commit d79072e9dff59f767cce2cda1caab80ce5a0815b
author Glenn Kasten <gkasten@google.com> Wed Jan 06 08:41:20 2016 -0800
committer Glenn Kasten <gkasten@google.com> Mon Jan 25 15:34:04 2016 -0800
tree cab9cab36ddf4bb61675c4508d923b283fd3edd6
parent a7b723715bab4ddf6db364c825135a9bc003c7c0

Remove TimedAudioTrack and associated code

Bug: 8278435
Change-Id: I095c1a4888e645e14d93b0b15fbef4524a831ca1

services/audioflinger/AudioResampler.cpp

diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp
index e49b7b1..7c20478 100644
--- a/services/audioflinger/AudioResampler.cpp
+++ b/services/audioflinger/AudioResampler.cpp
@@ -261,8 +261,8 @@
         int32_t sampleRate, src_quality quality) :
         mChannelCount(inChannelCount),
         mSampleRate(sampleRate), mInSampleRate(sampleRate), mInputIndex(0),
-        mPhaseFraction(0), mLocalTimeFreq(0),
-        mPTS(AudioBufferProvider::kInvalidPTS), mQuality(quality) {
+        mPhaseFraction(0),
+        mQuality(quality) {
 
     const int maxChannels = quality < DYN_LOW_QUALITY ? 2 : 8;
     if (inChannelCount < 1
@@ -304,23 +304,6 @@
     mVolume[1] = u4_12_from_float(clampFloatVol(right));
 }
 
-void AudioResampler::setLocalTimeFreq(uint64_t freq) {
-    mLocalTimeFreq = freq;
-}
-
-void AudioResampler::setPTS(int64_t pts) {
-    mPTS = pts;
-}
-
-int64_t AudioResampler::calculateOutputPTS(int outputFrameIndex) {
-
-    if (mPTS == AudioBufferProvider::kInvalidPTS) {
-        return AudioBufferProvider::kInvalidPTS;
-    } else {
-        return mPTS + ((outputFrameIndex * mLocalTimeFreq) / mSampleRate);
-    }
-}
-
 void AudioResampler::reset() {
     mInputIndex = 0;
     mPhaseFraction = 0;
@@ -368,8 +351,7 @@
         // buffer is empty, fetch a new one
         while (mBuffer.frameCount == 0) {
             mBuffer.frameCount = inFrameCount;
-            provider->getNextBuffer(&mBuffer,
-                                    calculateOutputPTS(outputIndex / 2));
+            provider->getNextBuffer(&mBuffer);
             if (mBuffer.raw == NULL) {
                 goto resampleStereo16_exit;
             }
@@ -465,8 +447,7 @@
         // buffer is empty, fetch a new one
         while (mBuffer.frameCount == 0) {
             mBuffer.frameCount = inFrameCount;
-            provider->getNextBuffer(&mBuffer,
-                                    calculateOutputPTS(outputIndex / 2));
+            provider->getNextBuffer(&mBuffer);
             if (mBuffer.raw == NULL) {
                 mInputIndex = inputIndex;
                 mPhaseFraction = phaseFraction;