32_t aBufferMax) {
int32_t aBufferMax) {
// TODO: adjust for mStop (see bug 913854 comment 9).
uint32_t availableInOutputBuffer = WEBAUDIO_BLOCK_SIZE - aOffsetWithinBlock;
SpeexResamplerState* resampler = Resampler(aStream, aChannels);
MOZ_ASSERT(aChannels > 0);
if (aBufferOffset < aBufferMax) {
uint32_t availableInInputBuffer = aBufferMax - aBufferOffset;
if (mBufferPosition < aBufferMax) {
uint32_t availableInInputBuffer = aBufferMax - mBufferPosition;
// 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
@ -210,7 +207,7 @@ public:
availableInOutputBuffer * num / den + 10);
for (uint32_t i = 0; true; ) {
uint32_t inSamples = inputLimit;
const float* inputData = mBuffer->GetData(i) + aBufferOffset;
const float* inputData = mBuffer->GetData(i) + mBufferPosition;
uint32_t outSamples = availableInOutputBuffer;
float* outputData =
@ -308,17 +305,16 @@ public:
uint32_t aChannels,
uint32_t* aOffsetWithinBlock,
TrackTicks* aCurrentPosition,
uint32_t aBufferOffset,
uint32_t aBufferMax)
int32_t aBufferMax)
{
MOZ_ASSERT(*aCurrentPosition < mStop);
uint32_t numFrames =
std::min<TrackTicks>(std::min(WEBAUDIO_BLOCK_SIZE - *aOffsetWithinBlock,
aBufferMax - aBufferOffset),
mStop - *aCurrentPosition);
std::min(std::min<TrackTicks>(WEBAUDIO_BLOCK_SIZE - *aOffsetWithinBlock,
aBufferMax - mBufferPosition),
mStop - *aCurrentPosition);
if (numFrames == WEBAUDIO_BLOCK_SIZE && !ShouldResample(aStream->SampleRate())) {
MOZ_ASSERT(aBufferOffset < aBufferMax);
BorrowFromInputBuffer(aOutput, aChannels, aBufferOffset);
MOZ_ASSERT(mBufferPosition < aBufferMax);
BorrowFromInputBuffer(aOutput, aChannels);
*aOffsetWithinBlock += numFrames;
*aCurrentPosition += numFrames;
mBufferPosition += numFrames;
@ -327,14 +323,14 @@ public:
AllocateAudioBlock(aChannels, aOutput);
}
if (!ShouldResample(aStream->SampleRate())) {
MOZ_ASSERT(aBufferOffset < aBufferMax);
CopyFromInputBuffer(aOutput, aChannels, aBufferOffset, *aOffsetWithinBlock, numFrames);
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, aBufferOffset, aBufferMax);
CopyFromInputBufferWithResampling(aStream, aOutput, aChannels, *aOffsetWithinBlock, framesWritten, aBufferMax);
*aOffsetWithinBlock += framesWritten;
*aCurrentPosition += framesWritten;
}
@ -424,15 +420,16 @@ public:
continue;
}
if (mLoop) {
if (mBufferPosition < mLoopEnd) {
CopyFromBuffer(aStream, aOutput, channels, &written, &streamPosition, mBufferPosition, mLoopEnd);
} else {
uint32_t offsetInLoop = (mBufferPosition - mLoopEnd) % (mLoopEnd - mLoopStart);
CopyFromBuffer(aStream, aOutput, channels, &written, &streamPosition, mLoopStart + offsetInLoop, mLoopEnd);
// mLoopEnd can become less than mBufferPosition when a LOOPEND engine
// parameter is received after "loopend" is changed on the node or a
// new buffer with lower samplerate is set.
if (mBufferPosition >= mLoopEnd) {
mBufferPosition = mLoopStart;
}
CopyFromBuffer(aStream, aOutput, channels, &written, &streamPosition, mLoopEnd);
} else {
if (mBufferPosition < mBufferEnd || mRemainingResamplerTail) {
CopyFromBuffer(aStream, aOutput, channels, &written, &streamPosition, mBufferPosition, mBufferEnd);
CopyFromBuffer(aStream, aOutput, channels, &written, &streamPosition, mBufferEnd);
} else {
FillWithZeroes(aOutput, channels, &written, &streamPosition, TRACK_TICKS_MAX);
}