/external/chromium_org/chromeos/audio/ |
audio_devices_pref_handler.h | 26 // only have either a gain or a volume for a device (depending on whether it 30 // Sets the audio volume or gain value to prefs for a device.
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/external/chromium_org/third_party/WebKit/Source/modules/webaudio/ |
AudioBuffer.idl | 36 attribute float gain; // linear gain (default 1.0)
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PannerNode.h | 43 // A distance effect will attenuate the gain as the position moves away from the listener. 44 // A cone effect will attenuate the gain as the orientation moves away from the listener. 125 // Accessors for dynamically calculated gain values. 135 // Returns the combined distance and cone gain attenuation. 149 // Gain
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/external/webrtc/src/modules/audio_coding/codecs/isac/fix/source/ |
spectrum_ar_model_tables.c | 14 * This file contains tables with AR coefficients, Gain coefficients 122 /******************** GAIN Coefficient Tables ***********************/ 124 /* cdf for Gain coefficient */ 130 /* representation levels for quantized squared Gain coefficient */ 136 /* quantization boundary levels for squared Gain coefficient */ 142 /* pointers to Gain cdf table */ 147 /* gain initial index for gain quantizer and cdf table search */
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/external/webrtc/src/modules/audio_coding/codecs/isac/main/source/ |
decode.c | 64 float gain; local 142 /* Convert AvgPitchGain back to float for computation of gain. */ 144 gain = 1.0f - 0.45f * (float)AvgPitchGain; 147 /* Reduce gain to compensate for pitch enhancer. */ 148 LPw_pf[k] *= gain; 153 /* Compensation for transcoding gain changes. */ 196 const WebRtc_Word16 kAveragePitchGain = 0; /* No pitch-gain for upper-band. */ 262 const WebRtc_Word16 kAveragePitchGain = 0; /* No pitch-gain for upper-band. */
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spectrum_ar_model_tables.c | 97 /******************** GAIN Coefficient Tables ***********************/ 98 /* cdf for Gain coefficient */ 103 /* representation levels for quantized squared Gain coefficient */ 107 /* quantization boundary levels for squared Gain coefficient */ 111 /* pointers to Gain cdf table */ 114 /* Gain initial index for gain quantizer and cdf table search */
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/external/webrtc/src/modules/audio_processing/agc/interface/ |
gain_control.h | 101 * It is a digital gain applied to the input signal and is used in the 133 * This function processes a 10/20ms frame and adjusts (normalizes) the gain 134 * both analog and digitally. The gain adjustments are done only during 159 * - out : Gain-adjusted near-end speech vector (L band) 161 * - out_H : Gain-adjusted near-end speech vector (H band) 241 * : 1 - Adaptive Analog Automatic Gain Control -3dBOv 242 * : 2 - Adaptive Digital Automatic Gain Control -3dBOv 243 * : 3 - Fixed Digital Gain 0dB
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/frameworks/av/media/libstagefright/codecs/aacenc/inc/ |
tns.h | 37 Word16 threshOn; /* min. prediction gain for using tns TABUL * 100*/ 71 Word16 threshold; /* min. prediction gain for using tns TABUL * 100 */
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/external/chromium_org/third_party/opus/src/silk/ |
LPC_inv_pred_gain.c | 39 /* Compute inverse of LPC prediction gain, and */ 41 static opus_int32 LPC_inverse_pred_gain_QA( /* O Returns inverse prediction gain in energy domain, Q30 */ 71 /* Update inverse gain */ 99 /* Update inverse gain */ 109 opus_int32 silk_LPC_inverse_pred_gain( /* O Returns inverse prediction gain in energy domain, Q30 */ 136 opus_int32 silk_LPC_inverse_pred_gain_Q24( /* O Returns inverse prediction gain in energy domain, Q30 */
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define.h | 113 /* dB level of lowest gain quantization level */ 115 /* dB level of highest gain quantization level */ 117 /* Number of gain quantization levels */ 119 /* Max increase in gain quantization index */ 121 /* Max decrease in gain quantization index */
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/external/webrtc/src/modules/audio_processing/agc/ |
analog_agc.h | 24 /* Analog Automatic Gain Control variables: 50 WebRtc_Word16 compressionGaindB; // Fixed gain level in dB 105 WebRtc_UWord16 gainTableIdx; // Current position in virtual gain table 106 WebRtc_Word32 micGainIdx; // Gain index of mic level to increase slowly 108 WebRtc_Word32 maxLevel; // Max possible vol level, incl dig gain 113 WebRtc_Word32 zeroCtrlMax; // Remember max gain => don't amp low input
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digital_agc.c | 32 // % Matlab code for plotting the gain and input/output level characteristic (copy/paste the following 3 lines): 34 // subplot(121); plot(in, out); axis([-30, 0, -5, 20]); grid on; xlabel('Input (dB)'); ylabel('Gain (dB)'); 67 // This function generates the compressor gain table used in the fixed digital part. 89 // Calculate maximum digital gain and zero gain level 104 // Calculate the difference between maximum gain and gain at 0dB0v: 266 // start at minimum to find correct gain faster 270 // start out with 0 dB gain 274 stt->gain = 65536 [all...] |
/frameworks/base/media/java/android/media/ |
FocusRequester.java | 44 * the audio focus gain request that caused the addition of this object in the focus stack. 111 return "GAIN"; 141 + " -- gain: " + focusGainToString() 166 * For a given audio focus gain request, return the audio focus loss type that will result 169 * @return the audio focus loss type that matches the gain request 223 Log.e(TAG, "Failure to signal gain of audio focus due to: ", e);
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/cts/tests/tests/security/src/android/security/cts/ |
ListeningPortsTest.java | 52 * Remotely accessible ports are often used by attackers to gain 61 * Remotely accessible ports are often used by attackers to gain 70 * Remotely accessible ports are often used by attackers to gain 79 * Remotely accessible ports are often used by attackers to gain 89 * installed programs to gain unauthorized access to program data or 105 * installed programs to gain unauthorized access to program data or 121 * installed programs to gain unauthorized access to program data or 137 * installed programs to gain unauthorized access to program data or 175 * attackers to gain unauthorized access to computers systems without 179 * malicious locally installed programs to gain unauthorized access t [all...] |
/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 [all...] |
/external/bluetooth/bluedroid/stack/include/ |
uipc_msg.h | 694 #define AUDIO_ROUTE_EQ_CONFIG_GAIN 0xFF /* Custion Gain Config */ 730 /* For custon equalizer gain configuration */ 733 UINT32 audio_l_g0; /* IIR biquad filter left ch gain 0 */ 734 UINT32 audio_l_g1; /* IIR biquad filter left ch gain 1 */ 735 UINT32 audio_l_g2; /* IIR biquad filter left ch gain 2 */ 736 UINT32 audio_l_g3; /* IIR biquad filter left ch gain 3 */ 737 UINT32 audio_l_g4; /* IIR biquad filter left ch gain 4 */ 738 UINT32 audio_l_gl; /* IIR biquad filter left ch global gain */ 739 UINT32 audio_r_g0; /* IIR biquad filter left ch gain 0 */ 740 UINT32 audio_r_g1; /* IIR biquad filter left ch gain 1 * [all...] |
/external/chromium_org/third_party/WebKit/Source/platform/audio/ |
DynamicsCompressor.cpp | 98 void DynamicsCompressor::setEmphasisStageParameters(unsigned stageIndex, float gain, float normalizedFrequency /* 0 -> 1 */) 100 float gk = 1 - gain / 20; 122 void DynamicsCompressor::setEmphasisParameters(float gain, float anchorFreq, float filterStageRatio) 124 setEmphasisStageParameters(0, gain, anchorFreq); 125 setEmphasisStageParameters(1, gain, anchorFreq / filterStageRatio); 126 setEmphasisStageParameters(2, gain, anchorFreq / (filterStageRatio * filterStageRatio)); 127 setEmphasisStageParameters(3, gain, anchorFreq / (filterStageRatio * filterStageRatio * filterStageRatio));
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AudioBus.cpp | 429 // If it is copying from the same bus and no need to change gain, just return. 442 // We don't want to suddenly change the gain from mixing one time slice to the next, 443 // so we "de-zipper" by slowly changing the gain each sample-frame until we've achieved the target gain. 445 // Take master bus gain into account as well as the targetGain. 449 float gain = static_cast<float>(m_isFirstTime ? totalDesiredGain : *lastMixGain); local 455 // If the gain is within epsilon of totalDesiredGain, we can skip dezippering. 458 float gainDiff = fabs(totalDesiredGain - gain); 460 // Number of frames to de-zipper before we are close enough to the target gain. 461 // FIXME: framesToDezipper could be smaller when target gain is close enough within this process loop [all...] |
/frameworks/base/media/java/android/media/audiofx/ |
LoudnessEnhancer.java | 29 * The processing is parametrized by a target gain value, which determines the maximum amount 46 * The maximum gain applied applied to the signal to process. 114 * Set the target gain for the audio effect. 115 * The target gain is the maximum value by which a sample value will be amplified when the 117 * @param gainmB the effect target gain expressed in mB. 0mB corresponds to no amplification. 128 * Return the target gain. 129 * @return the effect target gain expressed in mB.
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/external/sonivox/arm-fm-22k/lib_src/ |
eas_math.h | 155 /* the max positive gain used in the synth for EG1 */ 168 We implement the EG1 using a linear gain value, which means that the 169 attack segment is handled by incrementing (adding) the linear gain. 171 the Attack portion. For Decay, Sustain, and Release, the gain is 173 a linear scale. Because we use a linear gain for EG1, we implement 180 #define MULT_EG1_EG1(gain,damping) /*lint -e(704) <avoid divide for performance>*/ \ 183 ((EAS_I32)(gain)) * ((EAS_I32)(damping)) \ 199 #define MULT_EG1_EG1_X2(gain,damping) /*lint -e(702) <avoid divide for performance>*/ \ 202 ((EAS_I32)(gain)) * ((EAS_I32)(damping)) \ 237 For gain, the LFO generates a value that modulates in term [all...] |
/external/sonivox/arm-hybrid-22k/lib_src/ |
eas_math.h | 155 /* the max positive gain used in the synth for EG1 */ 168 We implement the EG1 using a linear gain value, which means that the 169 attack segment is handled by incrementing (adding) the linear gain. 171 the Attack portion. For Decay, Sustain, and Release, the gain is 173 a linear scale. Because we use a linear gain for EG1, we implement 180 #define MULT_EG1_EG1(gain,damping) /*lint -e(704) <avoid divide for performance>*/ \ 183 ((EAS_I32)(gain)) * ((EAS_I32)(damping)) \ 199 #define MULT_EG1_EG1_X2(gain,damping) /*lint -e(702) <avoid divide for performance>*/ \ 202 ((EAS_I32)(gain)) * ((EAS_I32)(damping)) \ 237 For gain, the LFO generates a value that modulates in term [all...] |
/external/sonivox/arm-wt-22k/lib_src/ |
eas_math.h | 155 /* the max positive gain used in the synth for EG1 */ 168 We implement the EG1 using a linear gain value, which means that the 169 attack segment is handled by incrementing (adding) the linear gain. 171 the Attack portion. For Decay, Sustain, and Release, the gain is 173 a linear scale. Because we use a linear gain for EG1, we implement 180 #define MULT_EG1_EG1(gain,damping) /*lint -e(704) <avoid divide for performance>*/ \ 183 ((EAS_I32)(gain)) * ((EAS_I32)(damping)) \ 199 #define MULT_EG1_EG1_X2(gain,damping) /*lint -e(702) <avoid divide for performance>*/ \ 202 ((EAS_I32)(gain)) * ((EAS_I32)(damping)) \ 237 For gain, the LFO generates a value that modulates in term [all...] |
/frameworks/av/media/libeffects/testlibs/ |
EffectsMath.h | 146 /* the max positive gain used in the synth for EG1 */ 159 We implement the EG1 using a linear gain value, which means that the 160 attack segment is handled by incrementing (adding) the linear gain. 162 the Attack portion. For Decay, Sustain, and Release, the gain is 164 a linear scale. Because we use a linear gain for EG1, we implement 171 #define MULT_EG1_EG1(gain,damping) /*lint -e(704) <avoid divide for performance>*/ \ 174 ((int32_t)(gain)) * ((int32_t)(damping)) \ 190 #define MULT_EG1_EG1_X2(gain,damping) /*lint -e(702) <avoid divide for performance>*/ \ 193 ((int32_t)(gain)) * ((int32_t)(damping)) \ 228 For gain, the LFO generates a value that modulates in term [all...] |
/external/chromium_org/third_party/opus/src/celt/ |
vq.c | 70 opus_val16 gain, theta; local 78 gain = celt_div((opus_val32)MULT16_16(Q15_ONE,len),(opus_val32)(len+factor*K)); 79 theta = HALF16(MULT16_16_Q15(gain,gain)); 110 /** Takes the pitch vector and the decoded residual vector, computes the gain 113 int N, opus_val32 Ryy, opus_val16 gain) 126 g = MULT16_16_P15(celt_rsqrt_norm(t),gain); 156 , opus_val16 gain 311 normalise_residual(iy, X, N, yy, gain); 323 ec_dec *dec, opus_val16 gain) [all...] |
/external/chromium_org/third_party/opus/src/silk/float/ |
SigProc_FLP.h | 51 /* compute inverse of LPC prediction gain, and */ 54 silk_float silk_LPC_inverse_pred_gain_FLP( /* O return inverse prediction gain, energy domain */ 111 const silk_float minInvGain, /* I minimum inverse prediction gain */ 120 silk_float gain, 128 silk_float gain,
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