Prepare to convert various types to size_t.

webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc

 /*
  *  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 71 matching lines @@
   for (int n = 0; n < frame.samples_per_channel_; ++n)
     out_buff[n] = (frame.data_[2 * n] + frame.data_[2 * n + 1]) >> 1;
   return 0;
 }
 
 // Mono-to-stereo can be used as in-place.
 int UpMix(const AudioFrame& frame, int length_out_buff, int16_t* out_buff) {
   if (length_out_buff < frame.samples_per_channel_) {
     return -1;
   }
-  for (int n = frame.samples_per_channel_ - 1; n >= 0; --n) {
-    out_buff[2 * n + 1] = frame.data_[n];
-    out_buff[2 * n] = frame.data_[n];
+  for (int n = frame.samples_per_channel_; n > 0; --n) {
+    int i = n - 1;
+    int16_t sample = frame.data_[i];
+    out_buff[2 * i + 1] = sample;
+    out_buff[2 * i] = sample;
   }
   return 0;
 }
 
 void ConvertEncodedInfoToFragmentationHeader(
     const AudioEncoder::EncodedInfo& info,
     RTPFragmentationHeader* frag) {
   if (info.redundant.empty()) {
     frag->fragmentationVectorSize = 0;
     return;
@@ skipping 1150 matching lines @@
       *channels = 1;
       break;
 #endif
     default:
       FATAL() << "Codec type " << codec_type << " not supported.";
   }
   return true;
 }
 
 }  // namespace webrtc