Start probes only after network is connected.

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 128 matching lines @@
         static_cast<internal::AudioState*>(config_.audio_state.get());
     if (audio_state)
       return audio_state->voice_engine();
     else
       return nullptr;
   }
 
   void UpdateSendHistograms() EXCLUSIVE_LOCKS_REQUIRED(&bitrate_crit_);
   void UpdateReceiveHistograms();
   void UpdateHistograms();
-  void UpdateAggregateNetworkState();
 
   Clock* const clock_;
 
   const int num_cpu_cores_;
   const std::unique_ptr<ProcessThread> module_process_thread_;
   const std::unique_ptr<ProcessThread> pacer_thread_;
   const std::unique_ptr<CallStats> call_stats_;
   const std::unique_ptr<BitrateAllocator> bitrate_allocator_;
   Call::Config config_;
   rtc::ThreadChecker configuration_thread_checker_;
@@ skipping 115 matching lines @@
   RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps,
                 config.bitrate_config.min_bitrate_bps);
   if (config.bitrate_config.max_bitrate_bps != -1) {
     RTC_DCHECK_GE(config.bitrate_config.max_bitrate_bps,
                   config.bitrate_config.start_bitrate_bps);
   }
   Trace::CreateTrace();
   call_stats_->RegisterStatsObserver(congestion_controller_.get());
 
   congestion_controller_->SetBweBitrates(
+      config_.bitrate_config.start_bitrate_bps,
       config_.bitrate_config.min_bitrate_bps,
-      config_.bitrate_config.start_bitrate_bps,
       config_.bitrate_config.max_bitrate_bps);
 
   module_process_thread_->Start();
   module_process_thread_->RegisterModule(call_stats_.get());
   module_process_thread_->RegisterModule(congestion_controller_.get());
   pacer_thread_->RegisterModule(congestion_controller_->pacer());
   pacer_thread_->RegisterModule(
       congestion_controller_->GetRemoteBitrateEstimator(true));
   pacer_thread_->Start();
 }
@@ skipping 101 matching lines @@
   AudioSendStream* send_stream = new AudioSendStream(
       config, config_.audio_state, &worker_queue_, congestion_controller_.get(),
       bitrate_allocator_.get(), event_log_);
   {
     WriteLockScoped write_lock(*send_crit_);
     RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) ==
                audio_send_ssrcs_.end());
     audio_send_ssrcs_[config.rtp.ssrc] = send_stream;
   }
   send_stream->SignalNetworkState(audio_network_state_);
-  UpdateAggregateNetworkState();
   return send_stream;
 }
 
 void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
   TRACE_EVENT0("webrtc", "Call::DestroyAudioSendStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_DCHECK(send_stream != nullptr);
 
   send_stream->Stop();
 
   webrtc::internal::AudioSendStream* audio_send_stream =
       static_cast<webrtc::internal::AudioSendStream*>(send_stream);
   {
     WriteLockScoped write_lock(*send_crit_);
     size_t num_deleted = audio_send_ssrcs_.erase(
         audio_send_stream->config().rtp.ssrc);
     RTC_DCHECK(num_deleted == 1);
   }
-  UpdateAggregateNetworkState();
   delete audio_send_stream;
 }
 
 webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
     const webrtc::AudioReceiveStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   event_log_->LogAudioReceiveStreamConfig(config);
   AudioReceiveStream* receive_stream = new AudioReceiveStream(
       congestion_controller_.get(), config, config_.audio_state, event_log_);
   {
     WriteLockScoped write_lock(*receive_crit_);
     RTC_DCHECK(audio_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
                audio_receive_ssrcs_.end());
     audio_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream;
     ConfigureSync(config.sync_group);
   }
   receive_stream->SignalNetworkState(audio_network_state_);
-  UpdateAggregateNetworkState();
   return receive_stream;
 }
 
 void Call::DestroyAudioReceiveStream(
     webrtc::AudioReceiveStream* receive_stream) {
   TRACE_EVENT0("webrtc", "Call::DestroyAudioReceiveStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_DCHECK(receive_stream != nullptr);
   webrtc::internal::AudioReceiveStream* audio_receive_stream =
       static_cast<webrtc::internal::AudioReceiveStream*>(receive_stream);
   {
     WriteLockScoped write_lock(*receive_crit_);
     size_t num_deleted = audio_receive_ssrcs_.erase(
         audio_receive_stream->config().rtp.remote_ssrc);
     RTC_DCHECK(num_deleted == 1);
     const std::string& sync_group = audio_receive_stream->config().sync_group;
     const auto it = sync_stream_mapping_.find(sync_group);
     if (it != sync_stream_mapping_.end() &&
         it->second == audio_receive_stream) {
       sync_stream_mapping_.erase(it);
       ConfigureSync(sync_group);
     }
   }
-  UpdateAggregateNetworkState();
   delete audio_receive_stream;
 }
 
 webrtc::VideoSendStream* Call::CreateVideoSendStream(
     webrtc::VideoSendStream::Config config,
     VideoEncoderConfig encoder_config) {
   TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
 
   video_send_delay_stats_->AddSsrcs(config);
@@ skipping 11 matching lines @@
 
   {
     WriteLockScoped write_lock(*send_crit_);
     for (uint32_t ssrc : ssrcs) {
       RTC_DCHECK(video_send_ssrcs_.find(ssrc) == video_send_ssrcs_.end());
       video_send_ssrcs_[ssrc] = send_stream;
     }
     video_send_streams_.insert(send_stream);
   }
   send_stream->SignalNetworkState(video_network_state_);
-  UpdateAggregateNetworkState();
 
   return send_stream;
 }
 
 void Call::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
   TRACE_EVENT0("webrtc", "Call::DestroyVideoSendStream");
   RTC_DCHECK(send_stream != nullptr);
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
 
   send_stream->Stop();
@@ skipping 15 matching lines @@
   RTC_CHECK(send_stream_impl != nullptr);
 
   VideoSendStream::RtpStateMap rtp_state =
       send_stream_impl->StopPermanentlyAndGetRtpStates();
 
   for (VideoSendStream::RtpStateMap::iterator it = rtp_state.begin();
        it != rtp_state.end(); ++it) {
     suspended_video_send_ssrcs_[it->first] = it->second;
   }
 
-  UpdateAggregateNetworkState();
   delete send_stream_impl;
 }
 
 webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
     webrtc::VideoReceiveStream::Config configuration) {
   TRACE_EVENT0("webrtc", "Call::CreateVideoReceiveStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   VideoReceiveStream* receive_stream = new VideoReceiveStream(
       num_cpu_cores_, congestion_controller_.get(), std::move(configuration),
       voice_engine(), module_process_thread_.get(), call_stats_.get(), &remb_);
 
   const webrtc::VideoReceiveStream::Config& config = receive_stream->config();
   {
     WriteLockScoped write_lock(*receive_crit_);
     RTC_DCHECK(video_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
                video_receive_ssrcs_.end());
     video_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream;
     // TODO(pbos): Configure different RTX payloads per receive payload.
     VideoReceiveStream::Config::Rtp::RtxMap::const_iterator it =
         config.rtp.rtx.begin();
     if (it != config.rtp.rtx.end())
       video_receive_ssrcs_[it->second.ssrc] = receive_stream;
     video_receive_streams_.insert(receive_stream);
     ConfigureSync(config.sync_group);
   }
   receive_stream->SignalNetworkState(video_network_state_);
-  UpdateAggregateNetworkState();
   event_log_->LogVideoReceiveStreamConfig(config);
   return receive_stream;
 }
 
 void Call::DestroyVideoReceiveStream(
     webrtc::VideoReceiveStream* receive_stream) {
   TRACE_EVENT0("webrtc", "Call::DestroyVideoReceiveStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_DCHECK(receive_stream != nullptr);
   VideoReceiveStream* receive_stream_impl = nullptr;
   {
     WriteLockScoped write_lock(*receive_crit_);
     // Remove all ssrcs pointing to a receive stream. As RTX retransmits on a
     // separate SSRC there can be either one or two.
     auto it = video_receive_ssrcs_.begin();
     while (it != video_receive_ssrcs_.end()) {
       if (it->second == static_cast<VideoReceiveStream*>(receive_stream)) {
         if (receive_stream_impl != nullptr)
           RTC_DCHECK(receive_stream_impl == it->second);
         receive_stream_impl = it->second;
         video_receive_ssrcs_.erase(it++);
       } else {
         ++it;
       }
     }
     video_receive_streams_.erase(receive_stream_impl);
     RTC_CHECK(receive_stream_impl != nullptr);
     ConfigureSync(receive_stream_impl->config().sync_group);
   }
-  UpdateAggregateNetworkState();
   delete receive_stream_impl;
 }
 
 webrtc::FlexfecReceiveStream* Call::CreateFlexfecReceiveStream(
     webrtc::FlexfecReceiveStream::Config configuration) {
   TRACE_EVENT0("webrtc", "Call::CreateFlexfecReceiveStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   FlexfecReceiveStream* receive_stream =
       new FlexfecReceiveStream(std::move(configuration), this);
 
@@ skipping 82 matching lines @@
           bitrate_config.max_bitrate_bps) {
     // Nothing new to set, early abort to avoid encoder reconfigurations.
     return;
   }
   config_.bitrate_config.min_bitrate_bps = bitrate_config.min_bitrate_bps;
   // Start bitrate of -1 means we should keep the old bitrate, which there is
   // no point in remembering for the future.
   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) {
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   switch (media) {
     case MediaType::AUDIO:
       audio_network_state_ = state;
       break;
     case MediaType::VIDEO:
       video_network_state_ = state;
       break;
     case MediaType::ANY:
     case MediaType::DATA:
       RTC_NOTREACHED();
       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_) {
       kv.second->SignalNetworkState(audio_network_state_);
     }
     for (auto& kv : video_send_ssrcs_) {
       kv.second->SignalNetworkState(video_network_state_);
     }
   }
   {
@@ skipping 13 matching lines @@
   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;
+  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);
+                 << " new remote network id "
+                 << network_route.remote_network_id;
+    congestion_controller_->OnNetworkRouteChanged(network_route);
   }
 }
 
-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;
-  }
-  {
-    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) {
   if (first_packet_sent_ms_ == -1)
     first_packet_sent_ms_ = clock_->TimeInMilliseconds();
   video_send_delay_stats_->OnSentPacket(sent_packet.packet_id,
                                         clock_->TimeInMilliseconds());
   congestion_controller_->OnSentPacket(sent_packet);
 }
 
 void Call::OnNetworkChanged(uint32_t target_bitrate_bps, uint8_t fraction_loss,
                             int64_t rtt_ms) {
@@ skipping 218 matching lines @@
   uint32_t ssrc = ByteReader<uint32_t>::ReadBigEndian(&packet[8]);
   ReadLockScoped read_lock(*receive_crit_);
   auto it = video_receive_ssrcs_.find(ssrc);
   if (it == video_receive_ssrcs_.end())
     return false;
   return it->second->OnRecoveredPacket(packet, length);
 }
 
 }  // namespace internal
 }  // namespace webrtc