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_ASYNCSOCKET_H_ 12 #define WEBRTC_LIBJINGLE_XMPP_ASYNCSOCKET_H_ 13 14 #include <string> 15 16 #include "webrtc/base/sigslot.h" 17 18 namespace rtc { 19 class SocketAddress; 20 } 21 22 namespace buzz { 23 24 class AsyncSocket { 25 public: 26 enum State { 27 STATE_CLOSED = 0, //!< Socket is not open. 28 STATE_CLOSING, //!< Socket is closing but can have buffered data 29 STATE_CONNECTING, //!< In the process of 30 STATE_OPEN, //!< Socket is connected 31 #if defined(FEATURE_ENABLE_SSL) 32 STATE_TLS_CONNECTING, //!< Establishing TLS connection 33 STATE_TLS_OPEN, //!< TLS connected 34 #endif 35 }; 36 37 enum Error { 38 ERROR_NONE = 0, //!< No error 39 ERROR_WINSOCK, //!< Winsock error 40 ERROR_DNS, //!< Couldn't resolve host name 41 ERROR_WRONGSTATE, //!< Call made while socket is in the wrong state 42 #if defined(FEATURE_ENABLE_SSL) 43 ERROR_SSL, //!< Something went wrong with OpenSSL 44 #endif 45 }; 46 47 virtual ~AsyncSocket() {} 48 virtual State state() = 0; 49 virtual Error error() = 0; 50 virtual int GetError() = 0; // winsock error code 51 52 virtual bool Connect(const rtc::SocketAddress& addr) = 0; 53 virtual bool Read(char * data, size_t len, size_t* len_read) = 0; 54 virtual bool Write(const char * data, size_t len) = 0; 55 virtual bool Close() = 0; 56 #if defined(FEATURE_ENABLE_SSL) 57 // We allow matching any passed domain. This allows us to avoid 58 // handling the valuable certificates for logins into proxies. If 59 // both names are passed as empty, we do not require a match. 60 virtual bool StartTls(const std::string & domainname) = 0; 61 #endif 62 63 sigslot::signal0<> SignalConnected; 64 sigslot::signal0<> SignalSSLConnected; 65 sigslot::signal0<> SignalClosed; 66 sigslot::signal0<> SignalRead; 67 sigslot::signal0<> SignalError; 68 }; 69 70 } 71 72 #endif // WEBRTC_LIBJINGLE_XMPP_ASYNCSOCKET_H_ 73
