| /external/sonivox/arm-hybrid-22k/lib_src/ |
| eas_fmengine.h | 56 /* LFO modulation to gain control */
67 EAS_U16 gain; /* current internal gain */ member in struct:__anon26294
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:__anon26296
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:__anon26297
[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/libeffects/lvm/lib/Common/src/ |
| AGC_MIX_VOL_2St1Mon_D32_WRA.c | 51 /* | Gain | |___| | Gain | | */
55 /* |-------------------------------| AGC Gain |<--| Peak |<--| */
90 LVM_INT16 AGC_Mult; /* Short AGC gain */
97 LVM_INT32 AGC_Gain = pInstance->AGC_Gain; /* Get the current AGC gain */
98 LVM_INT32 AGC_MaxGain = pInstance->AGC_MaxGain; /* Get maximum AGC gain */
118 AGC_Mult = (LVM_INT16)(AGC_Gain >> 16); /* Get the short AGC gain */
119 Vol_Mult = (LVM_INT16)(Vol_Current >> 16); /* Get the short volume gain */
131 * Apply the AGC gain to the mono input and mix with the stereo signal
154 * Update the AGC gain
[all...] |
| LVC_Mixer_VarSlope_SetTimeConstant.c | 28 /* This function calculates the step change for fractional gain for a */
37 /* go from linear fractional gain of 0 to 0.99999999 */
42 /* Delta - the step change for fractional gain per 4 samples */
71 /* Get gain values */
|
| /external/speex/include/speex/ |
| speex_preprocess.h | 7 * gain control (AGC) and voice activity detection (VAD).
42 * gain control (AGC) and voice activity detection (VAD).
103 /** Set preprocessor Automatic Gain Control state */
105 /** Get preprocessor Automatic Gain Control state */
113 /** Set preprocessor Automatic Gain Control level (float) */
115 /** Get preprocessor Automatic Gain Control level (float) */
163 /** Set maximal gain increase in dB/second (int32) */
166 /** Get maximal gain increase in dB/second (int32) */
169 /** Set maximal gain decrease in dB/second (int32) */
172 /** Get maximal gain decrease in dB/second (int32) *
[all...] |
| /frameworks/av/media/libeffects/lvm/lib/Bass/src/ |
| LVDBE_Tables.h | 62 /* Gain for use without the high pass filter */
65 /* Gain for use with the high pass filter */
70 /* Volume control gain and time constant tables */
|
| /frameworks/av/media/libeffects/lvm/lib/Common/lib/ |
| AGC.h | 43 LVM_INT32 AGC_Gain; /* The current AGC gain */
44 LVM_INT32 AGC_MaxGain; /* The maximum AGC gain */
50 LVM_INT16 AGC_GainShift; /* The gain shift */
|
| CompLim.h | 47 LVM_INT16 Shift; /* Shift gain */
56 LVM_INT16 SoftClipGain; /* Soft clip gain control */
70 void NonLinComp_D16(LVM_INT16 Gain,
|
| /frameworks/av/media/libstagefright/codecs/amrwbenc/src/ |
| updt_tar.c | 31 Word16 gain, /* (i) Q14 : adaptive codebook gain */
41 L_tmp -= (y[i] * gain)<<1;
|
| /external/chromium_org/third_party/opus/src/silk/ |
| gain_quant.c | 38 /* Gain scalar quantization with hysteresis, uniform on log scale */
40 opus_int8 ind[ MAX_NB_SUBFR ], /* O gain indices */
43 const opus_int conditional, /* I first gain is delta coded if 1 */
53 /* Round towards previous quantized gain (hysteresis) */
68 /* Double the quantization step size for large gain increases, so that the max gain level can be reached */
95 const opus_int8 ind[ MAX_NB_SUBFR ], /* I gain indices */
97 const opus_int conditional, /* I first gain is delta coded if 1 */
105 /* Gain index is not allowed to go down more than 16 steps (~21.8 dB) */
126 /* Compute unique identifier of gain indices vector *
[all...] |
| /external/jmonkeyengine/engine/src/core/com/jme3/audio/ |
| Environment.java | 52 private float gain = 0.316f; field in class:Environment
88 this.gain = source.gain;
97 public Environment(float density, float diffusion, float gain, float gainHf,
104 this.gain = gain;
123 gain = eaxDbToAmp(e[3]); // convert
202 return gain;
205 public void setGain(float gain) {
206 this.gain = gain
[all...] |
| /frameworks/av/media/libeffects/loudness/dsp/core/ |
| dynamic_range_compression.h | 30 // An adaptive dynamic range compression algorithm. The gain adaptation is made
39 // fed to the compressor. The compressor is tuned according to the target gain
42 // Target gain receives values between 0.0 and 10.0. The knee threshold is
43 // reduced as the target gain increases in order to fit the increased range of
71 // Sets knee threshold via the target gain using an experimentally derived
98 // the latest gain factor that was applied to the input signal
108 // This interpolator provides the function that relates target gain to knee
|
| /frameworks/av/media/libeffects/lvm/lib/Eq/src/ |
| LVEQNB_Tables.c | 70 * Gain table
72 const LVM_INT16 LVEQNB_GainTable[] = {LVEQNB_Gain_Neg15_dB, /* -15dB gain */
87 LVEQNB_Gain_0_dB, /* 0dB gain */
102 LVEQNB_Gain_15_dB}; /* +15dB gain */
106 * D table for 100 / (Gain + 1)
108 const LVM_INT16 LVEQNB_DTable[] = {LVEQNB_100D_Neg15_dB, /* -15dB gain */
123 LVEQNB_100D_0_dB}; /* 0dB gain */
|
| LVEQNB_CalcCoef.c | 63 /* GaindB is the gain in dBs, range -15dB to +15dB */
74 /* small errors in this value have a combined effect on the Q and Gain but not the */
93 LVM_INT16 Gain = pFilterDefinition->Gain;
118 if (Gain >= 0)
124 D = LVEQNB_DTable[Gain+15]; /* D = 1 / (1 + G) if GaindB < 0 */
166 pCoefficients->G = LVEQNB_GainTable[Gain+15];
201 /* GaindB is the gain in dBs, range -15dB to +15dB */
223 LVM_INT16 Gain = pFilterDefinition->Gain;
[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/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-wt-22k/lib_src/ |
| 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/libeffects/lvm/lib/StereoWidening/src/ |
| LVCS_Control.c | 128 LVM_UINT32 Gain;
130 Gain = (LVM_UINT32)(pOutputGainTable[Offset].Loss * LVM_MAXINT_16);
131 Gain = (LVM_UINT32)pOutputGainTable[Offset].UnprocLoss * (Gain >> 15);
132 Gain=Gain>>15;
134 * Apply the gain correction and shift, note the result is in Q3.13 format
136 Gain = (Gain * pInstance->VolCorrect.GainMin) >>12;
138 LVC_Mixer_Init(&pInstance->BypassMix.Mixer_Instance.MixerStream[1],0,Gain);
[all...] |
| LVCS_Tables.c | 369 /* Gain 100% effect */
370 /* Gain 0% effect */
372 /* The Compression gain is represented by a Q1.15 number to give a range of 0dB */
375 /* 5461 is 1dB compression gain */
376 /* 10923 is 2dB compression gain */
377 /* 32767 is 6dB compression gain */
379 /* The Gain is represented as a Q3.13 number to give a range of +8 to -infinity */
382 /* 32767 is +18dB (x8) gain */
383 /* 4096 is 0dB gain */
384 /* 1024 is -12dB gain */
[all...] |
| /external/webrtc/src/common_audio/signal_processing/ |
| vector_scaling_operations.c | 95 WebRtc_Word16 gain, WebRtc_Word16 in_vector_length,
98 // Performs vector operation: out_vector = (gain*in_vector)>>right_shifts
108 (*outptr++) = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(*inptr++, gain, right_shifts);
113 WebRtc_Word16 gain, WebRtc_Word16 in_vector_length,
116 // Performs vector operation: out_vector = (gain*in_vector)>>right_shifts
127 tmpW32 = WEBRTC_SPL_MUL_16_16_RSFT(*inptr++, gain, right_shifts);
|
| ilbc_specific_functions.c | 89 WebRtc_Word16 gain, WebRtc_Word32 add_constant,
100 (*outPtr++) += (WebRtc_Word16)((WEBRTC_SPL_MUL_16_16((*inPtr++), gain)
106 WebRtc_Word16 gain, WebRtc_Word32 add_constant,
117 (*outPtr++) = (WebRtc_Word16)((WEBRTC_SPL_MUL_16_16((*inPtr++), gain)
|
| /cts/suite/audio_quality/test_description/ |
| dut_playback_thd.xml | 25 <output device="DUT" id="sound1" gain="100" sync="start" waitforcompletion="0" />
28 <input device="host" id="dummy" gain="100" time="1000" sync="complete" />
29 <input device="host" id="host_in_$j" gain="100" time="2000" sync="complete" />
|
| /external/webrtc/src/modules/audio_coding/codecs/isac/fix/source/ |
| entropy_coding.h | 70 /* decode & dequantize squared Gain */
74 /* quantize & code squared Gain (input is squared gain) */
|
| /cts/suite/audio_quality/test_description/test/ |
| test_io.xml | 28 <output device="host" id="sound1" gain="100" sync="start"/>
29 <output device="DUT" id="sound1" gain="100" sync="start"/>
30 <input device="host" id="host1" gain="100" time="500" sync="start"/>
31 <input device="DUT" id="device1" gain="100" time="500" sync="start"/>
|
| /external/chromium_org/chrome/common/extensions/api/ |
| audio.idl | 32 // The input gain ranging from 0.0 to 1.0.
33 double gain;
42 // If this is an input device then this field indicates the input gain.
44 double? gain;
|
