Unify newapi::RtcpMode and RTCPMethod.

webrtc/modules/rtp_rtcp/source/rtcp_sender.h

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 61 matching lines @@
  };
 
  RTCPSender(bool audio,
             Clock* clock,
             ReceiveStatistics* receive_statistics,
             RtcpPacketTypeCounterObserver* packet_type_counter_observer);
  virtual ~RTCPSender();
 
  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,
                           bool enabled);  // combine the functions
 
  int32_t SetNackStatus(bool enable);
 
  void SetStartTimestamp(uint32_t start_timestamp);
 
  void SetLastRtpTime(uint32_t rtp_timestamp, int64_t capture_time_ms);
@@ skipping 125 matching lines @@
  BuildResult BuildNACK(RtcpContext* context)
      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
  BuildResult BuildReceiverReferenceTime(RtcpContext* context)
      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
  BuildResult BuildDlrr(RtcpContext* context)
      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
 
 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_);
 
  rtc::scoped_ptr<CriticalSectionWrapper> critical_section_rtcp_sender_;
  bool using_nack_ GUARDED_BY(critical_section_rtcp_sender_);
  bool sending_ GUARDED_BY(critical_section_rtcp_sender_);
  bool remb_enabled_ GUARDED_BY(critical_section_rtcp_sender_);
 
  int64_t next_time_to_send_rtcp_ GUARDED_BY(critical_section_rtcp_sender_);
@@ skipping 79 matching lines @@
  std::set<ReportFlag> report_flags_ GUARDED_BY(critical_section_rtcp_sender_);
 
  typedef BuildResult (RTCPSender::*Builder)(RtcpContext*);
  std::map<RTCPPacketType, Builder> builders_;
 
  class PacketBuiltCallback;
 };
 }  // namespace webrtc
 
 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_