Misc. small cleanups

webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_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 37 matching lines @@
 
   // Encoding and decode an audio of |audio_duration| (in seconds) and
   // record the runtime for encoding and decoding separately.
   void EncodeDecode(size_t audio_duration);
 
   int block_duration_ms_;
   int input_sampling_khz_;
   int output_sampling_khz_;
 
   // Number of samples-per-channel in a frame.
-  int input_length_sample_;
+  size_t input_length_sample_;
 
   // Expected output number of samples-per-channel in a frame.
-  int output_length_sample_;
+  size_t output_length_sample_;
 
   rtc::scoped_ptr<int16_t[]> in_data_;
   rtc::scoped_ptr<int16_t[]> out_data_;
   size_t data_pointer_;
   size_t loop_length_samples_;
   rtc::scoped_ptr<uint8_t[]> bit_stream_;
 
   // Maximum number of bytes in output bitstream for a frame of audio.
   size_t max_bytes_;
 
   size_t encoded_bytes_;
   float encoding_time_ms_;
   float decoding_time_ms_;
   FILE* out_file_;
 
   int channels_;
 
   // Bit rate is in bit-per-second.
   int bit_rate_;
 
   std::string in_filename_;
 
   // Determines whether to save the output to file.
   bool save_out_data_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_