Wire up audio packet loss to BWE.

webrtc/audio/audio_send_stream.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_AUDIO_AUDIO_SEND_STREAM_H_
 #define WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
 
 #include <memory>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/call/audio_send_stream.h"
 #include "webrtc/call/audio_state.h"
 #include "webrtc/call/bitrate_allocator.h"
 
 namespace webrtc {
 class CongestionController;
 class VoiceEngine;
 class RtcEventLog;
+class RtcpBandwidthObserver;
 class RtcpRttStats;
 class PacketRouter;
 
 namespace voe {
 class ChannelProxy;
 }  // namespace voe
 
 namespace internal {
 class AudioSendStream final : public webrtc::AudioSendStream,
                               public webrtc::BitrateAllocatorObserver {
@@ skipping 34 matching lines @@
   bool SetupSendCodec();
 
   rtc::ThreadChecker thread_checker_;
   rtc::TaskQueue* worker_queue_;
   const webrtc::AudioSendStream::Config config_;
   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
   std::unique_ptr<voe::ChannelProxy> channel_proxy_;
 
   BitrateAllocator* const bitrate_allocator_;
   CongestionController* const congestion_controller_;
+  std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
 };
 }  // namespace internal
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_