Move webrtc/{base => rtc_base}

webrtc/base/asynctcpsocket_unittest.cc

Index: webrtc/base/asynctcpsocket_unittest.cc
diff --git a/webrtc/base/asynctcpsocket_unittest.cc b/webrtc/base/asynctcpsocket_unittest.cc
deleted file mode 100644
index bd7dcfbdbf0ff1f296b5a836ae83b8d86c790a27..0000000000000000000000000000000000000000
--- a/webrtc/base/asynctcpsocket_unittest.cc
+++ /dev/null
@@ -1,50 +0,0 @@
-/*
- *  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.
- */
-
-#include <memory>
-#include <string>
-
-#include "webrtc/base/asynctcpsocket.h"
-#include "webrtc/base/gunit.h"
-#include "webrtc/base/virtualsocketserver.h"
-
-namespace rtc {
-
-class AsyncTCPSocketTest
-    : public testing::Test,
-      public sigslot::has_slots<> {
- public:
-  AsyncTCPSocketTest()
-      : vss_(new rtc::VirtualSocketServer()),
-        socket_(vss_->CreateAsyncSocket(SOCK_STREAM)),
-        tcp_socket_(new AsyncTCPSocket(socket_, true)),
-        ready_to_send_(false) {
-    tcp_socket_->SignalReadyToSend.connect(this,
-                                           &AsyncTCPSocketTest::OnReadyToSend);
-  }
-
-  void OnReadyToSend(rtc::AsyncPacketSocket* socket) {
-    ready_to_send_ = true;
-  }
-
- protected:
-  std::unique_ptr<VirtualSocketServer> vss_;
-  AsyncSocket* socket_;
-  std::unique_ptr<AsyncTCPSocket> tcp_socket_;
-  bool ready_to_send_;
-};
-
-TEST_F(AsyncTCPSocketTest, OnWriteEvent) {
-  EXPECT_FALSE(ready_to_send_);
-  socket_->SignalWriteEvent(socket_);
-  EXPECT_TRUE(ready_to_send_);
-}
-
-}  // namespace rtc