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audio_processing: Replaced WEBRTC_SPL_RSHIFT_U32 with >>

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A trivial macro that is replaced. Affected components:
* AGC
* NSx

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29599005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7366 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
bjornv@webrtc.org 2014-10-02 08:07:05 +00:00
parent 5fa8c458d8
commit 348eac641e
5 changed files with 62 additions and 76 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

10
webrtc/modules/audio_processing/agc/analog_agc.c
View File

@ -871,7 +871,7 @@ int32_t WebRtcAgc_ProcessAnalog(void *state, int32_t inMicLevel,
/* stt->micVol *= 0.903; */
tmp32 = inMicLevelTmp - stt->minLevel;
tmpU32 = WEBRTC_SPL_UMUL(29591, (uint32_t)(tmp32));
stt->micVol = (int32_t)WEBRTC_SPL_RSHIFT_U32(tmpU32, 15) + stt->minLevel;
stt->micVol = (tmpU32 >> 15) + stt->minLevel;
if (stt->micVol > lastMicVol - 2)
{
stt->micVol = lastMicVol - 2;
@ -1004,7 +1004,7 @@ int32_t WebRtcAgc_ProcessAnalog(void *state, int32_t inMicLevel,
/* 0.95 in Q15 */
tmp32 = inMicLevelTmp - stt->minLevel;
tmpU32 = WEBRTC_SPL_UMUL(31130, (uint32_t)(tmp32));
stt->micVol = (int32_t)WEBRTC_SPL_RSHIFT_U32(tmpU32, 15) + stt->minLevel;
stt->micVol = (tmpU32 >> 15) + stt->minLevel;
if (stt->micVol > lastMicVol - 1)
{
stt->micVol = lastMicVol - 1;
@ -1055,7 +1055,7 @@ int32_t WebRtcAgc_ProcessAnalog(void *state, int32_t inMicLevel,
/* 0.965 in Q15 */
tmp32 = inMicLevelTmp - stt->minLevel;
tmpU32 = WEBRTC_SPL_UMUL(31621, (uint32_t)(inMicLevelTmp - stt->minLevel));
stt->micVol = (int32_t)WEBRTC_SPL_RSHIFT_U32(tmpU32, 15) + stt->minLevel;
stt->micVol = (tmpU32 >> 15) + stt->minLevel;
if (stt->micVol > lastMicVol - 1)
{
stt->micVol = lastMicVol - 1;
@ -1109,7 +1109,7 @@ int32_t WebRtcAgc_ProcessAnalog(void *state, int32_t inMicLevel,
tmp32 = inMicLevelTmp - stt->minLevel;
tmpU32 = ((uint32_t)weightFIX * (uint32_t)(inMicLevelTmp - stt->minLevel));
stt->micVol = (int32_t)WEBRTC_SPL_RSHIFT_U32(tmpU32, 14) + stt->minLevel;
stt->micVol = (tmpU32 >> 14) + stt->minLevel;
if (stt->micVol < lastMicVol + 2)
{
stt->micVol = lastMicVol + 2;
@ -1170,7 +1170,7 @@ int32_t WebRtcAgc_ProcessAnalog(void *state, int32_t inMicLevel,
tmp32 = inMicLevelTmp - stt->minLevel;
tmpU32 = ((uint32_t)weightFIX * (uint32_t)(inMicLevelTmp - stt->minLevel));
stt->micVol = (int32_t)WEBRTC_SPL_RSHIFT_U32(tmpU32, 14) + stt->minLevel;
stt->micVol = (tmpU32 >> 14) + stt->minLevel;
if (stt->micVol < lastMicVol + 1)
{
stt->micVol = lastMicVol + 1;

14
webrtc/modules/audio_processing/agc/digital_agc.c
View File

@ -149,12 +149,12 @@ int32_t WebRtcAgc_CalculateGainTable(int32_t *gainTable, // Q16
absInLevel = (uint32_t)WEBRTC_SPL_ABS_W32(inLevel); // Q14
// LUT with interpolation
intPart = (uint16_t)WEBRTC_SPL_RSHIFT_U32(absInLevel, 14);
intPart = (uint16_t)(absInLevel >> 14);
fracPart = (uint16_t)(absInLevel & 0x00003FFF); // extract the fractional part
tmpU16 = kGenFuncTable[intPart + 1] - kGenFuncTable[intPart]; // Q8
tmpU32no1 = tmpU16 * fracPart; // Q22
tmpU32no1 += (uint32_t)kGenFuncTable[intPart] << 14; // Q22
logApprox = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 8); // Q14
logApprox = tmpU32no1 >> 8; // Q14
// Compensate for negative exponent using the relation:
// log2(1 + 2^-x) = log2(1 + 2^x) - x
if (inLevel < 0)
@ -164,25 +164,25 @@ int32_t WebRtcAgc_CalculateGainTable(int32_t *gainTable, // Q16
if (zeros < 15)
{
// Not enough space for multiplication
tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(absInLevel, 15 - zeros); // Q(zeros-1)
tmpU32no2 = absInLevel >> (15 - zeros); // Q(zeros-1)
tmpU32no2 = WEBRTC_SPL_UMUL_32_16(tmpU32no2, kLogE_1); // Q(zeros+13)
if (zeros < 9)
{
tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 9 - zeros); // Q(zeros+13)
zerosScale = 9 - zeros;
tmpU32no1 >>= zerosScale; // Q(zeros+13)
} else
{
tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(tmpU32no2, zeros - 9); // Q22
tmpU32no2 >>= zeros - 9; // Q22
}
} else
{
tmpU32no2 = WEBRTC_SPL_UMUL_32_16(absInLevel, kLogE_1); // Q28
tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(tmpU32no2, 6); // Q22
tmpU32no2 >>= 6; // Q22
}
logApprox = 0;
if (tmpU32no2 < tmpU32no1)
{
logApprox = WEBRTC_SPL_RSHIFT_U32(tmpU32no1 - tmpU32no2, 8 - zerosScale); //Q14
logApprox = (tmpU32no1 - tmpU32no2) >> (8 - zerosScale); //Q14
}
}
numFIX = (maxGain * constMaxGain) << 6; // Q14

94
webrtc/modules/audio_processing/ns/nsx_core.c
View File

@ -855,7 +855,7 @@ void WebRtcNsx_FeatureParameterExtraction(NsxInst_t* inst, int flag) {
}
// Spectral flatness
// (inst->featureSpecFlat*20)>>10 = (inst->featureSpecFlat*5)>>8
histIndex = WEBRTC_SPL_RSHIFT_U32(inst->featureSpecFlat * 5, 8);
histIndex = (inst->featureSpecFlat * 5) >> 8;
if (histIndex < HIST_PAR_EST) {
inst->histSpecFlat[histIndex]++;
}
@ -1058,7 +1058,7 @@ void WebRtcNsx_ComputeSpectralFlatness(NsxInst_t* inst, uint16_t* magn) {
} else {
//if at least one frequency component is zero, treat separately
tmpU32 = WEBRTC_SPL_UMUL_32_16(inst->featureSpecFlat, SPECT_FLAT_TAVG_Q14); // Q24
inst->featureSpecFlat -= WEBRTC_SPL_RSHIFT_U32(tmpU32, 14); // Q10
inst->featureSpecFlat -= tmpU32 >> 14; // Q10
return;
}
}
@ -1122,7 +1122,7 @@ void WebRtcNsx_ComputeSpectralDifference(NsxInst_t* inst, uint16_t* magnIn) {
}
// normalize by replacing div of "inst->magnLen" with "inst->stages-1" shifts
avgPauseFX = WEBRTC_SPL_RSHIFT_W32(avgPauseFX, inst->stages - 1);
avgMagnFX = (int32_t)WEBRTC_SPL_RSHIFT_U32(inst->sumMagn, inst->stages - 1);
avgMagnFX = inst->sumMagn >> (inst->stages - 1);
// Largest possible deviation in magnPause for (co)var calculations
tmp32no1 = WEBRTC_SPL_MAX(maxPause - avgPauseFX, avgPauseFX - minPause);
// Get number of shifts to make sure we don't get wrap around in varPause
@ -1142,8 +1142,8 @@ void WebRtcNsx_ComputeSpectralDifference(NsxInst_t* inst, uint16_t* magnIn) {
varPauseUFX += tmp32no1 * tmp32no1; // Q(2*(prevQMagn-minPause))
}
//update of average magnitude spectrum: Q(-2*stages) and averaging replaced by shifts
inst->curAvgMagnEnergy += WEBRTC_SPL_RSHIFT_U32(inst->magnEnergy, 2 * inst->normData
+ inst->stages - 1);
inst->curAvgMagnEnergy +=
inst->magnEnergy >> (2 * inst->normData + inst->stages - 1);
avgDiffNormMagnUFX = varMagnUFX; // Q(2*qMagn)
if ((varPauseUFX) && (covMagnPauseFX)) {
@ -1152,7 +1152,7 @@ void WebRtcNsx_ComputeSpectralDifference(NsxInst_t* inst, uint16_t* magnIn) {
if (norm32 > 0) {
tmpU32no1 <<= norm32; // Q(prevQMagn+qMagn+norm32)
} else {
tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, -norm32); // Q(prevQMagn+qMagn+norm32)
tmpU32no1 >>= -norm32; // Q(prevQMagn+qMagn+norm32)
}
tmpU32no2 = WEBRTC_SPL_UMUL(tmpU32no1, tmpU32no1); // Q(2*(prevQMagn+qMagn-norm32))
@ -1165,7 +1165,7 @@ void WebRtcNsx_ComputeSpectralDifference(NsxInst_t* inst, uint16_t* magnIn) {
if (varPauseUFX > 0) {
// Q(2*(qMagn+norm32-16+minPause))
tmpU32no1 = tmpU32no2 / varPauseUFX;
tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, nShifts);
tmpU32no1 >>= nShifts;
// Q(2*qMagn)
avgDiffNormMagnUFX -= WEBRTC_SPL_MIN(avgDiffNormMagnUFX, tmpU32no1);
@ -1174,15 +1174,15 @@ void WebRtcNsx_ComputeSpectralDifference(NsxInst_t* inst, uint16_t* magnIn) {
}
}
//normalize and compute time average update of difference feature
tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(avgDiffNormMagnUFX, 2 * inst->normData);
tmpU32no1 = avgDiffNormMagnUFX >> (2 * inst->normData);
if (inst->featureSpecDiff > tmpU32no1) {
tmpU32no2 = WEBRTC_SPL_UMUL_32_16(inst->featureSpecDiff - tmpU32no1,
SPECT_DIFF_TAVG_Q8); // Q(8-2*stages)
inst->featureSpecDiff -= WEBRTC_SPL_RSHIFT_U32(tmpU32no2, 8); // Q(-2*stages)
inst->featureSpecDiff -= tmpU32no2 >> 8; // Q(-2*stages)
} else {
tmpU32no2 = WEBRTC_SPL_UMUL_32_16(tmpU32no1 - inst->featureSpecDiff,
SPECT_DIFF_TAVG_Q8); // Q(8-2*stages)
inst->featureSpecDiff += WEBRTC_SPL_RSHIFT_U32(tmpU32no2, 8); // Q(-2*stages)
inst->featureSpecDiff += tmpU32no2 >> 8; // Q(-2*stages)
}
}
@ -1282,11 +1282,8 @@ void WebRtcNsx_DataAnalysis(NsxInst_t* inst, short* speechFrame, uint16_t* magnU
//
// Switch initMagnEst to Q(minNorm-stages)
inst->initMagnEst[0] = WEBRTC_SPL_RSHIFT_U32(inst->initMagnEst[0],
right_shifts_in_initMagnEst);
inst->initMagnEst[inst->anaLen2] =
WEBRTC_SPL_RSHIFT_U32(inst->initMagnEst[inst->anaLen2],
right_shifts_in_initMagnEst); // Q(minNorm-stages)
inst->initMagnEst[0] >>= right_shifts_in_initMagnEst;
inst->initMagnEst[inst->anaLen2] >>= right_shifts_in_initMagnEst;
// Shift magnU16 to same domain as initMagnEst
tmpU32no1 = WEBRTC_SPL_RSHIFT_W32((uint32_t)magnU16[0],
@ -1326,8 +1323,7 @@ void WebRtcNsx_DataAnalysis(NsxInst_t* inst, short* speechFrame, uint16_t* magnU
inst->sumMagn += (uint32_t)magnU16[i]; // Q(normData-stages)
// Switch initMagnEst to Q(minNorm-stages)
inst->initMagnEst[i] = WEBRTC_SPL_RSHIFT_U32(inst->initMagnEst[i],
right_shifts_in_initMagnEst);
inst->initMagnEst[i] >>= right_shifts_in_initMagnEst;
// Shift magnU16 to same domain as initMagnEst, i.e., Q(minNorm-stages)
tmpU32no1 = WEBRTC_SPL_RSHIFT_W32((uint32_t)magnU16[i],
@ -1360,16 +1356,15 @@ void WebRtcNsx_DataAnalysis(NsxInst_t* inst, short* speechFrame, uint16_t* magnU
// Estimate White noise
// Switch whiteNoiseLevel to Q(minNorm-stages)
inst->whiteNoiseLevel = WEBRTC_SPL_RSHIFT_U32(inst->whiteNoiseLevel,
right_shifts_in_initMagnEst);
inst->whiteNoiseLevel >>= right_shifts_in_initMagnEst;
// Update the average magnitude spectrum, used as noise estimate.
tmpU32no1 = WEBRTC_SPL_UMUL_32_16(inst->sumMagn, inst->overdrive);
tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, inst->stages + 8);
tmpU32no1 >>= inst->stages + 8;
// Replacing division above with 'stages' shifts
// Shift to same Q-domain as whiteNoiseLevel
tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, right_shifts_in_magnU16);
tmpU32no1 >>= right_shifts_in_magnU16;
// This operation is safe from wrap around as long as END_STARTUP_SHORT < 128
assert(END_STARTUP_SHORT < 128);
inst->whiteNoiseLevel += tmpU32no1; // Q(minNorm-stages)
@ -1405,14 +1400,14 @@ void WebRtcNsx_DataAnalysis(NsxInst_t* inst, short* speechFrame, uint16_t* magnU
// Calculate and update pinkNoiseNumerator. Result in Q11.
tmp_2_w32 = WEBRTC_SPL_MUL_16_U16(sum_log_i_square, sum_log_magn_u16); // Q(11-zeros)
tmpU32no1 = WEBRTC_SPL_RSHIFT_U32((uint32_t)sum_log_i_log_magn, 12); // Q5
tmpU32no1 = sum_log_i_log_magn >> 12; // Q5
// Shift the largest value of sum_log_i and tmp32no3 before multiplication
tmp_u16 = ((uint16_t)sum_log_i << 1); // Q6
if ((uint32_t)sum_log_i > tmpU32no1) {
tmp_u16 >>= zeros;
} else {
tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, zeros);
tmpU32no1 >>= zeros;
}
tmp_2_w32 -= (int32_t)WEBRTC_SPL_UMUL_32_16(tmpU32no1, tmp_u16); // Q(11-zeros)
matrix_determinant = WEBRTC_SPL_RSHIFT_W16(matrix_determinant, zeros); // Q(-zeros)
@ -1615,7 +1610,7 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
//noise estimate from previous frame
for (i = 0; i < inst->magnLen; i++) {
prevNoiseU16[i] = (uint16_t)WEBRTC_SPL_RSHIFT_U32(inst->prevNoiseU32[i], 11); // Q(prevQNoise)
prevNoiseU16[i] = (uint16_t)(inst->prevNoiseU32[i] >> 11); // Q(prevQNoise)
}
if (inst->blockIndex < END_STARTUP_SHORT) {
@ -1674,7 +1669,7 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
numerator <<= nShifts;
// Shift denominator to Q(nShifts-6+minNorm-stages)
tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(inst->initMagnEst[i], 6 - nShifts);
tmpU32no1 = inst->initMagnEst[i] >> (6 - nShifts);
if (tmpU32no1 == 0) {
// This is only possible if numerator = 0, in which case
// we don't need any division.
@ -1693,16 +1688,16 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
// may not.
// Shift 'noiseU32' to 'q_domain_to_use'
tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(noiseU32[i], (int)qNoise - q_domain_to_use);
tmpU32no1 = noiseU32[i] >> (qNoise - q_domain_to_use);
// Shift 'noise_estimate_avg' to 'q_domain_to_use'
tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(noise_estimate_avg, inst->minNorm - inst->stages
- q_domain_to_use);
tmpU32no2 = noise_estimate_avg >>
(inst->minNorm - inst->stages - q_domain_to_use);
// Make a simple check to see if we have enough room for weighting 'tmpU32no1'
// without wrap around
nShifts = 0;
if (tmpU32no1 & 0xfc000000) {
tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 6);
tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(tmpU32no2, 6);
tmpU32no1 >>= 6;
tmpU32no2 >>= 6;
nShifts = 6;
}
tmpU32no1 *= inst->blockIndex;
@ -1719,9 +1714,8 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
// used to normalize spectral difference measure
if (inst->blockIndex < END_STARTUP_LONG) {
// substituting division with shift ending up in Q(-2*stages)
inst->timeAvgMagnEnergyTmp
+= WEBRTC_SPL_RSHIFT_U32(inst->magnEnergy,
2 * inst->normData + inst->stages - 1);
inst->timeAvgMagnEnergyTmp +=
inst->magnEnergy >> (2 * inst->normData + inst->stages - 1);
inst->timeAvgMagnEnergy = WebRtcSpl_DivU32U16(inst->timeAvgMagnEnergyTmp,
inst->blockIndex + 1);
}
@ -1755,7 +1749,7 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
postLocSnr[i] = 2048; // 1.0 in Q11
tmpU32no1 = (uint32_t)magnU16[i] << 6; // Q(6+qMagn)
if (postShifts < 0) {
tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(noiseU32[i], -postShifts); // Q(6+qMagn)
tmpU32no2 = noiseU32[i] >> -postShifts; // Q(6+qMagn)
} else {
tmpU32no2 = noiseU32[i] << postShifts; // Q(6+qMagn)
}
@ -1774,7 +1768,7 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
// |nearMagnEst| in Q(prevQMagn + 14)
nearMagnEst = inst->prevMagnU16[i] * inst->noiseSupFilter[i];
tmpU32no1 = nearMagnEst << 3; // Q(prevQMagn+17)
tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(inst->prevNoiseU32[i], nShifts); // Q(prevQMagn+6)
tmpU32no2 = inst->prevNoiseU32[i] >> nShifts; // Q(prevQMagn+6)
if (tmpU32no2 > 0) {
tmpU32no1 /= tmpU32no2; // Q11
@ -1789,7 +1783,7 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
tmpU32no2 = WEBRTC_SPL_UMUL_32_16(postLocSnr[i] - 2048, ONE_MINUS_DD_PR_SNR_Q11); // Q22
priorSnr = tmpU32no1 + tmpU32no2 + 512; // Q22 (added 512 for rounding)
// priorLocSnr = 1 + 2*priorSnr
priorLocSnr[i] = 2048 + WEBRTC_SPL_RSHIFT_U32(priorSnr, 10); // Q11
priorLocSnr[i] = 2048 + (priorSnr >> 10); // Q11
} // end of loop over frequencies
// done with step 1: DD computation of prior and post SNR
@ -1811,7 +1805,7 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
// get normalization for spectral difference for next window estimate
// Shift to Q(-2*stages)
inst->curAvgMagnEnergy = WEBRTC_SPL_RSHIFT_U32(inst->curAvgMagnEnergy, STAT_UPDATES);
inst->curAvgMagnEnergy >>= STAT_UPDATES;
tmpU32no1 = (inst->curAvgMagnEnergy + inst->timeAvgMagnEnergy + 1) >> 1; //Q(-2*stages)
// Update featureSpecDiff
@ -1857,7 +1851,7 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
// noiseUpdate = noisePrev[i] + (1 - gammaNoise) * nonSpeechProb * (magn[i] - noisePrev[i])
if (postShifts < 0) {
tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(magnU16[i], -postShifts); // Q(prevQNoise)
tmpU32no2 = magnU16[i] >> -postShifts; // Q(prevQNoise)
} else {
tmpU32no2 = (uint32_t)magnU16[i] << postShifts; // Q(prevQNoise)
}
@ -1875,12 +1869,10 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
tmpU32no3 = WEBRTC_SPL_UMUL_32_16(tmpU32no1, nonSpeechProbFinal[i]); // Q(prevQNoise+8)
if (0x7c000000 & tmpU32no3) {
// Shifting required before multiplication
tmpU32no2
= WEBRTC_SPL_UMUL_32_16(WEBRTC_SPL_RSHIFT_U32(tmpU32no3, 5), gammaNoise); // Q(prevQNoise+11)
tmpU32no2 = (tmpU32no3 >> 5) * gammaNoise; // Q(prevQNoise+11)
} else {
// We can do shifting after multiplication
tmpU32no2
= WEBRTC_SPL_RSHIFT_U32(WEBRTC_SPL_UMUL_32_16(tmpU32no3, gammaNoise), 5); // Q(prevQNoise+11)
tmpU32no2 = (tmpU32no3 * gammaNoise) >> 5; // Q(prevQNoise+11)
}
if (sign > 0) {
noiseUpdateU32 += tmpU32no2; // Q(prevQNoise+11)
@ -1909,12 +1901,10 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
if (0x7c000000 & tmpU32no3) {
// Shifting required before multiplication
tmpU32no2
= WEBRTC_SPL_UMUL_32_16(WEBRTC_SPL_RSHIFT_U32(tmpU32no3, 5), gammaNoise); // Q(prevQNoise+11)
tmpU32no2 = (tmpU32no3 >> 5) * gammaNoise; // Q(prevQNoise+11)
} else {
// We can do shifting after multiplication
tmpU32no2
= WEBRTC_SPL_RSHIFT_U32(WEBRTC_SPL_UMUL_32_16(tmpU32no3, gammaNoise), 5); // Q(prevQNoise+11)
tmpU32no2 = (tmpU32no3 * gammaNoise) >> 5; // Q(prevQNoise+11)
}
if (sign > 0) {
tmpU32no1 = inst->prevNoiseU32[i] + tmpU32no2; // Q(prevQNoise+11)
@ -1983,7 +1973,7 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
tmpNoiseU32 = noiseU32[i] << -nShifts; // Q(qMagn)
} else if (nShifts > 17) {
tmpMagnU32 = (uint32_t)magnU16[i] << 17; // Q(qMagn+17)
tmpNoiseU32 = WEBRTC_SPL_RSHIFT_U32(noiseU32[i], nShifts - 17); // Q(qMagn+17)
tmpNoiseU32 = noiseU32[i] >> (nShifts - 17); // Q(qMagn+17)
} else {
tmpMagnU32 = (uint32_t)magnU16[i] << nShifts; // Q(qNoise_prev+11)
tmpNoiseU32 = noiseU32[i]; // Q(qNoise_prev+11)
@ -1992,7 +1982,7 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
tmpU32no1 = tmpMagnU32 - tmpNoiseU32; // Q(qCur)
norm32no2 = WEBRTC_SPL_MIN(11, WebRtcSpl_NormU32(tmpU32no1));
tmpU32no1 <<= norm32no2; // Q(qCur+norm32no2)
tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(tmpNoiseU32, 11 - norm32no2); // Q(qCur+norm32no2-11)
tmpU32no2 = tmpNoiseU32 >> (11 - norm32no2); // Q(qCur+norm32no2-11)
if (tmpU32no2 > 0) {
tmpU32no1 /= tmpU32no2; // Q11
}
@ -2008,8 +1998,7 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
priorSnr = tmpU32no1 + tmpU32no2; // Q22
//gain filter
tmpU32no1 = (uint32_t)(inst->overdrive)
+ WEBRTC_SPL_RSHIFT_U32(priorSnr + 8192, 14); // Q8
tmpU32no1 = inst->overdrive + ((priorSnr + 8192) >> 14); // Q8
assert(inst->overdrive > 0);
tmpU16no1 = (priorSnr + tmpU32no1 / 2) / tmpU32no1; // Q14
inst->noiseSupFilter[i] = WEBRTC_SPL_SAT(16384, tmpU16no1, inst->denoiseBound); // 16384 = Q14(1.0) // Q14
@ -2037,7 +2026,7 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
}
} else {
for (i = 0; i < inst->magnLen; i++) {
inst->prevNoiseU32[i] = WEBRTC_SPL_RSHIFT_U32(noiseU32[i], 5 - norm32no1); // Q(qNoise+11)
inst->prevNoiseU32[i] = noiseU32[i] >> (5 - norm32no1); // Q(qNoise+11)
inst->prevMagnU16[i] = magnU16[i]; // Q(qMagn)
}
}
@ -2071,8 +2060,7 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
}
assert(inst->stages >= 7);
avgProbSpeechHB = (4096 - (tmpU16no1 >> (inst->stages - 7))); // Q12
avgFilterGainHB = (int16_t)WEBRTC_SPL_RSHIFT_U32(
tmpU32no1, inst->stages - 3); // Q14
avgFilterGainHB = (int16_t)(tmpU32no1 >> (inst->stages - 3)); // Q14
// // original FLOAT code
// // gain based on speech probability:

11
webrtc/modules/audio_processing/ns/nsx_core_c.c
View File

@ -45,7 +45,7 @@ void WebRtcNsx_SpeechNoiseProb(NsxInst_t* inst,
if (normTmp > 10) {
den = priorLocSnr[i] << (normTmp - 11); // Q(normTmp)
} else {
den = WEBRTC_SPL_RSHIFT_U32(priorLocSnr[i], 11 - normTmp); // Q(normTmp)
den = priorLocSnr[i] >> (11 - normTmp); // Q(normTmp)
}
if (den > 0) {
besselTmpFX32 -= num / den; // Q11
@ -127,7 +127,7 @@ void WebRtcNsx_SpeechNoiseProb(NsxInst_t* inst,
// FLOAT code
// indicator1 = 0.5 * (tanh(sgnMap * widthPrior *
// (threshPrior1 - tmpFloat1)) + 1.0);
tableIndex = (int16_t)WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 14);
tableIndex = (int16_t)(tmpU32no1 >> 14);
if (tableIndex < 16) {
tmp16no2 = kIndicatorTable[tableIndex];
tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex];
@ -150,8 +150,7 @@ void WebRtcNsx_SpeechNoiseProb(NsxInst_t* inst,
WebRtcSpl_NormU32(inst->featureSpecDiff));
assert(normTmp >= 0);
tmpU32no1 = inst->featureSpecDiff << normTmp; // Q(normTmp-2*stages)
tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(inst->timeAvgMagnEnergy,
20 - inst->stages - normTmp);
tmpU32no2 = inst->timeAvgMagnEnergy >> (20 - inst->stages - normTmp);
if (tmpU32no2 > 0) {
// Q(20 - inst->stages)
tmpU32no1 /= tmpU32no2;
@ -170,12 +169,12 @@ void WebRtcNsx_SpeechNoiseProb(NsxInst_t* inst,
//widthPrior = widthPrior * 2.0;
nShifts--;
}
tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no2, nShifts);
tmpU32no1 = tmpU32no2 >> nShifts;
// compute indicator function: sigmoid map
/* FLOAT code
indicator2 = 0.5 * (tanh(widthPrior * (tmpFloat1 - threshPrior2)) + 1.0);
*/
tableIndex = (int16_t)WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 14);
tableIndex = (int16_t)(tmpU32no1 >> 14);
if (tableIndex < 16) {
tmp16no2 = kIndicatorTable[tableIndex];
tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex];

9
webrtc/modules/audio_processing/ns/nsx_core_mips.c
View File

@ -162,7 +162,7 @@ void WebRtcNsx_SpeechNoiseProb(NsxInst_t* inst,
// FLOAT code
// indicator1 = 0.5 * (tanh(sgnMap * widthPrior *
// (threshPrior1 - tmpFloat1)) + 1.0);
tableIndex = (int16_t)WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 14);
tableIndex = (int16_t)(tmpU32no1 >> 14);
if (tableIndex < 16) {
tmp16no2 = kIndicatorTable[tableIndex];
tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex];
@ -185,8 +185,7 @@ void WebRtcNsx_SpeechNoiseProb(NsxInst_t* inst,
WebRtcSpl_NormU32(inst->featureSpecDiff));
assert(normTmp >= 0);
tmpU32no1 = inst->featureSpecDiff << normTmp; // Q(normTmp-2*stages)
tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(inst->timeAvgMagnEnergy,
20 - inst->stages - normTmp);
tmpU32no2 = inst->timeAvgMagnEnergy >> (20 - inst->stages - normTmp);
if (tmpU32no2 > 0) {
// Q(20 - inst->stages)
tmpU32no1 /= tmpU32no2;
@ -205,12 +204,12 @@ void WebRtcNsx_SpeechNoiseProb(NsxInst_t* inst,
//widthPrior = widthPrior * 2.0;
nShifts--;
}
tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no2, nShifts);
tmpU32no1 = tmpU32no2 >> nShifts;
// compute indicator function: sigmoid map
/* FLOAT code
indicator2 = 0.5 * (tanh(widthPrior * (tmpFloat1 - threshPrior2)) + 1.0);
*/
tableIndex = (int16_t)WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 14);
tableIndex = (int16_t)(tmpU32no1 >> 14);
if (tableIndex < 16) {
tmp16no2 = kIndicatorTable[tableIndex];
tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex];