1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64
|
/*
* 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(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(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_
|