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authorGlenn Kasten <gkasten@google.com>2015-10-14 20:44:40 +0000
committerAndroid Git Automerger <android-git-automerger@android.com>2015-10-14 20:44:40 +0000
commit1f42a6cc1edd96030478a8bc1e05a0ff7710e539 (patch)
treebd3201353b9cc3582c32f9960817123c546c976b
parent9d1bc9f252edae3a31decf5c18e971c6ab424b26 (diff)
parentc57ad1ed177ab58ae759a5054d8a14a306715f85 (diff)
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am c57ad1ed: am 55d1ea47: Merge "DO NOT MERGE - AudioFlinger: Clear record buffers when starting RecordThread" into lmp-dev
* commit 'c57ad1ed177ab58ae759a5054d8a14a306715f85': DO NOT MERGE - AudioFlinger: Clear record buffers when starting RecordThread
-rw-r--r--services/audioflinger/FastCapture.cpp1
-rw-r--r--services/audioflinger/Threads.cpp4
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 5282ffe..4c90be5 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -6239,7 +6239,9 @@ 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?