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Rename LOG_ASSERT to ALOG_ASSERT
authorSteve Block <steveblock@google.com>
Mon, 9 Jan 2012 18:35:44 +0000 (18:35 +0000)
committerJean-Baptiste Queru <jbq@google.com>
Thu, 19 Jan 2012 22:45:08 +0000 (14:45 -0800)
Change-Id: Ie2c7ea6560656d65bad791a61996174c75677517

media/libmediaplayerservice/TestPlayerStub.cpp
services/audioflinger/AudioResampler.cpp
services/audioflinger/AudioResamplerCubic.cpp

index 0f0ff65..5d9728a 100644 (file)
@@ -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;
     }
index ee5bf66..4586b54 100644 (file)
@@ -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);
         }
     }
 
index 587c7be..47205ba 100644 (file)
@@ -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) {