| /frameworks/av/media/libeffects/lvm/lib/SpectrumAnalyzer/src/ |
| LVPSA_Init.c | 61 INST_ALLOC Coef;
95 InstAlloc_Init( &Coef , pMemoryTable->Region[LVPSA_MEMREGION_PERSISTENT_COEF].pBaseAddress);
159 pLVPSA_Inst->pBP_Instances = InstAlloc_AddMember( &Coef, pInitParams->nBands * sizeof(Biquad_Instance_t) );
160 pLVPSA_Inst->pQPD_States = InstAlloc_AddMember( &Coef, pInitParams->nBands * sizeof(QPD_State_t) );
162 pLVPSA_Inst->pBP_Instances = InstAlloc_AddMember( &Coef, pInitParams->nBands * \
164 pLVPSA_Inst->pQPD_States = InstAlloc_AddMember( &Coef, pInitParams->nBands * \
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| LVPSA_Control.c | 547 LVM_FLOAT coef; local
578 * Cos += coef(n) * t0^n For n = 0 to 6
586 coef = LVPSA_Float_CosCoef[i]; /* Get the nth coefficient */
587 COS_T0 += (factor * coef); /* The nth partial sum */
627 LVM_INT16 coef; local
659 * Cos += coef(n) * t0^n For n = 0 to 6
667 coef = LVPSA_CosCoef[i]; /* Get the nth coefficient */
668 COS_T0 += (factor * coef) >> 5; /* The nth partial sum */
751 LVM_FLOAT coef; local
834 LVM_INT16 coef; local
[all...] |
| /frameworks/av/media/libaudioprocessing/ |
| AudioResamplerFirProcess.h | 27 void mac(int32_t& l, int32_t& r, TC coef, const int16_t* samples)
30 l = mulAddRL(1, rl, coef, l);
31 r = mulAddRL(0, rl, coef, r);
36 void mac(int32_t& l, TC coef, const int16_t* samples)
38 l = mulAdd(samples[0], coef, l);
44 void mac(float& l, float& r, TC coef, const float* samples)
46 l += *samples++ * coef;
47 r += *samples * coef;
52 void mac(float& l, TC coef, const float* samples)
54 l += *samples * coef;
[all...] |
| AudioResamplerFirGen.h | 423 static inline double firTransfer(const T* coef, int L, int halfNumCoef, double w) {
424 double accum = static_cast<double>(coef[0])*0.5; // "center coefficient" from first bank
425 coef += halfNumCoef; // skip first filterbank (picked up by the last filterbank).
430 accum += cos(ix*w)*static_cast<double>(*coef++);
475 sc = static_cast<double>(*coef++) + dcos*sc - sp;
503 * @param coef is the designed polyphase filter banks
524 static void testFir(const T* coef, int L, int halfNumCoef,
530 double trf = firTransfer(coef, L, halfNumCoef, wstart);
537 trf = firTransfer(coef, L, halfNumCoef, wstart);
564 * @param coef is the designed polyphase filter bank
[all...] |
| AudioResamplerFirProcessNeon.h | 554 "vmlal.s16 q0, d4, d17 \n"// (1+0d) multiply (reversed)samples by coef
555 "vmlal.s16 q0, d5, d16 \n"// (1) multiply (reversed)samples by coef
690 "vmlal.s16 q0, d4, d17 \n"// (1+0d) multiply reversed samples by coef
691 "vmlal.s16 q0, d5, d16 \n"// (1) multiply reversed samples by coef
[all...] |
| /external/aac/libSBRdec/src/ |
| psdec.cpp | [all...] |
| /external/libopus/silk/arm/ |
| NSQ_neon.c | 72 opus_int32 silk_NSQ_noise_shape_feedback_loop_neon(const opus_int32 *data0, opus_int32 *data1, const opus_int16 *coef, opus_int order)
85 int16x8_t coef16 = vld1q_s16(coef);
111 return silk_NSQ_noise_shape_feedback_loop_c(data0, data1, coef, order);
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| arm_silk_map.c | 45 const opus_int32 *data0, opus_int32 *data1, const opus_int16 *coef,
|
| /frameworks/av/media/libeffects/lvm/lib/Eq/src/ |
| LVEQNB_CalcCoef.c | 112 LVM_INT16 coef; local
141 * CosErr += coef(n) * t0^n For n = 0 to 4
149 coef = LVEQNB_DPCosCoef[i]; /* Get the nth coefficient */
150 CosErr += (factor * coef) >> 5; /* The nth partial sum */
303 LVM_INT16 coef; local
332 * Cos += coef(n) * t0^n For n = 0 to 6
340 coef = LVEQNB_CosCoef[i]; /* Get the nth coefficient */
341 COS_T0 += (factor * coef) >> 5; /* The nth partial sum */
|
| /external/eigen/bench/btl/libs/ublas/ |
| ublas_interface.hh | 99 static inline void axpy_slow(const real coef, const gene_vector & X, gene_vector & Y, int N){
100 Y+=coef*X;
117 static inline void axpy(const real coef, const gene_vector & X, gene_vector & Y, int N){
118 Y.plus_assign(coef*X);
|
| /external/mesa3d/src/gallium/drivers/softpipe/ |
| sp_fs.h | 42 void sp_setup_pos_vector(const struct tgsi_interp_coef *coef,
|
| sp_fs_exec.c | 85 setup_pos_vector(const struct tgsi_interp_coef *coef,
104 const float dadx = coef->dadx[chan];
105 const float dady = coef->dady[chan];
106 const float a0 = coef->a0[chan] + dadx * x + dady * y;
|
| /external/libopus/silk/ |
| VAD.c | 311 opus_int coef, min_coef; local
335 coef = VAD_NOISE_LEVEL_SMOOTH_COEF_Q16 >> 3;
337 coef = VAD_NOISE_LEVEL_SMOOTH_COEF_Q16;
339 coef = silk_SMULWB( silk_SMULWW( inv_nrg, nl ), VAD_NOISE_LEVEL_SMOOTH_COEF_Q16 << 1 );
343 coef = silk_max_int( coef, min_coef );
346 psSilk_VAD->inv_NL[ k ] = silk_SMLAWB( psSilk_VAD->inv_NL[ k ], inv_nrg - psSilk_VAD->inv_NL[ k ], coef );
|
| /external/libopus/silk/float/ |
| prefilter_FLP.c | 53 const silk_float coef[], /* I Coefficients [order] */
73 acc = coef[ 0 ] * tmp2;
79 acc += coef[ i - 1 ] * tmp1;
83 acc += coef[ i ] * tmp2;
86 acc += coef[ order - 1 ] * tmp1;
|
| /frameworks/rs/cpu_ref/ |
| rsCpuIntrinsics_x86.cpp | 81 const short *coef, uint32_t count) {
89 x = _mm_loadl_epi64((const __m128i *)(coef+0));
92 x = _mm_loadl_epi64((const __m128i *)(coef+4));
95 x = _mm_loadl_epi64((const __m128i *)(coef+8));
143 const short *coef, uint32_t count) {
157 c0 = _mm_loadl_epi64((const __m128i *)(coef+0));
158 c1 = _mm_loadl_epi64((const __m128i *)(coef+4));
161 c2 = _mm_loadl_epi64((const __m128i *)(coef+8));
162 c3 = _mm_loadl_epi64((const __m128i *)(coef+12));
198 const short *coef, uint32_t count)
[all...] |
| /external/eigen/unsupported/Eigen/src/Skyline/ |
| SkylineInplaceLU.h | 141 const double coef = lIt.value(); local
147 uIt.valueRef() -= uItPivot.value() * coef;
159 const double coef = lIt3.value(); local
163 m_lu.coeffRefLower(rrow, row + i + 1) -= uItPivot.value() * coef;
174 const double coef = lIt2.value(); local
177 m_lu.coeffRefDiag(rrow) -= uItPivot.value() * coef;
|
| /external/eigen/bench/btl/libs/blitz/ |
| tiny_blitz_interface.hh | 99 static inline void axpy(const real coef, const gene_vector & X, gene_vector & Y, int N){
100 Y += coef * X;
|
| /external/eigen/bench/btl/libs/tensors/ |
| tensor_interface.hh | 88 static inline void axpy(real coef, const gene_vector & X, gene_vector & Y, int /*N*/){
89 Y += X.constant(coef) * X;
|
| /external/eigen/bench/btl/libs/tvmet/ |
| tvmet_interface.hh | 97 static inline void axpy(const real coef, const gene_vector & X, gene_vector & Y, int N){
98 Y+=coef*X;
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
| entropy_coding.h | 37 /* decode & dequantize LPC Coef */
49 /* quantize & code LPC Coef */
|
| /frameworks/av/media/libeffects/testlibs/ |
| AudioPeakingFilter.cpp | 28 // kCoefTable[freq][gain][bw][coef]
32 // coef - 0: b0
|
| AudioShelvingFilter.cpp | 28 // kCoefTable[freq][gain][coef]
32 // coef - 0: b0
|
| /frameworks/av/media/libstagefright/codecs/amrwbenc/src/ |
| az_isp.c | 69 Word16 *coef; local
110 coef = f1;
113 ylow = Chebps2(xlow, coef, order);
121 ylow = Chebps2(xlow, coef, order);
128 ymid = Chebps2(xmid, coef, order);
170 coef = f2;
175 coef = f1;
178 ylow = Chebps2(xlow, coef, order);
|
| /frameworks/base/media/jni/ |
| android_media_ResampleInputStream.cpp | 39 #define FIR_COEF(coef) (short)(0x10000 * (coef))
|
| /external/aac/libAACdec/src/ |
| pulsedata.cpp | 142 FIXP_DBL *coef) /*!< pointer to spectrum */
153 if (coef [k] > (FIXP_DBL)0) coef[k] += (FIXP_DBL)(int)PulseData->PulseAmp[i];
154 else coef[k] -= (FIXP_DBL)(int)PulseData->PulseAmp[i];
|
