1 /* 2 * Copyright 2004 The WebRTC Project Authors. All rights reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef WEBRTC_LIBJINGLE_XMPP_XMPPSOCKET_H_ 12 #define WEBRTC_LIBJINGLE_XMPP_XMPPSOCKET_H_ 13 14 #include "webrtc/libjingle/xmpp/asyncsocket.h" 15 #include "webrtc/libjingle/xmpp/xmppengine.h" 16 #include "webrtc/base/asyncsocket.h" 17 #include "webrtc/base/bytebuffer.h" 18 #include "webrtc/base/sigslot.h" 19 20 // The below define selects the SSLStreamAdapter implementation for 21 // SSL, as opposed to the SSLAdapter socket adapter. 22 // #define USE_SSLSTREAM 23 24 namespace rtc { 25 class StreamInterface; 26 class SocketAddress; 27 }; 28 extern rtc::AsyncSocket* cricket_socket_; 29 30 namespace buzz { 31 32 class XmppSocket : public buzz::AsyncSocket, public sigslot::has_slots<> { 33 public: 34 XmppSocket(buzz::TlsOptions tls); 35 ~XmppSocket(); 36 37 virtual buzz::AsyncSocket::State state(); 38 virtual buzz::AsyncSocket::Error error(); 39 virtual int GetError(); 40 41 virtual bool Connect(const rtc::SocketAddress& addr); 42 virtual bool Read(char * data, size_t len, size_t* len_read); 43 virtual bool Write(const char * data, size_t len); 44 virtual bool Close(); 45 virtual bool StartTls(const std::string & domainname); 46 47 sigslot::signal1<int> SignalCloseEvent; 48 49 private: 50 void CreateCricketSocket(int family); 51 #ifndef USE_SSLSTREAM 52 void OnReadEvent(rtc::AsyncSocket * socket); 53 void OnWriteEvent(rtc::AsyncSocket * socket); 54 void OnConnectEvent(rtc::AsyncSocket * socket); 55 void OnCloseEvent(rtc::AsyncSocket * socket, int error); 56 #else // USE_SSLSTREAM 57 void OnEvent(rtc::StreamInterface* stream, int events, int err); 58 #endif // USE_SSLSTREAM 59 60 rtc::AsyncSocket * cricket_socket_; 61 #ifdef USE_SSLSTREAM 62 rtc::StreamInterface *stream_; 63 #endif // USE_SSLSTREAM 64 buzz::AsyncSocket::State state_; 65 rtc::ByteBuffer buffer_; 66 buzz::TlsOptions tls_; 67 }; 68 69 } // namespace buzz 70 71 #endif // WEBRTC_LIBJINGLE_XMPP_XMPPSOCKET_H_ 72 73
