Index: webrtc/media/engine/webrtcvoiceengine.cc |
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc |
index 2cd19e7435e884ceefe2c801cd8070b12977948c..928787a8c7b2e008a64afd2470722ebadcad8ce5 100644 |
--- a/webrtc/media/engine/webrtcvoiceengine.cc |
+++ b/webrtc/media/engine/webrtcvoiceengine.cc |
@@ -1350,6 +1350,9 @@ WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine, |
RTC_DCHECK(call); |
engine->RegisterChannel(this); |
SetOptions(options); |
+ call_->SignalChannelNetworkState( |
+ webrtc::MediaType::AUDIO, |
+ webrtc::ChannelNetworkState::CHANNEL_NETWORK_UP); |
} |
WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel() { |
@@ -1365,6 +1368,9 @@ WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel() { |
RemoveRecvStream(recv_streams_.begin()->first); |
} |
engine()->UnregisterChannel(this); |
+ call_->SignalChannelNetworkState( |
+ webrtc::MediaType::AUDIO, |
+ webrtc::ChannelNetworkState::CHANNEL_NOT_PRESENT); |
} |
rtc::DiffServCodePoint WebRtcVoiceMediaChannel::PreferredDscp() const { |
@@ -2399,6 +2405,16 @@ bool WebRtcVoiceMediaChannel::SetSendBitrateInternal(int bps) { |
} |
} |
+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); |