Wire up bandwidth limitation info to GetStats and adapt_reason.

webrtc/video_send_stream.h

 /*
  *  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 51 matching lines @@
   };
 
   struct Stats {
     int input_frame_rate = 0;
     int encode_frame_rate = 0;
     int avg_encode_time_ms = 0;
     int encode_usage_percent = 0;
     int target_media_bitrate_bps = 0;
     int media_bitrate_bps = 0;
     bool suspended = false;
+    bool bw_limited_resolution = false;
     std::map<uint32_t, StreamStats> substreams;
   };
 
   struct Config {
     Config() = delete;
     explicit Config(Transport* send_transport)
         : send_transport(send_transport) {}
 
     std::string ToString() const;
 
@@ skipping 95 matching lines @@
   // in the config. Encoder settings are passed on to the encoder instance along
   // with the VideoStream settings.
   virtual bool ReconfigureVideoEncoder(const VideoEncoderConfig& config) = 0;
 
   virtual Stats GetStats() = 0;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_SEND_STREAM_H_