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
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

View File

@ -61,6 +61,7 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const WebRtc_Word32 id,
const bool audio): const bool audio):
TMMBRHelp(audio), TMMBRHelp(audio),
_rtpSender(id, audio), _rtpSender(id, audio),
_rtpReceiver(id, audio, *this),
_id(id), _id(id),
_audio(audio), _audio(audio),
_collisionDetected(false), _collisionDetected(false),
@ -76,7 +77,6 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const WebRtc_Word32 id,
_deadOrAliveActive(false), _deadOrAliveActive(false),
_deadOrAliveTimeoutMS(0), _deadOrAliveTimeoutMS(0),
_deadOrAliveLastTimer(0), _deadOrAliveLastTimer(0),
_rtpReceiver(id, audio, *this),
_rtcpReceiver(id,*this), _rtcpReceiver(id,*this),
_bandwidthManagement(id), _bandwidthManagement(id),
_receivedNTPsecsAudio(0), _receivedNTPsecsAudio(0),

View File

@ -526,6 +526,7 @@ protected:
TMMBRSet*& boundingSetRec); TMMBRSet*& boundingSetRec);
RTPSender _rtpSender; RTPSender _rtpSender;
RTPReceiver _rtpReceiver;
private: private:
void SendKeyFrame(); void SendKeyFrame();
@ -549,7 +550,6 @@ private:
WebRtc_UWord32 _deadOrAliveLastTimer; WebRtc_UWord32 _deadOrAliveLastTimer;
// receive side // receive side
RTPReceiver _rtpReceiver;
RTCPReceiver _rtcpReceiver; RTCPReceiver _rtcpReceiver;
BandwidthManagement _bandwidthManagement; BandwidthManagement _bandwidthManagement;