| 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;
|
| }
|
|
|
|
|