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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webrtc/rtc_base/asyncudpsocket.h
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webrtc/rtc_base/asyncudpsocket.h
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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_RTC_BASE_ASYNCUDPSOCKET_H_
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#define WEBRTC_RTC_BASE_ASYNCUDPSOCKET_H_
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#include <memory>
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#include "webrtc/base/asyncpacketsocket.h"
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#include "webrtc/base/socketfactory.h"
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namespace rtc {
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// Provides the ability to receive packets asynchronously. Sends are not
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// buffered since it is acceptable to drop packets under high load.
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class AsyncUDPSocket : public AsyncPacketSocket {
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public:
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// Binds |socket| and creates AsyncUDPSocket for it. Takes ownership
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// of |socket|. Returns null if bind() fails (|socket| is destroyed
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// in that case).
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static AsyncUDPSocket* Create(AsyncSocket* socket,
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const SocketAddress& bind_address);
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// Creates a new socket for sending asynchronous UDP packets using an
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// asynchronous socket from the given factory.
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static AsyncUDPSocket* Create(SocketFactory* factory,
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const SocketAddress& bind_address);
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explicit AsyncUDPSocket(AsyncSocket* socket);
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~AsyncUDPSocket() override;
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SocketAddress GetLocalAddress() const override;
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SocketAddress GetRemoteAddress() const override;
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int Send(const void* pv,
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size_t cb,
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const rtc::PacketOptions& options) override;
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int SendTo(const void* pv,
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size_t cb,
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const SocketAddress& addr,
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const rtc::PacketOptions& options) override;
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int Close() override;
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State GetState() const override;
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int GetOption(Socket::Option opt, int* value) override;
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int SetOption(Socket::Option opt, int value) override;
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int GetError() const override;
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void SetError(int error) override;
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private:
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// Called when the underlying socket is ready to be read from.
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void OnReadEvent(AsyncSocket* socket);
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// Called when the underlying socket is ready to send.
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void OnWriteEvent(AsyncSocket* socket);
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std::unique_ptr<AsyncSocket> socket_;
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char* buf_;
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size_t size_;
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};
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} // namespace rtc
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#endif // WEBRTC_RTC_BASE_ASYNCUDPSOCKET_H_
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