| Index: webrtc/libjingle/xmpp/xmppsocket.h
|
| diff --git a/webrtc/libjingle/xmpp/xmppsocket.h b/webrtc/libjingle/xmpp/xmppsocket.h
|
| new file mode 100644
|
| index 0000000000000000000000000000000000000000..02d645383c3b9774a291d0b43f0bdb70d485ac71
|
| --- /dev/null
|
| +++ b/webrtc/libjingle/xmpp/xmppsocket.h
|
| @@ -0,0 +1,72 @@
|
| +/*
|
| + * Copyright 2004 The WebRTC Project Authors. All rights reserved.
|
| + *
|
| + * Use of this source code is governed by a BSD-style license
|
| + * that can be found in the LICENSE file in the root of the source
|
| + * tree. An additional intellectual property rights grant can be found
|
| + * in the file PATENTS. All contributing project authors may
|
| + * be found in the AUTHORS file in the root of the source tree.
|
| + */
|
| +
|
| +#ifndef WEBRTC_LIBJINGLE_XMPP_XMPPSOCKET_H_
|
| +#define WEBRTC_LIBJINGLE_XMPP_XMPPSOCKET_H_
|
| +
|
| +#include "webrtc/libjingle/xmpp/asyncsocket.h"
|
| +#include "webrtc/libjingle/xmpp/xmppengine.h"
|
| +#include "webrtc/base/asyncsocket.h"
|
| +#include "webrtc/base/buffer.h"
|
| +#include "webrtc/base/sigslot.h"
|
| +
|
| +// The below define selects the SSLStreamAdapter implementation for
|
| +// SSL, as opposed to the SSLAdapter socket adapter.
|
| +// #define USE_SSLSTREAM
|
| +
|
| +namespace rtc {
|
| + class StreamInterface;
|
| + class SocketAddress;
|
| +};
|
| +extern rtc::AsyncSocket* cricket_socket_;
|
| +
|
| +namespace buzz {
|
| +
|
| +class XmppSocket : public buzz::AsyncSocket, public sigslot::has_slots<> {
|
| +public:
|
| + XmppSocket(buzz::TlsOptions tls);
|
| + ~XmppSocket();
|
| +
|
| + virtual buzz::AsyncSocket::State state();
|
| + virtual buzz::AsyncSocket::Error error();
|
| + virtual int GetError();
|
| +
|
| + virtual bool Connect(const rtc::SocketAddress& addr);
|
| + virtual bool Read(char * data, size_t len, size_t* len_read);
|
| + virtual bool Write(const char * data, size_t len);
|
| + virtual bool Close();
|
| + virtual bool StartTls(const std::string & domainname);
|
| +
|
| + sigslot::signal1<int> SignalCloseEvent;
|
| +
|
| +private:
|
| + void CreateCricketSocket(int family);
|
| +#ifndef USE_SSLSTREAM
|
| + void OnReadEvent(rtc::AsyncSocket * socket);
|
| + void OnWriteEvent(rtc::AsyncSocket * socket);
|
| + void OnConnectEvent(rtc::AsyncSocket * socket);
|
| + void OnCloseEvent(rtc::AsyncSocket * socket, int error);
|
| +#else // USE_SSLSTREAM
|
| + void OnEvent(rtc::StreamInterface* stream, int events, int err);
|
| +#endif // USE_SSLSTREAM
|
| +
|
| + rtc::AsyncSocket * cricket_socket_;
|
| +#ifdef USE_SSLSTREAM
|
| + rtc::StreamInterface *stream_;
|
| +#endif // USE_SSLSTREAM
|
| + buzz::AsyncSocket::State state_;
|
| + rtc::Buffer buffer_;
|
| + buzz::TlsOptions tls_;
|
| +};
|
| +
|
| +} // namespace buzz
|
| +
|
| +#endif // WEBRTC_LIBJINGLE_XMPP_XMPPSOCKET_H_
|
| +
|
|
|