Re-enable UBsan on AGC.

webrtc/modules/audio_processing/agc/legacy/analog_agc.c

 /*
  *  Copyright (c) 2012 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 458 matching lines @@
   }
 
   /* stt->envSum *= 0.99; */
   stt->envSum = (int16_t)((stt->envSum * 32440) >> 15);
 }
 
 void WebRtcAgc_ZeroCtrl(LegacyAgc* stt, int32_t* inMicLevel, int32_t* env) {
   int16_t i;
   int32_t tmp32 = 0;
   int32_t midVal;
+  const int kZeroThreshold = 500;
 
   /* Is the input signal zero? */
   for (i = 0; i < 10; i++) {
+    if (env[i] >= kZeroThreshold || tmp32 >= kZeroThreshold) {

[peah-webrtc, 2016/06/03 06:51:12]

What about my comment on what happened to the int64 sum variant?

+      break;
+    }
     tmp32 += env[i];
   }
 
   /* Each block is allowed to have a few non-zero
    * samples.
    */
-  if (tmp32 < 500) {
+  if (i == 10 && tmp32 < kZeroThreshold) {
     stt->msZero += 10;
   } else {
     stt->msZero = 0;
   }
 
   if (stt->muteGuardMs > 0) {
     stt->muteGuardMs -= 10;
   }
 
   if (stt->msZero > 500) {
@@ skipping 881 matching lines @@
     fprintf(stt->fpt, "minLevel, maxLevel value(s) are invalid\n\n");
 #endif
     return -1;
   } else {
 #ifdef WEBRTC_AGC_DEBUG_DUMP
     fprintf(stt->fpt, "\n");
 #endif
     return 0;
   }
 }