96khz Sample Rate

What's the reason for this demand?

well, i guess it's about recording in 96kHz when using the Virus as audio interface...
however, this reminds me of something i've been wondering about:
what is the internal sample rate the Ti is processing at and streaming via USB, same like it's ADCs?

best regards, gravity

Zitat

As to why, 96khz sounds better to me than 44.1

why 44.1khz, i'd bet the Virus is processing at least at it's maximum interface sample-rate, which is 48...
however, i work on 96k too, and the internal rate was exactly my point,
so marc or whoever knows for sure, please enlighten us!

best regards, gravity

I think Access will never tell us the full truth here so we can only speculate.

I don't think the TI's basic processing path works internally with more than 4x.xx kHz. Still there can be some stages oversampled and then things get difficult.

Remember how old the basic TI HW design is. TI2 isn't much more than the same thing with a new DSP claiming 25% more calculation power - which isn't a really huge change and nothing to trade in a TI for a TI2...

But assume you could switch to 96 kHz, what do you expect? There is just one thing for sure - half polyphony. Let's add all the new OS features I am curious when the TI has problems producing even one note... On the other side I hardly believe you will hear twice the sound quality.

To be honest, the 96 kHz is just a number. For live uses on stage this is totally useless as all locations have by far stronger and other issues than the perfect hi end spectrum. I don't say it's totally useless but to benefit from that you need to be picky almost everywhere. One of the points is the converters clock precision. If the jitter is to strong here dialing up the sampling rate simply gets you nothing more than higher calculation demands... My personal opinion here is go for 24bit instead of 96 kHz.

Coming to "production", do you really think the majority of music consumers do have high end audio equipment at home? I guess no, they have mp3 players and cheapoo earplugs... do you really think 96 kHz sampling rate make the cake here.... maybe better invest some bucks in cheap earplugs to get an impression on how your music might be really perceived by the masses... if you really have such an audience, do you?...

To finish with, the TI's DA converters are quality converters that sound really good. If you find this outdated you have to dump the TI as it is outdated too - HW wise. I personally wouldn't spend time to think about 96 kHz sampling rate until the tradeoff in polyphony would be much less important than it is today. But even if, I don't think anyone really basically needs 96 kHz - unless you do a lot of pitch shifting or similar extreme things with the output signal of your virus...

Finally - better spend your time making music than believing you can't go on without a very high sampling rate... and when you're a bit older your ears won't notice the difference anyway, even it was ever there...

there are algorithms which indeed create artifacts, for instance the sine wave oscillator. we didn't change that intentionally (so it keeps it's original sound back from 14 years ago). instead we've tried to offer an alternative, the sine wave of the wavetable oscillator does not feature those artifacts.
the virus uses internal oversampling where necessary.
best, marc

Thanks Mark.

In a way I can not understand people... who are never satisfied with what they have. Maybe it helps to re-lable the Virus as a "vintage VA synth" so people can change their mind without loosing their faces not using the latest hype... LOL

Even more strange I see those people who can't go without bit and sample frequency reducers - but of course best have a basic signal path of 192 kHz... But anyway, high end noise can't be discussed as taste can't be discussed...

Eh, what about a Virus in a golden housing? ...for a warmer sound...

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.

@ 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...

@ 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?

even 44.1 kHz gives a generous transition region above human hearing - like i said, i have fun surprising people that they cant hear anything above 18kHz (these are young people) - I myself can hear up to about 18.5, but I can't say I enjoy hearing frequencies that high (as a harmonic of a deeper sound or otherwise), its just tiring to the ears.

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.

Also, yes a lot of codecs use 1-bit sigma/delta decimationish type approaches - its a nice way of getting something like a super duper high input impedance without actually doing it (by mirroring the signal internally, effectively). It leaves the codec as 16/24-bit data, usually, however.

Heh, and don't get me started on 24-bit sampling... with dithering you really don't get quantisation noise that people care about - so why do 24-bit samples sound nicer? Probably because a bit more work went into making them sound shiny such as compressing them more intelligently to make use of the dynamic range.

I also muse at all of this since any distribution medium that most music will hit will probably be mp3 (sadly) or very loud in a club where quantisation noise is basically buried in the nonlinearities of speakers playing very loud...

As I said before, I don't want to spoil anybodies fun, and hard disks are equipped to take the big data bill, but for your own sake don't lose any sleep over it!

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.

Zitat von sdrr00

...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

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...

Zitat von TiUser

Eh, what about a Virus in a golden housing? ...for a warmer sound...

and i thought you mean this: the mighty goldfilter - i'm sure even openlabs wound't score as many bling bling points ...

marc

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...

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

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

the goldfilter, we did in 2008 and it was limited to ONE unit.

best, marc

...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

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

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.

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...

Zitat von AtonyB

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