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-rw-r--r--media/libmediaplayerservice/TestPlayerStub.cpp2
-rw-r--r--services/audioflinger/AudioFlinger.cpp2
-rw-r--r--services/audioflinger/AudioResampler.cpp8
-rw-r--r--services/audioflinger/AudioResamplerCubic.cpp2
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) {