Merge "Fix rounding error accumulation of audio timestamp"

am: cfa3f49644

Change-Id: I584c6b7b81f46ade75cb069ced5249b340b6f8de

media/libstagefright/AudioSource.cpp

@@ -427,19 +427,17 @@ status_t AudioSource::dataCallback(const AudioRecord::Buffer& audioBuffer) {
 void AudioSource::queueInputBuffer_l(MediaBuffer *buffer, int64_t timeUs) {
     const size_t bufferSize = buffer->range_length();
     const size_t frameSize = mRecord->frameSize();
-    const int64_t timestampUs =
-                mPrevSampleTimeUs +
-                    ((1000000LL * (bufferSize / frameSize)) +
-                        (mSampleRate >> 1)) / mSampleRate;
-
     if (mNumFramesReceived == 0) {
         buffer->meta_data().setInt64(kKeyAnchorTime, mStartTimeUs);
     }
-
+    mNumFramesReceived += bufferSize / frameSize;
+    const int64_t timestampUs =
+                mStartTimeUs +
+                    ((1000000LL * mNumFramesReceived) +
+                        (mSampleRate >> 1)) / mSampleRate;
     buffer->meta_data().setInt64(kKeyTime, mPrevSampleTimeUs);
     buffer->meta_data().setInt64(kKeyDriftTime, timeUs - mInitialReadTimeUs);
     mPrevSampleTimeUs = timestampUs;
-    mNumFramesReceived += bufferSize / frameSize;
     mBuffersReceived.push_back(buffer);
     mFrameAvailableCondition.signal();
 }