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Bugfix

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TBR
Review URL: https://webrtc-codereview.appspot.com/570007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2214 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pwestin@webrtc.org 2012-05-10 15:14:41 +00:00
parent 80f7ac23b0
commit fd3fef514b
2 changed files with 2 additions and 2 deletions
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3
src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@ -41,7 +41,7 @@ RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) {
return new ModuleRtpRtcpImpl(configuration);
} else {
RtpRtcp::Configuration configuration_copy;
memcpy(&configuration_copy, &configuration_copy,
memcpy(&configuration_copy, &configuration,
sizeof(RtpRtcp::Configuration));
configuration_copy.clock = ModuleRTPUtility::GetSystemClock();
ModuleRtpRtcpImpl* rtp_rtcp_instance =
@ -484,7 +484,6 @@ WebRtc_Word32 ModuleRtpRtcpImpl::IncomingPacket(
_id,
"IncomingPacket(packetLength:%u)",
incomingPacketLength);
// minimum RTP is 12 bytes
// minimum RTCP is 8 bytes (RTCP BYE)
if (incomingPacketLength < 8 || incomingPacket == NULL) {

1
src/video_engine/vie_channel.cc
View File

@ -93,6 +93,7 @@ ViEChannel::ViEChannel(WebRtc_Word32 channel_id,
configuration.bitrate_observer = bitrate_observer;
rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(configuration));
vie_receiver_.SetRtpRtcpModule(rtp_rtcp_.get());
}
WebRtc_Word32 ViEChannel::Init() {