| /* | 
 |  *  Copyright 2005 The WebRTC Project Authors. All rights reserved. | 
 |  * | 
 |  *  Use of this source code is governed by a BSD-style license | 
 |  *  that can be found in the LICENSE file in the root of the source | 
 |  *  tree. An additional intellectual property rights grant can be found | 
 |  *  in the file PATENTS.  All contributing project authors may | 
 |  *  be found in the AUTHORS file in the root of the source tree. | 
 |  */ | 
 |  | 
 | #ifndef WEBRTC_RTC_BASE_SOCKETSTREAM_H_ | 
 | #define WEBRTC_RTC_BASE_SOCKETSTREAM_H_ | 
 |  | 
 | #include "webrtc/rtc_base/asyncsocket.h" | 
 | #include "webrtc/rtc_base/constructormagic.h" | 
 | #include "webrtc/rtc_base/stream.h" | 
 |  | 
 | namespace rtc { | 
 |  | 
 | /////////////////////////////////////////////////////////////////////////////// | 
 |  | 
 | class SocketStream : public StreamInterface, public sigslot::has_slots<> { | 
 |  public: | 
 |   explicit SocketStream(AsyncSocket* socket); | 
 |   ~SocketStream() override; | 
 |  | 
 |   void Attach(AsyncSocket* socket); | 
 |   AsyncSocket* Detach(); | 
 |  | 
 |   AsyncSocket* GetSocket() { return socket_; } | 
 |  | 
 |   StreamState GetState() const override; | 
 |  | 
 |   StreamResult Read(void* buffer, | 
 |                     size_t buffer_len, | 
 |                     size_t* read, | 
 |                     int* error) override; | 
 |  | 
 |   StreamResult Write(const void* data, | 
 |                      size_t data_len, | 
 |                      size_t* written, | 
 |                      int* error) override; | 
 |  | 
 |   void Close() override; | 
 |  | 
 |  private: | 
 |   void OnConnectEvent(AsyncSocket* socket); | 
 |   void OnReadEvent(AsyncSocket* socket); | 
 |   void OnWriteEvent(AsyncSocket* socket); | 
 |   void OnCloseEvent(AsyncSocket* socket, int err); | 
 |  | 
 |   AsyncSocket* socket_; | 
 |  | 
 |   RTC_DISALLOW_COPY_AND_ASSIGN(SocketStream); | 
 | }; | 
 |  | 
 | /////////////////////////////////////////////////////////////////////////////// | 
 |  | 
 | }  // namespace rtc | 
 |  | 
 | #endif  // WEBRTC_RTC_BASE_SOCKETSTREAM_H_ |