Revert 6107 "Adds a modified copy of talk/base to webrtc/base. I..."

This breaks Chromium FYI builds and prevent roll of webrtc/libjingle to Chrome.

http://chromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Win%20Builder/builds/457


> Adds a modified copy of talk/base to webrtc/base. It is the first step in migrating talk/base to webrtc/base.
> 
> BUG=N/A
> R=andrew@webrtc.org, wu@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/12199004

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6116 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
perkj@webrtc.org
2014-05-13 08:15:48 +00:00
parent 3a5825909d
commit e9a604accd
386 changed files with 6 additions and 84973 deletions

View File

@@ -1,63 +0,0 @@
/*
* 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_BASE_ASYNCUDPSOCKET_H_
#define WEBRTC_BASE_ASYNCUDPSOCKET_H_
#include "webrtc/base/asyncpacketsocket.h"
#include "webrtc/base/scoped_ptr.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);
virtual ~AsyncUDPSocket();
virtual SocketAddress GetLocalAddress() const;
virtual SocketAddress GetRemoteAddress() const;
virtual int Send(const void *pv, size_t cb,
const rtc::PacketOptions& options);
virtual int SendTo(const void *pv, size_t cb, const SocketAddress& addr,
const rtc::PacketOptions& options);
virtual int Close();
virtual State GetState() const;
virtual int GetOption(Socket::Option opt, int* value);
virtual int SetOption(Socket::Option opt, int value);
virtual int GetError() const;
virtual void SetError(int error);
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);
scoped_ptr<AsyncSocket> socket_;
char* buf_;
size_t size_;
};
} // namespace rtc
#endif // WEBRTC_BASE_ASYNCUDPSOCKET_H_