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I found a usefull description on programming waveforms!

 
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#1 02-10-28 00:32

I found a usefull description on programming waveforms!

I found a usefull description on programming waveforms!

From: "Joe Wright" <joe@nyrsound.com>
To: <music-dsp@shoko.calarts.edu>
Subject: Re: waveform discussions
Date: Tue, 19 Oct 1999 14:45:56 +0100

After help from this group and reading various literature I have finished my
waveform engine.  As requested, I am now going to share some of the things
are have learnt from a practical viewpoint.

Problem:

The waveforms of interest are sawtooth, square and triangle.

The waveforms must be bandlimited (i.e. fitting under Nyquist).  This
precludes simple generation of the waveforms.  For example, the
analogue/continuous formular (which has infinite harmonics):

s(t) = (t*f0) mod 1  (t=time, f0=frequency of note)

procudues aliasing that cannot be filtered out when converted to the
digital/discrete form:

s(n) = (f0*n*Ts) mod 1 (n=sample, Ts equals 1/sampling rate)


The other condition of this problem is that the waveforms are generatable in
real-time.  Additionally, bonuses are given for solutions which allow
pulse-width modulation and triangle asymmetry.


The generation of these waves is non-triaval and below is discussed three
techniques to solve the problem - wavetables, approximation through
processing of |sinewave| and BLIT integration.  BLIT integration is
discussed in depth.


Wavetables:

You can generate a wavetable for the waveform by summing sine waves up to
the Nyquist frequency.  For example, the sawtooth waveform can be generated
by:

s(n) = Sum[k=1,n] (1/k *sin(2PI*f0*k*Ts)) where f0*n < Ts/2

The wavetable can then be played back, pitched up or down subject that
pitched f*n < Ts/2.  Anything lower will not alias but it may lack some
higher harmonics if pitched too low.

To cover these situations, use multiple wavetables describing different
frequency ranges within which it is fine to pitch up or down.

You may need to compromise between number of wavetables and accuracy because
of memory considerations (especially if over-sampling).  This means some
wavetables will have to cover larger ranges than they should.  As long as
the range is too far in the lower direction rather than higher, you will not
alias (you will just miss some higher harmonics).

With wavetables you can add a sawtooth to an inverted sawtooth offset in
time, to produce a pulse/square wave.  Vary the offset to vary the pulse
width.  Asymmetry of triangle waves is not possible (as far as I know)
though.


Approximation through processing of |sinewave|

This method is discussed in detail in 'Modeling Analog Synthesis with
DSPs' - Computer Music Journal, 21:4 pp.23 - 41, Winter 1997, Lane, Hoory,
Marinez and Wang.

The basic idea starts with the generation of a sawtooth by feeding
abs(sin(n)) into a lowpass followed by a highpass filter.  This approximates
(quite well supposedly) a bandlimited sawtooth.  Unfortunately, details of
what the cutoff for the lowpass should be were not precise in the paper
(although the highpass cutoff was given).

Square wave and triangle wave approximation was implemented by subtracting
abs(sin(n)) and abs(sin(n/2)).  With different gains, pulse width (and I
presume asymmetry) were possible.

For more information, consult the paper.


BLIT intergration

This topic refers to the paper 'Alias-Free Digital Synthesis of Classic
Analog Waveforms' by Tim Stilson and Julius Smith of CCRMA.  The paper can
be found at http://www-ccrma.stanford.edu/~stilti/papers

BLIT stands for bandlimited impluse train.  I'm not going to go into the
theory, you'll have to read the paper for that.  However, simply put, a
pulse train of the form 10000100001000 etc... is not bandlimited.

The formular for a BLIT is as follows:

BLIT(x) = (m/p) * (sin(PI*x*m/p)/(m*sin(PI*x/p))

x=sample number ranging from 1 to period
p=period in samples (fs/f0).  Although this should theorectically not be an
interger, I found pratically speaking it needs to be.
m=2*((int)p/2)+1 (i.e. when p=odd, m=p otherwise m=p+1)

[note] in the paper they describe m as the largest odd interger not
exceeding the period when in fact their formular for m (which is the one
that works in practive) makes it (int)p or (int)p +1

As an extension, we also have a bipolar version:

BP-BLIT k (x)=BLIT(x) - BLIT(x+k)   (where k is in the range [0,period])


Now for the clever bit.  Lets start with the square/rectangle wave.  Through
intergration we get:

Rect(n) = Sum(i=0,n) (BP-BLIT k0 (i) - C4)

C4 is a DC offset which for the BP-BLIT is zero.  This gives a nice
iteration for rect(n):

Rect(n) = Rect(n-1) + BP-BLIT k0 (n)  where k0 is the pulse width between
[0,period] or in practive [1,period-1]


A triangle wave is given by:

Tri(n) = Sum(i=0,n) (Rect(k) - C6)

Tri(n) = Tri(n-1) + Rect(n) - C6

C6 = k0/period

The triangle must also be scaled:

Tri(b) = Tri(n-1) + g(f,d)*(Rect(n) - C6)

where g(f,d) = 0.99 / (period* d * (d-1))   d=k0/period

Theorietcally it could be 1.00 / ... but I found numerical error sometimes
pushed it over the edge.  The paper actually states 2.00/... but for some
reason I find this to be incorrect.

Lets look at some rough and ready code.  I find the best thing to do is to
generate one period at a time and then reset everything.  The numerical
errors over one period are negligable (based on 32bit float) but if you keep
on going without resetting, the errors start to creep in.

    float period = samplingrate/frequency;
    float m=2*(int)(period/2)+1.0f;
    float k=(int)(k0*period);
    float g=0.99f/(periodg*(k/period)*(1-k/period));
    float t;
    float bpblit;
    float square=0.0f;
    float triangle=0.0f;
    for(t=1;t<=period;t++)
    {
        bpblit=sin(PI*(t+1)*m/period)/(m*sin(PI*(t+1)/period));
        bpblit-=sin(PI*(t+k)*m/period)/(m*sin(PI*(t+k)/period));
        square+=bpblit;
        triangle+=g*(square+k/period);
    }
    square=0;
    triangle=0;

Highly un-optimised code but you get the point.  At each sample(t) the
output values are square and triangle respectively.


Sawtooth:

This is given by:

Saw(n) = Sum(k=0,n) (BLIT(k) - C2)   [as opposed to BP-BLIT]
Saw(n) = Saw(n-1) + BLIT(n) - C2

Now, C2 is a bit tricky.  Its the average of BLIT for that period (i.e. 1/n
* Sum (k=0,n) (BLIT(k)).  [Note] Blit(0) = 0.

I found the best way to deal with this is to have a lookup table which you
have generated and saved to disk as a file which contains a value of C2 for
every period you are interested in.  This is because I know of no easy way
to generate C2 in real-time.

Last thing.  My implementation of BLIT gives negative values.  Therefore my
sawtooth is +C2 rather than -C2.


I hope this helps, any questions don't hestitate to contact me.


Joe Wright - Nyr Sound Ltd
http://www.nyrsound.com
info@nyrsound.com
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