QS effect unit basics
Posted by Markus Nentwig on September 04, 1998 at 20:42:51:
There have been several questions concerning the effects unit of the QS synth recently.
I've written down some information from my experience that I think may be useful.
(Maybe I'll title this)
What you always wanted to know about QS FX units
(but never dared to ask)
Although I'm not the most experienced QS programmer, I'll try to write down my own experience and summarize some of the
most frequently asked questions.
The first and most important tip is this: You NEED the manual. If you don't have one, get one from Alesis (as a .pdf file, it's
on their homepage available for download).
Although the manual section concerning the FX seems a little hard to comprehend at first reading, it contains some invaluable
diagrams. They are the 'key' to the FX unit. (Maybe you'll need the printed version to make out any details, though.I haven't
tried to print the .pdf file yet.)
And, although I may sound a little nerdy, don't expect to understand it all at the first time. I think I have spent several days
pondering over these pages myself (and being a student of electrical engineering surely helps, too)
A) Basic concept or 'why do I need FX at all?' or some blah-blah that you can skip over if you're in a hurry
As I read my own lines now, I come to the conclusion, that this section is propably the least basic, although it's right at the
start. So please don't despair if I'm taking big steps - the idea is simply to get a feeling for the subject.
The FX unit is there to achieve, simply said, modifications on the sound that the voices can't achieve alone.
An example to start with: reverb. In physical reality ('on stage') the natural reverb is caused by the sound being reflected
from walls, floor, ceiling and whatever is close. Plus, a part of the reflections is reflected again, part of those again and
again, giving us the impression of a life stage or wherever we are.
In the early days of electroacoustics, big theatres used to have special 'echo rooms' in the basement: large, tiled chambers
that were usually empty but some speakers and microphones.
The actor on stage got a microphone that fed its signal over speakers in this very room, where some microphones recorded
the artificial echo and brought the signal back on stage, driving some hidden speakers which fired into the audience, giving
the impression of depth.
Later, people used loose coils of steel wire or big sheets of metal that were fed with an audio signal by a special sort of
speaker, whose reverberations were recorded by microphones to create artificial reverb.
Ask a guitar player, he'll show you (or simply kick his amp - you'll hear the thrashing of the coils over the speaker- then run
like hell or he'll thrash you...)
Modern reverb units are no more based on mechanics but use digital delays and are so small that they fit into a pocket or into
a QS synth.
Well, so much for nostalgy, but back to the subject now.
The echo room as much as the coil or plate reverb and the digital reverb have a common concept:
They take some of the audio signal,
they process it into a 'reverb signal'
and sum this 'reverb signal' up with the original signal.
Here comes the connection to the QS: For programming the FX unit, you must take three principal steps:
You have to tell your QS
- to take what source signal from the voices and how much of it
- what to do with the branched-off signal - the QS has more to offer than artificial reverb
- to take how much of the now generated effects signal and put it on the output, combining it with the original signal the voice
This concept works for most, but not all applications.
In this second example I will show a slightly different approach. Here we'll have a look at a Rotary speaker cabinet, which is
usually known as a 'Leslie' cabinet and got misspelled in your QS as 'Lezlie' (for legal reasons, I guess)
By the way, before we begin spinning the speakers, some words to a general approach on FX programming.
I usually see the FX unit as a MODEL of reality (a real room, instrument, speaker,...). It tries to do the same as the original,
as good as it can. For achieving this, it usually is configured to replicate the STRUCTURE of the original.
If you were to design a model for a miked piano, for example, you would certainly employ a reverb unit.
Now we'd have a look at the available reverbs. Since the enclosed space of the real piano is quite small, the reverb setting
'large stage' may sound really fat, but since it models reflections coming from obstacles quite far away, it may not be the best
choice. 'Small room' would be more appropriate.
Usually, this 'physical modeling' approach works well up to a certain point. Then someone comes and says 'Hey, if I use the
'large stage' reverb, it sounds better' and maybe it really does, leaving you startled because you just left the concept of that
physical description you worked so hard to think up. Well, theory and practice...
Back to the leslie cabinet: For those who have never seen one:
Back in the good ol'days, maybe in the seventies, when rock'n'roll was made not by techno weenies but by real men with real
guitars (yeah ladies, and some real women, that's for sure...), rowdie wages were low.
Since every self-respecting rock band hauled around a 400 pound Hammond-organ, the additional weight for this Leslie
cabinet wasn't even worth mentioning.
But even today it may be well worth the drudgery, because these things sounded and sound GREAT.
It consists basically of a large wooden box containing an amplifier (tube or transistor) and two speakers.
The lower speaker has a diameter of about 15 '' (ca. 40 cm) and is mounted under a slitted drum that is - now the trick -
rotated by an electric motor.
The upper speaker is usually a one inch horn driver that fires into a perpendicular bakelite horn that is also driven by a
The motors are remote-controlled, usually by a foot-switch, that allows the organist (or sometimes guitar player) to set high
speed, usually low speed and brake.
The sound is - impossible to describe. Before I try, better listen to any Steppenwolf or Deep Purple record.
Now we want to catch this sound to our QS synth. Before, a warning:
Compared to a real Leslie or a good simulation, the QS Leslie is rather miserable. But it is still useful, especially for
background work on slow speed.
So now we'll examine what happens to our audio signal in reality.
At first there is the amplifier. Usually it's driven at full power or slightly overdriven. Some organists love to turn it up to the
max to get a real fat distortion sound, resulting in complaints from the guitar player that he can't hear his own playing
anymore (No, I've made that up!)
The amplifier powers the two speakers, whose output is guided and modulated by the rotating drum and the horn.
Now the sound leaves the box. Part of it is caught by the microphones, some will be reflected from walls and ceiling and find
its way to the microphones.
A leslie is usually miked using three microphones: one to the left and the right of the tweeter horn for stereo, one at the
Our QS has a 'model' for a leslie cabinet that has a mono input (two mono inputs, to be exact) and a stereo output.
A possible fx routing: (FX setting lezlie/overdrive)
The output of all the organ voices is turned off. This is especially important, since a real leslie cabinet has no way for the
sound to bypass the rotating drum/horn. As you see, we try to reproduce physical reality.
A drawback is that the overall volume will reduce.
Maybe you can have a look at the plan of this FX setup in the manual now. Trace the signal wires.
The signal from all organ voices is routed to the overdrive. It is set to a convenient (more or less distorted) sound. The output
from the overdrive to the A/B outputs is turned OFF!
The overdrive output will be connected to the reverb unit and to the lezlie effect block.
The reverb output is set to a low to medium value.
The 'Lezlie' output to A/B output is set to max (99).
Finally, a foot switch is set to modulate Lezlie speed, so that you can switch speed with your foot.
Aditionally, the motor on/off modulator can be set to the modulation wheel so that you can brake the 'Lezlie'.
This setup approximates the real organ/Leslie combination quite well: The distortion occurs before the rotating speaker, and
there is some reverb (although there is no Leslie effect on the reverb, but this can't be helped)
Maybe the problem will arise that the stereo effect is very present due to the nearly perfect separation the two 'microphones'
in the Lezlie effect block. This can be helped by opening the overdrive output to A/B to a low level, blurring the gigantic
stereo image somewhat (and giving away a little bit of our previous 'physical modeling' approach, since the signal can't leak
there in reality) .
We could also 'leak' some signal over delay or chorus.
Well, so much for an introduction to basic concepts. I have tried to give a general insight into 'FX engineering' and some
background to aid you in making fundamental decisions.
B) General knowledge about programming the FX unit
You can imagine the FX unit of the QS as a 'black box' that has FOUR! inputs and two outputs.
The inputs are labeled EFFECT SEND, or EFFECT SEND 1,2,3,4.
Please close your eyes for a moment and imagine a black box that has four jacks on the left side where you can plug in a
microphone, a guitar, a bass and whatever you think up at the moment that has a plug and makes music.
It also has two jacks on the right side, where you connect your P.A. amplifier with its left and right channel.
Now this black box contains a reverb modeling the previously mentioned echo chamber, but also the spring and plate reverb,
and reverb simulations of rooms of several sizes. It is possible to use several reverbs with independent settings, allowing for
example to use one setting for the snare-drum, another for the bass-drum, if you connect a (miked) drum
Then, there is a delay, a ping-pong delay and for the delight of any true 'heavy', a multi-tap delay!
Additionally we have several choruses, flangers, phasers, a cavity resonator and a leslie.
This image in your mind is a good starting point for understanding the concept:
-It has (up to) four inputs, but they are no 'real' jacks, instead a signal of any voice can be wired internally to one of them.
-It contains several effect blocks (reverb,chorus,...) that can either be connected to one of the inputs as to the output of each
-It sums up the output signals from the fx blocks and sends them to the output jacks of your QS synth, where they are
combined with any signal that bypasses the FX unit.
This concept allows a great deal of flexibility. It is also good for a great deal of confusion.
There are several effect configurations available. Each includes a different selection of FX blocks. Some use all four inputs,
Please take your manual and look up the routing diagrams for the FX unit. Don't let the many lines confuse you!
It shows what is inside this 'black box'.
The signal flow is generally from the left to the right and usually upward.
On the left side, there are the four inputs.
In the middle, there are the FX blocks, some signals splitting up and some points where signals are joined.
On top of all this, there are two 'rails' where the output signals from the single FX blocks are combined.
At points where signals are joined, you can determine the mixing ratio (set to 0: only signal A, set to 99: only signal B,
anything in between: both mixed)
In the top right of the schematic, there is the equalizer that allows to boost bass or treble (of the signal bypassing the FX unit
as well). Its output is sent to the A/B outputs.
Remember: usually, only a fraction of all those many lines is used. Most of my own FX configurations don't even use all the
Since there have been several questions concerning 'FX send' and 'FX bus', here an explanation
'FX bus' Another name for the input number of the whole FX unit.
There is FX bus 1,2,3,4
By choosing the FX bus you can determine to which of the four inputs of the 'black box' the signal is routed.
'FX send' Could be used in the same meaning as 'FX bus' (FX send 1,2,3,4), but it might also stand for 'FX send level'.
Depends on the context.
'FX send level': Here you can choose with which level the signal is applied to the chosen input.
If you set FX send level to 0, this voice will not contribute any FX input. If set to 99, it will be strongest.
Remember that you can turn off program output, when you don't want any signal to bypass the effect unit.
An additional source of confusion is the connection between effects blocks and buttons in FX EDIT mode, especially when
there are duplicate blocks.
The designers put the available FX blocks into groups.
Flanger, Detune, Chorus and resonator, for example, are grouped as PITCH.
The various delays (Ping-Pong, Stereo,Multi-Tap,...) are grouped as DELAY.
Reverbs are in 'REVERB',
Overdrives are found under 'OVERDRIVE', the Leslie simulator under 'LEZLIE'.
These group names are printed over the buttons.
Each effect setup has none, one or several blocks of most groups.
If there are several blocks (reverbs, for example), they will be assigned to different inputs.
You access them by pressing the REVERB button, then SEND1 (00), SEND2 (10) ,...
C) First steps / Experiments
Take any sound in program mode. Turn off all voices but one. Set the output parameter of this voice to OFF (you have the
choices R/L,AUX,OFF, don't mix it with the volume setting)
Play and listen, you should hear now the pure FX out signal (usually reverb, it will not sound very beautiful but that doesn't
Now switch to edit FX mode
Press the button labeled 'MIX' (120). Flip through the pages and turn off the output of all the FX blocks until you can't hear
your playing any more.
Since there may be some duplicate FX blocks for each input, you may have to change the block by pressing 00,10,20,30 and
again switching pages.
Turn that last output parameter up again so that you hear only the output signal of one single FX block.
Listen to the different reverbs, choruses, flangers, phasers and what else lies hidden in your QS, but isolate the sound so that
you hear only one block at a time.
Change the FX Input nr that you send your signal to and try to route a path so that you hear again something
Try to setup a routing so that a signal goes through two or more blocks in line.
(You have to establish a connection between the blocks and turn on the output of the last block.)
Use several inputs and assign different effects to them.
When you think, you've got it and start to do serious work, make some copies of the manual page with the FX setup. Maybe
redraw the intended connections with a pen to visualize what you aim for.
Basic tip, applies to ANY programming: if you change a parameter and don't hear the difference, always restore the old
value! (Unless you're sure of what you're doing)
C) General knowledge about managing progs, mixes and FX
The basic part of a QS 'sound', both in MIX and PROG mode, is a VOICE.
Each voice refers to a sample that is playable over a range of keys.
A PROGRAM consists of one to four voices.
The 'True Stereo' piano, for example, is split into two keyboard regions:
Voice 1 and 2 produce the left and right channel, Voice 3 and 4 replace them on the upper end of the keyboard.
A Program offers the possibility to determine to send HOW MUCH of the voice's output to the FX unit.
This can be done INDEPENDENTLY for EACH VOICE.
A MIX consists of one to sixteen programs.
The FX setup is stored in the program data. Each program has its own FX setup.
Since MIX mode involves usually more than one program, one of them must be named to the QS that provides the FX unit
This is where the main problems begin because now several programs have to share only one FX unit, requiring some
Each voice in each program sends to one FX input at a determined level.
(It is possible that more than one voice sends to one FX input, but one voice cannot send to more than one FX input)
Now in MIX mode, these settings may no more be valid, because the prog has to cooperate with another prog's FX setting.
A possible workaround is to alter all the involved programs and store them somewhere. Sometimes this may be the right
choice, because different voices can be assigned different FX inputs, but usually, it's too much work
In my experience, musicians usually don't care WHY it works as long it sounds good.
So a MIX setup contains the possibility to override the FX input and FX send level settings. If these parameters are set to
'PROG', nothing will change. Otherwise, all the prog's voices will be re-routed.
D) Handling of the FX setup:
Since it is a lot of work to create a well-sounding FX setup, it may be useful to make a special user program, that contains
only FX settings, because the voices are turned off. This program - or a copy if it is to be modified - is now included in every
new mix and the 'FX chan' parameter set to its channel.
Maybe making some notes really is a good idea.
Try to find a factory preset that has a decent FX setup. Put it in the mix, simply disable it (or delete the keyboard range)
Some (hardware) reverb units implement a 'test click' function. They offer a short, percussive 'standard' sound to test the
reverb setting. It may be a clever idea to include a drum sound for this (sticks?) .
Remember, the QS offers functions to copy the FX settings alone (somewhere in STORE).
You can store the FX setup in your sequencer if you dump this sound via MIDI and set destination to TEMPORARYx
Simply put it at the start of your sequence.
The FX unit contains two modulators.
You can, for example, toggle Leslie speed with the sustain pedal (but turn the voice's reaction to SUSTAIN off in program
mode or SUSTAIN OFF in MIX mode under Keyboard/Midi, otherwise notes will suspend)
Some parameters can be modulated, but create strange sounds when a signal is present. Either leave them be or modulate
only when all voices are off.
F) The exclamation mark !!!!
Sometimes your QS shows an exclamation mark in the display.
This means that internal signal levels in the FX unit are reaching their limits.
The result will be digital clipping in the FX path (it usually sounds harsh, but maybe it's desirable - there are factory presets
which produce exclamation marks)
Since it's a numeric effect that occurs in the signal processing, nothing will be damaged inside your QS.
To get rid of it, simply reduce the levels of all involved programs by the same amount. Repeat this until it appears no longer.
'Most of the information contained in this text may be correct.'
Since english is not my native language, (I already guessed that you had noticed) there may be some involuntary jokes in the
text. Please explain so that I can have a laugh, too.
If you think I'm talking bXllshit somewhere, please mail me, also, if you have some additional ideas or material to include.
My adress is
- Re: QS effect unit basics sam 9/15/98