Add more logging about the bwe state and VideoSendStream state.

webrtc/video/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 121 matching lines @@
   static const int kMaxLayers = 3;
   std::unique_ptr<IvfFileWriter> file_writers_[kMaxLayers];
 
   rtc::PlatformThread encoder_thread_;
   rtc::Event encoder_wakeup_event_;
   volatile int stop_encoder_thread_;
   rtc::CriticalSection encoder_settings_crit_;
   std::unique_ptr<EncoderSettings> pending_encoder_settings_
       GUARDED_BY(encoder_settings_crit_);
   uint32_t encoder_max_bitrate_bps_ GUARDED_BY(encoder_settings_crit_);
+  uint32_t encoder_target_rate_bps_ GUARDED_BY(encoder_settings_crit_);
 
   enum class State {
     kStopped,  // VideoSendStream::Start has not yet been called.
     kStarted,  // VideoSendStream::Start has been called.
     // VideoSendStream::Start has been called but the encoder have timed out.
     kEncoderTimedOut,
   };
   rtc::Optional<State> pending_state_change_ GUARDED_BY(encoder_settings_crit_);
 
   // Only used on the encoder thread.
@@ skipping 12 matching lines @@
   const std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_;
   // RtpRtcp modules, declared here as they use other members on construction.
   const std::vector<RtpRtcp*> rtp_rtcp_modules_;
   PayloadRouter payload_router_;
   VideoCaptureInput input_;
 };
 }  // namespace internal
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_VIDEO_SEND_STREAM_H_