Reformat existing code. There should be no functional effects.

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc

Index: webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index 1eeb5caa9f53795989d2ae38d5bfba7d136c7cd0..4452eefb8d5e89bdc142a9951f9a573652cc6479 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -114,10 +114,9 @@ int AudioEncoderOpus::NumChannels() const {
 size_t AudioEncoderOpus::MaxEncodedBytes() const {
   // Calculate the number of bytes we expect the encoder to produce,
   // then multiply by two to give a wide margin for error.
-  int frame_size_ms = num_10ms_frames_per_packet_ * 10;
   int bytes_per_millisecond = bitrate_bps_ / (1000 * 8) + 1;
-  size_t approx_encoded_bytes =
-      static_cast<size_t>(frame_size_ms * bytes_per_millisecond);
+  size_t approx_encoded_bytes = static_cast<size_t>(
+      num_10ms_frames_per_packet_ * 10 * bytes_per_millisecond);
   return 2 * approx_encoded_bytes;
 }