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

webrtc/common_audio/resampler/push_resampler.cc

Index: webrtc/common_audio/resampler/push_resampler.cc
diff --git a/webrtc/common_audio/resampler/push_resampler.cc b/webrtc/common_audio/resampler/push_resampler.cc
index 973c8f74f765e85cd7d51b01500fa356dd9a6d81..566acdeaa3c7ac670a768104f62a33546830c7c2 100644
--- a/webrtc/common_audio/resampler/push_resampler.cc
+++ b/webrtc/common_audio/resampler/push_resampler.cc
@@ -47,8 +47,10 @@ int PushResampler<T>::InitializeIfNeeded(int 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) {
@@ -64,10 +66,12 @@ int PushResampler<T>::InitializeIfNeeded(int src_sample_rate_hz,
 }
 
 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;
 
@@ -75,15 +79,15 @@ int PushResampler<T>::Resample(const T* src, int src_length, T* dst,
     // 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,
@@ -92,9 +96,10 @@ int PushResampler<T>::Resample(const T* src, int src_length, T* dst,
     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));
   }
 }