/* * 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 RTC_BASE_SOCKET_STREAM_H_ #define RTC_BASE_SOCKET_STREAM_H_ #include #include "rtc_base/async_socket.h" #include "rtc_base/constructor_magic.h" #include "rtc_base/stream.h" #include "rtc_base/third_party/sigslot/sigslot.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 // RTC_BASE_SOCKET_STREAM_H_