Replace Clock with timeutils in AudioEncoder.

webrtc/common_audio/smoothing_filter.h

 /*
  *  Copyright (c) 2016 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 22 matching lines @@
 // assumed to equal the last received sample.
 class SmoothingFilterImpl final : public SmoothingFilter {
  public:
   // |init_time_ms| is initialization time. It defines a period starting from
   // the arriving time of the first sample. During this period, the exponential
   // filter uses a varying time constant so that a smaller time constant will be
   // applied to the earlier samples. This is to allow the the filter to adapt to
   // earlier samples quickly. After the initialization period, the time constant
   // will be set to |init_time_ms| first and can be changed through
   // |SetTimeConstantMs|.
-  SmoothingFilterImpl(int init_time_ms, const Clock* clock);
+  explicit SmoothingFilterImpl(int init_time_ms);
   ~SmoothingFilterImpl() override;
 
   void AddSample(float sample) override;
   rtc::Optional<float> GetAverage() override;
   bool SetTimeConstantMs(int time_constant_ms) override;
 
   // Methods used for unittests.
   float alpha() const { return alpha_; }
 
  private:
   void UpdateAlpha(int time_constant_ms);
   void ExtrapolateLastSample(int64_t time_ms);
 
   const int init_time_ms_;
   const float init_factor_;
   const float init_const_;
-  const Clock* const clock_;
 
   rtc::Optional<int64_t> init_end_time_ms_;
   float last_sample_;
   float alpha_;
   float state_;
   int64_t last_state_time_ms_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(SmoothingFilterImpl);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_COMMON_AUDIO_SMOOTHING_FILTER_H_