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/*
* 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/socket.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(Socket* socket);
~SocketStream() override;
SocketStream(const SocketStream&) = delete;
SocketStream& operator=(const SocketStream&) = delete;
void Attach(Socket* socket);
Socket* Detach();
Socket* GetSocket() { return socket_; }
StreamState GetState() const override;
StreamResult Read(rtc::ArrayView<uint8_t> buffer,
size_t& read,
int& error) override;
StreamResult Write(rtc::ArrayView<const uint8_t> data,
size_t& written,
int& error) override;
void Close() override;
private:
void OnConnectEvent(Socket* socket);
void OnReadEvent(Socket* socket);
void OnWriteEvent(Socket* socket);
void OnCloseEvent(Socket* socket, int err);
Socket* socket_;
};
///////////////////////////////////////////////////////////////////////////////
} // namespace rtc
#endif // RTC_BASE_SOCKET_STREAM_H_
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