/frameworks/av/media/libstagefright/codecs/aacenc/inc/ |
quantize.h | 40 Word16 gain);
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/external/chromium_org/third_party/WebKit/Source/modules/webaudio/ |
BiquadDSPKernel.cpp | 49 double gain; local 55 gain = biquadProcessor()->parameter3()->finalValue(); 60 gain = biquadProcessor()->parameter3()->smoothedValue(); 65 gain = biquadProcessor()->parameter3()->value(); 92 m_biquad.setLowShelfParams(normalizedFrequency, gain); 96 m_biquad.setHighShelfParams(normalizedFrequency, gain); 100 m_biquad.setPeakingParams(normalizedFrequency, value2, gain);
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GainNode.h | 37 // GainNode is an AudioNode with one input and one output which applies a gain (volume) change to the audio signal. 38 // De-zippering (smoothing) is applied when the gain value is changed dynamically. 55 AudioParam* gain() { return m_gain.get(); } function in class:WebCore::GainNode
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/device/generic/goldfish/camera/fake-pipeline2/ |
Sensor.h | 30 * frame duration, and gain are set for the next frame to be captured. In stage 113 void setSensitivity(uint32_t gain); 233 void captureRaw(uint8_t *img, uint32_t gain, uint32_t stride); 234 void captureRGBA(uint8_t *img, uint32_t gain, uint32_t stride); 235 void captureRGB(uint8_t *img, uint32_t gain, uint32_t stride); 236 void captureNV21(uint8_t *img, uint32_t gain, uint32_t stride);
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/frameworks/av/media/libstagefright/codecs/amrnb/enc/src/ |
g_code.cpp | 107 pOverflow -> 1 if the innovative gain calculation resulted in overflow 110 gain = Gain of Innovation code (Word16) 121 This function computes the innovative codebook gain. 123 The innovative codebook gain is given by 142 Word16 G_code ( // out : Gain of innovation code 148 Word16 xy, yy, exp_xy, exp_yy, gain; 173 // If (xy < 0) gain = 0 188 // compute gain = xy/yy 191 gain = div_s (xy, yy) 236 Word16 xy, yy, exp_xy, exp_yy, gain; local [all...] |
g_pitch.cpp | 118 g_coeff = pointer to buffer of correlations needed for gain quantization 128 gain = ratio of dot products.(Word16) 139 This function computes the pitch (adaptive codebook) gain. The adaptive 140 codebook gain is given by 148 The gain is limited to the range [0,1.2] (=0..19661 Q14) 163 Word16 G_pitch ( // o : Gain of pitch lag saturated to 1.2 167 Word16 g_coeff[], // i : Correlations need for gain quantization 172 Word16 xy, yy, exp_xy, exp_yy, gain; 244 // If (xy < 4) gain = 0 251 // compute gain = xy/y 313 Word16 gain; local [all...] |
/external/chromium_org/third_party/opus/src/silk/float/ |
process_gains_FLP.c | 45 silk_float s, InvMaxSqrVal, gain, quant_offset; local 47 /* Gain reduction when LTP coding gain is high */ 60 gain = psEncCtrl->Gains[ k ]; 61 gain = ( silk_float )sqrt( gain * gain + psEncCtrl->ResNrg[ k ] * InvMaxSqrVal ); 62 psEncCtrl->Gains[ k ] = silk_min_float( gain, 32767.0f ); 70 /* Save unquantized gains and gain Index */ 83 /* Set quantizer offset for voiced signals. Larger offset when LTP coding gain is low or tilt is high (ie low-pass) * [all...] |
/external/sonivox/arm-fm-22k/lib_src/ |
eas_fmengine.c | 195 * Assign the left and right gain values corresponding to the given pan value. 283 EAS_I32 gain; local 291 /* establish local gain variable */ 292 gain = (EAS_I32) p->gain << 16; 294 /* calculate gain increment */ 296 gainInc = ((EAS_I32) gainTarget - (EAS_I32) p->gain) << (16 - SYNTH_UPDATE_PERIOD_IN_BITS); 337 /* internal gain for modulation effects */ 338 temp = FMUL_15x15(temp, (gain >> 16)); 340 /* output gain calculation * 393 EAS_I32 gain; local [all...] |
eas_mixer.c | 137 EAS_U16 gain; local 141 /* calculate the gain multiplier */ 149 gain = 32767; 151 gain = (EAS_U16) temp; 154 gain = (EAS_U16) pEASData->masterGain; 156 gain = (EAS_U16) pEASData->masterGain; 159 /* Not using all the gain bits for now 164 gain = gain >> 5; 166 gain = gain >> 4 [all...] |
eas_fmengine.h | 56 /* LFO modulation to gain control */ 67 EAS_U16 gain; /* current internal gain */ member in struct:__anon28054 68 EAS_U16 outputGain; /* current output gain */ 78 EAS_U16 gainLeft; /* left gain multiplier */ 79 EAS_U16 gainRight; /* right gain multiplier */ 87 EAS_U16 gain[4]; /* initial operator gain value */ member in struct:__anon28056 88 EAS_U16 outputGain[4]; /* initial operator output gain value */ 89 EAS_U16 voiceGain; /* initial voice gain */ 99 EAS_U16 gain[4]; \/* new operator gain value *\/ member in struct:__anon28057 [all...] |
/external/sonivox/arm-hybrid-22k/lib_src/ |
eas_fmengine.c | 195 * Assign the left and right gain values corresponding to the given pan value. 283 EAS_I32 gain; local 291 /* establish local gain variable */ 292 gain = (EAS_I32) p->gain << 16; 294 /* calculate gain increment */ 296 gainInc = ((EAS_I32) gainTarget - (EAS_I32) p->gain) << (16 - SYNTH_UPDATE_PERIOD_IN_BITS); 337 /* internal gain for modulation effects */ 338 temp = FMUL_15x15(temp, (gain >> 16)); 340 /* output gain calculation * 393 EAS_I32 gain; local [all...] |
eas_mixer.c | 137 EAS_U16 gain; local 141 /* calculate the gain multiplier */ 149 gain = 32767; 151 gain = (EAS_U16) temp; 154 gain = (EAS_U16) pEASData->masterGain; 156 gain = (EAS_U16) pEASData->masterGain; 159 /* Not using all the gain bits for now 164 gain = gain >> 5; 166 gain = gain >> 4 [all...] |
eas_fmengine.h | 56 /* LFO modulation to gain control */ 67 EAS_U16 gain; /* current internal gain */ member in struct:__anon28104 68 EAS_U16 outputGain; /* current output gain */ 78 EAS_U16 gainLeft; /* left gain multiplier */ 79 EAS_U16 gainRight; /* right gain multiplier */ 87 EAS_U16 gain[4]; /* initial operator gain value */ member in struct:__anon28106 88 EAS_U16 outputGain[4]; /* initial operator output gain value */ 89 EAS_U16 voiceGain; /* initial voice gain */ 99 EAS_U16 gain[4]; \/* new operator gain value *\/ member in struct:__anon28107 [all...] |
ARM-E_voice_gain_gnu.s | 59 gain .req r8
label 137 LDR gain, [pWTFrame, #m_prevGain]
138 MOV gain, gain, LSL #(NUM_MIXER_GUARD_BITS + 4)
141 SUB gainIncrement, gainIncrement, gain
149 ADD gain, gain, gainIncrement @ gain step to eliminate zipper noise
150 SMULWB tmp0, gain, tmp0 @ sample * local gain
[all...] |
/frameworks/av/media/libstagefright/codecs/aacenc/src/ |
quantize.c | 53 * quaSpectrum = mdctSpectrum^3/4*2^(-(3/16)*gain) 56 static Word16 quantizeSingleLine(const Word16 gain, const Word32 absSpectrum) 67 /* calculate the final fractional exponent times 16 (was 3*(4*e + gain) + (INT_BITS-1)*16) */ 68 minusFinalExp = (e << 2) + gain; 99 * quaSpectrum = mdctSpectrum^3/4*2^(-(3/16)*gain) 100 * input: global gain, number of lines to process, spectral data 104 static void quantizeLines(const Word16 gain, 110 Word32 m = gain&3; 111 Word32 g = (gain >> 2) + 4; 114 /* gain&3 * [all...] |
/external/sonivox/arm-wt-22k/lib_src/ |
eas_mixer.c | 137 EAS_U16 gain; local 141 /* calculate the gain multiplier */ 149 gain = 32767; 151 gain = (EAS_U16) temp; 154 gain = (EAS_U16) pEASData->masterGain; 156 gain = (EAS_U16) pEASData->masterGain; 159 /* Not using all the gain bits for now 164 gain = gain >> 5; 166 gain = gain >> 4 [all...] |
ARM-E_voice_gain_gnu.s | 59 gain .req r8
label 137 LDR gain, [pWTFrame, #m_prevGain]
138 MOV gain, gain, LSL #(NUM_MIXER_GUARD_BITS + 4)
141 SUB gainIncrement, gainIncrement, gain
149 ADD gain, gain, gainIncrement @ gain step to eliminate zipper noise
150 SMULWB tmp0, gain, tmp0 @ sample * local gain
[all...] |
/external/chromium_org/chrome/browser/extensions/api/audio/ |
audio_service.cc | 24 int gain) OVERRIDE; 51 int gain) {
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/external/chromium_org/third_party/WebKit/Source/core/platform/audio/ |
DynamicsCompressor.h | 109 void setEmphasisStageParameters(unsigned stageIndex, float gain, float normalizedFrequency /* 0 -> 1 */); 110 void setEmphasisParameters(float gain, float anchorFreq, float filterStageRatio);
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Cone.h | 36 // Cone gain is defined according to the OpenAL specification 42 // Returns scalar gain for the given source/listener positions/orientations 43 double gain(FloatPoint3D sourcePosition, FloatPoint3D sourceOrientation, FloatPoint3D listenerPosition);
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/external/webrtc/src/modules/audio_coding/codecs/isac/main/source/ |
lpc_analysis.h | 38 double* gain,
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/frameworks/av/media/libstagefright/codecs/amrwbenc/inc/ |
p_med_o.h | 33 Word16 * gain, /* output: normalize correlation of hp_wsp for the Lag */
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/external/aac/libAACenc/src/ |
quantize.cpp | 100 input: global gain, number of lines to process, spectral data 104 static void FDKaacEnc_quantizeLines(INT gain, 111 FIXP_QTD quantizer = FDKaacEnc_quantTableQ[(-gain)&3]; 112 INT quantizershift = ((-gain)>>2)+1; 156 mdctSpectrum = iquaSpectrum^4/3 *2^(0.25*gain) 157 input: global gain, number of lines to process,quantized spectrum 161 static void FDKaacEnc_invQuantizeLines(INT gain, 171 iquantizermod = gain&3; 172 iquantizershift = gain>>2; 292 input: gain, number of lines to process, spectral dat [all...] |
/external/speex/libspeex/ |
ltp.c | 173 void open_loop_nbest_pitch(spx_word16_t *sw, int start, int end, int len, int *pitch, spx_word16_t *gain, int N, char *stack) 260 /* Search for the best pitch prediction gain */ 291 /* Compute open-loop gain if necessary */ 292 if (gain) 302 gain[j]=g; 376 spx_word16_t gain[3]; local 487 gain[0] = ADD16(32,(spx_word16_t)gain_cdbk[best_cdbk*4]); 488 gain[1] = ADD16(32,(spx_word16_t)gain_cdbk[best_cdbk*4+1]); 489 gain[2] = ADD16(32,(spx_word16_t)gain_cdbk[best_cdbk*4+2]); 490 /*printf ("%d %d %d %d\n",gain[0],gain[1],gain[2], best_cdbk);* 676 spx_word16_t gain[3]; local [all...] |
/external/aac/libSBRenc/src/ |
resampler.h | 110 FIXP_DBL gain; /*! overall gain factor */ member in struct:__anon2667
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