| /frameworks/av/media/libeffects/lvm/lib/Common/src/ |
| H A D | BP_1I_D16F16Css_TRC_WRA_01_Private.h | 25 LVM_INT32 coefs[3]; /* pointer to the filter coefficients */ member in struct:_Filter_State_
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| H A D | BP_1I_D16F32Cll_TRC_WRA_01_Private.h | 25 LVM_INT32 coefs[3]; /* pointer to the filter coefficients */ member in struct:_Filter_State_
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| H A D | BP_1I_D32F32Cll_TRC_WRA_02_Private.h | 25 LVM_INT32 coefs[3]; /* pointer to the filter coefficients */ member in struct:_Filter_State_
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| H A D | BQ_1I_D16F16Css_TRC_WRA_01_Private.h | 25 LVM_INT16 coefs[5]; /* pointer to the filter coefficients */ member in struct:_Filter_State_
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| H A D | BQ_1I_D16F32Css_TRC_WRA_01_Private.h | 25 LVM_INT16 coefs[5]; /* pointer to the filter coefficients */ member in struct:_Filter_State_
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| H A D | BQ_2I_D16F16Css_TRC_WRA_01_Private.h | 26 LVM_INT16 coefs[5]; /* pointer to the filter coefficients */ member in struct:_Filter_State_
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| H A D | BQ_2I_D16F32Css_TRC_WRA_01_Private.h | 26 LVM_INT16 coefs[5]; /* pointer to the filter coefficients */ member in struct:_Filter_State_
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| H A D | BQ_2I_D32F32Cll_TRC_WRA_01_Private.h | 27 LVM_INT32 coefs[5]; /* pointer to the filter coefficients */ member in struct:_Filter_State_
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| H A D | FO_1I_D16F16Css_TRC_WRA_01_Private.h | 26 LVM_INT16 coefs[3]; /* pointer to the filter coefficients */ member in struct:_Filter_State_
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| H A D | FO_1I_D32F32Cll_TRC_WRA_01_Private.h | 27 LVM_INT32 coefs[3]; /* pointer to the filter coefficients */ member in struct:_Filter_State_
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| H A D | FO_2I_D16F32Css_LShx_TRC_WRA_01_Private.h | 26 LVM_INT16 coefs[3]; /* pointer to the filter coefficients */ member in struct:_Filter_State_
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| H A D | PK_2I_D32F32CllGss_TRC_WRA_01_Private.h | 26 LVM_INT32 coefs[5]; /* pointer to the filter coefficients */ member in struct:_Filter_State_
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| H A D | PK_2I_D32F32CssGss_TRC_WRA_01_Private.h | 27 LVM_INT32 coefs[5]; /* pointer to the filter coefficients */ member in struct:_Filter_State_
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| /frameworks/av/media/libeffects/testlibs/ |
| H A D | AudioPeakingFilter.cpp | 88 audio_coef_t coefs[5]; local
99 mCoefInterp.getCoef(intCoord, fracCoord, coefs);
100 mBiquad.setCoefs(coefs, immediate);
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| H A D | AudioShelvingFilter.cpp | 90 audio_coef_t coefs[5]; local
100 mHiCoefInterp.getCoef(intCoord, fracCoord, coefs);
102 mLoCoefInterp.getCoef(intCoord, fracCoord, coefs);
104 mBiquad.setCoefs(coefs, immediate);
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| H A D | AudioBiquadFilter.cpp | 53 void AudioBiquadFilter::setCoefs(const audio_coef_t coefs[NUM_COEFS], bool immediate) { argument
54 memcpy(mTargetCoefs, coefs, sizeof(mTargetCoefs));
57 memcpy(mCoefs, coefs, sizeof(mCoefs));
120 bool AudioBiquadFilter::updateCoefs(const audio_coef_t coefs[NUM_COEFS], argument
125 audio_coef_t diff = coefs[i] - mCoefs[i];
131 mCoefs[i] = coefs[i];
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| /frameworks/av/services/audioflinger/ |
| H A D | AudioResamplerSinc.cpp | 114 // we have 16 coefs samples per zero-crossing
252 // FIXME store current state (up or down sample) and only load the coefs when the state
412 const int32_t* coefs = mFirCoefs; local
416 interpolate<CHANNELS>(l, r, coefs+indexP, lerpP, sP);
417 interpolate<CHANNELS>(l, r, coefs+indexN, lerpN, sN);
418 sP -= CHANNELS; sN += CHANNELS; coefs += 1 << c->coefsBits;
419 interpolate<CHANNELS>(l, r, coefs+indexP, lerpP, sP);
420 interpolate<CHANNELS>(l, r, coefs+indexN, lerpN, sN);
421 sP -= CHANNELS; sN += CHANNELS; coefs += 1 << c->coefsBits;
422 interpolate<CHANNELS>(l, r, coefs
432 interpolate( int32_t& l, int32_t& r, const int32_t* coefs, int16_t lerp, const int16_t* samples) argument
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