Update a ton of audio code to use size_t more correctly and in general reduce

webrtc/modules/audio_conference_mixer/source/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 142 matching lines @@
 
     int32_t _id;
 
     Frequency _minimumMixingFreq;
 
     // Mix result callback
     AudioMixerOutputReceiver* _mixReceiver;
 
     // The current sample frequency and sample size when mixing.
     Frequency _outputFrequency;
-    uint16_t _sampleSize;
+    size_t _sampleSize;
 
     // Memory pool to avoid allocating/deallocating AudioFrames
     MemoryPool<AudioFrame>* _audioFramePool;
 
     // List of all participants. Note all lists are disjunct
     MixerParticipantList _participantList;              // May be mixed.
     // Always mixed, anonomously.
     MixerParticipantList _additionalParticipantList;
 
     size_t _numMixedParticipants;
     // Determines if we will use a limiter for clipping protection during
     // mixing.
     bool use_limiter_;
 
     uint32_t _timeStamp;
 
     // Metronome class.
     TimeScheduler _timeScheduler;
 
     // Counter keeping track of concurrent calls to process.
     // Note: should never be higher than 1 or lower than 0.
     int16_t _processCalls;
 
     // Used for inhibiting saturation in mixing.
     rtc::scoped_ptr<AudioProcessing> _limiter;
 };
 }  // namespace webrtc
 
 #endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_CONFERENCE_MIXER_IMPL_H_