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    Searched defs:sP (Results 1 - 3 of 3) sorted by null

  /frameworks/av/media/libaudioprocessing/
AudioResamplerFirProcess.h 184 const TI* sP,
200 // for (int j = 0; j < CHANNELS; ++j) mac(accum[j], c, sP + j);
201 const TI *tmp_data = sP; // tmp_ptr seems to work better
205 sP -= CHANNELS;
222 mac(l, r, TFUNC::interpolatep(coefsP[0], coefsP[count], lerpP), sP);
224 sP -= CHANNELS;
234 mac(l, TFUNC::interpolatep(coefsP[0], coefsP[count], lerpP), sP);
236 sP -= CHANNELS;
255 const TI* sP,
259 ProcessBase<CHANNELS, STRIDE, InterpNull>(out, count, coefsP, coefsN, sP, sN, 0, volumeLR)
    [all...]
AudioResamplerSinc.cpp 423 int16_t const* sP = samples;
432 interpolate<CHANNELS>(l, r, coefsP++, offset, lerpP, sP);
433 sP -= CHANNELS;
444 sP -= CHANNELS*3;
463 sampleP = vld1_s16(sP);
469 sP -= 4;
512 sP -= CHANNELS*3;
533 sampleP = vld2_s16(sP);
539 sP -= 8;
  /external/eclipse-basebuilder/basebuilder-3.6.2/org.eclipse.releng.basebuilder/plugins/org.apache.ant_1.7.1.v20090120-1145/lib/
ant-jdepend.jar 

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