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/socketstream.h

@@ -0,0 +1,61 @@
+/*
+ *  Copyright 2005 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_SOCKETSTREAM_H_
+#define WEBRTC_RTC_BASE_SOCKETSTREAM_H_
+
+#include "webrtc/base/asyncsocket.h"
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/stream.h"
+
+namespace rtc {
+
+///////////////////////////////////////////////////////////////////////////////
+
+class SocketStream : public StreamInterface, public sigslot::has_slots<> {
+ public:
+  explicit SocketStream(AsyncSocket* socket);
+  ~SocketStream() override;
+
+  void Attach(AsyncSocket* socket);
+  AsyncSocket* Detach();
+
+  AsyncSocket* GetSocket() { return socket_; }
+
+  StreamState GetState() const override;
+
+  StreamResult Read(void* buffer,
+                    size_t buffer_len,
+                    size_t* read,
+                    int* error) override;
+
+  StreamResult Write(const void* data,
+                     size_t data_len,
+                     size_t* written,
+                     int* error) override;
+
+  void Close() override;
+
+ private:
+  void OnConnectEvent(AsyncSocket* socket);
+  void OnReadEvent(AsyncSocket* socket);
+  void OnWriteEvent(AsyncSocket* socket);
+  void OnCloseEvent(AsyncSocket* socket, int err);
+
+  AsyncSocket* socket_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(SocketStream);
+};
+
+///////////////////////////////////////////////////////////////////////////////
+
+}  // namespace rtc
+
+#endif  // WEBRTC_RTC_BASE_SOCKETSTREAM_H_