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

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  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.
  */
 
@@ skipping 1332 matching lines @@
 
 WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
                                                  const MediaConfig& config,
                                                  const AudioOptions& options,
                                                  webrtc::Call* call)
     : VoiceMediaChannel(config), engine_(engine), call_(call) {
   LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel";
   RTC_DCHECK(call);
   engine->RegisterChannel(this);
   SetOptions(options);
+  call_->SignalChannelNetworkState(
+    webrtc::MediaType::AUDIO,
+    webrtc::ChannelNetworkState::CHANNEL_NETWORK_UP);
 }
 
 WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel() {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel";
   // TODO(solenberg): Should be able to delete the streams directly, without
   //                  going through RemoveNnStream(), once stream objects handle
   //                  all (de)configuration.
   while (!send_streams_.empty()) {
     RemoveSendStream(send_streams_.begin()->first);
   }
   while (!recv_streams_.empty()) {
     RemoveRecvStream(recv_streams_.begin()->first);
   }
   engine()->UnregisterChannel(this);
+  call_->SignalChannelNetworkState(
+    webrtc::MediaType::AUDIO,
+    webrtc::ChannelNetworkState::CHANNEL_NOT_PRESENT);
 }
 
 rtc::DiffServCodePoint WebRtcVoiceMediaChannel::PreferredDscp() const {
   return kAudioDscpValue;
 }
 
 bool WebRtcVoiceMediaChannel::SetSendParameters(
     const AudioSendParameters& params) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetSendParameters: "
@@ skipping 1014 matching lines @@
     if (bps < codec.rate) {
       LOG(LS_INFO) << "Failed to set codec " << codec.plname
                    << " to bitrate " << bps << " bps"
                    << ", requires at least " << codec.rate << " bps.";
       return false;
     }
     return true;
   }
 }
 
+void WebRtcVoiceMediaChannel::OnReadyToSend(bool ready)
+{
+  LOG(LS_VERBOSE) << "OnReadyToSend: " << (ready ? "Ready." : "Not ready.");
+  call_->SignalChannelNetworkState(
+    webrtc::MediaType::AUDIO,
+    ready ?
+    webrtc::ChannelNetworkState::CHANNEL_NETWORK_UP :
+    webrtc::ChannelNetworkState::CHANNEL_NETWORK_DOWN);
+}
+
 bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   RTC_DCHECK(info);
 
   // Get SSRC and stats for each sender.
   RTC_DCHECK(info->senders.size() == 0);
   for (const auto& stream : send_streams_) {
     webrtc::AudioSendStream::Stats stats = stream.second->GetStats();
     VoiceSenderInfo sinfo;
     sinfo.add_ssrc(stats.local_ssrc);
@@ skipping 108 matching lines @@
     }
   } else {
     LOG(LS_INFO) << "Stopping playout for channel #" << channel;
     engine()->voe()->base()->StopPlayout(channel);
   }
   return true;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE