Remove unneeded #defines from libjingle_xmpp.

third_party/libjingle_xmpp/xmpp/asyncsocket.h

 /*
  *  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.
  */
 
@@ skipping 10 matching lines @@
 
 namespace buzz {
 
 class AsyncSocket {
 public:
   enum State {
     STATE_CLOSED = 0,      //!< Socket is not open.
     STATE_CLOSING,         //!< Socket is closing but can have buffered data
     STATE_CONNECTING,      //!< In the process of
     STATE_OPEN,            //!< Socket is connected
-#if defined(FEATURE_ENABLE_SSL)
     STATE_TLS_CONNECTING,  //!< Establishing TLS connection
     STATE_TLS_OPEN,        //!< TLS connected
-#endif
   };
 
   enum Error {
     ERROR_NONE = 0,         //!< No error
     ERROR_WINSOCK,          //!< Winsock error
     ERROR_DNS,              //!< Couldn't resolve host name
     ERROR_WRONGSTATE,       //!< Call made while socket is in the wrong state
-#if defined(FEATURE_ENABLE_SSL)
     ERROR_SSL,              //!< Something went wrong with OpenSSL
-#endif
   };
 
   virtual ~AsyncSocket() {}
   virtual State state() = 0;
   virtual Error error() = 0;
   virtual int GetError() = 0;    // winsock error code
 
   virtual bool Connect(const rtc::SocketAddress& addr) = 0;
   virtual bool Read(char * data, size_t len, size_t* len_read) = 0;
   virtual bool Write(const char * data, size_t len) = 0;
   virtual bool Close() = 0;
-#if defined(FEATURE_ENABLE_SSL)
   // We allow matching any passed domain.  This allows us to avoid
   // handling the valuable certificates for logins into proxies.  If
   // both names are passed as empty, we do not require a match.
   virtual bool StartTls(const std::string & domainname) = 0;
-#endif
 
   sigslot::signal0<> SignalConnected;
   sigslot::signal0<> SignalSSLConnected;
   sigslot::signal0<> SignalClosed;
   sigslot::signal0<> SignalRead;
   sigslot::signal0<> SignalError;
 };
 
 }
 
 #endif  // WEBRTC_LIBJINGLE_XMPP_ASYNCSOCKET_H_