Merge "DO NOT MERGE - AudioFlinger: Clear record buffers when starting RecordThread" into lmp-dev

2 files changed, 4 insertions, 1 deletions

diff --git a/services/audioflinger/FastCapture.cpp b/services/audioflinger/FastCapture.cpp
index 0c9b976..9613e26 100644
--- a/services/audioflinger/FastCapture.cpp
+++ b/services/audioflinger/FastCapture.cpp
@@ -134,6 +134,7 @@ void FastCapture::onStateChange()
             unsigned channelCount = Format_channelCount(format);
             // FIXME frameSize
             readBuffer = new short[frameCount * channelCount];
+            memset(readBuffer, 0, frameCount * channelCount * sizeof(readBuffer[0]));
             periodNs = (frameCount * 1000000000LL) / sampleRate;    // 1.00
             underrunNs = (frameCount * 1750000000LL) / sampleRate;  // 1.75
             overrunNs = (frameCount * 500000000LL) / sampleRate;    // 0.50
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index b429cc2..63feeaa 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -6136,7 +6136,9 @@ void AudioFlinger::RecordThread::readInputParameters_l()
     // The current value is higher than necessary.  However it should not add to latency.
 
     // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer
-    mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount];
+    size_t bufferSizeInShorts = (mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount;
+    mRsmpInBuffer = new int16_t[bufferSizeInShorts];
+    memset(mRsmpInBuffer, 0, bufferSizeInShorts * sizeof(mRsmpInBuffer[0]));
 
     // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints.
     // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks?