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Unified Diff: webrtc/modules/rtp_rtcp/source/rtcp_sender.h

Issue 1373903003: Unify newapi::RtcpMode and RTCPMethod. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 3 months ago
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Index: webrtc/modules/rtp_rtcp/source/rtcp_sender.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
index 2051102a2d436e6dc4a99b0154cc73e551d77697..e756803d7c2ed93cdb52144a654afd45c6cf90a1 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
@@ -79,8 +79,8 @@ public:
int32_t RegisterSendTransport(Transport* outgoingTransport);
- RTCPMethod Status() const;
- void SetRTCPStatus(RTCPMethod method);
+ RtcpMode Status() const;
+ void SetRTCPStatus(RtcpMode method);
bool Sending() const;
int32_t SetSendingStatus(const FeedbackState& feedback_state,
@@ -226,7 +226,7 @@ private:
private:
const bool audio_;
Clock* const clock_;
- RTCPMethod method_ GUARDED_BY(critical_section_rtcp_sender_);
+ RtcpMode method_ GUARDED_BY(critical_section_rtcp_sender_);
rtc::scoped_ptr<CriticalSectionWrapper> critical_section_transport_;
Transport* cbTransport_ GUARDED_BY(critical_section_transport_);

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