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  /frameworks/av/services/audioflinger/
AudioResamplerDyn.h 65 void set(int L, int halfNumCoefs,
80 void resize(int CHANNELS, int halfNumCoefs);
92 inline void readAgain(TI*& impulse, const int halfNumCoefs,
96 inline void readAdvance(TI*& impulse, const int halfNumCoefs,
103 // in general, mRingFull = mState + mStateSize - halfNumCoefs*CHANNELS.
AudioResamplerFirProcess.h 268 * @param count should be half the size of the total filter length (halfNumCoefs), as we
271 * @param coefsP is one phase of the polyphase filter bank of size halfNumCoefs, corresponding
274 * @param coefsN is one phase of the polyphase filter bank of size halfNumCoefs, corresponding
329 * @param halfNumCoefs is the half the number of coefficients per polyphase filter. Since the
330 * overall filterbank is odd-length symmetric, only halfNumCoefs need be stored.
333 * and including the #polyphases. Each polyphase of the filter has half-length halfNumCoefs
335 * (#polyphases+1)*halfNumCoefs.
344 * the negative half of the filter is dot product from samples+1 to samples+halfNumCoefs;
345 * the positive half of the filter is dot product from samples to samples-halfNumCoefs+1.
378 const int coefShift, const int halfNumCoefs, const TC* const coefs
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AudioResamplerDyn.cpp 45 * Layout of the state buffer for halfNumCoefs=8.
82 void AudioResamplerDyn<TC, TI, TO>::InBuffer::resize(int CHANNELS, int halfNumCoefs)
85 size_t stateCount = halfNumCoefs * CHANNELS * 2 * kStateSizeMultipleOfFilterLength;
90 && mRingFull-mState == (ssize_t) (mStateCount-halfNumCoefs*CHANNELS)) {
101 TI* srcLo = mImpulse - halfNumCoefs*CHANNELS;
102 TI* srcHi = mImpulse + halfNumCoefs*CHANNELS;
119 mImpulse = state + halfNumCoefs*CHANNELS; // actually one sample greater than needed
120 mRingFull = state + mStateCount - halfNumCoefs*CHANNELS;
123 // copy in the input data into the head (impulse+halfNumCoefs) of the buffer.
126 void AudioResamplerDyn<TC, TI, TO>::InBuffer::readAgain(TI*& impulse, const int halfNumCoefs,
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AudioResamplerSinc.cpp 94 c->halfNumCoefs = RESAMPLE_FIR_NUM_COEF;
132 c->halfNumCoefs = readResampleFirNumCoeff();
134 ALOGV("halfNumCoefs = %d", c->halfNumCoefs);
243 const size_t numCoefs = 2 * c.halfNumCoefs;
247 mImpulse = mState + (c.halfNumCoefs-1)*mChannelCount;
290 const size_t headOffset = c.halfNumCoefs*CHANNELS;
381 const size_t stateSize = (c.halfNumCoefs*2)*CHANNELS;
386 int16_t* head = impulse + c.halfNumCoefs*CHANNELS;
410 const size_t offset = c.halfNumCoefs;
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AudioResamplerSinc.h 87 unsigned int halfNumCoefs;

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