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

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 18 matching lines @@
                       int output_sampling_khz);
   virtual void SetUp();
   virtual void TearDown();
 
   // EncodeABlock(...) does the following:
   // 1. encodes a block of audio, saved in |in_data|,
   // 2. save the bit stream to |bit_stream| of |max_bytes| bytes in size,
   // 3. assign |encoded_bytes| with the length of the bit stream (in bytes),
   // 4. return the cost of time (in millisecond) spent on actual encoding.
   virtual float EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
-                             int max_bytes, int* encoded_bytes) = 0;
+                             size_t max_bytes, size_t* encoded_bytes) = 0;
 
   // DecodeABlock(...) does the following:
   // 1. decodes the bit stream in |bit_stream| with a length of |encoded_bytes|
   // (in bytes),
   // 2. save the decoded audio in |out_data|,
   // 3. return the cost of time (in millisecond) spent on actual decoding.
-  virtual float DecodeABlock(const uint8_t* bit_stream, int encoded_bytes,
+  virtual float DecodeABlock(const uint8_t* bit_stream, size_t encoded_bytes,
                              int16_t* out_data) = 0;
 
   // 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_;
 
   // Expected output number of samples-per-channel in a frame.
   int 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.
-  int max_bytes_;
+  size_t max_bytes_;
 
-  int encoded_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_