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=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17479005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6129 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/base/asyncudpsocket.h b/webrtc/base/asyncudpsocket.h
new file mode 100644
index 0000000..ac64dca
--- /dev/null
+++ b/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_