blob: f9875fe41e171d7de429cd384ac2232efcbe7ddd [file] [log] [blame]
/*
* 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 <stddef.h>
#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_