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

webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.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 17 matching lines @@
                            0.5063f, 0.5190f, 0.5316f, 0.5443f,
                            0.5570f, 0.5696f, 0.5823f, 0.5949f,
                            0.6076f, 0.6203f, 0.6329f, 0.6456f,
                            0.6582f, 0.6709f, 0.6835f, 0.6962f,
                            0.7089f, 0.7215f, 0.7342f, 0.7468f,
                            0.7595f, 0.7722f, 0.7848f, 0.7975f,
                            0.8101f, 0.8228f, 0.8354f, 0.8481f,
                            0.8608f, 0.8734f, 0.8861f, 0.8987f,
                            0.9114f, 0.9241f, 0.9367f, 0.9494f,
                            0.9620f, 0.9747f, 0.9873f, 1.0000f};
-const int rampSize = sizeof(rampArray)/sizeof(rampArray[0]);
+const size_t rampSize = sizeof(rampArray)/sizeof(rampArray[0]);
 }  // namespace
 
 namespace webrtc {
 void CalculateEnergy(AudioFrame& audioFrame)
 {
     audioFrame.energy_ = 0;
-    for(int position = 0; position < audioFrame.samples_per_channel_;
+    for(size_t position = 0; position < audioFrame.samples_per_channel_;
         position++)
     {
         // TODO(andrew): this can easily overflow.
         audioFrame.energy_ += audioFrame.data_[position] *
                               audioFrame.data_[position];
     }
 }
 
 void RampIn(AudioFrame& audioFrame)
 {
     assert(rampSize <= audioFrame.samples_per_channel_);
-    for(int i = 0; i < rampSize; i++)
+    for(size_t i = 0; i < rampSize; i++)
     {
         audioFrame.data_[i] = static_cast<int16_t>(rampArray[i] *
                                                    audioFrame.data_[i]);
     }
 }
 
 void RampOut(AudioFrame& audioFrame)
 {
     assert(rampSize <= audioFrame.samples_per_channel_);
-    for(int i = 0; i < rampSize; i++)
+    for(size_t i = 0; i < rampSize; i++)
     {
-        const int rampPos = rampSize - 1 - i;
+        const size_t rampPos = rampSize - 1 - i;
         audioFrame.data_[i] = static_cast<int16_t>(rampArray[rampPos] *
                                                    audioFrame.data_[i]);
     }
     memset(&audioFrame.data_[rampSize], 0,
            (audioFrame.samples_per_channel_ - rampSize) *
            sizeof(audioFrame.data_[0]));
 }
 }  // namespace webrtc