Searched refs:phaseWrapLimit (Results 1 - 2 of 2) sorted by relevance

/frameworks/av/services/audioflinger/
H A DAudioResamplerDyn.cpp391 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
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H A DAudioResamplerFirProcess.h283 * 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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