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 e69b0c8fb3917be38b0c27851d763a4b77f8043e..17fa5b24b8ccdc97e3a096331ef01af82dfc6573 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 +114,10 @@ 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;
   size_t bytes_per_millisecond =
-      static_cast<size_t>(bitrate_bps_ / (1000 * 8) + 1);
-  size_t approx_encoded_bytes = frame_size_ms * bytes_per_millisecond;
+       static_cast<size_t>(bitrate_bps_ / (1000 * 8) + 1);
+  size_t approx_encoded_bytes =
+      num_10ms_frames_per_packet_ * 10 * bytes_per_millisecond;
   return 2 * approx_encoded_bytes;
 }