Convert channel counts to size_t.

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 69 matching lines @@
   }
 }
 
 // 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,
                   size_t samples_per_channel,
-                  int num_channels,
+                  size_t num_channels,
                   T* const* deinterleaved) {
-  for (int i = 0; i < num_channels; ++i) {
+  for (size_t i = 0; i < num_channels; ++i) {
     T* channel = deinterleaved[i];
-    int interleaved_idx = i;
+    size_t interleaved_idx = i;
     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,
                 size_t samples_per_channel,
-                int num_channels,
+                size_t num_channels,
                 T* interleaved) {
-  for (int i = 0; i < num_channels; ++i) {
+  for (size_t i = 0; i < num_channels; ++i) {
     const T* channel = deinterleaved[i];
-    int interleaved_idx = i;
+    size_t interleaved_idx = i;
     for (size_t j = 0; j < samples_per_channel; ++j) {
       interleaved[interleaved_idx] = channel[j];
       interleaved_idx += num_channels;
     }
   }
 }
 
 // Copies audio from a single channel buffer pointed to by |mono| to each
 // channel of |interleaved|. There must be sufficient space allocated in
 // |interleaved| (|samples_per_channel| * |num_channels|).
@@ skipping 56 matching lines @@
 
 template <>
 void DownmixInterleavedToMono<int16_t>(const int16_t* interleaved,
                                        size_t num_frames,
                                        int num_channels,
                                        int16_t* deinterleaved);
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_