Update audio code to use size_t more correctly,

webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc

Index: webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
diff --git a/webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc b/webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
index d94ceb60f812f9947154954595e42725af466408..0b89352308b533781fcda1fdfb7d67eccebf5ff8 100644
--- a/webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
+++ b/webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
@@ -59,11 +59,11 @@ class NetEqPcmuQualityTest : public NetEqQualityTest {
   }
 
   int EncodeBlock(int16_t* in_data,
-                  int block_size_samples,
+                  size_t block_size_samples,
                   uint8_t* payload,
-                  int max_bytes) override {
-    const int kFrameSizeSamples = 80;  // Samples per 10 ms.
-    int encoded_samples = 0;
+                  size_t max_bytes) override {
+    const size_t kFrameSizeSamples = 80;  // Samples per 10 ms.
+    size_t encoded_samples = 0;
     uint32_t dummy_timestamp = 0;
     AudioEncoder::EncodedInfo info;
     do {