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Merge branch 'improve-autovectorization' into 'master'

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Improvements to help compiler autovectorization

See merge request soundtouch/soundtouch!10
This commit is contained in:
Olli Parviainen 2020-10-13 18:19:08 +00:00
commit 1d42d899ab
6 changed files with 84 additions and 55 deletions
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11
include/STTypes.h
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@ -149,8 +149,9 @@ namespace soundtouch
// floating point samples
typedef float SAMPLETYPE;
// data type for sample accumulation: Use double to utilize full precision.
typedef double LONG_SAMPLETYPE;
// data type for sample accumulation: Use float also here to enable
// efficient autovectorization
typedef float LONG_SAMPLETYPE;
#ifdef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS
// Allow SSE optimizations
@ -159,6 +160,12 @@ namespace soundtouch
#endif // SOUNDTOUCH_INTEGER_SAMPLES
#if ((SOUNDTOUCH_ALLOW_SSE) || (__SSE__) || (SOUNDTOUCH_USE_NEON))
#if SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION
#define ST_SIMD_AVOID_UNALIGNED
#endif
#endif
};
// define ST_NO_EXCEPTION_HANDLING switch to disable throwing std exceptions:

70
source/SoundTouch/FIRFilter.cpp
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@ -60,12 +60,14 @@ FIRFilter::FIRFilter()
length = 0;
lengthDiv8 = 0;
filterCoeffs = NULL;
filterCoeffsStereo = NULL;
}
FIRFilter::~FIRFilter()
{
delete[] filterCoeffs;
delete[] filterCoeffsStereo;
}
@ -78,28 +80,26 @@ uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, ui
// because division is much slower operation than multiplying.
double dScaler = 1.0 / (double)resultDivider;
#endif
// hint compiler autovectorization that loop length is divisible by 8
int ilength = length & -8;
assert(length != 0);
assert(src != NULL);
assert(dest != NULL);
assert(filterCoeffs != NULL);
assert((length != 0) && (length == ilength) && (src != NULL) && (dest != NULL) && (filterCoeffs != NULL));
end = 2 * (numSamples - length);
end = 2 * (numSamples - ilength);
#pragma omp parallel for
for (j = 0; j < end; j += 2)
{
const SAMPLETYPE *ptr;
LONG_SAMPLETYPE suml, sumr;
uint i;
suml = sumr = 0;
ptr = src + j;
for (i = 0; i < length; i ++)
for (int i = 0; i < ilength; i ++)
{
suml += ptr[2 * i] * filterCoeffs[i];
sumr += ptr[2 * i + 1] * filterCoeffs[i];
suml += ptr[2 * i] * filterCoeffsStereo[2 * i];
sumr += ptr[2 * i + 1] * filterCoeffsStereo[2 * i + 1];
}
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
@ -109,14 +109,11 @@ uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, ui
suml = (suml < -32768) ? -32768 : (suml > 32767) ? 32767 : suml;
// saturate to 16 bit integer limits
sumr = (sumr < -32768) ? -32768 : (sumr > 32767) ? 32767 : sumr;
#else
suml *= dScaler;
sumr *= dScaler;
#endif // SOUNDTOUCH_INTEGER_SAMPLES
dest[j] = (SAMPLETYPE)suml;
dest[j + 1] = (SAMPLETYPE)sumr;
}
return numSamples - length;
return numSamples - ilength;
}
@ -130,18 +127,21 @@ uint FIRFilter::evaluateFilterMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint
double dScaler = 1.0 / (double)resultDivider;
#endif
assert(length != 0);
// hint compiler autovectorization that loop length is divisible by 8
int ilength = length & -8;
end = numSamples - length;
assert(ilength != 0);
end = numSamples - ilength;
#pragma omp parallel for
for (j = 0; j < end; j ++)
for (j = 0; j < end; j ++)
{
const SAMPLETYPE *pSrc = src + j;
LONG_SAMPLETYPE sum;
uint i;
int i;
sum = 0;
for (i = 0; i < length; i ++)
for (i = 0; i < ilength; i ++)
{
sum += pSrc[i] * filterCoeffs[i];
}
@ -149,8 +149,6 @@ uint FIRFilter::evaluateFilterMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint
sum >>= resultDivFactor;
// saturate to 16 bit integer limits
sum = (sum < -32768) ? -32768 : (sum > 32767) ? 32767 : sum;
#else
sum *= dScaler;
#endif // SOUNDTOUCH_INTEGER_SAMPLES
dest[j] = (SAMPLETYPE)sum;
}
@ -174,14 +172,18 @@ uint FIRFilter::evaluateFilterMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uin
assert(filterCoeffs != NULL);
assert(numChannels < 16);
end = numChannels * (numSamples - length);
// hint compiler autovectorization that loop length is divisible by 8
int ilength = length & -8;
end = numChannels * (numSamples - ilength);
#pragma omp parallel for
for (j = 0; j < end; j += numChannels)
{
const SAMPLETYPE *ptr;
LONG_SAMPLETYPE sums[16];
uint c, i;
uint c;
int i;
for (c = 0; c < numChannels; c ++)
{
@ -190,7 +192,7 @@ uint FIRFilter::evaluateFilterMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uin
ptr = src + j;
for (i = 0; i < length; i ++)
for (i = 0; i < ilength; i ++)
{
SAMPLETYPE coef=filterCoeffs[i];
for (c = 0; c < numChannels; c ++)
@ -204,13 +206,11 @@ uint FIRFilter::evaluateFilterMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uin
{
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
sums[c] >>= resultDivFactor;
#else
sums[c] *= dScaler;
#endif // SOUNDTOUCH_INTEGER_SAMPLES
dest[j+c] = (SAMPLETYPE)sums[c];
}
}
return numSamples - length;
return numSamples - ilength;
}
@ -222,6 +222,13 @@ void FIRFilter::setCoefficients(const SAMPLETYPE *coeffs, uint newLength, uint u
assert(newLength > 0);
if (newLength % 8) ST_THROW_RT_ERROR("FIR filter length not divisible by 8");
#ifdef SOUNDTOUCH_FLOAT_SAMPLES
// scale coefficients already here if using floating samples
double scale = 1.0 / resultDivider;
#else
short scale = 1;
#endif
lengthDiv8 = newLength / 8;
length = lengthDiv8 * 8;
assert(length == newLength);
@ -231,7 +238,16 @@ void FIRFilter::setCoefficients(const SAMPLETYPE *coeffs, uint newLength, uint u
delete[] filterCoeffs;
filterCoeffs = new SAMPLETYPE[length];
memcpy(filterCoeffs, coeffs, length * sizeof(SAMPLETYPE));
delete[] filterCoeffsStereo;
filterCoeffsStereo = new SAMPLETYPE[length*2];
for (uint i = 0; i < length; i ++)
{
filterCoeffs[i] = (SAMPLETYPE)(coeffs[i] * scale);
// create also stereo set of filter coefficients: this allows compiler
// to autovectorize filter evaluation much more efficiently
filterCoeffsStereo[2 * i] = (SAMPLETYPE)(coeffs[i] * scale);
filterCoeffsStereo[2 * i + 1] = (SAMPLETYPE)(coeffs[i] * scale);
}
}

1
source/SoundTouch/FIRFilter.h
View File

@ -57,6 +57,7 @@ protected:
// Memory for filter coefficients
SAMPLETYPE *filterCoeffs;
SAMPLETYPE *filterCoeffsStereo;
virtual uint evaluateFilterStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,

2
source/SoundTouch/InterpolateLinear.cpp
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@ -142,7 +142,7 @@ int InterpolateLinearInteger::transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE
LONG_SAMPLETYPE temp, vol1;
assert(iFract < SCALE);
vol1 = (SCALE - iFract);
vol1 = (LONG_SAMPLETYPE)(SCALE - iFract);
for (int c = 0; c < numChannels; c ++)
{
temp = vol1 * src[c] + iFract * src[c + numChannels];

53
source/SoundTouch/TDStretch.cpp
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@ -54,11 +54,6 @@ using namespace soundtouch;
#define max(x, y) (((x) > (y)) ? (x) : (y))
#if defined(SOUNDTOUCH_USE_NEON) && defined(SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION)
// SIMD mode, allow shortcuts to avoid operations that aren't aligned to 16-byte boundary
#define ST_SIMD_AVOID_UNALIGNED
#endif
/*****************************************************************************
*
* Constant definitions
@ -207,7 +202,7 @@ void TDStretch::overlapMono(SAMPLETYPE *pOutput, const SAMPLETYPE *pInput) const
m1 = (SAMPLETYPE)0;
m2 = (SAMPLETYPE)overlapLength;
for (i = 0; i < overlapLength ; i ++)
for (i = 0; i < overlapLength ; i ++)
{
pOutput[i] = (pInput[i] * m1 + pMidBuffer[i] * m2 ) / overlapLength;
m1 += 1;
@ -315,7 +310,7 @@ int TDStretch::seekBestOverlapPositionFull(const SAMPLETYPE *refPos)
bestCorr = (bestCorr + 0.1) * 0.75;
#pragma omp parallel for
for (i = 1; i < seekLength; i ++)
for (i = 1; i < seekLength; i ++)
{
double corr;
// Calculates correlation value for the mixing position corresponding to 'i'
@ -682,18 +677,16 @@ void TDStretch::processSamples()
isBeginning = false;
int skip = (int)(tempo * overlapLength + 0.5);
#ifdef SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION
#ifdef SOUNDTOUCH_ALLOW_SSE
// if SSE mode, round the skip amount to value corresponding to aligned memory address
if (channels == 1)
{
skip &= -4;
}
else if (channels == 2)
{
skip &= -2;
}
#endif
#ifdef ST_SIMD_AVOID_UNALIGNED
// in SIMD mode, round the skip amount to value corresponding to aligned memory address
if (channels == 1)
{
skip &= -4;
}
else if (channels == 2)
{
skip &= -2;
}
#endif
skipFract -= skip;
assert(nominalSkip >= -skipFract);
@ -823,7 +816,7 @@ void TDStretch::overlapStereo(short *poutput, const short *input) const
short temp;
int cnt2;
for (i = 0; i < overlapLength ; i ++)
for (i = 0; i < overlapLength ; i ++)
{
temp = (short)(overlapLength - i);
cnt2 = 2 * i;
@ -897,9 +890,12 @@ double TDStretch::calcCrossCorr(const short *mixingPos, const short *compare, do
if (((ulongptr)mixingPos) & 15) return -1e50;
#endif
// hint compiler autovectorization that loop length is divisible by 8
int ilength = (channels * overlapLength) & -8;
corr = lnorm = 0;
// Same routine for stereo and mono
for (i = 0; i < channels * overlapLength; i += 2)
for (i = 0; i < ilength; i += 2)
{
corr += (mixingPos[i] * compare[i] +
mixingPos[i + 1] * compare[i + 1]) >> overlapDividerBitsNorm;
@ -931,6 +927,9 @@ double TDStretch::calcCrossCorrAccumulate(const short *mixingPos, const short *c
long lnorm;
int i;
// hint compiler autovectorization that loop length is divisible by 8
int ilength = (channels * overlapLength) & -8;
// cancel first normalizer tap from previous round
lnorm = 0;
for (i = 1; i <= channels; i ++)
@ -940,7 +939,7 @@ double TDStretch::calcCrossCorrAccumulate(const short *mixingPos, const short *c
corr = 0;
// Same routine for stereo and mono.
for (i = 0; i < channels * overlapLength; i += 2)
for (i = 0; i < ilength; i += 2)
{
corr += (mixingPos[i] * compare[i] +
mixingPos[i + 1] * compare[i + 1]) >> overlapDividerBitsNorm;
@ -1053,9 +1052,12 @@ double TDStretch::calcCrossCorr(const float *mixingPos, const float *compare, do
if (((ulongptr)mixingPos) & 15) return -1e50;
#endif
// hint compiler autovectorization that loop length is divisible by 8
int ilength = (channels * overlapLength) & -8;
corr = norm = 0;
// Same routine for stereo and mono
for (i = 0; i < channels * overlapLength; i ++)
for (i = 0; i < ilength; i ++)
{
corr += mixingPos[i] * compare[i];
norm += mixingPos[i] * mixingPos[i];
@ -1080,8 +1082,11 @@ double TDStretch::calcCrossCorrAccumulate(const float *mixingPos, const float *c
norm -= mixingPos[-i] * mixingPos[-i];
}
// hint compiler autovectorization that loop length is divisible by 8
int ilength = (channels * overlapLength) & -8;
// Same routine for stereo and mono
for (i = 0; i < channels * overlapLength; i ++)
for (i = 0; i < ilength; i ++)
{
corr += mixingPos[i] * compare[i];
}

2
source/SoundTouch/sse_optimized.cpp
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@ -80,7 +80,7 @@ double TDStretchSSE::calcCrossCorr(const float *pV1, const float *pV2, double &a
// Compile-time define SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION is provided
// for choosing if this little cheating is allowed.
#ifdef SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION
#ifdef ST_SIMD_AVOID_UNALIGNED
// Little cheating allowed, return valid correlation only for
// aligned locations, meaning every second round for stereo sound.