Hi

I've found a pdf containing pictures of all the waveforms and wavetables the PPG wave synthesizer used, it's quite good really as I have wanted to build a PPG in Pure Data for a while, but I didn't think such a resource existed.

I have two questions
1. I have never really used pictures and visuals in Pd before, so I was wondering if it would be possible to take the picture of a waveform from the pdf and write it into a table in Pd, and then hopefully save it?

2. If this is possible, then I would like to build the wavetable, which from what I gather is just a list of waveforms that are added together over time, then played back at a given frequency. Is this possible, and if so, does anyone know of any good tutorials that would give me an idea of how to do it?

This is going to be my summer project, I might even try my hand at modeling the PPG filter then compiling it in C, although that is running before I can walk

I have attached the pdf for anyone interested in PPGs or wavetable synthesis in general

Cheers!

EDIT- I realised whilst I was uploading the pdf it was 2.5MB, which is a lot to stick on this site, so here is the link I got it from
www.seib.synth.net/documents/PPGWTbl.pdf

Hi
I've read about both of these, but I can't see the difference, [vd~] is a variable delayline reader, but I can vary the point at which [delread~] reads a delayline in real time with a number box.
Can anyone else offer an answer?!
Thanks

I was looking for a filter that would allow me to design it by drawing the desired frequency response in an array, something that a few people were messing around with on the pdlist, but from what I remember it was never finished.

on the floss manual site [filtersme2~] is mentioned as doing this, but it isn't built in to pd extended, and I can't find it anywhere to download.

Does anyone now where I can find it?

Thanks

Hi
I've been using a patch for a digital waveguide I found to try and build my own

I've just realized that the delay lines he made are bi-directional, so they include both the left and right traveling waves. In the patch I have connected both delay lines up via a scattering junction, as this is the only way I can select input and output position.

Basically I want to be able to select the output out of the same delay line that I am able to select the input, as I only want to model 1 string, not the sympathetic resonance and output of one string which is excited by another.

I have included bits of description in the patch, basically the left hand side is the waveguide I can select input position for, but I also want to select the output position for the same one, therefore not needing the scattering junction or a second waveguide

Any help would be greatly appreciated

(ps i haven't sorted out the loadbangs properly yet, so values for top and bottom filters, waveguide lengths and filter values need to be put in before it will make any sound, sorry)

Thank you

http://www.pdpatchrepo.info/hurleur/Digital_Waveguide_Pd.zip

Hi
I've been trying to make the analogue volume leveller mentioned in another thread
http://puredata.hurleur.com/sujet-1232-analogue-style-volume-levelling

I want to use it do do a saturation effect, but it just isn't working, I've looked at the table and it is just a flat line

Can anyone help me please?
I have attached what I have done

Thanks

http://www.pdpatchrepo.info/hurleur/clipping.pd

Hello!
I remember seeing a pd patch from stanford universities website a while ago of a digital waveguide. it had a graph (technically an array) with level on the y axis and frequency on the x axis, this meant you hit 'bang' and could see the frequencies in the sound, particularly the resonances of the waveguide, and see as they died down over time.
Does anyone know how to do this? I have a feeling an FFT is in there somewhere....?
Thanks

Hi

I found a paper published in 2003 entitled "digital implementation of the moog ladder filter" (the one muug~ is based on), and it made me wonder if there are any other papers like this

Does anyone know of any papers or sites that give information on how to design digital copies of old filters?

I'd really like to find the ARP filter, Oberheim filter and the Curtis filter out of the sequential circuits and dave smith synthesizers

Thanks

Hi, I've started using the resonant low pass filter lp2~, and I am having problems with the sound. When 'interpolation time' is set to zero, I get a loud click when I sound a note, this changes to a metallic sound when the value is set high, but the filter also tales a lot longer to react to the filter envelope I have hooked up to it.
Does anyone know why this is, and what exactly does the interpolation time of the filter do?
thanks

Hi

I am wanting to design a Biquad filter, and I know all the relevant equations for calculating the co-efficients, but the problem is I have only used the [lop~] object up until now, which whilst not being resonant, I am able to hook up an envelope generator to change the cutoff frequency.

I was wondering if anyone knew of a way of designing a biquad filter which would allow an enevlope generator/LFO to change it's co-efficients?

Thanks

I did my final year project for my degree on physical modeling before this summer. I chose the guitar, specifically electric, and I built my final synth in PureData. Is this what you mean? I can upload my final annotated patch if you want?
I read Julius smith's entire thing on his site, and it took a while to work out how to implement it in pd, especially with delay lines. Built one in Matlab first, then did it in pd because I hate programming in text like with C++. Built a delay with feedback, bit-crusher, analogue/digital overdrive and distortion, as well as ring mod and amplifier feedback loop. I ran out of time, so I didn't get to do pickup design-specific modeling, or sympathetic resonance. I did manage to get different types of excitation (pick or finger etc.)
I've got a pdf of my dissertation hanging around somewhere as well, puts all the physics stuff into laymen/musician's terms
anyway, let me know if this is what you're on about

maybe split the signal in to directions, one goes through a bandpass filter [bp~] and is then added to the original, that should be like a 1 band graphic equalizer. that might work

I was thinking more of the actual components of the compressor circuit, and how they won't behave as expected, certainly as they get hot they begin behaving differently.
If you are modeling a compressor with valves would be different to solid state, and I don't know how you'd go about modeling valves.
I suppose what I'm trying to say is your bog standard digital 1 pole filter response is a straight line, whereas the analogue equivalent would not be a perfect straight line, as the resistors and capacitors themselves don't behave linearly, and don't have a <perfect>linear relationship. And also, [tanh~] would create a saturation effect, but not one that sounds truly analogue, as the response of a real overdrive pedal would not be a perfect [tanh~] line.
I've never found any decent papers on Analogue Modeling, which is annoying, especially as it is such an important part of contemporary DSP.
I'm pretty interested in this topic myself now, it would be useful to be able to model electronic circuits in Pd

It will probably be a waste of time trying to faithfully model the compressor in Pd, it's easy to create a simple compressor, but what you need to do is physically model the circuitry. This will mean all the non-linearities present in the circuit will be recreated, and the compressor will sound like a real analogue compressor. I wanted to model analogue filters in Pd, but it can't be done, what people have done in the past is written them in C and compiled them into externals.

That's if you wanted to do a realistic imitation of it, which for a uni project I assume you would want to do. Analogue modeling is a difficult one, I just completed a uni project on physical modeling of a guitar using Pd and I wanted to model a Moog filter to pass the guitar signal through, but I can't write in C.

Actually, I remember reading a thesis where someone modeled the tone controls of a guitar in C, and to get a difference equation for a filter they built the circuit in SPICE, which is freeware for PC, but there is a Mac version. Once you build the circuit it plots graphs of the output, and you can swap around components. I don't know if that's any help, but it could be a start seeing as I assume you built it yourself so will have the circuit diagrams for it.

sorry, that was the wrong post, I actually found a bitcrusher patch attached on a page about analogue overdrive
http://puredata.hurleur.com/sujet-1232-analogue-style-volume-levelling
it is on the second post from the bottom

I couldn't get that help patch to work for some reason, I think that is what I'm after though. I will look into it a bit further, Thanks

Cheers BerengerRecoules, the hammond one will come in handy when designing new waveforms for the synth.

I haven't had time to check your patches katjav, but they sond like exactly what I was after. I can still see myself having to fiddle about with them in something like paint, or the equivalent on Mac, so all the waveforms are at zero corssing at the edges of the tables.

I've got a lot of work on this week, as I have done the past fortnight, but when I've finished that I will get straight on it and post up my first attempt.

Also I will need to get my head around interpolating between waveforms, as this is how many of the wavetables in the ppg work, will also save disk space.

One upside is the PPG was known for aliasing, so I won't need to worry about bandlimiting the waveforms!

Nau, this was posted for bit depth reduction originally by Maelstorm

[expr~ $v1*.5+.5] <-put the signal in 0-1 range
|
| [8\ <-bit-depth
| |
| [expr pow(2, $f1)-1]
| |\
[expr~ int($v1*$f2)] \
\ |
\ |
[expr~ $v1/$f2*2-1] <-bring it back to -1 - 1 range

I used it recently in a waveguide guitar patch I built and it worked perfectly, I'm going to put in the the ppg emulation.

The page I found it on: http://puredata.hurleur.com/sujet-1214-bit-crusher
One of the posts further down the page has a bitcrusher attachment

There is also a ring modulator in the help browser in audio examples, I think there might be two actually. I don't really know how they work though, I can't draw any similarities between the help patch and a standard ring mod

Thanks, I'll have a look into it, I think vd~ might work better in my waveguide than delread~ , I am getting a lot of clicking when I move the input and output position on the delayline using delread~

I thought maybe cutting all the parts from the output delay line that don't actually create a delay line then connecting them up to the input delay line, but this doesn't work.
Does anybody have any suggestions?
thanks

nice one, much appreciated!

That's exactly what I wanted, thanks a lot!