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-Original-Commit-Position: refs/heads/master@{#18820}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: ec78f1cebcbf1181d93450bd9c91efe8f6a7688d
diff --git a/base/asyncudpsocket.h b/base/asyncudpsocket.h
index c3212c0..e5535e0 100644
--- a/base/asyncudpsocket.h
+++ b/base/asyncudpsocket.h
@@ -11,9 +11,57 @@
 #ifndef WEBRTC_BASE_ASYNCUDPSOCKET_H_
 #define WEBRTC_BASE_ASYNCUDPSOCKET_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/asyncudpsocket.h"
+#include "webrtc/base/asyncpacketsocket.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);
+  ~AsyncUDPSocket() override;
+
+  SocketAddress GetLocalAddress() const override;
+  SocketAddress GetRemoteAddress() const override;
+  int Send(const void* pv,
+           size_t cb,
+           const rtc::PacketOptions& options) 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;
+
+ 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);
+
+  std::unique_ptr<AsyncSocket> socket_;
+  char* buf_;
+  size_t size_;
+};
+
+}  // namespace rtc
 
 #endif  // WEBRTC_BASE_ASYNCUDPSOCKET_H_