Reset the BWE when the network changes

webrtc/media/engine/webrtcvoiceengine.h

 /*
  *  Copyright (c) 2004 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 166 matching lines @@
   bool SetOutputVolume(uint32_t ssrc, double volume) override;
 
   bool CanInsertDtmf() override;
   bool InsertDtmf(uint32_t ssrc, int event, int duration) override;
 
   void OnPacketReceived(rtc::Buffer* packet,
                         const rtc::PacketTime& packet_time) override;
   void OnRtcpReceived(rtc::Buffer* packet,
                       const rtc::PacketTime& packet_time) override;
   void OnReadyToSend(bool ready) override {}
+  void OnBestConnectionChanged(Connection* connection) override;
   bool GetStats(VoiceMediaInfo* info) override;
 
   void SetRawAudioSink(
       uint32_t ssrc,
       std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
 
   // implements Transport interface
   bool SendRtp(const uint8_t* data,
                size_t len,
                const webrtc::PacketOptions& options) override {
@@ skipping 87 matching lines @@
     int cng_payload_type = -1;
     int cng_plfreq = -1;
     webrtc::CodecInst codec_inst;
   } send_codec_spec_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
 };
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_