Wire up audio packet loss to BWE.

webrtc/voice_engine/channel_proxy.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_VOICE_ENGINE_CHANNEL_PROXY_H_
 #define WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_
 
 #include "webrtc/api/audio/audio_mixer.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/race_checker.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/voice_engine/channel_manager.h"
 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
 
 #include <memory>
 #include <string>
 #include <vector>
 
 namespace webrtc {
 
 class AudioSinkInterface;
 class PacketRouter;
 class RtcEventLog;
+class RtcpBandwidthObserver;
 class RtcpRttStats;
 class RtpPacketSender;
 class RtpReceiver;
 class RtpRtcp;
 class Transport;
 class TransportFeedbackObserver;
 
 namespace voe {
 
 class Channel;
@@ skipping 15 matching lines @@
   virtual void SetLocalSSRC(uint32_t ssrc);
   virtual void SetRTCP_CNAME(const std::string& c_name);
   virtual void SetNACKStatus(bool enable, int max_packets);
   virtual void SetSendAudioLevelIndicationStatus(bool enable, int id);
   virtual void SetReceiveAudioLevelIndicationStatus(bool enable, int id);
   virtual void EnableSendTransportSequenceNumber(int id);
   virtual void EnableReceiveTransportSequenceNumber(int id);
   virtual void RegisterSenderCongestionControlObjects(
       RtpPacketSender* rtp_packet_sender,
       TransportFeedbackObserver* transport_feedback_observer,
-      PacketRouter* packet_router);
+      PacketRouter* packet_router,
+      RtcpBandwidthObserver* bandwidth_observer);
   virtual void RegisterReceiverCongestionControlObjects(
       PacketRouter* packet_router);
   virtual void ResetCongestionControlObjects();
   virtual CallStatistics GetRTCPStatistics() const;
   virtual std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const;
   virtual NetworkStatistics GetNetworkStatistics() const;
   virtual AudioDecodingCallStats GetDecodingCallStatistics() const;
   virtual int32_t GetSpeechOutputLevelFullRange() const;
   virtual uint32_t GetDelayEstimate() const;
   virtual bool SetSendTelephoneEventPayloadType(int payload_type,
@@ skipping 47 matching lines @@
   rtc::ThreadChecker thread_checker_;
   rtc::RaceChecker race_checker_;
   ChannelOwner channel_owner_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(ChannelProxy);
 };
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_