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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 | 39fd8eb956dea32692d3d72b10b64a8597df0a40 (patch) | |
| tree | 28e2a2d467c3f71d60114859641320f97c99c64a /services | |
| parent | 9a8ded7348c5b2302dd27b285b395416bc842c49 (diff) | |
| download | frameworks_av-39fd8eb956dea32692d3d72b10b64a8597df0a40.zip frameworks_av-39fd8eb956dea32692d3d72b10b64a8597df0a40.tar.gz frameworks_av-39fd8eb956dea32692d3d72b10b64a8597df0a40.tar.bz2 | |
Rename LOG_ASSERT to ALOG_ASSERT
Change-Id: Ie2c7ea6560656d65bad791a61996174c75677517
Diffstat (limited to 'services')
| -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) { |