Remove contention between RTCP packets and encoding.

webrtc/modules/video_coding/main/source/video_sender.cc

 /*
  *  Copyright (c) 2013 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 351 matching lines @@
   } else {
     threshold_bps = current_codec_.simulcastStream[0].minBitrate * 1000;
   }
   // Set the hysteresis window to be at 10% of the threshold, but at least
   // 10 kbps.
   int window_bps = std::max(threshold_bps / 10, 10000);
   _mediaOpt.SuspendBelowMinBitrate(threshold_bps, window_bps);
 }
 
 bool VideoSender::VideoSuspended() const {
-  rtc::CritScope lock(&send_crit_);
   return _mediaOpt.IsVideoSuspended();
 }
 }  // namespace vcm
 }  // namespace webrtc