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Refactoring common_audio: Replace trivial multiplication macro

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This multiplication macro literally use the '*' operator, so there is no need for it.

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

Review URL: https://webrtc-codereview.appspot.com/22109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6964 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
bjornv@webrtc.org
2014-08-25 11:42:42 +00:00
parent d32c4389ac
commit 926707b167
9 changed files with 18 additions and 24 deletions
Download Patch File Download Diff File Expand all files Collapse all files

1
webrtc/common_audio/signal_processing/include/signal_processing_library.h
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@ -85,7 +85,6 @@
(C + (B >> 16) * A + (((uint32_t)(0x0000FFFF & B) * A) >> 16))
#define WEBRTC_SPL_SAT(a, b, c) (b > a ? a : b < c ? c : b)
#define WEBRTC_SPL_MUL_32_16(a, b) ((a) * (b))
// Shifting with negative numbers allowed
// Positive means left shift

1
webrtc/common_audio/signal_processing/signal_processing_unittest.cc
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@ -60,7 +60,6 @@ TEST_F(SplTest, MacroTest) {
EXPECT_EQ(21, WEBRTC_SPL_SAT(a, A, B));
EXPECT_EQ(21, WEBRTC_SPL_SAT(a, B, A));
EXPECT_EQ(-49149, WEBRTC_SPL_MUL_32_16(a, b));
// Shifting with negative numbers allowed
int shift_amount = 1; // Workaround compiler warning using variable here.

2
webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
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@ -313,7 +313,7 @@ static void CalcRootInvArSpec(const int16_t *ARCoefQ12,
for (k = 1; k < (AR_ORDER); k += 2) {
for (n = 0; n < FRAMESAMPLES/8; n++)
summQ16[n] += WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_32_16(CorrQ11[k+1],WebRtcIsacfix_kCos[k][n]) + 2, 2);
summQ16[n] += ((CorrQ11[k + 1] * WebRtcIsacfix_kCos[k][n]) + 2) >> 2;
}
CS_ptrQ9 = WebRtcIsacfix_kCos[0];

5
webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.c
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@ -66,7 +66,7 @@ static __inline void Intrp1DQ8(int32_t *x, int32_t *fx, int32_t *y, int32_t *fy)
r32=fx[1]-fx[2];
q32=fx[0]-fx[1];
nom32=q32+r32;
den32=WEBRTC_SPL_MUL_32_16((q32-r32), 2);
den32 = (q32 - r32) * 2;
if (nom32<0)
sign1=-1;
if (den32<0)
@ -74,7 +74,8 @@ static __inline void Intrp1DQ8(int32_t *x, int32_t *fx, int32_t *y, int32_t *fy)
/* t = (q32+r32)/(2*(q32-r32)) = (fx[0]-fx[1] + fx[1]-fx[2])/(2 * fx[0]-fx[1] - (fx[1]-fx[2]))*/
/* (Signs are removed because WebRtcSpl_DivResultInQ31 can't handle negative numbers) */
t32=WebRtcSpl_DivResultInQ31(WEBRTC_SPL_MUL_32_16(nom32, sign1),WEBRTC_SPL_MUL_32_16(den32, sign2)); /* t in Q31, without signs */
/* t in Q31, without signs */
t32 = WebRtcSpl_DivResultInQ31(nom32 * sign1, den32 * sign2);
t16=(int16_t)WEBRTC_SPL_RSHIFT_W32(t32, 23); /* Q8 */
t16=t16*sign1*sign2; /* t in Q8 with signs */

3
webrtc/modules/audio_processing/aecm/aecm_core_c.c
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@ -460,8 +460,7 @@ int WebRtcAecm_ProcessBlock(AecmCore_t * aecm,
// Far end signal through channel estimate in Q8
// How much can we shift right to preserve resolution
tmp32no1 = echoEst32[i] - aecm->echoFilt[i];
aecm->echoFilt[i] += WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_32_16(tmp32no1,
50), 8);
aecm->echoFilt[i] += (tmp32no1 * 50) >> 8;
zeros32 = WebRtcSpl_NormW32(aecm->echoFilt[i]) + 1;
zeros16 = WebRtcSpl_NormW16(supGain) + 1;

3
webrtc/modules/audio_processing/aecm/aecm_core_mips.c
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@ -970,8 +970,7 @@ int WebRtcAecm_ProcessBlock(AecmCore_t* aecm,
// Far end signal through channel estimate in Q8
// How much can we shift right to preserve resolution
tmp32no1 = echoEst32[i] - aecm->echoFilt[i];
aecm->echoFilt[i] += WEBRTC_SPL_RSHIFT_W32(
WEBRTC_SPL_MUL_32_16(tmp32no1, 50), 8);
aecm->echoFilt[i] += (tmp32no1 * 50) >> 8;
zeros32 = WebRtcSpl_NormW32(aecm->echoFilt[i]) + 1;
zeros16 = WebRtcSpl_NormW16(supGain) + 1;

6
webrtc/modules/audio_processing/agc/digital_agc.c
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@ -187,7 +187,7 @@ int32_t WebRtcAgc_CalculateGainTable(int32_t *gainTable, // Q16
}
}
numFIX = WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_U16(maxGain, constMaxGain), 6); // Q14
numFIX -= WEBRTC_SPL_MUL_32_16((int32_t)logApprox, diffGain); // Q14
numFIX -= (int32_t)logApprox * diffGain; // Q14
// Calculate ratio
// Shift |numFIX| as much as possible.
@ -237,13 +237,13 @@ int32_t WebRtcAgc_CalculateGainTable(int32_t *gainTable, // Q16
{
tmp16 = WEBRTC_SPL_LSHIFT_W16(2, 14) - constLinApprox;
tmp32no2 = WEBRTC_SPL_LSHIFT_W32(1, 14) - fracPart;
tmp32no2 = WEBRTC_SPL_MUL_32_16(tmp32no2, tmp16);
tmp32no2 *= tmp16;
tmp32no2 = WEBRTC_SPL_RSHIFT_W32(tmp32no2, 13);
tmp32no2 = WEBRTC_SPL_LSHIFT_W32(1, 14) - tmp32no2;
} else
{
tmp16 = constLinApprox - WEBRTC_SPL_LSHIFT_W16(1, 14);
tmp32no2 = WEBRTC_SPL_MUL_32_16(fracPart, tmp16);
tmp32no2 = fracPart * tmp16;
tmp32no2 = WEBRTC_SPL_RSHIFT_W32(tmp32no2, 13);
}
fracPart = (uint16_t)tmp32no2;

12
webrtc/modules/audio_processing/ns/nsx_core.c
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@ -885,14 +885,14 @@ void WebRtcNsx_FeatureParameterExtraction(NsxInst_t* inst, int flag) {
tmp32 = WEBRTC_SPL_MUL_16_16(inst->histLrt[i], j);
avgHistLrtFX += tmp32;
numHistLrt += inst->histLrt[i];
avgSquareHistLrtFX += WEBRTC_SPL_MUL_32_16(tmp32, j);
avgSquareHistLrtFX += tmp32 * j;
}
avgHistLrtComplFX = avgHistLrtFX;
for (; i < HIST_PAR_EST; i++) {
j = (2 * i + 1);
tmp32 = WEBRTC_SPL_MUL_16_16(inst->histLrt[i], j);
avgHistLrtComplFX += tmp32;
avgSquareHistLrtFX += WEBRTC_SPL_MUL_32_16(tmp32, j);
avgSquareHistLrtFX += tmp32 * j;
}
fluctLrtFX = WEBRTC_SPL_MUL(avgSquareHistLrtFX, numHistLrt);
fluctLrtFX -= WEBRTC_SPL_MUL(avgHistLrtFX, avgHistLrtComplFX);
@ -1083,7 +1083,7 @@ void WebRtcNsx_ComputeSpectralFlatness(NsxInst_t* inst, uint16_t* magn) {
//time average update of spectral flatness feature
tmp32 = currentSpectralFlatness - (int32_t)inst->featureSpecFlat; // Q10
tmp32 = WEBRTC_SPL_MUL_32_16(SPECT_FLAT_TAVG_Q14, tmp32); // Q24
tmp32 *= SPECT_FLAT_TAVG_Q14; // Q24
inst->featureSpecFlat = (uint32_t)((int32_t)inst->featureSpecFlat
+ WEBRTC_SPL_RSHIFT_W32(tmp32, 14)); // Q10
// done with flatness feature
@ -1135,7 +1135,7 @@ void WebRtcNsx_ComputeSpectralDifference(NsxInst_t* inst, uint16_t* magnIn) {
tmp16no1 = (int16_t)((int32_t)magnIn[i] - avgMagnFX);
tmp32no2 = inst->avgMagnPause[i] - avgPauseFX;
varMagnUFX += (uint32_t)WEBRTC_SPL_MUL_16_16(tmp16no1, tmp16no1); // Q(2*qMagn)
tmp32no1 = WEBRTC_SPL_MUL_32_16(tmp32no2, tmp16no1); // Q(prevQMagn+qMagn)
tmp32no1 = tmp32no2 * tmp16no1; // Q(prevQMagn+qMagn)
covMagnPauseFX += tmp32no1; // Q(prevQMagn+qMagn)
tmp32no1 = WEBRTC_SPL_RSHIFT_W32(tmp32no2, nShifts); // Q(prevQMagn-minPause)
varPauseUFX += (uint32_t)WEBRTC_SPL_MUL(tmp32no1, tmp32no1); // Q(2*(prevQMagn-minPause))
@ -1940,12 +1940,12 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
if (nonSpeechProbFinal[i] > ONE_MINUS_PROB_RANGE_Q8) {
if (nShifts < 0) {
tmp32no1 = (int32_t)magnU16[i] - tmp32no2; // Q(qMagn)
tmp32no1 = WEBRTC_SPL_MUL_32_16(tmp32no1, ONE_MINUS_GAMMA_PAUSE_Q8); // Q(8+prevQMagn+nShifts)
tmp32no1 *= ONE_MINUS_GAMMA_PAUSE_Q8; // Q(8+prevQMagn+nShifts)
tmp32no1 = WEBRTC_SPL_RSHIFT_W32(tmp32no1 + 128, 8); // Q(qMagn)
} else {
tmp32no1 = WEBRTC_SPL_LSHIFT_W32((int32_t)magnU16[i], nShifts)
- inst->avgMagnPause[i]; // Q(qMagn+nShifts)
tmp32no1 = WEBRTC_SPL_MUL_32_16(tmp32no1, ONE_MINUS_GAMMA_PAUSE_Q8); // Q(8+prevQMagn+nShifts)
tmp32no1 *= ONE_MINUS_GAMMA_PAUSE_Q8; // Q(8+prevQMagn+nShifts)
tmp32no1 = WEBRTC_SPL_RSHIFT_W32(tmp32no1 + (128 << nShifts), 8 + nShifts); // Q(qMagn)
}
tmp32no2 += tmp32no1; // Q(qMagn)

9
webrtc/modules/audio_processing/ns/nsx_core_c.c
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@ -62,8 +62,7 @@ void WebRtcNsx_SpeechNoiseProb(NsxInst_t* inst,
frac32 = tmp32 + 37;
// tmp32 = log2(priorLocSnr[i])
tmp32 = (int32_t)(((31 - zeros) << 12) + frac32) - (11 << 12); // Q12
logTmp = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_32_16(tmp32, 178), 8);
// log2(priorLocSnr[i])*log(2)
logTmp = (tmp32 * 178) >> 8; // log2(priorLocSnr[i])*log(2)
tmp32no1 = WEBRTC_SPL_RSHIFT_W32(logTmp + inst->logLrtTimeAvgW32[i], 1);
// Q12
inst->logLrtTimeAvgW32[i] += (besselTmpFX32 - tmp32no1); // Q12
@ -248,14 +247,12 @@ void WebRtcNsx_SpeechNoiseProb(NsxInst_t* inst,
if (normTmp + normTmp2 < 15) {
invLrtFX = WEBRTC_SPL_RSHIFT_W32(invLrtFX, 15 - normTmp2 - normTmp);
// Q(normTmp+normTmp2-7)
tmp32no1 = WEBRTC_SPL_MUL_32_16(invLrtFX,
(16384 - inst->priorNonSpeechProb));
tmp32no1 = invLrtFX * (16384 - inst->priorNonSpeechProb);
// Q(normTmp+normTmp2+7)
invLrtFX = WEBRTC_SPL_SHIFT_W32(tmp32no1, 7 - normTmp - normTmp2);
// Q14
} else {
tmp32no1 = WEBRTC_SPL_MUL_32_16(invLrtFX,
(16384 - inst->priorNonSpeechProb));
tmp32no1 = invLrtFX * (16384 - inst->priorNonSpeechProb);
// Q22
invLrtFX = WEBRTC_SPL_RSHIFT_W32(tmp32no1, 8); // Q14
}