Reformat existing code. There should be no functional effects.

webrtc/modules/audio_device/android/audio_common.h

 /*
  *  Copyright (c) 2013 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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_COMMON_H_
 #define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_COMMON_H_
 
 namespace webrtc {
 
 enum {
   kDefaultSampleRate = 44100,
   kNumChannels = 1,
-  kDefaultBufSizeInSamples = kDefaultSampleRate * 10 / 1000,
   // Number of bytes per audio frame.
   // Example: 16-bit PCM in mono => 1*(16/8)=2 [bytes/frame]
   kBytesPerFrame = kNumChannels * (16 / 8),
   // Delay estimates for the two different supported modes. These values
   // are based on real-time round-trip delay estimates on a large set of
   // devices and they are lower bounds since the filter length is 128 ms,
   // so the AEC works for delays in the range [50, ~170] ms and [150, ~270] ms.
   // Note that, in most cases, the lowest delay estimate will not be utilized
   // since devices that support low-latency output audio often supports
   // HW AEC as well.
   kLowLatencyModeDelayEstimateInMilliseconds = 50,
   kHighLatencyModeDelayEstimateInMilliseconds = 150,
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_COMMON_H_