webrtc_m130/webrtc/rtc_base/asyncudpsocket.h
Henrik Kjellander c03627683f 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}
2017-06-29 06:04:25 +00:00

68 lines
2.3 KiB
C++

/*
* Copyright 2004 The WebRTC Project Authors. All rights reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_RTC_BASE_ASYNCUDPSOCKET_H_
#define WEBRTC_RTC_BASE_ASYNCUDPSOCKET_H_
#include <memory>
#include "webrtc/base/asyncpacketsocket.h"
#include "webrtc/base/socketfactory.h"
namespace rtc {
// Provides the ability to receive packets asynchronously. Sends are not
// buffered since it is acceptable to drop packets under high load.
class AsyncUDPSocket : public AsyncPacketSocket {
public:
// Binds |socket| and creates AsyncUDPSocket for it. Takes ownership
// of |socket|. Returns null if bind() fails (|socket| is destroyed
// in that case).
static AsyncUDPSocket* Create(AsyncSocket* socket,
const SocketAddress& bind_address);
// Creates a new socket for sending asynchronous UDP packets using an
// asynchronous socket from the given factory.
static AsyncUDPSocket* Create(SocketFactory* factory,
const SocketAddress& bind_address);
explicit AsyncUDPSocket(AsyncSocket* socket);
~AsyncUDPSocket() override;
SocketAddress GetLocalAddress() const override;
SocketAddress GetRemoteAddress() const override;
int Send(const void* pv,
size_t cb,
const rtc::PacketOptions& options) override;
int SendTo(const void* pv,
size_t cb,
const SocketAddress& addr,
const rtc::PacketOptions& options) override;
int Close() override;
State GetState() const override;
int GetOption(Socket::Option opt, int* value) override;
int SetOption(Socket::Option opt, int value) override;
int GetError() const override;
void SetError(int error) override;
private:
// Called when the underlying socket is ready to be read from.
void OnReadEvent(AsyncSocket* socket);
// Called when the underlying socket is ready to send.
void OnWriteEvent(AsyncSocket* socket);
std::unique_ptr<AsyncSocket> socket_;
char* buf_;
size_t size_;
};
} // namespace rtc
#endif // WEBRTC_RTC_BASE_ASYNCUDPSOCKET_H_