/external/webrtc/src/modules/audio_coding/codecs/isac/main/source/ |
encode_lpc_swb.h | 15 * encode LPC parameters (Shape & gain) of the upper band. 190 * Transform the LPC gain to log domain then remove the mean value. 193 * -lpcGain : pointer to LPC Gain, expecting 6 LPC gains 206 * multiplying gain vector with decorrelating matrix. 209 * -data : LPC gain in log-domain with mean removed. 274 * -lpcGain : LPC gain in log-domain & mean removed 277 * -lpcGain : LPC gain in normal domain.
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pitch_lag_tables.h | 25 /* tables for use with small pitch gain */ 53 /* tables for use with medium pitch gain */ 81 /* tables for use with large pitch gain */
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/frameworks/av/media/libeffects/testlibs/ |
AudioShelvingFilter.h | 26 // A shelving audio filter, with unity skirt gain, and controllable cutoff 27 // frequency and gain. 61 // gain: 0 68 // Sets gain value. Actual change will only take place upon commit(). 70 // millibel Gain value in millibel (1/100 of decibel). 73 // Gets the gain, in millibel, as set. 121 // Fractional index into the gain dimension of the coef table in
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AudioEqualizer.h | 30 // shelf, where each band has frequency and gain controls, and the peaking 36 // Gain in millibel. 37 int32_t gain; member in struct:android::AudioEqualizer::BandConfig 93 // gain: 0 105 // Sets gain value. Actual change will only take place upon commit(). 107 // band The band to set the gain for. 108 // millibel Gain value in millibel (1/100 of decibel). 111 // Gets gain of a certain band. This is always the last value set (or 113 // band The band to get the gain for. 114 // returns Gain value in millibel (1/100 of decibel) [all...] |
/cts/suite/audio_quality/test_description/ |
dut_recording_spectrum.xml | 27 <output device="host" id="sound1" gain="100" sync="start" waitforcompletion="0" /> 29 <input device="host" id="host_in_$j" gain="100" time="6000" sync="start" /> 30 <input device="DUT" id="dut_in_$j" gain="100" time="4000" sync="start" />
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dut_recording_thd.xml | 27 <output device="host" id="sound1" gain="100" sync="start" waitforcompletion="0" /> 29 <input device="host" id="host_in_$j" gain="100" time="4000" sync="start" /> 30 <input device="DUT" id="dut_in_$j" gain="100" time="2000" sync="start" />
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dut_speaker_calibration.xml | 18 <case name="dut_speaker_calibration" version="1.0" description="Calibrate host recording gain"> 30 <output device="DUT" id="sound1" gain="100" sync="start"/> 32 <input device="host" id="host_in" gain="100" time="250" sync="complete" />
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dut_speaker_calibration_no_pass.xml | 18 <case name="dut_speaker_calibration" version="1.0" description="Calibrate host recording gain"> 30 <output device="DUT" id="sound1" gain="100" sync="start" waitforcompletion="1" /> 32 <input device="host" id="host_in" gain="100" time="250" sync="complete" />
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/external/chromium_org/third_party/WebKit/Source/modules/webaudio/ |
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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/external/chromium_org/third_party/WebKit/Source/platform/audio/ |
Cone.h | 37 // Cone gain is defined according to the OpenAL specification 43 // Returns scalar gain for the given source/listener positions/orientations 44 double gain(FloatPoint3D sourcePosition, FloatPoint3D sourceOrientation, FloatPoint3D listenerPosition);
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Distance.cpp | 52 double DistanceEffect::gain(double distance) function in class:WebCore::DistanceEffect 75 // We want a gain that decreases linearly from m_refDistance to 76 // m_maxDistance. The gain is 1 at m_refDistance.
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/external/chromium_org/third_party/opus/src/celt/ |
vq.h | 52 , opus_val16 gain 64 ec_dec *dec, opus_val16 gain); 66 void renormalise_vector(celt_norm *X, int N, opus_val16 gain);
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/external/webrtc/src/modules/audio_coding/codecs/isac/fix/source/ |
pitch_lag_tables.h | 27 /* tables for use with small pitch gain */ 53 /* tables for use with medium pitch gain */ 78 /* tables for use with large pitch gain */
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/frameworks/av/media/libeffects/lvm/lib/Common/src/ |
LVC_Mixer_SetTimeConstant.c | 27 /* This function calculates the step change for fractional gain for a */ 36 /* go from linear fractional gain of 0 to 0.99999999 */ 41 /* Delta - the step change for fractional gain per 4 samples */
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/frameworks/av/media/libeffects/lvm/lib/Reverb/src/ |
LVREV_Tables.c | 53 output gain/attenuation is applied. This gain depends on T60 and Rooms 64 The gain is calculated using variable x=(T60*32767/7000)*32768;
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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, 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/chromium/chrome/browser/ui/cocoa/ |
keystone_infobar.h | 17 // gain authorization to perform the promotion. The info bar is not shown
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/external/chromium_org/chrome/browser/ui/cocoa/ |
keystone_infobar_delegate.h | 16 // gain authorization to perform the promotion. The info bar is not shown
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/frameworks/av/media/libeffects/lvm/lib/Bundle/src/ |
LVM_Tables.h | 45 /* Volume control gain and time constant tables */
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/frameworks/av/media/libstagefright/codecs/aacenc/inc/ |
quantize.h | 40 Word16 gain);
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/frameworks/base/media/java/android/media/audiofx/ |
AutomaticGainControl.java | 22 * Automatic Gain Control (AGC). 23 * <p>Automatic Gain Control (AGC) is an audio pre-processing which automatically normalizes the 47 * Checks if the device implements automatic gain control. 48 * @return true if the device implements automatic gain control, false otherwise.
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/frameworks/wilhelm/src/itf/ |
IOutputMixExt.c | 34 /** \brief Summary of the gain, as an optimization for the mixer */ 216 float gain = track->mGains[channel]; local 217 gains[channel] = gain; 219 if (gain <= 0.001) { 221 } else if (gain >= 0.999) { 241 // apply gain during add 247 // no gain adjustment needed, so do a simple add 255 // apply gain during copy 261 // no gain adjustment needed, so do a simple copy 426 /** \brief Called when a gain-related field (mute, solo, volume, stereo position, etc.) updated * 447 float gain; local [all...] |
/frameworks/wilhelm/src/ |
platform.h | 20 #define PLATFORM_MILLIBEL_MAX_VOLUME 0 // No more than unity gain
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/pdk/apps/CameraITS/tests/ |
test_param_noise_reduction.py | 27 and a high analog gain to ensure the captured image is noisy. 30 Also captures an image with low gain and NR off, and uses the variance 54 # NR mode 0 with low gain 69 # NR modes 0, 1, 2 with high gain
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/external/chromium_org/chrome/browser/extensions/api/audio/ |
audio_service.h | 50 // Set the muted and volume/gain properties of a device. 54 int gain) = 0;
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