Update audio code to use size_t more correctly, webrtc/common_audio/ portion.

webrtc/common_audio/include/audio_util.h

 /*
  *  Copyright (c) 2013 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 53 matching lines @@
 void FloatS16ToS16(const float* src, size_t size, int16_t* dest);
 void FloatToFloatS16(const float* src, size_t size, float* dest);
 void FloatS16ToFloat(const float* src, size_t size, float* dest);
 
 // Deinterleave audio from |interleaved| to the channel buffers pointed to
 // by |deinterleaved|. There must be sufficient space allocated in the
 // |deinterleaved| buffers (|num_channel| buffers with |samples_per_channel|
 // per buffer).
 template <typename T>
 void Deinterleave(const T* interleaved,
-                  int samples_per_channel,
+                  size_t samples_per_channel,
                   int num_channels,
                   T* const* deinterleaved) {
   for (int i = 0; i < num_channels; ++i) {
     T* channel = deinterleaved[i];
     int interleaved_idx = i;
-    for (int j = 0; j < samples_per_channel; ++j) {
+    for (size_t j = 0; j < samples_per_channel; ++j) {
       channel[j] = interleaved[interleaved_idx];
       interleaved_idx += num_channels;
     }
   }
 }
 
 // Interleave audio from the channel buffers pointed to by |deinterleaved| to
 // |interleaved|. There must be sufficient space allocated in |interleaved|
 // (|samples_per_channel| * |num_channels|).
 template <typename T>
 void Interleave(const T* const* deinterleaved,
-                int samples_per_channel,
+                size_t samples_per_channel,
                 int num_channels,
                 T* interleaved) {
   for (int i = 0; i < num_channels; ++i) {
     const T* channel = deinterleaved[i];
     int interleaved_idx = i;
-    for (int j = 0; j < samples_per_channel; ++j) {
+    for (size_t j = 0; j < samples_per_channel; ++j) {
       interleaved[interleaved_idx] = channel[j];
       interleaved_idx += num_channels;
     }
   }
 }
 
 template <typename T, typename Intermediate>
 void DownmixToMono(const T* const* input_channels,
-                   int num_frames,
+                   size_t num_frames,
                    int num_channels,
                    T* out) {
-  for (int i = 0; i < num_frames; ++i) {
+  for (size_t i = 0; i < num_frames; ++i) {
     Intermediate value = input_channels[0][i];
     for (int j = 1; j < num_channels; ++j) {
       value += input_channels[j][i];
     }
     out[i] = value / num_channels;
   }
 }
 
 // Downmixes an interleaved multichannel signal to a single channel by averaging
 // all channels.
 template <typename T, typename Intermediate>
 void DownmixInterleavedToMonoImpl(const T* interleaved,
-                                  int num_frames,
+                                  size_t num_frames,
                                   int num_channels,
                                   T* deinterleaved) {
   DCHECK_GT(num_channels, 0);
-  DCHECK_GT(num_frames, 0);
+  DCHECK_GT(num_frames, 0u);
 
   const T* const end = interleaved + num_frames * num_channels;
 
   while (interleaved < end) {
     const T* const frame_end = interleaved + num_channels;
 
     Intermediate value = *interleaved++;
     while (interleaved < frame_end) {
       value += *interleaved++;
     }
 
     *deinterleaved++ = value / num_channels;
   }
 }
 
 template <typename T>
 void DownmixInterleavedToMono(const T* interleaved,
-                              int num_frames,
+                              size_t num_frames,
                               int num_channels,
                               T* deinterleaved);
 
 template <>
 void DownmixInterleavedToMono<int16_t>(const int16_t* interleaved,
-                                       int num_frames,
+                                       size_t num_frames,
                                        int num_channels,
                                        int16_t* deinterleaved);
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_