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Merge "RTP: prevent buffer overflow in AudioRecord." into gingerbread

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This commit is contained in:
Chia-chi Yeh
2010-09-09 02:37:07 -07:00
committed by Android (Google) Code Review
parent 4d1e012127 557b04de23
commit f6936a3a52
1 changed files with 2 additions and 2 deletions
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4
voip/jni/rtp/AudioGroup.cpp
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@@ -588,7 +588,7 @@ bool AudioGroup::set(int sampleRate, int sampleCount)
// Give device socket a reasonable timeout and buffer size.
timeval tv;
tv.tv_sec = 0;
tv.tv_usec = 1000 * sampleCount / sampleRate * 1000;
tv.tv_usec = 1000 * sampleCount / sampleRate * 500;
if (setsockopt(pair[0], SOL_SOCKET, SO_RCVTIMEO, &tv, sizeof(tv)) ||
setsockopt(pair[0], SOL_SOCKET, SO_RCVBUF, &output, sizeof(output)) ||
setsockopt(pair[1], SOL_SOCKET, SO_SNDBUF, &output, sizeof(output))) {
@@ -793,7 +793,7 @@ bool AudioGroup::deviceLoop()
status_t status = mRecord.obtainBuffer(&buffer, 1);
if (status == NO_ERROR) {
int count = (buffer.frameCount < toRead) ?
int count = ((int)buffer.frameCount < toRead) ?
buffer.frameCount : toRead;
memcpy(&input[mSampleCount - toRead], buffer.i8, count * 2);
toRead -= count;