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RTP module: Make sure payloadName is null terminated.

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Review URL: http://webrtc-codereview.appspot.com/268006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@908 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
mflodman@webrtc.org
2011-11-09 14:51:18 +00:00
parent f3c1b87f00
commit 03a9eb1526
1 changed files with 3 additions and 3 deletions
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6
src/modules/rtp_rtcp/source/rtp_receiver.cc
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@ -648,7 +648,7 @@ RTPReceiver::RemotePayload(WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
ModuleRTPUtility::Payload* payload = (ModuleRTPUtility::Payload*)item->GetItem();
if(payload)
{
memcpy(payloadName, payload->name, RTP_PAYLOAD_NAME_SIZE);
memcpy(payloadName, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
if(payloadType )
{
@ -1134,7 +1134,7 @@ RTPReceiver::CheckSSRCChanged(const WebRtcRTPHeader* rtpHeader)
ModuleRTPUtility::Payload* payload = (ModuleRTPUtility::Payload*)item->GetItem();
if(payload)
{
memcpy(payloadName, payload->name, RTP_PAYLOAD_NAME_SIZE);
memcpy(payloadName, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
if(payload->audio)
{
frequency = payload->typeSpecific.Audio.frequency;
@ -1244,7 +1244,7 @@ RTPReceiver::CheckPayloadChanged(const WebRtcRTPHeader* rtpHeader,
return -1;
}
memcpy(payloadName, payload->name, RTP_PAYLOAD_NAME_SIZE);
memcpy(payloadName, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
_lastReceivedPayloadType = payloadType;
reInitializeDecoder = true;