Fixed the overflow in the AGC

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

 /*
  *  Copyright (c) 2011 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 506 matching lines @@
 
     gain32 += delta;
   }
   // iterate over subframes
   for (k = 1; k < 10; k++) {
     delta = (gains[k + 1] - gains[k]) * (1 << (4 - L2));
     gain32 = gains[k] * (1 << 4);
     // iterate over samples
     for (n = 0; n < L; n++) {
       for (i = 0; i < num_bands; ++i) {
-        tmp32 = out[i][k * L + n] * (gain32 >> 4);
-        out[i][k * L + n] = (int16_t)(tmp32 >> 16);
+        int64_t tmp64 = ((int64_t)(out[i][k * L + n])) * (gain32 >> 4);
+        tmp64 = tmp64 >> 16;
+        if (tmp64 > 32767) {
+          out[i][k * L + n] = 32767;
+        }
+        else if (tmp64 < -32768) {
+          out[i][k * L + n] = -32768;
+        }
+        else {
+          out[i][k * L + n] = (int16_t)(tmp64);
+        }
       }
       gain32 += delta;
     }
   }
 
   return 0;
 }
 
 void WebRtcAgc_InitVad(AgcVad* state) {
   int16_t k;
@@ skipping 140 matching lines @@
   // limit
   if (state->logRatio > 2048) {
     state->logRatio = 2048;
   }
   if (state->logRatio < -2048) {
     state->logRatio = -2048;
   }
 
   return state->logRatio;  // Q10
 }