I have done some searching but can’t seem to find what I assumed would be an obvious topic.

What I mean by ‘good sounding’ is that relatively immediate smoothness you get in Reaktor, or SuperCollider. (Apologies if saying those words in this forum is blasphemy!) Max seems glitchier and harsher for synthesis in comparison.

I am planning to build a basic synth tutorial for my Masters project, and for my BTEC class, to get them into synthesis and ultimately Max. (Hopefully, like a more basic, but better sounding and more user friendly of the excellent Huddersfield University’s SYBIL).

Is there a ‘synth tips’ type, post that I can draw from and add to? If so, please can someone point it out? If not, would it be too presumptuous as a forum noob to suggest that we make one?

In the spirit of sharing ideas, please check out my first basic attempt at smoothing out audio switching, job no. 1 in my ‘good sounding’ Max synth job list, below.

Any constructive comments greatly appreciated.

(If anyone is interested, I’ll happily breakdown the rest of my job list and continued ideas to solve).

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Let me know what the other “problems” you’re trying to solve are.

lh

My two ideas are:

1) Avoid cycle~. 512 samples is pretty small for this kind of wavetable. Try 65536 and playback using play~.

2) There’s an abstraction in the CNMAT MMJ Depot* called silictor~ that has a ramp time… it does a similar thing to what you’ve done. Also, matrix~ has a ramp time (I’d probably use that).

mz

* http://cnmat.berkeley.edu/downloads

My two ideas are:

1) Avoid cycle~. 512 samples is pretty small for this kind of wavetable. Try 65536 and playback using play~.

2) There’s an abstraction in the CNMAT MMJ Depot* called silictor~ that has a ramp time… it does a similar thing to what you’ve done. Also, matrix~ has a ramp time (I’d probably use that).

mz

* http://cnmat.berkeley.edu/downloads

Thanks, I’ll definitely try that – why not wave~ by the way?

I’ll have a crack at the matrix object – I did find that with a straight fade it got quite crunchy below 100ms, curve seems to give smoother results so far, but it may be due to my patching inexperience.

Here is my list so far, which has prompted a mini project of defining specifically, in engineering terms, what is meant by ‘warm’ (I assume even order harmonics, esp 2nd, but need to look into that) and ‘smooth’, ect. Anyway, the list:

1. High frequency glitches in turning audio on and off, switching between oscillators, and fast variations of frequency and amplitude. Pretty simple this one, log fades and basic smoothing
2. Aliasing/Foldover effects. Is this possible to fix mathematically rather than with a filter? See if AES have any papers on Yamaha DX7 EAS Foldover elimination.
3. Basic synthesis has a harsh sounding basic oscillators and esp. filters compared to Reaktor or SuperCollider. Dissect Core cell modules in Reaktor, see what makes them tick.
4. Overly ‘clean’ ‘digital’ sound how to model the ‘warmth’ inherent in analogue synths? Free running oscillators, modelling freq variations. layering with poly…

(solving notes in italics)

That is what I have so far, I am also planning to define ‘good analogue characteristics’ and reverse-engineer into Max. Planning to get my hands on the college SH09, and compare / analyze. Any other info – recommended reading into that is appreciated.

Huge and abject thanks again…

Tim

Thanks, I’ll definitely try that – why not wave~ by the way?

That’s probably better. Play around with the interpolation modes.

mz

My DX7 aliases like crazy; that’s part of the fun. I think it runs at some odd sample rate (57k, maybe?), which make the aliasing hard to reproduce.

Sorry I can’t help with Max anti-aliasing.

mz

honestly, max isn’t the best environment to build synths in. Reaktor is better suited for that. However, Max, in the larger picture, is a much better, much more versatile program.

Check out the ddg tutorial on synth building, if you haven’t already.

The trick is (as with most sound applications) is not to approach max with ‘how can I get this to sound like Reaktor’ rather than ‘what can I do with this that I can’t anywhere else’. Reaktor will always be better at sounding like Reaktor. Hands down, no competition there.

If you want to take advantage of max in using synthesis call on its strengths, which is largely data manipulation. Try creating synth tones that don’t rely on midi on/off messages but rather a message indicating is duration in milliseconds. Now you have a synthesizer that know’s how long the note is going to last (a huge advantage over midi). What can you do with that?

Think about the different kinds of sonic parameters you want to control. How would you sequence them? A straightforward way would be to write them out in a coll and use some sort of metro-counter combination for playback. Too tedious? Try and find a way to generate this data algorithmically.

Ultimately, I’ve found great synthesis comes from complex time varying timbral changes. If you can make an instrument that reacts differently to pitch, velocity, duration among other parameters then you are on the path to something interesting.

My two ideas are:

1) Avoid cycle~. 512 samples is pretty small for this kind of wavetable. Try 65536 and playback using play~.

mzed are you suggesting that the small (512) size of the buffer in a cycle~ (or an anti-aliased tri~ or saw~ or rect~ I take it) is a limitation for producing higher quality synthesis? I am just curious because i’ve never considered this as a cause for weaker sounding synthesis although it does make some sense… Would a 65536 sample wavetable of a sawtooth wave sound better than a 512? Or does this really depend on the waveform?

That depends on the samplerate and interpolation algorithm.

honestly, max isn’t the best environment to build synths in. Reaktor is better suited for that. However, Max, in the larger picture, is a much better, much more versatile program.

Check out the ddg tutorial on synth building, if you haven’t already.

That’s really interesting, thanks!
Do you know if wave~ in anti-aliased, and how that is managed? I assume it’s a filter of some kind, (arbitrary, or linked to sample rate?)
That would be useful to know for anti-aliasing modulated waves…

Yes, I know that both Reaktor and SuperCollider are better at ‘traditional’ synthesis, but I am curious to find out exactly why, hence this project…
That is mostly because you can’t make a standalone app in reaktor and it seems far easier to make a good looking front end in Max5 rather than SC (that and I am a complete noob at SC!).

I have checked out most of the ddg tutorial, but it doesn’t seem to offer any more than the Max tutorials – is it worth going through that as well as the tutorials?

Thanks again

//

Ultimately, I’ve found great synthesis comes from complex time varying timbral changes. If you can make an instrument that reacts differently to pitch, velocity, duration among other parameters then you are on the path to something interesting.

Thanks for that – I had not thought of that approach to synthesis…

At the moment though, what I am trying to do is to create ‘traditional’ synth tones – and it seems tricky to get anything particularly ‘lush’ (another word to nail down, thanks!).

It’s not so much that I am trying to make Max sound like reaktor, more I am trying to define, in programming terms what are generally regarded as ‘good synth’ characteristics, and how to apply them in Max.

That depends on the samplerate and interpolation algorithm.

(sorry about the flurry of replies, should have done some last night…)

What if you were to have a wave~, using interpolation 1 (high quality) for both a single cos wave at 512 and 65536 lengths at 44.1kHz? Would that give any noticeable advantages?

Yes, that’s what I think. But I’m 1/2 superstitious and 1/2 scientific in my reasons for holding this belief. I while ago there was a thread where the sound of Max was being compared to other things. There are those who believe Reaktor or SuperCollider “Sound Better.” I did some subjective tests here at the lab with a sine wave who’s frequency was being swept by another sine wave, and everyone felt that the SuperCollider version sounded “smoother.” Around that time, a posting by James McCartney was linked; he hypothesized that there is nothing fundamentally higher quality about SC, but that users were pointed to higher quality oscillators and filters more quickly.

Anyhow, 512 samples is a small table to synthesize a sine wave. Especially in the case of 0.2 Hz, or whatever LFO frequency I was using. There will be a lot of interpolation going on and the signal to noise ratio won’t be ideal.

This is frequency dependent. A 512 sample wavetable is perfect at around 86.3 Hz (when the period is the same as the size of the wavetable). Over many useful frequencies, the quality difference is negligible. But, noise adds up, especially with a lot of oscillators or with modulation synthesis.

The CNMAT objects are using 65536 sample wavetables. Given the computing resources of the day, it seems like throwing more samples at quality couldn’t hurt.

I wouldn’t extend this logic to every waveform. Linear Interpolation works fine for a sawtooth… the waveform is mostly lines anyhow. Also, I believe the anti-aliased oscillators in Max are generated out of blits and the form is frequency dependant. They aren’t reading out of a fixed table.

I’d be curious if other people have thoughts on this.

mz

the only limit max has is the 24 bits of signal precision – but thats true to SC and R4 as well. everything else can be done for the cost of CPU and brain.
(but even the 24 bits of precision can be partly extended – by using the “hr” objects by joshua, for example you can convert
50 signals to double precision using [hr.* 1.] or [hr. + 0.] and
then sum the 50 channels with a [hr.*] to archive a higher quality mix.)

cycle~? lores~?
we are not forced to use them.
we are not forced to use clip~ and overdrive~ either.

we are allowed to upsample in a poly~.
we are also allowed to dither, to bandlimit, or to build
symetric networks for phaselinear filters.

we are also allowed to buy a 7000 dollar DA interface with a
proper clock at 192 kHz but thats another story.

hey, we are even allowed to load a multiband compressor
VST plug-in (from some socalled “pro” company) inside each
voice of or polyphonic maxmsp patch. or three of them.

i had that time when i refused to visit any more live
acts which are using maxmsp, because they all sounded
the same: minimalistic and consequently repetive 16-bit
quality drumloops with distortion.

but it is absolutely legal to break with the monkey see
monkey do principle.

-110

While you’re allowed to upsample in poly~, there is no comfortable solution provided to deal with input/output filtering inside the poly~.

The ideal would to have built-in filter insides poly~ when oversampled … please C74 …

Salvator

While you’re allowed to upsample in poly~, there is no comfortable solution provided to deal with input/output filtering inside the poly~.

The ideal would to have built-in filter insides poly~ when oversampled … please C74 …

Salvator

thats danmed true, at the moment linear interpolation after downsampling is all what i actually use. (i use the method which, i think, ben nevile once told me, using count~ for windowing)

yet it can make sense to have a “poly up 2″ when you have something like this:

in~ in~
comb~ comb~
*~
comb
out~

so that at least between the 2 processes a higher rate is kept.

-[poly~ proposal up 110]

Roald Baudoux

Aliasing is especially a concern when altering the dynamic of the audio (i.e. waveshaping, compressors etc;..)

And unfortunately, cascade~ is far from being ideal.
For 44100 work, even 10 cascaded low pass (60db/octave) @nyquist are not steep enough because :

1. if corner frequency is nyquist biquad filter will alter the phase and gain bellow nyquist down to the audio range.
Higher Q don’t help at all because of ringing.

2. With such cascaded filter the slope is “only” -60 for frequency of 44100 hz. This means that all the audio that WILL alias between niquist and say 30000 Hz, are not that much attenuated…

Depending on your application, (in my case distortion/softcliping) a good antialiasing requiere at least -60db not within 1 octave but within 1/8 of octave !

So fft~ or buffir~ are the only “efficient” way.
Well, did I say efficient ???
I should say “effective”, because in a poly, both fft~ and buffir~ easily put your CPU onto it’s knees…

I actually found my way using polyphase buffir~, or mixture of biquad + notch + buffir~ etc…
but still a LOT of hassle & more CPU that needed.

So an optimized routine built inside poly~ would be great !

If only it was easy like this “poly~ mygreatpatch up 8 @antialias masteringquality”

Salvator

the question is if antialisasing is wanted in any situation.
i mean it needs quite some CPU if you do that on a 64 voice synth or some multitapdelay-poly patcher .. my construction using cout~with biquad~ or lores~ already sucks a lot of cycles and its only my very last option when i think i really need this.

>>>So fft~ or buffir~ are the only “efficient” way.

FIR with a steep slope is the next problem in a 32-bit signal system which also should not have 1000 samples delay if it is for realtime hehe .. actually there seem to be no proper solution for a brickwall filter except fft.
but fft inside a poly is something i wont do anyway ..

depending on the musicla and technical effect one want to archieve, i think its ok to go a bit lower than nyquist.
its also ok to cascacade biquads like salvstor described; one lowpass and one notch and move tham around until it fits somehow.
of course with loosing phaseaccuracy …

For polyphonic synth, I guess using lower corner frequency is fine.

BTW : Anyone seen that this message in plugconfig documentation ?
“oversampling Arguments: code number
This message is currently ignored by the runtime plug-in environment.”

Such ‘global’ oversampling would be better that filtering each voice !
But speaking of that , I guess a main upsampled poly~ containing the “polyphonic poly~ ‘s” may also work the same way…

In that case, you only need 1 set of filters (input & output of the global poly~) and it would then not requiere 64 set of filters

Last, latency for good FIR and FFT are about the same. And considering it’s high frequency filtering, the lengh of the filter/fft can be of only about 64 or 128 samples.
It’s also possible to use FIR that are not linear phase. For example take the fingerprint of 10 cascaded biquad lowpass @22050 , and then use that in buffir~ (it’s about 64 taps and no latency).

Hope this help some,

Salvator

I imagine starting with good prepared waves (with proper harmonics and antialiasing and whatever) and run it thru phasor~ -> wave~ would be the most efficient and pleasing.

Salvator

JMD

that would be a nice addition to [plug-in~] and [plugout~] indeed.
for poly~ your wishes will remain wet dreams, but
for the runtimes themselves something like “global poly upification with antialiasing” could be an option.

>>>But speaking of that , I guess a main upsampled poly~ containing the “polyphonic poly~ ‘s” may also work the same way…

the idea is not wrong, it is just a lot of work then
getting messages into the subpoly. i for example usually
use a very flexible, custom way of talking to voices from
outside the poly patcher.