Update video adaptation stats to support degradations in both resolution and framerate.

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 @@
     // limitations.
     int target_media_bitrate_bps = 0;
     // Bitrate the encoder is actually producing.
     int media_bitrate_bps = 0;
     // Media bitrate this VideoSendStream is configured to prefer if there are
     // no bandwidth limitations.
     int preferred_media_bitrate_bps = 0;
     bool suspended = false;
     bool bw_limited_resolution = false;
     bool cpu_limited_resolution = false;
+    bool bw_limited_framerate = false;
+    bool cpu_limited_framerate = false;
     // Total number of times resolution as been requested to be changed due to
     // CPU/quality adaptation.
     int number_of_cpu_adapt_changes = 0;
     int number_of_quality_adapt_changes = 0;
     std::map<uint32_t, StreamStats> substreams;
   };
 
   struct Config {
    public:
     Config() = delete;
@@ skipping 175 matching lines @@
     EnableEncodedFrameRecording(std::vector<rtc::PlatformFile>(), 0);
   }
 
  protected:
   virtual ~VideoSendStream() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_SEND_STREAM_H_