Validates sending RTCP before RTP.

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 168 matching lines @@
     }
   }
 
   // Get processed rtt.
   if (process_rtt) {
     last_rtt_process_time_ = now;
     if (rtt_stats_)
       set_rtt_ms(rtt_stats_->LastProcessedRtt());
   }
 
-  // For sending streams, make sure to not send a SR before media has been sent.
-  if (rtcp_sender_.TimeToSendRTCPReport()) {
-    RTCPSender::FeedbackState state = GetFeedbackState();
-    // Prevent sending streams to send SR before any media has been sent.
-    if (!rtcp_sender_.Sending() || state.packets_sent > 0)
-      rtcp_sender_.SendRTCP(state, kRtcpReport);
-  }
+  if (rtcp_sender_.TimeToSendRTCPReport())
+    rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
 
   if (UpdateRTCPReceiveInformationTimers()) {
     // A receiver has timed out
     rtcp_receiver_.UpdateTMMBR();
   }
   return 0;
 }
 
 void ModuleRtpRtcpImpl::SetRtxSendStatus(int mode) {
   rtp_sender_.SetRtxStatus(mode);
@@ skipping 787 matching lines @@
 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
     StreamDataCountersCallback* callback) {
   rtp_sender_.RegisterRtpStatisticsCallback(callback);
 }
 
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
   return rtp_sender_.GetRtpStatisticsCallback();
 }
 }  // namespace webrtc