webrtc_m130/webrtc/rtc_base/socketstream.h
Henrik Kjellander c03627683f Reland "Move webrtc/{base => rtc_base}" (https://codereview.webrtc.org/2877023002)
Reland the base->rtc_base without adding stub headers (will be
done in follow-up CL). This preserves git blame history of all files.

BUG=webrtc:7634
NOTRY=True
TBR=kwiberg@webrtc.org

Change-Id: Iea3bb6f3f67b8374c96337b63e8f5aa3e6181012
Reviewed-on: https://chromium-review.googlesource.com/554611
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18821}
2017-06-29 06:04:25 +00:00

62 lines
1.7 KiB
C++

/*
* 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/base/asyncsocket.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/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_