Revert "Move webrtc/{base => rtc_base}" (https://codereview.webrtc.org/2877023002)
Will reland in two different commits to preserve git blame history.
BUG=webrtc:7634
NOTRY=True
TBR=kwiberg@webrtc.org
Change-Id: I550da8525aeb9c5b8f96338fcf1c9714f3dcdab1
Reviewed-on: https://chromium-review.googlesource.com/554610
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18820}diff --git a/webrtc/base/asynctcpsocket.h b/webrtc/base/asynctcpsocket.h
index d64927b..2e4ff9a 100644
--- a/webrtc/base/asynctcpsocket.h
+++ b/webrtc/base/asynctcpsocket.h
@@ -11,9 +11,98 @@
#ifndef WEBRTC_BASE_ASYNCTCPSOCKET_H_
#define WEBRTC_BASE_ASYNCTCPSOCKET_H_
+#include <memory>
-// This header is deprecated and is just left here temporarily during
-// refactoring. See https://bugs.webrtc.org/7634 for more details.
-#include "webrtc/rtc_base/asynctcpsocket.h"
+#include "webrtc/base/asyncpacketsocket.h"
+#include "webrtc/base/buffer.h"
+#include "webrtc/base/constructormagic.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);
+ ~AsyncTCPSocketBase() override;
+
+ // Pure virtual methods to send and recv data.
+ int Send(const void *pv, size_t cb,
+ const rtc::PacketOptions& options) override = 0;
+ virtual void ProcessInput(char* data, size_t* len) = 0;
+ // Signals incoming connection.
+ virtual void HandleIncomingConnection(AsyncSocket* socket) = 0;
+
+ SocketAddress GetLocalAddress() const override;
+ SocketAddress GetRemoteAddress() const 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;
+
+ 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 outbuf_.size() == 0; }
+ void ClearOutBuffer() { outbuf_.Clear(); }
+
+ private:
+ // Called by the underlying socket
+ void OnConnectEvent(AsyncSocket* socket);
+ void OnReadEvent(AsyncSocket* socket);
+ void OnWriteEvent(AsyncSocket* socket);
+ void OnCloseEvent(AsyncSocket* socket, int error);
+
+ std::unique_ptr<AsyncSocket> socket_;
+ bool listen_;
+ Buffer inbuf_;
+ Buffer outbuf_;
+ size_t max_insize_;
+ size_t max_outsize_;
+
+ RTC_DISALLOW_COPY_AND_ASSIGN(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);
+ ~AsyncTCPSocket() override {}
+
+ int Send(const void* pv,
+ size_t cb,
+ const rtc::PacketOptions& options) override;
+ void ProcessInput(char* data, size_t* len) override;
+ void HandleIncomingConnection(AsyncSocket* socket) override;
+
+ private:
+ RTC_DISALLOW_COPY_AND_ASSIGN(AsyncTCPSocket);
+};
+
+} // namespace rtc
#endif // WEBRTC_BASE_ASYNCTCPSOCKET_H_
