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b=967972 include offset in buffer position r=padenot

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There is a change in behaviour re the offset parameter when changing the
buffer while playing to one with a different sample rate, but I'm not clear
that what we current do with mPosition in that situation is best anyway.  I
think the main thing is to ensure playback from the same place if the buffer
is changed to one with the same sample rate.

--HG--
extra : transplant_source : %17%40%15%12L%9E%05et%C2o%BE%15%D7%C7%F4%ED%5E%24q
This commit is contained in:
Karl Tomlinson 2014-02-06 08:28:42 +13:00
parent a2c021ffa5
commit 8dd5bcc7a7
1 changed files with 24 additions and 21 deletions
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45
content/media/webaudio/AudioBufferSourceNode.cpp
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@ -58,9 +58,9 @@ public:
AudioNodeEngine(aNode),
mStart(0), mStop(TRACK_TICKS_MAX),
mResampler(nullptr), mRemainingResamplerTail(0),
mOffset(0), mBufferEnd(0),
mBufferEnd(0),
mLoopStart(0), mLoopEnd(0),
mBufferSampleRate(0), mPosition(0), mChannels(0), mPlaybackRate(1.0f),
mBufferSampleRate(0), mBufferPosition(0), mChannels(0), mPlaybackRate(1.0f),
mDopplerShift(1.0f),
mDestination(static_cast<AudioNodeStream*>(aDestination->Stream())),
mPlaybackRateTimeline(1.0f), mLoop(false)
@ -114,7 +114,11 @@ public:
{
switch (aIndex) {
case AudioBufferSourceNode::SAMPLE_RATE: mBufferSampleRate = aParam; break;
case AudioBufferSourceNode::BUFFERSTART: mOffset = aParam; break;
case AudioBufferSourceNode::BUFFERSTART:
if (mBufferPosition == 0) {
mBufferPosition = aParam;
}
break;
case AudioBufferSourceNode::BUFFEREND: mBufferEnd = aParam; break;
case AudioBufferSourceNode::LOOP: mLoop = !!aParam; break;
case AudioBufferSourceNode::LOOPSTART: mLoopStart = aParam; break;
@ -217,8 +221,8 @@ public:
inputData, &inSamples,
outputData, &outSamples);
if (++i == aChannels) {
mPosition += inSamples;
MOZ_ASSERT(mOffset + mPosition <= mBufferEnd || mLoop);
mBufferPosition += inSamples;
MOZ_ASSERT(mBufferPosition <= mBufferEnd || mLoop);
aFramesWritten = outSamples;
if (inSamples == availableInInputBuffer && !mLoop) {
// If the available output space were unbounded then the input
@ -317,7 +321,7 @@ public:
BorrowFromInputBuffer(aOutput, aChannels, aBufferOffset);
*aOffsetWithinBlock += numFrames;
*aCurrentPosition += numFrames;
mPosition += numFrames;
mBufferPosition += numFrames;
} else {
if (*aOffsetWithinBlock == 0) {
AllocateAudioBlock(aChannels, aOutput);
@ -327,7 +331,7 @@ public:
CopyFromInputBuffer(aOutput, aChannels, aBufferOffset, *aOffsetWithinBlock, numFrames);
*aOffsetWithinBlock += numFrames;
*aCurrentPosition += numFrames;
mPosition += numFrames;
mBufferPosition += numFrames;
} else {
uint32_t framesWritten;
CopyFromInputBufferWithResampling(aStream, aOutput, aChannels, *aOffsetWithinBlock, framesWritten, aBufferOffset, aBufferMax);
@ -352,9 +356,10 @@ public:
bool ShouldResample(TrackRate aStreamSampleRate) const
{
// There is latency in the resampler. If there is already a resampler,
// then it will have moved mPosition to after the samples it has read, but
// it hasn't output its buffered samples. Keep using the resampler, even
// if the rates now match, so that this latency segment is output.
// then it will have moved mBufferPosition to after the samples it has
// read, but it hasn't output its buffered samples. Keep using the
// resampler, even if the rates now match, so that this latent segment is
// output.
return mResampler ||
(mPlaybackRate * mDopplerShift * mBufferSampleRate != aStreamSampleRate);
}
@ -418,17 +423,16 @@ public:
FillWithZeroes(aOutput, channels, &written, &streamPosition, mStart);
continue;
}
TrackTicks t = mPosition;
if (mLoop) {
if (mOffset + t < mLoopEnd) {
CopyFromBuffer(aStream, aOutput, channels, &written, &streamPosition, mOffset + t, mLoopEnd);
if (mBufferPosition < mLoopEnd) {
CopyFromBuffer(aStream, aOutput, channels, &written, &streamPosition, mBufferPosition, mLoopEnd);
} else {
uint32_t offsetInLoop = (mOffset + t - mLoopEnd) % (mLoopEnd - mLoopStart);
uint32_t offsetInLoop = (mBufferPosition - mLoopEnd) % (mLoopEnd - mLoopStart);
CopyFromBuffer(aStream, aOutput, channels, &written, &streamPosition, mLoopStart + offsetInLoop, mLoopEnd);
}
} else {
if (mOffset + t < mBufferEnd || mRemainingResamplerTail) {
CopyFromBuffer(aStream, aOutput, channels, &written, &streamPosition, mOffset + t, mBufferEnd);
if (mBufferPosition < mBufferEnd || mRemainingResamplerTail) {
CopyFromBuffer(aStream, aOutput, channels, &written, &streamPosition, mBufferPosition, mBufferEnd);
} else {
FillWithZeroes(aOutput, channels, &written, &streamPosition, TRACK_TICKS_MAX);
}
@ -438,7 +442,7 @@ public:
// We've finished if we've gone past mStop, or if we're past mDuration when
// looping is disabled.
if (streamPosition >= mStop ||
(!mLoop && mOffset + mPosition >= mBufferEnd && !mRemainingResamplerTail)) {
(!mLoop && mBufferPosition >= mBufferEnd && !mRemainingResamplerTail)) {
*aFinished = true;
}
}
@ -447,15 +451,14 @@ public:
TrackTicks mStop;
nsRefPtr<ThreadSharedFloatArrayBufferList> mBuffer;
SpeexResamplerState* mResampler;
// mRemainingResamplerTail, like mPosition, mOffset, and mBufferEnd, is
// measured in input buffer samples.
// mRemainingResamplerTail, like mBufferPosition, and
// mBufferEnd, is measured in input buffer samples.
int mRemainingResamplerTail;
int32_t mOffset;
int32_t mBufferEnd;
int32_t mLoopStart;
int32_t mLoopEnd;
int32_t mBufferSampleRate;
int32_t mPosition;
int32_t mBufferPosition;
uint32_t mChannels;
float mPlaybackRate;
float mDopplerShift;