RTP: prevent buffer overflow in AudioRecord.
This change simply reduces the receive timeout of DeviceSocket. It works
because AudioRecord will block us till there is enough data, which makes
AudioSocket overlap AudioRecord.
Change-Id: I4700224fb407e148ef359a9d99279e10240128d0
diff --git a/voip/jni/rtp/AudioGroup.cpp b/voip/jni/rtp/AudioGroup.cpp
index bb45a9a..3433dcf 100644
--- a/voip/jni/rtp/AudioGroup.cpp
+++ b/voip/jni/rtp/AudioGroup.cpp
@@ -588,7 +588,7 @@
// 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 @@
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;