Rewrite of sigslot that avoids vtables.

webrtc/media/sctp/sctptransport_unittest.cc

 /*
  *  Copyright (c) 2013 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.
  */
 
@@ skipping 32 matching lines @@
 class SctpFakeDataReceiver : public sigslot::has_slots<> {
  public:
   SctpFakeDataReceiver() : received_(false) {}
 
   void Clear() {
     received_ = false;
     last_data_ = "";
     last_params_ = ReceiveDataParams();
   }
 
-  virtual void OnDataReceived(const ReceiveDataParams& params,
-                              const rtc::CopyOnWriteBuffer& data) {
+  void OnDataReceived(const ReceiveDataParams& params,
+                      const rtc::CopyOnWriteBuffer& data) {
     received_ = true;
     last_data_ = std::string(data.data<char>(), data.size());
     last_params_ = params;
   }
 
   bool received() const { return received_; }
   std::string last_data() const { return last_data_; }
   ReceiveDataParams last_params() const { return last_params_; }
 
  private:
@@ skipping 489 matching lines @@
   AddStream(1);
   ASSERT_TRUE(SendData(transport1(), 1, "hi?", &result));
   EXPECT_EQ(SDR_SUCCESS, result);
   EXPECT_TRUE_WAIT(ReceivedData(receiver2(), 1, "hi?"), kDefaultTimeout);
   transport1()->ResetStream(1);
   EXPECT_TRUE_WAIT(transport2_sig_receiver.StreamCloseCount(1) == 2,
                    kDefaultTimeout);
 }
 
 }  // namespace cricket