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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6107 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/base/asyncudpsocket.h

@@ -0,0 +1,63 @@
+/*
+ *  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_