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..35a1f3a5591bfc1f3a30848f98f5b65c7c98446f 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 +198,10 @@ 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 |
+ y32 = (y32 + (y32 >= 0 ? 1 : -1)) / 2; // This is to do rounding in Q14. |
peah-webrtc
2016/06/03 08:00:56
Why not use:
if (y32 < 0)
y32 = -(((-y32) + 1) >
|
+ |
if (limiterEnable && (i < limiterIdx)) { |
tmp32 = WEBRTC_SPL_MUL_16_U16(i - 1, kLog10_2); // Q14 |
tmp32 -= limiterLvl * (1 << 14); // Q14 |