| 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
|
|
|