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

webrtc/common_audio/resampler/push_resampler.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 29 matching lines @@
     return 0;
 
   if (src_sample_rate_hz <= 0 || dst_sample_rate_hz <= 0 ||
       num_channels <= 0 || num_channels > 2)
     return -1;
 
   src_sample_rate_hz_ = src_sample_rate_hz;
   dst_sample_rate_hz_ = dst_sample_rate_hz;
   num_channels_ = num_channels;
 
-  const int src_size_10ms_mono = src_sample_rate_hz / 100;
-  const int dst_size_10ms_mono = dst_sample_rate_hz / 100;
+  const size_t src_size_10ms_mono =
+      static_cast<size_t>(src_sample_rate_hz / 100);
+  const size_t dst_size_10ms_mono =
+      static_cast<size_t>(dst_sample_rate_hz / 100);
   sinc_resampler_.reset(new PushSincResampler(src_size_10ms_mono,
                                               dst_size_10ms_mono));
   if (num_channels_ == 2) {
     src_left_.reset(new T[src_size_10ms_mono]);
     src_right_.reset(new T[src_size_10ms_mono]);
     dst_left_.reset(new T[dst_size_10ms_mono]);
     dst_right_.reset(new T[dst_size_10ms_mono]);
     sinc_resampler_right_.reset(new PushSincResampler(src_size_10ms_mono,
                                                       dst_size_10ms_mono));
   }
 
   return 0;
 }
 
 template <typename T>
-int PushResampler<T>::Resample(const T* src, int src_length, T* dst,
-                               int dst_capacity) {
-  const int src_size_10ms = src_sample_rate_hz_ * num_channels_ / 100;
-  const int dst_size_10ms = dst_sample_rate_hz_ * num_channels_ / 100;
+int PushResampler<T>::Resample(const T* src, size_t src_length, T* dst,
+                               size_t dst_capacity) {
+  const size_t src_size_10ms =
+      static_cast<size_t>(src_sample_rate_hz_ * num_channels_ / 100);
+  const size_t dst_size_10ms =
+      static_cast<size_t>(dst_sample_rate_hz_ * num_channels_ / 100);
   if (src_length != src_size_10ms || dst_capacity < dst_size_10ms)
     return -1;
 
   if (src_sample_rate_hz_ == dst_sample_rate_hz_) {
     // The old resampler provides this memcpy facility in the case of matching
     // sample rates, so reproduce it here for the sinc resampler.
     memcpy(dst, src, src_length * sizeof(T));
-    return src_length;
+    return static_cast<int>(src_length);
   }
   if (num_channels_ == 2) {
-    const int src_length_mono = src_length / num_channels_;
-    const int dst_capacity_mono = dst_capacity / num_channels_;
+    const size_t src_length_mono = src_length / num_channels_;
+    const size_t dst_capacity_mono = dst_capacity / num_channels_;
     T* deinterleaved[] = {src_left_.get(), src_right_.get()};
     Deinterleave(src, src_length_mono, num_channels_, deinterleaved);
 
-    int dst_length_mono =
+    size_t dst_length_mono =
         sinc_resampler_->Resample(src_left_.get(), src_length_mono,
                                   dst_left_.get(), dst_capacity_mono);
     sinc_resampler_right_->Resample(src_right_.get(), src_length_mono,
                                     dst_right_.get(), dst_capacity_mono);
 
     deinterleaved[0] = dst_left_.get();
     deinterleaved[1] = dst_right_.get();
     Interleave(deinterleaved, dst_length_mono, num_channels_, dst);
-    return dst_length_mono * num_channels_;
+    return static_cast<int>(dst_length_mono * num_channels_);
   } else {
-    return sinc_resampler_->Resample(src, src_length, dst, dst_capacity);
+    return static_cast<int>(
+        sinc_resampler_->Resample(src, src_length, dst, dst_capacity));
   }
 }
 
 // Explictly generate required instantiations.
 template class PushResampler<int16_t>;
 template class PushResampler<float>;
 
 }  // namespace webrtc