musicdsp.org source code archive - WSInf

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}*/ // Unrolled loop for faster execution /*samp -= c[0]*EH[HistPos] + c[1]*EH[HistPos+1] + c[2]*EH[HistPos+2] + c[3]*EH[HistPos+3] + c[4]*EH[HistPos+4] + c[5]*EH[HistPos+5] + c[6]*EH[HistPos+6] + c[7]*EH[HistPos+7] + c[8]*EH[HistPos+8];*/ // This arrangement seems to execute 3 clock cycles faster on a P-III samp -= c[8]*EH[HistPos+8] + c[7]*EH[HistPos+7] + c[6]*EH[HistPos+6] + c[5]*EH[HistPos+5] + c[4]*EH[HistPos+4] + c[3]*EH[HistPos+3] + c[2]*EH[HistPos+1] + c[1]*EH[HistPos+1] + c[0]*EH[HistPos]; output = round(samp + (frand() + frand() - 1)); //HistPos =(HistPos+8) % order;// The % operator is really slow HistPos = my_mod((HistPos+8), order); // Update buffer (both copies) EH[HistPos+9] = EH[HistPos] = output - samp; return ltos(output); } __forceinline int NS9dither16::processI(float samp) { int output; /*for(int x=0; x
Nonblocking multiprocessor/multithread algorithms in C++ (click this to go back to the index) Type : queue, stack, garbage collection, memory allocation, templates for atomic algorithms and types References : Posted by joshscholar[AT]yahoo[DOT]com Linked file : ATOMIC.H Notes : see linked file...
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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
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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
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: } __inline int round(float f) { int
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
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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
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musicdsp.org source code archive - WSInf

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