Reland "Move webrtc/{base => rtc_base}" (https://codereview.webrtc.org/2877023002)

Reland the base->rtc_base without adding stub headers (will be
done in follow-up CL). This preserves git blame history of all files.

BUG=webrtc:7634
NOTRY=True
TBR=kwiberg@webrtc.org

Change-Id: Iea3bb6f3f67b8374c96337b63e8f5aa3e6181012
Reviewed-on: https://chromium-review.googlesource.com/554611
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18821}

webrtc/rtc_base/asyncpacketsocket.cc

@@ -0,0 +1,29 @@
+/*
+ *  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.
+ */
+
+#include "webrtc/base/asyncpacketsocket.h"
+
+namespace rtc {
+
+PacketTimeUpdateParams::PacketTimeUpdateParams()
+    : rtp_sendtime_extension_id(-1),
+      srtp_auth_tag_len(-1),
+      srtp_packet_index(-1) {
+}
+
+PacketTimeUpdateParams::~PacketTimeUpdateParams() = default;
+
+AsyncPacketSocket::AsyncPacketSocket() {
+}
+
+AsyncPacketSocket::~AsyncPacketSocket() {
+}
+
+};  // namespace rtc