Removed extended jitter report from RtcpSender

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 116 matching lines @@
  void SetREMBStatus(bool enable);
 
  void SetREMBData(uint32_t bitrate, const std::vector<uint32_t>& ssrcs);
 
  bool TMMBR() const;
 
  void SetTMMBRStatus(bool enable);
 
  int32_t SetTMMBN(const TMMBRSet* boundingSet, uint32_t maxBitrateKbit);
 
- // Extended jitter report
- bool IJ() const;
-
- void SetIJStatus(bool enable);
-
  int32_t SetApplicationSpecificData(uint8_t subType,
                                     uint32_t name,
                                     const uint8_t* data,
                                     uint16_t length);
  int32_t SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric);
 
  void SendRtcpXrReceiverReferenceTime(bool enable);
 
  bool RtcpXrReceiverReferenceTime() const;
 
@@ skipping 39 matching lines @@
                  const uint16_t* nackList,
                  bool repeat,
                  uint64_t pictureID,
                  uint8_t* rtcp_buffer,
                  int buffer_size);
 
  BuildResult BuildSR(RtcpContext* context)
      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
  BuildResult BuildRR(RtcpContext* context)
      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
- BuildResult BuildExtendedJitterReport(RtcpContext* context)
-     EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
  BuildResult BuildSDES(RtcpContext* context)
      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
  BuildResult BuildPLI(RtcpContext* context)
      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
  BuildResult BuildREMB(RtcpContext* context)
      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
  BuildResult BuildTMMBR(RtcpContext* context)
      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
  BuildResult BuildTMMBN(RtcpContext* context)
      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
@@ skipping 22 matching lines @@
  Clock* const clock_;
  RTCPMethod 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_);
- bool extended_jitter_report_enabled_ GUARDED_BY(critical_section_rtcp_sender_);
 
  int64_t next_time_to_send_rtcp_ GUARDED_BY(critical_section_rtcp_sender_);
 
  uint32_t start_timestamp_ GUARDED_BY(critical_section_rtcp_sender_);
  uint32_t last_rtp_timestamp_ GUARDED_BY(critical_section_rtcp_sender_);
  int64_t last_frame_capture_time_ms_ GUARDED_BY(critical_section_rtcp_sender_);
  uint32_t ssrc_ GUARDED_BY(critical_section_rtcp_sender_);
  // SSRC that we receive on our RTP channel
  uint32_t remote_ssrc_ GUARDED_BY(critical_section_rtcp_sender_);
  std::string cname_ GUARDED_BY(critical_section_rtcp_sender_);
@@ skipping 71 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_