common_audio: Replaced WEBRTC_SPL_LSHIFT_U32 with << in audio_processing
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
diff --git a/webrtc/modules/audio_processing/aecm/aecm_core.c b/webrtc/modules/audio_processing/aecm/aecm_core.c
index 6b6d2f5..db81478 100644
--- a/webrtc/modules/audio_processing/aecm/aecm_core.c
+++ b/webrtc/modules/audio_processing/aecm/aecm_core.c
@@ -706,6 +706,15 @@
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(...)
//
// This function calculates the log of energies for nearend, farend and estimated
@@ -751,9 +760,7 @@
if (nearEner)
{
zeros = WebRtcSpl_NormU32(nearEner);
- frac = (int16_t)WEBRTC_SPL_RSHIFT_U32(
- (WEBRTC_SPL_LSHIFT_U32(nearEner, zeros) & 0x7FFFFFFF),
- 23);
+ frac = ExtractFractionPart(nearEner, zeros);
// log2 in Q8
tmp16 += WEBRTC_SPL_LSHIFT_W16((31 - zeros), 8) + frac;
tmp16 -= WEBRTC_SPL_LSHIFT_W16(aecm->dfaNoisyQDomain, 8);
@@ -774,8 +781,7 @@
if (tmpFar)
{
zeros = WebRtcSpl_NormU32(tmpFar);
- frac = (int16_t)WEBRTC_SPL_RSHIFT_U32((WEBRTC_SPL_LSHIFT_U32(tmpFar, zeros)
- & 0x7FFFFFFF), 23);
+ frac = ExtractFractionPart(tmpFar, zeros);
// log2 in Q8
tmp16 += WEBRTC_SPL_LSHIFT_W16((31 - zeros), 8) + frac;
tmp16 -= WEBRTC_SPL_LSHIFT_W16(far_q, 8);
@@ -787,8 +793,7 @@
if (tmpAdapt)
{
zeros = WebRtcSpl_NormU32(tmpAdapt);
- frac = (int16_t)WEBRTC_SPL_RSHIFT_U32((WEBRTC_SPL_LSHIFT_U32(tmpAdapt, zeros)
- & 0x7FFFFFFF), 23);
+ frac = ExtractFractionPart(tmpAdapt, zeros);
//log2 in Q8
tmp16 += WEBRTC_SPL_LSHIFT_W16((31 - zeros), 8) + frac;
tmp16 -= WEBRTC_SPL_LSHIFT_W16(RESOLUTION_CHANNEL16 + far_q, 8);
@@ -800,8 +805,7 @@
if (tmpStored)
{
zeros = WebRtcSpl_NormU32(tmpStored);
- frac = (int16_t)WEBRTC_SPL_RSHIFT_U32((WEBRTC_SPL_LSHIFT_U32(tmpStored, zeros)
- & 0x7FFFFFFF), 23);
+ frac = ExtractFractionPart(tmpStored, zeros);
//log2 in Q8
tmp16 += WEBRTC_SPL_LSHIFT_W16((31 - zeros), 8) + frac;
tmp16 -= WEBRTC_SPL_LSHIFT_W16(RESOLUTION_CHANNEL16 + far_q, 8);
diff --git a/webrtc/modules/audio_processing/agc/digital_agc.c b/webrtc/modules/audio_processing/agc/digital_agc.c
index b15b6e3..e439e09 100644
--- a/webrtc/modules/audio_processing/agc/digital_agc.c
+++ b/webrtc/modules/audio_processing/agc/digital_agc.c
@@ -154,7 +154,7 @@
fracPart = (uint16_t)(absInLevel & 0x00003FFF); // extract the fractional part
tmpU16 = kGenFuncTable[intPart + 1] - kGenFuncTable[intPart]; // Q8
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
// Compensate for negative exponent using the relation:
// log2(1 + 2^-x) = log2(1 + 2^x) - x
diff --git a/webrtc/modules/audio_processing/ns/nsx_core.c b/webrtc/modules/audio_processing/ns/nsx_core.c
index 5a88c12..19ad1a9 100644
--- a/webrtc/modules/audio_processing/ns/nsx_core.c
+++ b/webrtc/modules/audio_processing/ns/nsx_core.c
@@ -620,7 +620,7 @@
}
// Shift fractional part to Q(minNorm-stages)
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
*noise_estimate = (*noise_estimate_avg) * (uint32_t)(inst->blockIndex + 1);
}
@@ -1149,7 +1149,7 @@
tmpU32no1 = (uint32_t)WEBRTC_SPL_ABS_W32(covMagnPauseFX); // Q(prevQMagn+qMagn)
norm32 = WebRtcSpl_NormU32(tmpU32no1) - 16;
if (norm32 > 0) {
- tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(tmpU32no1, norm32); // Q(prevQMagn+qMagn+norm32)
+ tmpU32no1 <<= norm32; // Q(prevQMagn+qMagn+norm32)
} else {
tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, -norm32); // Q(prevQMagn+qMagn+norm32)
}
@@ -1660,7 +1660,7 @@
// numerator = (initMagnEst - noise_estimate * overdrive)
// Result in Q(8+minNorm-stages)
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) {
// Suppression filter coefficient larger than zero, so calculate.
numerator -= tmpU32no1;
@@ -1671,7 +1671,7 @@
nShifts = WEBRTC_SPL_SAT(6, nShifts, 0);
// 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)
tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(inst->initMagnEst[i], 6 - nShifts);
@@ -1710,7 +1710,7 @@
// Add them together and divide by startup length
noiseU32[i] = WebRtcSpl_DivU32U16(tmpU32no1 + tmpU32no2, END_STARTUP_SHORT);
// Shift back if necessary
- noiseU32[i] = WEBRTC_SPL_LSHIFT_U32(noiseU32[i], nShifts);
+ noiseU32[i] <<= nShifts;
}
// Update new Q-domain for 'noiseU32'
qNoise = q_domain_to_use;
@@ -1753,15 +1753,15 @@
// calculate post SNR: output 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) {
tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(noiseU32[i], -postShifts); // Q(6+qMagn)
} else {
- tmpU32no2 = WEBRTC_SPL_LSHIFT_U32(noiseU32[i], postShifts); // Q(6+qMagn)
+ tmpU32no2 = noiseU32[i] << postShifts; // Q(6+qMagn)
}
if (tmpU32no1 > tmpU32no2) {
// Current magnitude larger than noise
- tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(tmpU32no1, 11); // Q(17+qMagn)
+ tmpU32no1 <<= 11; // Q(17+qMagn)
if (tmpU32no2 > 0) {
tmpU32no1 /= tmpU32no2; // Q11
postLocSnr[i] = WEBRTC_SPL_MIN(satMax, tmpU32no1); // Q11
@@ -1772,7 +1772,7 @@
// 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)
- 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)
if (tmpU32no2 > 0) {
@@ -1833,7 +1833,7 @@
inst->featureSpecDiff = 0x007FFFFF;
} else {
inst->featureSpecDiff = WEBRTC_SPL_MIN(0x007FFFFF,
- WEBRTC_SPL_LSHIFT_U32(tmpU32no3, norm32no1));
+ tmpU32no3 << norm32no1);
}
}
@@ -1858,7 +1858,7 @@
if (postShifts < 0) {
tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(magnU16[i], -postShifts); // Q(prevQNoise)
} else {
- tmpU32no2 = WEBRTC_SPL_LSHIFT_U32(magnU16[i], postShifts); // Q(prevQNoise)
+ tmpU32no2 = (uint32_t)magnU16[i] << postShifts; // Q(prevQNoise)
}
if (prevNoiseU16[i] > tmpU32no2) {
sign = -1;
@@ -1979,18 +1979,18 @@
if (nShifts < 0) {
// This case is equivalent with magn < noise which implies curNearSnr = 0;
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) {
- 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)
} 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)
}
if (tmpMagnU32 > tmpNoiseU32) {
tmpU32no1 = tmpMagnU32 - tmpNoiseU32; // Q(qCur)
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)
if (tmpU32no2 > 0) {
tmpU32no1 /= tmpU32no2; // Q11
@@ -2033,7 +2033,7 @@
inst->prevQMagn = qMagn;
if (norm32no1 > 5) {
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)
}
} else {
diff --git a/webrtc/modules/audio_processing/ns/nsx_core_c.c b/webrtc/modules/audio_processing/ns/nsx_core_c.c
index 4472583..2fce49b 100644
--- a/webrtc/modules/audio_processing/ns/nsx_core_c.c
+++ b/webrtc/modules/audio_processing/ns/nsx_core_c.c
@@ -8,6 +8,8 @@
* 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/nsx_core.h"
@@ -39,9 +41,9 @@
for (i = 0; i < inst->magnLen; i++) {
besselTmpFX32 = (int32_t)postLocSnr[i]; // Q11
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) {
- den = WEBRTC_SPL_LSHIFT_U32(priorLocSnr[i], normTmp - 11); // Q(normTmp)
+ den = priorLocSnr[i] << (normTmp - 11); // Q(normTmp)
} else {
den = WEBRTC_SPL_RSHIFT_U32(priorLocSnr[i], 11 - normTmp); // Q(normTmp)
}
@@ -121,11 +123,7 @@
//widthPrior = widthPrior * 2.0;
nShifts++;
}
- tmp32no1 = (int32_t)WebRtcSpl_DivU32U16(WEBRTC_SPL_LSHIFT_U32(tmpU32no2,
- nShifts), 25);
- //Q14
- tmpU32no1 = WebRtcSpl_DivU32U16(WEBRTC_SPL_LSHIFT_U32(tmpU32no2, nShifts),
- 25); //Q14
+ tmpU32no1 = WebRtcSpl_DivU32U16(tmpU32no2 << nShifts, 25); // Q14
// compute indicator function: sigmoid map
// FLOAT code
// indicator1 = 0.5 * (tanh(sgnMap * widthPrior *
@@ -151,8 +149,8 @@
if (inst->featureSpecDiff) {
normTmp = WEBRTC_SPL_MIN(20 - inst->stages,
WebRtcSpl_NormU32(inst->featureSpecDiff));
- tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(inst->featureSpecDiff, normTmp);
- // Q(normTmp-2*stages)
+ assert(normTmp >= 0);
+ tmpU32no1 = inst->featureSpecDiff << normTmp; // Q(normTmp-2*stages)
tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(inst->timeAvgMagnEnergy,
20 - inst->stages - normTmp);
if (tmpU32no2 > 0) {
diff --git a/webrtc/modules/audio_processing/ns/nsx_core_mips.c b/webrtc/modules/audio_processing/ns/nsx_core_mips.c
index 0671627..47b1b5f 100644
--- a/webrtc/modules/audio_processing/ns/nsx_core_mips.c
+++ b/webrtc/modules/audio_processing/ns/nsx_core_mips.c
@@ -8,6 +8,8 @@
* 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/nsx_core.h"
@@ -155,11 +157,7 @@
//widthPrior = widthPrior * 2.0;
nShifts++;
}
- tmp32no1 = (int32_t)WebRtcSpl_DivU32U16(WEBRTC_SPL_LSHIFT_U32(tmpU32no2,
- nShifts), 25);
- //Q14
- tmpU32no1 = WebRtcSpl_DivU32U16(WEBRTC_SPL_LSHIFT_U32(tmpU32no2, nShifts),
- 25); //Q14
+ tmpU32no1 = WebRtcSpl_DivU32U16(tmpU32no2 << nShifts, 25); //Q14
// compute indicator function: sigmoid map
// FLOAT code
// indicator1 = 0.5 * (tanh(sgnMap * widthPrior *
@@ -185,8 +183,8 @@
if (inst->featureSpecDiff) {
normTmp = WEBRTC_SPL_MIN(20 - inst->stages,
WebRtcSpl_NormU32(inst->featureSpecDiff));
- tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(inst->featureSpecDiff, normTmp);
- // Q(normTmp-2*stages)
+ assert(normTmp >= 0);
+ tmpU32no1 = inst->featureSpecDiff << normTmp; // Q(normTmp-2*stages)
tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(inst->timeAvgMagnEnergy,
20 - inst->stages - normTmp);
if (tmpU32no2 > 0) {
