diff options
| author | Andy Hung <hunga@google.com> | 2015-09-24 16:36:56 -0700 |
|---|---|---|
| committer | Andy Hung <hunga@google.com> | 2015-09-24 16:49:57 -0700 |
| commit | 8c987fa71326eb0cc504959a5ebb440410d73180 (patch) | |
| tree | 5d7b75a17bf736d15f95880a06b564f03cd03ee1 | |
| parent | a8f90d57f5b3ad4ef7194501aa20f0a0bd903e8f (diff) | |
| download | frameworks_av-8c987fa71326eb0cc504959a5ebb440410d73180.zip frameworks_av-8c987fa71326eb0cc504959a5ebb440410d73180.tar.gz frameworks_av-8c987fa71326eb0cc504959a5ebb440410d73180.tar.bz2 | |
DO NOT MERGE - AudioFlinger: Clear record buffers when starting RecordThread
Bug: 24211743
Bug: 24267152
Change-Id: I58c55e56b85067b71e4e300f947b4dfc159637ba
| -rw-r--r-- | services/audioflinger/FastCapture.cpp | 1 | ||||
| -rw-r--r-- | services/audioflinger/Threads.cpp | 4 |
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? |