commit c1dc1cb1d1eaf84e88669f1a5f22579a0d9237c2
author Steve Block <steveblock@google.com> Mon Jan 09 18:35:44 2012 +0000
committer Steve Block <steveblock@google.com> Mon Jan 09 21:36:22 2012 +0000
tree a23167188913d4f7ae0673d5efc3a232d245d048
parent 5f29ca38b71506ad7c7cb9925efbddf588e9655b

Rename LOG_ASSERT to ALOG_ASSERT  DO NOT MERGE

See https://android-git.corp.google.com/g/157519

Bug: 5449033
Change-Id: I8ceb2dba1b031a0fd68d15d146960d9ced62bbf3

media/libmediaplayerservice/TestPlayerStub.cpp

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 @@
     mContentUrl = NULL;
 
     if (mPlayer) {
-        LOG_ASSERT(mDeletePlayer != NULL, "mDeletePlayer is null");
+        ALOG_ASSERT(mDeletePlayer != NULL, "mDeletePlayer is null");
         (*mDeletePlayer)(mPlayer);
         mPlayer = NULL;
     }

services/audioflinger/AudioFlinger.cpp

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 @@
                 // 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() &&

services/audioflinger/AudioResampler.cpp

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 @@
     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 @@
         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 @@
             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 @@
             provider->releaseBuffer(&mBuffer);
 
             // verify that the releaseBuffer resets the buffer frameCount
-            // LOG_ASSERT(mBuffer.frameCount == 0);
+            // ALOG_ASSERT(mBuffer.frameCount == 0);
         }
     }
 

services/audioflinger/AudioResamplerCubic.cpp

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 @@
         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) {