/external/chromium_org/third_party/webrtc/common_audio/signal_processing/ |
H A D | dot_product_with_scale.c | 16 int scaling) { 22 sum += (vector1[i + 0] * vector2[i + 0]) >> scaling; 23 sum += (vector1[i + 1] * vector2[i + 1]) >> scaling; 24 sum += (vector1[i + 2] * vector2[i + 2]) >> scaling; 25 sum += (vector1[i + 3] * vector2[i + 3]) >> scaling; 28 sum += (vector1[i] * vector2[i]) >> scaling; 13 WebRtcSpl_DotProductWithScale(const int16_t* vector1, const int16_t* vector2, int length, int scaling) argument
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H A D | energy.c | 24 int scaling = WebRtcSpl_GetScalingSquare(vector, vector_length, vector_length); local 30 en += WEBRTC_SPL_MUL_16_16_RSFT(*vectorptr, *vectorptr, scaling); 33 *scale_factor = scaling;
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H A D | auto_correlation.c | 21 int scaling = 0; local 36 scaling = 0; 44 scaling = 0; 46 scaling = nbits - t; 55 sum += (in_vector[j + 0] * in_vector[i + j + 0]) >> scaling; 56 sum += (in_vector[j + 1] * in_vector[i + j + 1]) >> scaling; 57 sum += (in_vector[j + 2] * in_vector[i + j + 2]) >> scaling; 58 sum += (in_vector[j + 3] * in_vector[i + j + 3]) >> scaling; 61 sum += (in_vector[j] * in_vector[i + j]) >> scaling; 66 *scale = scaling; [all...] |
/external/webrtc/src/common_audio/signal_processing/ |
H A D | energy.c | 24 int scaling = WebRtcSpl_GetScalingSquare(vector, vector_length, vector_length); local 30 en += WEBRTC_SPL_MUL_16_16_RSFT(*vectorptr, *vectorptr, scaling); 33 *scale_factor = scaling;
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H A D | dot_product_with_scale.c | 21 int length, int scaling) 35 sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1++, *vector2++, scaling); 38 if (scaling == 0) 67 sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1, *vector2, scaling); 70 sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1, *vector2, scaling); 73 sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1, *vector2, scaling); 76 sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1, *vector2, scaling); 83 sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1, *vector2, scaling); 20 WebRtcSpl_DotProductWithScale(WebRtc_Word16 *vector1, WebRtc_Word16 *vector2, int length, int scaling) argument
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H A D | auto_correlation.c | 32 int scaling = 0; local 50 scaling = 0; 58 scaling = 0; 61 scaling = nbits - t; 78 sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1++, *xptr2++, scaling); 83 if (scaling == 0) 112 sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1, *xptr2, scaling); 115 sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1, *xptr2, scaling); 118 sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1, *xptr2, scaling); 121 sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1, *xptr2, scaling); [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/ |
H A D | fft.h | 22 * scaling: normalizing constant by which the final result is *divided* 23 * scaling == -1, normalize by total dimension of the transform 24 * scaling < -1, normalize by the square-root of the total dimension 41 int isign, double scaling, FFTstr *fftstate);
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/external/webrtc/src/modules/audio_coding/codecs/isac/main/source/ |
H A D | fft.h | 22 * scaling: normalizing constant by which the final result is *divided* 23 * scaling == -1, normalize by total dimension of the transform 24 * scaling < -1, normalize by the square-root of the total dimension 41 int isign, double scaling, FFTstr *fftstate);
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/external/aac/libAACenc/src/ |
H A D | pre_echo_control.cpp | 119 int scaling; local 132 scaling = 2*(mdctScale-*mdctScalenm1); 136 FDK_ASSERT(scaling>=0); 137 tmpThreshold1 = maxAllowedIncreaseFactor * (pbThresholdNm1[i]>>scaling); 151 scaling = 2*(*mdctScalenm1-mdctScale); 161 FDK_ASSERT(scaling>=0); 162 if((pbThreshold[i]>>(scaling+1)) > tmpThreshold1) { 163 pbThreshold[i] = tmpThreshold1<<(scaling+1);
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
H A D | time_stretch.cc | 80 // Calculate scaling to ensure that |peak_index| samples can be square-summed 82 int scaling = 31 - WebRtcSpl_NormW32(max_input_value_ * max_input_value_) - local 84 scaling = std::max(0, scaling); 93 WebRtcSpl_DotProductWithScale(vec1, vec1, peak_index, scaling); 95 WebRtcSpl_DotProductWithScale(vec2, vec2, peak_index, scaling); 99 WebRtcSpl_DotProductWithScale(vec1, vec2, peak_index, scaling); 103 scaling); 116 // Make sure total scaling is even (to simplify scale factor after sqrt). 161 // Set scaling facto 162 int scaling = kLogCorrelationLen - WebRtcSpl_NormW32( local [all...] |
H A D | normal.cc | 81 int scaling = 6 + fs_shift local 83 scaling = std::max(scaling, 0); // |scaling| should always be >= 0. 85 energy_length, scaling); 86 if ((energy_length >> scaling) > 0) { 87 energy = energy / (energy_length >> scaling); 96 scaling = WebRtcSpl_NormW32(energy) - 16; 99 background_noise_.Energy(channel_ix) << (scaling+14); 100 int16_t energy_scaled = energy << scaling; [all...] |
/external/webrtc/src/modules/audio_coding/codecs/isac/fix/source/ |
H A D | filters_neon.c | 39 int16_t scaling = 0; local 42 // Step 1, calculate r[0] and how much scaling is needed. 61 // Calculate the value of shifting (scaling). 71 scaling = (32 - zeros_high + 1); 73 scaling = 1; 75 reg64x1b = -scaling; 90 sum = (int32_t)(prod >> scaling); 145 "mov %[tmp], %[scaling], asr #31\n\t" 146 "vmov.32 d16, %[scaling], %[tmp]\n\t" 155 [scaling]" [all...] |
H A D | filters.c | 34 int16_t scaling = 0; local 44 // Calculate scaling (the value of shifting). 47 scaling = 0; 49 scaling = 32 - WebRtcSpl_NormU32(temp); 51 r[0] = (int32_t)(prod >> scaling); 59 sum = (int32_t)(prod >> scaling); 63 *scale = scaling;
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
H A D | pitch_estimator_c.c | 23 int16_t scaling,n,k; local 32 scaling = WebRtcSpl_GetScalingSquare((int16_t*)in, 41 scaling); // Q0 44 scaling); // Q0 65 scaling); 68 scaling); 74 int32x4_t int_32x4_scale = vdupq_n_s32(-scaling); 95 if(scaling == 0) { 101 csum32 += (x[n] * inptr[n]) >> scaling;
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H A D | filters_mips.c | 21 int16_t scaling = 0; local 32 // Calculate r[0] and scaling needed. 102 // Calculate scaling (the value of shifting). 105 "subu %[scaling], $0, %[r1] \n\t" 107 "movn %[scaling], $0, %[r1] \n\t" 109 "extrv.w %[r0], $ac0, %[scaling] \n\t" 112 "addiu %[r1], %[scaling], -32 \n\t" 115 "srlv %[r0], %[r3], %[scaling] \n\t" 118 "slti %[r1], %[scaling], 32 \n\t" 126 [count] "+r" (count), [scaling] " [all...] |
H A D | filters.c | 24 int16_t scaling = 0; local 38 // Calculate scaling (the value of shifting). 41 scaling = 0; 43 scaling = 32 - WebRtcSpl_NormU32(temp); 45 r[0] = (int32_t)(prod >> scaling); 53 sum = (int32_t)(prod >> scaling); 57 *scale = scaling;
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H A D | pitch_estimator_mips.c | 18 int16_t scaling,n,k; local 26 scaling = WebRtcSpl_GetScalingSquare((int16_t*)in, 59 "srav %[tmp5], %[tmp5], %[scaling] \n\t" 60 "srav %[tmp1], %[tmp1], %[scaling] \n\t" 61 "srav %[tmp6], %[tmp6], %[scaling] \n\t" 62 "srav %[tmp2], %[tmp2], %[scaling] \n\t" 63 "srav %[tmp7], %[tmp7], %[scaling] \n\t" 64 "srav %[tmp3], %[tmp3], %[scaling] \n\t" 65 "srav %[tmp8], %[tmp8], %[scaling] \n\t" 66 "srav %[tmp4], %[tmp4], %[scaling] \ [all...] |
/external/chromium_org/third_party/webrtc/voice_engine/include/mock/ |
H A D | mock_voe_volume_control.h | 39 MOCK_METHOD2(SetChannelOutputVolumeScaling, int(int channel, float scaling)); 40 MOCK_METHOD2(GetChannelOutputVolumeScaling, int(int channel, float& scaling));
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/external/chromium_org/third_party/webrtc/voice_engine/ |
H A D | voe_volume_control_impl.h | 44 virtual int SetChannelOutputVolumeScaling(int channel, float scaling); 46 virtual int GetChannelOutputVolumeScaling(int channel, float& scaling);
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/external/speex/libspeex/ |
H A D | filterbank.h | 45 float *scaling; member in struct:__anon30988
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/external/chromium_org/third_party/freetype/src/cff/ |
H A D | cffparse.c | 133 FT_Long* scaling ) 144 if ( scaling ) 145 *scaling = 0; 279 if ( scaling ) 290 *scaling = exponent - fraction_length + 1; 299 /* Make `scaling' as small as possible. */ 320 *scaling = exponent; 328 *scaling = exponent - 4; 333 *scaling = exponent - 5; 415 FT_Long scaling ) 523 FT_Long scaling; local [all...] |
/external/freetype/src/cff/ |
H A D | cffparse.c | 129 FT_Long* scaling ) 140 if ( scaling ) 141 *scaling = 0; 275 if ( scaling ) 286 *scaling = exponent - fraction_length + 1; 295 /* Make `scaling' as small as possible. */ 316 *scaling = exponent; 324 *scaling = exponent - 4; 329 *scaling = exponent - 5; 411 FT_Long scaling ) 519 FT_Long scaling; local [all...] |
/external/pdfium/core/src/fxge/fx_freetype/fxft2.5.01/src/cff/ |
H A D | cffparse.c | 133 FT_Long* scaling ) 144 if ( scaling ) 145 *scaling = 0; 279 if ( scaling ) 290 *scaling = exponent - fraction_length + 1; 299 /* Make `scaling' as small as possible. */ 320 *scaling = exponent; 328 *scaling = exponent - 4; 333 *scaling = exponent - 5; 415 FT_Long scaling ) 523 FT_Long scaling; local [all...] |
/external/chromium_org/third_party/freetype/include/freetype/internal/ |
H A D | ftcalc.h | 91 * the 64bit multiplication. Let `sa' and `sb' be the scaling factors of 92 * `a' and `b', respectively, then the scaling factor of the result is 98 FT_Long scaling ); 108 FT_Long scaling );
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/external/chromium_org/third_party/webrtc/voice_engine/include/ |
H A D | voe_volume_control.h | 17 // - Additional stereo scaling methods. 94 // Sets a volume |scaling| applied to the outgoing signal of a specific 96 virtual int SetChannelOutputVolumeScaling(int channel, float scaling) = 0; 98 // Gets the current volume scaling for a specified |channel|. 99 virtual int GetChannelOutputVolumeScaling(int channel, float& scaling) = 0; 105 // Gets the current left and right scaling factors.
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