Reset the BWE when the network changes

webrtc/media/base/mediachannel.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.
  */
 
 #ifndef WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_
 #define WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_
 
 #include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/api/rtpparameters.h"
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/copyonwritebuffer.h"
 #include "webrtc/base/dscp.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/base/networkroute.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/socket.h"
 #include "webrtc/base/window.h"
 #include "webrtc/media/base/codec.h"
 #include "webrtc/media/base/mediaconstants.h"
 #include "webrtc/media/base/streamparams.h"
 #include "webrtc/media/base/videosinkinterface.h"
 // TODO(juberti): re-evaluate this include
 #include "webrtc/pc/audiomonitor.h"
@@ skipping 347 matching lines @@
     return rtc::DSCP_DEFAULT;
   }
   // Called when a RTP packet is received.
   virtual void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
                                 const rtc::PacketTime& packet_time) = 0;
   // Called when a RTCP packet is received.
   virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
                               const rtc::PacketTime& packet_time) = 0;
   // Called when the socket's ability to send has changed.
   virtual void OnReadyToSend(bool ready) = 0;
+  // Called when the network route used for sending packets changed.
+  virtual void OnNetworkRouteChanged(const std::string& transport_name,
+                                     const NetworkRoute& network_route) = 0;
   // Creates a new outgoing media stream with SSRCs and CNAME as described
   // by sp.
   virtual bool AddSendStream(const StreamParams& sp) = 0;
   // Removes an outgoing media stream.
   // ssrc must be the first SSRC of the media stream if the stream uses
   // multiple SSRCs.
   virtual bool RemoveSendStream(uint32_t ssrc) = 0;
   // Creates a new incoming media stream with SSRCs and CNAME as described
   // by sp.
   virtual bool AddRecvStream(const StreamParams& sp) = 0;
@@ skipping 697 matching lines @@
 
   virtual bool SetSendParameters(const DataSendParameters& params) = 0;
   virtual bool SetRecvParameters(const DataRecvParameters& params) = 0;
 
   // TODO(pthatcher): Implement this.
   virtual bool GetStats(DataMediaInfo* info) { return true; }
 
   virtual bool SetSend(bool send) = 0;
   virtual bool SetReceive(bool receive) = 0;
 
+  virtual void OnNetworkRouteChanged(const std::string& transport_name,
+                                     const NetworkRoute& network_route) {}
+
   virtual bool SendData(
       const SendDataParams& params,
       const rtc::CopyOnWriteBuffer& payload,
       SendDataResult* result = NULL) = 0;
   // Signals when data is received (params, data, len)
   sigslot::signal3<const ReceiveDataParams&,
                    const char*,
                    size_t> SignalDataReceived;
   // Signal when the media channel is ready to send the stream. Arguments are:
   //     writable(bool)
   sigslot::signal1<bool> SignalReadyToSend;
   // Signal for notifying that the remote side has closed the DataChannel.
   sigslot::signal1<uint32_t> SignalStreamClosedRemotely;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_