webrtc/talk/xmpp/xmppsocket.h

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/*
* libjingle
* Copyright 2004--2005, Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef TALK_XMPP_XMPPSOCKET_H_
#define TALK_XMPP_XMPPSOCKET_H_
#include "talk/xmpp/asyncsocket.h"
#include "talk/xmpp/xmppengine.h"
#include "webrtc/base/asyncsocket.h"
#include "webrtc/base/bytebuffer.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::ByteBuffer buffer_;
buzz::TlsOptions tls_;
};
} // namespace buzz
#endif // TALK_XMPP_XMPPSOCKET_H_