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}
diff --git a/webrtc/modules/audio_coding/neteq/normal.cc b/webrtc/modules/audio_coding/neteq/normal.cc
index bf455c9..ebecbf9 100644
--- a/webrtc/modules/audio_coding/neteq/normal.cc
+++ b/webrtc/modules/audio_coding/neteq/normal.cc
@@ -45,12 +45,12 @@
   output->PushBackInterleaved(input, length);
   int16_t* signal = &(*output)[0][0];
 
-  const unsigned fs_mult = fs_hz_ / 8000;
+  const int fs_mult = fs_hz_ / 8000;
   assert(fs_mult > 0);
   // fs_shift = log2(fs_mult), rounded down.
   // Note that |fs_shift| is not "exact" for 48 kHz.
   // TODO(hlundin): Investigate this further.
-  const int fs_shift = 30 - WebRtcSpl_NormW32(static_cast<int32_t>(fs_mult));
+  const int fs_shift = 30 - WebRtcSpl_NormW32(fs_mult);
 
   // Check if last RecOut call resulted in an Expand. If so, we have to take
   // care of some cross-fading and unmuting.
@@ -73,11 +73,11 @@
       int16_t* signal = &(*output)[channel_ix][0];
       size_t length_per_channel = length / output->Channels();
       // Find largest absolute value in new data.
-      int16_t decoded_max = WebRtcSpl_MaxAbsValueW16(
-        signal,  static_cast<int>(length_per_channel));
+      int16_t decoded_max =
+          WebRtcSpl_MaxAbsValueW16(signal, length_per_channel);
       // Adjust muting factor if needed (to BGN level).
-      int energy_length = std::min(static_cast<int>(fs_mult * 64),
-                                   static_cast<int>(length_per_channel));
+      size_t energy_length =
+          std::min(static_cast<size_t>(fs_mult * 64), length_per_channel);
       int scaling = 6 + fs_shift
           - WebRtcSpl_NormW32(decoded_max * decoded_max);
       scaling = std::max(scaling, 0);  // |scaling| should always be >= 0.
@@ -111,7 +111,7 @@
       }
 
       // If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14).
-      int increment = static_cast<int>(64 / fs_mult);
+      int increment = 64 / fs_mult;
       for (size_t i = 0; i < length_per_channel; i++) {
         // Scale with mute factor.
         assert(channel_ix < output->Channels());
@@ -131,7 +131,7 @@
       assert(fs_shift < 3);  // Will always be 0, 1, or, 2.
       increment = 4 >> fs_shift;
       int fraction = increment;
-      for (size_t i = 0; i < 8 * fs_mult; i++) {
+      for (size_t i = 0; i < static_cast<size_t>(8 * fs_mult); i++) {
         // TODO(hlundin): Add 16 instead of 8 for correct rounding. Keeping 8
         // now for legacy bit-exactness.
         assert(channel_ix < output->Channels());
@@ -144,7 +144,7 @@
     }
   } else if (last_mode == kModeRfc3389Cng) {
     assert(output->Channels() == 1);  // Not adapted for multi-channel yet.
-    static const int kCngLength = 32;
+    static const size_t kCngLength = 32;
     int16_t cng_output[kCngLength];
     // Reset mute factor and start up fresh.
     external_mute_factor_array[0] = 16384;
@@ -167,7 +167,7 @@
     assert(fs_shift < 3);  // Will always be 0, 1, or, 2.
     int16_t increment = 4 >> fs_shift;
     int16_t fraction = increment;
-    for (size_t i = 0; i < 8 * fs_mult; i++) {
+    for (size_t i = 0; i < static_cast<size_t>(8 * fs_mult); i++) {
       // TODO(hlundin): Add 16 instead of 8 for correct rounding. Keeping 8 now
       // for legacy bit-exactness.
       signal[i] =
@@ -178,7 +178,7 @@
     // Previous was neither of Expand, FadeToBGN or RFC3389_CNG, but we are
     // still ramping up from previous muting.
     // If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14).
-    int increment = static_cast<int>(64 / fs_mult);
+    int increment = 64 / fs_mult;
     size_t length_per_channel = length / output->Channels();
     for (size_t i = 0; i < length_per_channel; i++) {
       for (size_t channel_ix = 0; channel_ix < output->Channels();