Delete support for sending RTCP RPSI and SLI messages.

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

 /*
  *  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 736 matching lines @@
 int32_t ModuleRtpRtcpImpl::RequestKeyFrame() {
   switch (key_frame_req_method_) {
     case kKeyFrameReqPliRtcp:
       return SendRTCP(kRtcpPli);
     case kKeyFrameReqFirRtcp:
       return SendRTCP(kRtcpFir);
   }
   return -1;
 }
 
-int32_t ModuleRtpRtcpImpl::SendRTCPSliceLossIndication(uint8_t picture_id) {
-  return rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpSli, 0, 0, picture_id);
-}
-
 void ModuleRtpRtcpImpl::SetUlpfecConfig(int red_payload_type,
                                         int ulpfec_payload_type) {
   rtp_sender_.SetUlpfecConfig(red_payload_type, ulpfec_payload_type);
 }
 
 bool ModuleRtpRtcpImpl::SetFecParameters(
     const FecProtectionParams& delta_params,
     const FecProtectionParams& key_params) {
   return rtp_sender_.SetFecParameters(delta_params, key_params);
 }
@@ skipping 11 matching lines @@
   *total_rate = rtp_sender_.BitrateSent();
   *video_rate = rtp_sender_.VideoBitrateSent();
   *fec_rate = rtp_sender_.FecOverheadRate();
   *nack_rate = rtp_sender_.NackOverheadRate();
 }
 
 void ModuleRtpRtcpImpl::OnRequestSendReport() {
   SendRTCP(kRtcpSr);
 }
 
-int32_t ModuleRtpRtcpImpl::SendRTCPReferencePictureSelection(
-    const uint64_t picture_id) {
-  return rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpRpsi, 0, 0, picture_id);
-}
-
 void ModuleRtpRtcpImpl::OnReceivedNack(
     const std::vector<uint16_t>& nack_sequence_numbers) {
   for (uint16_t nack_sequence_number : nack_sequence_numbers) {
     send_loss_stats_.AddLostPacket(nack_sequence_number);
   }
   if (!rtp_sender_.StorePackets() ||
       nack_sequence_numbers.size() == 0) {
     return;
   }
   // Use RTT from RtcpRttStats class if provided.
@@ skipping 67 matching lines @@
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
   return rtp_sender_.GetRtpStatisticsCallback();
 }
 
 void ModuleRtpRtcpImpl::SetVideoBitrateAllocation(
     const BitrateAllocation& bitrate) {
   rtcp_sender_.SetVideoBitrateAllocation(bitrate);
 }
 }  // namespace webrtc