musicdsp.org source code archive - WSInf

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} */ return ((dtable1[i]*alpha + dtable2[i])*alpha + dtable3[i])*alpha+table[i];
Alias-free waveform generation with analog filtering (click this to go back to the index) Type : waveform generation References : Posted by Magnus Jonsson Linked file : synthesis001.txt (this linked file is included below) Notes : (see linkfile) Linked files alias-free waveform generation with analog filtering ---------------------------------------------------- Ok, here is how I did it. I'm not 100% sure that everything is correct. I'll demonstrate it on a square wave instead, although i havent tried it. The impulse response of an analog pole is: r(t) = exp(p*x) if t >= 0, 0 if t < 0 notice that if we know r(t) we can get r(t+1) = r(t)*exp(p). You all know what the waveform looks like. It's a constant -1 followed by constant 1 followed by ..... We need to convolve the impulse response with the waveform and sample it at discrete intervals. What if we assume that we already know the result of the last sample? Then we can "move" the previous result backwards by multiplying it with exp(p) since r(t+1) = r(t)*exp(p) Now the problem is reduced to integrate only between the last sample and the current sample, and add that to the result. some pseudo-<strong>code</strong>: while forever { result *= exp(pole); phase += freq; result += integrate(waveform(phase-freq*t), exp(t*pole), t=0..1); } integrate(square(phase-freq*t), exp(t*pole), t=0..1) The square is constant except for when it changes sign. Let's find out what you get if you integrate a constant multiplied with exp(t*pole) =) integrate(k*exp(t*pole)) = k*exp(t*pole)/pole k = square(phase-freq*t) and with t from 0 to 1, that becomes k*exp(pole)/pole-k/pole = (exp(pole)-1)*k/pole the only problem left to solve now is the jumps from +1 to -1 and vice versa. you first calculate (exp(pole)-1)*k/pole like you'd normally do then you detect if phase goes beyond 0.5 or 1. If so, find out exactly where between the samples this change takes place. subtract integrate(-2*exp(t*pole), t=0..place) from the result to undo the error Since I am terribly bad at explaining things, this is probably just a mess to you :) Here's the (unoptimised) <strong>code</strong> to do this with a saw: note that sum and pole are complex. float getsample() { sum *= exp(pole); phase += freq; sum += (exp(pole)*((phase-freq)*pole+freq)-(phase*pole+freq))/pole; if (phase >= 0.5) {
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Page 3 and 4: VST SDK GUI Switch without 16-Point
Page 5: (Allmost) Ready-to-use oscillators
Page 9 and 10: AM Formantic Synthesis (click this
Page 11 and 12: Another cheap sinusoidal LFO (click
Page 13 and 14: property Rate:Single read FRate wri
Page 15 and 16: procedure SetSampleRate(const Value
Page 17 and 18: antialiased square generator (click
Page 19 and 20: Audiable alias free waveform gen us
Page 21 and 22: Bandlimited sawtooth synthesis (cli
Page 23 and 24: } Nbeta = b->N * beta; cosbeta = co
Page 25 and 26: } lastnumpartials=partials; a=int(2
Page 27 and 28: Bandlimited waveform generation wit
Page 29 and 30: Approximation through processing of
Page 31 and 32: Bandlimited waveforms... (click thi
Page 33 and 34: I've already nearly done all the di
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Page 37 and 38: Cubic polynomial envelopes (click t
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Page 43 and 44: Fast Exponential Envelope Generator
Page 45 and 46: } coeff = 1.0f + (log(levelEnd) - l
Page 47 and 48: inc our 32bit integer LFO pos & let
Page 49 and 50: egin fSpeed:=v; iSpeed:=Round($1000
Page 51 and 52: Fast sine wave calculation (click t
Page 53 and 54: Fast Whitenoise Generator (click th
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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 +
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{ return (MulVect::calc(v, 0)); } }
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Measuring interpollation noise (cli
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Simple peak follower (click this to
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else tone[i] = 2*cos(A)*cos(A) - co
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Tone detection with Goertzel (x86 A
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1-RC and C filter (click this to go
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1st and 2nd order pink noise filter
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double f2p3; // Sample history buff
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303 type filter with saturation (cl
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All-Pass Filters, a good explanatio
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Another 4-pole lowpass... (click th
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fCoeffs[0]:=p; fCoeffs[1]:=4.0*p*s;
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Biquad C code (click this to go bac
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a1 = 2 * ((A - 1) - (A + 1) * cs);
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} // hishelf if(type==8) { b0=float
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C-Weighed Filter (click this to go
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Cool Sounding Lowpass With Decibel
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Delphi Class implementation of the
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0:=alpha; b1:=0.0; b2:=-alpha; a0:=
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Direct form II (click this to go ba
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Formant filter (click this to go ba
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frequency warped FIR lattice (click
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Hiqh quality /2 decimators (click t
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Karlsen (click this to go back to t
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end; procedure TKarlsen.SetQ(v:Sing
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float b_cut = ((fvar1 * fvar1) + ((
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Lowpass filter for parameter edge f
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} public float Process(float input)
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LPF 24dB/Oct (click this to go back
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float v2; v2 = 40000; // twice the
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..but it won't be much faster than
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Moog VCF (click this to go back to
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Moog VCF, variation 2 (click this t
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Notch filter (click this to go back
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One pole, one zero LP/HP (click thi
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Peak/Notch filter (click this to go
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Pink noise filter (click this to go
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Polyphase Filters (click this to go
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{0.2659685265210946 ,0.665104153263
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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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