Update audio code to use size_t more correctly,

webrtc/modules/audio_coding/main/test/PCMFile.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 132 matching lines @@
 void PCMFile::Write10MsData(AudioFrame& audio_frame) {
   if (audio_frame.num_channels_ == 1) {
     if (!save_stereo_) {
       if (fwrite(audio_frame.data_, sizeof(uint16_t),
                  audio_frame.samples_per_channel_, pcm_file_) !=
           static_cast<size_t>(audio_frame.samples_per_channel_)) {
         return;
       }
     } else {
       int16_t* stereo_audio = new int16_t[2 * audio_frame.samples_per_channel_];
-      int k;
+      size_t k;
       for (k = 0; k < audio_frame.samples_per_channel_; k++) {

[kwiberg-webrtc, 2015/07/12 17:39:26]

Merge lines 153 and 154 while you're at it?

[Peter Kasting, 2015/07/13 02:46:26]

I'm willing to add that to the misc. cleanup changeset if you want, but I'm
intentionally avoiding doing even a single "incidental cleanup" change in a CL
that touches some 450 files.

         stereo_audio[k << 1] = audio_frame.data_[k];
         stereo_audio[(k << 1) + 1] = audio_frame.data_[k];
       }
       if (fwrite(stereo_audio, sizeof(int16_t),
                  2 * audio_frame.samples_per_channel_, pcm_file_) !=
           static_cast<size_t>(2 * audio_frame.samples_per_channel_)) {
         return;
       }
       delete[] stereo_audio;
     }
   } else {
     if (fwrite(audio_frame.data_, sizeof(int16_t),
                audio_frame.num_channels_ * audio_frame.samples_per_channel_,
                pcm_file_) !=
         static_cast<size_t>(audio_frame.num_channels_ *
                             audio_frame.samples_per_channel_)) {
       return;
     }
   }
 }
 
-void PCMFile::Write10MsData(int16_t* playout_buffer, uint16_t length_smpls) {
+void PCMFile::Write10MsData(int16_t* playout_buffer, size_t length_smpls) {
   if (fwrite(playout_buffer, sizeof(uint16_t), length_smpls, pcm_file_) !=
       length_smpls) {
     return;
   }
 }
 
 void PCMFile::Close() {
   fclose(pcm_file_);
   pcm_file_ = NULL;
 }
 
 void PCMFile::Rewind() {
   rewind(pcm_file_);
   end_of_file_ = false;
 }
 
 bool PCMFile::Rewinded() {
   return rewinded_;
 }
 
 void PCMFile::SaveStereo(bool is_stereo) {
   save_stereo_ = is_stereo;
 }
 
 void PCMFile::ReadStereo(bool is_stereo) {
   read_stereo_ = is_stereo;
 }
 
 }  // namespace webrtc