Files
platform-external-webrtc/webrtc/rtc_base/asyncudpsocket.h
Henrik Kjellander 6776518bea Move webrtc/{base => rtc_base}
This refactoring takes a careful approach to avoid rushing the change:
* stub headers are left in all the old locations of webrtc/base
* existing GN targets are kept and now just forward to the moved ones
  using public_deps.
The only exception to the above is the base_java target and its .java files,
which were moved to webrtc/rtc_base right away since it's not possible
to use public_deps for android_library.
To avoid breaking builds, a temporary Dummy.java file was added to
the new intermediate target in webrtc/rtc_base:base_java as well to avoid
hitting a GN assert in the android_library template.

The above approach should make the transition smooth without breaking
downstream.

A helper script was created (https://codereview.webrtc.org/2879203002/)
and was run like this:
stub-headers.py -s webrtc/base -d webrtc/rtc_base -i 7634
stub-headers.py -s webrtc/base/numerics -d webrtc/rtc_base/numerics -i 7634

Fixed invalid header guards in the following files:
webrtc/base/base64.h
webrtc/base/cryptstring.h
webrtc/base/event.h
webrtc/base/flags.h
webrtc/base/httpbase.h
webrtc/base/httpcommon-inl.h
webrtc/base/httpcommon.h
webrtc/base/httpserver.h
webrtc/base/logsinks.h
webrtc/base/macutils.h
webrtc/base/nattypes.h
webrtc/base/openssladapter.h
webrtc/base/opensslstreamadapter.h
webrtc/base/pathutils.h
webrtc/base/physicalsocketserver.h
webrtc/base/proxyinfo.h
webrtc/base/sigslot.h
webrtc/base/sigslotrepeater.h
webrtc/base/socket.h
webrtc/base/socketaddresspair.h
webrtc/base/socketfactory.h
webrtc/base/stringutils.h
webrtc/base/testbase64.h
webrtc/base/testutils.h
webrtc/base/transformadapter.h
webrtc/base/win32filesystem.h

Added new header guards to:
sslroots.h
testbase64.h

BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True
R=kwiberg@webrtc.org

Review-Url: https://codereview.webrtc.org/2877023002 .
Cr-Commit-Position: refs/heads/master@{#18816}
2017-06-28 18:58:10 +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_