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