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https://github.com/JulianGro/overte.git
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AVX2 optimized audio resampler
This commit is contained in:
Ken Cooke
parent
f2bedc546f
commit
34f46b860b
2 changed files with 163 additions and 60 deletions
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@ -544,22 +544,9 @@ static const float prototypeFilter[PROTOTYPE_TAPS * PROTOTYPE_PHASES] = {
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8.99882100e-06f, 7.61267073e-06f, 6.57702907e-06f, 5.59829210e-06f, 4.27698546e-06f, 1.03248674e-05f,
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};
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//
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// polyphase filter
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//
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static const int SRC_PHASEBITS = 8;
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static const int SRC_PHASES = (1 << SRC_PHASEBITS);
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static const int SRC_FRACBITS = 32 - SRC_PHASEBITS;
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static const uint32_t SRC_FRACMASK = (1 << SRC_FRACBITS) - 1;
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static const float QFRAC_TO_FLOAT = 1.0f / (1 << SRC_FRACBITS);
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static const float Q32_TO_FLOAT = 1.0f / (1ULL << 32);
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// blocking size in frames, chosen so block processing fits in L1 cache
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static const int SRC_BLOCK = 1024;
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#define lo32(a) ((uint32_t)(a))
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#define hi32(a) ((int32_t)((a) >> 32))
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// high/low part of int64_t
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#define LO32(a) ((uint32_t)(a))
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#define HI32(a) ((int32_t)((a) >> 32))
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//
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// Portable aligned malloc/free
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@ -610,8 +597,8 @@ static void cubicInterpolation(const float* input, float* output, int inputSize,
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// Lagrange interpolation using Farrow structure
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for (int j = 0; j < outputSize; j++) {
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int32_t i = hi32(offset);
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uint32_t f = lo32(offset);
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int32_t i = HI32(offset);
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uint32_t f = LO32(offset);
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// values outside the window are zero
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float x0 = (i - 1 < 0) ? 0.0f : input[i - 1];
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@ -649,7 +636,7 @@ int AudioSRC::createRationalFilter(int upFactor, int downFactor, float gain) {
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numTaps = (numCoefs + upFactor - 1) / upFactor;
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gain *= (float)oldCoefs / numCoefs;
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}
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numTaps = (numTaps + 3) & ~3; // SIMD4
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numTaps = (numTaps + 7) & ~7; // SIMD8
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// interpolate the coefficients of the prototype filter
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float* tempFilter = new float[numTaps * numPhases];
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@ -658,7 +645,7 @@ int AudioSRC::createRationalFilter(int upFactor, int downFactor, float gain) {
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cubicInterpolation(prototypeFilter, tempFilter, prototypeCoefs, numCoefs, gain);
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// create the polyphase filter
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_polyphaseFilter = (float*)aligned_malloc(numTaps * numPhases * sizeof(float), 16); // SIMD4
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_polyphaseFilter = (float*)aligned_malloc(numTaps * numPhases * sizeof(float), 32); // SIMD8
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// rearrange into polyphase form, ordered by use
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for (int i = 0; i < numPhases; i++) {
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@ -699,7 +686,7 @@ int AudioSRC::createIrrationalFilter(int upFactor, int downFactor, float gain) {
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numTaps = (numCoefs + upFactor - 1) / upFactor;
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gain *= (float)oldCoefs / numCoefs;
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}
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numTaps = (numTaps + 3) & ~3; // SIMD4
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numTaps = (numTaps + 7) & ~7; // SIMD8
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// interpolate the coefficients of the prototype filter
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float* tempFilter = new float[numTaps * numPhases];
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@ -708,7 +695,7 @@ int AudioSRC::createIrrationalFilter(int upFactor, int downFactor, float gain) {
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cubicInterpolation(prototypeFilter, tempFilter, prototypeCoefs, numCoefs, gain);
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// create the polyphase filter, with extra phase at the end to simplify coef interpolation
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_polyphaseFilter = (float*)aligned_malloc(numTaps * (numPhases + 1) * sizeof(float), 16); // SIMD4
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_polyphaseFilter = (float*)aligned_malloc(numTaps * (numPhases + 1) * sizeof(float), 32); // SIMD8
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// rearrange into polyphase form, ordered by fractional delay
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for (int phase = 0; phase < numPhases; phase++) {
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@ -741,14 +728,14 @@ int AudioSRC::createIrrationalFilter(int upFactor, int downFactor, float gain) {
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#include <emmintrin.h>
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int AudioSRC::multirateFilter1(const float* input0, float* output0, int inputFrames) {
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int AudioSRC::multirateFilter1_SSE(const float* input0, float* output0, int inputFrames) {
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int outputFrames = 0;
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assert((_numTaps & 0x3) == 0); // SIMD4
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assert(_numTaps % 4 == 0); // SIMD4
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if (_step == 0) { // rational
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int32_t i = hi32(_offset);
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int32_t i = HI32(_offset);
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while (i < inputFrames) {
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@ -761,7 +748,7 @@ int AudioSRC::multirateFilter1(const float* input0, float* output0, int inputFra
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//float coef = c0[j];
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__m128 coef0 = _mm_loadu_ps(&c0[j]);
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//acc0 += input0[i + j] * coef;
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//acc += input[i + j] * coef;
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acc0 = _mm_add_ps(_mm_mul_ps(_mm_loadu_ps(&input0[i + j]), coef0), acc0);
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}
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@ -781,10 +768,10 @@ int AudioSRC::multirateFilter1(const float* input0, float* output0, int inputFra
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} else { // irrational
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while (hi32(_offset) < inputFrames) {
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while (HI32(_offset) < inputFrames) {
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int32_t i = hi32(_offset);
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uint32_t f = lo32(_offset);
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int32_t i = HI32(_offset);
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uint32_t f = LO32(_offset);
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uint32_t phase = f >> SRC_FRACBITS;
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__m128 frac = _mm_set1_ps((f & SRC_FRACMASK) * QFRAC_TO_FLOAT);
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@ -802,7 +789,7 @@ int AudioSRC::multirateFilter1(const float* input0, float* output0, int inputFra
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coef1 = _mm_sub_ps(coef1, coef0);
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coef0 = _mm_add_ps(_mm_mul_ps(coef1, frac), coef0);
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//acc0 += input0[i + j] * coef;
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//acc += input[i + j] * coef;
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acc0 = _mm_add_ps(_mm_mul_ps(_mm_loadu_ps(&input0[i + j]), coef0), acc0);
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}
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@ -821,14 +808,14 @@ int AudioSRC::multirateFilter1(const float* input0, float* output0, int inputFra
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return outputFrames;
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}
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int AudioSRC::multirateFilter2(const float* input0, const float* input1, float* output0, float* output1, int inputFrames) {
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int AudioSRC::multirateFilter2_SSE(const float* input0, const float* input1, float* output0, float* output1, int inputFrames) {
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int outputFrames = 0;
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assert((_numTaps & 0x3) == 0); // SIMD4
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assert(_numTaps % 4 == 0); // SIMD4
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if (_step == 0) { // rational
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int32_t i = hi32(_offset);
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int32_t i = HI32(_offset);
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while (i < inputFrames) {
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@ -842,7 +829,7 @@ int AudioSRC::multirateFilter2(const float* input0, const float* input1, float*
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//float coef = c0[j];
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__m128 coef0 = _mm_loadu_ps(&c0[j]);
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//acc0 += input0[i + j] * coef;
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//acc += input[i + j] * coef;
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acc0 = _mm_add_ps(_mm_mul_ps(_mm_loadu_ps(&input0[i + j]), coef0), acc0);
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acc1 = _mm_add_ps(_mm_mul_ps(_mm_loadu_ps(&input1[i + j]), coef0), acc1);
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}
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@ -866,10 +853,10 @@ int AudioSRC::multirateFilter2(const float* input0, const float* input1, float*
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} else { // irrational
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while (hi32(_offset) < inputFrames) {
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while (HI32(_offset) < inputFrames) {
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int32_t i = hi32(_offset);
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uint32_t f = lo32(_offset);
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int32_t i = HI32(_offset);
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uint32_t f = LO32(_offset);
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uint32_t phase = f >> SRC_FRACBITS;
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__m128 frac = _mm_set1_ps((f & SRC_FRACMASK) * QFRAC_TO_FLOAT);
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@ -888,7 +875,7 @@ int AudioSRC::multirateFilter2(const float* input0, const float* input1, float*
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coef1 = _mm_sub_ps(coef1, coef0);
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coef0 = _mm_add_ps(_mm_mul_ps(coef1, frac), coef0);
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//acc0 += input0[i + j] * coef;
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//acc += input[i + j] * coef;
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acc0 = _mm_add_ps(_mm_mul_ps(_mm_loadu_ps(&input0[i + j]), coef0), acc0);
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acc1 = _mm_add_ps(_mm_mul_ps(_mm_loadu_ps(&input1[i + j]), coef0), acc1);
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}
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@ -911,6 +898,107 @@ int AudioSRC::multirateFilter2(const float* input0, const float* input1, float*
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return outputFrames;
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}
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//
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// Detect AVX/AVX2 support
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//
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#if defined(_MSC_VER)
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#include <intrin.h>
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static bool cpuSupportsAVX() {
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int info[4];
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int mask = (1 << 27) | (1 << 28); // OSXSAVE and AVX
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__cpuidex(info, 0x1, 0);
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bool result = false;
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if ((info[2] & mask) == mask) {
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if ((_xgetbv(_XCR_XFEATURE_ENABLED_MASK) & 0x6) == 0x6) {
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result = true;
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}
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}
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return result;
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}
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static bool cpuSupportsAVX2() {
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int info[4];
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int mask = (1 << 5); // AVX2
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bool result = false;
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if (cpuSupportsAVX()) {
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__cpuidex(info, 0x7, 0);
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if ((info[1] & mask) == mask) {
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result = true;
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}
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}
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return result;
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}
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#elif defined(__GNUC__)
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#include <cpuid.h>
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static bool cpuSupportsAVX() {
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unsigned int eax, ebx, ecx, edx;
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unsigned int mask = (1 << 27) | (1 << 28); // OSXSAVE and AVX
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bool result = false;
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if (__get_cpuid(0x1, &eax, &ebx, &ecx, &edx) && ((ecx & mask) == mask)) {
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__asm__("xgetbv" : "=a"(eax), "=d"(edx) : "c"(0));
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if ((eax & 0x6) == 0x6) {
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result = true;
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}
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}
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return result;
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}
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static bool cpuSupportsAVX2() {
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unsigned int eax, ebx, ecx, edx;
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unsigned mask = (1 << 5); // AVX2
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bool result = false;
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if (cpuSupportsAVX()) {
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if (__get_cpuid(0x7, &eax, &ebx, &ecx, &edx) && ((ebx & mask) == mask)) {
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result = true;
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}
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}
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return result;
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}
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#else
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static bool cpuSupportsAVX() {
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return false;
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}
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static bool cpuSupportsAVX2() {
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return false;
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}
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#endif
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//
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// Runtime CPU dispatch
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//
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int AudioSRC::multirateFilter1(const float* input0, float* output0, int inputFrames) {
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static auto f = cpuSupportsAVX2() ? &AudioSRC::multirateFilter1_AVX2 : &AudioSRC::multirateFilter1_SSE;
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return (this->*f)(input0, output0, inputFrames); // dispatch
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}
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int AudioSRC::multirateFilter2(const float* input0, const float* input1, float* output0, float* output1, int inputFrames) {
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static auto f = cpuSupportsAVX2() ? &AudioSRC::multirateFilter2_AVX2 : &AudioSRC::multirateFilter2_SSE;
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return (this->*f)(input0, input1, output0, output1, inputFrames); // dispatch
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}
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// convert int16_t to float, deinterleave stereo
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void AudioSRC::convertInputFromInt16(const int16_t* input, float** outputs, int numFrames) {
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__m128 scale = _mm_set1_ps(1/32768.0f);
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@ -1069,7 +1157,7 @@ int AudioSRC::multirateFilter1(const float* input0, float* output0, int inputFra
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if (_step == 0) { // rational
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int32_t i = hi32(_offset);
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int32_t i = HI32(_offset);
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while (i < inputFrames) {
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@ -1096,10 +1184,10 @@ int AudioSRC::multirateFilter1(const float* input0, float* output0, int inputFra
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} else { // irrational
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while (hi32(_offset) < inputFrames) {
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while (HI32(_offset) < inputFrames) {
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int32_t i = hi32(_offset);
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uint32_t f = lo32(_offset);
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int32_t i = HI32(_offset);
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uint32_t f = LO32(_offset);
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uint32_t phase = f >> SRC_FRACBITS;
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float frac = (f & SRC_FRACMASK) * QFRAC_TO_FLOAT;
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@ -1132,7 +1220,7 @@ int AudioSRC::multirateFilter2(const float* input0, const float* input1, float*
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if (_step == 0) { // rational
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int32_t i = hi32(_offset);
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int32_t i = HI32(_offset);
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while (i < inputFrames) {
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@ -1162,10 +1250,10 @@ int AudioSRC::multirateFilter2(const float* input0, const float* input1, float*
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} else { // irrational
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while (hi32(_offset) < inputFrames) {
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while (HI32(_offset) < inputFrames) {
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int32_t i = hi32(_offset);
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uint32_t f = lo32(_offset);
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int32_t i = HI32(_offset);
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uint32_t f = LO32(_offset);
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uint32_t phase = f >> SRC_FRACBITS;
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float frac = (f & SRC_FRACMASK) * QFRAC_TO_FLOAT;
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@ -1320,7 +1408,7 @@ AudioSRC::AudioSRC(int inputSampleRate, int outputSampleRate, int numChannels) {
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assert(inputSampleRate > 0);
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assert(outputSampleRate > 0);
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assert(numChannels > 0);
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assert(numChannels <= MAX_CHANNELS);
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assert(numChannels <= SRC_MAX_CHANNELS);
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_inputSampleRate = inputSampleRate;
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_outputSampleRate = outputSampleRate;
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@ -1349,7 +1437,7 @@ AudioSRC::AudioSRC(int inputSampleRate, int outputSampleRate, int numChannels) {
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_numTaps = createIrrationalFilter(_upFactor, _downFactor, 1.0f);
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}
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//printf("up=%d down=%.3f taps=%d\n", _upFactor, _downFactor + (lo32(_step)<<SRC_PHASEBITS) * Q32_TO_FLOAT, _numTaps);
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//printf("up=%d down=%.3f taps=%d\n", _upFactor, _downFactor + (LO32(_step)<<SRC_PHASEBITS) * Q32_TO_FLOAT, _numTaps);
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// filter history buffers
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_numHistory = _numTaps - 1;
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@ -1357,10 +1445,10 @@ AudioSRC::AudioSRC(int inputSampleRate, int outputSampleRate, int numChannels) {
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_history[1] = new float[2 * _numHistory];
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// format conversion buffers
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_inputs[0] = (float*)aligned_malloc(SRC_BLOCK * sizeof(float), 16); // SIMD4
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_inputs[1] = (float*)aligned_malloc(SRC_BLOCK * sizeof(float), 16);
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_outputs[0] = (float*)aligned_malloc(SRC_BLOCK * sizeof(float), 16);
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_outputs[1] = (float*)aligned_malloc(SRC_BLOCK * sizeof(float), 16);
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_inputs[0] = (float*)aligned_malloc(4*SRC_BLOCK * sizeof(float), 16); // SIMD4
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_inputs[1] = _inputs[0] + 1*SRC_BLOCK;
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_outputs[0] = _inputs[0] + 2*SRC_BLOCK;
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_outputs[1] = _inputs[0] + 3*SRC_BLOCK;
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// input blocking size, such that input and output are both guaranteed not to exceed SRC_BLOCK frames
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_inputBlock = std::min(SRC_BLOCK, getMaxInput(SRC_BLOCK));
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@ -1380,9 +1468,6 @@ AudioSRC::~AudioSRC() {
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delete[] _history[1];
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aligned_free(_inputs[0]);
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aligned_free(_inputs[1]);
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aligned_free(_outputs[0]);
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aligned_free(_outputs[1]);
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}
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//
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@ -14,11 +14,23 @@
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#include <stdint.h>
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static const int SRC_MAX_CHANNELS = 2;
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// polyphase filter
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static const int SRC_PHASEBITS = 8;
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static const int SRC_PHASES = (1 << SRC_PHASEBITS);
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static const int SRC_FRACBITS = 32 - SRC_PHASEBITS;
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static const uint32_t SRC_FRACMASK = (1 << SRC_FRACBITS) - 1;
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static const float QFRAC_TO_FLOAT = 1.0f / (1 << SRC_FRACBITS);
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static const float Q32_TO_FLOAT = 1.0f / (1ULL << 32);
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// blocking size in frames, chosen so block processing fits in L1 cache
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static const int SRC_BLOCK = 256;
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class AudioSRC {
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public:
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static const int MAX_CHANNELS = 2;
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AudioSRC(int inputSampleRate, int outputSampleRate, int numChannels);
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~AudioSRC();
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@ -33,9 +45,9 @@ private:
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float* _polyphaseFilter;
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int* _stepTable;
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float* _history[MAX_CHANNELS];
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float* _inputs[MAX_CHANNELS];
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float* _outputs[MAX_CHANNELS];
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float* _history[SRC_MAX_CHANNELS];
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float* _inputs[SRC_MAX_CHANNELS];
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float* _outputs[SRC_MAX_CHANNELS];
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int _inputSampleRate;
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int _outputSampleRate;
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|
@ -57,6 +69,12 @@ private:
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int multirateFilter1(const float* input0, float* output0, int inputFrames);
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||||
int multirateFilter2(const float* input0, const float* input1, float* output0, float* output1, int inputFrames);
|
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|
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int multirateFilter1_SSE(const float* input0, float* output0, int inputFrames);
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||||
int multirateFilter2_SSE(const float* input0, const float* input1, float* output0, float* output1, int inputFrames);
|
||||
|
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int multirateFilter1_AVX2(const float* input0, float* output0, int inputFrames);
|
||||
int multirateFilter2_AVX2(const float* input0, const float* input1, float* output0, float* output1, int inputFrames);
|
||||
|
||||
void convertInputFromInt16(const int16_t* input, float** outputs, int numFrames);
|
||||
void convertOutputToInt16(float** inputs, int16_t* output, int numFrames);
|
||||
|
||||
|
|
|
|||
Loading…
Reference in a new issue