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_ASYNCTCPSOCKET_H_
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#define WEBRTC_BASE_ASYNCTCPSOCKET_H_
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#include <memory>
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#include "webrtc/base/asyncpacketsocket.h"
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#include "webrtc/base/buffer.h"
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/base/socketfactory.h"
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namespace rtc {
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// Simulates UDP semantics over TCP. Send and Recv packet sizes
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// are preserved, and drops packets silently on Send, rather than
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// buffer them in user space.
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class AsyncTCPSocketBase : public AsyncPacketSocket {
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public:
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AsyncTCPSocketBase(AsyncSocket* socket, bool listen, size_t max_packet_size);
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~AsyncTCPSocketBase() override;
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// Pure virtual methods to send and recv data.
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int Send(const void *pv, size_t cb,
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const rtc::PacketOptions& options) override = 0;
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virtual void ProcessInput(char* data, size_t* len) = 0;
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// Signals incoming connection.
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virtual void HandleIncomingConnection(AsyncSocket* socket) = 0;
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SocketAddress GetLocalAddress() const override;
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SocketAddress GetRemoteAddress() const 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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protected:
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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 AsyncSocket* ConnectSocket(AsyncSocket* socket,
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const SocketAddress& bind_address,
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const SocketAddress& remote_address);
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virtual int SendRaw(const void* pv, size_t cb);
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int FlushOutBuffer();
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// Add data to |outbuf_|.
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void AppendToOutBuffer(const void* pv, size_t cb);
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// Helper methods for |outpos_|.
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bool IsOutBufferEmpty() const { return outbuf_.size() == 0; }
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void ClearOutBuffer() { outbuf_.Clear(); }
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private:
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// Called by the underlying socket
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void OnConnectEvent(AsyncSocket* socket);
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void OnReadEvent(AsyncSocket* socket);
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void OnWriteEvent(AsyncSocket* socket);
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void OnCloseEvent(AsyncSocket* socket, int error);
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std::unique_ptr<AsyncSocket> socket_;
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bool listen_;
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Buffer inbuf_;
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Buffer outbuf_;
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size_t max_insize_;
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size_t max_outsize_;
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RTC_DISALLOW_COPY_AND_ASSIGN(AsyncTCPSocketBase);
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};
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class AsyncTCPSocket : public 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 AsyncTCPSocket* Create(AsyncSocket* socket,
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const SocketAddress& bind_address,
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const SocketAddress& remote_address);
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AsyncTCPSocket(AsyncSocket* socket, bool listen);
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~AsyncTCPSocket() 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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void ProcessInput(char* data, size_t* len) override;
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void HandleIncomingConnection(AsyncSocket* socket) override;
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private:
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RTC_DISALLOW_COPY_AND_ASSIGN(AsyncTCPSocket);
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};
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} // namespace rtc
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#endif // WEBRTC_BASE_ASYNCTCPSOCKET_H_
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