musicdsp.org source code archive - WSInf

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Remez Remez (Parks/McClellan) (click this to go back to the index) Type : FIR Remez (Parks/McClellan) References : Posted by Christian[AT]savioursofsoul[DOT]de Notes : Below you can find a Object Pascal / Delphi Translation of the Remez (Parks/McClellan) FIR Filter Design algorithm. It behaves slightly different from the c++ version, but the results work very well. Code : http://www.savioursofsoul.de/Christian/remez.zip
Resonant filter (click this to go back to the index) References : Posted by Paul Kellett Notes : This filter consists of two first order low-pass filters in series, with some of the difference between the two filter outputs fed back to give a resonant peak. You can use more filter stages for a steeper cutoff but the stability criteria get more complicated if the extra stages are within the feedback loop. Code : //set feedback amount given f and q between 0 and 1 fb = q + q/(1.0 - f); //for each sample... buf0 = buf0 + f * (in - buf0 + fb * (buf0 - buf1)); buf1 = buf1 + f * (buf0 - buf1); out = buf1; Comments from : mr.just starting comment : very nice! how could i turn that into a HPF? from : dsp@dsparsons.nospam.co.uk comment : The cheats way is to use HPF = sample - out; If you do a plot, you'll find that it isn't as good as designing an HPF from scratch, but it's good enuff for most ears. This would also mean that you have a quick method for splitting a signal and operating on the (in)discreet parts separately. :) DSP
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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 +
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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
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 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
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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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