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  /frameworks/av/services/audioflinger/
AudioResamplerDyn.cpp 391 const uint32_t phaseWrapLimit = c.mL << c.mShift;
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;
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);
    [all...]
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.

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