Searched refs:halfNumCoefs (Results 1 – 5 of 5) sorted by relevance
| /frameworks/av/media/libaudioprocessing/ |
| D | AudioResamplerDyn.cpp | 86 void AudioResamplerDyn<TC, TI, TO>::InBuffer::resize(int CHANNELS, int halfNumCoefs) in resize() argument
89 size_t stateCount = halfNumCoefs * CHANNELS * 2 * kStateSizeMultipleOfFilterLength; in resize()
94 && mRingFull-mState == (ssize_t) (mStateCount-halfNumCoefs*CHANNELS)) { in resize()
108 TI* srcLo = mImpulse - halfNumCoefs*CHANNELS; in resize()
109 TI* srcHi = mImpulse + halfNumCoefs*CHANNELS; in resize()
126 mImpulse = state + halfNumCoefs*CHANNELS; // actually one sample greater than needed in resize()
127 mRingFull = state + mStateCount - halfNumCoefs*CHANNELS; in resize()
133 void AudioResamplerDyn<TC, TI, TO>::InBuffer::readAgain(TI*& impulse, const int halfNumCoefs, in readAgain() argument
136 TI* head = impulse + halfNumCoefs*CHANNELS; in readAgain()
145 void AudioResamplerDyn<TC, TI, TO>::InBuffer::readAdvance(TI*& impulse, const int halfNumCoefs, in readAdvance() argument
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| D | AudioResamplerFirProcess.h | 378 const int coefShift, const int halfNumCoefs, const TC* const coefs, in fir() argument
391 const TC* coefsP = coefs + indexP*halfNumCoefs; in fir()
392 const TC* coefsN = coefs + indexN*halfNumCoefs; in fir()
398 halfNumCoefs, coefsP, coefsN, sP, sN, volumeLR); in fir()
404 const TC* coefsP = coefs + indexP*halfNumCoefs; in fir()
405 const TC* coefsN = coefs + indexN*halfNumCoefs; in fir()
406 const TC* coefsP1 = coefsP + halfNumCoefs; in fir()
407 const TC* coefsN1 = coefsN + halfNumCoefs; in fir()
426 halfNumCoefs, coefsP, coefsN, coefsP1, coefsN1, sP, sN, lerpP, volumeLR); in fir()
432 halfNumCoefs, coefsP, coefsN, coefsP1, coefsN1, sP, sN, lerpP, volumeLR); in fir()
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| D | AudioResamplerDyn.h | 98 void set(int L, int halfNumCoefs,
113 void resize(int CHANNELS, int halfNumCoefs);
125 inline void readAgain(TI*& impulse, const int halfNumCoefs,
129 inline void readAdvance(TI*& impulse, const int halfNumCoefs,
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| D | AudioResamplerSinc.cpp | 98 c->halfNumCoefs = RESAMPLE_FIR_NUM_COEF; in init_routine()
136 c->halfNumCoefs = readResampleFirNumCoeff(); in init_routine()
138 ALOGV("halfNumCoefs = %d", c->halfNumCoefs); in init_routine()
251 const size_t numCoefs = 2 * c.halfNumCoefs; in init()
255 mImpulse = mState + (c.halfNumCoefs-1)*mChannelCount; in init()
298 const size_t headOffset = c.halfNumCoefs*CHANNELS; in resample()
388 const size_t stateSize = (c.halfNumCoefs*2)*CHANNELS; in read()
393 int16_t* head = impulse + c.halfNumCoefs*CHANNELS; in read()
417 const size_t offset = c.halfNumCoefs; in filterCoefficient()
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| D | AudioResamplerSinc.h | 87 unsigned int halfNumCoefs; member
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