/frameworks/av/media/libstagefright/codecs/amrnb/dec/src/ |
d_gain_p.cpp | 64 Word16 gain -- (Q14) 76 Purpose : Decodes the pitch gain using the received index. 176 Word16 d_gain_pitch( /* return value: gain (Q14) */ 181 Word16 gain; local 183 gain = qua_gain_pitch[index]; 188 gain &= 0xFFFC; 191 return gain;
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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...] |
/external/webrtc/webrtc/modules/audio_coding/codecs/ilbc/ |
gain_dequant.c | 23 * decoder for quantized gains in the gain-shape coding of 28 /* (o) quantized gain value (Q14) */ 30 int16_t maxIn, /* (i) maximum of unquantized gain (Q14) */ 34 const int16_t *gain; local 42 gain = WebRtcIlbcfix_kGain[stage]; 44 return (int16_t)((scale * gain[index] + 8192) >> 14);
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cb_construct.c | 30 int16_t *gain_index, /* (i) Gain quantization indices */ 36 int16_t gain[CB_NSTAGES]; local 44 /* gain de-quantization */ 46 gain[0] = WebRtcIlbcfix_GainDequant(gain_index[0], 16384, 0); 47 gain[1] = WebRtcIlbcfix_GainDequant(gain_index[1], gain[0], 1); 48 gain[2] = WebRtcIlbcfix_GainDequant(gain_index[2], gain[1], 2); 57 gainPtr = &gain[0];
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/frameworks/av/media/libstagefright/codecs/amrwbenc/src/ |
g_pitch.c | 20 * Description:Compute the gain of pitch. Result in Q12 * 21 * if(gain < 0) gain = 0 * 22 * if(gain > 1.2) gain = 1.2 * 29 Word16 G_pitch( /* (o) Q14 : Gain of pitch lag saturated to 1.2 */ 32 Word16 g_coeff[], /* : Correlations need for gain quantization. */ 37 Word16 xy, yy, exp_xy, exp_yy, gain; local 56 /* If (xy < 0) gain = 0 */ 60 /* compute gain = xy/yy * [all...] |
gpclip.c | 20 * Description:To avoid unstable synthesis on frame erasure, the gain * 21 * need to be limited(gain pitch < 1.0) when the following * 38 Word16 mem[] /* (o) : memory of gain of pitch clipping algorithm */ 47 Word16 mem[] /* (i/o) : memory of gain of pitch clipping algorithm */ 60 Word16 mem[] /* (i/o) : memory of gain of pitch clipping algorithm */ 90 Word16 gain_pit, /* (i) Q14 : gain of quantized pitch */ 91 Word16 mem[] /* (i/o) : memory of gain of pitch clipping algorithm */ 94 Word16 gain; local 99 gain = extract_h(L_tmp); 101 if(gain < GAIN_PIT_MIN [all...] |
/external/libopus/silk/fixed/ |
process_gains_FIX.c | 44 opus_int32 s_Q16, InvMaxSqrVal_Q16, gain, gain_squared, ResNrg, ResNrgPart, quant_offset_Q10; local 46 /* Gain reduction when LTP coding gain is high */ 73 gain = psEncCtrl->Gains_Q16[ k ]; 74 gain_squared = silk_ADD_SAT32( ResNrgPart, silk_SMMUL( gain, gain ) ); 77 gain_squared = silk_SMLAWW( silk_LSHIFT( ResNrgPart, 16 ), gain, gain ); 79 gain = silk_SQRT_APPROX( gain_squared ); /* Q8 */ 80 gain = silk_min( gain, silk_int32_MAX >> 8 ) [all...] |
/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) * [all...] |
/external/aac/libAACdec/src/ |
ldfiltbank.cpp | 175 FIXP_DBL gain = (FIXP_DBL)0; local 190 imdct_gain(&gain, &scale, N); 194 if (gain != (FIXP_DBL)0) { 195 scaleValuesWithFactor(mdctData, gain, N, scale);
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/external/aac/libSBRenc/src/ |
resampler.h | 110 FIXP_DBL gain; /*! overall gain factor */ member in struct:__anon7003
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/external/skia/bench/ |
MatrixConvolutionBench.cpp | 24 SkScalar gain = 0.3f, bias = SkIntToScalar(100); local 26 fFilter = SkMatrixConvolutionImageFilter::Create(kernelSize, kernel, gain, bias, kernelOffset, tileMode, convolveAlpha);
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/external/sonivox/arm-wt-22k/lib_src/ |
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 [all...] |
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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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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/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...] |
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 [all...] |
/frameworks/base/media/java/android/media/ |
AudioDevicePortConfig.java | 30 int format, AudioGainConfig gain) { 31 super((AudioPort)devicePort, samplingRate, channelMask, format, gain); 36 config.gain());
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AudioPortConfig.java | 45 static final int GAIN = 0x8; 49 AudioGainConfig gain) { 54 mGain = gain; 87 * The gain configuration if this port supports gain control, null otherwise 90 public AudioGainConfig gain() { method in class:AudioPortConfig
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/external/skia/gm/ |
matrixconvolution.cpp | 59 SkScalar gain = 0.3f, bias = SkIntToScalar(100); local 64 gain,
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imagefiltersgraph.cpp | 200 SkScalar gain = 1.0f, bias = SkIntToScalar(0); variable 207 gain,
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/external/webrtc/webrtc/modules/audio_processing/agc/ |
agc_unittest.cc | 33 // The tested values depend on this assumed gain. 52 // A gain of <= -100 will zero out the signal. 64 float gain = Db2Linear(gain_db); local 66 gain = 0; 73 ApplyGainLinear(gain, &frame);
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/external/opencv3/modules/video/include/opencv2/video/ |
tracking.hpp | 373 CV_PROP_RW Mat gain; //!< Kalman gain matrix (K(k)): K(k)=P'(k)*Ht*inv(H*P'(k)*Ht+R) member in class:cv::KalmanFilter
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/external/skia/src/gpu/effects/ |
GrMatrixConvolutionEffect.h | 25 SkScalar gain, 30 return new GrMatrixConvolutionEffect(texture, bounds, kernelSize, kernel, gain, bias, 37 SkScalar gain, 49 float gain() const { return fGain; } function in class:GrMatrixConvolutionEffect 61 SkScalar gain,
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/external/webrtc/webrtc/modules/audio_processing/agc/legacy/ |
digital_agc.h | 43 int32_t gain; member in struct:__anon27858
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/frameworks/av/media/libeffects/testlibs/ |
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...] |