Hi, Long time since I got back to music... I bought a Macbook pro retina 15 2014.
I downloaded updated and installed TI software.
When I open vst or au in Ableton. I have the same message/problem... The TI is connected to one usb port (only device).
Definitely worth a look by the guys at access. I am pretty handy with computers but I have no idea how to fix this...
My TI worked for years on my old Q6600 Windows...
Is there already a (simple) solution???
ahaaa... tnx
Well then they have to spend more bits and by
tes... however that works I don't know...
Hey,
When I was programming. I stumbled many times into some limitations. For Example I like to modulate in the filterbank the freq knob of varislope in the matrix or a lfo. The knob is not in the list.
Or like modulate chorus rotary speaker knob: distance. Also not possible.
Isn't it possible to put all knobs that exist into those assign lists?
tnx for replying. And give me a little more insight into how virus calculates 
Hmm... yes technically it will give headaches.
How about some new routing possibility's ?
like:
Use part two to mix with part 1... What I mean is... Making layers... But choose to send audio to stream to post fx of part one. Then you can layer the sound of part 2 but use the same fx's like reverb r delay of part one...
or maybe post it to filter section of part one...
Look at it as a chain... OSCsection... Filtersection...FXsection From the point you post it to another part... the rest will automatically go OFF...
Think about it... you can sacrifice sections... to use the same distortion or reverb room like the first layer. You will even safe calculation power. Cause other wise if you want the same effect used in layer two it would be a pain in the a** to copy it...
not even talked about tweaking... it really would open doors
Really it would be great idea...
Because every time I tweak cutoff with a higher resonance. I cannot bring it instantly to the perfect digit number thats on the screen (with the original value)
At a higher resonance the cutoff point could be set to a specific frequency which gives that specific colour... A good reason to install this function. Click and bang perfect back on the hotspot 
i.e. it doesn't instantly revert the controls - they take time to get back to the original settings and you can set that to instant or a long time (like portamento). Perhaps we are even wandering into an A-B fade type setup, even...
Yes... good idea... a sort of portamento or glide function... lets call it the ''smootness'' degree
Perhaps even some 'elasticity' to it so you can get some cool transient effects going on?
Explain...
switching to another preset and back again resets the patch with two clicks. doesn't mean your idea is bad though.
best, marc
I know But take for example a Trancy Acid loop is playing with ARP or via Sequencer. And (for example) you start tweaking some knobs like Delay time or Filter Decay or Filter cutoff or Freq shifter... to lift up or lift down the music or something like that...
And when all knobs are far far away from original point. Throwing it back in or out with a simple click.
I know Virus remembers the original knob value, because it is shown in the screen when you start tweaking like [55] or something This also was a great idea to tweak it back to original position. But that's just another effect you see?
I'll give another simple example: cut the smoothly away like with lo pass filter (silence) ... and 1, 2, 3, 4 BANG back in like it was
There would be a ton of effect possibility's possible
And Maybe it could be arranged that in config menu there is an option to choose how panic button will respond: 'cutting notes' ot 'reset to patch default'
And last but not least. In some kind of way it is a sort of panic option too. Because when tweaking something to much with to much knobs and you can't get the sound back as it was in time. You start to panic 
just click and everything sounds as normal 
Just trying to sell it to you, Marc 
When tweaking knobs live. It would be a great feature to use the panic button to reset all knobs to original patch levels after you got way to insane on the knobs
Instead of cutting notes. It's fast... and this way your sound will get back in the mix like it was before by a simple click
This would be a very useful TOOL on stage. Don't you think Access?
recursive mod is good. But when I look to DECAY it isn't linear drawn. It already has some character with it. It would be maybe better to make it linear to and then if you want a convex or concave one. To start from a linear one.
I Also would be very happy if they could be shaped in vstmode
Don't get me wrong... I already am very happy... Just try to brainstorm a little here...
...oh, I simply wished the patch names would be in a larger font... most of the time I need a magnifying glass to identify the names...
Pick a lower resolution for your screen Or Buy A large TV screen for your computer :p
Hah... 96kHz.... I direct you to the Shanon sampling theorem.
The only benefit of higher sample rates lies in noise handling and since it's a digital synthesizer that is not an issue. But I suppose hard drives are pretty big these days, so if you want to enjoy the placebo I won't stop you...
the Heisenberg uncertainty principle is very annoying, but still true, deal with it.
Maybe a little late to react but...
I had many discussions about this... Today I recorded a Sine Square Saw and triangle. Playing 10 octaves of note C. Each 1sec. I recorded at 44.1kHz 16bit at 44.1kHz 24bits at 88.2kHz 16bits 88.2khz 24bits.
Guess what... I should post you a foto about how my waves looked likek at not c6 c7 and c8. Dawm... If my speaker moves like that ugly thing on the screen. I think I start to realize why some people don't give a sh** about
Polyphony. Some people just want to bounce things like fill in effects and things like that. And yes the higher the frequency goes the more artificial is sounded at the higher samplerate 88.1 24 bit.
Do the experiment and take a good look for yourself. Frequencies start getting real alien like around 8000hz
I never use all 16 parts... And everything I do not tweak live I will bounce to make a loop for using with apc40. So... If possible... it still would be worth asking if the feature would be possible. Even if we could bounce one track a time... it would be worth being an option
Hey... I am just a beginner... but what I see is whatever wave I played, it would be a monster very soon at some higher pitch...
44100kHz only gives 8 measuring points at one cicle: 5512Hz. Try some complex waves And guess what... Those numbers like 192kHz are not just propaganda... Yes... it is truth... we need atleast 2 measuring points for drawing everything a human can hear (22.5kHz)... really think about it... how can you draw a complex wave like the most simplest sine with two measuring points? How can you draw a wave at 5512Hz with only 8 frames? Lets say 20 frames? at 2205Hz?
No no... I do believe that theory... 2points for each cicle... It doesn't make any sence how to pick two points in time to measure amplitude while they are (analog) infinite to measure...
I take a look to what my scope is telling me... and it says... MORE, MORE samplerates Need more... 44.1 is OLD stuff... Take a look around... Artists are selling now on USB sticks... No turning devices any more... Solid disks... with 4gb or so... 44.1 16bit is old. And music is getting on stick with more Mbites to spend for quality.
sory for my bad english :p
