| Index: webrtc/modules/audio_processing/agc/legacy/digital_agc.c
 | 
| diff --git a/webrtc/modules/audio_processing/agc/legacy/digital_agc.c b/webrtc/modules/audio_processing/agc/legacy/digital_agc.c
 | 
| index 2ca967a4aae1a0d0a00758b4861a4838b1d11d46..231a2044a3d56d50e03a05955dcbb4bbd3324188 100644
 | 
| --- a/webrtc/modules/audio_processing/agc/legacy/digital_agc.c
 | 
| +++ b/webrtc/modules/audio_processing/agc/legacy/digital_agc.c
 | 
| @@ -189,7 +189,7 @@ int32_t WebRtcAgc_CalculateGainTable(int32_t* gainTable,       // Q16
 | 
|      // Calculate ratio
 | 
|      // Shift |numFIX| as much as possible.
 | 
|      // Ensure we avoid wrap-around in |den| as well.
 | 
| -    if (numFIX > (den >> 8))  // |den| is Q8.
 | 
| +    if (numFIX > (den >> 8) || -numFIX > (den >> 8))  // |den| is Q8.
 | 
|      {
 | 
|        zeros = WebRtcSpl_NormW32(numFIX);
 | 
|      } else {
 | 
| @@ -198,13 +198,11 @@ int32_t WebRtcAgc_CalculateGainTable(int32_t* gainTable,       // Q16
 | 
|      numFIX *= 1 << zeros;  // Q(14+zeros)
 | 
|  
 | 
|      // Shift den so we end up in Qy1
 | 
| -    tmp32no1 = WEBRTC_SPL_SHIFT_W32(den, zeros - 8);  // Q(zeros)
 | 
| -    if (numFIX < 0) {
 | 
| -      numFIX -= tmp32no1 / 2;
 | 
| -    } else {
 | 
| -      numFIX += tmp32no1 / 2;
 | 
| -    }
 | 
| -    y32 = numFIX / tmp32no1;  // in Q14
 | 
| +    tmp32no1 = WEBRTC_SPL_SHIFT_W32(den, zeros - 9);  // Q(zeros - 1)
 | 
| +    y32 = numFIX / tmp32no1;  // in Q15
 | 
| +    // This is to do rounding in Q14.
 | 
| +    y32 = y32 >= 0 ? (y32 + 1) >> 1 : -((-y32 + 1) >> 1);
 | 
| +
 | 
|      if (limiterEnable && (i < limiterIdx)) {
 | 
|        tmp32 = WEBRTC_SPL_MUL_16_U16(i - 1, kLog10_2);  // Q14
 | 
|        tmp32 -= limiterLvl * (1 << 14);                 // Q14
 | 
| 
 |