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}
diff --git a/webrtc/rtc_base/bufferqueue.h b/webrtc/rtc_base/bufferqueue.h
new file mode 100644
index 0000000..7db2c8c
--- /dev/null
+++ b/webrtc/rtc_base/bufferqueue.h
@@ -0,0 +1,61 @@
+/*
+ *  Copyright 2015 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_BUFFERQUEUE_H_
+#define WEBRTC_RTC_BASE_BUFFERQUEUE_H_
+
+#include <deque>
+#include <vector>
+
+#include "webrtc/base/buffer.h"
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/criticalsection.h"
+
+namespace rtc {
+
+class BufferQueue {
+ public:
+  // Creates a buffer queue with a given capacity and default buffer size.
+  BufferQueue(size_t capacity, size_t default_size);
+  virtual ~BufferQueue();
+
+  // Return number of queued buffers.
+  size_t size() const;
+
+  // Clear the BufferQueue by moving all Buffers from |queue_| to |free_list_|.
+  void Clear();
+
+  // ReadFront will only read one buffer at a time and will truncate buffers
+  // that don't fit in the passed memory.
+  // Returns true unless no data could be returned.
+  bool ReadFront(void* data, size_t bytes, size_t* bytes_read);
+
+  // WriteBack always writes either the complete memory or nothing.
+  // Returns true unless no data could be written.
+  bool WriteBack(const void* data, size_t bytes, size_t* bytes_written);
+
+ protected:
+  // These methods are called when the state of the queue changes.
+  virtual void NotifyReadableForTest() {}
+  virtual void NotifyWritableForTest() {}
+
+ private:
+  size_t capacity_;
+  size_t default_size_;
+  CriticalSection crit_;
+  std::deque<Buffer*> queue_ GUARDED_BY(crit_);
+  std::vector<Buffer*> free_list_ GUARDED_BY(crit_);
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(BufferQueue);
+};
+
+}  // namespace rtc
+
+#endif  // WEBRTC_RTC_BASE_BUFFERQUEUE_H_