Delete support for sending RTCP RPSI and SLI messages.

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 95 matching lines @@
   int32_t AddMixedCNAME(uint32_t SSRC, const char* c_name);
 
   int32_t RemoveMixedCNAME(uint32_t SSRC);
 
   bool TimeToSendRTCPReport(bool sendKeyframeBeforeRTP = false) const;
 
   int32_t SendRTCP(const FeedbackState& feedback_state,
                    RTCPPacketType packetType,
                    int32_t nackSize = 0,
                    const uint16_t* nackList = 0,
                    uint64_t pictureID = 0);

[danilchap, 2017/03/14 13:31:47]

may be remove this parameter too

[nisse-webrtc, 2017/03/14 13:41:09]

Done.

 
   int32_t SendCompoundRTCP(const FeedbackState& feedback_state,
                            const std::set<RTCPPacketType>& packetTypes,
                            int32_t nackSize = 0,
                            const uint16_t* nackList = 0,
                            uint64_t pictureID = 0);

[danilchap, 2017/03/14 13:31:46]

may be remove this parameter too

[nisse-webrtc, 2017/03/14 13:41:09]

Done.

 
   bool REMB() const;
 
   void SetREMBStatus(bool enable);
 
   void SetREMBData(uint32_t bitrate, const std::vector<uint32_t>& ssrcs);
 
   bool TMMBR() const;
 
   void SetTMMBRStatus(bool enable);
@@ skipping 46 matching lines @@
       EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
   std::unique_ptr<rtcp::RtcpPacket> BuildAPP(const RtcpContext& context)
       EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
   std::unique_ptr<rtcp::RtcpPacket> BuildExtendedReports(
       const RtcpContext& context)
       EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
   std::unique_ptr<rtcp::RtcpPacket> BuildBYE(const RtcpContext& context)
       EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
   std::unique_ptr<rtcp::RtcpPacket> BuildFIR(const RtcpContext& context)
       EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
-  std::unique_ptr<rtcp::RtcpPacket> BuildSLI(const RtcpContext& context)
-      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
-  std::unique_ptr<rtcp::RtcpPacket> BuildRPSI(const RtcpContext& context)
-      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
   std::unique_ptr<rtcp::RtcpPacket> BuildNACK(const RtcpContext& context)
       EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
 
  private:
   const bool audio_;
   Clock* const clock_;
   Random random_ GUARDED_BY(critical_section_rtcp_sender_);
   RtcpMode method_ GUARDED_BY(critical_section_rtcp_sender_);
 
   RtcEventLog* const event_log_;
@@ skipping 85 matching lines @@
   typedef std::unique_ptr<rtcp::RtcpPacket> (RTCPSender::*BuilderFunc)(
       const RtcpContext&);
   // Map from RTCPPacketType to builder.
   std::map<uint32_t, BuilderFunc> builders_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTCPSender);
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_