Reformat existing code. There should be no functional effects.

webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h

 /*
  *  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 209 matching lines @@
    *                            in samples.
    *      - arr_ts            : the arrival time of the packet (from NetEq)
    *                            in samples.
    *
    * Return value             :  0 - Ok
    *                            -1 - Error
    */
 
   int16_t WebRtcIsacfix_UpdateBwEstimate(ISACFIX_MainStruct *ISAC_main_inst,
                                          const uint8_t* encoded,
-                                         int32_t          packet_size,
-                                         uint16_t         rtp_seq_number,
-                                         uint32_t         send_ts,
-                                         uint32_t         arr_ts);
+                                         int32_t packet_size,
+                                         uint16_t rtp_seq_number,
+                                         uint32_t send_ts,
+                                         uint32_t arr_ts);
 
   /****************************************************************************
    * WebRtcIsacfix_Decode(...)
    *
    * This function decodes an ISAC frame. Output speech length
    * will be a multiple of 480 samples: 480 or 960 samples,
    * depending on the framesize (30 or 60 ms).
    *
    * Input:
    *      - ISAC_main_inst    : ISAC instance.
@@ skipping 383 matching lines @@
   int16_t WebRtcIsacfix_GetNewFrameLen(ISACFIX_MainStruct *ISAC_main_inst);
 
 
 #if defined(__cplusplus)
 }
 #endif
 
 
 
 #endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_ISACFIX_H_ */