Update audio code to use size_t more correctly,

webrtc/modules/audio_processing/vad/vad_audio_proc.h

Index: webrtc/modules/audio_processing/vad/vad_audio_proc.h
diff --git a/webrtc/modules/audio_processing/vad/vad_audio_proc.h b/webrtc/modules/audio_processing/vad/vad_audio_proc.h
index 6cf3937f79f6e290af482625665ebe8fbbb23179..85500aed845f117d6ec8dac9d47fd8dcdbce38a1 100644
--- a/webrtc/modules/audio_processing/vad/vad_audio_proc.h
+++ b/webrtc/modules/audio_processing/vad/vad_audio_proc.h
@@ -30,46 +30,51 @@ class VadAudioProc {
   ~VadAudioProc();
 
   int ExtractFeatures(const int16_t* audio_frame,
-                      int length,
+                      size_t length,
                       AudioFeatures* audio_features);
 
-  static const int kDftSize = 512;
+  static const size_t kDftSize = 512;
 
  private:
-  void PitchAnalysis(double* pitch_gains, double* pitch_lags_hz, int length);
-  void SubframeCorrelation(double* corr, int length_corr, int subframe_index);
-  void GetLpcPolynomials(double* lpc, int length_lpc);
-  void FindFirstSpectralPeaks(double* f_peak, int length_f_peak);
-  void Rms(double* rms, int length_rms);
+  void PitchAnalysis(double* pitch_gains, double* pitch_lags_hz, size_t length);
+  void SubframeCorrelation(double* corr,
+                           size_t length_corr,
+                           size_t subframe_index);
+  void GetLpcPolynomials(double* lpc, size_t length_lpc);
+  void FindFirstSpectralPeaks(double* f_peak, size_t length_f_peak);
+  void Rms(double* rms, size_t length_rms);
   void ResetBuffer();
 
   // To compute spectral peak we perform LPC analysis to get spectral envelope.
   // For every 30 ms we compute 3 spectral peak there for 3 LPC analysis.
   // LPC is computed over 15 ms of windowed audio. For every 10 ms sub-frame
   // we need 5 ms of past signal to create the input of LPC analysis.
-  static const int kNumPastSignalSamples = kSampleRateHz / 200;
+  static const size_t kNumPastSignalSamples =
+      static_cast<size_t>(kSampleRateHz / 200);
 
   // TODO(turajs): maybe defining this at a higher level (maybe enum) so that
   // all the code recognize it as "no-error."
   static const int kNoError = 0;
 
-  static const int kNum10msSubframes = 3;
-  static const int kNumSubframeSamples = kSampleRateHz / 100;
-  static const int kNumSamplesToProcess =
+  static const size_t kNum10msSubframes = 3;
+  static const size_t kNumSubframeSamples =
+      static_cast<size_t>(kSampleRateHz / 100);
+  static const size_t kNumSamplesToProcess =
       kNum10msSubframes *
       kNumSubframeSamples;  // Samples in 30 ms @ given sampling rate.
-  static const int kBufferLength = kNumPastSignalSamples + kNumSamplesToProcess;
-  static const int kIpLength = kDftSize >> 1;
-  static const int kWLength = kDftSize >> 1;
+  static const size_t kBufferLength =
+      kNumPastSignalSamples + kNumSamplesToProcess;
+  static const size_t kIpLength = kDftSize >> 1;
+  static const size_t kWLength = kDftSize >> 1;
 
-  static const int kLpcOrder = 16;
+  static const size_t kLpcOrder = 16;
 
-  int ip_[kIpLength];

[aluebs-webrtc, 2015/07/17 01:04:42]

This is a helper array for the fft4g, which is actually an int*, so I think it
should remain as it is, unless there is a good reason for changing this as well.

[Peter Kasting, 2015/07/17 18:57:40]

This is changing in sync with that -- see
https://codereview.webrtc.org/1227203003/diff/1/webrtc/common_audio/fft4g.h .

[aluebs-webrtc, 2015/07/17 23:16:58]

Oh, I see. I was not aware of that. Then it looks good.

+  size_t ip_[kIpLength];
   float w_fft_[kWLength];
 
   // A buffer of 5 ms (past audio) + 30 ms (one iSAC frame ).
   float audio_buffer_[kBufferLength];
-  int num_buffer_samples_;
+  size_t num_buffer_samples_;
 
   double log_old_gain_;
   double old_lag_;