Add WebrtcAudioModule

remoting/protocol/webrtc_transport.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_TRANSPORT_H_
 #define REMOTING_PROTOCOL_WEBRTC_TRANSPORT_H_
 
 #include <memory>
 #include <string>
 
 #include "base/macros.h"
 #include "base/memory/ref_counted.h"
 #include "base/memory/scoped_vector.h"
 #include "base/memory/weak_ptr.h"
 #include "base/threading/thread_checker.h"
 #include "base/timer/timer.h"
 #include "crypto/hmac.h"
 #include "remoting/protocol/transport.h"
 #include "remoting/protocol/webrtc_data_stream_adapter.h"
 #include "remoting/protocol/webrtc_dummy_video_encoder.h"
 #include "remoting/signaling/signal_strategy.h"
 #include "third_party/webrtc/api/peerconnectioninterface.h"
 
 namespace remoting {
 namespace protocol {
 
 class TransportContext;
 class MessagePipe;
+class WebrtcAudioModule;
 
 class WebrtcTransport : public Transport {
  public:
   class EventHandler {
    public:
     // Called after |peer_connection| has been created but before handshake. The
     // handler should create data channels and media streams. Renegotiation will
     // be required in two cases after this method returns:
     //   1. When the first data channel is created, if it wasn't created by this
     //      event handler.
@@ skipping 105 matching lines @@
 
   base::WeakPtrFactory<WebrtcTransport> weak_factory_;
 
   DISALLOW_COPY_AND_ASSIGN(WebrtcTransport);
 };
 
 }  // namespace protocol
 }  // namespace remoting
 
 #endif  // REMOTING_PROTOCOL_WEBRTC_TRANSPORT_H_