musicdsp.org source code archive - WSInf

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Cascaded resonant lp/hp filter (click this to go back to the index) Type : lp+hp References : Posted by tobybear[AT]web[DOT]de Notes : // Cascaded resonant lowpass/hipass combi-filter // The original source for this filter is from Paul Kellet from // the archive. This is a cascaded version in Delphi where the // output of the lowpass is fed into the highpass filter. // Cutoff frequencies are in the range of 0
Cool Sounding Lowpass With Decibel Measured Resonance (click this to go back to the index) Type : LP 2-pole resonant tweaked butterworth References : Posted by daniel_jacob_werner [AT] yaho [DOT] com [DOT] au Notes : This algorithm is a modified version of the tweaked butterworth lowpass filter by Patrice Tarrabia posted on musicdsp.org's archives. It calculates the coefficients for a second order IIR filter. The resonance is specified in decibels above the DC gain. It can be made suitable to use as a SoundFont 2.0 filter by scaling the output so the overall gain matches the specification (i.e. if resonance is 6dB then you should scale the output by -3dB). Note that you can replace the sqrt(2) values in the standard butterworth highpass algorithm with my "q =" line of code to get a highpass also. How it works: normally q is the constant sqrt(2), and this value controls resonance. At sqrt(2) resonance is 0dB, smaller values increase resonance. By multiplying sqrt(2) by a power ratio we can specify the resonant gain at the cutoff frequency. The resonance power ratio is calculated with a standard formula to convert between decibels and power ratios (the powf statement...). Good Luck, Daniel Werner http://experimentalscene.com/ Code : float c, csq, resonance, q, a0, a1, a2, b1, b2; c = 1.0f / (tanf(pi * (cutoff / samplerate))); csq = c * c; resonance = powf(10.0f, -(resonancedB * 0.1f)); q = sqrt(2.0f) * resonance; a0 = 1.0f / (1.0f + (q * c) + (csq)); a1 = 2.0f * a0; a2 = a0; b1 = (2.0f * a0) * (1.0f - csq); b2 = a0 * (1.0f - (q * c) + csq); Comments from : acid_mutant[aat]yahoo[doot]com comment : For some reason when I tested this algorithm, even though the frequency response looked OK in my graphs - i.e. it should resonate the output didn't seem to be very resonant - it could be a phase issue, I'll keep checking. (BTW: I use an impulse, then FFT, then display the power bands returned) from : dans@dans.com comment : shouldn't it be resonance = powf(10.0f, -(resonancedB * 0.05f)); instead of resonance = powf(10.0f, -(resonancedB * 0.1f)); to get correct dB gain? ... since gain = 10^(dB/20) ...
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musicdsp.org source code archive Th
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VST SDK GUI Switch without 16-Point
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(Allmost) Ready-to-use oscillators
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Alias-free waveform generation with
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AM Formantic Synthesis (click this
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Another cheap sinusoidal LFO (click
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property Rate:Single read FRate wri
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procedure SetSampleRate(const Value
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antialiased square generator (click
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Audiable alias free waveform gen us
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Bandlimited sawtooth synthesis (cli
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} Nbeta = b->N * beta; cosbeta = co
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} lastnumpartials=partials; a=int(2
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Bandlimited waveform generation wit
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Approximation through processing of
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Bandlimited waveforms... (click thi
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I've already nearly done all the di
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C++ gaussian noise generation (clic
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Cubic polynomial envelopes (click t
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Drift generator (click this to go b
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Easy noise generation (click this t
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Fast Exponential Envelope Generator
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} coeff = 1.0f + (log(levelEnd) - l
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inc our 32bit integer LFO pos & let
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egin fSpeed:=v; iSpeed:=Round($1000
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Fast sine wave calculation (click t
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Fast Whitenoise Generator (click th
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Gaussian White Noise (click this to
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Inverted parabolic envelope (click
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if (nargin < 2 ) thresh = -100; end
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Phase modulation Vs. Frequency modu
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Phase modulation Vs. Frequency modu
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Pulsewidth modulation (click this t
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RBJ Wavetable 101 (click this to go
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SawSin (click this to go back to th
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Square Waves (click this to go back
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Waveform generator using MinBLEPS (
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Weird synthesis (click this to go b
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} } // Step 4 : rising edge detecto
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Coefficients for Daubechies wavelet
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-3.22448695846383746484797550621349
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9.323261308672633862226517802548514
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2.135815619103406884039052814341926
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-7.70690988123119623288037272295551
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1.509692082823910867903367712096001
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-8.36549047125880079934928979439790
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1.089584350416766882738651833752634
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4.784787462793710621468610706120519
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-5.65781324505881838042401697351671
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-4.17514164854039779729632506577571
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Envelope detector (click this to go
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Envelope follower with different at
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from : (1< - ??????????????????????
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while(1){ if(i!=0) { while((i&1)==0
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l=size-1; m=size>>1; for (i=0,j=0;
Page 111 and 112: //Source: see the routines it calls
Page 113 and 114: long dl, dl2, i, j; double d0,d1,d2
Page 115 and 116: i3+=n8; i4+=n8; t1=(data[i3]+data[i
Page 117 and 118: i4=i3+n4; i5=i+n4-j+1; i6=i5+n4; i7
Page 119 and 120: free(data2); }
Page 121 and 122: Frequency response from biquad coef
Page 123 and 124: c = ((c * i) + Math.abs(a[i])) / (i
Page 125 and 126: LPC analysis (autocorrelation + Lev
Page 127 and 128: egin for j:=0 to P do begin A[0]:=0
Page 129 and 130: } (b0*b0 + b1*b1 + b2*b2 + b3*b3 +
Page 131 and 132: { return (MulVect::calc(v, 0)); } }
Page 133 and 134: Measuring interpollation noise (cli
Page 135 and 136: Simple peak follower (click this to
Page 137 and 138: else tone[i] = 2*cos(A)*cos(A) - co
Page 139 and 140: Tone detection with Goertzel (x86 A
Page 141 and 142: 1-RC and C filter (click this to go
Page 143 and 144: 1st and 2nd order pink noise filter
Page 145 and 146: double f2p3; // Sample history buff
Page 147 and 148: 303 type filter with saturation (cl
Page 149 and 150: All-Pass Filters, a good explanatio
Page 151 and 152: Another 4-pole lowpass... (click th
Page 153 and 154: fCoeffs[0]:=p; fCoeffs[1]:=4.0*p*s;
Page 155 and 156: Biquad C code (click this to go bac
Page 157 and 158: a1 = 2 * ((A - 1) - (A + 1) * cs);
Page 159 and 160: } // hishelf if(type==8) { b0=float
Page 161: C-Weighed Filter (click this to go
Page 165 and 166: Delphi Class implementation of the
Page 167 and 168: 0:=alpha; b1:=0.0; b2:=-alpha; a0:=
Page 169 and 170: Direct form II (click this to go ba
Page 171 and 172: Formant filter (click this to go ba
Page 173 and 174: frequency warped FIR lattice (click
Page 175 and 176: Hiqh quality /2 decimators (click t
Page 177 and 178: Karlsen (click this to go back to t
Page 179 and 180: end; procedure TKarlsen.SetQ(v:Sing
Page 181 and 182: float b_cut = ((fvar1 * fvar1) + ((
Page 183 and 184: Lowpass filter for parameter edge f
Page 185 and 186: } public float Process(float input)
Page 188 and 189: LPF 24dB/Oct (click this to go back
Page 190 and 191: float v2; v2 = 40000; // twice the
Page 192 and 193: ..but it won't be much faster than
Page 194 and 195: Moog VCF (click this to go back to
Page 196 and 197: Moog VCF, variation 2 (click this t
Page 198 and 199: Notch filter (click this to go back
Page 200 and 201: One pole, one zero LP/HP (click thi
Page 202 and 203: Peak/Notch filter (click this to go
Page 204 and 205: Pink noise filter (click this to go
Page 206 and 207: Polyphase Filters (click this to go
Page 208 and 209: {0.2659685265210946 ,0.665104153263
Page 210 and 211: delete filter_b; }; double CHalfBan
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------------ AllPass Filter Cascade
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end; end.
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RBJ Audio-EQ-Cookbook (click this t
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1 = -(1 + cos(w0)) b2 = (1 + cos(w0
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RBJ Audio-EQ-Cookbook (click this t
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Remez Remez (Parks/McClellan) (clic
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Resonant IIR lowpass (12dB/oct) (cl
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{ unsigned int i; float *hist1_ptr,
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* For n=4, or two second order sect
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{ } /* Calculate a1 and a2 and over
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Reverb Filter Generator (click this
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State Variable Filter (Chamberlin v
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State Variable Filter (Double Sampl
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} lowpass = output; highpass = inpu
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} if (size == 1) { out[0] = in1[0]
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Time domain convolution with O(n^lo
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Various Biquad filters (click this
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void setfilter_shelvelowpass(f,freq
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Windowed Sinc FIR Generator (click
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do W[i]:=W[i]*(0.35875-0.48829*cos(
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int i; double *h1 = new double[N];
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Zoelzer biquad filters (click this
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So tan(x) would be x - x^3/6 + x^5/
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2 Wave shaping things (click this t
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#define infile "a.raw" //input file
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Band Limited PWM Generator (click t
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Bit quantization/reduction effect (
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Compressor (click this to go back t
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Decimator (click this to go back to
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Delay time calculation for reverber
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dynamic convolution (click this to
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Early echo's with image-mirror tech
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for (y=-ceil(dist_max/(2*breite));y
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} } j=0; for(i=0;i
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Raummodell: H - Hoererposition L -
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} } } angle=atan(y_pos/x_pos); if (
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ECE320 project: Reverberation w/ pa
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move#ser_data,r6; incoming data buf
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movex:(r0)+,a; er delay asr#20,a,a
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fold back distortion (click this to
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Lo-Fi Crusher (click this to go bac
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Most simple static delay (click thi
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Parallel combs delay calculation (c
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inherited; fA1:=0; fZM1:=0; end; de
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implementation { TPhaser } function
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end; a[1]:=b-fY[5]; b:=a[1]*fA1; fY
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Polynominal Waveshaper (click this
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Reverberation techniques (click thi
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smsPitchScale Source Code (click th
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Stereo Enhancer (click this to go b
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Time compression-expansion using st
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transistor differential amplifier s
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WaveShaper (click this to go back t
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Waveshaper (click this to go back t
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Waveshaper :: Gloubi-boulga (click
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VST SDK GUI Switch without (click t
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private final int POINTS = 1 15; a
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3rd order Spline interpollation (cl
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5-point spline interpollation (clic
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Antialiased Lines (click this to go
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Automatic PDC system (click this to
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} return ret; }
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Block/Loop Benchmarking (click this
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END; {$ELSE} BEGIN RESULT:=(Value S
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} } throw new IllegalArgumentExcept
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Center separation in a stereo mixdo
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Center separation in a stereo mixdo
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Clipping without branching (click t
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Conversions on a PowerPC (click thi
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Cubic interpollation (click this to
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Denormal DOUBLE variables, macro (c
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Use this eq. to calculate the appro
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Denormalization preventer (click th
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Anyway this version gives more expl
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Dithering (click this to go back to
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Envelope Follower (click this to go
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fast abs/neg/sign for 32bit floats
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Fast cube root, square root, and re
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float z; _asm { rsqrtss xmm0, x rcp
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1 - 1.5 (using fastexp6, the maximu
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Fast log2 (click this to go back to
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fast power and root estimates for 3
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Float to int (more intel asm) (clic
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Float-to-int, coverting an array of
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Gaussian random numbers (click this
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Pentium 4 and above, but not using
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HTH DSP from : antiprosynthesis@hot
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Matlab Time Domain Impulse Response
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MATLAB-Tools for SNDAN (click this
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end; // converts from MIDI note to
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Motorola 56300 Disassembler (click
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} __inline int round(float f) { int
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Nonblocking multiprocessor/multithr
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pow(x,4) approximation (click this
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Real basic DSP with Matlab (+ GUI)
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Really fast x86 floating point sin/
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from : Christian@savioursofsoul.de
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pfmul mm1, mm0 //mm1=a movq mm2, mm
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esampling (click this to go back to
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depending on whether you're looking
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Sin, Cos, Tan approximation (click
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fResult += 0.180141f; fResult *= fV
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musicdsp.org source code archive - WSInf

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