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

webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc

Index: webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
index c60b993ad8c7f8096d4899dffbf7acc196434ff3..1c028c97443c6a1e59de242e651d380ad155a49e 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
@@ -218,8 +218,9 @@ NetEqQualityTest::NetEqQualityTest(int block_duration_ms,
       block_duration_ms_(block_duration_ms),
       in_sampling_khz_(in_sampling_khz),
       out_sampling_khz_(out_sampling_khz),
-      in_size_samples_(in_sampling_khz_ * block_duration_ms_),
-      out_size_samples_(out_sampling_khz_ * kOutputSizeMs),
+      in_size_samples_(
+          static_cast<size_t>(in_sampling_khz_ * block_duration_ms_)),
+      out_size_samples_(static_cast<size_t>(out_sampling_khz_ * kOutputSizeMs)),
       payload_size_bytes_(0),
       max_payload_bytes_(0),
       in_file_(new ResampleInputAudioFile(FLAGS_in_filename,
@@ -392,7 +393,7 @@ int NetEqQualityTest::Transmit() {
 
 int NetEqQualityTest::DecodeBlock() {
   int channels;
-  int samples;
+  size_t samples;
   int ret = neteq_->GetAudio(out_size_samples_ * channels_, &out_data_[0],
                              &samples, &channels, NULL);
 
@@ -400,9 +401,9 @@ int NetEqQualityTest::DecodeBlock() {
     return -1;
   } else {
     assert(channels == channels_);
-    assert(samples == kOutputSizeMs * out_sampling_khz_);
+    assert(samples == static_cast<size_t>(kOutputSizeMs * out_sampling_khz_));
     CHECK(output_->WriteArray(out_data_.get(), samples * channels));
-    return samples;
+    return static_cast<int>(samples);
   }
 }