Enable -Winconsistent-missing-override flag.

webrtc/media/engine/fakewebrtcvoiceengine.h

 /*
  *  Copyright (c) 2010 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 292 matching lines @@
   }
   WEBRTC_STUB(StopSend, (int channel));
   WEBRTC_STUB(GetVersion, (char version[1024]));
   WEBRTC_STUB(LastError, ());
   WEBRTC_FUNC(AssociateSendChannel, (int channel,
                                      int accociate_send_channel)) {
     WEBRTC_CHECK_CHANNEL(channel);
     channels_[channel]->associate_send_channel = accociate_send_channel;
     return 0;
   }
-  webrtc::RtcEventLog* GetEventLog() { return nullptr; }
+  webrtc::RtcEventLog* GetEventLog() override { return nullptr; }
 
   // webrtc::VoECodec
   WEBRTC_STUB(NumOfCodecs, ());
   WEBRTC_STUB(GetCodec, (int index, webrtc::CodecInst& codec));
   WEBRTC_FUNC(SetSendCodec, (int channel, const webrtc::CodecInst& codec)) {
     WEBRTC_CHECK_CHANNEL(channel);
     // To match the behavior of the real implementation.
     if (_stricmp(codec.plname, "telephone-event") == 0 ||
         _stricmp(codec.plname, "audio/telephone-event") == 0 ||
         _stricmp(codec.plname, "CN") == 0 ||
@@ skipping 130 matching lines @@
   WEBRTC_STUB(GetPlayoutDeviceName, (int i, char* name, char* guid));
   WEBRTC_STUB(SetRecordingDevice, (int, webrtc::StereoChannel));
   WEBRTC_STUB(SetPlayoutDevice, (int));
   WEBRTC_STUB(SetAudioDeviceLayer, (webrtc::AudioLayers));
   WEBRTC_STUB(GetAudioDeviceLayer, (webrtc::AudioLayers&));
   WEBRTC_STUB(SetRecordingSampleRate, (unsigned int samples_per_sec));
   WEBRTC_STUB_CONST(RecordingSampleRate, (unsigned int* samples_per_sec));
   WEBRTC_STUB(SetPlayoutSampleRate, (unsigned int samples_per_sec));
   WEBRTC_STUB_CONST(PlayoutSampleRate, (unsigned int* samples_per_sec));
   WEBRTC_STUB(EnableBuiltInAEC, (bool enable));
-  virtual bool BuiltInAECIsAvailable() const { return false; }
+  bool BuiltInAECIsAvailable() const override { return false; }
   WEBRTC_STUB(EnableBuiltInAGC, (bool enable));
-  virtual bool BuiltInAGCIsAvailable() const { return false; }
+  bool BuiltInAGCIsAvailable() const override { return false; }
   WEBRTC_STUB(EnableBuiltInNS, (bool enable));
-  virtual bool BuiltInNSIsAvailable() const { return false; }
+  bool BuiltInNSIsAvailable() const override { return false; }
 
   // webrtc::VoENetwork
   WEBRTC_FUNC(RegisterExternalTransport, (int channel,
                                           webrtc::Transport& transport)) {
     WEBRTC_CHECK_CHANNEL(channel);
     channels_[channel]->external_transport = true;
     return 0;
   }
   WEBRTC_FUNC(DeRegisterExternalTransport, (int channel)) {
     WEBRTC_CHECK_CHANNEL(channel);
@@ skipping 187 matching lines @@
     enabled = typing_detection_enabled_;
     return 0;
   }
 
   WEBRTC_STUB(TimeSinceLastTyping, (int& seconds));
   WEBRTC_STUB(SetTypingDetectionParameters, (int timeWindow,
                                              int costPerTyping,
                                              int reportingThreshold,
                                              int penaltyDecay,
                                              int typeEventDelay));
-  int EnableHighPassFilter(bool enable) {
+  int EnableHighPassFilter(bool enable) override {
     highpass_filter_enabled_ = enable;
     return 0;
   }
-  bool IsHighPassFilterEnabled() {
+  bool IsHighPassFilterEnabled() override {
     return highpass_filter_enabled_;
   }
-  bool IsStereoChannelSwappingEnabled() {
+  bool IsStereoChannelSwappingEnabled() override {
     return stereo_swapping_enabled_;
   }
-  void EnableStereoChannelSwapping(bool enable) {
+  void EnableStereoChannelSwapping(bool enable) override {
     stereo_swapping_enabled_ = enable;
   }
   int GetNetEqCapacity() const {
     auto ch = channels_.find(last_channel_);
     ASSERT(ch != channels_.end());
     return ch->second->neteq_capacity;
   }
   bool GetNetEqFastAccelerate() const {
     auto ch = channels_.find(last_channel_);
     ASSERT(ch != channels_.end());
@@ skipping 19 matching lines @@
   webrtc::NsModes ns_mode_ = webrtc::kNsDefault;
   webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault;
   webrtc::AgcConfig agc_config_;
   int playout_fail_channel_ = -1;
   FakeAudioProcessing audio_processing_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_