Add histogram for percentage of sent frames that are limited in resolution due to bandwidth.

webrtc/video_frame.h

 /*
  *  Copyright (c) 2014 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 149 matching lines @@
 
  private:
   // An opaque reference counted handle that stores the pixel data.
   rtc::scoped_refptr<webrtc::VideoFrameBuffer> video_frame_buffer_;
   uint32_t timestamp_;
   int64_t ntp_time_ms_;
   int64_t render_time_ms_;
   VideoRotation rotation_;
 };
 
+
 // TODO(pbos): Rename EncodedFrame and reformat this class' members.
 class EncodedImage {
  public:
   EncodedImage() : EncodedImage(nullptr, 0, 0) {}
   EncodedImage(uint8_t* buffer, size_t length, size_t size)
       : _buffer(buffer), _length(length), _size(size) {}
 
   struct AdaptReason {
     AdaptReason()
-        : quality_resolution_downscales(-1) {}
+        : quality_resolution_downscales(-1),
+          bw_resolutions_disabled(-1) {}
 
     int quality_resolution_downscales;  // Number of times this frame is down
                                         // scaled in resolution due to quality.
                                         // Or -1 if information is not provided.
+    int bw_resolutions_disabled;  // Number of resolutions that are not sent
+                                  // due to bandwidth for this frame.
+                                  // Or -1 if information is not provided.
   };
   uint32_t _encodedWidth = 0;
   uint32_t _encodedHeight = 0;
   uint32_t _timeStamp = 0;
   // NTP time of the capture time in local timebase in milliseconds.
   int64_t ntp_time_ms_ = 0;
   int64_t capture_time_ms_ = 0;
   FrameType _frameType = kDeltaFrame;
   uint8_t* _buffer;
   size_t _length;
   size_t _size;
   bool _completeFrame = false;
   AdaptReason adapt_reason_;
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_VIDEO_FRAME_H_