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239 lines
7.5 KiB
239 lines
7.5 KiB
4 months ago
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/*
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* Copyright (C) 2016 The Android Open Source Project
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*
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* Licensed under the Apache License, Version 2.0 (the "License");
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* you may not use this file except in compliance with the License.
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* You may obtain a copy of the License at
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*
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* http://www.apache.org/licenses/LICENSE-2.0
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*
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* Unless required by applicable law or agreed to in writing, software
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* distributed under the License is distributed on an "AS IS" BASIS,
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* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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* See the License for the specific language governing permissions and
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* limitations under the License.
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*/
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#ifndef ANDROID_AUDIO_RESAMPLER_FIR_PROCESS_SSE_H
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#define ANDROID_AUDIO_RESAMPLER_FIR_PROCESS_SSE_H
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namespace android {
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// depends on AudioResamplerFirOps.h, AudioResamplerFirProcess.h
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#if USE_SSE
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#define TO_STRING2(x) #x
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#define TO_STRING(x) TO_STRING2(x)
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// uncomment to print GCC version, may be relevant for intrinsic optimizations
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/* #pragma message ("GCC version: " TO_STRING(__GNUC__) \
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"." TO_STRING(__GNUC_MINOR__) \
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"." TO_STRING(__GNUC_PATCHLEVEL__)) */
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//
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// SSEx specializations are enabled for Process() and ProcessL() in AudioResamplerFirProcess.h
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//
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template <int CHANNELS, int STRIDE, bool FIXED>
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static inline void ProcessSSEIntrinsic(float* out,
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int count,
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const float* coefsP,
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const float* coefsN,
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const float* sP,
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const float* sN,
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const float* volumeLR,
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float lerpP,
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const float* coefsP1,
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const float* coefsN1)
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{
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ALOG_ASSERT(count > 0 && (count & 7) == 0); // multiple of 8
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static_assert(CHANNELS == 1 || CHANNELS == 2, "CHANNELS must be 1 or 2");
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sP -= CHANNELS*(4-1); // adjust sP for a loop iteration of four
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__m128 interp;
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if (!FIXED) {
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interp = _mm_set1_ps(lerpP);
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}
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__m128 accL, accR;
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accL = _mm_setzero_ps();
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if (CHANNELS == 2) {
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accR = _mm_setzero_ps();
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}
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do {
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__m128 posCoef = _mm_load_ps(coefsP);
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__m128 negCoef = _mm_load_ps(coefsN);
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coefsP += 4;
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coefsN += 4;
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if (!FIXED) { // interpolate
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__m128 posCoef1 = _mm_load_ps(coefsP1);
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__m128 negCoef1 = _mm_load_ps(coefsN1);
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coefsP1 += 4;
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coefsN1 += 4;
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// Calculate the final coefficient for interpolation
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// posCoef = interp * (posCoef1 - posCoef) + posCoef
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// negCoef = interp * (negCoef - negCoef1) + negCoef1
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posCoef1 = _mm_sub_ps(posCoef1, posCoef);
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negCoef = _mm_sub_ps(negCoef, negCoef1);
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#if USE_AVX2
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posCoef = _mm_fmadd_ps(posCoef1, interp, posCoef);
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negCoef = _mm_fmadd_ps(negCoef, interp, negCoef1);
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#else
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posCoef1 = _mm_mul_ps(posCoef1, interp);
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negCoef = _mm_mul_ps(negCoef, interp);
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posCoef = _mm_add_ps(posCoef1, posCoef);
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negCoef = _mm_add_ps(negCoef, negCoef1);
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#endif //USE_AVX2
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}
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switch (CHANNELS) {
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case 1: {
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__m128 posSamp = _mm_loadu_ps(sP);
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__m128 negSamp = _mm_loadu_ps(sN);
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sP -= 4;
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sN += 4;
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posSamp = _mm_shuffle_ps(posSamp, posSamp, 0x1B);
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#if USE_AVX2
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accL = _mm_fmadd_ps(posSamp, posCoef, accL);
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accL = _mm_fmadd_ps(negSamp, negCoef, accL);
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#else
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posSamp = _mm_mul_ps(posSamp, posCoef);
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negSamp = _mm_mul_ps(negSamp, negCoef);
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accL = _mm_add_ps(accL, posSamp);
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accL = _mm_add_ps(accL, negSamp);
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#endif
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} break;
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case 2: {
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__m128 posSamp0 = _mm_loadu_ps(sP);
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__m128 posSamp1 = _mm_loadu_ps(sP+4);
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__m128 negSamp0 = _mm_loadu_ps(sN);
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__m128 negSamp1 = _mm_loadu_ps(sN+4);
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sP -= 8;
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sN += 8;
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// deinterleave everything and reverse the positives
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__m128 posSampL = _mm_shuffle_ps(posSamp1, posSamp0, 0x22);
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__m128 posSampR = _mm_shuffle_ps(posSamp1, posSamp0, 0x77);
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__m128 negSampL = _mm_shuffle_ps(negSamp0, negSamp1, 0x88);
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__m128 negSampR = _mm_shuffle_ps(negSamp0, negSamp1, 0xDD);
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#if USE_AVX2
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accL = _mm_fmadd_ps(posSampL, posCoef, accL);
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accR = _mm_fmadd_ps(posSampR, posCoef, accR);
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accL = _mm_fmadd_ps(negSampL, negCoef, accL);
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accR = _mm_fmadd_ps(negSampR, negCoef, accR);
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#else
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posSampL = _mm_mul_ps(posSampL, posCoef);
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posSampR = _mm_mul_ps(posSampR, posCoef);
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negSampL = _mm_mul_ps(negSampL, negCoef);
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negSampR = _mm_mul_ps(negSampR, negCoef);
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accL = _mm_add_ps(accL, posSampL);
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accR = _mm_add_ps(accR, posSampR);
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accL = _mm_add_ps(accL, negSampL);
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accR = _mm_add_ps(accR, negSampR);
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#endif
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} break;
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}
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} while (count -= 4);
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// multiply by volume and save
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__m128 vLR = _mm_setzero_ps();
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__m128 outSamp;
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vLR = _mm_loadl_pi(vLR, reinterpret_cast<const __m64*>(volumeLR));
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outSamp = _mm_loadl_pi(vLR, reinterpret_cast<__m64*>(out));
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// combine and funnel down accumulator
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__m128 outAccum = _mm_setzero_ps();
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if (CHANNELS == 1) {
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// duplicate accL to both L and R
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outAccum = _mm_add_ps(accL, _mm_movehl_ps(accL, accL));
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outAccum = _mm_add_ps(outAccum, _mm_shuffle_ps(outAccum, outAccum, 0x11));
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} else if (CHANNELS == 2) {
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// accR contains R, fold in
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outAccum = _mm_hadd_ps(accL, accR);
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outAccum = _mm_hadd_ps(outAccum, outAccum);
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}
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#if USE_AVX2
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outSamp = _mm_fmadd_ps(outAccum, vLR,outSamp);
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#else
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outAccum = _mm_mul_ps(outAccum, vLR);
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outSamp = _mm_add_ps(outSamp, outAccum);
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#endif
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_mm_storel_pi(reinterpret_cast<__m64*>(out), outSamp);
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}
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template<>
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inline void ProcessL<1, 16>(float* const out,
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int count,
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const float* coefsP,
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const float* coefsN,
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const float* sP,
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const float* sN,
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const float* const volumeLR)
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{
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ProcessSSEIntrinsic<1, 16, true>(out, count, coefsP, coefsN, sP, sN, volumeLR,
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0 /*lerpP*/, NULL /*coefsP1*/, NULL /*coefsN1*/);
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}
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template<>
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inline void ProcessL<2, 16>(float* const out,
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int count,
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const float* coefsP,
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const float* coefsN,
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const float* sP,
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const float* sN,
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const float* const volumeLR)
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{
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ProcessSSEIntrinsic<2, 16, true>(out, count, coefsP, coefsN, sP, sN, volumeLR,
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0 /*lerpP*/, NULL /*coefsP1*/, NULL /*coefsN1*/);
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}
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template<>
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inline void Process<1, 16>(float* const out,
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int count,
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const float* coefsP,
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const float* coefsN,
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const float* coefsP1,
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const float* coefsN1,
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const float* sP,
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const float* sN,
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float lerpP,
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const float* const volumeLR)
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{
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ProcessSSEIntrinsic<1, 16, false>(out, count, coefsP, coefsN, sP, sN, volumeLR,
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lerpP, coefsP1, coefsN1);
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}
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template<>
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inline void Process<2, 16>(float* const out,
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int count,
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const float* coefsP,
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const float* coefsN,
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const float* coefsP1,
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const float* coefsN1,
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const float* sP,
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const float* sN,
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float lerpP,
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const float* const volumeLR)
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{
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ProcessSSEIntrinsic<2, 16, false>(out, count, coefsP, coefsN, sP, sN, volumeLR,
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lerpP, coefsP1, coefsN1);
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}
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#endif //USE_SSE
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} // namespace android
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#endif /*ANDROID_AUDIO_RESAMPLER_FIR_PROCESS_SSE_H*/
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