Reformat existing code. There should be no functional effects.

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

Index: webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
index 11dd20a8f9a375b2a5a0ba93e8023a8e5cea04d7..6bcd717279b8b5bf4ce8423eaec62b825908997c 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
@@ -552,8 +552,8 @@ int main(int argc, char* argv[]) {
 
     // 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;