Auto pause video streams based on encoder target bitrate.

webrtc/video/video_quality_test.cc

 /*
  *  Copyright (c) 2015 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.
  */
 #include <stdio.h>
@@ skipping 1107 matching lines @@
   // TODO(ivica): Use two calls to be able to merge with RunWithAnalyzer or at
   // least share as much code as possible. That way this test would also match
   // the full stack tests better.
   transport.SetReceiver(call->Receiver());
 
   SetupCommon(&transport, &transport);
 
   video_send_config_.local_renderer = local_preview.get();
   video_receive_configs_[stream_id].renderer = loopback_video.get();
 
+  video_send_config_.suspend_below_min_bitrate =
+      params_.common.suspend_below_min_bitrate;
+
   if (params.common.fec) {
     video_send_config_.rtp.fec.red_payload_type = kRedPayloadType;
     video_send_config_.rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
     video_receive_configs_[stream_id].rtp.fec.red_payload_type =
       kRedPayloadType;
     video_receive_configs_[stream_id].rtp.fec.ulpfec_payload_type =
       kUlpfecPayloadType;
   }
 
   if (params_.screenshare.enabled)
@@ skipping 15 matching lines @@
   video_send_stream_->Stop();
   receive_stream->Stop();
 
   call->DestroyVideoReceiveStream(receive_stream);
   call->DestroyVideoSendStream(video_send_stream_);
 
   transport.StopSending();
 }
 
 }  // namespace webrtc