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https://github.com/JulianGro/overte.git
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Added ARM NEON optimized audio sample rate conversion
This commit is contained in:
Ken Cooke
parent
9e7cd92a96
commit
b47000eab2
1 changed files with 216 additions and 0 deletions
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@ -593,6 +593,202 @@ void AudioSRC::convertOutput(float** inputs, float* output, int numFrames) {
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#else
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#if defined(__ARM_NEON__) || defined(__ARM_NEON)
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#include <arm_neon.h>
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int AudioSRC::multirateFilter1(const float* input0, float* output0, int inputFrames) {
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int outputFrames = 0;
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assert(_numTaps % 8 == 0); // SIMD8
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if (_step == 0) { // rational
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int32_t i = HI32(_offset);
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while (i < inputFrames) {
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const float* c0 = &_polyphaseFilter[_numTaps * _phase];
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float32x4_t acc0 = vdupq_n_f32(0);
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float32x4_t acc1 = vdupq_n_f32(0);
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for (int j = 0; j < _numTaps; j += 8) {
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//float coef = c0[j];
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float32x4_t coef0 = vld1q_f32(&c0[j + 0]); // aligned
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float32x4_t coef1 = vld1q_f32(&c0[j + 4]); // aligned
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//acc += input[i + j] * coef;
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acc0 = vmlaq_f32(acc0, vld1q_f32(&input0[i + j + 0]), coef0);
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acc1 = vmlaq_f32(acc1, vld1q_f32(&input0[i + j + 4]), coef1);
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}
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acc0 = vaddq_f32(acc0, acc1);
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// horizontal sum
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float32x2_t t0 = vadd_f32(vget_low_f32(acc0), vget_high_f32(acc0));
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t0 = vpadd_f32(t0, t0);
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vst1_lane_f32(&output0[outputFrames], t0, 0);
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outputFrames += 1;
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i += _stepTable[_phase];
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if (++_phase == _upFactor) {
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_phase = 0;
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}
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}
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_offset = (int64_t)(i - inputFrames) << 32;
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} else { // irrational
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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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uint32_t phase = f >> SRC_FRACBITS;
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float32x4_t frac = vdupq_n_f32((f & SRC_FRACMASK) * QFRAC_TO_FLOAT);
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const float* c0 = &_polyphaseFilter[_numTaps * (phase + 0)];
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const float* c1 = &_polyphaseFilter[_numTaps * (phase + 1)];
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float32x4_t acc0 = vdupq_n_f32(0);
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float32x4_t acc1 = vdupq_n_f32(0);
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for (int j = 0; j < _numTaps; j += 8) {
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float32x4_t coef0 = vld1q_f32(&c0[j + 0]); // aligned
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float32x4_t coef1 = vld1q_f32(&c0[j + 4]); // aligned
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float32x4_t coef2 = vld1q_f32(&c1[j + 0]); // aligned
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float32x4_t coef3 = vld1q_f32(&c1[j + 4]); // aligned
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//float coef = c0[j] + frac * (c1[j] - c0[j]);
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coef2 = vsubq_f32(coef2, coef0);
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coef3 = vsubq_f32(coef3, coef1);
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coef0 = vmlaq_f32(coef0, coef2, frac);
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coef1 = vmlaq_f32(coef1, coef3, frac);
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//acc += input[i + j] * coef;
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acc0 = vmlaq_f32(acc0, vld1q_f32(&input0[i + j + 0]), coef0);
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acc1 = vmlaq_f32(acc1, vld1q_f32(&input0[i + j + 4]), coef1);
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}
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acc0 = vaddq_f32(acc0, acc1);
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// horizontal sum
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float32x2_t t0 = vadd_f32(vget_low_f32(acc0), vget_high_f32(acc0));
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t0 = vpadd_f32(t0, t0);
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vst1_lane_f32(&output0[outputFrames], t0, 0);
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outputFrames += 1;
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_offset += _step;
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}
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_offset -= (int64_t)inputFrames << 32;
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}
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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 outputFrames = 0;
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assert(_numTaps % 8 == 0); // SIMD8
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if (_step == 0) { // rational
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int32_t i = HI32(_offset);
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while (i < inputFrames) {
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const float* c0 = &_polyphaseFilter[_numTaps * _phase];
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float32x4_t acc0 = vdupq_n_f32(0);
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float32x4_t acc1 = vdupq_n_f32(0);
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for (int j = 0; j < _numTaps; j += 8) {
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//float coef = c0[j];
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float32x4_t coef0 = vld1q_f32(&c0[j + 0]); // aligned
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float32x4_t coef1 = vld1q_f32(&c0[j + 4]); // aligned
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//acc += input[i + j] * coef;
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acc0 = vmlaq_f32(acc0, vld1q_f32(&input0[i + j + 0]), coef0);
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acc1 = vmlaq_f32(acc1, vld1q_f32(&input1[i + j + 0]), coef0);
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acc0 = vmlaq_f32(acc0, vld1q_f32(&input0[i + j + 4]), coef1);
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acc1 = vmlaq_f32(acc1, vld1q_f32(&input1[i + j + 4]), coef1);
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}
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// horizontal sum
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float32x2_t t0 = vadd_f32(vget_low_f32(acc0), vget_high_f32(acc0));
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float32x2_t t1 = vadd_f32(vget_low_f32(acc1), vget_high_f32(acc1));
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t0 = vpadd_f32(t0, t1);
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vst1_lane_f32(&output0[outputFrames], t0, 0);
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vst1_lane_f32(&output1[outputFrames], t0, 1);
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outputFrames += 1;
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i += _stepTable[_phase];
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if (++_phase == _upFactor) {
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_phase = 0;
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}
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}
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_offset = (int64_t)(i - inputFrames) << 32;
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} else { // irrational
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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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uint32_t phase = f >> SRC_FRACBITS;
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float32x4_t frac = vdupq_n_f32((f & SRC_FRACMASK) * QFRAC_TO_FLOAT);
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const float* c0 = &_polyphaseFilter[_numTaps * (phase + 0)];
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const float* c1 = &_polyphaseFilter[_numTaps * (phase + 1)];
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float32x4_t acc0 = vdupq_n_f32(0);
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float32x4_t acc1 = vdupq_n_f32(0);
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for (int j = 0; j < _numTaps; j += 8) {
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float32x4_t coef0 = vld1q_f32(&c0[j + 0]); // aligned
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float32x4_t coef1 = vld1q_f32(&c0[j + 4]); // aligned
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float32x4_t coef2 = vld1q_f32(&c1[j + 0]); // aligned
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float32x4_t coef3 = vld1q_f32(&c1[j + 4]); // aligned
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//float coef = c0[j] + frac * (c1[j] - c0[j]);
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coef2 = vsubq_f32(coef2, coef0);
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coef3 = vsubq_f32(coef3, coef1);
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coef0 = vmlaq_f32(coef0, coef2, frac);
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coef1 = vmlaq_f32(coef1, coef3, frac);
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//acc += input[i + j] * coef;
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acc0 = vmlaq_f32(acc0, vld1q_f32(&input0[i + j + 0]), coef0);
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acc1 = vmlaq_f32(acc1, vld1q_f32(&input1[i + j + 0]), coef0);
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acc0 = vmlaq_f32(acc0, vld1q_f32(&input0[i + j + 4]), coef1);
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acc1 = vmlaq_f32(acc1, vld1q_f32(&input1[i + j + 4]), coef1);
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}
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// horizontal sum
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float32x2_t t0 = vadd_f32(vget_low_f32(acc0), vget_high_f32(acc0));
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float32x2_t t1 = vadd_f32(vget_low_f32(acc1), vget_high_f32(acc1));
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t0 = vpadd_f32(t0, t1);
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vst1_lane_f32(&output0[outputFrames], t0, 0);
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vst1_lane_f32(&output1[outputFrames], t0, 1);
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outputFrames += 1;
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_offset += _step;
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}
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_offset -= (int64_t)inputFrames << 32;
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}
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return outputFrames;
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}
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#else
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int AudioSRC::multirateFilter1(const float* input0, float* output0, int inputFrames) {
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int outputFrames = 0;
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@ -725,6 +921,8 @@ int AudioSRC::multirateFilter2(const float* input0, const float* input1, float*
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return outputFrames;
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}
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#endif
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// convert int16_t to float, deinterleave stereo
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void AudioSRC::convertInput(const int16_t* input, float** outputs, int numFrames) {
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const float scale = 1/32768.0f;
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@ -1030,3 +1228,21 @@ int AudioSRC::getMaxInput(int outputFrames) {
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return (int)(((int64_t)outputFrames * _step) >> 32);
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}
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}
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// the input frames that will produce exactly outputFrames
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int AudioSRC::getExactInput(int outputFrames) {
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//
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// For upsampling, a correct implementation is more complicated
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// because it requires early exit of the multirate filter.
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// This is not currently supported.
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//
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if (_upFactor > _downFactor) {
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return -1;
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}
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if (_step == 0) {
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int64_t offset = ((int64_t)_phase * _downFactor) % _upFactor;
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return (int)(((int64_t)outputFrames * _downFactor + offset) / _upFactor);
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} else {
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return (int)(((int64_t)outputFrames * _step + _offset) >> 32);
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}
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}
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