Wire up audio packet loss to BWE.

webrtc/voice_engine/channel.h

 /*
  *  Copyright (c) 2012 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 286 matching lines @@
   int SetLocalSSRC(unsigned int ssrc);
   int GetLocalSSRC(unsigned int& ssrc);
   int GetRemoteSSRC(unsigned int& ssrc);
   int SetSendAudioLevelIndicationStatus(bool enable, unsigned char id);
   int SetReceiveAudioLevelIndicationStatus(bool enable, unsigned char id);
   int SetSendAbsoluteSenderTimeStatus(bool enable, unsigned char id);
   int SetReceiveAbsoluteSenderTimeStatus(bool enable, unsigned char id);
   void EnableSendTransportSequenceNumber(int id);
   void EnableReceiveTransportSequenceNumber(int id);
 
-    void RegisterSenderCongestionControlObjects(
-        RtpPacketSender* rtp_packet_sender,
-        TransportFeedbackObserver* transport_feedback_observer,
-        PacketRouter* packet_router);
-    void RegisterReceiverCongestionControlObjects(PacketRouter* packet_router);
-    void ResetCongestionControlObjects();
+  void RegisterSenderCongestionControlObjects(
+      RtpPacketSender* rtp_packet_sender,
+      TransportFeedbackObserver* transport_feedback_observer,
+      PacketRouter* packet_router,
+      RtcpBandwidthObserver* bandwidth_observer);
+  void RegisterReceiverCongestionControlObjects(PacketRouter* packet_router);
+  void ResetCongestionControlObjects();
 
   void SetRTCPStatus(bool enable);
   int GetRTCPStatus(bool& enabled);
   int SetRTCP_CNAME(const char cName[256]);
   int GetRemoteRTCP_CNAME(char cName[256]);
   int GetRemoteRTCPData(unsigned int& NTPHigh,
                         unsigned int& NTPLow,
                         unsigned int& timestamp,
                         unsigned int& playoutTimestamp,
                         unsigned int* jitter,
@@ skipping 230 matching lines @@
   std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
 
   // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed.
   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_