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

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

Index: webrtc/modules/audio_coding/main/acm2/acm_receive_test.cc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receive_test.cc b/webrtc/modules/audio_coding/main/acm2/acm_receive_test.cc
index dc59984a95304fbb0133c17f37a4b710365f457e..b5a86d0bde3e2f6c68dbe915699dbc2463d6f967 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receive_test.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receive_test.cc
@@ -93,7 +93,8 @@ void AcmReceiveTest::Run() {
       AudioFrame output_frame;
       EXPECT_TRUE(acm_->Get10MsAudio(&output_frame));
       EXPECT_EQ(output_freq_hz_, output_frame.sample_rate_hz_);
-      const int samples_per_block = output_freq_hz_ * 10 / 1000;
+      const size_t samples_per_block =
+          static_cast<size_t>(output_freq_hz_ * 10 / 1000);
       EXPECT_EQ(samples_per_block, output_frame.samples_per_channel_);
       if (expected_output_channels_ != kArbitraryChannels) {
         if (output_frame.speech_type_ == webrtc::AudioFrame::kPLC) {