Solaris and the XITE-1 @96k

[YISH313z]

Im looking to only record audio into Reaper once I'm done sequencing in Cubase. Cubase LE5 only allows 8 tracks to be recorded at a time and a Max of 48 audio tracks total(Stereo or Mono), I purchased Reaper a couple years ago when I was heavy into VST as it kicked ass over the other DAWs, and still does for mixing in comparison.(per clip effects, skins, better workflow customizing, etc ), but...Since Scope all I need is a Sequencer and a Multitrack tape recorder.

I would have used Reaper for midi, but its tied into the Asio Buffer Stack creating dela....hold that thought while I look back into that midi thing for a moment...

[dawman]

Reapers MIDI playback is tight and doesn't suffer from huge amounts of data in mutliple streams.
A good test is to send active Sensing from all MIDI sources into a Scope project window, then w/o filtering allow that information into Sequencer MIDI Destination module, then back through the Sequencer MIDI Source module.
Notice that Reaper can still play tightly even with this huge amount of useless/ space taking data.
Now when you use Reaper as a source of MIDI automation, and merge it with realtime controller data from the XITE-1 MIDI In and see everything working smoothly, it's proof of the Pudding.
But make sure that in the Preferences you have checked the box allowing multiple threads instead of 1 thread of MIDI.

Also check out Kenny Goias Tops and Tricks.....
http://www.groove3.com/str/reaper-tips-and-tricks.html
I use a splash of verb on his voice, it makes me feel as if I am at as seminar..

[YISH313z]

yep...Just as I remembered there was an option in the MIDI settings that every time I reinstalled Reaper to use it as a MIDI only Sequencer I always forgot to enable it ...Low Latency/Low Precision mode, this bypassed the ASIO buffer issue and allowed midi to work with no delay/latency, and right on time as Ive got my STDM cable parts from DIGI-KEY, well Jimmy looks like I might be putting Reaper in my Scope tonight, thanks for making me take another look.

[tonatona]

sure, that message just means that the stuff loaded needs to be reorganized.

I hit OK up to nine times. Then i can sometimes lower the polyphony and reorganize the allocation, sometimes i have to restart the application and asio crashed too so i have to restart cubase.

If i use several small modular patches i can load more although there are interconnection errors.

I can load more than 90% of the Xite-1 power using the STM 2448 only with EQs and compressors. This is not bad though.

The Sixstring sounds better at 48KHz, it sounds different, a little bit dumb and unnatural at 96KHz.

Clear message: Stay away from 96KHz.


With 96KHz i have just a little bit better stage and depth, i can distinguish sounds in the mix a little bit better and it is a little bit easier to focus on quiet sounds in a busy mix, there are some advantages using multieffects with time-stretching and convolution reverbs with 96KHz impulse responses. The difference to 41KHz is very small. I get 30% more asio power in Cubase by overclocking the cheap Intel I7 920 CPU from 2,67GHz to 3,6GHz so power and money investment were not very important.

What if not sound quality was the reason to make the Solaris hardware synth 96KHz?

I do have sample libraries in 41KHz, 48KHz and 96KHz. I wish Jesus Christ did make 100KHz obligatory and the crusaders, CIA, KGB or Al-Qaida would eliminate everyone who would even dare to think about another sample rate.

[tonatona]

sure, that message just means that the stuff loaded needs to be reorganized.

I hit OK up to nine times. Then i can sometimes lower the polyphony and reorganize the allocation, sometimes i have to restart the application and asio crashed too so i have to restart cubase.
If i use several small modular patches i can load more although there are interconnection errors.
I can load more than 90% of the Xite-1 power using the STM 2448 only with EQs and compressors. This is not bad though.
The Sixstring sounds better at 48KHz, it sounds different, a little bit dumb and unnatural at 96KHz.
Clear message: Stay away from 96KHz.

With 96KHz i have just a little bit better stage and depth, i can distinguish sounds in the mix a little bit better and it is a little bit easier to focus on quiet sounds in a busy mix, there are some advantages using multieffects with time-stretching and convolution reverbs with 96KHz impulse responses. The difference to 44,1KHz is very small. I get 30% more asio power in Cubase by overclocking the cheap Intel I7 920 CPU from 2,67GHz to 3,6GHz so power and money investment were not important.

What if not sound quality was the reason to make the Solaris hardware synth 96KHz?

I own sample libraries in 44,1KHz, 48KHz and 96KHz. I wish Jesus Christ did make 100KHz obligatory and the crusaders, CIA, KGB or Al-Qaida would eliminate everyone who would even dare to think about another sample rate.

[dawman]

Playback at 96k isn't worth the resources needed. But 48k allows the outside signals into a project and add more clarity and depth. The analog synths coming into a project that is 48k only seems to allow better control of the effects and mixing accuracy, but in Solaris case it really is very noticable and I could never use 44.1k again unless the DAW shuts down. Even then I switch banks on the Solaris and the Privia PX3S are then routed into the powered monitors so the show must go on.
I lose the effects on the Piano and EP's but it's pretty crappy anyways.
DAW reboots and I am back in business.
After 14 years though never a crash once, so the advice I recieved I consider delivered from the Lord of sound himself..

[garyb]

when you go from 96k to 44.1k for cd, you will lose any advantage you might have had anyway.

[jksuperstar]

That's very true for mixing, but not really for EQs and Synthesis.

Anti-aliasing filters have a roll-off and phase shift to them near any transition band jut like any other filter. 44.1 gives almost no headroom unless you use either a very steep filter at 18kHz, or something smoother at 12kHz (notice that's the cutoff of many of the oscillator frequencies?). Using a steep filter makes for very bad phase distortion, going up to 96Khz means one can use either a smoother low-order filter, or place the steeper filter so far out of audible range that the distortion is not heard. These filters are built-in to the oscillators and some filters, so the sound generated is actually different. Besides, the Nyquist theorem is theoretical, not practical. The same applies to A/D converters.

I wouldn't say it's a deal breaker for me at all. But I would say in the long run it's worth the effort to get it working well. That's no small task, and certainly will take a long time of making small tweaks. It's a very nice option to have when the synth sound you are working with is right up front in the mix, and just sucks you in as a result of the extra cream. It will also keep the scope platform more relevant for synthesis, as the return of analog is making waves again in production. But SCOPE isn't a fixed architecture like other synths (even the Nord G2, which only has an 8 signal global bus connecting all DSPs.), so the issues are compounded when you consider routing, 3rd party devices, making almost any connection you want, etc. Like I said, it's a good long term goal.

Jimmy, wouldn't you like to run the Minimax at both 48kHz and 96kHz, and nullify them to see the difference?

[garyb]

no argument there.

[siriusbliss]

There was a small attempted trend a while back to push HDCD and Blueray audio, but alas, people are attached (literally/figuratively) to their i-devices playing mp3's.

The so-called high-res movement is still going on, but new compression schemes may have to be developed first for distribution over the net.

Otherwise for most music (IMO) you don't need high-res.
But that won't stop us from doing high-res synth or orchestra, or acoustic, etc.

My understanding is that a lot of film is already done at 192K anyways, so...

Greg

[Fluxpod]

Film is 48 K Land and will be for a very long time.Very very few exeptions in 96 and 88.1 but that is r-a-r-e.The track count in Post is still stressing systems.Even hdx satelite system clusters are not capable of 200+ tracks @192. You could make it work with a really really big system but the cost and crew for that wouldnt fit 99.9% of the budgets.

[neuromantik]

Sorry to resurrect an old thread, but has any progress been made regarding the 96kHz support on the Xite? I'm not complaining that much, but I've been wanting to generate some content using ModIV @ 96kHz (and compare it to my G2 which runs @ 96kHz by default flawlessly) but I've been getting DSP errors, despite the fact that my consumed DSP count is low. I'm only using modules from S|C, no Flexor ones (I'm aware they are not and will never be optimized fpr Xite). Are there S|C stock modules which I might avoid as well?

Additionally, on a related note, is there a matrix somewhere listing the plugins currently being sold on the S|C shop which are Xite-1 optimized (I know the Zarg ones aren't unfortunately, would have loved an 8 voice Quantum Wave)?

Thanks

ps. I don't want to stir up any feathers, I'm very happy with the Xite-1 Just want to sell the idea to friends, and they are serious about 96kHz ever since the Nord Lead 4 came out...

[djmicron]

not to be off topic, but while i can hear a big improvement switching from 16 bit to 32 bit, i can't hear a so big difference switching from 48 khz to 96 khz.
On native plugins it's another story, 96 khz is much smoother, but virtual synths such as U-He Diva, have a nice smooth sound even at 44.1 khz, so i don't think it's all about sample rate and the 96 khz is often stereotyped.
Very often mixing engineers are forced to convert the 96 khz tracks of their clients to 48 khz, because with a mainstream protools system it's too much to be processed and when the client is not there the engineer says "why the f@#k they use 96 khz!".
I like scope, because of this, it sounds good even at 44.1.
Then i agree about xite optimization is needed in the future.

[jhulk]

headroom is the only reason to use 96khz

and most audio processing is done in 32bits floating point and theirs no difference in 64bit

as seeing most content is 80-20khz nyquist states to record a frequency you need double the frequency so 44.1 is ample as 20khz double would be 40khz

now i have been sampling for 27+ years and most audio content does not even go above 12-16khz so i sample at lofi 22.5/24/32khz as those are the frequencies used by most older hardware samplers

and my latest sets for the scarcs sample osc are in those frequencies and they dont sound any less than there higher sample equivelents as when sharc was testing them i ask how they sounded in the modular and he said they sounded great even great sounding workstations from korg trinity triton oasys use 8bit compression on there sound roms to get double the amount of rom content at sample rates of 48khz

in fact when we used to sample for the EII on my digidesign soundtools macII we used to sample in frequency to conserve memory as its pointless in sampling a drum sample that only has frequency content from 80hz to 1khz so instead of sampling at the max 27khz we would sample at a lower frequency which conserved memory and allowed us to get the most out of the 400k 8bit compressed format for the floppy system as it compressed 1mb into 400k and then would compound that 8bits compressed sound into a 12bit audio output into the sample and holds to the filters

[Fluxpod]

headroom is the only reason to use 96khz

and most audio processing is done in 32bits floating point and theirs no difference in 64bit

as seeing most content is 80-20khz nyquist states to record a frequency you need double the frequency so 44.1 is ample as 20khz double would be 40khz

now i have been sampling for 27+ years and most audio content does not even go above 12-16khz so i sample at lofi 22.5/24/32khz as those are the frequencies used by most older hardware samplers

and my latest sets for the scarcs sample osc are in those frequencies and they dont sound any less than there higher sample equivelents as when sharc was testing them i ask how they sounded in the modular and he said they sounded great even great sounding workstations from korg trinity triton oasys use 8bit compression on there sound roms to get double the amount of rom content at sample rates of 48khz

in fact when we used to sample for the EII on my digidesign soundtools macII we used to sample in frequency to conserve memory as its pointless in sampling a drum sample that only has frequency content from 80hz to 1khz so instead of sampling at the max 27khz we would sample at a lower frequency which conserved memory and allowed us to get the most out of the 400k 8bit compressed format for the floppy system as it compressed 1mb into 400k and then would compound that 8bits compressed sound into a 12bit audio output into the sample and holds to the filters

Sorry but headroom and 96K arent related.Process depth for synths and plugins yes,headroom no.Headroom is entirely bit depth. In a modern daw with 32 bit float you have ~ 1400db to work with.No idea if thats enought for dubstep

+Having Speakers that exceed 40khz.I can tell you that you can feel it.Listen to a 128-192-256-320 kbit mp3 on atc 150 and you can not only hear but feel the difference.You can hear the algorithm work.This was demonstrated 2009 by George Massenburg to me and it certanly left an Impression!

[fra77x]

Headroom is related to the bit depth.

About the sample rates: Most people are confused. Does 96kHz gives better sound? How is that if i can only hear up to 20kHz?
Does downsampling a 96 kHz wave to media sample rates (cd, dat) makes the "extra content" dissapear?

The subject is quite different. Ok logic says that 96kHz perhaps articulates better the upper frequencies. But how about the low frequencies? How on earth these also sound better?

The answer is a lot simpler and has nothing to do with niquist, more audio content or anything like that. It's simple as hell by the way.

The sound is produced by machines. At the end of the story these machines just calculate sound and at higher sample rates the calculation is more steep and these processors just provide better sound when they provide more processing power for the same calculation.
That applies to the oscillators, the filters and the mixing process. Most of the time you can use lower sample rate envelopes and noone will feel dissapointed. Of course audio envelopes at 96 are more detailed than at lower sample rates but thats not really a problem. Its the oscillators and the filters that play the most important role. Some times i use 24kHz software envelopes just for their sample accuracy. No prob.

So all synthesizer developers at least the big ones just use 96 kHz to do their job. It's the matter that you just want to have better sound, dedicate your processing to the job and use most of it.

A simple test.

Put scope at 96. Record a simple audio from the modular (in VDAT please). Do the same with lower sample rates. By hearing the result you can immediatelly spot the difference. (cleaner) Downsample the 96kHz wave to 44100. The difference is still there. It's not a matter of something magic with the 96kHz, it's just that these machines (processors) provide better sound at that sample rate. That better sound remains at the downsampled audio because it has nothing to do with content beyond 20000. It's just the machines sound that has been produced and has been captured. (96kHz does not extend our hearing)

My 2 cents.

[dawman]

Solaris being internally processed @ 96k sure makes a big difference.
Running that @ 48k into the AES/EBU I/Os of the XITE-1 gives me an incredibly detailed powerful sound.
Had some old Barbetta Cabs, but then heard Solaris through the QSC k12s with modifications.
Went and played the stock cabinets at GC, then bought them and did the Reinkus Heinz mod and it's the best sound I ever had with synths.
To top that off, I also use the powerful subtractive analog SE-1 for it's smack down low end, and using the BX Digital post mixer/pre cabinets is the icing on the cake.
The mono maker takes all frequencies below a certain point and makes the woofier stereo signal mono. This really adds a tighter, more focused low end.
There's no such thing as work when things sound this good....

[Sounddesigner]

neuromantik i suggest you and others who want better 96khz compatibility with XITE-1 write Sonic Core and request it so they know a large enough group of customers want or need it. I work at 96khz and i wrote them years ago, Ralf Dressel at Sonic Core confirmed that some synths benifit from 96khz such as Poison, Vectron and Modular 3, etc but said some synths don't such as Minimax, Profit 5, Prodissey, etc cause they do oversampling. So Sonic Core confirmed what i was hearing.

SCOPE synths at 44khz still sound better than Native at higher rates simply because you still need a good algorithm apart from samplerates. Alot of Native plugins do extreme-oversampling and have 1001 parameter controls in substitute for a good algorithm and they just become more expensive crap sound due to the huge amount of cpu power they need.

96khz project Operation has MANY benifits. I know the realtime oversampling algorithm in many plugins sound like crap, linear-phase filter of oversampling algorithm can sound too constrained and make music sound more closed and minimum-phase has its own artifacts as well so like everything in audio its a trade-off when you use them and sometimes its not a good trade-off depending on the developers algorithm and the situation, thus often i get better results running my projects at 96khz and leaving the artifact ridden oversampling off. When working at 44khz and using plugins that oversample your more at the mercy of the developers realtime oversampling/conversion algorithm wich may not be that good, and running your audio thru several plugins with terrible OS algorithms may yield much worse results than just running your whole project at 96khz, and at 96khz you get much less latency than what the Oversampling plugins causes.

Latency is greatly reduced at 96khz. My Native Asio round-trip latency at 96khz is as low as 2.7ms round-trip compared to 5.9ms at 44khz, thus more than half the latency is gone at 96khz. Plugin latency, converter latency, plugin-chaining-latency, and some routing-latency is reduced as well thus there can be a HUGE difference latency wise with projects ran at 96khz vs 44khz. I'm fairly certain XITE-1 dsp platform latency is less than 1ms round-trip at 96khz, probably around .7ms .

There are other benifits of 96khz such as some poorer quality converters benifit sonicly from higher samplerates when recording. But for all converters even Dan Lavry who's not that caring for high samplerates says that the optimum rate for converters is 60khz cause there are some sounds that the humane ear can hear above 44khz wich is generally sufficient and the amount correlating with the human 22khz hearing. But 60khz is'nt a standard and 96khz is the nearest standard rate to that. The article is on lavry's website, i can post a link if needed.

There are even more benifits to higher samplerates such as future mastering and re-mastering, video compatibility, etc . I hope Sonic Core do work on 96khz compatibility with XITE-1 cause higher rates are for higher-end audio. Higher-end digital generally can't be achieved without this component (either thru oversampling or runnining whole project at higher samplerates is needed for high-end generally). You and others need to contact Sonic Core and let them know many customers want or need this.

That said, ATM i simply avoide alot of Modular 4 patches at 96khz and use the ones that have no problem. I avoide lots of voices with some synths and do other things to make 96khz work. Mod 4 has many patches that are more problematic than previouse versions of modular so you'll need to do alot of testing to figure out wich ones work best and can delete the ones that work worse.

I'm assuming the load balancing will improve with SCOPE 6 wich alone should improve 96khz operation to some extent, but that's my assumption only, no one knows what SCOPE 6 will bring regarding higher samplerate operation, but we can make request to Sonic Core who's doing the development.



EDITED

[Sounddesigner]

Solaris being internally processed @ 96k sure makes a big difference.
Running that @ 48k into the AES/EBU I/Os of the XITE-1 gives me an incredibly detailed powerful sound.
Had some old Barbetta Cabs, but then heard Solaris through the QSC k12s with modifications.
Went and played the stock cabinets at GC, then bought them and did the Reinkus Heinz mod and it's the best sound I ever had with synths.
To top that off, I also use the powerful subtractive analog SE-1 for it's smack down low end, and using the BX Digital post mixer/pre cabinets is the icing on the cake.
The mono maker takes all frequencies below a certain point and makes the woofier stereo signal mono. This really adds a tighter, more focused low end.
There's no such thing as work when things sound this good....

The Solaris is the proof of 96khz necessity for high-end sound. Soon as Sonic Core decided to build a high-end synth they made sure to use higher-rate so we know their opinions on this matter and we've heard the evidence . Sonicly Solaris to many i'm sure is considered the best sounding digital synth out there. I know my ears love it.

[garyb]

*warning* gross generalizations follow....

higher samplerates are not necessary for better sound. they are needed to make programming of filters easier, as aliasing can be ignored as the samplerate increases since the aliasing will be so far beyond human hearing. lower samplerates are more likely to experience aliasing that occurs near the range of human hearing. it doesn't matter if there are frequencies that a human can experience beyond 20k. almost NO playback speakers can reliably produce sounds above 18k and most only are usable in the 12-15k range at best.

basicallly 96k is something that makes crappy programming sound better. it also allows even good programming to require less attention from the programmer. if gear was made the way it used to be, in the early 70's and before when bic lighters were unacceptable and nobody had the stupid idea that they should get useful things for free, 44.1k would be more than acceptable. what proves this is that some early DACs, while having specs that are nowhere near the specs of the current hardware, actually SOUNDED better than almost everything made now. they're just too expensive to make that way. quality is always preferrable to quantity, except in the pocketbook.