Update audio code to use size_t more correctly,

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

 /*
  *  Copyright (c) 2014 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 58 matching lines @@
                    enum NetEqDecoder decoder_type);
   virtual ~NetEqQualityTest();
 
   void SetUp() override;
 
   // EncodeBlock(...) does the following:
   // 1. encodes a block of audio, saved in |in_data| and has a length of
   // |block_size_samples| (samples per channel),
   // 2. save the bit stream to |payload| of |max_bytes| bytes in size,
   // 3. returns the length of the payload (in bytes),
-  virtual int EncodeBlock(int16_t* in_data, int block_size_samples,
-                          uint8_t* payload, int max_bytes) = 0;
+  virtual int EncodeBlock(int16_t* in_data, size_t block_size_samples,
+                          uint8_t* payload, size_t max_bytes) = 0;
 
   // PacketLost(...) determines weather a packet sent at an indicated time gets
   // lost or not.
   bool PacketLost();
 
   // DecodeBlock() decodes a block of audio using the payload stored in
   // |payload_| with the length of |payload_size_bytes_| (bytes). The decoded
   // audio is to be stored in |out_data_|.
   int DecodeBlock();
 
@@ skipping 13 matching lines @@
  private:
   int decoded_time_ms_;
   int decodable_time_ms_;
   double drift_factor_;
   int packet_loss_rate_;
   const int block_duration_ms_;
   const int in_sampling_khz_;
   const int out_sampling_khz_;
 
   // Number of samples per channel in a frame.
-  const int in_size_samples_;
+  const size_t in_size_samples_;
 
   // Expected output number of samples per channel in a frame.
-  const int out_size_samples_;
+  const size_t out_size_samples_;
 
   size_t payload_size_bytes_;
-  int max_payload_bytes_;
+  size_t max_payload_bytes_;
 
   rtc::scoped_ptr<InputAudioFile> in_file_;
   rtc::scoped_ptr<AudioSink> output_;
   std::ofstream log_file_;
 
   rtc::scoped_ptr<RtpGenerator> rtp_generator_;
   rtc::scoped_ptr<NetEq> neteq_;
   rtc::scoped_ptr<LossModel> loss_model_;
 
   rtc::scoped_ptr<int16_t[]> in_data_;
   rtc::scoped_ptr<uint8_t[]> payload_;
   rtc::scoped_ptr<int16_t[]> out_data_;
   WebRtcRTPHeader rtp_header_;
 
   size_t total_payload_size_bytes_;
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_