| /external/sonivox/arm-hybrid-22k/lib_src/ |
| eas_wtengine.c | 65 * Output gain for individual voice
78 EAS_I32 gain; local
99 gain = pWTIntFrame->prevGain << 16;
108 /* incremental gain step to prevent zipper noise */
110 gain += gainIncrement;
112 tmp2 = gain >> 16;
114 /* scale sample by gain */
464 * optimizations. It calls the interpolator, filter, and gain routines
535 /* apply gain, and left and right gain */
562 EAS_I32 gain; local
[all...] |
| eas_fmsynth.c | 164 if ((pRegion->oper[operIndex].gain & 0xfc) == 0)
338 /* if level control or envelope gain is zero, skip this envelope */
339 if (((pRegion->oper[operIndex].gain & 0xfc) == 0) ||
345 /* if the envelope gain is above the sustain level, we need to catch this voice */
420 /* calculate pan gain values only if stereo output */
434 /* initialize gain value for anti-zipper filter */
442 /* establish operator output gain level */
444 pFMVoice->oper[operIndex].outputGain = EAS_LogToLinear16(((EAS_I16) (pRegion->oper[operIndex].gain & 0xfc) - 0xfc) << 7);
464 /* save static gain parameters */
511 * - the given channel's static gain and static pitch are update
[all...] |
| eas_reverbdata.h | 179 EAS_I16 m_nApGain; // gain for ap
191 EAS_I16 m_nLpfFwd; // lpf forward gain
193 EAS_I16 m_nLpfFbk; // lpf feedback gain
197 EAS_I16 m_nGain[REVERB_MAX_NUM_REFLECTIONS]; // gain for ap
314 EAS_I16 m_nSin; // gain for self taps
316 EAS_I16 m_nCos; // gain for cross taps
318 EAS_I16 m_nSinIncrement; // increment for gain
320 EAS_I16 m_nCosIncrement; // increment for gain
322 EAS_I16 m_nLpfFwd; // lpf forward gain (includes scaling for mixer)
324 EAS_I16 m_nLpfFbk; // lpf feedback gain
[all...] |
| /external/sonivox/arm-wt-22k/lib_src/ |
| eas_wtengine.c | 65 * Output gain for individual voice
78 EAS_I32 gain; local
99 gain = pWTIntFrame->prevGain << 16;
108 /* incremental gain step to prevent zipper noise */
110 gain += gainIncrement;
112 tmp2 = gain >> 16;
114 /* scale sample by gain */
464 * optimizations. It calls the interpolator, filter, and gain routines
535 /* apply gain, and left and right gain */
562 EAS_I32 gain; local
[all...] |
| eas_reverbdata.h | 179 EAS_I16 m_nApGain; // gain for ap
191 EAS_I16 m_nLpfFwd; // lpf forward gain
193 EAS_I16 m_nLpfFbk; // lpf feedback gain
197 EAS_I16 m_nGain[REVERB_MAX_NUM_REFLECTIONS]; // gain for ap
314 EAS_I16 m_nSin; // gain for self taps
316 EAS_I16 m_nCos; // gain for cross taps
318 EAS_I16 m_nSinIncrement; // increment for gain
320 EAS_I16 m_nCosIncrement; // increment for gain
322 EAS_I16 m_nLpfFwd; // lpf forward gain (includes scaling for mixer)
324 EAS_I16 m_nLpfFbk; // lpf feedback gain
[all...] |
| /external/chromium_org/third_party/WebKit/Source/platform/audio/ |
| DynamicsCompressorKernel.cpp | 233 // Makeup gain.
285 // Calculate desired gain
418 // Exponential approach to desired gain.
420 // Attack - reduce gain to desired.
423 // Release - exponentially increase gain to 1.0
428 // Warp pre-compression gain to smooth out sharp exponential transition points.
431 // Calculate total gain using master gain and effect blend.
441 // Apply final gain.
|
| DynamicsCompressor.h | 110 void setEmphasisStageParameters(unsigned stageIndex, float gain, float normalizedFrequency /* 0 -> 1 */);
111 void setEmphasisParameters(float gain, float anchorFreq, float filterStageRatio);
|
| /external/sonivox/arm-fm-22k/lib_src/ |
| eas_fmsynth.c | 164 if ((pRegion->oper[operIndex].gain & 0xfc) == 0)
338 /* if level control or envelope gain is zero, skip this envelope */
339 if (((pRegion->oper[operIndex].gain & 0xfc) == 0) ||
345 /* if the envelope gain is above the sustain level, we need to catch this voice */
420 /* calculate pan gain values only if stereo output */
434 /* initialize gain value for anti-zipper filter */
442 /* establish operator output gain level */
444 pFMVoice->oper[operIndex].outputGain = EAS_LogToLinear16(((EAS_I16) (pRegion->oper[operIndex].gain & 0xfc) - 0xfc) << 7);
464 /* save static gain parameters */
511 * - the given channel's static gain and static pitch are update
[all...] |
| eas_reverbdata.h | 179 EAS_I16 m_nApGain; // gain for ap
191 EAS_I16 m_nLpfFwd; // lpf forward gain
193 EAS_I16 m_nLpfFbk; // lpf feedback gain
197 EAS_I16 m_nGain[REVERB_MAX_NUM_REFLECTIONS]; // gain for ap
314 EAS_I16 m_nSin; // gain for self taps
316 EAS_I16 m_nCos; // gain for cross taps
318 EAS_I16 m_nSinIncrement; // increment for gain
320 EAS_I16 m_nCosIncrement; // increment for gain
322 EAS_I16 m_nLpfFwd; // lpf forward gain (includes scaling for mixer)
324 EAS_I16 m_nLpfFbk; // lpf feedback gain
[all...] |
| /external/webrtc/src/modules/audio_coding/codecs/isac/main/source/ |
| lpc_analysis.c | 147 /* Find average pitch gain */
156 /* If pitch gain is low and energy constant - increase noise level*/
199 /* If pitch gain is low and energy constant - increase noise level*/
323 /* add hearing threshold and compute the gain */
350 /* add hearing threshold and compute of the gain */
483 * -gain : pointer to a buffer where LP gains are written.
491 double* gain,
532 /* add hearing threshold and compute the gain */
533 gain[subFrameCntr] = S_N_R / (sqrt(res_nrg) / *varscale + H_T_H);
|
| /frameworks/av/media/libeffects/lvm/lib/StereoWidening/src/ |
| LVCS_Headphone_Coeffs.h | 244 /* Reverb Gain Settings */
245 #define LVCS_HEADPHONE_DELAYGAIN 0.800000 /* Algorithm delay path gain */
246 #define LVCS_HEADPHONE_OUTPUTGAIN 1.000000 /* Algorithm output gain */
247 #define LVCS_HEADPHONE_PROCGAIN 18403 /* Processed path gain */
248 #define LVCS_HEADPHONE_UNPROCGAIN 18403 /* Unprocessed path gain */
249 #define LVCS_HEADPHONE_GAINCORRECT 1.009343 /* Delay mixer gain correction */
387 /* The Output Gain Correction */
393 #define LVCS_HEADPHONE_GAIN 6840 /* Unprocessed path gain */
396 #define LVCS_EX_HEADPHONE_GAIN 5108 /* EX Unprocessed path gain */
|
| /frameworks/av/media/libstagefright/codecs/amrnb/enc/src/ |
| g_adapt.h | 107 Word16 prev_gc; /* previous code gain, Q1 */
109 Word16 ltpg_mem[LTPG_MEM_SIZE]; /* LTP coding gain history, Q13 */
118 /* initialize one instance of the gain adaptor
125 /* reset of gain adaptor state (i.e. set state memory to zero)
130 /* de-initialize gain adaptor state (i.e. free state struct)
137 * Purpose: calculate pitch/codebook gain adaptation factor alpha
144 Word16 ltpg, /* i : ltp coding gain (log2()), Q */
145 Word16 gain_cod, /* i : code gain, Q13 */
146 Word16 *alpha, /* o : gain adaptation factor, Q15 */
|
| qgain795.h | 108 GainAdaptState *adapt_st, /* i/o: gain adapter state structure */
117 Word16 exp_gcode0, /* i : predicted CB gain (exponent), Q0 */
118 Word16 frac_gcode0, /* i : predicted CB gain (fraction), Q15 */
120 Word16 cod_gain_frac, /* i : opt. codebook gain (fraction),Q15 */
121 Word16 cod_gain_exp, /* i : opt. codebook gain (exponent), Q0 */
122 Word16 gp_limit, /* i : pitch gain limit */
123 Word16 *gain_pit, /* i/o: Pitch gain (unquant/quant), Q14 */
124 Word16 *gain_cod, /* o : Code gain, Q1 */
130 /* (first gain pitch, then code pitch)*/
|
| /frameworks/av/media/libstagefright/codecs/amrwbenc/src/ |
| dtx.c | 167 Word16 log_en, gain, level, exp, exp0, tmp; local
238 /* the result corresponds to log2(gain) in Q10 */
247 /* Subtract 2 from log_en in Q9, i.e divide the gain by 2 (energy by 4) */
263 /* gain = level / sqrt(ener) * sqrt(L_FRAME) */
270 gain = extract_h(ener32);
272 gain = mult(level, gain); /* gain in Q15 */
281 tmp = mult(exc2[i], gain); /* Q0 * Q15 */
|
| /external/chromium_org/third_party/WebKit/Source/modules/webaudio/ |
| PannerNode.cpp | 120 // Get the distance and cone gain.
123 // Snap to desired gain at the beginning.
127 // Apply gain in-place with de-zippering.
137 m_lastGain = -1.0; // force to snap to initial gain
377 double distanceGain = m_distanceEffect.gain(listenerDistance);
382 double coneGain = m_coneEffect.gain(m_position, m_orientation, listenerPosition);
|
| /frameworks/av/media/libstagefright/codecs/amrnb/dec/src/ |
| agc.cpp | 607 st->past_gain = gain
625 sig_out[n] = sig_out[n] * gain[n]
626 gain[n] = agc_fac * gain[n-1] + (1 - agc_fac) g_in/g_out
628 where: gain[n] = gain at the nth sample given by
654 Word16 gain_in, gain_out, g0, gain;
696 // compute gain[n] = agc_fac * gain[n-1]
698 // sig_out[n] = gain[n] * sig_out[n
753 Word16 gain; local
[all...] |
| /frameworks/av/media/libeffects/testlibs/ |
| EffectReverb.h | 109 // The diffusion expressed in permilles changes the Allpass gain in a linear manner in the range defined by
128 int16_t m_nApGain; // gain for ap
140 int16_t m_nGain[REVERB_MAX_NUM_REFLECTIONS]; // gain for ap
220 int16_t m_nSin; // gain for self taps
222 int16_t m_nCos; // gain for cross taps
224 int16_t m_nSinIncrement; // increment for gain
226 int16_t m_nCosIncrement; // increment for gain
228 int16_t m_nRvbLpfFwd; // reverb feedback lpf forward gain (includes scaling for mixer)
230 int16_t m_nRvbLpfFbk; // reverb feedback lpf feedback gain
232 int16_t m_nRoomLpfFwd; // room lpf forward gain (includes scaling for mixer
[all...] |
| /cts/suite/audio_quality/lib/src/audio/ |
| AudioLocal.cpp | 20 bool AudioLocal::prepare(AudioHardware::SamplingRate samplingRate, int gain, int /*mode*/)
23 // gain control not necessary in MobilePre as there is no control.
|
| /device/asus/flo/voice_processing/ |
| voice_processing_descriptors.c | 50 // Automatic Gain Control
58 // "Automatic Gain Control",
|
| /device/lge/hammerhead/voice_processing/ |
| voice_processing_descriptors.c | 50 // Automatic Gain Control 2d416a80-1fcf-11e3-b0b7-0002a5d5c51b
58 // "Automatic Gain Control",
|
| /external/chromium_org/third_party/opus/src/silk/ |
| main.h | 66 /* Find least-squares prediction gain for one signal based on another and quantize it */
169 /* Gain scalar quantization with hysteresis, uniform on log scale */
171 opus_int8 ind[ MAX_NB_SUBFR ], /* O gain indices */
174 const opus_int conditional, /* I first gain is delta coded if 1 */
181 const opus_int8 ind[ MAX_NB_SUBFR ], /* I gain indices */
183 const opus_int conditional, /* I first gain is delta coded if 1 */
187 /* Compute unique identifier of gain indices vector */
189 const opus_int8 ind[ MAX_NB_SUBFR ], /* I gain indices */
|
| resampler_private_up2_HQ.c | 78 /* Apply gain in Q15, convert back to int16 and store to output */
99 /* Apply gain in Q15, convert back to int16 and store to output */
|
| /external/chromium_org/third_party/skia/include/effects/ |
| SkMatrixConvolutionImageFilter.h | 35 @param gain A scale factor applied to each pixel after
55 SkScalar gain,
|
| /external/skia/gm/ |
| matrixconvolution.cpp | 59 SkScalar gain = 0.3f, bias = SkIntToScalar(100); local
64 gain,
|
| /external/skia/include/effects/ |
| SkMatrixConvolutionImageFilter.h | 35 @param gain A scale factor applied to each pixel after
55 SkScalar gain,
|
