Fix perf issue when timinig out receivers infos in RTCP.

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 193 matching lines @@
       // Make sure we have a valid RTT before setting.
       int64_t last_rtt = rtt_stats_->LastProcessedRtt();
       if (last_rtt >= 0)
         set_rtt_ms(last_rtt);
     }
   }
 
   if (rtcp_sender_.TimeToSendRTCPReport())
     rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
 
-  if (UpdateRTCPReceiveInformationTimers()) {
+  if (TMMBR() && rtcp_receiver_.UpdateRTCPReceiveInformationTimers()) {
     // A receiver has timed out.
     rtcp_receiver_.UpdateTmmbr();
   }
 }
 
 void ModuleRtpRtcpImpl::SetRtxSendStatus(int mode) {
   rtp_sender_.SetRtxStatus(mode);
 }
 
 int ModuleRtpRtcpImpl::RtxSendStatus() const {
@@ skipping 635 matching lines @@
                           rtcp_arrival_time_secs,
                           rtcp_arrival_time_frac,
                           NULL)) {
     return false;
   }
   *remote_sr =
       ((ntp_secs & 0x0000ffff) << 16) + ((ntp_frac & 0xffff0000) >> 16);
   return true;
 }
 
-bool ModuleRtpRtcpImpl::UpdateRTCPReceiveInformationTimers() {
-  // If this returns true this channel has timed out.
-  // Periodically check if this is true and if so call UpdateTMMBR.
-  return rtcp_receiver_.UpdateRTCPReceiveInformationTimers();
-}
-
 // Called from RTCPsender.
 std::vector<rtcp::TmmbItem> ModuleRtpRtcpImpl::BoundingSet(bool* tmmbr_owner) {
   return rtcp_receiver_.BoundingSet(tmmbr_owner);
 }
 
 int64_t ModuleRtpRtcpImpl::RtcpReportInterval() {
   if (audio_)
     return RTCP_INTERVAL_AUDIO_MS;
   else
     return RTCP_INTERVAL_VIDEO_MS;
@@ skipping 25 matching lines @@
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
   return rtp_sender_.GetRtpStatisticsCallback();
 }
 
 void ModuleRtpRtcpImpl::SetVideoBitrateAllocation(
     const BitrateAllocation& bitrate) {
   rtcp_sender_.SetVideoBitrateAllocation(bitrate);
 }
 }  // namespace webrtc