Remove dependency on AudioStub in ConnectionToClient

remoting/protocol/webrtc_connection_to_client.cc

 // Copyright 2015 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "remoting/protocol/webrtc_connection_to_client.h"
 
 #include <utility>
 
 #include "base/bind.h"
 #include "base/location.h"
 #include "jingle/glue/thread_wrapper.h"
 #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"
 #include "remoting/protocol/host_stub.h"
 #include "remoting/protocol/input_stub.h"
 #include "remoting/protocol/message_pipe.h"
 #include "remoting/protocol/transport_context.h"
 #include "remoting/protocol/webrtc_transport.h"
 #include "remoting/protocol/webrtc_video_stream.h"
 #include "third_party/webrtc/api/mediastreaminterface.h"
@@ skipping 56 matching lines @@
   std::unique_ptr<VideoEncoder> video_encoder =
       WebrtcVideoEncoderVpx::CreateForVP8();
   std::unique_ptr<WebrtcVideoStream> stream(new WebrtcVideoStream());
   if (!stream->Start(std::move(desktop_capturer), transport_.get(),
                      video_encode_task_runner_, std::move(video_encoder))) {
     return nullptr;
   }
   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;
 }
 
 // Return pointer to ClientStub.
 ClientStub* WebrtcConnectionToClient::client_stub() {
   DCHECK(thread_checker_.CalledOnValidThread());
   return control_dispatcher_.get();
 }
 
 void WebrtcConnectionToClient::set_clipboard_stub(
@@ skipping 28 matching lines @@
       event_handler_->OnConnectionAuthenticating(this);
       break;
 
     case Session::AUTHENTICATED: {
       base::WeakPtr<WebrtcConnectionToClient> self = weak_factory_.GetWeakPtr();
       event_handler_->OnConnectionAuthenticated(this);
 
       // OnConnectionAuthenticated() call above may result in the connection
       // being torn down.
       if (self)
-        event_handler_->CreateVideoStreams(this);
+        event_handler_->CreateMediaStreams(this);
       break;
     }
 
     case Session::CLOSED:
     case Session::FAILED:
       control_dispatcher_.reset();
       event_dispatcher_.reset();
       transport_->Close(state == Session::CLOSED ? OK : session_->error());
       event_handler_->OnConnectionClosed(
           this, state == Session::CLOSED ? OK : session_->error());
@@ skipping 51 matching lines @@
     ChannelDispatcherBase* channel_dispatcher) {
   DCHECK(thread_checker_.CalledOnValidThread());
 
   LOG(ERROR) << "Channel " << channel_dispatcher->channel_name()
              << " was closed unexpectedly.";
   Disconnect(INCOMPATIBLE_PROTOCOL);
 }
 
 }  // namespace protocol
 }  // namespace remoting