Rename (IF_)LOGE(_IF) to (IF_)ALOGE(_IF) DO NOT MERGE

See https://android-git.corp.google.com/g/#/c/157220

Bug: 5449033
Change-Id: Ic9c19d30693bd56755f55906127cd6bd7126096c

1 files changed, 15 insertions, 15 deletions

diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp
index 7205045..fbdcb62 100644
--- a/services/audioflinger/AudioResampler.cpp
+++ b/services/audioflinger/AudioResampler.cpp
@@ -121,7 +121,7 @@ AudioResampler::AudioResampler(int bitDepth, int inChannelCount,
             mPhaseFraction(0) {
     // sanity check on format
     if ((bitDepth != 16) ||(inChannelCount < 1) || (inChannelCount > 2)) {
-        LOGE("Unsupported sample format, %d bits, %d channels", bitDepth,
+        ALOGE("Unsupported sample format, %d bits, %d channels", bitDepth,
                 inChannelCount);
         // LOG_ASSERT(0);
     }
@@ -190,7 +190,7 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
     size_t outputSampleCount = outFrameCount * 2;
     size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
 
-    // LOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d\n",
+    // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d\n",
     //      outFrameCount, inputIndex, phaseFraction, phaseIncrement);
 
     while (outputIndex < outputSampleCount) {
@@ -203,7 +203,7 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
                 goto resampleStereo16_exit;
             }
 
-            // LOGE("New buffer fetched: %d frames\n", mBuffer.frameCount);
+            // ALOGE("New buffer fetched: %d frames\n", mBuffer.frameCount);
             if (mBuffer.frameCount > inputIndex) break;
 
             inputIndex -= mBuffer.frameCount;
@@ -217,7 +217,7 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
 
         // handle boundary case
         while (inputIndex == 0) {
-            // LOGE("boundary case\n");
+            // ALOGE("boundary case\n");
             out[outputIndex++] += vl * Interp(mX0L, in[0], phaseFraction);
             out[outputIndex++] += vr * Interp(mX0R, in[1], phaseFraction);
             Advance(&inputIndex, &phaseFraction, phaseIncrement);
@@ -226,7 +226,7 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
         }
 
         // process input samples
-        // LOGE("general case\n");
+        // ALOGE("general case\n");
 
 #ifdef ASM_ARM_RESAMP1  // asm optimisation for ResamplerOrder1
         if (inputIndex + 2 < mBuffer.frameCount) {
@@ -248,13 +248,13 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
             Advance(&inputIndex, &phaseFraction, phaseIncrement);
         }
 
-        // LOGE("loop done - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
+        // ALOGE("loop done - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
 
         // if done with buffer, save samples
         if (inputIndex >= mBuffer.frameCount) {
             inputIndex -= mBuffer.frameCount;
 
-            // LOGE("buffer done, new input index %d", inputIndex);
+            // ALOGE("buffer done, new input index %d", inputIndex);
 
             mX0L = mBuffer.i16[mBuffer.frameCount*2-2];
             mX0R = mBuffer.i16[mBuffer.frameCount*2-1];
@@ -265,7 +265,7 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
         }
     }
 
-    // LOGE("output buffer full - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
+    // ALOGE("output buffer full - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
 
 resampleStereo16_exit:
     // save state
@@ -286,7 +286,7 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
     size_t outputSampleCount = outFrameCount * 2;
     size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
 
-    // LOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d\n",
+    // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d\n",
     //      outFrameCount, inputIndex, phaseFraction, phaseIncrement);
     while (outputIndex < outputSampleCount) {
         // buffer is empty, fetch a new one
@@ -298,7 +298,7 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
                 mPhaseFraction = phaseFraction;
                 goto resampleMono16_exit;
             }
-            // LOGE("New buffer fetched: %d frames\n", mBuffer.frameCount);
+            // ALOGE("New buffer fetched: %d frames\n", mBuffer.frameCount);
             if (mBuffer.frameCount >  inputIndex) break;
 
             inputIndex -= mBuffer.frameCount;
@@ -310,7 +310,7 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
 
         // handle boundary case
         while (inputIndex == 0) {
-            // LOGE("boundary case\n");
+            // ALOGE("boundary case\n");
             int32_t sample = Interp(mX0L, in[0], phaseFraction);
             out[outputIndex++] += vl * sample;
             out[outputIndex++] += vr * sample;
@@ -320,7 +320,7 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
         }
 
         // process input samples
-        // LOGE("general case\n");
+        // ALOGE("general case\n");
 
 #ifdef ASM_ARM_RESAMP1  // asm optimisation for ResamplerOrder1
         if (inputIndex + 2 < mBuffer.frameCount) {
@@ -343,13 +343,13 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
         }
 
 
-        // LOGE("loop done - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
+        // ALOGE("loop done - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
 
         // if done with buffer, save samples
         if (inputIndex >= mBuffer.frameCount) {
             inputIndex -= mBuffer.frameCount;
 
-            // LOGE("buffer done, new input index %d", inputIndex);
+            // ALOGE("buffer done, new input index %d", inputIndex);
 
             mX0L = mBuffer.i16[mBuffer.frameCount-1];
             provider->releaseBuffer(&mBuffer);
@@ -359,7 +359,7 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
         }
     }
 
-    // LOGE("output buffer full - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
+    // ALOGE("output buffer full - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
 
 resampleMono16_exit:
     // save state