Miscellaneous changes split from https://codereview.webrtc.org/1230503003 .

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

 /*
  *  Copyright (c) 2014 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 97 matching lines @@
 }
 
 int AudioEncoderOpus::NumChannels() const {
   return num_channels_;
 }
 
 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.
   size_t bytes_per_millisecond =
-       static_cast<size_t>(bitrate_bps_ / (1000 * 8) + 1);
+      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;
 }
 
 int AudioEncoderOpus::Num10MsFramesInNextPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
 int AudioEncoderOpus::Max10MsFramesInAPacket() const {
@@ skipping 131 matching lines @@
   return Reconstruct(conf);
 }
 
 bool AudioEncoderMutableOpus::SetMaxPlaybackRate(int frequency_hz) {
   auto conf = config();
   conf.max_playback_rate_hz = frequency_hz;
   return Reconstruct(conf);
 }
 
 }  // namespace webrtc