From e9f0e2eb20d9e6af7dcfc652a2776d90fcf7761d Mon Sep 17 00:00:00 2001 From: "pwestin@webrtc.org" Date: Wed, 31 Aug 2011 13:16:52 +0000 Subject: [PATCH] Moved _rtpReceiver to protected Review URL: http://webrtc-codereview.appspot.com/132005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@495 4adac7df-926f-26a2-2b94-8c16560cd09d --- src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc | 2 +- src/modules/rtp_rtcp/source/rtp_rtcp_impl.h | 2 +- 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 a1d76738a..97e699a76 100644 --- a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc +++ b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc @@ -61,6 +61,7 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const WebRtc_Word32 id, const bool audio): TMMBRHelp(audio), _rtpSender(id, audio), + _rtpReceiver(id, audio, *this), _id(id), _audio(audio), _collisionDetected(false), @@ -76,7 +77,6 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const WebRtc_Word32 id, _deadOrAliveActive(false), _deadOrAliveTimeoutMS(0), _deadOrAliveLastTimer(0), - _rtpReceiver(id, audio, *this), _rtcpReceiver(id,*this), _bandwidthManagement(id), _receivedNTPsecsAudio(0), diff --git a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h index a79e3ee7c..b0be027c0 100644 --- a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h +++ b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h @@ -526,6 +526,7 @@ protected: TMMBRSet*& boundingSetRec); RTPSender _rtpSender; + RTPReceiver _rtpReceiver; private: void SendKeyFrame(); @@ -549,7 +550,6 @@ private: WebRtc_UWord32 _deadOrAliveLastTimer; // receive side - RTPReceiver _rtpReceiver; RTCPReceiver _rtcpReceiver; BandwidthManagement _bandwidthManagement;