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

webrtc/common_audio/vad/vad_sp.h

 /*
  *  Copyright (c) 2011 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 15 matching lines @@
 // Input & Output:
 //      - filter_state  : Current filter states of the two all-pass filters. The
 //                        |filter_state| is updated after all samples have been
 //                        processed.
 //
 // Output:
 //      - signal_out    : Downsampled signal (of length |in_length| / 2).
 void WebRtcVad_Downsampling(const int16_t* signal_in,
                             int16_t* signal_out,
                             int32_t* filter_state,
-                            int in_length);
+                            size_t in_length);
 
 // Updates and returns the smoothed feature minimum. As minimum we use the
 // median of the five smallest feature values in a 100 frames long window.
 // As long as |handle->frame_counter| is zero, that is, we haven't received any
 // "valid" data, FindMinimum() outputs the default value of 1600.
 //
 // Inputs:
 //      - feature_value : New feature value to update with.
 //      - channel       : Channel number.
 //
 // Input & Output:
 //      - handle        : State information of the VAD.
 //
 // Returns:
 //                      : Smoothed minimum value for a moving window.
 int16_t WebRtcVad_FindMinimum(VadInstT* handle,
                               int16_t feature_value,
                               int channel);
 
 #endif  // WEBRTC_COMMON_AUDIO_VAD_VAD_SP_H_