Reformat existing code. There should be no functional effects.

webrtc/modules/audio_coding/codecs/g711/g711_interface.c

 /*
  *  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.
  */
 #include <string.h>
@@ skipping 34 matching lines @@
                            int16_t len,
                            int16_t* decoded,
                            int16_t* speechType) {
   int n;
   for (n = 0; n < len; n++)
     decoded[n] = ulaw_to_linear(encoded[n]);
   *speechType = 1;
   return len;
 }
 
-int WebRtcG711_DurationEst(const uint8_t* payload,
-                           int payload_length_bytes) {
-  (void) payload;
-  /* G.711 is one byte per sample, so we can just return the number of bytes. */
-  return payload_length_bytes;
-}
-
 int16_t WebRtcG711_Version(char* version, int16_t lenBytes) {
   strncpy(version, "2.0.0", lenBytes);
   return 0;
 }