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
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webrtc/base/socketstream.h
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webrtc/base/socketstream.h
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/*
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* Copyright 2005 The WebRTC Project Authors. All rights reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_BASE_SOCKETSTREAM_H_
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#define WEBRTC_BASE_SOCKETSTREAM_H_
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#include "webrtc/base/asyncsocket.h"
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#include "webrtc/base/common.h"
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#include "webrtc/base/stream.h"
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namespace rtc {
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///////////////////////////////////////////////////////////////////////////////
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class SocketStream : public StreamInterface, public sigslot::has_slots<> {
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public:
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explicit SocketStream(AsyncSocket* socket);
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virtual ~SocketStream();
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void Attach(AsyncSocket* socket);
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AsyncSocket* Detach();
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AsyncSocket* GetSocket() { return socket_; }
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virtual StreamState GetState() const;
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virtual StreamResult Read(void* buffer, size_t buffer_len,
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size_t* read, int* error);
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virtual StreamResult Write(const void* data, size_t data_len,
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size_t* written, int* error);
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virtual void Close();
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private:
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void OnConnectEvent(AsyncSocket* socket);
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void OnReadEvent(AsyncSocket* socket);
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void OnWriteEvent(AsyncSocket* socket);
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void OnCloseEvent(AsyncSocket* socket, int err);
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AsyncSocket* socket_;
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DISALLOW_EVIL_CONSTRUCTORS(SocketStream);
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};
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///////////////////////////////////////////////////////////////////////////////
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} // namespace rtc
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#endif // WEBRTC_BASE_SOCKETSTREAM_H_
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