Avoid rescheduling the next RTCP packet if the RTCP sender status doesn't change.

webrtc/modules/rtp_rtcp/source/rtcp_sender.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 187 matching lines @@
   builders_[kRtcpXrDlrrReportBlock] = &RTCPSender::BuildDlrr;
 }
 
 RTCPSender::~RTCPSender() {}
 
 RtcpMode RTCPSender::Status() const {
   rtc::CritScope lock(&critical_section_rtcp_sender_);
   return method_;
 }
 
-void RTCPSender::SetRTCPStatus(RtcpMode method) {
+void RTCPSender::SetRTCPStatus(RtcpMode new_method) {
   rtc::CritScope lock(&critical_section_rtcp_sender_);
-  method_ = method;
 
-  if (method == RtcpMode::kOff)
-    return;
-  next_time_to_send_rtcp_ =
+  if (method_ == RtcpMode::kOff && new_method != RtcpMode::kOff) {
+    // When switching on, reschedule the next packet
+    next_time_to_send_rtcp_ =
       clock_->TimeInMilliseconds() +
       (audio_ ? RTCP_INTERVAL_AUDIO_MS / 2 : RTCP_INTERVAL_VIDEO_MS / 2);
+  }
+  method_ = new_method;
 }
 
 bool RTCPSender::Sending() const {
   rtc::CritScope lock(&critical_section_rtcp_sender_);
   return sending_;
 }
 
 int32_t RTCPSender::SetSendingStatus(const FeedbackState& feedback_state,
                                      bool sending) {
   bool sendRTCPBye = false;
@@ skipping 802 matching lines @@
     // RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Sender);
     // but we can't because of an incorrect warning (C4822) in MVS 2013.
   } sender(transport_, event_log_);
 
   uint8_t buffer[IP_PACKET_SIZE];
   return packet.BuildExternalBuffer(buffer, IP_PACKET_SIZE, &sender) &&
          !sender.send_failure_;
 }
 
 }  // namespace webrtc