[cleanup] Remove unused fields/functions from rtcp module.

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.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 283 matching lines @@
 
   bool LastReceivedXrReferenceTimeInfo(RtcpReceiveTimeInfo* info) const;
 
   int32_t BoundingSet(bool* tmmbr_owner, TMMBRSet* bounding_set_rec);
 
   void BitrateSent(uint32_t* total_rate,
                    uint32_t* video_rate,
                    uint32_t* fec_rate,
                    uint32_t* nackRate) const override;
 
-  int64_t SendTimeOfSendReport(uint32_t send_report);
-
-  bool SendTimeOfXrRrReport(uint32_t mid_ntp, int64_t* time_ms) const;
-
   // Good state of RTP receiver inform sender.
   int32_t SendRTCPReferencePictureSelection(uint64_t picture_id) override;
 
   void RegisterSendChannelRtpStatisticsCallback(
       StreamDataCountersCallback* callback) override;
   StreamDataCountersCallback* GetSendChannelRtpStatisticsCallback()
       const override;
 
-  void OnReceivedTMMBR();
-
-  // Bad state of RTP receiver request a keyframe.
-  void OnRequestIntraFrame();
-
-  // Received a request for a new SLI.
-  void OnReceivedSliceLossIndication(uint8_t picture_id);
-
-  // Received a new reference frame.
-  void OnReceivedReferencePictureSelectionIndication(uint64_t picture_id);
-
   void OnReceivedNACK(const std::list<uint16_t>& nack_sequence_numbers);
 
   void OnRequestSendReport();
 
  protected:
   bool UpdateRTCPReceiveInformationTimers();
 
-  uint32_t BitrateReceivedNow() const;
-
-  // Get remote SequenceNumber.
-  uint16_t RemoteSequenceNumber() const;
-
   RTPSender rtp_sender_;
 
   RTCPSender rtcp_sender_;
   RTCPReceiver rtcp_receiver_;
 
   Clock* clock_;
 
  private:
   FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, Rtt);
   FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, RttForReceiverOnly);
   int64_t RtcpReportInterval();
   void SetRtcpReceiverSsrcs(uint32_t main_ssrc);
 
   void set_rtt_ms(int64_t rtt_ms);
   int64_t rtt_ms() const;
 
   bool TimeToSendFullNackList(int64_t now) const;
 
   const bool audio_;
   bool collision_detected_;
   int64_t last_process_time_;
   int64_t last_bitrate_process_time_;
   int64_t last_rtt_process_time_;
   uint16_t packet_overhead_;
 
-  size_t padding_index_;
-
   // Send side
-  NACKMethod nack_method_;
   int64_t nack_last_time_sent_full_;
   uint32_t nack_last_time_sent_full_prev_;
   uint16_t nack_last_seq_number_sent_;
 
-  VideoCodec send_video_codec_;
   KeyFrameRequestMethod key_frame_req_method_;
 
   RemoteBitrateEstimator* remote_bitrate_;
 
   RtcpRttStats* rtt_stats_;
 
   PacketLossStats send_loss_stats_;
   PacketLossStats receive_loss_stats_;
 
   // The processed RTT from RtcpRttStats.
   rtc::scoped_ptr<CriticalSectionWrapper> critical_section_rtt_;
   int64_t rtt_ms_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_