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| author | Steve Block <steveblock@google.com> | 2012-01-05 23:22:43 +0000 |
|---|---|---|
| committer | Steve Block <steveblock@google.com> | 2012-01-06 10:07:54 +0000 |
| commit | 5ff1dd576bb93c45b44088a51544a18fc43ebf58 (patch) | |
| tree | c002dcda87f08329197b01395539bd25f10ca737 /services/audioflinger/AudioResamplerCubic.cpp | |
| parent | 8a08dcc0a5de19a904e77d5f31bed3dff9a59890 (diff) | |
| download | frameworks_av-5ff1dd576bb93c45b44088a51544a18fc43ebf58.zip frameworks_av-5ff1dd576bb93c45b44088a51544a18fc43ebf58.tar.gz frameworks_av-5ff1dd576bb93c45b44088a51544a18fc43ebf58.tar.bz2 | |
Rename (IF_)LOGW(_IF) to (IF_)ALOGW(_IF) DO NOT MERGE
See https://android-git.corp.google.com/g/157065
Bug: 5449033
Change-Id: I00a4b904f9449e6f93b7fd35eac28640d7929e69
Diffstat (limited to 'services/audioflinger/AudioResamplerCubic.cpp')
| -rw-r--r-- | services/audioflinger/AudioResamplerCubic.cpp | 12 |
1 files changed, 6 insertions, 6 deletions
diff --git a/services/audioflinger/AudioResamplerCubic.cpp b/services/audioflinger/AudioResamplerCubic.cpp index 4d721f6..587c7be 100644 --- a/services/audioflinger/AudioResamplerCubic.cpp +++ b/services/audioflinger/AudioResamplerCubic.cpp @@ -68,7 +68,7 @@ void AudioResamplerCubic::resampleStereo16(int32_t* out, size_t outFrameCount, provider->getNextBuffer(&mBuffer); if (mBuffer.raw == NULL) return; - // LOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount); + // ALOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount); } int16_t *in = mBuffer.i16; @@ -99,7 +99,7 @@ void AudioResamplerCubic::resampleStereo16(int32_t* out, size_t outFrameCount, if (mBuffer.raw == NULL) goto save_state; // ugly, but efficient in = mBuffer.i16; - // LOGW("New buffer: offset=%p, frames=%d\n", mBuffer.raw, mBuffer.frameCount); + // ALOGW("New buffer: offset=%p, frames=%d\n", mBuffer.raw, mBuffer.frameCount); } // advance sample state @@ -109,7 +109,7 @@ void AudioResamplerCubic::resampleStereo16(int32_t* out, size_t outFrameCount, } save_state: - // LOGW("Done: index=%d, fraction=%u", inputIndex, phaseFraction); + // ALOGW("Done: index=%d, fraction=%u", inputIndex, phaseFraction); mInputIndex = inputIndex; mPhaseFraction = phaseFraction; } @@ -133,7 +133,7 @@ void AudioResamplerCubic::resampleMono16(int32_t* out, size_t outFrameCount, provider->getNextBuffer(&mBuffer); if (mBuffer.raw == NULL) return; - // LOGW("New buffer: offset=%p, frames=%d\n", mBuffer.raw, mBuffer.frameCount); + // ALOGW("New buffer: offset=%p, frames=%d\n", mBuffer.raw, mBuffer.frameCount); } int16_t *in = mBuffer.i16; @@ -163,7 +163,7 @@ void AudioResamplerCubic::resampleMono16(int32_t* out, size_t outFrameCount, provider->getNextBuffer(&mBuffer); if (mBuffer.raw == NULL) goto save_state; // ugly, but efficient - // LOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount); + // ALOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount); in = mBuffer.i16; } @@ -173,7 +173,7 @@ void AudioResamplerCubic::resampleMono16(int32_t* out, size_t outFrameCount, } save_state: - // LOGW("Done: index=%d, fraction=%u", inputIndex, phaseFraction); + // ALOGW("Done: index=%d, fraction=%u", inputIndex, phaseFraction); mInputIndex = inputIndex; mPhaseFraction = phaseFraction; } |