Posts by nms

    After the recent testing we did I wouldn't second guess the conversion quality of the analog inputs on my Ti2 as I once did. I don't need to use it normally but if I had to they're up for the task.
    Here's a link to the converter testing we're doing on gearslutz:…t-thread.html#post7153080
    Although limited in features and i/o, in terms of sound quality (and so long as you're not getting popping or glitching, which is system & OS dependent) the Virus functions fine as a basic audio interface. The AD conversion is sure more transparent than on my Steinberg MR816x!

    So you're setting your audio interface to 48k in logic.. and then trying to open the plugin? You're not using an odd buffer size are you? the newer OS versions are pushy about you using even ones ie 64, 128, 256,512.
    Are you using the virus as your audio interface or somethign else? I can change my interface's sample rate inside ableton's audio preferences no problem with the plugin already instantiated and no problems. You should be able to.

    Anyone using their Virus with an Apogee Duet? I'm trying to use the analog outs but they clip easily
    and in order for them not to clip I have to turn the keyboard or the Duet down very low but then I can barely hear it.
    Should Maestro be set to "instrument" input? Thanks!

    Sorry to say it guys but it's just a simple matter of RTFM and understanding your sound card's different input types ;)
    You want it set to line in. If You're ever experiencing that kind of volume siscrepency or distortion that's usually a good sign you have it plugged into the wrong place or have it set to the wrong input type.
    I have never even laid eyes on a duet btw.

    Ah ok i got it :)
    The final downsampling from 88.2 -to-> 44.1 must be done before or after the limiter/ditering?

    For anyone wondering about this stuff... for best quality (despite the fact that DAWs like ableton will allow you to just drop it in) convert any audio files in your projects all to the same sample rate. This would be your working sample rate. For most people that's 44.1khz, but a lot of the top pros work at higher sample rates like 88.2k or 96k. I prefer 88.2k as there are occasional quality benefits to working at an even multiple of your final product. 96k was more intended for video application where 48k is final format. SRCs (sample rate conversions) are unrelated to dither rules as dithering relates to bit depth conversions. You only dither at the point you are stepping down to a lower bit depth. So for example lets say your track is mastered and no other processing is to be done on it.. you'd SRC to 44.1khz and from there export your to the final 16bit format with dithering applied. So in regard to your final limiter yes I'd definitely do that at 88.2k as it'll perform better. The final 2 stages should be SRC then dither to 16bit.

    Yes. Your virus can only be sat to one clock rate at a time so if your project is at 48 or 96 then the Virus will generate at 48k internally whether usb or analog outs are used.
    Personally I avoid working at 48k and my preference is 88.2k as there are occasional benefits to working at exactly double the rate of your final product. The cleanest way for me to work it seems is switch my project to 48khz when I want to bounce audio and do it via usb, then use Izotope's 64bit SRC to convert it to my project's 88.2k sample rate.

    this really shouldn't be your problem its a disgrace..shelling out nearly 2 large for what amounts to shoddy software...regardless of how old or new your computer seems that its accepted for people to spend large amounts of time fiddling and tweaking and is this the responsibility of the consumer?...we want to make music..not give ourselves high blood pressure screwing over something that is either unsolved or unreleased..either way it amounts to a lot of money and a waste of effort

    Haha.. I would like to remind you guys the new OS is a BETA release.. making you BETA testers ;)
    I hear you with the crackling.. my Virus was crackling like a campfire upon installing this new release and making it work. I haven't had crackling from my virus in a LONG time before this.. and I ran a 96k buffer before which according to this new box that pops up is a no no? Plenty of crackling in the new one at 128k buffer though.
    I'm on win 7x64 and OS4.1 is the best I've used so far. I might try out the other OS4.5 beta pretty quick here though as this new one may be unusable.

    With the virus it will work identically to the normal saw, just in reverse. So it functions 100% as a typical reverse saw would. At maximum modulation it starts around 2 2/3 octaves below the root pitch and modulate up to around 2 2/3 octaves above the root.

    If you use in envelope mode the signal starts high, goes to the root, then back to the high point. Same thing with reverse saw.. it starts low, goes to the root then back to the low point where it holds. Nothing unusual here and everything operates the same way it would if there was a button for reverse saw.

    How frustrating, I've spent the last hour or so trying to find this article for you. My main computer has just had OSX re-installed so there's no history to look through & I mustn't have read this on my Macbook as it's not in any history there. I've found a few things on google but not the article I'm thinking about.

    It was saying something along the lines of some synths work behind the scenes at much higher sample rates pushing the nyquist frequency higher to avoid audible aliasing, then convert the signal back to the required sample rate at the outputs. Say for example you set your sample rate to 48khz, internally the processing would be happening at 192khz. I was going to post it in the forums here in response to a post Marc had made about sample rates, I'm not sure but I got the impression that the Virus was maybe a synth that did this.

    I'm sorry that I can't give you anything more solid to go on than a half recalled article I read, the paragraph I have written above may be complete rubbish - but I think I've got the main idea behind the article. If I do come across it again I will definitely post it here.

    No need.. I've actually written pages worth on the topic of sample rates and tested and discussed it to death so it wouldn't be anything new ;)
    What you're talking about is oversampling.. where a digital signal is processed at a rate several multiples of itself above to avoid digital aliasing and then downsampled to working sample rate.

    it does, at least where it really counts. please understand that i will not get into details.

    Why aren't you able to get into details on this Marc to clarify what it does? This is definitely relevant information! I had assumed it didn't since even jumping up to 48khz rendering gives an instantly noticable difference to the top end as I've shown here.

    There are most definitely benefits of reduced aliasing by going much further up in the frequency range than just 48k. In the jump from 44k to 48k you'll hear a difference from the filter at the top of the available frequency range raised higher to a level where it's no longer audible as well as reduced aliasing from some, but not all, aliasing being moved out of the audible range. From that point the filter is inaudible but aliasing can be reduced much further still. Try taking a softsynth that hasn't been rendered yet and using bounce a part or just a saw tone to audio at 44k, 48k, 88k, and 192k using your DAW's offline bounce/export feature and you will hear the results of that. Just get your hands on a top notch SRC like the Izotope 64bit SRC and downsample all the results to 44k after they're rendered (simulating the final output of the public release format). Plugins work better at high sample rates. This is why I work at 88.2khz normally.

    Admittedly I did play it off zippy share off some laptop speakers! Just seems to fly in the face of years of industry studies showing no human being on earth ever passes the blind listening test between 44.1 kHz and 48 kHz...but I see you're talking about something else: filters behaving differently due to sampling rate.

    LOL! classic. Man you aren't supposed to do critical listening in such a manner!
    It sounds like you are confused about studies and people not being able to tell 44 vs 48k though. I don't know that I've heard of these studies but if they are legit they are only true for particular circumstances.. like the recording of an analog source, not digitally generated sound via digital instruments & fx.

    The reason for the audible difference is that with the 44khz audio format the filter that bookends the available frequency limit stretches down within audible range whereas with 48k and higher that bookend reaches well above audible range which is also where a lot of the digital aliasing gets pushed.. so you end up with a clearer signal and more open top end.

    Anyway.. I absolutely love my Virus but one weakness for me has been the raw oscillators in some instances. Generating at 48k gives me the top end sizzle and strength I was missing on a lot of occasions. It's better to have the option to make a sound darker rather than to have only the choice of using the darker sound. Check the sample on your studio monitors.. it may catch your interest.

    Yeah I read the original post. I maintain my stance. If the choice is between allowing users to experience the difference between 3.9 kHz of barely-audible sound fidelity or having a Virus TI plugin mode without audio (as discussed elsewhere), I choose the latter. I think a lot of other people would too.

    As well, you have no idea if it's a slight alteration or a major undertaking, because you don't work at Access. Only they know.

    I do think the level of effort you're putting into this is impressive though.

    If the difference is barely audible for you and appears only at 3.9khz it's probably time for a new pair of monitors my friend! Haha.. what did you do, listen to it off the zippyshare player through laptop speakers?
    The difference is clear as day. In that example the difference is 4db at 15khz and only gets larger. By the time it reaches 19khz it's an 8 db difference. Ever hear people gripe about the dark sound of the virus? That would be a potential cure for that. You can always filter stuff down, but you can't open a filter more than full. The Analog 4 pole would benefit here for sure as it's a lot darker itself. I like the sound of it but it's too dark for me to use a lot of times.
    Also, where I suggested you went terribly wrong was that my post is primarily to illustrate the difference so that those who want more top end clarity and sizzle can make use of the existing ways you can operate the Virus at 48k. It's not a new development whatsoever. It's simply making use of the different clock rates in the existing functionality regardless of if usb is connected. The footnote to all that is that yes, it would be nice to be able to also have that with usb running. How do I know it wouldn't be a big undertaking? Well, because it's already implemented in the virus and happens currently in VC. The only prob is you don't have the option to choose it since it runs on auotopilot. So yes, one day a little box where you can choose between generating at 44 & 48k would be GREAT but until then you know how to get that sound if you want it.

    Just offering another view so it doesn't look like everyone on the board is agreeing with you by not responding.
    I would prefer Access prioritize other issues than this.

    Hmmm.. You may want to re-read the original post? This is about the difference in sound you can get with the virus generating at 44 vs 48k internally! As in.. with the unit as is currently available.
    As for one day allowing us to retain control over the selection of clock rates once usb is plugged in, well that's just a slight alteration, not developing a new feature or hunting down bugs.