musicdsp.org source code archive - WSInf

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{ if(fabs(*(data+b))>max) //normalise to richest waveform (0) max=(float)fabs(*(data+b)); } max*=0.5; for(b=0;b
Bandlimited waveform generation with hard sync (click this to go back to the index) References : Posted by Emanuel Landeholm Linked file : http://www.algonet.se/~e-san/hardsync.tar.gz Comments from : emanuel.landeholm@telia.com comment : The tar-gz <strong>archive</strong> has moved to http://web.telia.com/~u84605054/hardsync.tar.gz
Page 1 and 2: musicdsp.org source code archive Th
Page 3 and 4: VST SDK GUI Switch without 16-Point
Page 5 and 6: (Allmost) Ready-to-use oscillators
Page 7 and 8: Alias-free waveform generation with
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: } lastnumpartials=partials; a=int(2
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
Page 35 and 36: C++ gaussian noise generation (clic
Page 37 and 38: Cubic polynomial envelopes (click t
Page 39 and 40: Drift generator (click this to go b
Page 41 and 42: Easy noise generation (click this t
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
Page 55 and 56: Gaussian White Noise (click this to
Page 57 and 58: Inverted parabolic envelope (click
Page 59 and 60: if (nargin < 2 ) thresh = -100; end
Page 61 and 62: Phase modulation Vs. Frequency modu
Page 63 and 64: Phase modulation Vs. Frequency modu
Page 65 and 66: Pulsewidth modulation (click this t
Page 67 and 68: RBJ Wavetable 101 (click this to go
Page 69 and 70: SawSin (click this to go back to th
Page 71 and 72: Square Waves (click this to go back
Page 73 and 74: Waveform generator using MinBLEPS (
Page 75 and 76: Weird synthesis (click this to go b
Page 77 and 78:
} } // 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
Page 257 and 258:
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
Page 269 and 270:
Decimator (click this to go back to
Page 271 and 272:
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
Page 313 and 314:
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
Page 323 and 324:
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
Page 382 and 383:
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
Page 417:
fResult += 0.180141f; fResult *= fV
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musicdsp.org source code archive - WSInf

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