Reformat existing code. There should be no functional effects.

webrtc/modules/audio_processing/aec/aec_resampler.c

 /*
  *  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 56 matching lines @@
                               int size,
                               float skew,
                               float* outspeech,
                               int* size_out) {
   AecResampler* obj = (AecResampler*)resampInst;
 
   float* y;
   float be, tnew;
   int tn, mm;
 
-  assert(!(size < 0 || size > 2 * FRAME_LEN));
+  assert(size >= 0);
+  assert(size <= 2 * FRAME_LEN);
   assert(resampInst != NULL);
   assert(inspeech != NULL);
   assert(outspeech != NULL);
   assert(size_out != NULL);
 
   // Add new frame data in lookahead
   memcpy(&obj->buffer[FRAME_LEN + kResamplingDelay],
          inspeech,
          size * sizeof(inspeech[0]));
 
@@ skipping 112 matching lines @@
   xAvg = x / n;
   denom = x2 - xAvg * x;
 
   if (denom != 0) {
     skew = (xy - xAvg * y) / denom;
   }
 
   *skewEst = skew;
   return 0;
 }