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_); |