Posts by AtonyB

    I was just thinking today how fantastic it would be if i could set parts on multi mode to 'midi out parts'.


    Same key range controls


    Same patch settings (i suppose the virus would sent the patch changes upon change of multi, or when you edit it)


    Only difference being the virus controls other synths based on these controls rather than originating the sound itelf.


    Random example; part 3, set to midi channel 2 between G2 and F#3, sends midi out messages on channel 1 (using the MIDI OUT port).



    It would be even better if it were possible to enable the AUDIO IN on the same part so the virus can process the generated synth sounds if you connect it up.


    This would save me endless trouble with midi mergers and splitters, etc. if i could just use the virus for it all.

    I would like to stress how cool it would be to have extra analogue filters on there, though. Something a little more mellow than the Moog type sound - im thinking ARP Odyssey, here (not mk2).

    Noop - i guess they just have a tap right before the DAC for output 1 that goes to an S/PDIF device (the specific architecture of which i haven't a clue)

    You are entitled to you preference, however as this is my field of research - i will say with little doubt, there is nothing about analgue filters that cannot be modelled - they just aren't that complex. True enough linear techniques aren't the way to go about modelling them, but for a long time (and to a certain extent, now) linear techniques have been all that was available in an affordable system or any system at all.


    But I'm quite happy for the industry to have their doubts - and i know that a lot of the industry have their interests lying in people believeing true analogue tones are the special realm of analogue circuitry - this is fine. It is fine because when it is finally achieved (and im sure this will happen gradually over a series of generations) the competition will be few and far between...


    Delay, phaser and chorus effects have already been well covered as linear techniques can easily model these (mainly because they use sampling techniques anyway) - though im sure they can be adorned with more modern modelling to enrich the tonality.


    I'm sure you aren't to be convinced, anyway, so I'll cease my efforts in the forum, but just remember for later on...


    I told you so...

    You can use S/PDIF out and output 1 at the same time - and they are identical outputs (ie you will hear the same from both)

    People forget that the trick is not to perfectly replicate the interaction of an analogue filter - but to provide a program complex enough to fool a human ear into not being able to distinguish.


    The human hearing system is not as complex as people would like to think, and one day, possibly not too far in the future, electronics will be powerful enough to provide such complexity in a practical environment...

    Gimme a week or so with them and i would be happy to produce a piece of software to emulate them - it would be a challenge, however, to get them streamlined enough not to require a significant enough portion of a DSPs MMACs....

    In fact, the ONLY realm of analogue left to be conquered by digital is in saturation/distortion, where the interactions become highly non-linear, and this is where vast amounts of MMACCs and bandwidth are needed to pull up the model for various different situations

    and I introduce you to the Nyquist-Shannon sampling theorem, on top of that, if you look at the SNR of sampling quantization of anything above and including 16-bit, it is well beyond the realms of human perception. Thus you have a finite problem, thus there is a discrete and digitally programmable solution.

    as far as i know, it will switch to 48kHz automatically if that is what your sequencer is running at...


    The SPDIF runs as an alternative to output 1, that is, the same audio comes out both - output 2 and ouput 3 remain the same as analogue outputs

    Yeah, even cds at 44.1kHz provide frequencies above what humans can hear - people who can hear above 18kHz, let alone 22kHz, are few and far between, and these frequencies provide little in terms of aesthetics...


    Fact is that human hearing has a finite bandwidth, and is in many ways discrete, and certainly has an SNR limit well beyond the SNR of the subtleties in an analogue filter that may be missed compared to the volume of a mix as a whole (if you treat the rest of the mix as noise). Esentially im saying in any produced track the subtleties would be drowned out - and, since nearly all music listened to is stored digitally, and as mp3, these subtleties are discarded by being rounded off anyway.


    A little back on topic, however, I would love to see (or hear) an analogue filter inspired by the ARP odysseys mellow low pass - It really is an elusive tone to replicate....

    Firstly, yes a dedicated USB card is advised (but not crucial if you are light on USB devices).


    You want to check your ASIO settings - see how small a latency your system can cope with...


    Also, there is a 'live mode' button in the bottom left of the VST - if this isnt activated, you will find that clicking it will improve things a heck of a lot....

    The separate USB device is a good suggestion - i think you might need to divulge a little more of what you are trying to do....


    Are you using hardware outs or USB outs?


    Your clicks and pops may be buffer underruns in your ASIO settings, check them out...


    Or do you get it on really simple projects, too?