RTP Receiver is now only deals with a receiver strategy. Cleaned up dependencies.
BUG= TESTED=vie/voe_auto_test, rtp_rtcp_unittests Review URL: https://webrtc-codereview.appspot.com/1058004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3397 4adac7df-926f-26a2-2b94-8c16560cd09d
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@ -17,6 +17,7 @@
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#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_payload_registry.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_receiver.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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@ -471,8 +472,10 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
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// Only for internal testing.
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WebRtc_UWord32 LastSendReport(WebRtc_UWord32& last_rtcptime);
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RTPPayloadRegistry rtp_payload_registry_;
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RTPSender rtp_sender_;
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RTPReceiver rtp_receiver_;
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scoped_ptr<RTPReceiver> rtp_receiver_;
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RTCPSender rtcp_sender_;
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RTCPReceiver rtcp_receiver_;
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@ -482,6 +485,8 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
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private:
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int64_t RtcpReportInterval();
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RTPReceiverAudio* rtp_telephone_event_handler_;
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WebRtc_Word32 id_;
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const bool audio_;
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bool collision_detected_;
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