Beiträge von nms

Zitat von tomski1978

a lot of us Chris are frustrated over the tech side of things...getting the Virus to work 100% is quite a task.
but can you compare the Virus in fatness to say a minimoog or a voyager...this is the point i was making....mixed in with a bit of late night annoyance from wasting so many man hours on this thing.

Really man? Talk about comparing apples to oranges. If you want a minimoog sound you get a minimoog. If you want virus sounds you get a virus. One is digital, one is legendary analog. They are very different tools that each have their place. I have both.

I don't know what kind of computer setup you're running over there but you should really sort it out. Are you on the newest mac OS or something? most people aren't complaining about the newest TIOS. As for me, I haven't encountered problems with mine that kept me from being able to work on it ever. I have a really solid computer though that was built for what I do.

No idea how on earth your virus could ever damage your orpheus btw.. nothing I can possibly think of would be able to cause that to happen.

As to the OP.. your virus is only as good and fat as it's presets or your ability to program it. Maybe you should download an oscilloscope and start learning more about programming the virus to sound more fat and analog. Find images of moog saws, squares or other analog synths so you understand what kind of characteristics thick analog waveforms have.

should be all good. Ask for pics of any wear if you can. Shouldn't be too much to ask.

You have 3 areas to increase the volume in the virus..

the amplifier patch volume, the part volume, and the osc volume.

Between those 3 you should be able to get your level where it needs to be. If you need more than that just do it in your DAW.

Sure converters are out of the picture on the usb version, but with the analog version you're comparing to the converters will never be out of the picture since the analog is subject to the Virus DA converters followed by your interface's AD converters. You may very well prefer the slightly less clean sound of the analog outs though and there's absolutely nothing wrong with that. I find it easiest to compare though when you break it down to single tones and listen to them side by side where it becomes more obvious exactly what difference there is. I like to take a look at the waveform as well for any changes and throw a spectrum analyzer on it. Ears can be fooled a lot easier than eyes can. That weird little ridge that shows up on the waveform in the image I posted earlier this thread is what spooked me from using tha analog outs. I don't know what that is and intend to investigate that a bit more. There's something else weird that I encountered and I'm going to put a separate post up soon looking into it. I have found odd glitches though.. one that happens via analog outs only and one that only happens when the virus is connected via midi.

If you're talking about how the DA converters are 24bit 192khz...I'd say either that only speaks of the quality of the converters (like running a 192khz interface at 44k) or the signal is actually being upsampled to 192khz before conversion to analog. In either case though my money is on marketing stats, not performance increase. Marc & Jorg from Access have both said flat out you won't get a better signal from the analog outputs. You will get the same signal, but subjected to the DA converter of the Virus followed by the AD converter of your interface. The best sound starts at the source.. from there it's up to you to keep anything from degrading it in undesirable ways.

Unlike yourself I have a lot more experience with both high and typical sample rates and have discussed it with a few of the top authorities on the subject. I do not whatsoever come from a place of bias. You know who instigated my switch? Andy Simper (developer & owner of Cytomic) when he released a new version of his compressor "The Glue" (best vst compressor in existence in my experience) with oversample settings. I heard the difference right away and that's when I revisited the idea of working at a higher rate across the board so I could reap as much of that effect wherever possible. Talking with he and Alexey Lukin (Izotope developer & developer of the best dithering & SRC I know of on the planet) and reading what they've said on the topic gave me much of the understanding I now have, combined with proof from results of my own testing. Tests similar to the result I posted with the clearly audible and better Virus tone at 48k. I'm surprised you can sit and with a straight face tell me about being a competitive engineer trying to outdo the sound of others and then say that difference is negligable or that CPU takes precedence over sound quality gains.

Part of being a competitive professional is having good gear and so long as you have a computer that is adequate for someone in that position you have all the power you need for working at 88khz. The best engineers know quality comes first. Btw, if you had looked into it properly you'd know your ram doesn't get taxed any greater than it does at 44/48.

I love my virus, but I'm not blind. The synth engine itself is 15 yrs old which is not ideal for digital technology and the oscillators aren't the best sounding on the market. Andy Simper also developed Dcam Synth Squad and their Strobe synth beats the Virus in terms of sound quality, aliasing, and sound of the raw oscillators. Having oversampling settings available helps see to that. If you think you know better than the guy who developed that as well as the Glue.. you'd be delusional to say so without a lot more experience and testing than you have done.
The interface of the virus and FX, programmability, mod matrix and everything else are where the Virus pulls ahead. I demand the best from my gear though so rendering at 48k was the needed way to bridge the gap and strengthen its weak spot a bit.

If you spent half as much time actually testing the results of a project done at higher rates rather than 44/48 (assuming you have the ears, monitors and production skills that won't hold you back from getting results) as you did dreaming up theories about how it's surely better to avoid going higher than 48k, despite how most of the top guys in the industry would disagree.. then you might actually get somewhere with this. A smart person puts a great deal of effort into trying to find ways to prove himself wrong.. not stubbornly cling to his theories with every breath despite evidence to the contrary. And yes, computers are more than able to handle it.. or so many people wouldn't be doing it.

Yes you can use the Virus as your midi and audio interface. I'd trust the converters in it more than any in a cheap $300 without question.

It's a fool's argument because the issue boils down to two options:

a) work at 44/48 and only have access to oversampling in the plugins that do feature it (so many don't) and allow your audio to be up & down sampled everytime it goes through one of these plugins at whatever quality realtime SRC each plugin is capable of and filtered with whatever antialias filter it uses. Oversampling adds latency as well. Benefits restricted to only the plugins that have oversampling available. Plugin oversampling also drains cpu don't forget.

b) work at 88khz where all audio stays at a higher rate thereby not being subject to artifacts created from being up & downsampled multiple times at varying quality or adding latency from the process. Old favorites, DAW plugins, the whole lot operates as if running x2 oversampling. Antialias filters become 100% invisible because they are so high up in the spectrum we can't possibly hear them.

With the giant leaps in the current multicore processors available and cheap storage in abundance there is no excuse for sacrificing audio quality and latency just to save some CPU or hard drive space. Computers are way ahead of where they were 4 yrs ago and the capability is there. If you're a professional who takes audio quality very seriously then no doubt you'll have a fast & powerful computer that is more than capable of handling the task.

Anyways.. this is after all the feature request forum and this has veered off track. I'm not holding my breath for VC to cater to any of this and I'm just going to use my workaround which as I mentioned previously is to switch to 48khz when it's time to track something then SRC it to my working rate. Alternately if I want to use the analog outs I have to disconnect the virus usb and sequence it via midi so I can set the rate to 48 internally and then record it into my interface at 88khz.
It'd be nicer to just be able to choose 48k internal operation reagardless of circumstances.. and I feel like it's a significant margin for quality that goes untapped in its current state to many people.. who knows.. maybe one day.

There is a difference between usb & analog outs.. with that you're correct. It's just not as clean as the source. Here's an odd example I found with the difference in the same saw waveform sent via analog out vs usb. The one with the odd lip at the peak is the analog out:

[Blockierte Grafik: http://www.bass-skidz.com/virus-odd-waveform.gif]

I'm not sure what's going on there, but after finding this I stick to the USB and fully believe what the Access crew said flat out which is that there's no cleaner output than the source sent out via USB.
Personally I prefer to go with the purest source and process it and effect it from there. I'm somewhat of a control freak though and prefer that anything that's going to affect the sound be by my own doing.

A few things Antony.. first, 48khz is not at all a standard in audio for anything aside from video production. Distributable music is all at 44, all commercially available sample packs are almost always at 44. If you're working in music you'll get the cleanest most efficient conversions by going in even multiples. Some SRC's handle it better than others and are near flawless, but many aren't so even multiples is a good way of playing it safe..

One other thing here.. shame on you for coming in throwing around Nyquist theory! I happened to catch that in a thread from a yr ago where you were telling people higher SR's absolutely hold no improvement and are nothing but a placebo effect to those that think otherwise. You couldn't be more wrong and dropping Nyquist theory has no place in discussions about digital audio production. Its only relevance is in recording.

Why you're talking about ability to reproduce frequencies higher than we can hear I do not understand. It's got nothing to do with that. As I previously mentioned it's about raising the upper band limit so that the filters and the aliasing created by digital processing further out of the audible range away from the music. Yes, further than the extra 2.5khz that you get from working at 48khz. Digital aliasing is NOT something that occurs only in the top 2.5k of the frequency spectrum.

I'm done debating this though as I don't have the time and it'd only go in circles from here. To have the stance that nothing can be gained by going above 48khz is to imply that every company who's oversampling multiple times above that is doing it for no good reason and they're simply fooling evreyone with what's nothing more than a marketing gimmick. You know what the developer of IMO the best vst compressor ever made recommended to me recently? Despite his plugin offering up to 16x oversampling, he recommended I just run it without oversampling since I'm already working at 88.2khz where the biggest difference is already achieved.

Ionis - Your ears may be deceiving you.. it's easy to allow ourselves to be tricked now and then. Expectation bias and whatnot. I would strongly doubt your firebox would have anything positive to add from the AD conversion. The cleanest you can get is the source.. which is exactly what is streamed directly into your computer with the usb. Your converters will only color the sound if anything but if you like the sound then go for it I say. The Access team have said so several times and said directly that running it through the analog outs will not give you better sound quality. I admit I was skeptical of that myself til about a month ago but I was able to confirm it and find some distortion to the waveform that I wasn't happy with. I'd rather track the source directly and handle the SRC myself.. then run it out through any of my hardware compression or distortion if I choose while still retaining the original. I have to print to disk first anyways if I'm processing stereo stuff since my outboard units are mono so I wouldn't really want to deal with a source that's gone DA/AD/DA/AD.

Anyways.. are you using other outboard synths or gear? I would avoid the FF400 myself. Duets sound good. A Motu ultralight mk3 would get you good sound.. people are a little clueless as to just how good the converters are in those. People are so easily fooled by expectation bias and believing something's good or not because they read it somewhere. We've done testing on a lot of the top interfaces and the transparency of the Motus in test results (using proper test software) compared to the rest were shocking. As for the Virus, the inputs on it tested much more transparent than my MR816x. Do you have any issue with using the Virus as an interface? Unless you need extra inputs or the ability to work above 48khz I'd use that if it works with your system. I'd bet on the DA/AD quality being better than the firebox.

excellent, if you're doing a PhD on this then you'll find this all the more useful.

That sample I posted is particularly relevant because it's not a high frequency sound.. it's a 147hz sound with a lot of harmonics yet listen to how much it's affected when you isolate and compare it in a way that makes the difference obvious.

I used a 2 pole filter fully open for that. If you look in a high res spectrum analyzer you can see the differences extend down pretty far and not just 1 or 2k. You should know better than to assume an eq can easily reverse that. You can't put back what's been filtered out, only artifically boost what remains. If you're talking about some background pads it's probably not going to be an issue. If you're talking about synth leads and elements that are in the forefront then there is most definitely capability to produce better, sharper, clearer, more detailed sounds. Do you like the sound of the analog 4 pole filter but hate how dark it makes a lot of stuff sound? Try it at 48khz.

You have to understand that while 48k is an improvement over 44, it still only raises the available frequency ceiling 2.5khz higher than the usual 20k limit and is certainly not the be all end all sample rate where digital processing and synthesis are concerned.

Working at 88/96 isn't about capturing frequencies above 20k.. not in the least. It's about having enough frequency headroom to push the aliasing and filters WAY above the range our music exists in so it's not messing with it.
Do you want to listen to the music? Or do you want to listen to the music through a bunch of stacked filters you have no control over and digital garbage that's been folded back into the audible range?

It's only been a year since I switched to 88.2k from working exclusively at 44. In the past year and a half I have researched, tested, studied, and debated the topic to death. I also learned a lot discussing it with Andy Simper (owner of Cytomic & developer of 2 of the best plugins released - The Glue compressor & FXPansion's Dcam synth squad) as well as Alexey Lukin (of Izotope fame and developer of the Izotope 64bit SRC).. and have also bugged Marc & Jorg from Access a lot about the different aspects relating to the Virus. The things I've said on the topic are a result of what I have learned first hand. Chalk it up to my perfectionist nature and crusade towards producing the best sounding music I am able to. Rest assured, I don't just buy into theory or marketing hype here.. every piece of my rig from hardware to software gets rubber glove tested from top to bottom.

No offense Antonyb, but you are way over your head here with this discussion.

Zitat von AntonyB

You can't force a plugin to do anything it's no programmed to do. all increasing the sample rate will do is make it take up more memory in sample buffers

Oversampling is when a source is upsampled by a given number of multiples of itself. 88.2 yields the equivalent benefits of oversampling every plugin in your project x2 except without the downsampling back to 44 that would happen every time. Any plugins that oversample will have at least x4 oversampling. Your theory that it's all a big hoax by software developers and that the best engineers in our industry are being fooled by it all when really there's no difference is quite a stretch and just plain wrong. That's coming from someone who's tested it extensively and directly discussed it with a few of the best developers in the industry. Are you not familliar with aliasing or brick wall antialias filters at all?

Zitat von AtonyB

If your plugin doesn't sound great at 44.1kHz then you are doing something wrong. The Virus is the best digital synthesizer i've ever heard whether at 44.1 or 48, so Access manage just fine with their internal upsampling and subsequent downsampling.All that aside - this is nonsense, the Virus will only generate sample rates of either 44.1 or 48. If your project works at 88.2 then you will be better served (with your desire to pursue inperceptible differences) by upsampling 44.1 by a factor of 2 rather than 48 up by a factor of 1.8375 - clearly the integer scalar will give you better results.

Is that right... so what do you have to say to this: Virus-Saw-44k-vs-48k-source-downsampled.wav
That's the sound of a 147hz saw from the Virus.. it's what I like to call an idiot proof test since it exposes the differences in a clearly audible way. The first half of the loop is generated at 44k, second half is 48k downsampled to 44. I think you'll find those "imperceptible differences" audible even on crappy laptop speakers.

Zitat von Ace17

Agreed. x2 isn't a big oversampling factor.

Actually it is by far the most relevant and audible oversampling factor. From there it's majorly diminishing returns.
As for SRC, with a top quality SRC like Izotope's 64bit SRC there's nothing to worry about from the SRC upsampling from either 44/48k to 88/96k.

Zitat von Ionis

nms: What I do to combat this lo-fi reality is I use direct outs on the Virus TI and input them into my interface at a higher resolution. This solves two problems at once:
1. The ability to get the raw undithered hi-res Virus TI's 192 kHz outputs.
2. The ability to resample the higher outputs into a lower resolution without aliasing your audio.
In other words, until this baby can pump out 192 kHz through USB, this is the best way to "hear" the Virus TI.

Despite previous theories, the cleanest signal you can possibly get from the virus is with your project rate set to 48khz and running via usb. The audio that goes out the virus DA is taken from that same source. The "192khz DA converter" on the way out is essentially just for marketing. It's not producing a better sound than what comes out the usb. The converter chip may be capable of processing that high quality of a source, but the source has already been created at 48khz. If you hear a difference between the analog and digital outputs it's because you're fooling yourself or listening to the degradation of the DA to AD conversion. Maybe you like the coloration from your interface, or maybe you are listening to the specs in your head about the 192khz DA.. but you won't get any clearer signal than the digital source piped through the usb direclty and the Access team will say the same. If you are running your project at 44khz and the usb is plugged in, the source that feeds the analog output will only be 44k unfortunately. The one way the analog outputs can lead the pack is if you don't have USB plugged in and run it via midi, allowing you to choose 48k synthesis internally which then goes to the analog outs and gets streamed into your interface at whatever rate your project is running at. You're still sending it through a full DA/AD cycle though unlike if you'd ran it usb. This is why I've decided to just switch up to 48khz in my project when it's time to track something and use the usb, then switch back to 88k and resume use. It takes me about 15 seconds to open that take in soundforge and resample to 88k using the Izoptope SRC. That way I get the cleanest source possibile tracked to audio. I do like to process with my outboard hardware.. but doing it this way allows me to route it out for outboard processing and try whatever I like, while meanwhile I've already captured the take I wanted from the virus perfectly.

Ace has it exactly.
AntonyB - Do you really think that so many of the top guys use higher rates because they're simply caught up in sales hype though? That's silly as we're talking about the best in the business. I don't buy into going above 96k though personally which is why I recently picked up a Mytek 96 ADC rather than spent more for a 192k converter.

Here's what I'm talking about regarding slight converter improvements in 44 vs 88k. I get the same differences whether going MR816 DA into Mytek AD, or MR816x DA/AD. Taken from Right Mark Audio Analyzer via DA/AD loopback test:

Noise level: -97.5db @44k, -100.6db @88k (3.1db difference)
Dynamic Range: 101.4 @44k, 103.6db @88k (2.4db difference)

To me none of that would be any deciding factor though. The deciding factor and most audible difference comes from the stacked effect of all your plugins and digital synthesis. There's no easier way to get a better sound from them as a whole than going up to 88.2k. The leading software developers and high end gear manufacturers most always recommend production at 88/96k. But anyways this thread isn't meant to be a debate about the topic, rather a small feature request.

"Aside from plugins..."
So you can't find anyone to explain to you the benefits aside from the main reason for it? haha. That's not a note to put in brackets as that's the entire point of it. We live in a world where projects are stacked with digital operations and that's where the difference lies. If you were recording all in analog then there would be nothing to gain there past the potential of slightly beter AD encoding from your converter.. which actually a lot of converters do see btw. I get 6db higher noise floor with mine for instance. It's not silly at all since there are limitations involved with 44 & 48k which can only be surpassed by operating at a higher rate or using oversampling.

I won't ever work at 48k. My system is peachy at 88.2k as I've built a destroyer of a PC to handle it, but going to 96k has no benefit to me and it's preferable to stick with even multiples of your 44k final format as some processes will benefit and handle the conversion better. I can still max it out by working with unnecessarily bloated projects but I'd trim down the fat before moving anywhere from 88.2k.
Most of the best in the industry work at higher rates in the digital environment for a reason. If there were no gains to be had past 48k then you wouldn't find some of the best plugins for instance offering 4x oversampling. It's better to work at a higher rate though than to rely on plugins sampling your audio up and down multiple times through a project. You also get lower latency which is very nice.

doesn't sound like you guys have properly got the hang of working with your virus yet! The virus isn't a softsynth. It's a Virus.
It's good the way it is. You can have 16 parts all lined up in one instance so you have access to everything in one place as opposed to any vst where it would take you 16 channels. You can then setup separate audio channels to receive the 3 individual audio out streams. You've got your whole browser there so you can easily swap patches back and forth into the 16 slots or make an empty bank to use for the track you're working on if you feel like it. The only way the current system would realistically hold you back is if you need more than 3 separate streams going at once and can't have any of them doubling up on your 3 available output streams. It's not something I've ever run into in the 2 years I've had mine. If you're going to really go to town on the virus like it's the only instrument in your studio then you'll have to use the 3 streams as groups.. like Bass, Synths, FX or something. That or just track the damn thing to audio as that's a normal part of producing a song anyways.

yep sasha definitely belongs to the uk!!

Zitat von Knoggs

Keep getting this message:

"Virus TI has discovered a setup problem. The host is using an odd audio
interface buffer size which causes communication problems. Please
adjust the buffer size to any of the following sample values instead:
64, 128, 256, 512."

hoped the official version would get rid of that problem, but it does'nt.

My soundcard does not adjust its buffer by sample values, it adjusts in ms.
I know that there are other soundcards that adjust in ms too.

Obviously, it's a fault from the virus software.
Please fix it.

Alles anzeigen

Dear Access .. that message needs to have a box for "do not show me this message again"! Either that, or have it only come up the first time VC runs.

nice one bob!

Sometimes no news is good news!
Seriously, if this thread goes 2 weeks with no confirmable bugs the Access team should get drunk and celebrate! Gotta say it's looking good so far.