269fb4bc90
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p. Also makes libjingle use those version instead of the one in the talk folder. BUG=3379 Review URL: https://webrtc-codereview.appspot.com/26999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7549 4adac7df-926f-26a2-2b94-8c16560cd09d
51 lines
1.8 KiB
C++
51 lines
1.8 KiB
C++
/*
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* Copyright 2013 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_P2P_BASE_ASYNCSTUNTCPSOCKET_H_
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#define WEBRTC_P2P_BASE_ASYNCSTUNTCPSOCKET_H_
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#include "webrtc/base/asynctcpsocket.h"
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/base/socketfactory.h"
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namespace cricket {
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class AsyncStunTCPSocket : public rtc::AsyncTCPSocketBase {
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public:
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// Binds and connects |socket| and creates AsyncTCPSocket for
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// it. Takes ownership of |socket|. Returns NULL if bind() or
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// connect() fail (|socket| is destroyed in that case).
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static AsyncStunTCPSocket* Create(
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rtc::AsyncSocket* socket,
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const rtc::SocketAddress& bind_address,
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const rtc::SocketAddress& remote_address);
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AsyncStunTCPSocket(rtc::AsyncSocket* socket, bool listen);
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virtual ~AsyncStunTCPSocket() {}
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virtual int Send(const void* pv, size_t cb,
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const rtc::PacketOptions& options);
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virtual void ProcessInput(char* data, size_t* len);
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virtual void HandleIncomingConnection(rtc::AsyncSocket* socket);
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private:
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// This method returns the message hdr + length written in the header.
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// This method also returns the number of padding bytes needed/added to the
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// turn message. |pad_bytes| should be used only when |is_turn| is true.
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size_t GetExpectedLength(const void* data, size_t len,
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int* pad_bytes);
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DISALLOW_EVIL_CONSTRUCTORS(AsyncStunTCPSocket);
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};
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} // namespace cricket
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#endif // WEBRTC_P2P_BASE_ASYNCSTUNTCPSOCKET_H_
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