Adds a modified copy of talk/base to webrtc/base. It is the first step in migrating talk/base to webrtc/base.

BUG=N/A
R=andrew@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6107 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrike@webrtc.org
2014-05-12 18:03:09 +00:00
parent f3a5e6afc4
commit 2c7d1b39b9
386 changed files with 84973 additions and 6 deletions

View File

@ -0,0 +1,57 @@
/*
* 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_BASE_SOCKETSTREAM_H_
#define WEBRTC_BASE_SOCKETSTREAM_H_
#include "webrtc/base/asyncsocket.h"
#include "webrtc/base/common.h"
#include "webrtc/base/stream.h"
namespace rtc {
///////////////////////////////////////////////////////////////////////////////
class SocketStream : public StreamInterface, public sigslot::has_slots<> {
public:
explicit SocketStream(AsyncSocket* socket);
virtual ~SocketStream();
void Attach(AsyncSocket* socket);
AsyncSocket* Detach();
AsyncSocket* GetSocket() { return socket_; }
virtual StreamState GetState() const;
virtual StreamResult Read(void* buffer, size_t buffer_len,
size_t* read, int* error);
virtual StreamResult Write(const void* data, size_t data_len,
size_t* written, int* error);
virtual void Close();
private:
void OnConnectEvent(AsyncSocket* socket);
void OnReadEvent(AsyncSocket* socket);
void OnWriteEvent(AsyncSocket* socket);
void OnCloseEvent(AsyncSocket* socket, int err);
AsyncSocket* socket_;
DISALLOW_EVIL_CONSTRUCTORS(SocketStream);
};
///////////////////////////////////////////////////////////////////////////////
} // namespace rtc
#endif // WEBRTC_BASE_SOCKETSTREAM_H_