Make the audio channel communicate network state changes to the call.

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 68 matching lines @@
 
   Stats GetStats() const override;
 
   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 SignalNetworkState(NetworkState state) override;
+
+  void SignalChannelNetworkState(MediaType media,
+      ChannelNetworkState state) 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);
@@ skipping 20 matching lines @@
   Clock* const clock_;
 
   const int num_cpu_cores_;
   const rtc::scoped_ptr<ProcessThread> module_process_thread_;
   const rtc::scoped_ptr<ProcessThread> pacer_thread_;
   const rtc::scoped_ptr<CallStats> call_stats_;
   const rtc::scoped_ptr<BitrateAllocator> bitrate_allocator_;
   Call::Config config_;
   rtc::ThreadChecker configuration_thread_checker_;
 
-  bool network_enabled_;
+  ChannelNetworkState audio_network_state_;
+  ChannelNetworkState video_network_state_;
 
   rtc::scoped_ptr<RWLockWrapper> receive_crit_;
   // Audio and Video receive streams are owned by the client that creates them.
   std::map<uint32_t, AudioReceiveStream*> audio_receive_ssrcs_
       GUARDED_BY(receive_crit_);
   std::map<uint32_t, VideoReceiveStream*> video_receive_ssrcs_
       GUARDED_BY(receive_crit_);
   std::set<VideoReceiveStream*> video_receive_streams_
       GUARDED_BY(receive_crit_);
   std::map<std::string, AudioReceiveStream*> sync_stream_mapping_
@@ skipping 30 matching lines @@
   const rtc::scoped_ptr<CongestionController> congestion_controller_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(Call);
 };
 }  // namespace internal
 
 Call* Call::Create(const Call::Config& config) {
   return new internal::Call(config);
 }
 
+NetworkState ChannelStateToNetworkState(ChannelNetworkState channelState) {
+  return (channelState == ChannelNetworkState::CHANNEL_NETWORK_DOWN ?
+      kNetworkDown : kNetworkUp);
+}
+
+NetworkState AggregateNetworkState(ChannelNetworkState channelA,
+                                   ChannelNetworkState channelB) {
+  NetworkState aggregateState = kNetworkDown;
+  if (channelA == ChannelNetworkState::CHANNEL_NOT_PRESENT &&
+    channelB == ChannelNetworkState::CHANNEL_NOT_PRESENT) {
+    // When no channels are configured - such as in tests -
+    //  the network is assumed to be up
+    aggregateState = kNetworkUp;
+  } else if (channelA == ChannelNetworkState::CHANNEL_NETWORK_UP ||
+    channelB == ChannelNetworkState::CHANNEL_NETWORK_UP) {
+    // There is at least one channel that reports the network to be up
+    aggregateState = kNetworkUp;
+  } else {
+    // All the channels report that the network is down
+    aggregateState = kNetworkDown;
+  }
+  return aggregateState;
+}
+
 namespace internal {
 
 Call::Call(const Call::Config& config)
     : clock_(Clock::GetRealTimeClock()),
       num_cpu_cores_(CpuInfo::DetectNumberOfCores()),
       module_process_thread_(ProcessThread::Create("ModuleProcessThread")),
       pacer_thread_(ProcessThread::Create("PacerThread")),
       call_stats_(new CallStats(clock_)),
       bitrate_allocator_(new BitrateAllocator()),
       config_(config),
-      network_enabled_(true),
+      audio_network_state_(ChannelNetworkState::CHANNEL_NOT_PRESENT),
+      video_network_state_(ChannelNetworkState::CHANNEL_NOT_PRESENT),
       receive_crit_(RWLockWrapper::CreateRWLock()),
       send_crit_(RWLockWrapper::CreateRWLock()),
       received_video_bytes_(0),
       received_audio_bytes_(0),
       received_rtcp_bytes_(0),
       first_rtp_packet_received_ms_(-1),
       last_rtp_packet_received_ms_(-1),
       first_packet_sent_ms_(-1),
       estimated_send_bitrate_sum_kbits_(0),
       pacer_bitrate_sum_kbits_(0),
@@ skipping 107 matching lines @@
   // RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   return this;
 }
 
 webrtc::AudioSendStream* Call::CreateAudioSendStream(
     const webrtc::AudioSendStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   AudioSendStream* send_stream = new AudioSendStream(
       config, config_.audio_state, congestion_controller_.get());
-  if (!network_enabled_)
-    send_stream->SignalNetworkState(kNetworkDown);
+
+  send_stream->SignalNetworkState(
+      ChannelStateToNetworkState(audio_network_state_));
   {
     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;
   }
   return send_stream;
 }
 
 void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
@@ skipping 59 matching lines @@
   TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
 
   // TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if
   // the call has already started.
   VideoSendStream* send_stream = new VideoSendStream(
       num_cpu_cores_, module_process_thread_.get(), call_stats_.get(),
       congestion_controller_.get(), &remb_, bitrate_allocator_.get(), config,
       encoder_config, suspended_video_send_ssrcs_);
 
-  if (!network_enabled_)
-    send_stream->SignalNetworkState(kNetworkDown);
+  send_stream->SignalNetworkState(
+      ChannelStateToNetworkState(video_network_state_));
 
   WriteLockScoped write_lock(*send_crit_);
   for (uint32_t ssrc : config.rtp.ssrcs) {
     RTC_DCHECK(video_send_ssrcs_.find(ssrc) == video_send_ssrcs_.end());
     video_send_ssrcs_[ssrc] = send_stream;
   }
   video_send_streams_.insert(send_stream);
 
   if (event_log_)
     event_log_->LogVideoSendStreamConfig(config);
@@ skipping 49 matching lines @@
   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);
 
-  if (!network_enabled_)
-    receive_stream->SignalNetworkState(kNetworkDown);
+  receive_stream->SignalNetworkState(

[the sun, 2016/03/04 12:40:48]

For the send streams, the network state is signaled *before* the streams are
added to the internal data structures, for the video receive stream, the state
is signaled *after*. Unless there is some subtle reason for that I suggest we
make it consistent.

Also, should the network state should be signaled for AudioReceiveStream?

+      ChannelStateToNetworkState(video_network_state_));
 
   if (event_log_)
     event_log_->LogVideoReceiveStreamConfig(config);
 
   return receive_stream;
 }
 
 void Call::DestroyVideoReceiveStream(
     webrtc::VideoReceiveStream* receive_stream) {
   TRACE_EVENT0("webrtc", "Call::DestroyVideoReceiveStream");
@@ skipping 58 matching lines @@
           bitrate_config.max_bitrate_bps) {
     // Nothing new to set, early abort to avoid encoder reconfigurations.
     return;
   }
   config_.bitrate_config = bitrate_config;
   congestion_controller_->SetBweBitrates(bitrate_config.min_bitrate_bps,
                                          bitrate_config.start_bitrate_bps,
                                          bitrate_config.max_bitrate_bps);
 }
 
-void Call::SignalNetworkState(NetworkState state) {
+void Call::SignalChannelNetworkState(MediaType media,
+                                     ChannelNetworkState state) {
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
-  network_enabled_ = state == kNetworkUp;
-  congestion_controller_->SignalNetworkState(state);
+  if (media == MediaType::AUDIO)
+    audio_network_state_ = state;
+  if (media == MediaType::VIDEO)
+    video_network_state_ = state;
+
+  NetworkState aggregateState = AggregateNetworkState(audio_network_state_,
+                                                      video_network_state_);
+  LOG(LS_INFO) << "SignalChannelNetworkState: aggregateState=" <<
+      (aggregateState == kNetworkUp ? "up" : "down");
+  congestion_controller_->SignalNetworkState(aggregateState);
   {
     ReadLockScoped write_lock(*send_crit_);
     for (auto& kv : audio_send_ssrcs_) {
-      kv.second->SignalNetworkState(state);
+      kv.second->SignalNetworkState(
+          ChannelStateToNetworkState(audio_network_state_));
     }
     for (auto& kv : video_send_ssrcs_) {
-      kv.second->SignalNetworkState(state);
+      kv.second->SignalNetworkState(
+          ChannelStateToNetworkState(video_network_state_));
     }
   }
   {
     ReadLockScoped write_lock(*receive_crit_);
     for (auto& kv : video_receive_ssrcs_) {
-      kv.second->SignalNetworkState(state);
+      kv.second->SignalNetworkState(
+          ChannelStateToNetworkState(video_network_state_));
     }
   }
 }
 
 void Call::OnSentPacket(const rtc::SentPacket& sent_packet) {
   if (first_packet_sent_ms_ == -1)
     first_packet_sent_ms_ = clock_->TimeInMilliseconds();
   congestion_controller_->OnSentPacket(sent_packet);
 }
 
@@ skipping 166 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