musicdsp.org source code archive - WSInf

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_fbuf3 = ((b_fbuf3 - (b_fbuf3 * b_cutsc)) + (b_in * b_cutsc)); b_in = b_fbuf3; b_fbuf4 = ((b_fbuf4 - (b_fbuf4 * b_cutsc)) + (b_in * b_cutsc)); b_in = b_fbuf4; Soundwise, it's somewhere between a transistor ladder, and a diode ladder.. Enjoy! Ove Karlsen. PS: I prefer IRL communication these days, so if you need to reach me, please dial my cellphone, +047 928 50 803.
Lowpass filter for parameter edge filtering (click this to go back to the index) References : Olli Niemitalo Linked file : filter001.gif Notes : use this filter to smooth sudden parameter changes (see linkfile!) Code : /* - Three one-poles combined in parallel * - Output stays within input limits * - 18 dB/oct (approx) frequency response rolloff * - Quite fast, 2x3 parallel multiplications/sample, no internal buffers * - Time-scalable, allowing use with different samplerates * - Impulse and edge responses have continuous differential * - Requires high internal numerical precision */ { /* Parameters */ // Number of samples from start of edge to halfway to new value const double scale = 100; // 0 < Smoothness < 1. High is better, but may cause precision problems const double smoothness = 0.999; } /* Precalc variables */ double a = 1.0-(2.4/scale); // Could also be set directly double b = smoothness; // -"double acoef = a; double bcoef = a*b; double ccoef = a*b*b; double mastergain = 1.0 / (-1.0/(log(a)+2.0*log(b))+2.0/ (log(a)+log(b))-1.0/log(a)); double again = mastergain; double bgain = mastergain * (log(a*b*b)*(log(a)-log(a*b)) / ((log(a*b*b)-log(a*b))*log(a*b)) - log(a)/log(a*b)); double cgain = mastergain * (-(log(a)-log(a*b)) / (log(a*b*b)-log(a*b))); /* Runtime variables */ long streamofs; double areg = 0; double breg = 0; double creg = 0; /* Main loop */ for (streamofs = 0; streamofs < streamsize; streamofs++) { /* Update filters */ areg = acoef * areg + fromstream [streamofs]; breg = bcoef * breg + fromstream [streamofs]; creg = ccoef * creg + fromstream [streamofs]; } /* Combine filters in parallel */ long temp = again * areg + bgain * breg + cgain * creg; /* Check clipping */ if (temp > 32767) { temp = 32767; } else if (temp < -32768) { temp = -32768; } /* Store new value */ tostream [streamofs] = temp;
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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;
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//Source: see the routines it calls
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long dl, dl2, i, j; double d0,d1,d2
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i3+=n8; i4+=n8; t1=(data[i3]+data[i
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i4=i3+n4; i5=i+n4-j+1; i6=i5+n4; i7
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free(data2); }
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Frequency response from biquad coef
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c = ((c * i) + Math.abs(a[i])) / (i
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LPC analysis (autocorrelation + Lev
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egin for j:=0 to P do begin A[0]:=0
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} (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 and 162: C-Weighed Filter (click this to go
Page 163 and 164: Cool Sounding Lowpass With Decibel
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: float b_cut = ((fvar1 * fvar1) + ((
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
Page 212 and 213: ------------ AllPass Filter Cascade
Page 214 and 215: end; end.
Page 216 and 217: RBJ Audio-EQ-Cookbook (click this t
Page 218 and 219: 1 = -(1 + cos(w0)) b2 = (1 + cos(w0
Page 220 and 221: RBJ Audio-EQ-Cookbook (click this t
Page 222 and 223: Remez Remez (Parks/McClellan) (clic
Page 224 and 225: Resonant IIR lowpass (12dB/oct) (cl
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Page 230 and 231: { } /* 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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