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Searched refs:sP (Results 1 – 5 of 5) sorted by relevance

/frameworks/av/media/libaudioprocessing/
DAudioResamplerFirProcessNeon.h78 const int16_t* sP, in ProcessNeonIntrinsic() argument
88 sP -= CHANNELS*((STRIDE>>1)-1); in ProcessNeonIntrinsic()
129 int16x8_t posSamp = vld1q_s16(sP); in ProcessNeonIntrinsic()
139 sP -= 8; in ProcessNeonIntrinsic()
142 int16x8x2_t posSamp = vld2q_s16(sP); in ProcessNeonIntrinsic()
157 sP -= 16; in ProcessNeonIntrinsic()
186 const int16_t* sP, in ProcessNeonIntrinsic() argument
196 sP -= CHANNELS*((STRIDE>>1)-1); in ProcessNeonIntrinsic()
268 int16x8_t posSamp = vld1q_s16(sP); in ProcessNeonIntrinsic()
289 sP -= 8; in ProcessNeonIntrinsic()
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DAudioResamplerFirProcessSSE.h42 const float* sP, in ProcessSSEIntrinsic() argument
52 sP -= CHANNELS*(4-1); // adjust sP for a loop iteration of four in ProcessSSEIntrinsic()
96 __m128 posSamp = _mm_loadu_ps(sP); in ProcessSSEIntrinsic()
98 sP -= 4; in ProcessSSEIntrinsic()
115 __m128 posSamp0 = _mm_loadu_ps(sP); in ProcessSSEIntrinsic()
116 __m128 posSamp1 = _mm_loadu_ps(sP+4); in ProcessSSEIntrinsic()
119 sP -= 8; in ProcessSSEIntrinsic()
181 const float* sP,
185 ProcessSSEIntrinsic<1, 16, true>(out, count, coefsP, coefsN, sP, sN, volumeLR,
194 const float* sP,
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DAudioResamplerFirProcess.h184 const TI* sP, in ProcessBase() argument
201 const TI *tmp_data = sP; // tmp_ptr seems to work better in ProcessBase()
205 sP -= CHANNELS; in ProcessBase()
222 mac(l, r, TFUNC::interpolatep(coefsP[0], coefsP[count], lerpP), sP); in ProcessBase()
224 sP -= CHANNELS; in ProcessBase()
234 mac(l, TFUNC::interpolatep(coefsP[0], coefsP[count], lerpP), sP); in ProcessBase()
236 sP -= CHANNELS; in ProcessBase()
255 const TI* sP, in ProcessL() argument
259 ProcessBase<CHANNELS, STRIDE, InterpNull>(out, count, coefsP, coefsN, sP, sN, 0, volumeLR); in ProcessL()
303 const TI* sP, in Process() argument
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DAudioResamplerSinc.cpp424 int16_t const* sP = samples; in filterCoefficient() local
433 interpolate<CHANNELS>(l, r, coefsP++, offset, lerpP, sP); in filterCoefficient()
434 sP -= CHANNELS; in filterCoefficient()
445 sP -= CHANNELS*3; in filterCoefficient()
464 sampleP = vld1_s16(sP); in filterCoefficient()
470 sP -= 4; in filterCoefficient()
513 sP -= CHANNELS*3; in filterCoefficient()
534 sampleP = vld2_s16(sP); in filterCoefficient()
540 sP -= 8; in filterCoefficient()
/frameworks/base/media/tests/contents/media_api/music/
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