Searched refs:phaseWrapLimit (Results 1 - 2 of 2) sorted by relevance
/frameworks/av/services/audioflinger/ |
H A D | AudioResamplerDyn.cpp | 391 const uint32_t phaseWrapLimit = c.mL << c.mShift; local 394 * phaseWrapLimit / oldPhaseWrapLimit; 395 mPhaseFraction %= phaseWrapLimit; // should not do anything, but just in case. 396 mPhaseIncrement = static_cast<uint32_t>(static_cast<uint64_t>(phaseWrapLimit) 501 const uint32_t phaseWrapLimit = c.mL << c.mShift; local 503 / phaseWrapLimit; 508 // " phaseIncrement:%u phaseFraction:%u phaseWrapLimit:%u", 509 // inFrameCount, outFrameCount, phaseIncrement, phaseFraction, phaseWrapLimit); 520 // " phaseFraction:%u phaseWrapLimit:%u", 521 // inFrameCount, outputIndex, outFrameCount, phaseFraction, phaseWrapLimit); 573 ALOG_ASSERT(phaseFraction < phaseWrapLimit); fir<CHANNELS, LOCKED, STRIDE>( &out[outputIndex], phaseFraction, phaseWrapLimit, coefShift, halfNumCoefs, coefs, impulse, volumeSimd); outputIndex += OUTPUT_CHANNELS; phaseFraction += phaseIncrement; while (phaseFraction >= phaseWrapLimit) argument [all...] |
H A D | AudioResamplerFirProcess.h | 283 * phase >= 0 && phase < phaseWrapLimit. It can be thought of as a rational fraction 284 * of phase/phaseWrapLimit. 286 * @param phaseWrapLimit is #polyphases<<coefShift, where #polyphases is the number of polyphases 287 * in the polyphase filter. Likewise, #polyphases can be obtained as (phaseWrapLimit>>coefShift). 315 * phaseIncrement is calculated as inputSampling * phaseWrapLimit / outputSampling. 339 const uint32_t phase, const uint32_t phaseWrapLimit, 352 uint32_t indexN = (phaseWrapLimit - phase) >> coefShift; 365 uint32_t indexN = (phaseWrapLimit - phase - 1) >> coefShift; // one's complement. 338 fir(TO* const out, const uint32_t phase, const uint32_t phaseWrapLimit, const int coefShift, const int halfNumCoefs, const TC* const coefs, const TI* const samples, const TO* const volumeLR) argument
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