commit 4ff14bae91075eb274eb1c2975982358946e7e63
author John Grossman <johngro@google.com> Wed Feb 08 16:37:41 2012 -0800
committer John Grossman <johngro@google.com> Thu Feb 16 13:45:11 2012 -0800
tree e9e54fddb9832d30b69c2a11c9ed2884397f2f95
parent 951bd8d1ad9581a414e171ad8605a9515d0ad667

Upintegrate Audio Flinger changes from ICS_AAH

Bring in changes to audio flinger made to support timed audio tracks
and HW master volume control.

Change-Id: Ide52d48809bdbed13acf35fd59b24637e35064ae
Signed-off-by: John Grossman <johngro@google.com>

services/audioflinger/AudioResampler.cpp

diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp
index 9486b9c..398ba0b 100644
--- a/services/audioflinger/AudioResampler.cpp
+++ b/services/audioflinger/AudioResampler.cpp
@@ -122,7 +122,8 @@
         int32_t sampleRate) :
     mBitDepth(bitDepth), mChannelCount(inChannelCount),
             mSampleRate(sampleRate), mInSampleRate(sampleRate), mInputIndex(0),
-            mPhaseFraction(0) {
+            mPhaseFraction(0), mLocalTimeFreq(0),
+            mPTS(AudioBufferProvider::kInvalidPTS) {
     // sanity check on format
     if ((bitDepth != 16) ||(inChannelCount < 1) || (inChannelCount > 2)) {
         ALOGE("Unsupported sample format, %d bits, %d channels", bitDepth,
@@ -150,6 +151,23 @@
     mVolume[1] = 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;
@@ -196,7 +214,8 @@
         // buffer is empty, fetch a new one
         while (mBuffer.frameCount == 0) {
             mBuffer.frameCount = inFrameCount;
-            provider->getNextBuffer(&mBuffer);
+            provider->getNextBuffer(&mBuffer,
+                                    calculateOutputPTS(outputIndex / 2));
             if (mBuffer.raw == NULL) {
                 goto resampleStereo16_exit;
             }
@@ -290,7 +309,8 @@
         // buffer is empty, fetch a new one
         while (mBuffer.frameCount == 0) {
             mBuffer.frameCount = inFrameCount;
-            provider->getNextBuffer(&mBuffer);
+            provider->getNextBuffer(&mBuffer,
+                                    calculateOutputPTS(outputIndex / 2));
             if (mBuffer.raw == NULL) {
                 mInputIndex = inputIndex;
                 mPhaseFraction = phaseFraction;