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
73 lines
2.1 KiB
C++
73 lines
2.1 KiB
C++
/*
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* Copyright 2004 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_LIBJINGLE_XMPP_XMPPSOCKET_H_
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#define WEBRTC_LIBJINGLE_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 // WEBRTC_LIBJINGLE_XMPP_XMPPSOCKET_H_
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