Update audio code to use size_t more correctly,

webrtc/modules/audio_coding/neteq/dsp_helper.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 60 matching lines @@
                         size_t start_index,
                         size_t length,
                         int factor,
                         int increment);
 
   // Peak detection with parabolic fit. Looks for |num_peaks| maxima in |data|,
   // having length |data_length| and sample rate multiplier |fs_mult|. The peak
   // locations and values are written to the arrays |peak_index| and
   // |peak_value|, respectively. Both arrays must hold at least |num_peaks|
   // elements.
-  static void PeakDetection(int16_t* data, int data_length,
-                            int num_peaks, int fs_mult,
-                            int* peak_index, int16_t* peak_value);
+  static void PeakDetection(int16_t* data, size_t data_length,
+                            size_t num_peaks, size_t fs_mult,

[hlundin-webrtc, 2015/08/10 11:30:01]

Again, fs_mult should not be a size_t.

+                            size_t* peak_index, int16_t* peak_value);
 
   // Estimates the height and location of a maximum. The three values in the
   // array |signal_points| are used as basis for a parabolic fit, which is then
   // used to find the maximum in an interpolated signal. The |signal_points| are
   // assumed to be from a 4 kHz signal, while the maximum, written to
   // |peak_index| and |peak_value| is given in the full sample rate, as
   // indicated by the sample rate multiplier |fs_mult|.
-  static void ParabolicFit(int16_t* signal_points, int fs_mult,
-                           int* peak_index, int16_t* peak_value);
+  static void ParabolicFit(int16_t* signal_points, size_t fs_mult,

[hlundin-webrtc, 2015/08/10 11:30:00]

And here.

+                           size_t* peak_index, int16_t* peak_value);
 
   // Calculates the sum-abs-diff for |signal| when compared to a displaced
   // version of itself. Returns the displacement lag that results in the minimum
   // distortion. The resulting distortion is written to |distortion_value|.
   // The values of |min_lag| and |max_lag| are boundaries for the search.
-  static int MinDistortion(const int16_t* signal, int min_lag,
-                           int max_lag, int length, int32_t* distortion_value);
+  static size_t MinDistortion(const int16_t* signal, size_t min_lag,
+                           size_t max_lag, size_t length,
+                           int32_t* distortion_value);
 
   // Mixes |length| samples from |input1| and |input2| together and writes the
   // result to |output|. The gain for |input1| starts at |mix_factor| (Q14) and
   // is decreased by |factor_decrement| (Q14) for each sample. The gain for
   // |input2| is the complement 16384 - mix_factor.
   static void CrossFade(const int16_t* input1, const int16_t* input2,
                         size_t length, int16_t* mix_factor,
                         int16_t factor_decrement, int16_t* output);
 
   // Scales |input| with an increasing gain. Applies |factor| (Q14) to the first
   // sample and increases the gain by |increment| (Q20) for each sample. The
   // result is written to |output|. |length| samples are processed.
   static void UnmuteSignal(const int16_t* input, size_t length, int16_t* factor,
                            int increment, int16_t* output);
 
   // Starts at unity gain and gradually fades out |signal|. For each sample,
   // the gain is reduced by |mute_slope| (Q14). |length| samples are processed.
   static void MuteSignal(int16_t* signal, int mute_slope, size_t length);
 
   // Downsamples |input| from |sample_rate_hz| to 4 kHz sample rate. The input
   // has |input_length| samples, and the method will write |output_length|
   // samples to |output|. Compensates for the phase delay of the downsampling
   // filters if |compensate_delay| is true. Returns -1 if the input is too short
   // to produce |output_length| samples, otherwise 0.
   static int DownsampleTo4kHz(const int16_t* input, size_t input_length,
-                              int output_length, int input_rate_hz,
+                              size_t output_length, int input_rate_hz,
                               bool compensate_delay, int16_t* output);
 
  private:
   // Table of constants used in method DspHelper::ParabolicFit().
   static const int16_t kParabolaCoefficients[17][3];
 
   DISALLOW_COPY_AND_ASSIGN(DspHelper);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_