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authorAndy Hung <hunga@google.com>2015-09-21 12:44:54 -0700
committerGlenn Kasten <gkasten@google.com>2015-10-13 12:17:53 -0700
commit4c6e77ff8e18a1551320a6b42f6a45e19dcce748 (patch)
tree97d86a674d1cecec6da60dfb0ec0c2eae1e2b644 /services
parent5076a51e0572246f7062e48761766c01f17233c2 (diff)
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AudioFlinger: Clear record buffers when starting RecordThread
Bug: 24211743 Bug: 24267152 Change-Id: I58c55e56b85067b71e4e300f947b4dfc159637ba
Diffstat (limited to 'services')
-rw-r--r--services/audioflinger/FastCapture.cpp4
-rw-r--r--services/audioflinger/Threads.cpp5
2 files changed, 7 insertions, 2 deletions
diff --git a/services/audioflinger/FastCapture.cpp b/services/audioflinger/FastCapture.cpp
index 79ac12b..1bba5f6 100644
--- a/services/audioflinger/FastCapture.cpp
+++ b/services/audioflinger/FastCapture.cpp
@@ -131,7 +131,9 @@ void FastCapture::onStateChange()
// FIXME new may block for unbounded time at internal mutex of the heap
// implementation; it would be better to have normal capture thread allocate for
// us to avoid blocking here and to prevent possible priority inversion
- (void)posix_memalign(&mReadBuffer, 32, frameCount * Format_frameSize(mFormat));
+ size_t bufferSize = frameCount * Format_frameSize(mFormat);
+ (void)posix_memalign(&mReadBuffer, 32, bufferSize);
+ memset(mReadBuffer, 0, bufferSize); // if posix_memalign fails, will segv here.
mPeriodNs = (frameCount * 1000000000LL) / mSampleRate; // 1.00
mUnderrunNs = (frameCount * 1750000000LL) / mSampleRate; // 1.75
mOverrunNs = (frameCount * 500000000LL) / mSampleRate; // 0.50
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 0a7d4a2..246f6ba 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -6926,6 +6926,7 @@ void AudioFlinger::RecordThread::readInputParameters_l()
mRsmpInFrames = mFrameCount * 7;
mRsmpInFramesP2 = roundup(mRsmpInFrames);
free(mRsmpInBuffer);
+ mRsmpInBuffer = NULL;
// TODO optimize audio capture buffer sizes ...
// Here we calculate the size of the sliding buffer used as a source
@@ -6935,7 +6936,9 @@ 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
- (void)posix_memalign(&mRsmpInBuffer, 32, (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize);
+ size_t bufferSize = (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize;
+ (void)posix_memalign(&mRsmpInBuffer, 32, bufferSize);
+ memset(mRsmpInBuffer, 0, bufferSize); // if posix_memalign fails, will segv here.
// 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?