Reformat existing code. There should be no functional effects.

webrtc/common_audio/resampler/resampler.cc

 /*
  *  Copyright (c) 2011 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 822 matching lines @@
               // data is now actually in 96 kHz sampling rate, despite the
               // function name, and with a resampling factor of 1/3 becomes
               // 32 kHz.
               WebRtcSpl_Resample48khzTo16khz(samplesIn + i, tmp + i / 3,
                                              (WebRtcSpl_State48khzTo16khz*) state1_,
                                              tmp_mem);
             }
             outLen = lengthIn / 3;
             free(tmp_mem);
             // 4:2
-            WebRtcSpl_DownsampleBy2(tmp, outLen, tmp_2,
-                                    (int32_t*) state2_);
+            WebRtcSpl_DownsampleBy2(tmp, outLen, tmp_2, (int32_t*) state2_);
             outLen = outLen / 2;
             free(tmp);
             // 2:1
             WebRtcSpl_DownsampleBy2(tmp_2, outLen, samplesOut,
                                     (int32_t*) state3_);
             free(tmp_2);
             outLen = outLen / 2;
             return 0;
         case kResamplerMode3To2:
             if (maxLen < (lengthIn * 2 / 3))
@@ skipping 38 matching lines @@
             tmp = (int16_t*)malloc((lengthIn * 4) / 11 * sizeof(int16_t));
 
             for (int i = 0; i < lengthIn; i += 220)
             {
                 WebRtcSpl_Resample22khzTo8khz(samplesIn + i, tmp + (i * 4) / 11,
                                               (WebRtcSpl_State22khzTo8khz *)state1_,
                                               tmp_mem);
             }
             lengthIn = (lengthIn * 4) / 11;
 
-            WebRtcSpl_DownsampleBy2(tmp, lengthIn, samplesOut, (int32_t*)state2_);
+            WebRtcSpl_DownsampleBy2(tmp, lengthIn, samplesOut,
+                                    (int32_t*)state2_);
             outLen = lengthIn / 2;
 
             free(tmp_mem);
             free(tmp);
             return 0;
         case kResamplerMode11To4:
             // We can only handle blocks of 220 samples
             // Can be fixed, but I don't think it's needed
             if ((lengthIn % 220) != 0)
             {
@@ skipping 36 matching lines @@
             outLen = (lengthIn * 8) / 11;
             free(tmp_mem);
             return 0;
             break;
 
     }
     return 0;
 }
 
 }  // namespace webrtc