| /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:__anon28050
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:__anon28052
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:__anon28053
[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:__anon28100
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:__anon28102
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:__anon28103
[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) {
|
| audio_service.h | 50 // Set the muted and volume/gain properties of a device.
54 int gain) = 0;
|
| /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);
|
| 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);
|
| /external/webrtc/src/modules/audio_coding/codecs/isac/main/source/ |
| lpc_analysis.h | 38 double* gain,
|
| /frameworks/av/media/libstagefright/codecs/amrwbenc/inc/ |
| p_med_o.h | 33 Word16 * gain, /* output: normalize correlation of hp_wsp for the Lag */
|
| /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/aac/libSBRenc/src/ |
| resampler.h | 110 FIXP_DBL gain; /*! overall gain factor */ member in struct:__anon2667
|
