From fd3fef514bd68c33c4c9302bf2be81c6592b8268 Mon Sep 17 00:00:00 2001 From: "pwestin@webrtc.org" Date: Thu, 10 May 2012 15:14:41 +0000 Subject: [PATCH] Bugfix TBR Review URL: https://webrtc-codereview.appspot.com/570007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2214 4adac7df-926f-26a2-2b94-8c16560cd09d --- src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc | 3 +-- src/video_engine/vie_channel.cc | 1 + 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc index bbd4e298be..e2ab811ca5 100644 --- a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc +++ b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc @@ -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) { diff --git a/src/video_engine/vie_channel.cc b/src/video_engine/vie_channel.cc index fb0891f547..fcd1e30d64 100644 --- a/src/video_engine/vie_channel.cc +++ b/src/video_engine/vie_channel.cc @@ -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() {