Posts by Merlin

    Indeed, on many softsynths curving an envelope stage is a parameter in the envelope itself. By using the modulation matrix, the virus offers some flexibility though:

    Modulating, for example, filter decay rate by the filter envelope allows for snappier decay rates, but negative modulation also allows for more convex behaviour. So positive or negative modulation allows for both concave and convex curves.

    On top of that, every source in the modulation matrix can be used to modulate the depth of another slot in the matrix. In the above example, one could use a tempo synced lfo which modulates wether the decay rate modulation is positive or negative, allowing for a tempo synced change from convex to concave decay slopes.

    Also, lfo' s can modulate their own modulation depth, their own amplitude, their own speed, etc. etc.

    By using the modulation matrix for recursive stuff an enormous amount of flexibility emerges since such an approach is more generic than using dedicated parameters.


    lfo 1 is modulating the depth of lfo 2 which modulates speed of lfo 1 which modulates depth of lfo 3 which modulates both filter cutoff and lfo 2 and 3 speed.

    On the virus the above insanity is possible. On most softsynths it is not.



    Indeed Ableton will not render the virus portion of your song and this has a simple technical reason: When ableton renders a song, it makes all calculations on all tracks for producing a wav file. Making those calculations is not a problem for abletons effects and vst's but the Virus makes it' s calculations in the virus hardware. This cannot be controlled by Ableton, so the only way to "render" the virus is by recording the usb sound streams as a wav in realtime.


    The Virus can give you up to 3 seperate audio streams through the usb cable. To obtain those streams:

    -create an audio track
    -Every Ableton audio track has an audio input: set the input to the virus host track. In other words: your audio track draws the audio from the track which hosts the Virus plugin.
    -Under the audio input, you'll see another combobox: select the second (or possibly third) virus usb output.
    -On the virus plugin, make sure that a part dumps it's audio on the corresponding usb output, you set this in the common page.



    throwing something out is not the big issue but predicting which people required legacy destinations in the past and want to use it in the future

    If I had to do this I would write a small parser program which would extract which source, destination and combo' s are used inside one patch. Once that program is written, I would encapsulate that inside a bigger program which eats one midi file of patches and returns how many times which source, destination and their combo' s are used.

    After that, I would collect all patch banks which exist on the Access website, commercial banks, user banks, etc. ,categorize and analyse them using the above program.

    Finally, you obtain a lot of plain text files full of numbers on which several simple shell scripts are run which tell you, for example:

    -Which mod destinations are used most throughout the years and whether changes/trends occur after the introduction of new os versions.

    -Are some destinations used more in basses than in pads or other categories and is the category system used frequently int the first place? If so, what category of sounds is built most on the Virus?

    -Do people from Access music have habbits in using specific source/destination combo's and are these different from what commercial sounddesigners and/or artists outside Access use? Has this changed over the years?

    May other questions could be answered using the above approach and although this still makes it hard to predict what people will do in the future, it at least gives you some insight in what has been done in the past by various categories of users. Over 5000 patches is enough to justify a statistical approach.

    For writing a parser program, a functional language like Haskell and the parsec library could be used. For plotting statistics in graphs, perhaps good old gnuplot. All available fo free :)



    Download and read the manual and look for the shape parameter in the lfo menu. If I recall correctly, the shape parameter turns a sawtooth lfo in a more convex or concave shape. The shape parameter also has effect on the other waveforms so read the manual for details and experiment.



    I can't help you with the emx specs, so you are your own with that one, but asumed that it can sequence 5 different midi channels, that is is indeed one more than the virus snow can deliver. The snow has 4 parts multitimbrality, so it indeed can play 4 different sounds simultaneously at max. The TI desktop models have 16 parts and have two dsp processors instead of one for the snow.

    Reading about your dilemma, I think you should think a little bit about how you are going to use that synth. To put it a little bit rough: are you an in depth programmer or someone who wanders through presets and adapts a found sound to his own needs. This is important since the virus has an insane amount of parameters to mess around with. If you look at the desktop model, you'll see that the the synth is divided into sections like oscillators, filters, fx, etc. Each of these sections has an edit button and once you press that, you enter the extra parameters of that section. There are at least (roughly) 15 extra parameters per section and most of the sections have a lot more...

    So if you are going deep with programming, you really want those extra knobs of a desktop model. I would therefore advise you to save and get the desktop model. It has roughly twice the processing power of a snow and those knobs are welcome if you are a programmer. Even if you don' t want to save I would still recommend a second hand TI1 desktop over a snow since (in Europe) a second hand ti1 in mint condition is about just as expensive as a new snow. I have owned a TI1 desktop for several years now and the raw power that thing is already a beast in it's own. I never considered updating it to a ti2 since it's dsp power is already way more than I ever need.

    In my view the only reason for getting a snow is when portability is an issue or when you are programming the synth mainly through the virus control software from a daw. I am not saying that Access did not work hard to make all the parameters of the virus easily accessable on the snow, but the intuitive layout/more knobs+more dsp power+more multitimbrality of the desktop models is something to consider.



    Hi and welcome :)

    Ok, I' ll give you a push, but:

    >This is not THE method! Other methods will probably work as well. I don' t even know how the original was built. What follows comes from my own experience.
    >This is not a complete walkthrough, but a push in some direction.
    >I am not in front of my synth right now, so this comes from the mind ;)

    -Start out from an init patch

    -Set oscillator balance to the full left so only the first oscillator is heard. You should have a clean single sawtooth now.

    -Change osc 1' s waveshape so you end up with a pulse wave. The Virus can morph between sine->saw->pulse.

    -Setup a sequence of chords in your sequencer. The chord should consist of 4 notes: first note should have fairly low pitch. Somewhere between c1 and c3. Second note is 5 semitones above the first. Third note is 8 semitones above the first. Fourth note is a full octave (12 semitones) above the first note. So: Base+5+8+12. Possibly add a fifth note one octave lower. Experiment with that.

    -Tweak the filter by closing cutoff frequency somewhere around the middle and giving a mild amount of resonance.

    -Increase filter envelope depth to taste.

    -Experiment with the above three parameters and, if needed, the decay rate of the filter.

    -Change the pulse width of the first oscillator so the pulse wave becomes thinner.

    -Experiment to taste with:

    *oscillator 1 pulse width
    *oscillator 1 waveshape

    *filter cutoff
    *filter envelope depth
    *filter decay rate

    Once this basic sound is set, you can enhance it further:

    -Add some analog boost for a warmer sound
    -Add a little bit of distortion for some more rawness
    -Use the eq to emphasize specific frequencies and beef it up even further
    -Dial in the second oscillator, off course set to pulse and perhaps detuned or synced, one octave higher or lower...perhaps in combination with sub osc?
    -Unison can be used as well, but since this is a fairly raw sound, you should avoid things that make the sound wooly/smudgy/blurry/vague. So avoid reverb, detuning, chorus and unison or use them subtle.




    Ummm, Merlin, that is incorrect. Unipolar does not have an inverted half-cycle, it's applied as an offset.

    Indeed. Just tried it and watched the bootcamp video nr. 10...I stand corrected ;)

    Fairly easy:

    Since you understand the definition of unipolar vs. bipolar, try some simple experiments to see what it can do:

    -Set up a simple sawtooh patch and put filter 1 cutoff in the middle (=64). Also add some resonace. Now go for the modulation matrix and let lfo 1 modulate the filter 1 cutoff bipolar. Keep it simple and predictable so set lfo 1 waveform to sine or triangle. Play and hold a note and now you should hear the filter open and close fully. That is: it moves filter cutoff from fully closed (0) to fully opened (128). Once you put the modulation in unipolar, the negative part of the lfo cycle is inverted and therefore also positive. Result: the filter doesn' t close anymore. Instead it moves from 64 to 128 twice in one cycle which allows for a different rhythmic structure.

    -A practical, little more complicated example and walkthrough:

    Start from an init patch and set the following parameters:

    -set both oscillators to sinewave
    -Set fm type to wave
    -Raise oscillator 1 pitch with two octaves
    -Set oscillator balance to 128, so you only hear the right oscillator.
    -Set oscillator phase to 1
    -Set detuning of oscillator 2 to 0 (=no detuning)

    -Play a couple of low notes and twist a little with the env->fm amount. That parameter controls how much the filter envelope injects fm into oscillator 2.

    -Experiment with env->fm amount, fm amount and the filter envelope parameters until you obtain a classic fm bass sound.

    -Turn on the arpeggiator and configure the arp so that it generates a nice arp.

    -Time for some modulation:

    -Move into the modulation matrix and and let lfo 1 modulate env->fm amount or fm amount or both.
    -Set lfo 1 to tempo synced and choose a speed that is rhythmic, for example 1/2, 1/4, 1/8 or something.
    -experiment with different lfo waveforms, the shape parameter and unipolar vs. bipolar modulation.

    Once you play and hold a few notes, the arp keeps repeating those notes and the tempo synced lfo injects various amounts of fm into the signal. Since the lfo is tempo synced, this should sound fairly rhythmic and changing from uni to bipolar gives different variations.

    If you are still reading, enhance this patch a little further:

    -set lfo 2 to a tempo of 1/6 or 1/3 or something like that.
    -in the modulation matrix let lfo 2 modulate the depth of modulation of lfo 1. You do that by selecting the correct slot destination.

    Now lfo 1 directly modulates fm amount, but at the same time lfo 2 controls how much lfo 1 affects the signal. Since lfo 2 and lfo 1 have different speeds, you have two "rhythms" in one signal. Again create variations by changing uni and bipolar modulation, tempo, lfo waveforms, shapes, etc...

    A final further enhancement to top it off:

    -Let lfo 1 modulate lfo 2 depth. Meaning: lfo 1 modulates lfo 2 which modulates lfo 1 which modulates fm.
    -Let lfo 2 or 1 or both modulate oscillator 1 pitch. This gives wild results with a lot of harmonics. If it gets too wild, use the filter to keep it under control.

    -There happens to be a third lfo...



    When it comes to documentation I think such a list should indeed be provided, just for reasons of completeness, although I can understand that Access has other priorities.

    On the other hand, when I want to modulate a specific parameter on the Virus I use a different approach:

    The modulation matrix of the Virus allows you to use a midi CC number as source. So if I want to modulate the cutoff frequency from an external sequencer I select, for example, midi CC 12 as source and filter1 cutoff as destination. This way of working has a couple of advantages:

    -You don' t need a list anymore, so no lists you have to search for in your studio and no data that has to be updated in the event Access changes something in their o.s. By (ab)using the modulation matrix you have become independent of what Access changes to to their CC' s.
    -Every destination in the modulation matrix can be used.
    -It always works in the same way, so no need to remember numbers.
    -Since the modulation matrix has 6 sources with 3 destinations per source, you can modulate up to 18 parameters with only one incoming midi CC. I have never gone that far though...
    -By using modulation intensity in the modulation matrix, you can define upper and lower limits to how far a parameter is affected by the incoming midi CC.

    Finally a small enhancement:

    Select a midi CC as source (say 12) and assign one of the soft knobs to the same CC number. Now you have a soft knob as source in the modulation matrix, allowing you to modulate up to 18 parameters with the turn of only one knob...


    Marc: Thanks! I' ll take your advice but I won' t stay away from using that technique...I have been warned though ;)

    I am willing to compile a soundset with some basic templates, but it will take some time to set up as I have to change several parameters per patch in order to get some coherence in the set and also I have to compile an accompanying pdf file which provides a small description of what each patch does, what is the way of thinking behind it and what to try and look out for/suggestions for enhancements.

    I will, however, only make this effort if such a set is actually considered useful throughout this forum. Compiling something for one person only doesn' t make much sense, considering the work I' ll put into it and in the upcoming weeks, time is not on my side.

    So...if there is some response to this thread and people are interested, I' ll do my homework.

    Is there something like a file section on this forum? Would be great to have for things like this.


    Modulating volume with a fast random lfo...hmm...doesn' t look that spectacular on paper, so let' s try that one :)

    On a side note, this whole recursive modulation issue was taken from the virus tutorial by Howard Scarr (p.26). Since he wrote it in a booklet that came with the TI, I thought it was a "clean" technique to use. Could you clarify a little under which circumstances it should not be used? Thanks!


    I have some hints here and perhaps a little advice on how to speed up some things in working with the Virus. First a couple of modulations:

    source: filter envelope
    destination: filter decay

    By letting the filter modulate itself, you obtain so called recursive modulation. The result is that the decay will now get a concave or convex shape which makes the envelope more subtle or aggressive. You can do the same with other stages of the adsr on both filter and amp envelope.

    source: velocity
    destination: wavetable position

    By making this setting, every time you hit a note, the velocity at which you hit the note will select a different position in the wavetable. By carefully selecting wavetable, initial position and modulation depth, you can bring melodies way more to life. For instance playing rough makes the sound more harsh while playing soft gives you a more sweet and civilised kind of sound or vice versa.

    source: filter envelope
    destination: wavetable position

    Every time you hit a note, the wavetable is swept. By carefully selecting wavetable, initial position and modulation depth, you can control how subtle or agressive the sweep is.
    The effect can be enhanced further by letting the velocity modulate the depth of the above modulation. Striking a note hard gives a broader sweep while playing gently gives you a fairly static note. This, again, can be enhanced by letting the mod wheel modulate the wavetable position.
    While playing you select which part of the wavetable is swept, while at same time the velocity defines how far the sweep goes.

    There are a lot more of tricks like this, but let me describe a couple of things which make life easier by giving one example:

    Start with a basic sawtooth patch. Now dive in the mod matrix and configure it in such a way that:

    -the first softknob selects the wavetable of both oscillators.
    -the second softknob selects wavetable position of both wavetables
    -lfo x modulates the wavetable position of both oscillators
    -the third softknob controls lfo depth for both oscillators in the above modulation
    -lfo x is selected.
    -set up fx detuning etc. at taste.

    Now save this patch to a ram slot which you can easily find back and call it "wavetable start" or something. Now you have created a patch with which you can wander through all the wavetables to see how they sound and what a wavetable sweep (using the lfo) will sound like.

    this patch can be used as a starting point for wavetable based sounds. Every time you want to search for a wavetable based sound, just load this patch as a starting point and use the softknobs for searching through those tables until you find what you need.

    I have many more of these starting points in my virus, for example a basic fm patch, a basic hypersaw patch. a basic tb303 patch, a basic spectral wave patch, a basic sawbased bass patch, and so on and so on.

    It will take some effort in the beginning to set up a collection like this, but it pays off in the long run as it saves a lot of stupid work.



    SPdif means Sony/Philips digital interface. It is a connection standard developed by those two companies. SPdif is a digital standard so sound travels through the cables in a digital way. This means that by using spdif, you are bypassing the digital analog converters inside the ti itself. So using spdif gives you the cleanest/purest sound the ti engine can come up with.

    spdif is both input and output, so you can route an external spdif signal into the ti, just as you would when using the analog inputs. As far as I know you cannot use atomizer on the spdif input but for all other things like filters, fx, etc. it works fine.

    Since spdif is digital, both units that are connected to the spdif cable have to be synchronised. Usually one of those units is the master who sets the samplerate and the other one is slave and adapts to the dictated setting.

    Just type spdif on English wikipedia for a full explanation on that standard.

    There should be a configuration menu in the software of your soundcard which gives you the possibility to change whether the sound card is master or slave and at what samplerate data is transmitted/received. The same goes for the TI.

    So, for correct usage:

    -Put samplerate of both units at the same value.
    -See to it that one is master and the other slave.


    Hello everyone,

    When the virus forum closed down a couple of years ago, I was known as Merlin. I learned a lot back then and I hope this forum will be a good source of information and tips 'n tricks.