Prevent RTCP SR to be sent with bogus timestamp.

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 165 matching lines @@
     }
   }
 
   // Get processed rtt.
   if (process_rtt) {
     last_rtt_process_time_ = now;
     if (rtt_stats_)
       set_rtt_ms(rtt_stats_->LastProcessedRtt());
   }
 
-  if (rtcp_sender_.TimeToSendRTCPReport())
-    rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
+  // 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 (UpdateRTCPReceiveInformationTimers()) {
     // A receiver has timed out
     rtcp_receiver_.UpdateTMMBR();
   }
   return 0;
 }
 
 void ModuleRtpRtcpImpl::SetRtxSendStatus(int mode) {
   rtp_sender_.SetRtxStatus(mode);
@@ skipping 197 matching lines @@
 int32_t ModuleRtpRtcpImpl::SendOutgoingData(
     FrameType frame_type,
     int8_t payload_type,
     uint32_t time_stamp,
     int64_t capture_time_ms,
     const uint8_t* payload_data,
     size_t payload_size,
     const RTPFragmentationHeader* fragmentation,
     const RTPVideoHeader* rtp_video_hdr) {
   rtcp_sender_.SetLastRtpTime(time_stamp, capture_time_ms);
+  // Make sure an RTCP report isn't queued behind a key frame.
   if (rtcp_sender_.TimeToSendRTCPReport(kVideoFrameKey == frame_type)) {
       rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
   }
   return rtp_sender_.SendOutgoingData(
       frame_type, payload_type, time_stamp, capture_time_ms, payload_data,
       payload_size, fragmentation, rtp_video_hdr);
 }
 
 bool ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc,
                                          uint16_t sequence_number,
@@ skipping 570 matching lines @@
 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
     StreamDataCountersCallback* callback) {
   rtp_sender_.RegisterRtpStatisticsCallback(callback);
 }
 
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
   return rtp_sender_.GetRtpStatisticsCallback();
 }
 }  // namespace webrtc