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

webrtc/voice_engine/output_mixer.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 571 matching lines @@
 
     // replace mixed audio with Dtmf tone
     if (_audioFrame.num_channels_ == 1)
     {
         // mono
         memcpy(_audioFrame.data_, toneBuffer, sizeof(int16_t)
             * toneSamples);
     } else
     {
         // stereo
-        for (int i = 0; i < _audioFrame.samples_per_channel_; i++)
+        for (size_t i = 0; i < _audioFrame.samples_per_channel_; i++)
         {
             _audioFrame.data_[2 * i] = toneBuffer[i];
             _audioFrame.data_[2 * i + 1] = 0;
         }
     }
     assert(_audioFrame.samples_per_channel_ == toneSamples);
 
     return 0;
 }
 
 }  // namespace voe
 }  // namespace webrtc