Reland "Move webrtc/{base => rtc_base}" (https://codereview.webrtc.org/2877023002)
Reland the base->rtc_base without adding stub headers (will be done in follow-up CL). This preserves git blame history of all files. BUG=webrtc:7634 NOTRY=True TBR=kwiberg@webrtc.org Change-Id: Iea3bb6f3f67b8374c96337b63e8f5aa3e6181012 Reviewed-on: https://chromium-review.googlesource.com/554611 Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#18821}
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/*
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* Copyright 2004 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_BASE_SOCKETFACTORY_H__
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#define WEBRTC_BASE_SOCKETFACTORY_H__
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#include "webrtc/base/socket.h"
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#include "webrtc/base/asyncsocket.h"
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namespace rtc {
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class SocketFactory {
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public:
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virtual ~SocketFactory() {}
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// Returns a new socket for blocking communication. The type can be
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// SOCK_DGRAM and SOCK_STREAM.
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// TODO: C++ inheritance rules mean that all users must have both
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// CreateSocket(int) and CreateSocket(int,int). Will remove CreateSocket(int)
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// (and CreateAsyncSocket(int) when all callers are changed.
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virtual Socket* CreateSocket(int type) = 0;
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virtual Socket* CreateSocket(int family, int type) = 0;
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// Returns a new socket for nonblocking communication. The type can be
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// SOCK_DGRAM and SOCK_STREAM.
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virtual AsyncSocket* CreateAsyncSocket(int type) = 0;
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virtual AsyncSocket* CreateAsyncSocket(int family, int type) = 0;
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};
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} // namespace rtc
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#endif // WEBRTC_BASE_SOCKETFACTORY_H__
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