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

use of int16_t/uint16_t.

This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.

This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002

The change is being landed as TBR to all the folks who reviewed the above.

BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher

Review URL: https://codereview.webrtc.org/1230503003 .

Cr-Commit-Position: refs/heads/master@{#9768}

webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc

@@ -65,7 +65,8 @@ void AudioCodecSpeedTest::SetUp() {
   memcpy(&in_data_[loop_length_samples_], &in_data_[0],
          input_length_sample_ * channels_ * sizeof(int16_t));
 
-  max_bytes_ = input_length_sample_ * channels_ * sizeof(int16_t);
+  max_bytes_ =
+      static_cast<size_t>(input_length_sample_ * channels_ * sizeof(int16_t));
   out_data_.reset(new int16_t[output_length_sample_ * channels_]);
   bit_stream_.reset(new uint8_t[max_bytes_]);
 

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_;