f048872e91
migrating talk/base to webrtc/base. BUG=N/A R=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17479005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6129 4adac7df-926f-26a2-2b94-8c16560cd09d
45 lines
1.2 KiB
C++
45 lines
1.2 KiB
C++
/*
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* Copyright 2010 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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#include "webrtc/base/asyncsocket.h"
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namespace rtc {
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AsyncSocket::AsyncSocket() {
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}
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AsyncSocket::~AsyncSocket() {
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}
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AsyncSocketAdapter::AsyncSocketAdapter(AsyncSocket* socket) : socket_(NULL) {
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Attach(socket);
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}
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AsyncSocketAdapter::~AsyncSocketAdapter() {
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delete socket_;
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}
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void AsyncSocketAdapter::Attach(AsyncSocket* socket) {
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ASSERT(!socket_);
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socket_ = socket;
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if (socket_) {
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socket_->SignalConnectEvent.connect(this,
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&AsyncSocketAdapter::OnConnectEvent);
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socket_->SignalReadEvent.connect(this,
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&AsyncSocketAdapter::OnReadEvent);
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socket_->SignalWriteEvent.connect(this,
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&AsyncSocketAdapter::OnWriteEvent);
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socket_->SignalCloseEvent.connect(this,
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&AsyncSocketAdapter::OnCloseEvent);
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}
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}
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} // namespace rtc
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