Update audio code to use size_t more correctly, webrtc/modules/audio_processing/

webrtc/modules/audio_processing/transient/transient_suppressor.cc

 /*
  *  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 106 matching lines @@
   detection_buffer_.reset(new float[detection_length_]);
   memset(detection_buffer_.get(),
          0,
          detection_length_ * sizeof(detection_buffer_[0]));
   out_buffer_.reset(new float[analysis_length_ * num_channels_]);
   memset(out_buffer_.get(),
          0,
          analysis_length_ * num_channels_ * sizeof(out_buffer_[0]));
   // ip[0] must be zero to trigger initialization using rdft().
   size_t ip_length = 2 + sqrtf(analysis_length_);
-  ip_.reset(new int[ip_length]());
+  ip_.reset(new size_t[ip_length]());
   memset(ip_.get(), 0, ip_length * sizeof(ip_[0]));
   wfft_.reset(new float[complex_analysis_length_ - 1]);
   memset(wfft_.get(), 0, (complex_analysis_length_ - 1) * sizeof(wfft_[0]));
   spectral_mean_.reset(new float[complex_analysis_length_ * num_channels_]);
   memset(spectral_mean_.get(),
          0,
          complex_analysis_length_ * num_channels_ * sizeof(spectral_mean_[0]));
   fft_buffer_.reset(new float[analysis_length_ + 2]);
   memset(fft_buffer_.get(), 0, (analysis_length_ + 2) * sizeof(fft_buffer_[0]));
   magnitudes_.reset(new float[complex_analysis_length_]);
@@ skipping 277 matching lines @@
       const float magnitude_ratio = new_magnitude / magnitudes_[i];
 
       fft_buffer_[i * 2] *= magnitude_ratio;
       fft_buffer_[i * 2 + 1] *= magnitude_ratio;
       magnitudes_[i] = new_magnitude;
     }
   }
 }
 
 }  // namespace webrtc