Reformat existing code. There should be no functional effects.

webrtc/modules/audio_processing/audio_buffer.cc

 /*
  *  Copyright (c) 2012 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.
  */
 
 #include "webrtc/modules/audio_processing/audio_buffer.h"
 
 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/common_audio/resampler/push_sinc_resampler.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/modules/audio_processing/common.h"
 
 namespace webrtc {
 namespace {
 
-enum {
-  kSamplesPer16kHzChannel = 160,
-  kSamplesPer32kHzChannel = 320,
-  kSamplesPer48kHzChannel = 480
-};
+const int kSamplesPer16kHzChannel = 160;
+const int kSamplesPer32kHzChannel = 320;
+const int kSamplesPer48kHzChannel = 480;
 
 bool HasKeyboardChannel(AudioProcessing::ChannelLayout layout) {
   switch (layout) {
     case AudioProcessing::kMono:
     case AudioProcessing::kStereo:
       return false;
     case AudioProcessing::kMonoAndKeyboard:
     case AudioProcessing::kStereoAndKeyboard:
       return true;
   }
@@ skipping 40 matching lines @@
                          int process_num_frames,
                          int num_process_channels,
                          int output_num_frames)
   : input_num_frames_(input_num_frames),
     num_input_channels_(num_input_channels),
     proc_num_frames_(process_num_frames),
     num_proc_channels_(num_process_channels),
     output_num_frames_(output_num_frames),
     num_channels_(num_process_channels),
     num_bands_(NumBandsFromSamplesPerChannel(proc_num_frames_)),
-    num_split_frames_(rtc::CheckedDivExact(
-        proc_num_frames_, num_bands_)),
+    num_split_frames_(rtc::CheckedDivExact(proc_num_frames_, num_bands_)),
     mixed_low_pass_valid_(false),
     reference_copied_(false),
     activity_(AudioFrame::kVadUnknown),
     keyboard_data_(NULL),
     data_(new IFChannelBuffer(proc_num_frames_, num_proc_channels_)) {
   assert(input_num_frames_ > 0);
   assert(proc_num_frames_ > 0);
   assert(output_num_frames_ > 0);
   assert(num_input_channels_ > 0 && num_input_channels_ <= 2);
   assert(num_proc_channels_ > 0 && num_proc_channels_ <= num_input_channels_);
@@ skipping 293 matching lines @@
   return input_num_frames_;
 }
 
 int AudioBuffer::num_bands() const {
   return num_bands_;
 }
 
 // The resampler is only for supporting 48kHz to 16kHz in the reverse stream.
 void AudioBuffer::DeinterleaveFrom(AudioFrame* frame) {
   assert(frame->num_channels_ == num_input_channels_);
-  assert(frame->samples_per_channel_ ==  input_num_frames_);
+  assert(frame->samples_per_channel_ == input_num_frames_);
   InitForNewData();
   // Initialized lazily because there's a different condition in CopyFrom.
   if ((input_num_frames_ != proc_num_frames_) && !input_buffer_) {
     input_buffer_.reset(
         new IFChannelBuffer(input_num_frames_, num_proc_channels_));
   }
   activity_ = frame->vad_activity_;
 
   int16_t* const* deinterleaved;
   if (input_num_frames_ == proc_num_frames_) {
@@ skipping 60 matching lines @@
 
 void AudioBuffer::SplitIntoFrequencyBands() {
   splitting_filter_->Analysis(data_.get(), split_data_.get());
 }
 
 void AudioBuffer::MergeFrequencyBands() {
   splitting_filter_->Synthesis(split_data_.get(), data_.get());
 }
 
 }  // namespace webrtc