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/frameworks/av/services/audioflinger/ |
H A D | AudioResamplerSinc.cpp | 413 const int16_t *sP = samples; local 416 interpolate<CHANNELS>(l, r, coefs+indexP, lerpP, sP); 418 sP -= CHANNELS; sN += CHANNELS; coefs += 1 << c->coefsBits; 419 interpolate<CHANNELS>(l, r, coefs+indexP, lerpP, sP); 421 sP -= CHANNELS; sN += CHANNELS; coefs += 1 << c->coefsBits; 422 interpolate<CHANNELS>(l, r, coefs+indexP, lerpP, sP); 424 sP -= CHANNELS; sN += CHANNELS; coefs += 1 << c->coefsBits; 425 interpolate<CHANNELS>(l, r, coefs+indexP, lerpP, sP); 427 sP -= CHANNELS; sN += CHANNELS; coefs += 1 << c->coefsBits;
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