| /hardware/bsp/intel/peripheral/libupm/src/grovevdiv/ |
| grovevdiv.cxx | 62 float GroveVDiv::computedValue(uint8_t gain, unsigned int val, int vref, int res)
64 return ((float(gain) * float(val) * float(vref) / float(res)) / 1000.0);
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| grovevdiv.h | 85 * @param gain Gain switch, either 3 or 10 for Grove
92 float computedValue(uint8_t gain, unsigned int val, int vref=GROVEVDIV_VREF,
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| /hardware/bsp/intel/peripheral/libupm/src/hx711/ |
| hx711.h | 63 * @param gain Defines the gain factor
64 * Valid values are 128 or 64 for channel A; channel B works with a 32-gain factor only
66 HX711(uint8_t data, uint8_t sck, uint8_t gain = 128);
81 * Sets the gain factor; takes effect only after a call to read()
82 * channel A can be set for a 128 or 64 gain; channel B has a fixed 32-gain
84 * @param gain Defines the gain factor
86 void setGain(uint8_t gain = 128)
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| /system/bt/btif/include/ |
| btif_avrcp_audio_track.h | 53 * Sets audio track gain.
55 void BtifAvrcpSetAudioTrackGain(void *handle, float gain);
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| /external/sonivox/arm-wt-22k/lib_src/ |
| eas_dlssynth.c | 182 * Calculate the gain for the next frame
185 static EAS_I32 DLS_UpdateGain (S_WT_VOICE *pWTVoice, const S_DLS_ARTICULATION *pDLSArt, S_SYNTH_CHANNEL *pChannel, EAS_I32 gain, EAS_U8 velocity)
201 gain += FMUL_15x15(temp, pWTVoice->modLFO.lfoValue);
202 if (gain > 0)
203 gain = 0;
205 /* convert to linear gain including EG1 */
208 gain = (DLS_GAIN_FACTOR * gain) >> DLS_GAIN_SHIFT;
211 gain += (pWTVoice->eg1Value - 32767) >> 1;
212 gain = EAS_LogToLinear16(gain)
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| eas_wt_IPC_frame.h | 74 EAS_I16 gainLeft; /* left channel gain */
75 EAS_I16 gainRight; /* right channel gain */
78 EAS_I16 gain; /* current voice gain */ member in struct:s_wt_config_tag
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| 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
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| 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
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| /device/google/dragon/audio/hal/dsp/ |
| biquad.h | 48 * gain - The value is in dB. See Web Audio API for details.
51 double gain);
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| eq.h | 36 * Q, gain - The meaning depends on the type of the filter. See Web Audio
42 float gain);
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| eq2.h | 37 * Q, gain - The meaning depends on the type of the filter. See Web Audio
43 enum biquad_type type, float freq, float Q, float gain);
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| /external/webrtc/webrtc/tools/agc/ |
| test_utils.h | 19 void ApplyGainLinear(float gain, float last_gain, AudioFrame* frame);
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| /frameworks/av/media/libstagefright/codecs/aacenc/inc/ |
| quantize.h | 40 Word16 gain);
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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
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| 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
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| /external/libopus/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) *
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| /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 *
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| /external/fec/ |
| sim.c | 33 unsigned char addnoise(int sym,double amp,double gain,double offset,int clip){
36 sample = offset + gain*normal_rand(sym?amp:-amp,1.0);
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| /frameworks/base/media/java/android/media/ |
| AudioMixPort.java | 46 AudioGainConfig gain) {
47 return new AudioMixPortConfig(this, samplingRate, channelMask, format, gain);
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| AudioPort.java | 27 * - gain: a port can be associated with one or more gain controllers (see AudioGain).
156 * Get the list of gain descriptors
157 * Empty array if this port does not have gain control
164 * Get the gain descriptor at a given index
166 AudioGain gain(int index) { method in class:AudioPort
180 * @param gain The desired gain. null if no gain changed requested.
183 AudioGainConfig gain) {
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| /external/webrtc/webrtc/common_audio/signal_processing/ |
| ilbc_specific_functions.c | 68 int16_t gain, int32_t add_constant,
76 out[i] += (int16_t)((in[i] * gain + add_constant) >> right_shifts);
81 int16_t gain, int32_t add_constant,
88 out[i] = (int16_t)((in[i] * gain + add_constant) >> right_shifts);
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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,
112 FIXP_QTD quantizer = FDKaacEnc_quantTableQ[(-gain)&3];
113 INT quantizershift = ((-gain)>>2)+1;
162 mdctSpectrum = iquaSpectrum^4/3 *2^(0.25*gain)
163 input: global gain, number of lines to process,quantized spectrum
167 static void FDKaacEnc_invQuantizeLines(INT gain,
177 iquantizermod = gain&3;
178 iquantizershift = gain>>2;
300 input: gain, number of lines to process, spectral dat
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| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
| pitch_filter_mips.c | 14 int16_t gain,
36 // Load coefficients outside the loop and sign-extend gain and sign
49 "seh %[gain32], %[gain] \n\t"
55 : [coefficient] "r" (coefficient), [gain] "r" (gain),
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| /external/aac/libSBRenc/src/ |
| resampler.h | 110 FIXP_DBL gain; /*! overall gain factor */ member in struct:__anon6415
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