Add audio support in WebrtcConnectionToHost, audio unittest

remoting/protocol/webrtc_connection_to_host.h

 // 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.
 
 #ifndef REMOTING_PROTOCOL_WEBRTC_CONNECTION_TO_HOST_H_
 #define REMOTING_PROTOCOL_WEBRTC_CONNECTION_TO_HOST_H_
 
 #include <memory>
 #include <string>
 
 #include "base/macros.h"
 #include "remoting/protocol/channel_dispatcher_base.h"
 #include "remoting/protocol/clipboard_filter.h"
 #include "remoting/protocol/connection_to_host.h"
 #include "remoting/protocol/errors.h"
 #include "remoting/protocol/input_filter.h"
 #include "remoting/protocol/session.h"
 #include "remoting/protocol/webrtc_transport.h"
 
 namespace remoting {
 namespace protocol {
 
 class ClientControlDispatcher;
 class ClientEventDispatcher;
 class SessionConfig;
 class WebrtcVideoRendererAdapter;
+class WebrtcAudioSinkAdapter;
 
 class WebrtcConnectionToHost : public ConnectionToHost,
                                public Session::EventHandler,
                                public WebrtcTransport::EventHandler,
                                public ChannelDispatcherBase::EventHandler {
  public:
   WebrtcConnectionToHost();
   ~WebrtcConnectionToHost() override;
 
   // ConnectionToHost interface.
   void set_client_stub(ClientStub* client_stub) override;
   void set_clipboard_stub(ClipboardStub* clipboard_stub) override;
   void set_video_renderer(VideoRenderer* video_renderer) override;
   void InitializeAudio(
       scoped_refptr<base::SingleThreadTaskRunner> audio_decode_task_runner,
-      base::WeakPtr<AudioStub> audio_stub) override;
+      base::WeakPtr<AudioStub> audio_consumer) override;
   void Connect(std::unique_ptr<Session> session,
                scoped_refptr<TransportContext> transport_context,
                HostEventCallback* event_callback) override;
   const SessionConfig& config() override;
   ClipboardStub* clipboard_forwarder() override;
   HostStub* host_stub() override;
   InputStub* input_stub() override;
   State state() const override;
 
  private:
@@ skipping 23 matching lines @@
   void CloseChannels();
 
   void OnFrameRendered(uint32_t frame_id,
                        base::TimeTicks event_timestamp,
                        base::TimeTicks frame_rendered_time);
 
   void SetState(State state, ErrorCode error);
 
   HostEventCallback* event_callback_ = nullptr;
 
+  scoped_refptr<base::SingleThreadTaskRunner> audio_decode_task_runner_;
+
   // Stub for incoming messages.
   ClientStub* client_stub_ = nullptr;
   VideoRenderer* video_renderer_ = nullptr;
+  base::WeakPtr<AudioStub> audio_consumer_;
   ClipboardStub* clipboard_stub_ = nullptr;
 
   std::unique_ptr<Session> session_;
   std::unique_ptr<WebrtcTransport> transport_;
 
   std::unique_ptr<ClientControlDispatcher> control_dispatcher_;
   std::unique_ptr<ClientEventDispatcher> event_dispatcher_;
   ClipboardFilter clipboard_forwarder_;
   InputFilter event_forwarder_;
 
   std::unique_ptr<WebrtcVideoRendererAdapter> video_adapter_;
+  std::unique_ptr<WebrtcAudioSinkAdapter> audio_adapter_;
 
   // Internal state of the connection.
   State state_ = INITIALIZING;
   ErrorCode error_ = OK;
 
- private:
   DISALLOW_COPY_AND_ASSIGN(WebrtcConnectionToHost);
 };
 
 }  // namespace protocol
 }  // namespace remoting
 
 #endif  // REMOTING_PROTOCOL_WEBRTC_CONNECTION_TO_HOST_H_