Update the call when the network route changes

webrtc/call/call.cc

 /*
  *  Copyright (c) 2013 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 71 matching lines @@
   DeliveryStatus DeliverPacket(MediaType media_type,
                                const uint8_t* packet,
                                size_t length,
                                const PacketTime& packet_time) override;
 
   void SetBitrateConfig(
       const webrtc::Call::Config::BitrateConfig& bitrate_config) override;
 
   void SignalChannelNetworkState(MediaType media, NetworkState state) override;
 
+  void OnNetworkRouteChanged(const std::string& transport_name,
+                             const rtc::NetworkRoute& network_route) override;
+
   void OnSentPacket(const rtc::SentPacket& sent_packet) override;
 
   // Implements BitrateObserver.
   void OnNetworkChanged(uint32_t bitrate_bps, uint8_t fraction_loss,
                         int64_t rtt_ms) override;
 
  private:
   DeliveryStatus DeliverRtcp(MediaType media_type, const uint8_t* packet,
                              size_t length);
   DeliveryStatus DeliverRtp(MediaType media_type,
                             const uint8_t* packet,
                             size_t length,
                             const PacketTime& packet_time);
-
   void ConfigureSync(const std::string& sync_group)
       EXCLUSIVE_LOCKS_REQUIRED(receive_crit_);
 
   VoiceEngine* voice_engine() {
     internal::AudioState* audio_state =
         static_cast<internal::AudioState*>(config_.audio_state.get());
     if (audio_state)
       return audio_state->voice_engine();
     else
       return nullptr;
@@ skipping 47 matching lines @@
   int64_t last_rtp_packet_received_ms_;
   int64_t first_packet_sent_ms_;
 
   // TODO(holmer): Remove this lock once BitrateController no longer calls
   // OnNetworkChanged from multiple threads.
   rtc::CriticalSection bitrate_crit_;
   int64_t estimated_send_bitrate_sum_kbits_ GUARDED_BY(&bitrate_crit_);
   int64_t pacer_bitrate_sum_kbits_ GUARDED_BY(&bitrate_crit_);
   int64_t num_bitrate_updates_ GUARDED_BY(&bitrate_crit_);
 
+  std::map<std::string, rtc::NetworkRoute> network_routes_;
+
   VieRemb remb_;
   const std::unique_ptr<CongestionController> congestion_controller_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(Call);
 };
 }  // namespace internal
 
 Call* Call::Create(const Call::Config& config) {
   return new internal::Call(config);
 }
@@ skipping 401 matching lines @@
     ReadLockScoped read_lock(*receive_crit_);
     for (auto& kv : audio_receive_ssrcs_) {
       kv.second->SignalNetworkState(audio_network_state_);
     }
     for (auto& kv : video_receive_ssrcs_) {
       kv.second->SignalNetworkState(video_network_state_);
     }
   }
 }
 
+// TODO(honghaiz): Add tests for this method.
+void Call::OnNetworkRouteChanged(const std::string& transport_name,
+                                 const rtc::NetworkRoute& network_route) {
+  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  // Check if the network route is connected.
+  if (!network_route.connected) {
+    LOG(LS_INFO) << "Transport " << transport_name << " is disconnected";
+    // TODO(honghaiz): Perhaps handle this in SignalChannelNetworkState and
+    // consider merging these two methods.
+    return;
+  }
+
+  // Check whether the network route has changed on each transport.
+  auto result =
+      network_routes_.insert(std::make_pair(transport_name, network_route));
+  auto kv = result.first;
+  bool inserted = result.second;
+  if (inserted) {
+    // No need to reset BWE if this is the first time the network connects.
+    return;
+  }
+  if (kv->second != network_route) {
+    kv->second = network_route;
+    LOG(LS_INFO) << "Network route changed on transport " << transport_name
+                 << ": new local network id " << network_route.local_network_id
+                 << " new remote network id "
+                 << network_route.remote_network_id;
+    // TODO(holmer): Update the BWE bitrates.
+  }
+}
+
 void Call::UpdateAggregateNetworkState() {
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
 
   bool have_audio = false;
   bool have_video = false;
   {
     ReadLockScoped read_lock(*send_crit_);
     if (audio_send_ssrcs_.size() > 0)
       have_audio = true;
     if (video_send_ssrcs_.size() > 0)
@@ skipping 194 matching lines @@
   // thread. Then this check can be enabled.
   // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread());
   if (RtpHeaderParser::IsRtcp(packet, length))
     return DeliverRtcp(media_type, packet, length);
 
   return DeliverRtp(media_type, packet, length, packet_time);
 }
 
 }  // namespace internal
 }  // namespace webrtc