269fb4bc90
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p. Also makes libjingle use those version instead of the one in the talk folder. BUG=3379 Review URL: https://webrtc-codereview.appspot.com/26999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7549 4adac7df-926f-26a2-2b94-8c16560cd09d
90 lines
3.1 KiB
C++
90 lines
3.1 KiB
C++
/*
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* libjingle
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* Copyright 2004--2005, Google Inc.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#ifndef TALK_XMPP_XMPPSOCKET_H_
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#define TALK_XMPP_XMPPSOCKET_H_
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#include "webrtc/libjingle/xmpp/asyncsocket.h"
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#include "webrtc/libjingle/xmpp/xmppengine.h"
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#include "webrtc/base/asyncsocket.h"
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#include "webrtc/base/bytebuffer.h"
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#include "webrtc/base/sigslot.h"
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// The below define selects the SSLStreamAdapter implementation for
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// SSL, as opposed to the SSLAdapter socket adapter.
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// #define USE_SSLSTREAM
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namespace rtc {
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class StreamInterface;
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class SocketAddress;
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};
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extern rtc::AsyncSocket* cricket_socket_;
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namespace buzz {
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class XmppSocket : public buzz::AsyncSocket, public sigslot::has_slots<> {
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public:
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XmppSocket(buzz::TlsOptions tls);
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~XmppSocket();
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virtual buzz::AsyncSocket::State state();
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virtual buzz::AsyncSocket::Error error();
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virtual int GetError();
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virtual bool Connect(const rtc::SocketAddress& addr);
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virtual bool Read(char * data, size_t len, size_t* len_read);
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virtual bool Write(const char * data, size_t len);
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virtual bool Close();
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virtual bool StartTls(const std::string & domainname);
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sigslot::signal1<int> SignalCloseEvent;
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private:
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void CreateCricketSocket(int family);
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#ifndef USE_SSLSTREAM
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void OnReadEvent(rtc::AsyncSocket * socket);
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void OnWriteEvent(rtc::AsyncSocket * socket);
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void OnConnectEvent(rtc::AsyncSocket * socket);
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void OnCloseEvent(rtc::AsyncSocket * socket, int error);
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#else // USE_SSLSTREAM
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void OnEvent(rtc::StreamInterface* stream, int events, int err);
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#endif // USE_SSLSTREAM
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rtc::AsyncSocket * cricket_socket_;
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#ifdef USE_SSLSTREAM
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rtc::StreamInterface *stream_;
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#endif // USE_SSLSTREAM
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buzz::AsyncSocket::State state_;
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rtc::ByteBuffer buffer_;
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buzz::TlsOptions tls_;
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};
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} // namespace buzz
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#endif // TALK_XMPP_XMPPSOCKET_H_
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