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/asynctcpsocket.h b/webrtc/base/asynctcpsocket.h
new file mode 100644
index 0000000..ddee261
--- /dev/null
+++ b/webrtc/base/asynctcpsocket.h
@@ -0,0 +1,100 @@
+/*
+ * 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_ASYNCTCPSOCKET_H_
+#define WEBRTC_BASE_ASYNCTCPSOCKET_H_
+
+#include "webrtc/base/asyncpacketsocket.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/socketfactory.h"
+
+namespace rtc {
+
+// Simulates UDP semantics over TCP. Send and Recv packet sizes
+// are preserved, and drops packets silently on Send, rather than
+// buffer them in user space.
+class AsyncTCPSocketBase : public AsyncPacketSocket {
+ public:
+ AsyncTCPSocketBase(AsyncSocket* socket, bool listen, size_t max_packet_size);
+ virtual ~AsyncTCPSocketBase();
+
+ // Pure virtual methods to send and recv data.
+ virtual int Send(const void *pv, size_t cb,
+ const rtc::PacketOptions& options) = 0;
+ virtual void ProcessInput(char* data, size_t* len) = 0;
+ // Signals incoming connection.
+ virtual void HandleIncomingConnection(AsyncSocket* socket) = 0;
+
+ virtual SocketAddress GetLocalAddress() const;
+ virtual SocketAddress GetRemoteAddress() const;
+ 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);
+
+ protected:
+ // Binds and connects |socket| and creates AsyncTCPSocket for
+ // it. Takes ownership of |socket|. Returns NULL if bind() or
+ // connect() fail (|socket| is destroyed in that case).
+ static AsyncSocket* ConnectSocket(AsyncSocket* socket,
+ const SocketAddress& bind_address,
+ const SocketAddress& remote_address);
+ virtual int SendRaw(const void* pv, size_t cb);
+ int FlushOutBuffer();
+ // Add data to |outbuf_|.
+ void AppendToOutBuffer(const void* pv, size_t cb);
+
+ // Helper methods for |outpos_|.
+ bool IsOutBufferEmpty() const { return outpos_ == 0; }
+ void ClearOutBuffer() { outpos_ = 0; }
+
+ private:
+ // Called by the underlying socket
+ void OnConnectEvent(AsyncSocket* socket);
+ void OnReadEvent(AsyncSocket* socket);
+ void OnWriteEvent(AsyncSocket* socket);
+ void OnCloseEvent(AsyncSocket* socket, int error);
+
+ scoped_ptr<AsyncSocket> socket_;
+ bool listen_;
+ char* inbuf_, * outbuf_;
+ size_t insize_, inpos_, outsize_, outpos_;
+
+ DISALLOW_EVIL_CONSTRUCTORS(AsyncTCPSocketBase);
+};
+
+class AsyncTCPSocket : public AsyncTCPSocketBase {
+ public:
+ // Binds and connects |socket| and creates AsyncTCPSocket for
+ // it. Takes ownership of |socket|. Returns NULL if bind() or
+ // connect() fail (|socket| is destroyed in that case).
+ static AsyncTCPSocket* Create(AsyncSocket* socket,
+ const SocketAddress& bind_address,
+ const SocketAddress& remote_address);
+ AsyncTCPSocket(AsyncSocket* socket, bool listen);
+ virtual ~AsyncTCPSocket() {}
+
+ virtual int Send(const void* pv, size_t cb,
+ const rtc::PacketOptions& options);
+ virtual void ProcessInput(char* data, size_t* len);
+ virtual void HandleIncomingConnection(AsyncSocket* socket);
+
+ private:
+ DISALLOW_EVIL_CONSTRUCTORS(AsyncTCPSocket);
+};
+
+} // namespace rtc
+
+#endif // WEBRTC_BASE_ASYNCTCPSOCKET_H_