Enable -Winconsistent-missing-override flag.

webrtc/video/video_send_stream_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.
  */
 #include <algorithm>  // max
@@ skipping 690 matching lines @@
 
         rtcp_sender.SetRTCPStatus(RtcpMode::kReducedSize);
         rtcp_sender.SetRemoteSSRC(kVideoSendSsrcs[0]);
 
         RTCPSender::FeedbackState feedback_state;
 
         EXPECT_EQ(0, rtcp_sender.SendRTCP(feedback_state, kRtcpRr));
       }
     }
 
-    virtual void EncodedFrameCallback(const EncodedFrame& encoded_frame) {
+    void EncodedFrameCallback(const EncodedFrame& encoded_frame) override {
       // Increase frame size for next encoded frame, in the context of the
       // encoder thread.
       if (!use_fec_ &&
           current_size_frame_.Value() < static_cast<int32_t>(stop_size_)) {
         ++current_size_frame_;
       }
       encoder_.SetFrameSize(static_cast<size_t>(current_size_frame_.Value()));
     }
 
     Call::Config GetSenderCallConfig() override {
@@ skipping 270 matching lines @@
       test::PacketTransport* transport = new test::PacketTransport(
           nullptr, this, test::PacketTransport::kReceiver,
           FakeNetworkPipe::Config());
       transport_adapter_.reset(new internal::TransportAdapter(transport));
       transport_adapter_->Enable();
       return transport;
     }
 
     size_t GetNumVideoStreams() const override { return 3; }
 
-    virtual void OnFrameGeneratorCapturerCreated(
-        test::FrameGeneratorCapturer* frame_generator_capturer) {
+    void OnFrameGeneratorCapturerCreated(
+        test::FrameGeneratorCapturer* frame_generator_capturer) override {
       rtc::CritScope lock(&crit_);
       capturer_ = frame_generator_capturer;
     }
 
     void PerformTest() override {
       EXPECT_TRUE(Wait())
           << "Timed out while waiting for RTP packets to stop being sent.";
     }
 
     Clock* const clock_;
@@ skipping 19 matching lines @@
   static const int kRembBitrateBps = 80000;
   static const int kRembRespectedBitrateBps = 100000;
   class BitrateObserver : public test::SendTest {
    public:
     BitrateObserver()
         : SendTest(kDefaultTimeoutMs),
           bitrate_capped_(false) {
     }
 
    private:
-    virtual Action OnSendRtp(const uint8_t* packet, size_t length) {
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
       if (RtpHeaderParser::IsRtcp(packet, length))
         return DROP_PACKET;
 
       RTPHeader header;
       if (!parser_->Parse(packet, length, &header))
         return DROP_PACKET;
       RTC_DCHECK(stream_);
       VideoSendStream::Stats stats = stream_->GetStats();
       if (!stats.substreams.empty()) {
         EXPECT_EQ(1u, stats.substreams.size());
@@ skipping 1238 matching lines @@
         observation_complete_.Set();
       }
     }
   } test;
 
   RunBaseTest(&test);
 }
 #endif
 
 }  // namespace webrtc