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

webrtc/modules/audio_processing/transient/transient_suppressor.cc

Index: webrtc/modules/audio_processing/transient/transient_suppressor.cc
diff --git a/webrtc/modules/audio_processing/transient/transient_suppressor.cc b/webrtc/modules/audio_processing/transient/transient_suppressor.cc
index 2f79a20ac79da6b19e9cbc2dbd21f66fd1a0a0c0..206d14db755c03baefc64b0435cfb68d512bd947 100644
--- a/webrtc/modules/audio_processing/transient/transient_suppressor.cc
+++ b/webrtc/modules/audio_processing/transient/transient_suppressor.cc
@@ -124,7 +124,7 @@ int TransientSuppressor::Initialize(int sample_rate_hz,
          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]));