Tim Canfer

Hi, I am trying to track down information, tips, resources etc to optimize Max MSP for 'good sounding' synthesis.

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).

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. 

making-max-msp-synths-sound-good

If all you are trying to do in this patch is fade one sound source out and another one in then look at [amtrix~] which is like [selector~] but you can set a ramp time for making and breaking connections and thus avoid nasty clicks.

Let me know what the other "problems" you're trying to solve are.

Tim Canfer wrote on Thu, 19 March 2009 06:05

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).

Tim Canfer wrote on Thu, 19 March 2009 06:05Hi, I am trying to track down information, tips, resources etc to optimize Max MSP for 'good sounding' synthesis.

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).

mzed wrote on Thu, 19 March 2009 10:41Tim Canfer wrote on Thu, 19 March 2009 06:05

Thanks, I'll definitely try that - why not wave~ by the way?

mzed wrote on Thu, 19 March 2009 10:41Tim Canfer wrote on Thu, 19 March 2009 06:05Hi, I am trying to track down information, tips, resources etc to optimize Max MSP for 'good sounding' synthesis.

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

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...

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.

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)

Tim Canfer wrote on Thu, 19 March 2009 12:25

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

Tim Canfer wrote on Thu, 19 March 2009 12:26

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.

Tim Canfer wrote on Thu, 19 March 2009 12:26
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.

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.

the waveforms in max objects are already anti-aliased (saw~, tri~, rect~).

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.

Don't let the kids over at the NI forum read that! I'd like to disagree here. Max can get some pretty nice lush tones, as well as doing super harsh aliased goodness that I haven't heard anywhere else.

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. 

Don't let the kids over at the NI forum read that!  I'd like to disagree here. Max can get some pretty nice lush tones, as well as doing super harsh aliased goodness that I haven't heard anywhere else. 

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?  

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.  


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?

"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.

ComfortableInClouds wrote on Thu, 19 March 2009 18:57

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 

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?

ComfortableInClouds wrote on Thu, 19 March 2009 18:57the waveforms in max objects are already anti-aliased (saw~, tri~, rect~).

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!).

Nick Inhofe wrote on Fri, 20 March 2009 11:15

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.

Nick Inhofe wrote on Fri, 20 March 2009 11:15Don't let the kids over at the NI forum read that!  I'd like to disagree here. Max can get some pretty nice lush tones, as well as doing super harsh aliased goodness that I haven't heard anywhere else. 

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.  

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.


marlon brando wrote on Sat, 21 March 2009 04:41

(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?

marlon brando wrote on Sat, 21 March 2009 04:41"Would a 65536 sample wavetable of a sawtooth wave sound better than a 512? Or does this really depend on the waveform?"

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?


Kevin Shea Adams wrote on Fri, 20 March 2009 15:29

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.

Kevin Shea Adams wrote on Fri, 20 March 2009 15:29
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?

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.

you can be sure many people understand what you mean when

you say max sounds harsher than SC or reaktor.

of course it doesnt - but let me tell you i thought the 

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.)

we are not forced to use clip~ and overdrive~ either.

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 

but it is absolutely legal to break with the monkey see 

you can be sure many people understand what you mean when
you say max sounds harsher than SC or reaktor.
of course it doesnt - but let me tell you i thought the 
same - for severeal years!

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.

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 wrote on Mon, 23 March 2009 14:11

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:

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

Salvator wrote on Mon, 23 March 2009 14:11Roman,

Yes, the lack of a good antialiasing filter for upsampled polys is really a big drawback. You may use cascade~ but the cutoff frequency is still far from the Nyquist frequency if you want to avoid most aliasing even if this object is the lowpass filter with the steepest slope. It's a poor man's solution. I appreciate how Reaktor handles this.

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. 

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"

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.

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" 

>> a good antialiasing requiere at least -60db not within 1 octave but within 1/8 of octave !

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.

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.

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 ... 

Sure, good oversampling solution depend on the application...

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

BTW : Anyone seen that this message in plugconfig documentation ? 

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).

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). 

BTW : I'm not a synth specialist, but good sounding synth may also be more about repartition of the partial/harmonics than alisasing, which is only 1 obvious part of the puzzle.

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.

BTW : I'm not a synth specialist, but good sounding synth may also be more about repartition of the partial/harmonics than alisasing, which is only 1 obvious part of the puzzle. 

Use csound~ and/or rtcmix~ inside maxMSP as sound sources.

They are hard to beat in terms of sound quality and flexibility.

On the other hand the learning curve is pretty steep, specially CSound. But it worth it if you want to achieve a professional sound and even further

Use csound~ and/or rtcmix~ inside maxMSP as sound sources.
They are hard to beat in terms of sound quality and flexibility.
On the other hand the learning curve is pretty steep, specially CSound. But it worth it if you want to achieve a professional sound and even further 

>>>Such 'global' oversampling would be better that filtering each voice !

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

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.

I second the use of ~csound inside Max. Only just scratching the surface now myself, but the returns have been more than worth it.

I also use csound~, it hands down has the best quality of sound.

Learning it is definitely a well worth investment of time.

I also use csound~, it hands down has the best quality of sound.
Learning it is definitely a well worth investment of time.


I just use reaktor as a vst in max, reaktor makes building nice synths easy, easier than msp has ever made it imho. Max is much better for all my algorithmic composition and control and 

"connect x to y" needs so I let them both do what they do best. Just my two cents worth, not looking to start a flame war over Reaktor vs MSP.

I just use reaktor as a vst in max, reaktor makes building nice synths easy, easier than msp has ever made it imho. Max is much better for all my algorithmic composition and control and 
"connect x to y" needs so I let them both do what they do best. Just my two cents worth, not looking to start a flame war over Reaktor vs MSP.


Can I just say that after years of thinking that max was bad for making synths, after truly learning synthesis inside and out from playing with analogue modular systems like doepfer, as well as my nord modular g2, absynth, other virtual analogue synths and reading computer music tutorial, I know know that max can sound as good as any synth, and its not as hard as I thought (especially now that there is poly~, when I started max there was no such thing)

making max MSP synths sound good...