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 @@
             // 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 @@
     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 @@
     // 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?