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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/buffer.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::Buffer buffer_; | |
66 buzz::TlsOptions tls_; | |
67 }; | |
68 | |
69 } // namespace buzz | |
70 | |
71 #endif // WEBRTC_LIBJINGLE_XMPP_XMPPSOCKET_H_ | |
72 | |
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