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  /frameworks/av/media/libaudioprocessing/
AudioResamplerFirProcess.h 288 * below coefShift. See fir() for details.
324 * @param phaseWrapLimit is #polyphases<<coefShift, where #polyphases is the number of polyphases
325 * in the polyphase filter. Likewise, #polyphases can be obtained as (phaseWrapLimit>>coefShift).
327 * @param coefShift gives the bit alignment of the polyphase index in the phase parameter.
355 * The filter polyphase index is given by indexP = phase >> coefShift. Due to
359 * The fractional siting between the polyphase indices is given by the bits below coefShift:
361 * lerpP = phase << 32 - coefShift >> 1; // for 32 bit unsigned phase multiply
362 * lerpP = phase << 32 - coefShift >> 17; // for 16 bit unsigned phase multiply
366 * lerpP = phase << sizeof(phase)*8 - coefShift
371 * lerpP = (phase << 32 - coefShift) / (1 << 32); // floating point equivalen
    [all...]
AudioResamplerDyn.cpp 569 const int coefShift = c.mShift;
588 coefShift, halfNumCoefs, coefs,

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