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

webrtc/modules/audio_processing/vad/voice_activity_detector.h

 /*
  *  Copyright (c) 2015 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 13 matching lines @@
 
 // A Voice Activity Detector (VAD) that combines the voice probability from the
 // StandaloneVad and PitchBasedVad to get a more robust estimation.
 class VoiceActivityDetector {
  public:
   VoiceActivityDetector();
 
   // Processes each audio chunk and estimates the voice probability. The maximum
   // supported sample rate is 32kHz.
   // TODO(aluebs): Change |length| to size_t.
-  void ProcessChunk(const int16_t* audio, int length, int sample_rate_hz);
+  void ProcessChunk(const int16_t* audio, size_t length, int sample_rate_hz);
 
   // Returns a vector of voice probabilities for each chunk. It can be empty for
   // some chunks, but it catches up afterwards returning multiple values at
   // once.
   const std::vector<double>& chunkwise_voice_probabilities() const {
     return chunkwise_voice_probabilities_;
   }
 
   // Returns a vector of RMS values for each chunk. It has the same length as
   // chunkwise_voice_probabilities().
@@ skipping 16 matching lines @@
   rtc::scoped_ptr<StandaloneVad> standalone_vad_;
   PitchBasedVad pitch_based_vad_;
 
   int16_t resampled_[kLength10Ms];
   AudioFeatures features_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_VAD_VOICE_ACTIVITY_DETECTOR_H_