Posts by TiUser

    Well, for starters the filtering you are talking about is only for ADC, not DAC - so it's entirely independant of the virus.

    Maybe true with a filter for ADC but I wasn't about that. I was about reconstruction that needs a lowpass to avoid the backfolding issue evil from digital domain.

    Second of all - using a decimating input (sima-delta or whatever) gives you a really high bandwidth to start with, now you can filter that digitally or with an analogue filter - incidentally 96dB/octave wouldn't be a great effort. Analog Devices quote for one of their off the shelf AC'97 CODECs a pass band (+/- 0.09dB) of 0.4*fs (for 44.1kHz that's just shy of 18kHz) and a transition band from there up to 26kHz. (-74dB at the nyquist point - full scale sinusoids at just over nyquist point could only occupy last 4 bits, by the way thats over 150dB/octave, so the aliasing could only occupy 2 bits at best by the time you ramped up the frequency of the sinusoid so you could hear its aliasing). 48kHz makes aliasing disappear underneath quantization noise by the time it folds back into audiable territory - and this is with a full scale sinusoid, which would be unlikely to occur ever.

    Not sure if you try making things much more complicated to cover that you are not so sure - or whatever?
    I also didn't say it's impossible to get closer to what I showed up with as an example to make the basic effect clear - I just said you can't simply expect the theoretical limit the theorem promises... or a great construction from the two magic numbers.

    Also, realtek, just because they quote 24/96 it doesnt mean you're actually getting that resolution - its like when they stick 12MP digital camera sensors in mobile phones, the lens instrumentation just doesnt have that kind of resolving power. BUT that's the standard sensor they are flogging, so they stick it in anyway because it's cheap (especially if you are sony and you are making millions of them).

    Practical numbers are fun...

    Oh, now you contradict yourself a way, referring to real hardware and "practical numbers", that's indeed funny. :D

    I do not like to jump into the digital camera thing here - also there are some parallels - and some even more complicated other things ....

    That's somewhat silly to suggest using a low screen resolution...

    Despite the fact that the digital like font isn't really a joy to read imho the design might be for a 800x600 screen or so and even small for this... So you want me to use my full HD screen running in 800x600?...

    I think my point is more likely thinking of a redesign and going for a pixel independent gui that can be scaled without any side effects... :rolleyes:

    If VST technology could handle this is another story I do not know. :S

    Recursive mod eats up one modulation slot in the matrix.
    It is also not an intuitive way shaping the envelopes - it's a pure technical approach you have to know about.

    Why not using multi mode to layer sounds and sending on two midi channels - that needs no new cumbersome features.

    I guess reusing the effects via the routing you suggest will not work - as thanks heaven effects are part of a patch in a virus and not independent, mixer like routed stuff... which will create many headaches making patches sound the same in any mode.

    I am really surprised that this is an issue to some. This is for sure not at all a bug.

    It is a remarkable feature that the previous patch sustains naturally even the next one is already selected. Many other devices and most softsynths can't do this. It's also the natural behavior of all acoustic instruments - you need to damp them explicitely when you want something else. IMHO not having this feature is really troubling as workarounds for not having this are always cumbersome.

    If this is really an issue to anyone that must be sounds with insane release settings, mad reverb or whatever...

    Well, there's always space for another option... :D

    But please no harsh cutoff then but a strong forced eg damping... :whistling:

    ...the more I think about it the more I believe Access should probably offer limited signature mod versions...

    ...and don't forget the diamond option... (diamonds are forever) LOL :D:D:D

    Mark... does Access consider producing this? LOL, but there are always people who like such things, so well... maybe it's not at all such a bad idea... just take care to make it a limited signature edition... :D:D:D

    But indeed that's the kind of thing OpenLabs already did! A housing with real gold finish... seriously!!!

    It's about a $50.000 item if I remember right... LOL... There was also a competition in Texas where a golden Open Labs keyboard was #1 prize giveaway... OpenLabs now has a special section for custom modded gear - and you can easily spend more for mods than the whole thing... heck just I miss the diamond option... impossible to get one without that. LOL :D:D:D

    Honestly, I find that ridiculous but well, maybe I understand a s**tload of showbiz... ;(

    ...not satisfied with what i have? golden housing for a warmer sound? nowhere have i said that im not satisfied or that the virus is inferior in anyway. just because your elitist ego likes to assume things and talk down to others, while try to show how much you "know", says much more about you than it does about me. now fuck off

    You must have overlooked some smileys... and the golden housing was a joke related to OpenLabs gear... I'm so desperately waiting for OL diamond edition... with golden housing of course... LOL :D

    I hate posts without offering arguments. Pick up for yourself what is important to you. I am not interested in offending anyone nor to show off anything. But if I'll need a psychiatrist I'm going to remember your brilliant analysis... :D

    First of all I am not riding any high horse... sorry that I have to come back but I see you didn't understand my point.

    "so the 44.1kHz gives us a frequency range of 22kHz (ignore the extra little bit, thats a margain for nyquist point issues) leaving up to 4kHz transition, thats pretty healthy i'd say, im sure i could manage that with like 128 taps or something...

    The Shannon sampling theorem is not just theory - its pretty damn rigorous."

    You are quoting exactly just the theory. There is no lowpass that cuts the level at 22kHz with infinite slope down to minus infinite dB. You might know what a 24bd/oct filter is... do you? It lowers level 24dB per octave - that's again a simplification, but for the sake of an example - by now you have to built the DA thing that alisaing - which occurs by backfolding for frequencies above nyquist - gets unnoticeable. One octave lower than 22kHz is 11kHz, right? And aliasing is just damped by 24dB @ 22kHz, right? Still we can take a complicated 96dB/oct filter and still we are at 11KHz... great?... theory and real thing are different - there is still something happening between 11kHz and 22kHz affecting the sound. You do practically not get the theoretical numbers you take for so sure and approved... It depends a lot on the practical construction of the D/A system. That's also a reason why a good system on 44.1kHz can sound better than another on 96 KHz... it depends... and you are right - the latter can sound better because theory gives more headroom... but only if you are always committed to a great practical construction as well.

    Now to finish with, there is more practical inaccuracy than just filtering for the sake of nyquist... usually in the analog domain, like clocking stability. You can do some more tricks first in the digital domain before conversion but I guess this has not much to do with nyquist then.

    But I am really no DSP processing expert - It's just my personal experience with these things that make it obvious to me that these simple numbers are not enough. Why does a 24bit/192kHz built in Realtek soundchipset in a PC not sound much better than our Virus with poor 16bit @ 44.1kHz transported via USB?... Referring to your view the realtek chipset should sound by far superior - but strange, practically it doesn't... so nyquist must be wrong? No, it isn't, but it's the theoretical maximum approach only - and that's not so simple to come close to.

    Distribution media formats is a totally different story - I think we are talking pro audio and signals from a virus might be processed in a daw or host, mixed and degraded more or less so that any headroom during production is welcome. When I think of usual PA's I don't care much about all that in live applications.

    I also do not have sleepless nights - I personally do not need 96 kHz - as long as I use quality equipment on 4x.xx kHz.

    50% of these points are inherent to VST technology and are not TI specific.

    There is no latency free PC based ASIO/VST audio. Not even practically with buffer sizes of 128 samples - also that works somehow acceptable.

    It's maybe more obvious with TI because it can send audio to your daw or host and getting it back at the same time for audio out which doubles latency timing compared to your usual view on this. Additionally there might be issues with your daw or host a stand alone unit might not suffer from.

    You can minimize latency by using VC just for midi - not audio. So you can still use VC to edit sounds and put them into the slots but latency is as low as possible... which is probably a compromise for live applications.

    PC's with windows are not designed for realtime computing. Win Vista and Win 7 seem to behave even worse compared to NT because of more inefficient drivers running on highest scheduling priorities. See OpenLabs is still using NT and I am sure they know pretty well why they do not upgrade yet. And do not tell me there is no 64bit... there is NT 64bit too if you wish.

    There is a lot of features on pure midi arena I wished to have - not just octaves or chord memory. Unfortunately midi processing seems to be very unpopular to developers...
    What about these:

    • Automatic voice spread: Play single notes with 4 voices (unisono), two notes with 2 voices and spread cord notes over just one voice. So leveling a sound would be much more "band like"
    • Harmonizing: play a chord in real time and harmonize the melody line with additional notes matching the tonal space of the cord.
    • Cord recognition and transposing a total midi stream through a NTT (note transposition table)?

    Many cool things live performers could use beyond octaves, cord memory or boring arpeggiators.

    @ AtonyB:
    The shannon theorem is just one piece of the theory. All D/A conversions need some type of LP filter and it's hard to make a LP filter with endless cutoff slope... so in this regard 96kHz sampling rate can practically lead to a LP construction that is somewhat less critical and finally producing "better", "smoother" audio... but again, implying that 96 kHz conversion is performed with double clock precision as 48kHz conversion is a practical assumption where we have no proof for, do we?... and makers of converters are just too willingly fooling people with numbers that only tell part of the story. I also think that most todays converters are internally 1bit and everything else we were told is just math to give us the numbers we are familiar with...

    Heisenberg? Hope we'll find soon a theorem for good music too... :D

    @ sdrr
    It's hard to me to rate your knowledge or experience by just seeing your few short comments in this thread. I'm sorry that I sounded "teaching"... I've learned myself that there is a difference between theory and quality of practical devices. Numbers can quickly be misleading. Ever trusted in Behringers great audio specs?... LOL I hope you won't mind if I assume that probably other people without your experience might read this thread too... it's hard to make it right for everyone . so apologize. Coming back to the original question - I think TI with 96kHz is simply difficult to implement because of USB bandwidth restriction and therefore new HW is the only serious solution. Or would you be happy with just one stereo out pair at 96KHz coming from Virus into your host?