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Moved _rtpReceiver to protected

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Review URL: http://webrtc-codereview.appspot.com/132005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@495 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pwestin@webrtc.org 2011-08-31 13:16:52 +00:00
parent c7d5f6249b
commit e9f0e2eb20
2 changed files with 2 additions and 2 deletions
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2
src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@ -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),

2
src/modules/rtp_rtcp/source/rtp_rtcp_impl.h
View File

@ -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;