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

webrtc/modules/audio_coding/main/acm2/acm_resampler.cc

Index: webrtc/modules/audio_coding/main/acm2/acm_resampler.cc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc b/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc
index 97d87b1b3a4b357939d27b089f94c70598728a26..2650725331b94e5db5e110836704d11082657ec6 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc
@@ -29,9 +29,9 @@ int ACMResampler::Resample10Msec(const int16_t* in_audio,
                                  int in_freq_hz,
                                  int out_freq_hz,
                                  int num_audio_channels,
-                                 int out_capacity_samples,
+                                 size_t out_capacity_samples,
                                  int16_t* out_audio) {
-  int in_length = in_freq_hz * num_audio_channels / 100;
+  size_t in_length = static_cast<size_t>(in_freq_hz * num_audio_channels / 100);
   int out_length = out_freq_hz * num_audio_channels / 100;
   if (in_freq_hz == out_freq_hz) {
     if (out_capacity_samples < in_length) {
@@ -39,7 +39,7 @@ int ACMResampler::Resample10Msec(const int16_t* in_audio,
       return -1;
     }
     memcpy(out_audio, in_audio, in_length * sizeof(int16_t));
-    return in_length / num_audio_channels;
+    return static_cast<int>(in_length / num_audio_channels);
   }
 
   if (resampler_.InitializeIfNeeded(in_freq_hz, out_freq_hz,