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| author | John Grossman <johngro@google.com> | 2012-02-08 16:37:41 -0800 |
|---|---|---|
| committer | John Grossman <johngro@google.com> | 2012-02-16 13:45:11 -0800 |
| commit | 4ff14bae91075eb274eb1c2975982358946e7e63 (patch) | |
| tree | e9e54fddb9832d30b69c2a11c9ed2884397f2f95 /services/audioflinger/AudioResamplerCubic.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/AudioResamplerCubic.cpp')
| -rw-r--r-- | services/audioflinger/AudioResamplerCubic.cpp | 11 |
1 files changed, 6 insertions, 5 deletions
diff --git a/services/audioflinger/AudioResamplerCubic.cpp b/services/audioflinger/AudioResamplerCubic.cpp index c0e760e..18e59e9 100644 --- a/services/audioflinger/AudioResamplerCubic.cpp +++ b/services/audioflinger/AudioResamplerCubic.cpp @@ -65,7 +65,7 @@ void AudioResamplerCubic::resampleStereo16(int32_t* out, size_t outFrameCount, // fetch first buffer if (mBuffer.frameCount == 0) { mBuffer.frameCount = inFrameCount; - provider->getNextBuffer(&mBuffer); + provider->getNextBuffer(&mBuffer, mPTS); if (mBuffer.raw == NULL) return; // ALOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount); @@ -95,7 +95,8 @@ void AudioResamplerCubic::resampleStereo16(int32_t* out, size_t outFrameCount, inputIndex = 0; provider->releaseBuffer(&mBuffer); mBuffer.frameCount = inFrameCount; - provider->getNextBuffer(&mBuffer); + provider->getNextBuffer(&mBuffer, + calculateOutputPTS(outputIndex / 2)); if (mBuffer.raw == NULL) goto save_state; // ugly, but efficient in = mBuffer.i16; @@ -130,7 +131,7 @@ void AudioResamplerCubic::resampleMono16(int32_t* out, size_t outFrameCount, // fetch first buffer if (mBuffer.frameCount == 0) { mBuffer.frameCount = inFrameCount; - provider->getNextBuffer(&mBuffer); + provider->getNextBuffer(&mBuffer, mPTS); if (mBuffer.raw == NULL) return; // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount); @@ -160,7 +161,8 @@ void AudioResamplerCubic::resampleMono16(int32_t* out, size_t outFrameCount, inputIndex = 0; provider->releaseBuffer(&mBuffer); mBuffer.frameCount = inFrameCount; - provider->getNextBuffer(&mBuffer); + provider->getNextBuffer(&mBuffer, + calculateOutputPTS(outputIndex / 2)); if (mBuffer.raw == NULL) goto save_state; // ugly, but efficient // ALOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount); @@ -181,4 +183,3 @@ save_state: // ---------------------------------------------------------------------------- } ; // namespace android - |