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

Index: webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
diff --git a/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h b/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
index 35ac69e8ab7369055d35384c6173f7933e7cc86c..2736c2912e8bca2ef6dfe14ae1b9bdacd6e63f6e 100644
--- a/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
+++ b/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
@@ -36,14 +36,14 @@ class AudioCodecSpeedTest : public testing::TestWithParam<coding_param> {
   // 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
@@ -67,9 +67,9 @@ class AudioCodecSpeedTest : public testing::TestWithParam<coding_param> {
   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_;