Reformat existing code. There should be no functional effects.

webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc

 /*
  *  Copyright (c) 2013 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 534 matching lines @@
         if (packet_available != next_packet_available) {
           packet_available = !packet_available;
           next_packet_available = !next_packet_available;
         }
       }
       next_input_time_ms = packet->time_ms();
     }
 
     // Check if it is time to get output audio.
     if (time_now_ms >= next_output_time_ms) {
-      static const int kOutDataLen = kOutputBlockSizeMs * kMaxSamplesPerMs *
-          kMaxChannels;
+      static const int kOutDataLen =
+          kOutputBlockSizeMs * kMaxSamplesPerMs * kMaxChannels;
       int16_t out_data[kOutDataLen];
       int num_channels;
       int samples_per_channel;
       int error = neteq->GetAudio(kOutDataLen, out_data, &samples_per_channel,
                                    &num_channels, NULL);
       if (error != NetEq::kOK) {
         std::cerr << "GetAudio returned error code " <<
             neteq->LastError() << std::endl;
       } else {
         // Calculate sample rate from output size.
@@ skipping 14 matching lines @@
     time_now_ms = std::min(next_input_time_ms, next_output_time_ms);
   }
 
   printf("Simulation done\n");
   printf("Produced %i ms of audio\n", time_now_ms - start_time_ms);
 
   delete neteq;
   webrtc::Trace::ReturnTrace();
   return 0;
 }