Android/frameworks_base
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Merge "Remove aliasing"

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This commit is contained in:
Glenn Kasten
2012-02-10 15:29:35 -08:00
committed by Android (Google) Code Review
parent 5bb3090a18 d8b2e2b9e3
commit 2a67e1a6c0
1 changed files with 11 additions and 12 deletions
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23
services/audioflinger/AudioResamplerSinc.cpp
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@@ -199,33 +199,32 @@ void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount,
size_t outputSampleCount = outFrameCount * 2; size_t outputSampleCount = outFrameCount * 2;
size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate; size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
AudioBufferProvider::Buffer& buffer(mBuffer);
while (outputIndex < outputSampleCount) { while (outputIndex < outputSampleCount) {
// buffer is empty, fetch a new one // buffer is empty, fetch a new one
while (buffer.frameCount == 0) { while (mBuffer.frameCount == 0) {
buffer.frameCount = inFrameCount; mBuffer.frameCount = inFrameCount;
provider->getNextBuffer(&buffer); provider->getNextBuffer(&mBuffer);
if (buffer.raw == NULL) { if (mBuffer.raw == NULL) {
goto resample_exit; goto resample_exit;
} }
const uint32_t phaseIndex = phaseFraction >> kNumPhaseBits; const uint32_t phaseIndex = phaseFraction >> kNumPhaseBits;
if (phaseIndex == 1) { if (phaseIndex == 1) {
// read one frame // read one frame
read<CHANNELS>(impulse, phaseFraction, buffer.i16, inputIndex); read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex);
} else if (phaseIndex == 2) { } else if (phaseIndex == 2) {
// read 2 frames // read 2 frames
read<CHANNELS>(impulse, phaseFraction, buffer.i16, inputIndex); read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex);
inputIndex++; inputIndex++;
if (inputIndex >= mBuffer.frameCount) { if (inputIndex >= mBuffer.frameCount) {
inputIndex -= mBuffer.frameCount; inputIndex -= mBuffer.frameCount;
provider->releaseBuffer(&buffer); provider->releaseBuffer(&mBuffer);
} else { } else {
read<CHANNELS>(impulse, phaseFraction, buffer.i16, inputIndex); read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex);
} }
} }
} }
int16_t *in = buffer.i16; int16_t *in = mBuffer.i16;
const size_t frameCount = buffer.frameCount; const size_t frameCount = mBuffer.frameCount;
// Always read-in the first samples from the input buffer // Always read-in the first samples from the input buffer
int16_t* head = impulse + halfNumCoefs*CHANNELS; int16_t* head = impulse + halfNumCoefs*CHANNELS;
@@ -264,7 +263,7 @@ void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount,
// if done with buffer, save samples // if done with buffer, save samples
if (inputIndex >= frameCount) { if (inputIndex >= frameCount) {
inputIndex -= frameCount; inputIndex -= frameCount;
provider->releaseBuffer(&buffer); provider->releaseBuffer(&mBuffer);
} }
} }