/frameworks/av/media/libaudioprocessing/ |
AudioResamplerFirProcessNeon.h | 79 const int16_t* sN, 130 int16x8_t negSamp = vld1q_s16(sN); 131 sN += 8; 143 int16x8x2_t negSamp = vld2q_s16(sN); 144 sN += 16; 187 const int16_t* sN, 269 int16x8_t negSamp = vld1q_s16(sN); 270 sN += 8; 293 int16x8x2_t negSamp = vld2q_s16(sN); 294 sN += 16 [all...] |
AudioResamplerFirProcessSSE.h | 43 const float* sN, 92 __m128 negSamp = _mm_loadu_ps(sN); 94 sN += 4; 106 __m128 negSamp0 = _mm_loadu_ps(sN); 107 __m128 negSamp1 = _mm_loadu_ps(sN+4); 109 sN += 8; 159 const float* sN, 162 ProcessSSEIntrinsic<1, 16, true>(out, count, coefsP, coefsN, sP, sN, volumeLR, 172 const float* sN, 175 ProcessSSEIntrinsic<2, 16, true>(out, count, coefsP, coefsN, sP, sN, volumeLR [all...] |
AudioResamplerFirProcess.h | 185 const TI* sN, 208 // for (int j = 0; j < CHANNELS; ++j) mac(accum[j], c, sN + j); 209 tmp_data = sN; // tmp_ptr seems faster than directly using sN 213 sN += CHANNELS; 225 mac(l, r, TFUNC::interpolaten(coefsN[count], coefsN[0], lerpP), sN); 227 sN += CHANNELS; 237 mac(l, TFUNC::interpolaten(coefsN[count], coefsN[0], lerpP), sN); 239 sN += CHANNELS; 256 const TI* sN, [all...] |
AudioResamplerSinc.cpp | 424 int16_t const* sN = samples + CHANNELS; 434 interpolate<CHANNELS>(l, r, coefsN++, offset, lerpN, sN); 435 sN += CHANNELS; 464 sampleN = vld1_s16(sN); 470 sN += 4; 534 sampleN = vld2_s16(sN); 540 sN += 8;
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/external/pdfium/third_party/lcms2-2.6/src/ |
cmssm.c | 220 cmsFloat64Number sc, sN, sD; 237 sN = 0.0; // force using point P0 on segment S1 244 sN = (b*e - c*d); 247 if (sN < 0.0) { // sc < 0 => the s=0 edge is visible 249 sN = 0.0; 253 else if (sN > sD) { // sc > 1 => the s=1 edge is visible 254 sN = sD; 265 sN = 0.0; 267 sN = sD; 269 sN = -d [all...] |
/external/webrtc/data/voice_engine/stereo_rtp_files/ |
stereo_pcmu_vad.rtp | [all...] |
stereo_pcmu_vad_jitter.rtp | [all...] |
stereo_pcmu.rtp | [all...] |
stereo_g729_jitter.rtp | 86 A?? < |