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common_audio: Replaced WEBRTC_SPL_LSHIFT_U32 with << in audio_processing

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Affected components:
* AECMobile
  - Added a help function since the same operation was performed several times.
* Auto Gain Control
* Noise Suppression (fixed point)

BUG=3348,3353
TESTED=locally on Linux
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7076 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
bjornv@webrtc.org 2014-09-05 06:01:53 +00:00
parent 2c03a97d37
commit c2c4117477
5 changed files with 42 additions and 42 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

22
webrtc/modules/audio_processing/aecm/aecm_core.c
View File

@ -706,6 +706,15 @@ int16_t WebRtcAecm_AsymFilt(const int16_t filtOld, const int16_t inVal,
return retVal; return retVal;
} }
// ExtractFractionPart(a, zeros)
//
// returns the fraction part of |a|, with |zeros| number of leading zeros, as an
// int16_t scaled to Q8. There is no sanity check of |a| in the sense that the
// number of zeros match.
static int16_t ExtractFractionPart(uint32_t a, int zeros) {
return (int16_t)(((a << zeros) & 0x7FFFFFFF) >> 23);
}
// WebRtcAecm_CalcEnergies(...) // WebRtcAecm_CalcEnergies(...)
// //
// This function calculates the log of energies for nearend, farend and estimated // This function calculates the log of energies for nearend, farend and estimated
@ -751,9 +760,7 @@ void WebRtcAecm_CalcEnergies(AecmCore_t * aecm,
if (nearEner) if (nearEner)
{ {
zeros = WebRtcSpl_NormU32(nearEner); zeros = WebRtcSpl_NormU32(nearEner);
frac = (int16_t)WEBRTC_SPL_RSHIFT_U32( frac = ExtractFractionPart(nearEner, zeros);
(WEBRTC_SPL_LSHIFT_U32(nearEner, zeros) & 0x7FFFFFFF),
23);
// log2 in Q8 // log2 in Q8
tmp16 += WEBRTC_SPL_LSHIFT_W16((31 - zeros), 8) + frac; tmp16 += WEBRTC_SPL_LSHIFT_W16((31 - zeros), 8) + frac;
tmp16 -= WEBRTC_SPL_LSHIFT_W16(aecm->dfaNoisyQDomain, 8); tmp16 -= WEBRTC_SPL_LSHIFT_W16(aecm->dfaNoisyQDomain, 8);
@ -774,8 +781,7 @@ void WebRtcAecm_CalcEnergies(AecmCore_t * aecm,
if (tmpFar) if (tmpFar)
{ {
zeros = WebRtcSpl_NormU32(tmpFar); zeros = WebRtcSpl_NormU32(tmpFar);
frac = (int16_t)WEBRTC_SPL_RSHIFT_U32((WEBRTC_SPL_LSHIFT_U32(tmpFar, zeros) frac = ExtractFractionPart(tmpFar, zeros);
& 0x7FFFFFFF), 23);
// log2 in Q8 // log2 in Q8
tmp16 += WEBRTC_SPL_LSHIFT_W16((31 - zeros), 8) + frac; tmp16 += WEBRTC_SPL_LSHIFT_W16((31 - zeros), 8) + frac;
tmp16 -= WEBRTC_SPL_LSHIFT_W16(far_q, 8); tmp16 -= WEBRTC_SPL_LSHIFT_W16(far_q, 8);
@ -787,8 +793,7 @@ void WebRtcAecm_CalcEnergies(AecmCore_t * aecm,
if (tmpAdapt) if (tmpAdapt)
{ {
zeros = WebRtcSpl_NormU32(tmpAdapt); zeros = WebRtcSpl_NormU32(tmpAdapt);
frac = (int16_t)WEBRTC_SPL_RSHIFT_U32((WEBRTC_SPL_LSHIFT_U32(tmpAdapt, zeros) frac = ExtractFractionPart(tmpAdapt, zeros);
& 0x7FFFFFFF), 23);
//log2 in Q8 //log2 in Q8
tmp16 += WEBRTC_SPL_LSHIFT_W16((31 - zeros), 8) + frac; tmp16 += WEBRTC_SPL_LSHIFT_W16((31 - zeros), 8) + frac;
tmp16 -= WEBRTC_SPL_LSHIFT_W16(RESOLUTION_CHANNEL16 + far_q, 8); tmp16 -= WEBRTC_SPL_LSHIFT_W16(RESOLUTION_CHANNEL16 + far_q, 8);
@ -800,8 +805,7 @@ void WebRtcAecm_CalcEnergies(AecmCore_t * aecm,
if (tmpStored) if (tmpStored)
{ {
zeros = WebRtcSpl_NormU32(tmpStored); zeros = WebRtcSpl_NormU32(tmpStored);
frac = (int16_t)WEBRTC_SPL_RSHIFT_U32((WEBRTC_SPL_LSHIFT_U32(tmpStored, zeros) frac = ExtractFractionPart(tmpStored, zeros);
& 0x7FFFFFFF), 23);
//log2 in Q8 //log2 in Q8
tmp16 += WEBRTC_SPL_LSHIFT_W16((31 - zeros), 8) + frac; tmp16 += WEBRTC_SPL_LSHIFT_W16((31 - zeros), 8) + frac;
tmp16 -= WEBRTC_SPL_LSHIFT_W16(RESOLUTION_CHANNEL16 + far_q, 8); tmp16 -= WEBRTC_SPL_LSHIFT_W16(RESOLUTION_CHANNEL16 + far_q, 8);

2
webrtc/modules/audio_processing/agc/digital_agc.c
View File

@ -154,7 +154,7 @@ int32_t WebRtcAgc_CalculateGainTable(int32_t *gainTable, // Q16
fracPart = (uint16_t)(absInLevel & 0x00003FFF); // extract the fractional part fracPart = (uint16_t)(absInLevel & 0x00003FFF); // extract the fractional part
tmpU16 = kGenFuncTable[intPart + 1] - kGenFuncTable[intPart]; // Q8 tmpU16 = kGenFuncTable[intPart + 1] - kGenFuncTable[intPart]; // Q8
tmpU32no1 = WEBRTC_SPL_UMUL_16_16(tmpU16, fracPart); // Q22 tmpU32no1 = WEBRTC_SPL_UMUL_16_16(tmpU16, fracPart); // Q22
tmpU32no1 += WEBRTC_SPL_LSHIFT_U32((uint32_t)kGenFuncTable[intPart], 14); // Q22 tmpU32no1 += (uint32_t)kGenFuncTable[intPart] << 14; // Q22
logApprox = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 8); // Q14 logApprox = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 8); // Q14
// Compensate for negative exponent using the relation: // Compensate for negative exponent using the relation:
// log2(1 + 2^-x) = log2(1 + 2^x) - x // log2(1 + 2^-x) = log2(1 + 2^x) - x

32
webrtc/modules/audio_processing/ns/nsx_core.c
View File

@ -620,7 +620,7 @@ void WebRtcNsx_CalcParametricNoiseEstimate(NsxInst_t* inst,
} }
// Shift fractional part to Q(minNorm-stages) // Shift fractional part to Q(minNorm-stages)
tmp32no2 = WEBRTC_SPL_SHIFT_W32(tmp32no2, int_part - 11); tmp32no2 = WEBRTC_SPL_SHIFT_W32(tmp32no2, int_part - 11);
*noise_estimate_avg = WEBRTC_SPL_LSHIFT_U32(1, int_part) + (uint32_t)tmp32no2; *noise_estimate_avg = (1 << int_part) + (uint32_t)tmp32no2;
// Scale up to initMagnEst, which is not block averaged // Scale up to initMagnEst, which is not block averaged
*noise_estimate = (*noise_estimate_avg) * (uint32_t)(inst->blockIndex + 1); *noise_estimate = (*noise_estimate_avg) * (uint32_t)(inst->blockIndex + 1);
} }
@ -1149,7 +1149,7 @@ void WebRtcNsx_ComputeSpectralDifference(NsxInst_t* inst, uint16_t* magnIn) {
tmpU32no1 = (uint32_t)WEBRTC_SPL_ABS_W32(covMagnPauseFX); // Q(prevQMagn+qMagn) tmpU32no1 = (uint32_t)WEBRTC_SPL_ABS_W32(covMagnPauseFX); // Q(prevQMagn+qMagn)
norm32 = WebRtcSpl_NormU32(tmpU32no1) - 16; norm32 = WebRtcSpl_NormU32(tmpU32no1) - 16;
if (norm32 > 0) { if (norm32 > 0) {
tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(tmpU32no1, norm32); // Q(prevQMagn+qMagn+norm32) tmpU32no1 <<= norm32; // Q(prevQMagn+qMagn+norm32)
} else { } else {
tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, -norm32); // Q(prevQMagn+qMagn+norm32) tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, -norm32); // Q(prevQMagn+qMagn+norm32)
} }
@ -1660,7 +1660,7 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
// numerator = (initMagnEst - noise_estimate * overdrive) // numerator = (initMagnEst - noise_estimate * overdrive)
// Result in Q(8+minNorm-stages) // Result in Q(8+minNorm-stages)
tmpU32no1 = WEBRTC_SPL_UMUL_32_16(noise_estimate, inst->overdrive); tmpU32no1 = WEBRTC_SPL_UMUL_32_16(noise_estimate, inst->overdrive);
numerator = WEBRTC_SPL_LSHIFT_U32(inst->initMagnEst[i], 8); numerator = inst->initMagnEst[i] << 8;
if (numerator > tmpU32no1) { if (numerator > tmpU32no1) {
// Suppression filter coefficient larger than zero, so calculate. // Suppression filter coefficient larger than zero, so calculate.
numerator -= tmpU32no1; numerator -= tmpU32no1;
@ -1671,7 +1671,7 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
nShifts = WEBRTC_SPL_SAT(6, nShifts, 0); nShifts = WEBRTC_SPL_SAT(6, nShifts, 0);
// Shift numerator to Q(nShifts+8+minNorm-stages) // Shift numerator to Q(nShifts+8+minNorm-stages)
numerator = WEBRTC_SPL_LSHIFT_U32(numerator, nShifts); numerator <<= nShifts;
// Shift denominator to Q(nShifts-6+minNorm-stages) // Shift denominator to Q(nShifts-6+minNorm-stages)
tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(inst->initMagnEst[i], 6 - nShifts); tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(inst->initMagnEst[i], 6 - nShifts);
@ -1710,7 +1710,7 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
// Add them together and divide by startup length // Add them together and divide by startup length
noiseU32[i] = WebRtcSpl_DivU32U16(tmpU32no1 + tmpU32no2, END_STARTUP_SHORT); noiseU32[i] = WebRtcSpl_DivU32U16(tmpU32no1 + tmpU32no2, END_STARTUP_SHORT);
// Shift back if necessary // Shift back if necessary
noiseU32[i] = WEBRTC_SPL_LSHIFT_U32(noiseU32[i], nShifts); noiseU32[i] <<= nShifts;
} }
// Update new Q-domain for 'noiseU32' // Update new Q-domain for 'noiseU32'
qNoise = q_domain_to_use; qNoise = q_domain_to_use;
@ -1753,15 +1753,15 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
// calculate post SNR: output in Q11 // calculate post SNR: output in Q11
postLocSnr[i] = 2048; // 1.0 in Q11 postLocSnr[i] = 2048; // 1.0 in Q11
tmpU32no1 = WEBRTC_SPL_LSHIFT_U32((uint32_t)magnU16[i], 6); // Q(6+qMagn) tmpU32no1 = (uint32_t)magnU16[i] << 6; // Q(6+qMagn)
if (postShifts < 0) { if (postShifts < 0) {
tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(noiseU32[i], -postShifts); // Q(6+qMagn) tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(noiseU32[i], -postShifts); // Q(6+qMagn)
} else { } else {
tmpU32no2 = WEBRTC_SPL_LSHIFT_U32(noiseU32[i], postShifts); // Q(6+qMagn) tmpU32no2 = noiseU32[i] << postShifts; // Q(6+qMagn)
} }
if (tmpU32no1 > tmpU32no2) { if (tmpU32no1 > tmpU32no2) {
// Current magnitude larger than noise // Current magnitude larger than noise
tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(tmpU32no1, 11); // Q(17+qMagn) tmpU32no1 <<= 11; // Q(17+qMagn)
if (tmpU32no2 > 0) { if (tmpU32no2 > 0) {
tmpU32no1 /= tmpU32no2; // Q11 tmpU32no1 /= tmpU32no2; // Q11
postLocSnr[i] = WEBRTC_SPL_MIN(satMax, tmpU32no1); // Q11 postLocSnr[i] = WEBRTC_SPL_MIN(satMax, tmpU32no1); // Q11
@ -1772,7 +1772,7 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
// calculate prevNearSnr[i] and save for later instead of recalculating it later // calculate prevNearSnr[i] and save for later instead of recalculating it later
nearMagnEst = WEBRTC_SPL_UMUL_16_16(inst->prevMagnU16[i], inst->noiseSupFilter[i]); // Q(prevQMagn+14) nearMagnEst = WEBRTC_SPL_UMUL_16_16(inst->prevMagnU16[i], inst->noiseSupFilter[i]); // Q(prevQMagn+14)
tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(nearMagnEst, 3); // Q(prevQMagn+17) tmpU32no1 = nearMagnEst << 3; // Q(prevQMagn+17)
tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(inst->prevNoiseU32[i], nShifts); // Q(prevQMagn+6) tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(inst->prevNoiseU32[i], nShifts); // Q(prevQMagn+6)
if (tmpU32no2 > 0) { if (tmpU32no2 > 0) {
@ -1833,7 +1833,7 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
inst->featureSpecDiff = 0x007FFFFF; inst->featureSpecDiff = 0x007FFFFF;
} else { } else {
inst->featureSpecDiff = WEBRTC_SPL_MIN(0x007FFFFF, inst->featureSpecDiff = WEBRTC_SPL_MIN(0x007FFFFF,
WEBRTC_SPL_LSHIFT_U32(tmpU32no3, norm32no1)); tmpU32no3 << norm32no1);
} }
} }
@ -1858,7 +1858,7 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
if (postShifts < 0) { if (postShifts < 0) {
tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(magnU16[i], -postShifts); // Q(prevQNoise) tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(magnU16[i], -postShifts); // Q(prevQNoise)
} else { } else {
tmpU32no2 = WEBRTC_SPL_LSHIFT_U32(magnU16[i], postShifts); // Q(prevQNoise) tmpU32no2 = (uint32_t)magnU16[i] << postShifts; // Q(prevQNoise)
} }
if (prevNoiseU16[i] > tmpU32no2) { if (prevNoiseU16[i] > tmpU32no2) {
sign = -1; sign = -1;
@ -1979,18 +1979,18 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
if (nShifts < 0) { if (nShifts < 0) {
// This case is equivalent with magn < noise which implies curNearSnr = 0; // This case is equivalent with magn < noise which implies curNearSnr = 0;
tmpMagnU32 = (uint32_t)magnU16[i]; // Q(qMagn) tmpMagnU32 = (uint32_t)magnU16[i]; // Q(qMagn)
tmpNoiseU32 = WEBRTC_SPL_LSHIFT_U32(noiseU32[i], -nShifts); // Q(qMagn) tmpNoiseU32 = noiseU32[i] << -nShifts; // Q(qMagn)
} else if (nShifts > 17) { } else if (nShifts > 17) {
tmpMagnU32 = WEBRTC_SPL_LSHIFT_U32(magnU16[i], 17); // Q(qMagn+17) tmpMagnU32 = (uint32_t)magnU16[i] << 17; // Q(qMagn+17)
tmpNoiseU32 = WEBRTC_SPL_RSHIFT_U32(noiseU32[i], nShifts - 17); // Q(qMagn+17) tmpNoiseU32 = WEBRTC_SPL_RSHIFT_U32(noiseU32[i], nShifts - 17); // Q(qMagn+17)
} else { } else {
tmpMagnU32 = WEBRTC_SPL_LSHIFT_U32((uint32_t)magnU16[i], nShifts); // Q(qNoise_prev+11) tmpMagnU32 = (uint32_t)magnU16[i] << nShifts; // Q(qNoise_prev+11)
tmpNoiseU32 = noiseU32[i]; // Q(qNoise_prev+11) tmpNoiseU32 = noiseU32[i]; // Q(qNoise_prev+11)
} }
if (tmpMagnU32 > tmpNoiseU32) { if (tmpMagnU32 > tmpNoiseU32) {
tmpU32no1 = tmpMagnU32 - tmpNoiseU32; // Q(qCur) tmpU32no1 = tmpMagnU32 - tmpNoiseU32; // Q(qCur)
norm32no2 = WEBRTC_SPL_MIN(11, WebRtcSpl_NormU32(tmpU32no1)); norm32no2 = WEBRTC_SPL_MIN(11, WebRtcSpl_NormU32(tmpU32no1));
tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(tmpU32no1, norm32no2); // Q(qCur+norm32no2) tmpU32no1 <<= norm32no2; // Q(qCur+norm32no2)
tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(tmpNoiseU32, 11 - norm32no2); // Q(qCur+norm32no2-11) tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(tmpNoiseU32, 11 - norm32no2); // Q(qCur+norm32no2-11)
if (tmpU32no2 > 0) { if (tmpU32no2 > 0) {
tmpU32no1 /= tmpU32no2; // Q11 tmpU32no1 /= tmpU32no2; // Q11
@ -2033,7 +2033,7 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
inst->prevQMagn = qMagn; inst->prevQMagn = qMagn;
if (norm32no1 > 5) { if (norm32no1 > 5) {
for (i = 0; i < inst->magnLen; i++) { for (i = 0; i < inst->magnLen; i++) {
inst->prevNoiseU32[i] = WEBRTC_SPL_LSHIFT_U32(noiseU32[i], norm32no1 - 5); // Q(qNoise+11) inst->prevNoiseU32[i] = noiseU32[i] << (norm32no1 - 5); // Q(qNoise+11)
inst->prevMagnU16[i] = magnU16[i]; // Q(qMagn) inst->prevMagnU16[i] = magnU16[i]; // Q(qMagn)
} }
} else { } else {

16
webrtc/modules/audio_processing/ns/nsx_core_c.c
View File

@ -8,6 +8,8 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include <assert.h>
#include "webrtc/modules/audio_processing/ns/include/noise_suppression_x.h" #include "webrtc/modules/audio_processing/ns/include/noise_suppression_x.h"
#include "webrtc/modules/audio_processing/ns/nsx_core.h" #include "webrtc/modules/audio_processing/ns/nsx_core.h"
@ -39,9 +41,9 @@ void WebRtcNsx_SpeechNoiseProb(NsxInst_t* inst,
for (i = 0; i < inst->magnLen; i++) { for (i = 0; i < inst->magnLen; i++) {
besselTmpFX32 = (int32_t)postLocSnr[i]; // Q11 besselTmpFX32 = (int32_t)postLocSnr[i]; // Q11
normTmp = WebRtcSpl_NormU32(postLocSnr[i]); normTmp = WebRtcSpl_NormU32(postLocSnr[i]);
num = WEBRTC_SPL_LSHIFT_U32(postLocSnr[i], normTmp); // Q(11+normTmp) num = postLocSnr[i] << normTmp; // Q(11+normTmp)
if (normTmp > 10) { if (normTmp > 10) {
den = WEBRTC_SPL_LSHIFT_U32(priorLocSnr[i], normTmp - 11); // Q(normTmp) den = priorLocSnr[i] << (normTmp - 11); // Q(normTmp)
} else { } else {
den = WEBRTC_SPL_RSHIFT_U32(priorLocSnr[i], 11 - normTmp); // Q(normTmp) den = WEBRTC_SPL_RSHIFT_U32(priorLocSnr[i], 11 - normTmp); // Q(normTmp)
} }
@ -121,11 +123,7 @@ void WebRtcNsx_SpeechNoiseProb(NsxInst_t* inst,
//widthPrior = widthPrior * 2.0; //widthPrior = widthPrior * 2.0;
nShifts++; nShifts++;
} }
tmp32no1 = (int32_t)WebRtcSpl_DivU32U16(WEBRTC_SPL_LSHIFT_U32(tmpU32no2, tmpU32no1 = WebRtcSpl_DivU32U16(tmpU32no2 << nShifts, 25); // Q14
nShifts), 25);
//Q14
tmpU32no1 = WebRtcSpl_DivU32U16(WEBRTC_SPL_LSHIFT_U32(tmpU32no2, nShifts),
25); //Q14
// compute indicator function: sigmoid map // compute indicator function: sigmoid map
// FLOAT code // FLOAT code
// indicator1 = 0.5 * (tanh(sgnMap * widthPrior * // indicator1 = 0.5 * (tanh(sgnMap * widthPrior *
@ -151,8 +149,8 @@ void WebRtcNsx_SpeechNoiseProb(NsxInst_t* inst,
if (inst->featureSpecDiff) { if (inst->featureSpecDiff) {
normTmp = WEBRTC_SPL_MIN(20 - inst->stages, normTmp = WEBRTC_SPL_MIN(20 - inst->stages,
WebRtcSpl_NormU32(inst->featureSpecDiff)); WebRtcSpl_NormU32(inst->featureSpecDiff));
tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(inst->featureSpecDiff, normTmp); assert(normTmp >= 0);
// Q(normTmp-2*stages) tmpU32no1 = inst->featureSpecDiff << normTmp; // Q(normTmp-2*stages)
tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(inst->timeAvgMagnEnergy, tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(inst->timeAvgMagnEnergy,
20 - inst->stages - normTmp); 20 - inst->stages - normTmp);
if (tmpU32no2 > 0) { if (tmpU32no2 > 0) {

12
webrtc/modules/audio_processing/ns/nsx_core_mips.c
View File

@ -8,6 +8,8 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include <assert.h>
#include "webrtc/modules/audio_processing/ns/include/noise_suppression_x.h" #include "webrtc/modules/audio_processing/ns/include/noise_suppression_x.h"
#include "webrtc/modules/audio_processing/ns/nsx_core.h" #include "webrtc/modules/audio_processing/ns/nsx_core.h"
@ -155,11 +157,7 @@ void WebRtcNsx_SpeechNoiseProb(NsxInst_t* inst,
//widthPrior = widthPrior * 2.0; //widthPrior = widthPrior * 2.0;
nShifts++; nShifts++;
} }
tmp32no1 = (int32_t)WebRtcSpl_DivU32U16(WEBRTC_SPL_LSHIFT_U32(tmpU32no2, tmpU32no1 = WebRtcSpl_DivU32U16(tmpU32no2 << nShifts, 25); //Q14
nShifts), 25);
//Q14
tmpU32no1 = WebRtcSpl_DivU32U16(WEBRTC_SPL_LSHIFT_U32(tmpU32no2, nShifts),
25); //Q14
// compute indicator function: sigmoid map // compute indicator function: sigmoid map
// FLOAT code // FLOAT code
// indicator1 = 0.5 * (tanh(sgnMap * widthPrior * // indicator1 = 0.5 * (tanh(sgnMap * widthPrior *
@ -185,8 +183,8 @@ void WebRtcNsx_SpeechNoiseProb(NsxInst_t* inst,
if (inst->featureSpecDiff) { if (inst->featureSpecDiff) {
normTmp = WEBRTC_SPL_MIN(20 - inst->stages, normTmp = WEBRTC_SPL_MIN(20 - inst->stages,
WebRtcSpl_NormU32(inst->featureSpecDiff)); WebRtcSpl_NormU32(inst->featureSpecDiff));
tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(inst->featureSpecDiff, normTmp); assert(normTmp >= 0);
// Q(normTmp-2*stages) tmpU32no1 = inst->featureSpecDiff << normTmp; // Q(normTmp-2*stages)
tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(inst->timeAvgMagnEnergy, tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(inst->timeAvgMagnEnergy,
20 - inst->stages - normTmp); 20 - inst->stages - normTmp);
if (tmpU32no2 > 0) { if (tmpU32no2 > 0) {