Reformat existing code. There should be no functional effects.

webrtc/modules/utility/source/file_recorder_impl.cc

 /*
  *  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 209 matching lines @@
                              (int16_t*)_audioBuffer,
                              MAX_AUDIO_BUFFER_IN_BYTES, outLen);
         encodedLenInBytes = outLen * sizeof(int16_t);
     }
 
     // Codec may not be operating at a frame rate of 10 ms. Whenever enough
     // 10 ms chunks of data has been pushed to the encoder an encoded frame
     // will be available. Wait until then.
     if (encodedLenInBytes)
     {
-        uint16_t msOfData =
-            ptrAudioFrame->samples_per_channel_ /
-            uint16_t(ptrAudioFrame->sample_rate_hz_ / 1000);
-        if (WriteEncodedAudioData(_audioBuffer, encodedLenInBytes, msOfData,
-                                  playoutTS) == -1)
+        if (WriteEncodedAudioData(_audioBuffer, encodedLenInBytes) == -1)
         {
             return -1;
         }
     }
     return 0;
 }
 
 int32_t FileRecorderImpl::SetUpAudioEncoder()
 {
     if (_fileFormat == kFileFormatPreencodedFile ||
@@ skipping 12 matching lines @@
 int32_t FileRecorderImpl::codec_info(CodecInst& codecInst) const
 {
     if(codec_info_.plfreq == 0)
     {
         return -1;
     }
     codecInst = codec_info_;
     return 0;
 }
 
-int32_t FileRecorderImpl::WriteEncodedAudioData(
-    const int8_t* audioBuffer,
-    size_t bufferLength,
-    uint16_t /*millisecondsOfData*/,
-    const TickTime* /*playoutTS*/)
+int32_t FileRecorderImpl::WriteEncodedAudioData(const int8_t* audioBuffer,
+                                                size_t bufferLength)
 {
     return _moduleFile->IncomingAudioData(audioBuffer, bufferLength);
 }
 }  // namespace webrtc