Index: remoting/protocol/webrtc_connection_to_client.cc |
diff --git a/remoting/protocol/webrtc_connection_to_client.cc b/remoting/protocol/webrtc_connection_to_client.cc |
index 92430d6d0d0df94171d73760c6e6ed3cd79ca07d..bb7fa3cb74deed6172060cbb355e135ea6f6d07c 100644 |
--- a/remoting/protocol/webrtc_connection_to_client.cc |
+++ b/remoting/protocol/webrtc_connection_to_client.cc |
@@ -12,7 +12,8 @@ |
#include "net/base/io_buffer.h" |
#include "remoting/codec/video_encoder.h" |
#include "remoting/codec/webrtc_video_encoder_vpx.h" |
-#include "remoting/protocol/audio_writer.h" |
+#include "remoting/protocol/audio_source.h" |
+#include "remoting/protocol/audio_stream.h" |
#include "remoting/protocol/clipboard_stub.h" |
#include "remoting/protocol/host_control_dispatcher.h" |
#include "remoting/protocol/host_event_dispatcher.h" |
@@ -89,8 +90,9 @@ std::unique_ptr<VideoStream> WebrtcConnectionToClient::StartVideoStream( |
return std::move(stream); |
} |
-AudioStub* WebrtcConnectionToClient::audio_stub() { |
- DCHECK(thread_checker_.CalledOnValidThread()); |
+std::unique_ptr<AudioStream> WebrtcConnectionToClient::StartAudioStream( |
+ std::unique_ptr<AudioSource> audio_source) { |
+ NOTIMPLEMENTED(); |
return nullptr; |
} |
@@ -139,7 +141,7 @@ void WebrtcConnectionToClient::OnSessionStateChange(Session::State state) { |
// OnConnectionAuthenticated() call above may result in the connection |
// being torn down. |
if (self) |
- event_handler_->CreateVideoStreams(this); |
+ event_handler_->CreateMediaStreams(this); |
break; |
} |