Move webrtc/{base => rtc_base}

This refactoring takes a careful approach to avoid rushing the change:
* stub headers are left in all the old locations of webrtc/base
* existing GN targets are kept and now just forward to the moved ones
  using public_deps.
The only exception to the above is the base_java target and its .java files,
which were moved to webrtc/rtc_base right away since it's not possible
to use public_deps for android_library.
To avoid breaking builds, a temporary Dummy.java file was added to
the new intermediate target in webrtc/rtc_base:base_java as well to avoid
hitting a GN assert in the android_library template.

The above approach should make the transition smooth without breaking
downstream.

A helper script was created (https://codereview.webrtc.org/2879203002/)
and was run like this:
stub-headers.py -s webrtc/base -d webrtc/rtc_base -i 7634
stub-headers.py -s webrtc/base/numerics -d webrtc/rtc_base/numerics -i 7634

Fixed invalid header guards in the following files:
webrtc/base/base64.h
webrtc/base/cryptstring.h
webrtc/base/event.h
webrtc/base/flags.h
webrtc/base/httpbase.h
webrtc/base/httpcommon-inl.h
webrtc/base/httpcommon.h
webrtc/base/httpserver.h
webrtc/base/logsinks.h
webrtc/base/macutils.h
webrtc/base/nattypes.h
webrtc/base/openssladapter.h
webrtc/base/opensslstreamadapter.h
webrtc/base/pathutils.h
webrtc/base/physicalsocketserver.h
webrtc/base/proxyinfo.h
webrtc/base/sigslot.h
webrtc/base/sigslotrepeater.h
webrtc/base/socket.h
webrtc/base/socketaddresspair.h
webrtc/base/socketfactory.h
webrtc/base/stringutils.h
webrtc/base/testbase64.h
webrtc/base/testutils.h
webrtc/base/transformadapter.h
webrtc/base/win32filesystem.h

Added new header guards to:
sslroots.h
testbase64.h

BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True
R=kwiberg@webrtc.org

Review-Url: https://codereview.webrtc.org/2877023002 .
Cr-Commit-Position: refs/heads/master@{#18816}

webrtc/rtc_base/asyncresolverinterface.h

@@ -0,0 +1,47 @@
+/*
+ *  Copyright 2013 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_RTC_BASE_ASYNCRESOLVERINTERFACE_H_
+#define WEBRTC_RTC_BASE_ASYNCRESOLVERINTERFACE_H_
+
+#include "webrtc/base/sigslot.h"
+#include "webrtc/base/socketaddress.h"
+
+namespace rtc {
+
+// This interface defines the methods to resolve the address asynchronously.
+class AsyncResolverInterface {
+ public:
+  AsyncResolverInterface();
+  virtual ~AsyncResolverInterface();
+
+  // Start address resolve process.
+  virtual void Start(const SocketAddress& addr) = 0;
+  // Returns top most resolved address of |family|
+  virtual bool GetResolvedAddress(int family, SocketAddress* addr) const = 0;
+  // Returns error from resolver.
+  virtual int GetError() const = 0;
+  // Delete the resolver.
+  virtual void Destroy(bool wait) = 0;
+  // Returns top most resolved IPv4 address if address is resolved successfully.
+  // Otherwise returns address set in SetAddress.
+  SocketAddress address() const {
+    SocketAddress addr;
+    GetResolvedAddress(AF_INET, &addr);
+    return addr;
+  }
+
+  // This signal is fired when address resolve process is completed.
+  sigslot::signal1<AsyncResolverInterface*> SignalDone;
+};
+
+}  // namespace rtc
+
+#endif