musicdsp.org source code archive - WSInf

Recommendations

Info

comment : afact is used as an amplitude factor for fm or fb depending on the carrier frequency. The higher frequency the lower afact. It's not completely resolving the problem with aliasing but it is a cheap way that dramatically reduces it.
Bandlimited sawtooth synthesis (click this to go back to the index) Type : DSF BLIT References : Posted by emanuel.landeholm [AT] telia.com Linked file : synthesis002.txt (this linked file is included below) Notes : This is working <strong>code</strong> for synthesizing a bandlimited sawtooth waveform. The algorithm is DSF BLIT + leaky integrator. Includes driver <strong>code</strong>. There are two parameters you may tweak: 1) Desired attenuation at nyquist. A low value yields a duller sawtooth but gets rid of those annoying CLICKS when sweeping the frequency up real high. Must be strictly less than 1.0! 2) Integrator leakiness/cut off. Affects the shape of the waveform to some extent, esp. at the low end. Ideally you would want to set this low, but too low a setting will give you problems with DC. Have fun! /Emanuel Landeholm (see linked file) Comments from : rainbow_stash@hotmail.com comment : there is no need to use a butterworth design for a simple leaky integrator, in this case actually the variable curcps can be used directly in a simple: leak += curcps * (blit - leak); this produces a nearly perfect saw shape in almost all cases Linked files #include #include #include /* Bandlimited synthesis of sawtooth by * leaky integration of a DSF BLIT * * Emanuel Landeholm, March 2002 * emanuel.landeholm@telia.com * * Provided \"as is\". * Free as in Ef Are Ee Ee. */ double pi = 3.1415926535897932384626433832795029L; double twopi = 6.2831853071795864769252867665590058L; /* Leaky integrator/first order lowpass which * shapes the impulse train into a nice * -6dB/octave spectrum * * The cutoff frequency needs to be pretty lowish * or the sawtooth will suffer phase distortion * at the low end. */ typedef struct { double x1, y1; double a, b; } lowpass_t; /* initializes a lowpass, sets cutoff/leakiness */ void init_lowpass(lowpass_t *lp, double cutoff) { double Omega; } lp->x1 = lp->y1 = 0.0; Omega = atan(pi * cutoff); lp->a = -(1.0 - Omega) / (1.0 + Omega); lp->b = (1.0 - lp->b) / 2.0; double update_lowpass(lowpass_t *lp, double x) {
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: Audiable alias free waveform gen us
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
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
Page 79 and 80:
Coefficients for Daubechies wavelet
Page 81 and 82:
-3.22448695846383746484797550621349
Page 83 and 84:
9.323261308672633862226517802548514
Page 85 and 86:
2.135815619103406884039052814341926
Page 87 and 88:
-7.70690988123119623288037272295551
Page 89 and 90:
1.509692082823910867903367712096001
Page 91 and 92:
-8.36549047125880079934928979439790
Page 93 and 94:
1.089584350416766882738651833752634
Page 95 and 96:
4.784787462793710621468610706120519
Page 97 and 98:
-5.65781324505881838042401697351671
Page 99 and 100:
-4.17514164854039779729632506577571
Page 101 and 102:
Envelope detector (click this to go
Page 103 and 104:
Envelope follower with different at
Page 105 and 106:
from : (1< - ??????????????????????
Page 107 and 108:
while(1){ if(i!=0) { while((i&1)==0
Page 109 and 110:
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 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 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
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
Page 226 and 227:
{ unsigned int i; float *hist1_ptr,
Page 228 and 229:
* For n=4, or two second order sect
Page 230 and 231:
{ } /* Calculate a1 and a2 and over
Page 232:
Reverb Filter Generator (click this
Page 235 and 236:
State Variable Filter (Chamberlin v
Page 237 and 238:
State Variable Filter (Double Sampl
Page 239 and 240:
} lowpass = output; highpass = inpu
Page 241 and 242:
} if (size == 1) { out[0] = in1[0]
Page 243 and 244:
Time domain convolution with O(n^lo
Page 245 and 246:
Various Biquad filters (click this
Page 247 and 248:
void setfilter_shelvelowpass(f,freq
Page 249 and 250:
Windowed Sinc FIR Generator (click
Page 251 and 252:
do W[i]:=W[i]*(0.35875-0.48829*cos(
Page 253 and 254:
int i; double *h1 = new double[N];
Page 255 and 256:
Zoelzer biquad filters (click this
Page 257 and 258:
So tan(x) would be x - x^3/6 + x^5/
Page 259 and 260:
2 Wave shaping things (click this t
Page 261 and 262:
#define infile "a.raw" //input file
Page 263 and 264:
Band Limited PWM Generator (click t
Page 265 and 266:
Bit quantization/reduction effect (
Page 267 and 268:
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
Page 273 and 274:
dynamic convolution (click this to
Page 275 and 276:
Early echo's with image-mirror tech
Page 277 and 278:
for (y=-ceil(dist_max/(2*breite));y
Page 279 and 280:
} } j=0; for(i=0;i
Page 281 and 282:
Raummodell: H - Hoererposition L -
Page 283 and 284:
} } } angle=atan(y_pos/x_pos); if (
Page 285 and 286:
ECE320 project: Reverberation w/ pa
Page 287 and 288:
move#ser_data,r6; incoming data buf
Page 289 and 290:
movex:(r0)+,a; er delay asr#20,a,a
Page 291 and 292:
fold back distortion (click this to
Page 293 and 294:
Lo-Fi Crusher (click this to go bac
Page 295 and 296:
Most simple static delay (click thi
Page 297 and 298:
Parallel combs delay calculation (c
Page 299 and 300:
inherited; fA1:=0; fZM1:=0; end; de
Page 301 and 302:
implementation { TPhaser } function
Page 303 and 304:
end; a[1]:=b-fY[5]; b:=a[1]*fA1; fY
Page 305 and 306:
Polynominal Waveshaper (click this
Page 307 and 308:
Reverberation techniques (click thi
Page 309 and 310:
smsPitchScale Source Code (click th
Page 311 and 312:
Stereo Enhancer (click this to go b
Page 313 and 314:
Time compression-expansion using st
Page 315 and 316:
transistor differential amplifier s
Page 317 and 318:
WaveShaper (click this to go back t
Page 319 and 320:
Waveshaper (click this to go back t
Page 321 and 322:
Waveshaper :: Gloubi-boulga (click
Page 323 and 324:
VST SDK GUI Switch without (click t
Page 325 and 326:
private final int POINTS = 1 15; a
Page 327 and 328:
3rd order Spline interpollation (cl
Page 329 and 330:
5-point spline interpollation (clic
Page 331 and 332:
Antialiased Lines (click this to go
Page 333 and 334:
Automatic PDC system (click this to
Page 335 and 336:
} return ret; }
Page 337 and 338:
Block/Loop Benchmarking (click this
Page 339 and 340:
END; {$ELSE} BEGIN RESULT:=(Value S
Page 341 and 342:
} } throw new IllegalArgumentExcept
Page 343 and 344:
Center separation in a stereo mixdo
Page 345 and 346:
Center separation in a stereo mixdo
Page 347 and 348:
Clipping without branching (click t
Page 349 and 350:
Conversions on a PowerPC (click thi
Page 351 and 352:
Cubic interpollation (click this to
Page 353 and 354:
Denormal DOUBLE variables, macro (c
Page 355 and 356:
Use this eq. to calculate the appro
Page 357 and 358:
Denormalization preventer (click th
Page 359 and 360:
Anyway this version gives more expl
Page 361 and 362:
Dithering (click this to go back to
Page 363 and 364:
Envelope Follower (click this to go
Page 365:
fast abs/neg/sign for 32bit floats
Page 368 and 369:
Fast cube root, square root, and re
Page 370 and 371:
float z; _asm { rsqrtss xmm0, x rcp
Page 372 and 373:
1 - 1.5 (using fastexp6, the maximu
Page 374 and 375:
Fast log2 (click this to go back to
Page 376 and 377:
fast power and root estimates for 3
Page 378 and 379:
Float to int (more intel asm) (clic
Page 380 and 381:
Float-to-int, coverting an array of
Page 382 and 383:
Gaussian random numbers (click this
Page 384 and 385:
Pentium 4 and above, but not using
Page 386 and 387:
HTH DSP from : antiprosynthesis@hot
Page 388 and 389:
Matlab Time Domain Impulse Response
Page 390 and 391:
MATLAB-Tools for SNDAN (click this
Page 392 and 393:
end; // converts from MIDI note to
Page 395 and 396:
Motorola 56300 Disassembler (click
Page 397 and 398:
} __inline int round(float f) { int
Page 399 and 400:
Nonblocking multiprocessor/multithr
Page 401 and 402:
pow(x,4) approximation (click this
Page 403 and 404:
Real basic DSP with Matlab (+ GUI)
Page 405 and 406:
Really fast x86 floating point sin/
Page 407 and 408:
from : Christian@savioursofsoul.de
Page 409 and 410:
pfmul mm1, mm0 //mm1=a movq mm2, mm
Page 411 and 412:
esampling (click this to go back to
Page 413 and 414:
depending on whether you're looking
Page 415 and 416:
Sin, Cos, Tan approximation (click
Page 417:
fResult += 0.180141f; fResult *= fV
show all

musicdsp.org source code archive - WSInf

You also want an ePaper? Increase the reach of your titles

Delete template?

Save as template?