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/testechoserver.h

@@ -0,0 +1,75 @@
+/*
+ *  Copyright 2004 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_TESTECHOSERVER_H_
+#define WEBRTC_RTC_BASE_TESTECHOSERVER_H_
+
+#include <list>
+#include <memory>
+#include "webrtc/base/asynctcpsocket.h"
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/socketaddress.h"
+#include "webrtc/base/sigslot.h"
+#include "webrtc/base/thread.h"
+
+namespace rtc {
+
+// A test echo server, echoes back any packets sent to it.
+// Useful for unit tests.
+class TestEchoServer : public sigslot::has_slots<> {
+ public:
+  TestEchoServer(Thread* thread, const SocketAddress& addr)
+      : server_socket_(thread->socketserver()->CreateAsyncSocket(addr.family(),
+                                                                 SOCK_STREAM)) {
+    server_socket_->Bind(addr);
+    server_socket_->Listen(5);
+    server_socket_->SignalReadEvent.connect(this, &TestEchoServer::OnAccept);
+  }
+  ~TestEchoServer() {
+    for (ClientList::iterator it = client_sockets_.begin();
+         it != client_sockets_.end(); ++it) {
+      delete *it;
+    }
+  }
+
+  SocketAddress address() const { return server_socket_->GetLocalAddress(); }
+
+ private:
+  void OnAccept(AsyncSocket* socket) {
+    AsyncSocket* raw_socket = socket->Accept(nullptr);
+    if (raw_socket) {
+      AsyncTCPSocket* packet_socket = new AsyncTCPSocket(raw_socket, false);
+      packet_socket->SignalReadPacket.connect(this, &TestEchoServer::OnPacket);
+      packet_socket->SignalClose.connect(this, &TestEchoServer::OnClose);
+      client_sockets_.push_back(packet_socket);
+    }
+  }
+  void OnPacket(AsyncPacketSocket* socket, const char* buf, size_t size,
+                const SocketAddress& remote_addr,
+                const PacketTime& packet_time) {
+    rtc::PacketOptions options;
+    socket->Send(buf, size, options);
+  }
+  void OnClose(AsyncPacketSocket* socket, int err) {
+    ClientList::iterator it =
+        std::find(client_sockets_.begin(), client_sockets_.end(), socket);
+    client_sockets_.erase(it);
+    Thread::Current()->Dispose(socket);
+  }
+
+  typedef std::list<AsyncTCPSocket*> ClientList;
+  std::unique_ptr<AsyncSocket> server_socket_;
+  ClientList client_sockets_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(TestEchoServer);
+};
+
+}  // namespace rtc
+
+#endif  // WEBRTC_RTC_BASE_TESTECHOSERVER_H_