/frameworks/av/media/libstagefright/codecs/amrnb/dec/src/ |
H A D | 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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H A D | 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[ 753 Word16 gain; local [all...] |
/frameworks/av/media/libstagefright/codecs/amrwbenc/src/ |
H A D | 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 * 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 */ 63 gain [all...] |
H A D | updt_tar.c | 31 Word16 gain, /* (i) Q14 : adaptive codebook gain */ 41 L_tmp2 = L_mult(y[i], gain); 27 Updt_tar( Word16 * x, Word16 * x2, Word16 * y, Word16 gain, Word16 L ) argument
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H A D | 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_MI [all...] |
H A D | p_med_ol.c | 47 Word16 *gain = &(st->ol_gain); /* normalize correlation of hp_wsp for the lag */ local 120 /* gain = R0/ sqrt(R1*R2) */ 147 *gain = vo_round(L_shl(R0, exp_R0));
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H A D | 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 */
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/frameworks/base/media/java/android/media/ |
H A D | AudioDevicePortConfig.java | 30 int format, AudioGainConfig gain) { 31 super((AudioPort)devicePort, samplingRate, channelMask, format, gain); 36 config.gain()); 29 AudioDevicePortConfig(AudioDevicePort devicePort, int samplingRate, int channelMask, int format, AudioGainConfig gain) argument
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H A D | AudioMixPortConfig.java | 30 AudioGainConfig gain) { 31 super((AudioPort)mixPort, samplingRate, channelMask, format, gain); 29 AudioMixPortConfig(AudioMixPort mixPort, int samplingRate, int channelMask, int format, AudioGainConfig gain) argument
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H A D | AudioGainConfig.java | 20 * The AudioGainConfig is used by APIs setting or getting values on a given gain 21 * controller. It contains a valid configuration (value, channels...) for a gain controller 35 AudioGainConfig(int index, AudioGain gain, int mode, int channelMask, argument 38 mGain = gain; 46 * get the index of the parent gain. 62 * Indicates for which channels the gain is set.
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H A D | AudioMixPort.java | 46 AudioGainConfig gain) { 47 return new AudioMixPortConfig(this, samplingRate, channelMask, format, gain); 45 buildConfig(int samplingRate, int channelMask, int format, AudioGainConfig gain) argument
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H A D | AudioPortConfig.java | 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 48 AudioPortConfig(AudioPort port, int samplingRate, int channelMask, int format, AudioGainConfig gain) argument
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H A D | AudioDevicePort.java | 77 AudioGainConfig gain) { 78 return new AudioDevicePortConfig(this, samplingRate, channelMask, format, gain); 76 buildConfig(int samplingRate, int channelMask, int format, AudioGainConfig gain) argument
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H A D | AudioPort.java | 27 * - gain: a port can be associated with one or more gain controllers (see AudioGain). 156 * Get the list of gain descriptors 157 * Empty array if this port does not have gain control 164 * Get the gain descriptor at a given index 166 AudioGain gain(int index) { method in class:AudioPort 180 * @param gain The desired gain. null if no gain changed requested. 183 AudioGainConfig gain) { 182 buildConfig(int samplingRate, int channelMask, int format, AudioGainConfig gain) argument [all...] |
/frameworks/av/media/libstagefright/codecs/amrnb/enc/src/ |
H A D | 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 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); 197 gain = shl (shr (gain, 236 Word16 xy, yy, exp_xy, exp_yy, gain; local [all...] |
H A D | q_gain_c.cpp | 64 Scalar quantization of the innovative codebook gain. 126 exp_gcode0 -- Word16 -- predicted CB gain (exponent), Q0 127 frac_gcode0 -- Word16 -- predicted CB gain (fraction), Q15 128 gain -- Pointer to Word16 -- quantized fixed codebook gain, Q1 131 gain -- Pointer to Word16 -- quantized fixed codebook gain, Q1 152 Scalar quantization of the innovative codebook gain. 193 Word16 exp_gcode0, /* i : predicted CB gain (exponent), Q0 */ 194 Word16 frac_gcode0, /* i : predicted CB gain (fractio 191 q_gain_code( enum Mode mode, Word16 exp_gcode0, Word16 frac_gcode0, Word16 *gain, Word16 *qua_ener_MR122, Word16 *qua_ener, Flag *pOverflow ) argument [all...] |
H A D | q_gain_p.cpp | 115 gp_limit -- Word16 -- pitch gain limit 116 gain -- Pointer to Word16 -- Pitch gain (unquant/quant), Q14 119 gain -- Pointer to Word16 -- Pitch gain (unquant/quant), Q14 121 gain_cand -- Array of type Word16 -- pitch gain candidates (3), 124 gain_cind -- Array of type Word16 -- pitch gain cand. indices (3), 181 Word16 gp_limit, /* i : pitch gain limit */ 182 Word16 *gain, /* i/o: Pitch gain (unquan 179 q_gain_pitch( enum Mode mode, Word16 gp_limit, Word16 *gain, Word16 gain_cand[], Word16 gain_cind[], Flag *pOverflow ) argument [all...] |
H A D | 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) 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/yy 254 gain 313 Word16 gain; local [all...] |
/frameworks/av/media/libeffects/testlibs/ |
H A D | AudioEqualizer.h | 30 // shelf, where each band has frequency and gain controls, and the peaking 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. 111 // Gets gain of a certain band. This is always the last value set (or 113 // band The band to get the gain for. 149 // For the peaking filters, they are the gain[dB]/2 points.
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/frameworks/base/media/mca/filterfw/native/core/ |
H A D | statistics.h | 50 explicit RCFilter(float gain) argument 51 : gain_(gain), n_(0), value_(0.0f) {}
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/frameworks/av/media/libstagefright/codecs/aacenc/src/ |
H A D | quantize.c | 53 * quaSpectrum = mdctSpectrum^3/4*2^(-(3/16)*gain) 56 static Word16 quantizeSingleLine(const Word16 gain, const Word32 absSpectrum) argument 67 /* calculate the final fractional exponent times 16 (was 3*(4*e + gain) + (INT_BITS-1)*16) */ 68 minusFinalExp = (e << 2) + gain; 99 * quaSpectrum = mdctSpectrum^3/4*2^(-(3/16)*gain) 100 * input: global gain, number of lines to process, spectral data 104 static void quantizeLines(const Word16 gain, argument 110 Word32 m = gain&3; 111 Word32 g = (gain >> 2) + 4; 114 /* gain 228 iquantizeLines(const Word16 gain, const Word16 noOfLines, const Word16 *quantSpectrum, Word32 *mdctSpectrum) argument 327 calcSfbDist(const Word32 *spec, Word16 sfbWidth, Word16 gain) argument [all...] |
/frameworks/av/media/libstagefright/codecs/amrwb/src/ |
H A D | dtx_decoder_amr_wb.cpp | 208 int16 gain; local 375 /* L_log_en_int corresponds to log2(E)+2 in Q24, i.e log2(gain)+1 in Q25 */ 385 /* Subtract 2 from L_log_en_int in Q9, i.e divide the gain by 2 (energy by 4) */ 403 /* gain = level / sqrt(ener) * sqrt(L_FRAME) */ 410 gain = extract_h(ener32); 412 gain = mult_int16(level, gain); /* gain in Q15 */ 421 tmp = mult_int16(exc2[i], gain); /* Q0 * Q15 */
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/frameworks/av/services/audioflinger/ |
H A D | AudioResamplerFirProcess.h | 74 * Unrolling the loops achieves about 2x gain. 76 * values is an additional 10-20% gain. 92 inline void volume(TO*& out, TO gain) { argument 93 *out++ = volumeAdjust(value, gain); 94 Accumulator<CHANNELS-1, TO>::volume(out, gain); 108 inline void volume(TO*& out __unused, TO gain __unused) { 293 * A typical value for volume is 0x1000 to align to a unity gain output of 20.12. 350 * A typical value for volume is 0x1000 to align to a unity gain output of 20.12.
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/frameworks/av/services/audiopolicy/common/managerdefinitions/src/ |
H A D | ConfigParsingUtils.cpp | 38 sp<AudioGain> gain = new AudioGain(index, audioPort.useInputChannelMask()); local 42 gain->setMode(GainModeConverter::maskFromString(node->value)); 47 gain->setChannelMask(mask); 51 gain->setChannelMask(mask); 55 gain->setMinValueInMb(atoi(node->value)); 57 gain->setMaxValueInMb(atoi(node->value)); 59 gain->setDefaultValueInMb(atoi(node->value)); 61 gain->setStepValueInMb(atoi(node->value)); 63 gain->setMinRampInMs(atoi(node->value)); 65 gain [all...] |
/frameworks/wilhelm/src/itf/ |
H A D | 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 447 float gain; local [all...] |