Fix chromium-style errors in IntelligibilityEnhancer

webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h

 /*
  *  Copyright (c) 2014 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 17 matching lines @@
 // audio streams and modifies the render stream with a set of gains per
 // frequency bin to enhance speech against the noise background.
 // Details of the model and algorithm can be found in the original paper:
 // http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=6882788
 class IntelligibilityEnhancer : public LappedTransform::Callback {
  public:
   IntelligibilityEnhancer(int sample_rate_hz,
                           size_t num_render_channels,
                           size_t num_noise_bins);
 
-  ~IntelligibilityEnhancer();
+  ~IntelligibilityEnhancer() override;
 
   // Sets the capture noise magnitude spectrum estimate.
   void SetCaptureNoiseEstimate(std::vector<float> noise, float gain);
 
   // Reads chunk of speech in time domain and updates with modified signal.
   void ProcessRenderAudio(float* const* audio,
                           int sample_rate_hz,
                           size_t num_channels);
   bool active() const;
 
@@ skipping 69 matching lines @@
   unsigned long int num_active_chunks_;
 
   std::vector<float> noise_estimation_buffer_;
   SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>
       noise_estimation_queue_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_INTELLIGIBILITY_INTELLIGIBILITY_ENHANCER_H_