| /frameworks/av/media/libeffects/lvm/lib/Bundle/src/ |
| LVM_Tables.c | 36 {HPF_Fs22050_Gain1_A1, /* Gain setting 1 */
40 {HPF_Fs22050_Gain2_A1, /* Gain setting 2 */
44 {HPF_Fs22050_Gain3_A1, /* Gain setting 3 */
48 {HPF_Fs22050_Gain4_A1, /* Gain setting 4 */
52 {HPF_Fs22050_Gain5_A1, /* Gain setting 5 */
56 {HPF_Fs22050_Gain6_A1, /* Gain setting 6 */
60 {HPF_Fs22050_Gain7_A1, /* Gain setting 7 */
64 {HPF_Fs22050_Gain8_A1, /* Gain setting 8 */
68 {HPF_Fs22050_Gain9_A1, /* Gain setting 9 */
72 {HPF_Fs22050_Gain10_A1, /* Gain setting 10 *
[all...] |
| LVM_Coeffs.h | 34 /* Gain = 1.000000 dB */
41 /* Gain = 2.000000 dB */
48 /* Gain = 3.000000 dB */
55 /* Gain = 4.000000 dB */
62 /* Gain = 5.000000 dB */
69 /* Gain = 6.000000 dB */
76 /* Gain = 7.000000 dB */
83 /* Gain = 8.000000 dB */
90 /* Gain = 9.000000 dB */
97 /* Gain = 10.000000 dB *
[all...] |
| /frameworks/av/media/libstagefright/codecs/amrwbenc/src/ |
| g_pitch.c | 20 * Description:Compute the gain of pitch. Result in Q12 *
21 * if(gain < 0) gain = 0 *
22 * if(gain > 1.2) gain = 1.2 *
29 Word16 G_pitch( /* (o) Q14 : Gain of pitch lag saturated to 1.2 */
32 Word16 g_coeff[], /* : Correlations need for gain quantization. */
37 Word16 xy, yy, exp_xy, exp_yy, gain; local
56 /* If (xy < 0) gain = 0 */
60 /* compute gain = xy/yy *
[all...] |
| gpclip.c | 20 * Description:To avoid unstable synthesis on frame erasure, the gain *
21 * need to be limited(gain pitch < 1.0) when the following *
38 Word16 mem[] /* (o) : memory of gain of pitch clipping algorithm */
47 Word16 mem[] /* (i/o) : memory of gain of pitch clipping algorithm */
60 Word16 mem[] /* (i/o) : memory of gain of pitch clipping algorithm */
90 Word16 gain_pit, /* (i) Q14 : gain of quantized pitch */
91 Word16 mem[] /* (i/o) : memory of gain of pitch clipping algorithm */
94 Word16 gain; local
99 gain = extract_h(L_tmp);
101 if(gain < GAIN_PIT_MIN
[all...] |
| /external/qemu/distrib/sdl-1.2.15/docs/man3/ |
| SDL_ActiveEvent.3 | 9 Uint8 gain;
20 0 if the event is a loss or 1 if it is a gain\&.
|
| /external/webrtc/src/modules/audio_coding/codecs/isac/main/source/ |
| spectrum_ar_model_tables.h | 15 * Gain coefficients and cosine tables.
56 /******************** GAIN Coefficient Tables ***********************/
57 /* cdf for Gain coefficient */
60 /* representation levels for quantized Gain coefficient */
63 /* squared quantization boundary levels for Gain coefficient */
66 /* pointer to Gain cdf table */
69 /* Gain initial index for gain quantizer and cdf table search */
|
| /external/libnfc-nci/src/udrv/include/ |
| udac.h | 59 ** Read current DAC gain
68 ** Current gain setting
79 ** Set the DAC gain
82 ** gain Gain setting
91 void UDAC_Set(tUDAC_GAIN gain);
|
| /frameworks/av/media/libeffects/lvm/lib/StereoWidening/src/ |
| LVCS_Tables.h | 85 /* Gain 100% effect */
86 /* Gain 0% effect */
88 /* The Compression gain is represented by a Q1.15 number to give a range of 0dB */
91 /* 5461 is 1dB compression gain */
92 /* 10923 is 2dB compression gain */
93 /* 32767 is 6dB compression gain */
95 /* The Gain is represented as a Q3.13 number to give a range of +8 to -infinity */
98 /* 32767 is +18dB (x8) gain */
99 /* 4096 is 0dB gain */
100 /* 1024 is -12dB gain */
[all...] |
| LVCS_Process.c | 39 /* ----->| Stereo |->| Reverb |->| Equ |->| Alpha |-->| + |-| Gain |----> */
48 /* their gain to give a near peak to peak output (-0.1dBFS) with a worst case */
50 /* the gain block folloing the sum. */
200 LVM_INT16 Gain = pInstance->VolCorrect.CompMin;
204 Gain = (LVM_INT16)( pInstance->VolCorrect.CompMin
210 NonLinComp_D16(Gain, /* Compressor gain setting */
222 /* Large changes in Gain can cause clicks in output
223 Split data into small blocks and use interpolated gain values */
225 GainStep = (LVM_INT16)(((Gain-pInstance->CompressGain) * LVCS_COMPGAINFRAME)/NumSamples)
[all...] |
| LVCS_BypassMix.c | 46 /* The overall gain of the processed path is set by the gains in the individual */
47 /* processing blocks and by the effect level gain. */
49 /* The unprocessed path must have matching gain for the processed path to ensure */
53 /* The overall gain is corrected by a combination of a shift with saturation and a */
55 /* and also corrects for any excess gain in the shift. */
73 LVM_UINT32 Gain;
81 * Set the transition gain
97 * Calculate the output gain table offset
103 * Setup the mixer gain for the processed path
105 Gain = (LVM_UINT32)(pOutputGainTable[Offset].Loss * pInstance->TransitionGain)
[all...] |
| LVCS_BypassMix.h | 46 LVM_UINT16 Output_Shift; /* Correcting gain output shift */
51 /* Output gain type */
54 /* Output gain settings, Gain = (Loss/32768) * 2^Shift */
|
| /external/chromium_org/third_party/opus/src/silk/float/ |
| scale_vector_FLP.c | 37 silk_float gain,
46 data1[ i + 0 ] *= gain;
47 data1[ i + 1 ] *= gain;
48 data1[ i + 2 ] *= gain;
49 data1[ i + 3 ] *= gain;
54 data1[ i ] *= gain;
|
| scale_copy_vector_FLP.c | 38 silk_float gain,
47 data_out[ i + 0 ] = gain * data_in[ i + 0 ];
48 data_out[ i + 1 ] = gain * data_in[ i + 1 ];
49 data_out[ i + 2 ] = gain * data_in[ i + 2 ];
50 data_out[ i + 3 ] = gain * data_in[ i + 3 ];
55 data_out[ i ] = gain * data_in[ i ];
|
| /external/webrtc/src/modules/audio_coding/codecs/isac/fix/source/ |
| spectrum_ar_model_tables.h | 15 * Gain coefficients and cosine tables.
76 /******************** GAIN Coefficient Tables ***********************/
77 /* cdf for Gain coefficient */
80 /* representation levels for quantized Gain coefficient */
83 /* squared quantization boundary levels for Gain coefficient */
86 /* pointer to Gain cdf table */
89 /* Gain initial index for gain quantizer and cdf table search */
|
| /external/chromium_org/third_party/opus/src/silk/fixed/ |
| process_gains_FIX.c | 44 opus_int32 s_Q16, InvMaxSqrVal_Q16, gain, gain_squared, ResNrg, ResNrgPart, quant_offset_Q10; local
46 /* Gain reduction when LTP coding gain is high */
73 gain = psEncCtrl->Gains_Q16[ k ];
74 gain_squared = silk_ADD_SAT32( ResNrgPart, silk_SMMUL( gain, gain ) );
77 gain_squared = silk_SMLAWW( silk_LSHIFT( ResNrgPart, 16 ), gain, gain );
79 gain = silk_SQRT_APPROX( gain_squared ); /* Q8 */
80 gain = silk_min( gain, silk_int32_MAX >> 8 )
[all...] |
| /external/chromium_org/third_party/WebKit/Source/platform/audio/ |
| Cone.cpp | 45 double ConeEffect::gain(FloatPoint3D sourcePosition, FloatPoint3D sourceOrientation, FloatPoint3D listenerPosition) function in class:WebCore::ConeEffect
48 return 1.0; // no cone specified - unity gain
65 double gain = 1.0; local
69 gain = 1.0;
72 gain = m_outerGain;
77 gain = (1.0 - x) + m_outerGain * x;
80 return gain;
|
| /frameworks/av/media/libeffects/lvm/lib/Common/src/ |
| NonLinComp_D16.c | 42 /* Gain (fractional) Gain (integer) Compression Pk-Pk THD */
52 /* Gain - compression control parameter */
64 void NonLinComp_D16(LVM_INT16 Gain,
91 * gain factor
98 Sample = (LVM_INT16)(Sample + ((Gain * (Sample - Temp)) >> 15));
102 Sample = (LVM_INT16)(Sample + ((Gain * (Sample + Temp)) >> 15));
|
| LVC_Mixer_Init.c | 31 /* integer part of TargetGain and fractional gain values "Target" and */
35 /* Therefore integer gain of 4 is provided by Left Shift of 2 and */
36 /* fraction gain is provided through Current=0.475 and Target=0.625 */
59 while(MaxGain>0){ // Update Shift required to provide integer gain
63 pInstance->Target=TargetGain<<(16-Shift); // Update fractional gain Target
64 pInstance->Current=CurrentGain<<(16-Shift); // Update fractional gain Current
|
| LVC_Mixer_Private.h | 32 LVM_INT32 Target; /* 32 bit number specifying fractional value of Target Gain */
33 LVM_INT32 Current; /* 32 bit number specifying fractional valude of Current Gain */
34 LVM_INT32 Shift; /* Left Shift for Integer part of Gain */
35 LVM_INT32 Delta; /* 32 bit number specifying the fractional value of Delta Gain */
74 /* Gain values should not be more that 1.0 */
87 /* Gain values should not be more that 1.0 */
|
| LVC_Mixer_SetTarget.c | 31 /* integer part of TargetGain and fractional gain values "Target" and */
35 /* Therefore integer gain of 4 is provided by Left Shift of 2 and */
36 /* fraction gain is provided through Current=0.475 and Target=0.625 */
59 while(MaxGain>0){ // Update Shift required to provide integer gain
63 pInstance->Target=TargetGain<<(16-Shift); // Update fractional gain Target
64 pInstance->Current=CurrentGain<<(16-Shift); // Update fractional gain Current
|
| /external/chromium_org/third_party/WebKit/Source/modules/webaudio/ |
| GainNode.cpp | 43 m_gain = AudioParam::create(context, "gain", 1.0, 0.0, 1.0);
55 // FIXME: for some cases there is a nice optimization to avoid processing here, and let the gain change
67 if (gain()->hasSampleAccurateValues()) {
68 // Apply sample-accurate gain scaling for precise envelopes, grain windows, etc.
72 gain()->calculateSampleAccurateValues(gainValues, framesToProcess);
76 // Apply the gain with de-zippering into the output bus.
77 outputBus->copyWithGainFrom(*inputBus, &m_lastGain, gain()->value());
84 // Snap directly to desired gain.
85 m_lastGain = gain()->value();
88 // FIXME: this can go away when we do mixing with gain directly in summing junction of AudioNodeInpu
[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/qemu/distrib/sdl-1.2.15/src/events/ |
| SDL_active.c | 52 int SDL_PrivateAppActive(Uint8 gain, Uint8 state)
58 if ( gain ) {
78 event.active.gain = gain;
87 if ( (state & SDL_APPINPUTFOCUS) && !gain ) {
91 if ( (state & SDL_APPACTIVE) && !gain ) {
|
| /frameworks/av/media/libstagefright/codecs/amrnb/common/include/ |
| d_gain_p.h | 39 * Purpose : Decodes the pitch gain using the received index.
63 * Purpose : Decodes the pitch gain using the received index.
64 * Description : In case of no frame erasure, the gain is obtained
66 * otherwise, a downscaled past gain is used.
67 * Returns : Quantized pitch gain
71 Word16 d_gain_pitch( /* return value: gain (Q14) */
|
| /external/sonivox/arm-fm-22k/lib_src/ |
| eas_fmengine.h | 56 /* LFO modulation to gain control */
67 EAS_U16 gain; /* current internal gain */ member in struct:__anon26244
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:__anon26246
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:__anon26247
[all...] |
