73 lines
2.2 KiB
C
73 lines
2.2 KiB
C
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/*
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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_ASYNCSOCKET_H_
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#define WEBRTC_LIBJINGLE_XMPP_ASYNCSOCKET_H_
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#include <string>
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#include "webrtc/base/sigslot.h"
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namespace rtc {
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class SocketAddress;
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}
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namespace buzz {
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class AsyncSocket {
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public:
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enum State {
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STATE_CLOSED = 0, //!< Socket is not open.
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STATE_CLOSING, //!< Socket is closing but can have buffered data
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STATE_CONNECTING, //!< In the process of
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STATE_OPEN, //!< Socket is connected
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#if defined(FEATURE_ENABLE_SSL)
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STATE_TLS_CONNECTING, //!< Establishing TLS connection
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STATE_TLS_OPEN, //!< TLS connected
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#endif
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};
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enum Error {
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ERROR_NONE = 0, //!< No error
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ERROR_WINSOCK, //!< Winsock error
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ERROR_DNS, //!< Couldn't resolve host name
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ERROR_WRONGSTATE, //!< Call made while socket is in the wrong state
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#if defined(FEATURE_ENABLE_SSL)
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ERROR_SSL, //!< Something went wrong with OpenSSL
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#endif
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};
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virtual ~AsyncSocket() {}
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virtual State state() = 0;
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virtual Error error() = 0;
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virtual int GetError() = 0; // winsock error code
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virtual bool Connect(const rtc::SocketAddress& addr) = 0;
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virtual bool Read(char * data, size_t len, size_t* len_read) = 0;
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virtual bool Write(const char * data, size_t len) = 0;
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virtual bool Close() = 0;
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#if defined(FEATURE_ENABLE_SSL)
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// We allow matching any passed domain. This allows us to avoid
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// handling the valuable certificates for logins into proxies. If
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// both names are passed as empty, we do not require a match.
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virtual bool StartTls(const std::string & domainname) = 0;
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#endif
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sigslot::signal0<> SignalConnected;
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sigslot::signal0<> SignalSSLConnected;
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sigslot::signal0<> SignalClosed;
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sigslot::signal0<> SignalRead;
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sigslot::signal0<> SignalError;
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};
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}
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#endif // WEBRTC_LIBJINGLE_XMPP_ASYNCSOCKET_H_
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