Start probes only after network is connected.

webrtc/call/call.cc

Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index d7272170841c665883fff1e4ceae0adb1e12d66b..d6e57659e0481a0aa0b11d9c1f0d640693b6c2f1 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -146,7 +146,6 @@ class Call : public webrtc::Call,
   void UpdateSendHistograms() EXCLUSIVE_LOCKS_REQUIRED(&bitrate_crit_);
   void UpdateReceiveHistograms();
   void UpdateHistograms();
-  void UpdateAggregateNetworkState();
 
   Clock* const clock_;
 
@@ -282,8 +281,8 @@ Call::Call(const Call::Config& config)
   call_stats_->RegisterStatsObserver(congestion_controller_.get());
 
   congestion_controller_->SetBweBitrates(
-      config_.bitrate_config.min_bitrate_bps,
       config_.bitrate_config.start_bitrate_bps,
+      config_.bitrate_config.min_bitrate_bps,
       config_.bitrate_config.max_bitrate_bps);
 
   module_process_thread_->Start();
@@ -405,7 +404,6 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream(
     audio_send_ssrcs_[config.rtp.ssrc] = send_stream;
   }
   send_stream->SignalNetworkState(audio_network_state_);
-  UpdateAggregateNetworkState();
   return send_stream;
 }
 
@@ -424,7 +422,6 @@ void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
         audio_send_stream->config().rtp.ssrc);
     RTC_DCHECK(num_deleted == 1);
   }
-  UpdateAggregateNetworkState();
   delete audio_send_stream;
 }
 
@@ -443,7 +440,6 @@ webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
     ConfigureSync(config.sync_group);
   }
   receive_stream->SignalNetworkState(audio_network_state_);
-  UpdateAggregateNetworkState();
   return receive_stream;
 }
 
@@ -467,7 +463,6 @@ void Call::DestroyAudioReceiveStream(
       ConfigureSync(sync_group);
     }
   }
-  UpdateAggregateNetworkState();
   delete audio_receive_stream;
 }
 
@@ -499,7 +494,6 @@ webrtc::VideoSendStream* Call::CreateVideoSendStream(
     video_send_streams_.insert(send_stream);
   }
   send_stream->SignalNetworkState(video_network_state_);
-  UpdateAggregateNetworkState();
 
   return send_stream;
 }
@@ -535,7 +529,6 @@ void Call::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
     suspended_video_send_ssrcs_[it->first] = it->second;
   }
 
-  UpdateAggregateNetworkState();
   delete send_stream_impl;
 }
 
@@ -562,7 +555,6 @@ webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
     ConfigureSync(config.sync_group);
   }
   receive_stream->SignalNetworkState(video_network_state_);
-  UpdateAggregateNetworkState();
   event_log_->LogVideoReceiveStreamConfig(config);
   return receive_stream;
 }
@@ -592,7 +584,6 @@ void Call::DestroyVideoReceiveStream(
     RTC_CHECK(receive_stream_impl != nullptr);
     ConfigureSync(receive_stream_impl->config().sync_group);
   }
-  UpdateAggregateNetworkState();
   delete receive_stream_impl;
 }
 
@@ -695,9 +686,10 @@ void Call::SetBitrateConfig(
   if (bitrate_config.start_bitrate_bps > 0)
     config_.bitrate_config.start_bitrate_bps = bitrate_config.start_bitrate_bps;
   config_.bitrate_config.max_bitrate_bps = bitrate_config.max_bitrate_bps;
-  congestion_controller_->SetBweBitrates(bitrate_config.min_bitrate_bps,
-                                         bitrate_config.start_bitrate_bps,
-                                         bitrate_config.max_bitrate_bps);
+  congestion_controller_->SetBweBitrates(
+      config_.bitrate_config.start_bitrate_bps,
+      config_.bitrate_config.min_bitrate_bps,
+      config_.bitrate_config.max_bitrate_bps);
 }
 
 void Call::SignalChannelNetworkState(MediaType media, NetworkState state) {
@@ -715,7 +707,6 @@ void Call::SignalChannelNetworkState(MediaType media, NetworkState state) {
       break;
   }
 
-  UpdateAggregateNetworkState();

[honghaiz3, 2016/11/03 06:10:20]

I don't think it is appropriate to remove this here. 
It was called when sending failed here: 
https://cs.chromium.org/chromium/src/third_party/webrtc/pc/channel.cc?rcl=147...

If you remove this, when sending failed, it won't update the network state. 

   {
     ReadLockScoped read_lock(*send_crit_);
     for (auto& kv : audio_send_ssrcs_) {
@@ -749,61 +740,15 @@ void Call::OnNetworkRouteChanged(const std::string& transport_name,
   }
 
   // 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;
+  auto& old_route = network_routes_[transport_name];
+  if (old_route != network_route) {
+    old_route = 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
-                 << " Reset bitrates to min: "
-                 << config_.bitrate_config.min_bitrate_bps
-                 << " bps, start: " << config_.bitrate_config.start_bitrate_bps
-                 << " bps,  max: " << config_.bitrate_config.start_bitrate_bps
-                 << " bps.";
-    congestion_controller_->ResetBweAndBitrates(
-        config_.bitrate_config.start_bitrate_bps,
-        config_.bitrate_config.min_bitrate_bps,
-        config_.bitrate_config.max_bitrate_bps);
-  }
-}
-
-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)
-      have_video = true;
+                 << " new remote network id "
+                 << network_route.remote_network_id;
+    congestion_controller_->OnNetworkRouteChanged(network_route);
   }
-  {
-    ReadLockScoped read_lock(*receive_crit_);
-    if (audio_receive_ssrcs_.size() > 0)
-      have_audio = true;
-    if (video_receive_ssrcs_.size() > 0)
-      have_video = true;
-  }
-
-  NetworkState aggregate_state = kNetworkDown;
-  if ((have_video && video_network_state_ == kNetworkUp) ||
-      (have_audio && audio_network_state_ == kNetworkUp)) {
-    aggregate_state = kNetworkUp;
-  }
-
-  LOG(LS_INFO) << "UpdateAggregateNetworkState: aggregate_state="
-               << (aggregate_state == kNetworkUp ? "up" : "down");
-
-  congestion_controller_->SignalNetworkState(aggregate_state);
 }
 
 void Call::OnSentPacket(const rtc::SentPacket& sent_packet) {