Reset the BWE when the network changes

webrtc/media/base/fakemediaengine.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_FAKEMEDIAENGINE_H_
 #define WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_
 
 #include <list>
 #include <map>
 #include <memory>
 #include <set>
 #include <string>
 #include <vector>
 
 #include "webrtc/audio_sink.h"
 #include "webrtc/base/copyonwritebuffer.h"
+#include "webrtc/base/networkroute.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/media/base/audiosource.h"
 #include "webrtc/media/base/mediaengine.h"
 #include "webrtc/media/base/rtputils.h"
 #include "webrtc/media/base/streamparams.h"
 #include "webrtc/p2p/base/sessiondescription.h"
 
 namespace cricket {
 
 class FakeMediaEngine;
@@ skipping 138 matching lines @@
   const std::string rtcp_cname() {
     if (send_streams_.empty())
       return "";
     return send_streams_[0].cname;
   }
 
   bool ready_to_send() const {
     return ready_to_send_;
   }
 
+  NetworkRoute last_network_route() const { return last_network_route_; }
+  int num_network_route_changes() const { return num_network_route_changes_; }
+  void set_num_network_route_changes(int changes) {
+    num_network_route_changes_ = changes;
+  }
+
  protected:
   bool MuteStream(uint32_t ssrc, bool mute) {
     if (!HasSendStream(ssrc) && ssrc != 0) {
       return false;
     }
     if (mute) {
       muted_streams_.insert(ssrc);
     } else {
       muted_streams_.erase(ssrc);
     }
@@ skipping 18 matching lines @@
                                 const rtc::PacketTime& packet_time) {
     rtp_packets_.push_back(std::string(packet->data<char>(), packet->size()));
   }
   virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
                               const rtc::PacketTime& packet_time) {
     rtcp_packets_.push_back(std::string(packet->data<char>(), packet->size()));
   }
   virtual void OnReadyToSend(bool ready) {
     ready_to_send_ = ready;
   }
+  virtual void OnNetworkRouteChanged(const std::string& transport_name,
+                                     const NetworkRoute& network_route) {
+    last_network_route_ = network_route;
+    ++num_network_route_changes_;
+  }
   bool fail_set_send_codecs() const { return fail_set_send_codecs_; }
   bool fail_set_recv_codecs() const { return fail_set_recv_codecs_; }
 
  private:
   bool sending_;
   bool playout_;
   std::vector<RtpHeaderExtension> recv_extensions_;
   std::vector<RtpHeaderExtension> send_extensions_;
   std::list<std::string> rtp_packets_;
   std::list<std::string> rtcp_packets_;
   std::vector<StreamParams> send_streams_;
   std::vector<StreamParams> receive_streams_;
   std::set<uint32_t> muted_streams_;
   std::map<uint32_t, webrtc::RtpParameters> rtp_parameters_;
   bool fail_set_send_codecs_;
   bool fail_set_recv_codecs_;
   uint32_t send_ssrc_;
   std::string rtcp_cname_;
   bool ready_to_send_;
+  NetworkRoute last_network_route_;
+  int num_network_route_changes_ = 0;
 };
 
 class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
  public:
   struct DtmfInfo {
     DtmfInfo(uint32_t ssrc, int event_code, int duration)
         : ssrc(ssrc),
           event_code(event_code),
           duration(duration) {}
     uint32_t ssrc;
@@ skipping 644 matching lines @@
 
  private:
   std::vector<FakeDataMediaChannel*> channels_;
   std::vector<DataCodec> data_codecs_;
   DataChannelType last_channel_type_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_