Enable -Winconsistent-missing-override flag.

webrtc/call/call_perf_tests.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 395 matching lines @@
       int time_offset_ms = real_capture_timestamp - estimated_capture_timestamp;
       time_offset_ms = time_offset_ms / 90;
       std::stringstream ss;
       ss << time_offset_ms;
 
       webrtc::test::PrintResult(
           "capture_ntp_time", "", "real - estimated", ss.str(), "ms", true);
       EXPECT_TRUE(std::abs(time_offset_ms) < threshold_ms_);
     }
 
-    virtual Action OnSendRtp(const uint8_t* packet, size_t length) {
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
       rtc::CritScope lock(&crit_);
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
 
       if (!rtp_start_timestamp_set_) {
         // Calculate the rtp timestamp offset in order to calculate the real
         // capture time.
         uint32_t first_capture_timestamp =
             90 * static_cast<uint32_t>(capturer_->first_frame_capture_time());
         rtp_start_timestamp_ = header.timestamp - first_capture_timestamp;
@@ skipping 284 matching lines @@
     int encoder_inits_;
     uint32_t last_set_bitrate_;
     VideoSendStream* send_stream_;
     VideoEncoderConfig encoder_config_;
   } test;
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc