Remove aliasing

Code was aliasing mBuffer as buffer, but continuing to use both buffer
and mBuffer after that point.  This was at best misleading, and at worst
could confuse the compiler into generating bad code.  There was no
performance advantage to the alias, in fact removing it saves 16 bytes.

Change-Id: I55023ddba465d9be82f66745b088d18af658ac60

1 files changed, 11 insertions, 12 deletions

diff --git a/services/audioflinger/AudioResamplerSinc.cpp b/services/audioflinger/AudioResamplerSinc.cpp
index d012433..7a27b81 100644
--- a/services/audioflinger/AudioResamplerSinc.cpp
+++ b/services/audioflinger/AudioResamplerSinc.cpp
@@ -199,33 +199,32 @@ void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount,
     size_t outputSampleCount = outFrameCount * 2;
     size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
 
-    AudioBufferProvider::Buffer& buffer(mBuffer);
     while (outputIndex < outputSampleCount) {
         // buffer is empty, fetch a new one
-        while (buffer.frameCount == 0) {
-            buffer.frameCount = inFrameCount;
-            provider->getNextBuffer(&buffer);
-            if (buffer.raw == NULL) {
+        while (mBuffer.frameCount == 0) {
+            mBuffer.frameCount = inFrameCount;
+            provider->getNextBuffer(&mBuffer);
+            if (mBuffer.raw == NULL) {
                 goto resample_exit;
             }
             const uint32_t phaseIndex = phaseFraction >> kNumPhaseBits;
             if (phaseIndex == 1) {
                 // read one frame
-                read<CHANNELS>(impulse, phaseFraction, buffer.i16, inputIndex);
+                read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex);
             } else if (phaseIndex == 2) {
                 // read 2 frames
-                read<CHANNELS>(impulse, phaseFraction, buffer.i16, inputIndex);
+                read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex);
                 inputIndex++;
                 if (inputIndex >= mBuffer.frameCount) {
                     inputIndex -= mBuffer.frameCount;
-                    provider->releaseBuffer(&buffer);
+                    provider->releaseBuffer(&mBuffer);
                 } else {
-                    read<CHANNELS>(impulse, phaseFraction, buffer.i16, inputIndex);
+                    read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex);
                 }
            }
         }
-        int16_t *in = buffer.i16;
-        const size_t frameCount = buffer.frameCount;
+        int16_t *in = mBuffer.i16;
+        const size_t frameCount = mBuffer.frameCount;
 
         // Always read-in the first samples from the input buffer
         int16_t* head = impulse + halfNumCoefs*CHANNELS;
@@ -264,7 +263,7 @@ void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount,
         // if done with buffer, save samples
         if (inputIndex >= frameCount) {
             inputIndex -= frameCount;
-            provider->releaseBuffer(&buffer);
+            provider->releaseBuffer(&mBuffer);
         }
     }