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Refactoring common_audio/signal_processing: Remove macro WEBRTC_SPL_UDIV

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This macro is a direct use of the division operator without checking for division by zero. Hence, it is dangerous to use.
This CL replaces the macro with '/' at place.

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/14169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6976 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
bjornv@webrtc.org 2014-08-26 10:25:10 +00:00
parent 1de0cc4079
commit 4f71e22bf9
7 changed files with 23 additions and 31 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
webrtc/common_audio/signal_processing/include/signal_processing_library.h
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@ -48,8 +48,6 @@
((int32_t)(int16_t)(a) * (uint16_t)(b))
#define WEBRTC_SPL_DIV(a, b) \
((int32_t) ((int32_t)(a) / (int32_t)(b)))
#define WEBRTC_SPL_UDIV(a, b) \
((uint32_t) ((uint32_t)(a) / (uint32_t)(b)))
#ifndef WEBRTC_ARCH_ARM_V7
// For ARMv7 platforms, these are inline functions in spl_inl_armv7.h

1
webrtc/common_audio/signal_processing/signal_processing_unittest.cc
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@ -48,7 +48,6 @@ TEST_F(SplTest, MacroTest) {
a = b;
b = -3;
EXPECT_EQ(-5461, WEBRTC_SPL_DIV(a, b));
EXPECT_EQ(0u, WEBRTC_SPL_UDIV(a, b));
EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_32_RSFT16(a, b));
EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_32_RSFT15(a, b));

17
webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c
View File

@ -196,7 +196,7 @@ int32_t WebRtcIsacfix_UpdateUplinkBwImpl(BwEstimatorstr *bweStr,
bweStr->maxBwInv = kInvBandwidth[3];
bweStr->minBwInv = kInvBandwidth[2];
bweStr->recBwInv = WEBRTC_SPL_UDIV(1073741824, (bweStr->recBw + bweStr->recHeaderRate));
bweStr->recBwInv = 1073741824 / (bweStr->recBw + bweStr->recHeaderRate);
}
/* kBitsByteSec is in Q15 */
@ -211,7 +211,7 @@ int32_t WebRtcIsacfix_UpdateUplinkBwImpl(BwEstimatorstr *bweStr,
bweStr->maxBwInv = kInvBandwidth[1];
bweStr->minBwInv = kInvBandwidth[0];
bweStr->recBwInv = WEBRTC_SPL_UDIV(1073741824, (bweStr->recBw + bweStr->recHeaderRate));
bweStr->recBwInv = 1073741824 / (bweStr->recBw + bweStr->recHeaderRate);
}
/* kBitsByteSec is in Q14 */
@ -265,7 +265,7 @@ int32_t WebRtcIsacfix_UpdateUplinkBwImpl(BwEstimatorstr *bweStr,
if ((arrivalTime - bweStr->lastUpdate) > FS3) {
/* Calculate expected number of received packets since last update */
numPktsExpected = WEBRTC_SPL_UDIV(arrivalTime - bweStr->lastUpdate, frameSizeSampl);
numPktsExpected = (arrivalTime - bweStr->lastUpdate) / frameSizeSampl;
/* If received number of packets is more than 90% of expected (922 = 0.9 in Q10): */
/* do the update, else not */
@ -417,8 +417,7 @@ int32_t WebRtcIsacfix_UpdateUplinkBwImpl(BwEstimatorstr *bweStr,
and NOT right shifting recBwAvg 5 bits to an integer
At max 13 bits are used
shift to Q5 */
recBwAvgInv = WEBRTC_SPL_UDIV((uint32_t)(0x80000000 + WEBRTC_SPL_RSHIFT_U32(bweStr->recBwAvg, 1)),
bweStr->recBwAvg);
recBwAvgInv = (0x80000000 + bweStr->recBwAvg / 2) / bweStr->recBwAvg;
/* Calculate Projected arrival time difference */
@ -513,7 +512,7 @@ int32_t WebRtcIsacfix_UpdateUplinkBwImpl(BwEstimatorstr *bweStr,
bweStr->prevSendTime = sendTime;
/* Replace bweStr->recBw by the new value */
bweStr->recBw = WEBRTC_SPL_UDIV(1073741824, bweStr->recBwInv) - bweStr->recHeaderRate;
bweStr->recBw = 1073741824 / bweStr->recBwInv - bweStr->recHeaderRate;
if (immediateSet) {
/* delay correction factor is in Q10 */
@ -530,7 +529,7 @@ int32_t WebRtcIsacfix_UpdateUplinkBwImpl(BwEstimatorstr *bweStr,
bweStr->recJitterShortTerm = 0;
bweStr->recBwInv = WEBRTC_SPL_UDIV(1073741824, bweStr->recBw + bweStr->recHeaderRate);
bweStr->recBwInv = 1073741824 / (bweStr->recBw + bweStr->recHeaderRate);
immediateSet = 0;
}
@ -725,7 +724,7 @@ uint16_t WebRtcIsacfix_GetDownlinkBandwidth(const BwEstimatorstr *bweStr)
/* Q18 rec jitter short term abs is in Q13, multiply it by 2^13 to save precision
2^18 then needs to be shifted 13 bits to 2^31 */
rec_jitter_short_term_abs_inv = WEBRTC_SPL_UDIV(0x80000000, bweStr->recJitterShortTermAbs);
rec_jitter_short_term_abs_inv = 0x80000000u / bweStr->recJitterShortTermAbs;
/* Q27 = 9 + 18 */
jitter_sign = WEBRTC_SPL_MUL(WEBRTC_SPL_RSHIFT_W32(bweStr->recJitterShortTerm, 4), (int32_t)rec_jitter_short_term_abs_inv);
@ -887,7 +886,7 @@ uint16_t WebRtcIsacfix_GetMinBytes(RateModel *State,
//round and shift before conversion
MinRate += 256;
MinRate = WEBRTC_SPL_RSHIFT_W32(MinRate, 9);
MinBytes = (uint16_t)WEBRTC_SPL_UDIV(WEBRTC_SPL_MUL(MinRate, FrameSamples), FS8);
MinBytes = MinRate * FrameSamples / FS8;
/* StreamSize will be adjusted if less than MinBytes */
if (StreamSize < MinBytes) {

2
webrtc/modules/audio_coding/codecs/isac/fix/source/decode_bwe.c
View File

@ -53,7 +53,7 @@ int WebRtcIsacfix_EstimateBandwidth(BwEstimatorstr *bwest_str,
err = WebRtcIsacfix_UpdateUplinkBwImpl(
bwest_str,
rtp_seq_number,
(uint16_t)WEBRTC_SPL_UDIV(WEBRTC_SPL_UMUL(frame_samples,1000), FS),
frame_samples * 1000 / FS,
send_ts,
arr_ts,
(int16_t) packet_size, /* in bytes */

18
webrtc/modules/audio_processing/ns/nsx_core.c
View File

@ -864,8 +864,8 @@ void WebRtcNsx_FeatureParameterExtraction(NsxInst_t* inst, int flag) {
// Guard against division by zero
// If timeAvgMagnEnergy == 0 we have no normalizing statistics and
// therefore can't update the histogram
histIndex = WEBRTC_SPL_UDIV((inst->featureSpecDiff * 5) >> inst->stages,
inst->timeAvgMagnEnergy);
histIndex = ((inst->featureSpecDiff * 5) >> inst->stages) /
inst->timeAvgMagnEnergy;
}
if (histIndex < HIST_PAR_EST) {
inst->histSpecDiff[histIndex]++;
@ -1163,7 +1163,7 @@ void WebRtcNsx_ComputeSpectralDifference(NsxInst_t* inst, uint16_t* magnIn) {
}
if (varPauseUFX > 0) {
// Q(2*(qMagn+norm32-16+minPause))
tmpU32no1 = WEBRTC_SPL_UDIV(tmpU32no2, varPauseUFX);
tmpU32no1 = tmpU32no2 / varPauseUFX;
tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, nShifts);
// Q(2*qMagn)
@ -1681,7 +1681,7 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
// we don't need any division.
tmpU32no1 = 1;
}
tmpU32no2 = WEBRTC_SPL_UDIV(numerator, tmpU32no1); // Q14
tmpU32no2 = numerator / tmpU32no1; // Q14
noiseSupFilterTmp[i] = (uint16_t)WEBRTC_SPL_SAT(16384, tmpU32no2,
(uint32_t)(inst->denoiseBound)); // Q14
}
@ -1764,7 +1764,7 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
// Current magnitude larger than noise
tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(tmpU32no1, 11); // Q(17+qMagn)
if (tmpU32no2 > 0) {
tmpU32no1 = WEBRTC_SPL_UDIV(tmpU32no1, tmpU32no2); // Q11
tmpU32no1 /= tmpU32no2; // Q11
postLocSnr[i] = WEBRTC_SPL_MIN(satMax, tmpU32no1); // Q11
} else {
postLocSnr[i] = satMax;
@ -1777,7 +1777,7 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(inst->prevNoiseU32[i], nShifts); // Q(prevQMagn+6)
if (tmpU32no2 > 0) {
tmpU32no1 = WEBRTC_SPL_UDIV(tmpU32no1, tmpU32no2); // Q11
tmpU32no1 /= tmpU32no2; // Q11
tmpU32no1 = WEBRTC_SPL_MIN(satMax, tmpU32no1); // Q11
} else {
tmpU32no1 = satMax; // Q11
@ -1829,7 +1829,7 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
norm32no1++;
}
tmpU32no3 = WEBRTC_SPL_UMUL(tmpU32no3, tmpU32no2);
tmpU32no3 = WEBRTC_SPL_UDIV(tmpU32no3, inst->timeAvgMagnEnergy);
tmpU32no3 /= inst->timeAvgMagnEnergy;
if (WebRtcSpl_NormU32(tmpU32no3) < norm32no1) {
inst->featureSpecDiff = 0x007FFFFF;
} else {
@ -1994,7 +1994,7 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(tmpU32no1, norm32no2); // Q(qCur+norm32no2)
tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(tmpNoiseU32, 11 - norm32no2); // Q(qCur+norm32no2-11)
if (tmpU32no2 > 0) {
tmpU32no1 = WEBRTC_SPL_UDIV(tmpU32no1, tmpU32no2); // Q11
tmpU32no1 /= tmpU32no2; // Q11
}
curNearSnr = WEBRTC_SPL_MIN(satMax, tmpU32no1); // Q11
}
@ -2011,7 +2011,7 @@ int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFram
tmpU32no1 = (uint32_t)(inst->overdrive)
+ WEBRTC_SPL_RSHIFT_U32(priorSnr + 8192, 14); // Q8
assert(inst->overdrive > 0);
tmpU16no1 = (uint16_t)WEBRTC_SPL_UDIV(priorSnr + (tmpU32no1 >> 1), tmpU32no1); // Q14
tmpU16no1 = (priorSnr + tmpU32no1 / 2) / tmpU32no1; // Q14
inst->noiseSupFilter[i] = WEBRTC_SPL_SAT(16384, tmpU16no1, inst->denoiseBound); // 16384 = Q14(1.0) // Q14
// Weight in the parametric Wiener filter during startup

8
webrtc/modules/audio_processing/ns/nsx_core_c.c
View File

@ -46,7 +46,7 @@ void WebRtcNsx_SpeechNoiseProb(NsxInst_t* inst,
den = WEBRTC_SPL_RSHIFT_U32(priorLocSnr[i], 11 - normTmp); // Q(normTmp)
}
if (den > 0) {
besselTmpFX32 -= WEBRTC_SPL_UDIV(num, den); // Q11
besselTmpFX32 -= num / den; // Q11
} else {
besselTmpFX32 -= num; // Q11
}
@ -157,14 +157,12 @@ void WebRtcNsx_SpeechNoiseProb(NsxInst_t* inst,
20 - inst->stages - normTmp);
if (tmpU32no2 > 0) {
// Q(20 - inst->stages)
tmpU32no1 = WEBRTC_SPL_UDIV(tmpU32no1, tmpU32no2);
tmpU32no1 /= tmpU32no2;
} else {
tmpU32no1 = (uint32_t)(0x7fffffff);
}
}
tmpU32no3 = WEBRTC_SPL_UDIV(WEBRTC_SPL_LSHIFT_U32(inst->thresholdSpecDiff,
17),
25);
tmpU32no3 = (inst->thresholdSpecDiff << 17) / 25;
tmpU32no2 = tmpU32no1 - tmpU32no3;
nShifts = 1;
tmpIndFX = 16384; // Q14(1.0)

6
webrtc/modules/audio_processing/ns/nsx_core_mips.c
View File

@ -190,14 +190,12 @@ void WebRtcNsx_SpeechNoiseProb(NsxInst_t* inst,
20 - inst->stages - normTmp);
if (tmpU32no2 > 0) {
// Q(20 - inst->stages)
tmpU32no1 = WEBRTC_SPL_UDIV(tmpU32no1, tmpU32no2);
tmpU32no1 /= tmpU32no2;
} else {
tmpU32no1 = (uint32_t)(0x7fffffff);
}
}
tmpU32no3 = WEBRTC_SPL_UDIV(WEBRTC_SPL_LSHIFT_U32(inst->thresholdSpecDiff,
17),
25);
tmpU32no3 = (inst->thresholdSpecDiff << 17) / 25;
tmpU32no2 = tmpU32no1 - tmpU32no3;
nShifts = 1;
tmpIndFX = 16384; // Q14(1.0)