rhubarb-lip-sync/rhubarb/lib/webrtc-8d2248ff/webrtc/libjingle/xmpp/xmppsocket.h

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2016-06-21 20:13:05 +00:00
/*
* Copyright 2004 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_LIBJINGLE_XMPP_XMPPSOCKET_H_
#define WEBRTC_LIBJINGLE_XMPP_XMPPSOCKET_H_
#include "webrtc/libjingle/xmpp/asyncsocket.h"
#include "webrtc/libjingle/xmpp/xmppengine.h"
#include "webrtc/base/asyncsocket.h"
#include "webrtc/base/buffer.h"
#include "webrtc/base/sigslot.h"
// The below define selects the SSLStreamAdapter implementation for
// SSL, as opposed to the SSLAdapter socket adapter.
// #define USE_SSLSTREAM
namespace rtc {
class StreamInterface;
class SocketAddress;
};
extern rtc::AsyncSocket* cricket_socket_;
namespace buzz {
class XmppSocket : public buzz::AsyncSocket, public sigslot::has_slots<> {
public:
XmppSocket(buzz::TlsOptions tls);
~XmppSocket();
virtual buzz::AsyncSocket::State state();
virtual buzz::AsyncSocket::Error error();
virtual int GetError();
virtual bool Connect(const rtc::SocketAddress& addr);
virtual bool Read(char * data, size_t len, size_t* len_read);
virtual bool Write(const char * data, size_t len);
virtual bool Close();
virtual bool StartTls(const std::string & domainname);
sigslot::signal1<int> SignalCloseEvent;
private:
void CreateCricketSocket(int family);
#ifndef USE_SSLSTREAM
void OnReadEvent(rtc::AsyncSocket * socket);
void OnWriteEvent(rtc::AsyncSocket * socket);
void OnConnectEvent(rtc::AsyncSocket * socket);
void OnCloseEvent(rtc::AsyncSocket * socket, int error);
#else // USE_SSLSTREAM
void OnEvent(rtc::StreamInterface* stream, int events, int err);
#endif // USE_SSLSTREAM
rtc::AsyncSocket * cricket_socket_;
#ifdef USE_SSLSTREAM
rtc::StreamInterface *stream_;
#endif // USE_SSLSTREAM
buzz::AsyncSocket::State state_;
rtc::Buffer buffer_;
buzz::TlsOptions tls_;
};
} // namespace buzz
#endif // WEBRTC_LIBJINGLE_XMPP_XMPPSOCKET_H_