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| author | Steve Block <steveblock@google.com> | 2012-01-09 18:35:44 +0000 |
|---|---|---|
| committer | Steve Block <steveblock@google.com> | 2012-01-09 21:36:22 +0000 |
| commit | c1dc1cb1d1eaf84e88669f1a5f22579a0d9237c2 (patch) | |
| tree | a23167188913d4f7ae0673d5efc3a232d245d048 /services/audioflinger/AudioResampler.cpp | |
| parent | 5f29ca38b71506ad7c7cb9925efbddf588e9655b (diff) | |
| download | frameworks_av-c1dc1cb1d1eaf84e88669f1a5f22579a0d9237c2.zip frameworks_av-c1dc1cb1d1eaf84e88669f1a5f22579a0d9237c2.tar.gz frameworks_av-c1dc1cb1d1eaf84e88669f1a5f22579a0d9237c2.tar.bz2 | |
Rename LOG_ASSERT to ALOG_ASSERT DO NOT MERGE
See https://android-git.corp.google.com/g/157519
Bug: 5449033
Change-Id: I8ceb2dba1b031a0fd68d15d146960d9ced62bbf3
Diffstat (limited to 'services/audioflinger/AudioResampler.cpp')
| -rw-r--r-- | services/audioflinger/AudioResampler.cpp | 8 |
1 files changed, 4 insertions, 4 deletions
diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp index fbdcb62..feacd96 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); } } |