Update audio code to use size_t more correctly,

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

 /*
  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 41 matching lines @@
 
   void SetUp() override {
     ASSERT_EQ(1, channels_) << "PCMu supports only mono audio.";
     AudioEncoderPcmU::Config config;
     config.frame_size_ms = FLAGS_frame_size_ms;
     encoder_.reset(new AudioEncoderPcmU(config));
     NetEqQualityTest::SetUp();
   }
 
   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 {
       info = encoder_->Encode(dummy_timestamp, &in_data[encoded_samples],
                               kFrameSizeSamples, max_bytes, payload);
       encoded_samples += kFrameSizeSamples;
     } while (info.encoded_bytes == 0);
     return rtc::checked_cast<int>(info.encoded_bytes);
   }
 
  private:
   rtc::scoped_ptr<AudioEncoderPcmU> encoder_;
 };
 
 TEST_F(NetEqPcmuQualityTest, Test) {
   Simulate();
 }
 
 }  // namespace test
 }  // namespace webrtc