Make use of new APM statistics interface.

webrtc/modules/audio_processing/include/audio_processing.h

 /*
  *  Copyright (c) 2012 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 468 matching lines @@
     int average = 0;  // Long-term average.
     int maximum = 0;  // Long-term maximum.
     int minimum = 0;  // Long-term minimum.
   };
 
   struct Stat {
     void Set(const Statistic& other) {
       Set(other.instant, other.average, other.maximum, other.minimum);
     }
     void Set(float instant, float average, float maximum, float minimum) {
-      RTC_DCHECK_LE(instant, maximum);
-      RTC_DCHECK_GE(instant, minimum);
-      RTC_DCHECK_LE(average, maximum);
-      RTC_DCHECK_GE(average, minimum);
       instant_ = instant;
       average_ = average;
       maximum_ = maximum;
       minimum_ = minimum;
     }
     float instant() const { return instant_; }
     float average() const { return average_; }
     float maximum() const { return maximum_; }
     float minimum() const { return minimum_; }
 
@@ skipping 569 matching lines @@
   // This does not impact the size of frames passed to |ProcessStream()|.
   virtual int set_frame_size_ms(int size) = 0;
   virtual int frame_size_ms() const = 0;
 
  protected:
   virtual ~VoiceDetection() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_