Wire up send-side bandwidth estimation.

webrtc/video/video_quality_test.h

 /*
  *  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.
  */
 #ifndef WEBRTC_VIDEO_VIDEO_QUALITY_TEST_H_
@@ skipping 20 matching lines @@
       int32_t fps;
       int min_bitrate_bps;
       int target_bitrate_bps;
       int max_bitrate_bps;
       std::string codec;
       size_t num_temporal_layers;
 
       int min_transmit_bps;
       Call::Config::BitrateConfig call_bitrate_config;
       size_t tl_discard_threshold;
+      bool send_side_bwe;
     } common;
     struct {  // Video-specific settings.
       std::string clip_name;
     } video;
     struct {  // Screenshare-specific settings.
       bool enabled;
       int32_t slide_change_interval;
       int32_t scroll_duration;
     } screenshare;
     struct {  // Analyzer settings.
@@ skipping 26 matching lines @@
   rtc::scoped_ptr<test::TraceToStderr> trace_to_stderr_;
   rtc::scoped_ptr<test::FrameGenerator> frame_generator_;
   rtc::scoped_ptr<VideoEncoder> encoder_;
   VideoCodecUnion codec_settings_;
   Clock* const clock_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_VIDEO_QUALITY_TEST_H_