We can’t know everything. That’s a given, but there are some things we don’t know that we think we should, particularly when on a forum like this. When other people talk about these mysterious things we may nod politely, hoping that they will explain what they mean without us having to ask. And on Elektronauts, where many people are synthesis experts, I don’t know about you but some of the technical talk can go a bit over my head.
So here’s the thread to lay it on the line. Tell us what you need to know about synthesis or music production - the questions you’ve always wondered about but have been too embarrassed to ask in case you look stupid. Hopefully another forumite will be able to answer.
The one rule of this thread is that there should be no judgement cast on the questioner. “It’s only easy if you know the answer”, after all!
Here’s mine:
Envelopes! I know how an ADSR envelope works, but I’m really struggling when it comes to their practical effect in certain situations. Consider a standard envelope affecting filter cutoff on the AK, A4 or Rytm. The Attack is the time taken for the filter to reach its maximum value, right? But does that scale with the position of the cutoff knob? If it’s at 127 then obviously the attack ends at a cutoff level of 127, but if I set the cutoff knob to 64, will the Attack reach its maximum at a cutoff value of 64? And how does the position of the envelope knob (ie. how much I’ve opened the filter envelope) affect this?
And then I’m often not properly hearing the differences between different Decay, Sustain and Release values. Maybe I’m having problems with this because I’ve used the Rytm more, and the sounds generally happen too quickly for subtle changes in the later DSR elements of the filter envelope to make much of a difference.
So, what is the best way to properly explore how the filter envelope changes a sound, on both the Rytm and AK/A4?
Regarding envelopes, attack is the time it takes from when you first hold the note, for it to get from nothing to the maximum value. Decay is the time from that maximum value to the designated sustain level.
When you’re setting the sustain value, that’s the level it will stay at while you’re holding the note. So if it’s at maximum, the note will have its attach and then stay at the maximum value while you’re holding it down. If the sustain is at 0, the note will go through the attack and decay ramps, and then go back to 0.
The release part of the envelope sets the time it takes from when you let go of a note, for the envelope to go back to 0, from wherever it was when you let go. Long release times means the note will take a long time to trail off which is good for pad sounds. A release of 0 means the envelope will close instantly.
A good way to check out envelopes on the AR would be to use a cymbal, piano or other long sound so you can hear when the envelope takes a long time to close. Check out some preset sounds on the A4 to see how different envelope settings affect the sound, have a play and experiment!
One thing I really should know more about is compressors. I have a very musical ear but have a hard time discerning between changes on it, can’t say they make much of a difference for me.
With most ADSRs (waldorf do things differently sometimes) the start point 0 and first destination 127(generally) define the maximum range of modulation control, the env amount scales this range effectively from 0>1x (using a 7bit 128 range)
So if you add a modulation to a static value the proportion added is defined by the env amount, anything from none to full range, but if your existing value (say cutoff) is already half way up, you could sorta clip the modulation influence if env amount is too high, this just results in a bit of a flatline max cutoff value until the env shape returns below the max-cutoff level
Generally A D and R are defined as a time value and the profile of the ramp can be shaped, typically linear/exponential and you can also force the start point of the env to be relative to the modulated param val or to retrigger taking into consideration its current position if incomplete, the a4 is comprehensive and has a whole bunch of additional envelopes for percussive profiles and so on
Sustain level is the level that the env will reach if you hold long enough into the profile and it passed A and D, when you lift it will go to R, if you lift early during A/D stage, it will drop at the D rate, not the R rate, this is how you could get different ‘decays’ by pressing the key briefly or slightly longer (longer than A+D) to reach the sustain S part (specified as a level, not a time)
Either way, that was just a quick thought to get the ball rolling, but i really just wanted to say that for me, by far the best way to audition envelopes is to modulate pitch, try to have a decaying gate type env for amp (full instant volume and long release) and use a spare env to explore shape possibilities and retriggering influence, this is easy with A4 as it has modulation options aplenty
Sync. From what I’ve understood, two waveforms out of phase and once the sync source waveform completes its cycle, it resets the destination waveform? Then there’s hard sync and soft sync…
AM -> something modulating the volume really really fast, which somehow creates something audible.
Ring modulation -> at a total loss… I know it sounds cool.
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Seems like a precision tool for the concepts I’m struggling with – thanks Avantronica! Always meant to read up on the concepts, instead of going by gut feeling, now I have no excuse not to.
Re: Envelopes: Think of the initial cutoff value as an offset. If you have a positive ENV Amount, it gets “added” to the initial cutoff value (much like moving the cutoff knob clockwise at the envelope’s rate). The inverse goes for a negative ENV amount - it subtracts from the initial value (i.e. it moves the cutoff knob counter clockwise).
Experiment here: on the A4 AMP page, set VOL to 0, A=0, DEC=INF, S= Max, R=Inf. Press a key - nothing is heard. Go in your ENVF page and set DEST to AMP VOL fully clockwise - now the ENVF controls your voice volume. Much like on an SH101 where you only have one envelope to control filter cutoff and (VCA) volume…
I’m no synthesis expert by any means. I only know the basics and usually just fiddle around until things sound good. It takes me a while, but I eventually can get to where I want to go. I’ll stay tuned to this thread for the real synth guru advice!
Some things I wonder about but haven’t had the time yet to dig in and learn (or RTFM).
Sync
Sample and Hold
Polyphony modes
On LFOs, SPD vs MUL. I have a hard time determining practical difference. They both just seem to increase the intensity of the LFO effect on the sound.
On some patches, I always struggle with trying to figure out if I should be adjusting the filter cutoff param vs. the envelope amount param. Ha.
Maybe this is not a specific scientific parameter of synthesis, and if it’s outside of the intended scope of this thread, please ignore, but I am always wondering how to use Parameter locks on the A4 to the fullest. On the Machinedrum, using P-locks came naturally to me and it always sounded musical and interesting. On the A4, I am always at a loss as to what to P-lock, plus whenever I start doing it, it always ends up sounding like random bug-noise crap, imo. I have yet to effectively, and musically, use P-locks on the A4.
Sheesh, writing this stuff out and then reading it - I feel pretty stupid. Or maybe just lazy. Or maybe just extremely busy with other life duties and no time to study synthesis. Either way, kind of embarrassing.
I love AM. Usually you’ll think of amplitude modulation as slow modulation that gives dynamics or event over a timeline… think of that as notes on a written piece of music.
Really fast modulation of the amplitude effects the amplitude so fast that it no long is perceived as a timeline event, but as tonal, timbral, frequency information.
Good FM is hard to achieve if you’re after “musical” results. I tend to use a lot of FM on my modular synth for interesting percussion… where “correct musical notes” don’t apply as much. There’s a bit of math involved between the modulated vco and the modulator vco.
A S&H has two inputs and one output.
Input 1: is a signal input to be sampled. It can be noise, or any other waveform. In your A4 LFO it is one of the early waveform you can select (sine, triangle, square, exponential etc)
Input 2: is the “start to sample Input 1 now” input. Whenever there is a high signal here (can be a pulse, clock etc, A4 trigger), it “samples” the value of input 1 and “holds” that particular value of the input 1 signal at that time.
Output: The value of Input 1 signal held at the time the signal was sampled. The value is held until Input 2 receives the next command to sample.
Oftentimes S&H is only thought of as a random stepped modulation. However, you can Sample and Hold a triangle wave e.g.; then you will get a stair stepped rising and falling value.
On the A4 LFO, put its MOD to “Hold”. Select triangle as a waveform. Program a sequence of same 16th notes. Set DEST to pitch. Increase DEP slowly. You should hear a stepped melody emerge. Increase the SPD, MULT and SHP parameters…
when I’m unsure of what my LFO is doing – I’ll assign it to pitch, because for me, it’s easier to hear all the LFO action – then assign it back to the parameter I actually want to modulate.
I love AM. Usually you’ll think of amplitude modulation as slow modulation that gives dynamics or event over a timeline… think of that as notes on a written piece of music.
Really fast modulation of the amplitude effects the amplitude so fast that it no long is perceived as a timeline event, but as tonal, timbral, frequency information.
Good FM is hard to achieve if you’re after “musical” results. I tend to use a lot of FM on my modular synth for interesting percussion… where “correct musical notes” don’t apply as much. There’s a bit of math involved between the modulated vco and the modulator vco.
This is the best resource I could find on FM – http://www.soundonsound.com/sos/apr00/articles/synthsecrets.htm[/quote]
Must get up an’ learn those rules. Weather man and the crazy chief. One says sun and one says sleet A.M., the F.M. the P.M. too. Churning out that boogaloo
Same here exactly !
There was a great post about compressor settings here (or maybe the old forum) which explained them well. It said something like, “attack is like your top end controll” or something.
Anyone remember this and link it for us ?
Im pretty happy with synthesis knowledge, but mixing is a bit of a dark art to me - even though I used to do it for a living for a while. Probably as much because I start losing interest beyond creating patches and jamming, mixing for others is much easier because you arent tempted to go back and mess with the source so much.
Am interested though, who here describes themselves as a musician first, or would you call yourself an engineer or synthesist?
Do the types of wires used effect sound to the point I should put much thought into it?
Sometimes I wonder if using a coiled 1/4" cord is worse than a straight one because it’s bunched up. Sometimes I use an old wire from my TV for component video as an audio wire because it has the same RCA style jacks.
Does the length of a wire matter? Does using an 1/8th inch with a 1/4’ adaptor introduce noise where a 1/4’ wouldn’t?
Always meant to read up on the concepts, instead of going by gut feeling, now I have no excuse not to.
