commit 97ae82408469286426d8dc57a361350f74e24a8b
author Andy Hung <hunga@google.com> Fri May 30 10:35:47 2014 -0700
committer Andy Hung <hunga@google.com> Fri May 30 16:44:32 2014 -0700
tree 09d2ac62b57d06d0d100ce5bc93fb8a05a27398d
parent 5866a3b9689035580c31f1a02037b8176d7508a7

Rename UNITY_GAIN to UNITY_GAIN_INT in AudioMixer

Change-Id: Ic040311305026f0b4c4280a5b3bef7a447ac1da3

services/audioflinger/AudioMixer.cpp

diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 407694b..2a8ece6 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -265,8 +265,8 @@
         // assume default parameters for the track, except where noted below
         track_t* t = &mState.tracks[n];
         t->needs = 0;
-        t->volume[0] = UNITY_GAIN;
-        t->volume[1] = UNITY_GAIN;
+        t->volume[0] = UNITY_GAIN_INT;
+        t->volume[1] = UNITY_GAIN_INT;
         // no initialization needed
         // t->prevVolume[0]
         // t->prevVolume[1]
@@ -930,7 +930,7 @@
         // always resample with unity gain when sending to auxiliary buffer to be able
         // to apply send level after resampling
         // TODO: modify each resampler to support aux channel?
-        t->resampler->setVolume(UNITY_GAIN, UNITY_GAIN);
+        t->resampler->setVolume(UNITY_GAIN_INT, UNITY_GAIN_INT);
         memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
         t->resampler->resample(temp, outFrameCount, t->bufferProvider);
         if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) {
@@ -940,7 +940,7 @@
         }
     } else {
         if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) {
-            t->resampler->setVolume(UNITY_GAIN, UNITY_GAIN);
+            t->resampler->setVolume(UNITY_GAIN_INT, UNITY_GAIN_INT);
             memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
             t->resampler->resample(temp, outFrameCount, t->bufferProvider);
             volumeRampStereo(t, out, outFrameCount, temp, aux);
@@ -1500,7 +1500,7 @@
             } while (--outFrames);
             break;
         case AUDIO_FORMAT_PCM_16_BIT:
-            if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN || uint32_t(vr) > UNITY_GAIN)) {
+            if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN_INT || uint32_t(vr) > UNITY_GAIN_INT)) {
                 // volume is boosted, so we might need to clamp even though
                 // we process only one track.
                 do {

services/audioflinger/AudioMixer.h

diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index 573ba96..43fe664 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -31,7 +31,7 @@
 #include <media/nbaio/NBLog.h>
 
 // FIXME This is actually unity gain, which might not be max in future, expressed in U.12
-#define MAX_GAIN_INT AudioMixer::UNITY_GAIN
+#define MAX_GAIN_INT AudioMixer::UNITY_GAIN_INT
 
 namespace android {
 
@@ -58,7 +58,7 @@
     // maximum number of channels supported for the content
     static const uint32_t MAX_NUM_CHANNELS_TO_DOWNMIX = 8;
 
-    static const uint16_t UNITY_GAIN = 0x1000;
+    static const uint16_t UNITY_GAIN_INT = 0x1000;
 
     enum { // names
 

services/audioflinger/Tracks.cpp

diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 7ddc71c..64ae0b0 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -1873,7 +1873,7 @@
         mResampler = AudioResampler::create(16, thread->mChannelCount, sampleRate);
         // source SR
         mResampler->setSampleRate(thread->mSampleRate);
-        mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
+        mResampler->setVolume(AudioMixer::UNITY_GAIN_INT, AudioMixer::UNITY_GAIN_INT);
         mResamplerBufferProvider = new ResamplerBufferProvider(this);
     }
 }