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 |