Index: webrtc/libjingle/xmpp/xmppsocket.h |
diff --git a/webrtc/libjingle/xmpp/xmppsocket.h b/webrtc/libjingle/xmpp/xmppsocket.h |
deleted file mode 100644 |
index 02d645383c3b9774a291d0b43f0bdb70d485ac71..0000000000000000000000000000000000000000 |
--- a/webrtc/libjingle/xmpp/xmppsocket.h |
+++ /dev/null |
@@ -1,72 +0,0 @@ |
-/* |
- * 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_ |
- |