Reformat existing code. There should be no functional effects.

webrtc/modules/audio_processing/audio_processing_impl.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 31 matching lines @@
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
 namespace audioproc {
 
 class Event;
 
 }  // namespace audioproc
 #endif
 
 class AudioRate {
  public:
-  explicit AudioRate(int sample_rate_hz)
-      : rate_(sample_rate_hz),
-        samples_per_channel_(AudioProcessing::kChunkSizeMs * rate_ / 1000) {}
+  explicit AudioRate(int sample_rate_hz) { set(sample_rate_hz); }
   virtual ~AudioRate() {}
 
   void set(int rate) {
     rate_ = rate;
     samples_per_channel_ = AudioProcessing::kChunkSizeMs * rate_ / 1000;
   }
 
   int rate() const { return rate_; }
   int samples_per_channel() const { return samples_per_channel_; }
 
@@ skipping 159 matching lines @@
   const bool beamformer_enabled_;
   rtc::scoped_ptr<Beamformer<float>> beamformer_;
   const std::vector<Point> array_geometry_;
 
   const bool supports_48kHz_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_