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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 | |
| 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
| -rw-r--r-- | media/libmediaplayerservice/TestPlayerStub.cpp | 2 | ||||
| -rw-r--r-- | services/audioflinger/AudioFlinger.cpp | 2 | ||||
| -rw-r--r-- | services/audioflinger/AudioResampler.cpp | 8 | ||||
| -rw-r--r-- | services/audioflinger/AudioResamplerCubic.cpp | 2 |
4 files changed, 7 insertions, 7 deletions
diff --git a/media/libmediaplayerservice/TestPlayerStub.cpp b/media/libmediaplayerservice/TestPlayerStub.cpp index 0f0ff65..5d9728a 100644 --- a/media/libmediaplayerservice/TestPlayerStub.cpp +++ b/media/libmediaplayerservice/TestPlayerStub.cpp @@ -176,7 +176,7 @@ status_t TestPlayerStub::resetInternal() mContentUrl = NULL; if (mPlayer) { - LOG_ASSERT(mDeletePlayer != NULL, "mDeletePlayer is null"); + ALOG_ASSERT(mDeletePlayer != NULL, "mDeletePlayer is null"); (*mDeletePlayer)(mPlayer); mPlayer = NULL; } diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp index 72525cd..4ddefdb 100644 --- a/services/audioflinger/AudioFlinger.cpp +++ b/services/audioflinger/AudioFlinger.cpp @@ -2127,7 +2127,7 @@ uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track // the minimum track buffer size is normally twice the number of frames necessary // to fill one buffer and the resampler should not leave more than one buffer worth // of unreleased frames after each pass, but just in case... - LOG_ASSERT(minFrames <= cblk->frameCount); + ALOG_ASSERT(minFrames <= cblk->frameCount); } } if ((cblk->framesReady() >= minFrames) && track->isReady() && 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); } } 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) { |