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

webrtc/modules/audio_coding/neteq/dsp_helper.cc

Index: webrtc/modules/audio_coding/neteq/dsp_helper.cc
diff --git a/webrtc/modules/audio_coding/neteq/dsp_helper.cc b/webrtc/modules/audio_coding/neteq/dsp_helper.cc
index 3e5c61d87b559586b889ef0826527c278c677ecd..4188914c86c9bec701c40d92de3ee11be43569bc 100644
--- a/webrtc/modules/audio_coding/neteq/dsp_helper.cc
+++ b/webrtc/modules/audio_coding/neteq/dsp_helper.cc
@@ -99,13 +99,13 @@ int DspHelper::RampSignal(AudioMultiVector* signal,
   return end_factor;
 }
 
-void DspHelper::PeakDetection(int16_t* data, int data_length,
-                              int num_peaks, int fs_mult,
-                              int* peak_index, int16_t* peak_value) {
-  int16_t min_index = 0;
-  int16_t max_index = 0;
+void DspHelper::PeakDetection(int16_t* data, size_t data_length,
+                              size_t num_peaks, int fs_mult,
+                              size_t* peak_index, int16_t* peak_value) {
+  size_t min_index = 0;
+  size_t max_index = 0;
 
-  for (int i = 0; i <= num_peaks - 1; i++) {
+  for (size_t i = 0; i <= num_peaks - 1; i++) {
     if (num_peaks == 1) {
       // Single peak.  The parabola fit assumes that an extra point is
       // available; worst case it gets a zero on the high end of the signal.
@@ -148,7 +148,7 @@ void DspHelper::PeakDetection(int16_t* data, int data_length,
 }
 
 void DspHelper::ParabolicFit(int16_t* signal_points, int fs_mult,
-                             int* peak_index, int16_t* peak_value) {
+                             size_t* peak_index, int16_t* peak_value) {
   uint16_t fit_index[13];
   if (fs_mult == 1) {
     fit_index[0] = 0;
@@ -235,16 +235,16 @@ void DspHelper::ParabolicFit(int16_t* signal_points, int fs_mult,
   }
 }
 
-int DspHelper::MinDistortion(const int16_t* signal, int min_lag,
-                             int max_lag, int length,
-                             int32_t* distortion_value) {
-  int best_index = 0;
+size_t DspHelper::MinDistortion(const int16_t* signal, size_t min_lag,
+                                size_t max_lag, size_t length,
+                                int32_t* distortion_value) {
+  size_t best_index = 0;
   int32_t min_distortion = WEBRTC_SPL_WORD32_MAX;
-  for (int i = min_lag; i <= max_lag; i++) {
+  for (size_t i = min_lag; i <= max_lag; i++) {
     int32_t sum_diff = 0;
     const int16_t* data1 = signal;
     const int16_t* data2 = signal - i;
-    for (int j = 0; j < length; j++) {
+    for (size_t j = 0; j < length; j++) {
       sum_diff += WEBRTC_SPL_ABS_W32(data1[j] - data2[j]);
     }
     // Compare with previous minimum.
@@ -293,15 +293,15 @@ void DspHelper::MuteSignal(int16_t* signal, int mute_slope, size_t length) {
 }
 
 int DspHelper::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) {
   // Set filter parameters depending on input frequency.
   // NOTE: The phase delay values are wrong compared to the true phase delay
   // of the filters. However, the error is preserved (through the +1 term) for
   // consistency.
   const int16_t* filter_coefficients;  // Filter coefficients.
-  int16_t filter_length;  // Number of coefficients.
-  int16_t filter_delay;  // Phase delay in samples.
+  size_t filter_length;  // Number of coefficients.
+  size_t filter_delay;  // Phase delay in samples.
   int16_t factor;  // Conversion rate (inFsHz / 8000).
   switch (input_rate_hz) {
     case 8000: {
@@ -345,9 +345,8 @@ int DspHelper::DownsampleTo4kHz(const int16_t* input, size_t input_length,
 
   // Returns -1 if input signal is too short; 0 otherwise.
   return WebRtcSpl_DownsampleFast(
-      &input[filter_length - 1], static_cast<int>(input_length) -
-      (filter_length - 1), output, output_length, filter_coefficients,
-      filter_length, factor, filter_delay);
+      &input[filter_length - 1], input_length - filter_length + 1, output,
+      output_length, filter_coefficients, filter_length, factor, filter_delay);
 }
 
 }  // namespace webrtc