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

webrtc/modules/audio_processing/transient/transient_suppressor.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 68 matching lines @@
   size_t buffer_delay_;
   size_t complex_analysis_length_;
   int num_channels_;
   // Input buffer where the original samples are stored.
   rtc::scoped_ptr<float[]> in_buffer_;
   rtc::scoped_ptr<float[]> detection_buffer_;
   // Output buffer where the restored samples are stored.
   rtc::scoped_ptr<float[]> out_buffer_;
 
   // Arrays for fft.
-  rtc::scoped_ptr<int[]> ip_;
+  rtc::scoped_ptr<size_t[]> ip_;
   rtc::scoped_ptr<float[]> wfft_;
 
   rtc::scoped_ptr<float[]> spectral_mean_;
 
   // Stores the data for the fft.
   rtc::scoped_ptr<float[]> fft_buffer_;
 
   rtc::scoped_ptr<float[]> magnitudes_;
 
   const float* window_;
@@ skipping 11 matching lines @@
   int chunks_since_voice_change_;
 
   uint32_t seed_;
 
   bool using_reference_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_TRANSIENT_TRANSIENT_SUPPRESSOR_H_