Chromium Code Reviews| Index: remoting/protocol/webrtc_transport.cc |
| diff --git a/remoting/protocol/webrtc_transport.cc b/remoting/protocol/webrtc_transport.cc |
| index ac3b6b32398af1d83681118b97f228692b22d623..76b557c94150cd6421634059432348c025aaceee 100644 |
| --- a/remoting/protocol/webrtc_transport.cc |
| +++ b/remoting/protocol/webrtc_transport.cc |
| @@ -135,7 +135,6 @@ class SetSessionDescriptionObserver |
| DISALLOW_COPY_AND_ASSIGN(SetSessionDescriptionObserver); |
| }; |
| - |
| } // namespace |
| WebrtcTransport::WebrtcTransport( |
| @@ -150,7 +149,13 @@ WebrtcTransport::WebrtcTransport( |
| transport_context_->set_relay_mode(TransportContext::RelayMode::TURN); |
| } |
| -WebrtcTransport::~WebrtcTransport() {} |
| +WebrtcTransport::~WebrtcTransport() { |
| + // Destroy |fake_audio_device_module_| asynchronously. PeerConnection may be |
| + // on the stack and so it will be destroyed later. We need to ensures that |
|
Jamie
2016/07/23 00:03:48
s/ensures/ensure/
Sergey Ulanov
2016/07/23 06:30:04
reworded this comment
|
| + // |fake_audio_device_module_| is destroyed only after that. |
| + base::ThreadTaskRunnerHandle::Get()->DeleteSoon( |
| + FROM_HERE, fake_audio_device_module_.release()); |
| +} |
| std::unique_ptr<MessagePipe> WebrtcTransport::CreateOutgoingChannel( |
| const std::string& name) { |