Minor cosmetic improvements to the new mixer.

webrtc/modules/audio_mixer/source/new_audio_conference_mixer_impl.h

 /*
  *  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.
  */
 
@@ skipping 35 matching lines @@
   // Returns true if the audio source was mixed previous mix
   // iteration.
   bool WasMixed() const;
 
   // Updates the mixed status.
   int32_t SetIsMixed(bool mixed);
 
   void ResetMixedStatus();
 
  private:
-  bool _isMixed;
+  bool is_mixed_;
 };
 
 class NewAudioConferenceMixerImpl : public NewAudioConferenceMixer {
  public:
   // AudioProcessing only accepts 10 ms frames.
   enum { kProcessPeriodicityInMs = 10 };
 
   explicit NewAudioConferenceMixerImpl(int id);
 
   // Must be called after ctor.
@@ skipping 49 matching lines @@
                              bool use_limiter);
 
   // Mix the AudioFrames stored in audioFrameList into mixedAudio. No
   // record will be kept of this mix (e.g. the corresponding MixerAudioSources
   // will not be marked as IsMixed()
   int32_t MixAnonomouslyFromList(AudioFrame* mixedAudio,
                                  const AudioFrameList& audioFrameList) const;
 
   bool LimitMixedAudio(AudioFrame* mixedAudio) const;
 
-  std::unique_ptr<CriticalSectionWrapper> _crit;
-  std::unique_ptr<CriticalSectionWrapper> _cbCrit;
+  std::unique_ptr<CriticalSectionWrapper> crit_;
+  std::unique_ptr<CriticalSectionWrapper> cb_crit_;
 
-  int32_t _id;
+  int32_t id_;
 
   // The current sample frequency and sample size when mixing.
-  Frequency _outputFrequency;
-  size_t _sampleSize;
+  Frequency output_frequency_;
+  size_t sample_size_;
 
   // List of all audio sources. Note all lists are disjunct
   MixerAudioSourceList audio_source_list_;  // May be mixed.
 
   // Always mixed, anonomously.
   MixerAudioSourceList additional_audio_source_list_;
 
   size_t num_mixed_audio_sources_;
   // Determines if we will use a limiter for clipping protection during
   // mixing.
   bool use_limiter_;
 
-  uint32_t _timeStamp;
+  uint32_t time_stamp_;
 
   // Ensures that Mix is called from the same thread.
   rtc::ThreadChecker thread_checker_;
 
   // Used for inhibiting saturation in mixing.
-  std::unique_ptr<AudioProcessing> _limiter;
+  std::unique_ptr<AudioProcessing> limiter_;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_MIXER_SOURCE_NEW_AUDIO_CONFERENCE_MIXER_IMPL_H_