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

webrtc/modules/audio_coding/codecs/ilbc/state_construct.c

Index: webrtc/modules/audio_coding/codecs/ilbc/state_construct.c
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/state_construct.c b/webrtc/modules/audio_coding/codecs/ilbc/state_construct.c
index 324b670c9bd25bf62dd1ffff563855a2a90cfbf3..29fe91b87ea9f5eed086044e9842369ae4ab98f2 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/state_construct.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/state_construct.c
@@ -24,14 +24,14 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_StateConstruct(
-    int16_t idxForMax,   /* (i) 6-bit index for the quantization of
+    size_t idxForMax,   /* (i) 6-bit index for the quantization of
                                            max amplitude */
     int16_t *idxVec,   /* (i) vector of quantization indexes */
     int16_t *syntDenum,  /* (i) synthesis filter denumerator */
     int16_t *Out_fix,  /* (o) the decoded state vector */
-    int16_t len    /* (i) length of a state vector */
+    size_t len    /* (i) length of a state vector */
                                   ) {
-  int k;
+  size_t k;
   int16_t maxVal;
   int16_t *tmp1, *tmp2, *tmp3;
   /* Stack based */
@@ -96,7 +96,7 @@ void WebRtcIlbcfix_StateConstruct(
   /* Run MA filter + AR filter */
   WebRtcSpl_FilterMAFastQ12(
       sampleVal, sampleMa,
-      numerator, LPC_FILTERORDER+1, (int16_t)(len + LPC_FILTERORDER));
+      numerator, LPC_FILTERORDER+1, len + LPC_FILTERORDER);
   WebRtcSpl_MemSetW16(&sampleMa[len + LPC_FILTERORDER], 0, (len - LPC_FILTERORDER));
   WebRtcSpl_FilterARFastQ12(
       sampleMa, sampleAr,