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| author | Steve Block <steveblock@google.com> | 2012-01-09 18:35:44 +0000 |
|---|---|---|
| committer | Jean-Baptiste Queru <jbq@google.com> | 2012-01-19 14:45:08 -0800 |
| commit | f68633da3ced654a9344b5c3b82f4d1c79b6bd09 (patch) | |
| tree | a3ce92223f977611f6a1ab6710901a68fad8a968 /services/audioflinger | |
| parent | c6aacce37191e1cc79cfeba13b39899f59c68c3b (diff) | |
| download | frameworks_base-f68633da3ced654a9344b5c3b82f4d1c79b6bd09.zip frameworks_base-f68633da3ced654a9344b5c3b82f4d1c79b6bd09.tar.gz frameworks_base-f68633da3ced654a9344b5c3b82f4d1c79b6bd09.tar.bz2 | |
Rename LOG_ASSERT to ALOG_ASSERT
Change-Id: Ie2c7ea6560656d65bad791a61996174c75677517
Diffstat (limited to 'services/audioflinger')
| -rw-r--r-- | services/audioflinger/AudioResampler.cpp | 8 | ||||
| -rw-r--r-- | services/audioflinger/AudioResamplerCubic.cpp | 2 |
2 files changed, 5 insertions, 5 deletions
diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp index ee5bf66..4586b54 100644 --- a/services/audioflinger/AudioResampler.cpp +++ b/services/audioflinger/AudioResampler.cpp @@ -123,7 +123,7 @@ AudioResampler::AudioResampler(int bitDepth, int inChannelCount, if ((bitDepth != 16) ||(inChannelCount < 1) || (inChannelCount > 2)) { ALOGE("Unsupported sample format, %d bits, %d channels", bitDepth, inChannelCount); - // LOG_ASSERT(0); + // ALOG_ASSERT(0); } // initialize common members @@ -164,7 +164,7 @@ void AudioResamplerOrder1::resample(int32_t* out, size_t outFrameCount, AudioBufferProvider* provider) { // should never happen, but we overflow if it does - // LOG_ASSERT(outFrameCount < 32767); + // ALOG_ASSERT(outFrameCount < 32767); // select the appropriate resampler switch (mChannelCount) { @@ -261,7 +261,7 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount, provider->releaseBuffer(&mBuffer); // verify that the releaseBuffer resets the buffer frameCount - // LOG_ASSERT(mBuffer.frameCount == 0); + // ALOG_ASSERT(mBuffer.frameCount == 0); } } @@ -355,7 +355,7 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount, provider->releaseBuffer(&mBuffer); // verify that the releaseBuffer resets the buffer frameCount - // LOG_ASSERT(mBuffer.frameCount == 0); + // ALOG_ASSERT(mBuffer.frameCount == 0); } } diff --git a/services/audioflinger/AudioResamplerCubic.cpp b/services/audioflinger/AudioResamplerCubic.cpp index 587c7be..47205ba 100644 --- a/services/audioflinger/AudioResamplerCubic.cpp +++ b/services/audioflinger/AudioResamplerCubic.cpp @@ -36,7 +36,7 @@ void AudioResamplerCubic::resample(int32_t* out, size_t outFrameCount, AudioBufferProvider* provider) { // should never happen, but we overflow if it does - // LOG_ASSERT(outFrameCount < 32767); + // ALOG_ASSERT(outFrameCount < 32767); // select the appropriate resampler switch (mChannelCount) { |