Move audio decoding to protocol layer

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>
 
@@ skipping 19 matching lines @@
                                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 set_audio_stub(AudioStub* audio_stub) override;
+  void InitializeAudio(
+      scoped_refptr<base::SingleThreadTaskRunner> audio_decode_task_runner,
+      base::WeakPtr<AudioStub> audio_stub) 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 50 matching lines @@
   ErrorCode error_ = OK;
 
  private:
   DISALLOW_COPY_AND_ASSIGN(WebrtcConnectionToHost);
 };
 
 }  // namespace protocol
 }  // namespace remoting
 
 #endif  // REMOTING_PROTOCOL_WEBRTC_CONNECTION_TO_HOST_H_