| /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/chromium_org/third_party/WebKit/Source/platform/audio/ |
| 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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| Cone.cpp | 45 double ConeEffect::gain(FloatPoint3D sourcePosition, FloatPoint3D sourceOrientation, FloatPoint3D listenerPosition) function in class:WebCore::ConeEffect
48 return 1.0; // no cone specified - unity gain
65 double gain = 1.0; local
69 gain = 1.0;
72 gain = m_outerGain;
77 gain = (1.0 - x) + m_outerGain * x;
80 return gain;
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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/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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| AudioBuffer.h | 62 // Scalar gain
63 double gain() const { return m_gain; } function in class:WebCore::AudioBuffer
64 void setGain(double gain) { m_gain = gain; }
78 double m_gain; // scalar gain
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| BiquadDSPKernel.cpp | 48 double gain; local
54 gain = biquadProcessor()->parameter3()->finalValue();
59 gain = biquadProcessor()->parameter3()->smoothedValue();
64 gain = biquadProcessor()->parameter3()->value();
91 m_biquad.setLowShelfParams(normalizedFrequency, gain);
95 m_biquad.setHighShelfParams(normalizedFrequency, gain);
99 m_biquad.setPeakingParams(normalizedFrequency, value2, gain);
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| BiquadFilterNode.h | 60 AudioParam* gain() { return biquadProcessor()->parameter3(); } function in class:WebCore::BiquadFilterNode
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| /external/chromium_org/third_party/opus/src/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/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/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:__anon2667
|
| /external/skia/bench/ |
| MatrixConvolutionBench.cpp | 24 SkScalar gain = 0.3f, bias = SkIntToScalar(100); local
26 fFilter = new SkMatrixConvolutionImageFilter(kernelSize, kernel, gain, bias, target, tileMode, convolveAlpha);
|
| /external/sonivox/arm-fm-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...] |
| /external/sonivox/arm-hybrid-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
[all...] |
| /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
|
| 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/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...] |
| /external/chromium_org/content/browser/speech/endpointer/ |
| endpointer_unittest.cc | 36 float gain = 0.0; local
42 gain = 2000.0;
44 gain = 1.0;
50 samples[i] = static_cast<int16>(gain * randNum);
|
