| commit | bd4a2feddb3002ecb9c9b67b812d2648fd02cfa7 | [log] [tgz] |
|---|---|---|
| author | sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | Fri May 03 18:11:36 2013 +0000 |
| committer | sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | Fri May 03 18:11:36 2013 +0000 |
| tree | 5df88e2516aa250509df9bfa49acbd9449820127 | |
| parent | d3cd565ecf6e2c19769368ebd9bebe1122753c3e [diff] [blame] |
Fix off-by-one buffer overflow in WebRtcNetEQ_PacketBufferInsert(). BUG=1725 R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1395004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3953 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/neteq/packet_buffer.c b/webrtc/modules/audio_coding/neteq/packet_buffer.c index 83079d7..a542333 100644 --- a/webrtc/modules/audio_coding/neteq/packet_buffer.c +++ b/webrtc/modules/audio_coding/neteq/packet_buffer.c
@@ -324,8 +324,8 @@ { /* Payload is 16-bit aligned => just copy it */ - WEBRTC_SPL_MEMCPY_W16(bufferInst->currentMemoryPos, - RTPpacket->payload, (RTPpacket->payloadLen + 1) >> 1); + WEBRTC_SPL_MEMCPY_W8(bufferInst->currentMemoryPos, + RTPpacket->payload, RTPpacket->payloadLen); } else {