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  /frameworks/av/media/libaudioprocessing/
AudioResamplerFirProcess.h 390 uint32_t indexN = (phaseWrapLimit - phase) >> coefShift;
392 const TC* coefsN = coefs + indexN*halfNumCoefs;
403 uint32_t indexN = (phaseWrapLimit - phase - 1) >> coefShift; // one's complement.
405 const TC* coefsN = coefs + indexN*halfNumCoefs;
AudioResamplerSinc.cpp 414 uint32_t indexN = ((ONE-phase) & c.cMask) >> c.cShift;
419 indexN *= offset;
422 int32_t const* coefsN = mFirCoefs + indexN;
  /prebuilts/jdk/jdk8/darwin-x86/lib/
tools.jar 
  /prebuilts/jdk/jdk8/linux-x86/lib/
tools.jar 

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