Delete support for sending RTCP RPSI and SLI messages.

webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.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 126 matching lines @@
   MOCK_METHOD1(RemoveMixedCNAME, int32_t(uint32_t ssrc));
   MOCK_CONST_METHOD5(RTT,
                      int32_t(uint32_t remote_ssrc,
                              int64_t* rtt,
                              int64_t* avg_rtt,
                              int64_t* min_rtt,
                              int64_t* max_rtt));
   MOCK_METHOD1(SendRTCP, int32_t(RTCPPacketType packet_type));
   MOCK_METHOD1(SendCompoundRTCP,
                int32_t(const std::set<RTCPPacketType>& packet_types));
-  MOCK_METHOD1(SendRTCPReferencePictureSelection, int32_t(uint64_t picture_id));
-  MOCK_METHOD1(SendRTCPSliceLossIndication, int32_t(uint8_t picture_id));
   MOCK_CONST_METHOD2(DataCountersRTP,
                      int32_t(size_t* bytes_sent, uint32_t* packets_sent));
   MOCK_CONST_METHOD2(GetSendStreamDataCounters,
                      void(StreamDataCounters*, StreamDataCounters*));
   MOCK_CONST_METHOD3(GetRtpPacketLossStats,
                      void(bool, uint32_t, struct RtpPacketLossStats*));
   MOCK_METHOD1(RemoteRTCPStat, int32_t(RTCPSenderInfo* sender_info));
   MOCK_CONST_METHOD1(RemoteRTCPStat,
                      int32_t(std::vector<RTCPReportBlock>* receive_blocks));
   MOCK_METHOD4(SetRTCPApplicationSpecificData,
@@ skipping 46 matching lines @@
   MOCK_CONST_METHOD0(GetSendChannelRtpStatisticsCallback,
                      StreamDataCountersCallback*(void));
   MOCK_METHOD1(SetVideoBitrateAllocation, void(const BitrateAllocation&));
   // Members.
   unsigned int remote_ssrc_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_