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

webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc

Index: webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc b/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
index f0ef70a3ba46be4dec24fbef2641d1d9e12d5b21..c86fab766066e11ae5c18d199d98299c86e452f5 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
@@ -45,15 +45,15 @@ class OpusFecTest : public TestWithParam<coding_param> {
 
   int block_duration_ms_;
   int sampling_khz_;
-  int block_length_sample_;
+  size_t block_length_sample_;
 
   int channels_;
   int bit_rate_;
 
   size_t data_pointer_;
   size_t loop_length_samples_;
-  int max_bytes_;
-  int encoded_bytes_;
+  size_t max_bytes_;
+  size_t encoded_bytes_;
 
   WebRtcOpusEncInst* opus_encoder_;
   WebRtcOpusDecInst* opus_decoder_;
@@ -122,7 +122,8 @@ void OpusFecTest::TearDown() {
 OpusFecTest::OpusFecTest()
     : block_duration_ms_(kOpusBlockDurationMs),
       sampling_khz_(kOpusSamplingKhz),
-      block_length_sample_(block_duration_ms_ * sampling_khz_),
+      block_length_sample_(
+          static_cast<size_t>(block_duration_ms_ * sampling_khz_)),
       data_pointer_(0),
       max_bytes_(0),
       encoded_bytes_(0),
@@ -137,7 +138,7 @@ void OpusFecTest::EncodeABlock() {
                                 max_bytes_, &bit_stream_[0]);
   EXPECT_GT(value, 0);
 
-  encoded_bytes_ = value;
+  encoded_bytes_ = static_cast<size_t>(value);
 }
 
 void OpusFecTest::DecodeABlock(bool lost_previous, bool lost_current) {
@@ -154,14 +155,14 @@ void OpusFecTest::DecodeABlock(bool lost_previous, bool lost_current) {
     } else {
       value_1 = WebRtcOpus_DecodePlc(opus_decoder_, &out_data_[0], 1);
     }
-    EXPECT_EQ(block_length_sample_, value_1);
+    EXPECT_EQ(static_cast<int>(block_length_sample_), value_1);
   }
 
   if (!lost_current) {
     // Decode current frame.
     value_2 = WebRtcOpus_Decode(opus_decoder_, &bit_stream_[0], encoded_bytes_,
                                 &out_data_[value_1 * channels_], &audio_type);
-    EXPECT_EQ(block_length_sample_, value_2);
+    EXPECT_EQ(static_cast<int>(block_length_sample_), value_2);
   }
 }