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author | John Grossman <johngro@google.com> | 2012-02-08 16:37:41 -0800 |
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committer | John Grossman <johngro@google.com> | 2012-02-16 13:45:11 -0800 |
commit | 4ff14bae91075eb274eb1c2975982358946e7e63 (patch) | |
tree | e9e54fddb9832d30b69c2a11c9ed2884397f2f95 /services/audioflinger/AudioResampler.cpp | |
parent | 951bd8d1ad9581a414e171ad8605a9515d0ad667 (diff) | |
download | frameworks_av-4ff14bae91075eb274eb1c2975982358946e7e63.zip frameworks_av-4ff14bae91075eb274eb1c2975982358946e7e63.tar.gz frameworks_av-4ff14bae91075eb274eb1c2975982358946e7e63.tar.bz2 |
Upintegrate Audio Flinger changes from ICS_AAH
Bring in changes to audio flinger made to support timed audio tracks
and HW master volume control.
Change-Id: Ide52d48809bdbed13acf35fd59b24637e35064ae
Signed-off-by: John Grossman <johngro@google.com>
Diffstat (limited to 'services/audioflinger/AudioResampler.cpp')
-rw-r--r-- | services/audioflinger/AudioResampler.cpp | 26 |
1 files changed, 23 insertions, 3 deletions
diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp index 9486b9c..398ba0b 100644 --- a/services/audioflinger/AudioResampler.cpp +++ b/services/audioflinger/AudioResampler.cpp @@ -122,7 +122,8 @@ AudioResampler::AudioResampler(int bitDepth, int inChannelCount, int32_t sampleRate) : mBitDepth(bitDepth), mChannelCount(inChannelCount), mSampleRate(sampleRate), mInSampleRate(sampleRate), mInputIndex(0), - mPhaseFraction(0) { + mPhaseFraction(0), mLocalTimeFreq(0), + mPTS(AudioBufferProvider::kInvalidPTS) { // sanity check on format if ((bitDepth != 16) ||(inChannelCount < 1) || (inChannelCount > 2)) { ALOGE("Unsupported sample format, %d bits, %d channels", bitDepth, @@ -150,6 +151,23 @@ void AudioResampler::setVolume(int16_t left, int16_t right) { mVolume[1] = right; } +void AudioResampler::setLocalTimeFreq(uint64_t freq) { + mLocalTimeFreq = freq; +} + +void AudioResampler::setPTS(int64_t pts) { + mPTS = pts; +} + +int64_t AudioResampler::calculateOutputPTS(int outputFrameIndex) { + + if (mPTS == AudioBufferProvider::kInvalidPTS) { + return AudioBufferProvider::kInvalidPTS; + } else { + return mPTS + ((outputFrameIndex * mLocalTimeFreq) / mSampleRate); + } +} + void AudioResampler::reset() { mInputIndex = 0; mPhaseFraction = 0; @@ -196,7 +214,8 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount, // buffer is empty, fetch a new one while (mBuffer.frameCount == 0) { mBuffer.frameCount = inFrameCount; - provider->getNextBuffer(&mBuffer); + provider->getNextBuffer(&mBuffer, + calculateOutputPTS(outputIndex / 2)); if (mBuffer.raw == NULL) { goto resampleStereo16_exit; } @@ -290,7 +309,8 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount, // buffer is empty, fetch a new one while (mBuffer.frameCount == 0) { mBuffer.frameCount = inFrameCount; - provider->getNextBuffer(&mBuffer); + provider->getNextBuffer(&mBuffer, + calculateOutputPTS(outputIndex / 2)); if (mBuffer.raw == NULL) { mInputIndex = inputIndex; mPhaseFraction = phaseFraction; |