Virus TI running at 96khz

  • If a higher sample rate can't happen would better filtering be a potential solution?


    Once the higher frequencies have been reflected back to the audible range, it's too late : you can't filter them without also filtering "real" frequencies (the harmonics of your sound).


    A filtering that would work would need a higher sampling rate : you would then naively synthesize your "sharp" signal at 4x your final sampling rate, let's say 192kHz. Then, you would apply a low pass filter to remove everything above, let's say 20kHz : so now your 192kHz signal does not contain anymore frequencies that can't be represented by a 48kHz sampling rate. You can then peacefully downsample it to 48kHz, aliasing will not occur. So you see that to apply this method, you need to work internally at a higher sampling rate.


    Please note that there are A LOT of other methods to simulate an oscillator, this one being the simplest! I don't know how the Virus works. For example, another method works by synthesizing all the harmonics (sines) and then summing them together. So basically, as you're summing sinusoids here, you don't have to worry about aliasing. This method, compared to the previous one, would benefit a lot less from a higher internal sampling rate.

  • :thumbup:


    sorry guys little oftopic:
    ok. i never want to be offensive.
    yes u might be right but in the same time that man sounds exectly like thouse who will figh to death and will tell
    $ 200 guitar sound exectly the same as $5000 and u do not need to buy 5000$ guitar, or thouse people who
    tell that aliens does not exist with no knowladge about that topic and all their will guesses are based on just looking in to the sky.
    and if they did not see anything that means it does not exist. and what makes me more angree that they state their
    willguesses as a truth.
    so i am stoping on that and will better ignore that man.
    now lets back on topic
    There are a reasons why modern Phisical moddeling VSTs alow u to use a huge upsampling and alow u to run
    at 96khz rate. they did not put thouse staff just for fan, but there re is a benefit of using it. and even if u will down
    sample the end result to 48khz there still will be benefit

  • nicely seid,Totaly agree :thumbup:


  • There are a reasons why modern physical modelling VSTs allow you to use a huge upsampling and allow you to run
    at 96khz rate. They did not put this staff just for fun, but there re is a benefit of using it. And even if you will down
    sample the end result to 48khz there still will be benefit


    Exactly.


    Processing your audio at a higher sampling rate than the output sampling rate is called "oversampling". As I said above, and as Marc also said, not all sound processing methods will benefit equally from oversampling.


    For example, physical modelling greatly benefits from oversampling : that's why the VSTs you're talking about have such options.
    But a delay unit will benefit a lot less from it. Worse: now, your delay needs more memory! And memory is a scarce resource on the Virus (not so much on PCs). So you don't improve sound quality, but you increase memory usage.


    The point is, blindly oversampling the whole synth is far from optimal. In terms of memory, in terms of polyphony, and in terms of quality.


    As Marc said, some parts of the Virus already use oversampling. Why not let those who know, i.e the Access Team, make these choices for us?

  • Here's my technical understanding, please correct me if I'm wrong :



    1- the output sampling rate and the internal processing sampling rate are two very different things.


    Absolutely - oversampling may be necessary for non-linear models.



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    2- an output sampling rate high enough to allow non-audible frequencies is a waste of space


    Correct - It contains no useful information, even if you plug it into an amp you are going to mic up, it is not conceivable that you could construct a situation where the non-audible frequencies have any impact on the audible ones, mainly because it would have to be a headroom issue and you wouldn't be using a signal biased in that way. Nonlinearities such as distortion tend to only create frequencies above individual input frequencies (note this is not true for AM or FM).


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    3- a wav file at 96kHz can be downsampled to 48kHz without any audible difference (for a human).


    Correct - a causal (ie zero latency) AA filter would be more than sufficient for this owing to the massive guard band available.



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    4- depending on how the internal audio processing is done, a processing sampling rate higher than the output sampling rate can be beneficial : this is what we call oversampling.


    Indeed - as is the case for the Virus.





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    5- Are there good reasons for wanting an output sampling rate higher than 48kHz?


    Bigger numbers sell better. Technically you are spreading bit quantisation noise over a broader spectrum (some of it in the inaudible part, which can be filtered), but its already at -96dB for 16-bit so the cost of extra bandwidth really isn't worth it.



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    6- Wouldn't running the virus at 96kHz (internally) enhance synthesis/processing quality?


    Only in certain parts, where they already oversample, elsewhere it makes no difference beyond increasing the buffer sizes required and the number of MACs for no real benefit.



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    7- What about processing latency? A lot of processing can't be done in parallel, so halving the polyphony may not suffice. I mean, if one of your processing units is sequential and can only process up to 60k samples every second, no matter what you do to polyphony, you can't achieve real-time performance in 96kHz.


    In general DSP algorithms can be very easily multi-threaded, if you have more MAC (multiply and accumulate) units you almost unconditionally get more usable MACs (this is what CUDA and OpenCL are all about), but the Virus has 2 DSPs in it (I don't know how many multipliers that makes but probably either 2 or 4).


    For traditional DSP systems it's all about clock speed vs sample clock - if you have a 120 MHz DSP with a dual MAC unit you get 240 MACs, so if you run at 48 kHz you get 5000 MACs per sample. This means if you run at 96 kHz you get 2500 MACs per sample, BUT for the same spectral resolution (ie an 'equivalent' filter) you also need twice as many filter terms which means an effective increase in load of 4x (so, very loosely, if you could manage a polyphony of 4 before, you can now only manage 1) - not only that, but your sample buffers need to be twice as long, which may result in you even running out of memory (making it impossible, so your polyphony may be reduced further). All the while you will not hear any difference in the output (in general, not specific to the virus).


    Looking at some of the other posts I don't think this is possible to overstate - this has nothing to do with oversampling which you may do on an individual effect because the model requires it... with a robust enough interpolation algorithm the benefit of already having the higher sample rate does not go anywhere towards offsetting the above disadvantage. Delay/Phaser/Chorus/Flanger, etc. effects all interpolate... for those high frequencies in phaser effects you need to do subsample delays and the quality of your interpolation really pays off in a great sound (interestingly this is why 'analogue' phaser effects sound great as they have a fixed number of samples delay and they modulate their 'sample clock' speed continuously; a luxury you cannot enjoy in a complex system like the Virus.



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    On the non-technical side, market adapts to what people will buy, not to what people will benefit from ; it's a fact that (for the same price) a 96kHz soundcard will sell better than a 48kHz one. Wether it makes an audible difference or not. Because, in everyone minds, bigger numbers are better. So I think arguments based on what recent gear do are irrelevant here.


    Amen - I mean the audio bandwidth on blu-ray is ridiculous 192 kHz with 24-bit - as if any consumer equipment is low noise enough to even achieve 24-bit resolution let alone them being able to tell if it could. The only reason they get away with it is because the demands of the video processing make the audio a peripheral issue so it makes no real difference cost-wise.



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    Ruari: it's very hard to estimate the quality of any oscillator only by looking at the waveform. At least, look at the spectrum, which is more important here.

    You do not want to generate a signal whose waveform looks close to the reference signal : you want to generate a signal whose frequency content is close to the reference signal frequency content. Because this is the only thing your ears will pay attention to.

    And guess what : in a digital world, the closest waveform generally sounds like shit.


    I hate to pick up on this here - but if that were absolutely true then audio sampling would also sound like shit. Looking at audio spectra is also a fuzzy art - you have to be very careful about what window size you use and what windowing technique you use as the results can vary profoundly - with windowing you have to trade off SNR performance with the ability to distinguish close peaks which merge due to spectral leakage (also a function of window length).


    Getting the frequency content just right is all well and good so long as you get the PHASE information right, too, otherwise your waveform won't be the right shape - and if you put your signal through a non-linear effect you CAN hear a difference. Phase information is an all to often ignored part of signal processing despite the fact that if you don't look after it you can get unpredicted drops in frequency response and trash your SNR performance. Just look at what happens if you apply a Hilbert transform to a square wave (which sort of gives you the 'loudest' sound for a given headroom).

  • I hate to pick up on this here - but if that were absolutely true then audio sampling would also sound like shit


    By audio sampling, do you mean what an ADC does? Correct me if I'm wrong, but isn't there a lowpass filter before the actual sampling in a ADC?


    Anyway, thanks for your answers! :thumbup:

  • Yes all codecs need AA filtering (although sigma-delta ADCs have very loose external filtering requirements). What I mean is that if sampling an oscillator (which is effectively what matching the waveform in the time domain does) were to sound crap, then any sampling would - which is clearly not true. It does lack the inter-cycle variation you may see on any oscillator, however.

  • as Marc also said, not all sound processing methods will benefit equally from oversampling.
    For example, physical modelling greatly benefits from oversampling : that's why the VSTs you're talking about have such options.

    You mean like the modelling in a virtual analog synth called the Virus? Were you trying to imply that the Virus doesn't need what those other synths need and have?

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    memory is a scarce resource on the Virus (not so much on PCs).

    For who? Snow users? Niether polyphony nor memory have ever been an issue with my Ti2. My PC has to handle an infinitely heavier load in comparison.


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    The point is, blindly oversampling the whole synth is far from optimal. In terms of memory, in terms of polyphony, and in terms of quality.

    As mentioned, polyphony & memory are a non issue to me. The majority of professionals aren't using the Virus for more than one or two sounds in a track. We really lose nothing of interest by running the entire thing at 88khz but the gains where applicable would give a better sounding result. The pursuit of quality often does not share the same path as quantity. This is nothing new. As an example do you know how deadmau5 recommends using the Virus? Use it on one sound at a time, bounce it down to audio, and don't use the onboard FX.


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    As Marc said, some parts of the Virus already use oversampling. Why not let those who know, i.e the Access Team, make these choices for us?

    There's no polite way to say this, but are you aware of how brainwashed that sounds? You know how many times you could find virus & alias in the same review/sentence if you googled it? I bought my Ti2 new in a store in 2009... yet it came hampered with USB 1.1 which was superseded back in 2000. While there are many things I love about my virus, it has always had its flaws and I don't swallow the "I'm sure they got the design right all those yrs ago" notion. My Ti2 came with a USB cable and manual but I didn't get the magic Kool-Aid.


    I paid $3450 for my Ti2 new but that's never given me any reason to put my head in the sand when it comes to knowing its strengths & weaknesses.This forum is probably the wrong place for discussing higher quality audio for studio recording applications though I'm realizing so I'm going to drop it.


    Does this sound to you like the perfect design and pinnacle of performance? Virus vs Sylenth aliasing.


    Hey Antony.. care to explain that one?


    Does anyone here actually understand aliasing?


    I can't help but shake my head when I hear the oblivious comments about not needing to capture audio above what we hear and how it's a waste. It's not about that.


    It's about having enough bandwidth to keep the stuff just above your hearing limit from being reflected back down well into your hearing range... and not having to interfere with your audible range with the necessary filtering to prevent it. It's also about avoiding phase distortion. The further away your fitlers are, the less chance of phase distortion.

  • You mean like the modelling in a virtual analog synth called the Virus?

    No, not at all! Maybe I wasn't clear ; I was talking about this. There are simpler methods to implement the building blocks of a VA synth (and yes, many of them will also benefit a lot from oversampling). These methods will try to match the behaviour of analog parts, but without simulating the inner workings of these parts.


    Were you trying to imply that the Virus doesn't need what those other synths need and have?

    No. I was saying that synths that do physical modelling are designed to benefit from oversampling. Because physical simulation requires you to discretize time anyway, and most of the time, a smaller timestep means a bigger accuracy.
    But there are other methods, some of them called "bandlimited". For example, when summing sine waves to make a square wave you don't need to oversample (see here). Yes, you're simulating a VCO, but this is not physical modelling.


    For who? Snow users? Niether polyphony nor memory have ever been an issue with my Ti2. My PC has to handle an infinitely heavier load in comparison.

    I wasn't necessarily talking about the Virus here. I was saying that oversampling a fixed time delay was non-desirable, because there would be no quality benefit and it would use more memory.


    As for the memory on the Virus, you can hit the limits when playing with delay at low tempos. The max delay time for one part of the Virus is 693.6ms, which roughly corresponds to 30000 audio samples at 48kHz. Do the math, we're talking about allocating something like 100kbytes for one part, that's very small. Not the kind of choices you would make when designing something for a PC! I think polyphony is irrelevant here.

  • NMS i don't think you get the difference between aliasing due to oscillator design - which remains in the virus due to backwards compatibility (I guess they want peoples' old presets to keep sounding the same) and aliasing as a function of sample rate - the two are independant. They also fixed/improved it using the wavetable oscillators if you even care. There is ALWAYS a trade-off between precision and performance and different products/organisations will be comfortable with different points. For VST programmers its easy because if it underperforms they can just blame your pc for not being powerful enough, regardless of how inefficient their code is.


    I don't care what deadmau5 recommends about the virus - he may have found a workflow that works for him but it's very arrogant for you to assume you know the working practice of the entire, lets say 'pro', virus user community or that you can suggest there is a right and wrong way to do it. Personally I need more than one or two notes at a time, I don't know if thats particularly unusual...


    Besides which, you will be completely oblivious to the varying usage of RAM on the Virus, it's not like it has a RAM usage meter (or maybe it does and thats what the load meter is...) - you just don't know... unless you have a debugger attached.



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    You mean like the modelling in a virtual analog synth called the Virus? Were you trying to imply that the Virus doesn't need what those other synths need and have?


    I don't know what you're talking about. The virus uses oversampling only when it needs to - I really don't get how this point seems to slip you by every time.

  • Incidentally, I do record my virus one part at a time for projects - but I do use the onboard FX (Duh! why would you rob yourself of the TIs fantastic FX section??) and, like I said before, most parts consist of more than a couple of notes, so some polyphony is appreciated, so I would be upset if Access came along and said hey we've quartered the polyphony on the virus just to satisfy a couple of nutters... can you imagine??


    And while we're at it.


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    It's about having enough bandwidth to keep the stuff just above your hearing limit from being reflected back down well into your hearing range... and not having to interfere with your audible range with the necessary filtering to prevent it. It's also about avoiding phase distortion. The further away your fitlers are, the less chance of phase distortion.


    That is what guard bands are for - 48 kHz gives you a more than generous guard band.

  • Right.. so then you'll have no problem explaining how this happens then? You seem to have selectively skipped over this in your reply quotes:


    Virus vs Sylenth aliasing.


    Is that the perfect design you speak of at work with the more than generous guard band?


    What I said isn't arrogant at all to assume. It's true. The majority of professionals (clarify- well regarded artists commercially releasing music of high sounding quality) aren't using the Virus for more than one or two sounds (not notes, different patches) in a track.
    This is a reality, not an assumption. You say "some polyphony is appreciated".. well the TI (especially the TI2) is equipped with a LOT of polyphony and if you halved that you'd still have plenty to work with especially since you commonly don't have a huge mess of sounds all playing at once.


    I won't go into the mau5's opinion of the virus' fx but again it comes down to an issue of quality vs the latest vst offerings. There are obviously plenty of delay & reverb plugins that are far better in their sound & flexibility.


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    I would be upset if Access came along and said hey we've quartered the polyphony on the virus just to satisfy a couple of nutters... can you imagine??

    Can I imagine Access removing all other clock rates in order to add one additional rate in the options of 88khz? Uh... no.. and the fact you would even mention that as if that's what anyone was ever asking (or would be necessary in order to add functionality) really paints a clear picture of your logic & contribution to this topic.

  • If you read above:


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    NMS i don't think you get the difference between aliasing due to oscillator design - which remains in the virus due to backwards compatibility (I guess they want peoples' old presets to keep sounding the same) and aliasing as a function of sample rate - the two are independant. They also fixed/improved it using the wavetable oscillators if you even care.


    That oscillator design has been around for some time (not sure if it goes ALL the way back) - and they have superceded it.


    Also, if you are using the virus for traditional mainstream sounds, you will get oodles of polyphony out of it (maybe, it really depends on what you are using it for which makes discussions about how much polyphony is 'enough' moot) - but if you are using it to get true VA sounds you need the polyphony because of how much is left at the end of all the effects. By the time you are emulating 3 minimoogs, two polymoogs and an ARP Odyssey you really have to think about how many notes you are going to get out of it.


    For example, when we did this: http://youtu.be/gdjECAvwJFk?hd=1 - every sound beside vocals, bass guitar and drums (obv) is coming from the Virus. None of the presets, besides the claptrap, are lightweight and I've had to do a lot of fine tuning to get enough polyphony out of just one Virus for that. I'm working on a TI and the extra power from a TI 2 wouldn't go amiss, but I don't have £2000 to chuck around right now.


    Lets set aside the complete rewrite and jigging around of memory, etc. programming a higher sample rate might entail, the reduction in polyphony, even on a TI 2 would make this impossible.


    The best you could hope for is that they find a way of doing SRC right on the end of the signal chain to get you your exotic sample rate - but then if that's what they do, you can do that yourself - especially if you are recording it part by part with no FX - I'm assuming your wave file editor has a simple enough sample rate conversion...


    OR - you could just work at 96 kHz... just a thought...

  • 88.2khz is an exotic sample rate? lol.. alright that's my cue.. I'm out. You can hang out and post in this topic but it's relating to something beyond your needs or understanding. You seem to have too much time on your hands but you could attempt to do other people the courtesy of instead involving yourself in discussions relating to things that DO tie into your needs and understanding.


    I'm not going to bother bringing up sample rates again in this forum. The people who are concerned with that are generally not sitting around on the Virus forum killing time. .

  • Can I imagine Access removing all other clock rates in order to add one additional rate in the options of 88khz? Uh... no.


    No, but you have to admit it might raise some new issues. Like having different max delay times depending on the current sample rate (which is currently not the case). Nothing impossible, at first sight.


    Anyway, do you have an idea of why Access didn't already make these sampling rates available?

  • so I would be upset if Access came along and said hey we've quartered the polyphony on the virus just to satisfy a couple of nutters... can you imagine??

    why u puting nonsence again?!
    didn't we seid that 96khz should an option. 48 woulkd be still awailable if u need more poliphony.
    p.s. ok, please leave this thread now!!!

  • No, but you have to admit it might raise some new issues. Like having different max delay times depending on the current sample rate (which is currently not the case). Nothing impossible, at first sight.


    Anyway, do you have an idea of why Access didn't already make these sampling rates available?

    ok u can ask the same question tape in 1000 BC why Microsoft did no bring Windows 7 already?
    get it?!
    p.s. if something is still not implemented already it does not meen that it is not a good feature....

  • ok u can ask the same question tape in 1000 BC why Microsoft did no bring Windows 7 already?
    get it?!
    p.s. if something is still not implemented already it does not meen that it is not a good feature....


    Bagpula - The Return
    Coming soon


  • Seriously dude, what's wrong with you?

    i do not like people who twist or change words or meaning of other people have seid.
    even a singel changed or tiwsted word can change the meaning radicaly, and it is what that man does all the time or
    at list in this thread. That man try to look like he has a big knowladge and give us alot of technical staff which
    might be correct in most cases but he twist or change one or a fwe words or numbers in that info he gives and that the over meaning what he saying is complitely
    falls then. and he wants that falls meaning to give to awrybody.
    as it was seid by the God thouse who will change or twist even a singel word in original Bible will be in Hell.
    Well that man does not change the Bible but that man gives a correct information in which one or couple changed or twisted words which maked the overal meaning falls.
    that what makes me too angre