Definitely a processor issue, get something that can run the latest graphics intensive games smoothly and you're good to go... Also changing the audio interface wont do anything to remedy this issue, unless you invest in an interface with onboard DSP, but even then the DSP usually only works for the plugins that the DSP was designed to handle i.e. McDSP only uses DSP on McDSP plugins, so all your other plugins still hog up CPU processing power to do their tasks...

 

Just this at all the posts after my post...



40 Ozones in one project, great job mate, good luck... 192kHz probably recorded at 64 bits for purity? But made through a cheap preamp/cheap audio interface? Yeah, it makes so much difference with such an interface. Good luck to you, too... try 384kHz I know just the right interface - DAD [http://www.digitalaudio.dk/AX24-ADDA-Converter.1492.aspx] It's just measly 7000 quid but it does do it properly as opposed to a 150$ USB audio interface ...then some unknown high end USB interface, good luck mate!... monster cables... banana oil... bird milk... Earth magnets... Earth minerals... elephant fart for potency... giraffe milk for hair growth... get a life people! [meaning read around and learn, not screw around and talk through your behind hole].

To top it all - USB makes noise? good luck mate. How does USB make noise? USB hard disks, you mean? Who would use that for audio?? eSATA is made for that.

 

@ SineWave,

'DOUBLE FACEPALM' ​

It exactly confirms the 'so famous' Theorem of Mr. Nyquist !!! *yes* ​

​

For some users, and as short as possible, because there's hundreds of articles on this topic through Internet...

Sampling Theorem

The 'Nyquist–Shannon' sampling theorem states that perfect reconstruction of a signal is possible when the sampling frequency is greater than twice the maximum frequency of the signal being sampled, or equivalently, when the Nyquist frequency (half the sample rate) exceeds the highest frequency of the signal being sampled. If lower sampling rates are used, the original signal's information may not be completely recoverable from the sampled signal.
For example, if a signal has an upper band limit of 100 Hz, a sampling frequency greater than 200 Hz will avoid aliasing and would theoretically allow perfect reconstruction.

The full range of human hearing is between 20 Hz and 20 kHz.
(Apart some very rare exceptions... )​

The minimum sampling rate that satisfies the sampling theorem for this full bandwidth is 40 kHz. The 44.1 kHz sampling rate used for Compact Disc was chosen for this and other technical reasons.

So, I'm agree that we can't stop progress and technology... BUT why this unbridled race to such spectacular (and foolish !) Sampling Frequencies... apart perhaps for some rare species of bat (?)
Although, it's true (in some occasions) that Sampling Rates of 96kHz can mainly help with Orchestral, Acoustic,... Instruments to get (perceive) the accurate, detailed, subtle nuances played by an Instrumentist (or Orchestra), but above all, it's his/her subtleties and playing nuances that will do the 'trick'...


I personally think that the 'trick' here is rather the 'Bit Depth (Resolution)', far more important, to get more and more accurate and detailed sonic and sounding qualities as the progress and technology goes on... but this is another topic... *yes*

 

Is really a I7 950 3.1ghz a problem??? I work with a old 1055t 6core from amd which is similar to I5 as performance and i really don't bother with latency problems.
He asked for advices about a good interface to work with, not to upgrade his computer parts in conjunction to his latency optimisation.

Look Olymoon, let me give you a straight answer regarding to your topic:
My recommendation would be, RME Fireface UC or UCX, they both have the Hammerfall core which provides ultra-low latency that really compete with PCi-e cards, and also goes up to your desired 192khz sample rate having great ad/da converters. In my opinion, they are the best possible low-latency interfaces in the USB World.

 

Very nice discussion.I have to say my experience to add to the pile.When I upgraded my computer to a MSI gaming laptop is when I ceased to have major issues with latency.My laptop uses Windows 8, 120Gb SSD,1Tb standard HD for samples,32Gb DDR3 1600Mhz RAM with 3rd Gen i7 and 3Gb Nvidia GTX680 chip.CPU is only 2.4Ghz with hyperthreading to make it act like an 8 core and I am amazed at the results for recording music.I can load more plugins than one really needs without overloading the CPU or getting dropouts all the time.I am using a simple cheap Edirol UA1-G interface,Yamaha HS50 monitors and Sony MDR-7506 headphones.When playing guitar live or recording I can run at 64 samples using Amplitube on high resolution,Addictive Drums and Kontakt with bass and anything else I want with no crackling or dropouts.I also use ASIO4ALL.I only use 44.1k as well.A rock song can use up to 40 plugins....effects,kontakt,etc and I have never had to freeze tracks from VST overload.I will reiterate the finer points as those before me have,so well I might add...........It goes like this-What OS/CPU/RAM/Drive are you using,what monitors/headphones do you have.I have tested out various usb interfaces and they all do well on this system.I stuck with the Edirol UA1-G cuz it is portable,works well and has alot to offer in such a compact interface and doesn't use an adapter for power.I don't need anything beyond 2 inputs.For playback.....here is the clincher......my onboard realtek sound chip sounds great to my monitors direct too!!!! and also doesnt suffer latency issues.Although I do use the interace with my monitors.So.........if you dont need inputs chances are you should be focusing on a new system that uses 3rd or 4th Gen i7,a motherboard noted for having killer onboard sound and lots of RAM,32Gb or more and top it off with some nice monitors.I recommend the Yamaha HS series........really awesome for the money.I picked up my HS50's $200 for the pair.I also do classical music and have projects with 40 instruments in 1 instance of Kontakt.I have had the occasional skip in the sound but it seems this only happens when I am tweaking in realtime on some plugins and will happen even if it is the only plugin.Hope this helps you out some.

 

CapnCrunch Thank you for telling your EXPERIENCE.

I wouldn't want to seem harsh but at the beginning of this post, I said that I am looking for EXPERIENCED users.
Not about opinions.

I said I WANT 192Khz interface, I don't ask if I am wrong.
But as many people seems not to understand I explain.

Sonically: Sound is a sum of frequencies modulating each other, its not because you cant hear a frequency that it is not here and modulating the other.
Every body understand that about low frequency, you have to know that the same happens with higher frequency.
When you listen to a piano a lot of its sound is determined by its highest harmonics that interact with the other harmonics and this give part of its rich sound, even if you cant hear this harmonics.
So when using vsti, it is true, as someone said here that you are limited by the frequency that this particular vsti is programed to emit, but when in a mix you have several vsts, the inter-modulation between their sound may produce higher frequency than they are able to produce alone.
So with a higher sample rate, you are closer to get this richness in sound, because it will modulate frequency under 20khz which human can hear.

Latency: Higher sample rate = lower latency

Investment: I don't buy an audio interface thinking that I will buy another one tomorrow, when the rest had understood what 192khz is good for and it became a standard. So last technology allows for 192khz, why not get it?

Looking for a lot of information I come to the conclusion that, yes of course there is a calculation problem. But I doubt that this is from the computer part.
(And yes I can play the last games at higher resolution, like some says).
My CPU Load is never beyond half, this is a 4 processor that goes 8 with hyper threading. My Ram either.

But it seems that the calculation problem will be solved with specialized calculators: High end audio interface/DSP. (not sure)

All articles I find are sound engineers oriented, they speak about latency when recording, not when using a lot of vst/vsti.

They are not cheap, this is why I asked for EXPERIENCED users to tell me if they used an Apollo, RME etc.., composing with a lot of vst, did they had a change that worth the price?

Some notes:

The monitors are important? Yes but this is not the question here.

I am working on movie music.

 

Then buy what I have as DA - TC Konnekt 48. You can get it for like 600 quid these days, maybe less, with a remote. It's on sale now. I can only speak highly of it. The only thing that it's lacking is 8 preamps, it has 4, but they're worth 8 . I have additional 8 channel preamp anyway so I don't mind. The converters are great, both DA and AD. Look it up. It's much much cheaper than RME UFX but about the same value and the ADDA sounds better or the same as any RME interface. Or buy RME Fireface, but it's more expensive and you don't get TC's great DSP FX and a remote. I love its remote, it comes so handy and you can even send MIDI CCs with it... VERY versatile audio interface. I haven't used it at 192kHz yet as my whole system is 96kHz "oriented". It took me a very long time to decide what to buy for DA, Oly. Very long time. Years. You won't find anything better within the price range of about 1000-1500 quid, and Konnekts 48 are on sale now. Grab it while you can. They're probably making MKII version which will cost around 1000 quid or more what Konnekt 48 cost before the sale.

I've been also thinking about Motu 2408 for years, too. What I've been looking for is a handy audio interface that has great ADDA and can work as a standalone for location recording. Either would do. Motu 2408 also has OK DSP, but not nearly as good as TC and the ADDA is not that great. It is also pricier than TC Konnekt 48 now.

Cheers!

 

SineWave Ok, this is concrete information. Please could you tell me do you have sessions with a lot of tracks calculating?
What are your computer specs?
What sample latency buffer do you use? (128, 256?)

I like the TC Electronic interface, and the remote control, but in the TC Electronic web site they say firewire 1394 that mean 400 so this is old interface.
If I go firewire, I'll go 800.

 

I use it as a computer interface only when I'm recording on location and that's like once per month. In studio, I only use it as AD for microphones so it is connected to a RME 9652 with ADAT cables and WordClock. I use Konnekt 48 with a little single rack Intel atom based computer that is not really powerful but it can record many tracks at once anyway. I always use 256 and only sometimes 128 samples of latency. Usually I go with 128 when recording singers. Speaking of which, Konnekt 48 has great DSP reverb that I can control with the remote and use it as "comfort reverb" for singers or guitar players, and it has 2 headphone outputs so I can have headphones and the singer or a guitar player can have headphones, too.

Why do I have such a setup? Because there's no card like PCI or PCIe card and there's no ASIO driver like RME driver. However, I never had any problems recording at low latency with an atom [!] based computer, so that tells you a lot. If I had to, I would always gladly use it as my main audio card now. There were some problems with the TC drivers before but they have ironed them out and it's rock solid now. It also has a great and easy to understand mixer app.

Firewire 400 is more than enough bandwidth for 48 channels [at 44.1 or 48kHz] that you can record and play simultaneously through it. It's not old technology, but rather tested and tried. ;) Newest technologies can only get you in trouble... say, if I bought this TC in 2008, I would have such problems, too. You can find many old threads about it having problems with drivers and even knobs. I can safely say that ALL new interfaces have some problems of various kind, so it's better to buy a few years old technology and avoid the headaches. I like to play it safe and it works, I tell ya. ;) RME HDSP 9652 is even older tech, but damn it works good. Just damn it works so god damn good. and RME still update the drivers regularly.

Read the manual, get to know it. When I researched audio interfaces I had downloaded all the manuals and studied them. It's never a waste of time as it might come useful some day to know how to operate Fireface 800 or UFX, or MOTU 2408 etc.

 

I recon that going beyond collecting knowledge, I see that I'll probably have to go for pcie card, (there for abandon the USB criteria) because these are the one that really come to speed and high calculation.

SineWave So the way you use is very different of want I do.
I almost don't record. If I need to record I prefer to go to a good studio, with high end mics and acoustic, then I come back to my DAW with a high sampled raw tracks.
You don't need so much calculation power.

I agree that RME is one of the brand that may fit my needs.

 

Yap, I use RME 9652 for digital playback at 96k, connected with SPDIF to TC BMC-2 DA which has the same converter as Konnekt 48, but it's more useful since it's got big volume knob and VU meter and speaker controller... and I just love to have a DA on my desktop like that. Although, I could actually use Konnekt 48 instead both for AD and DA.

USB and Firewire both always mean higher latency. There's also latency they don't mention and it's usually around 64 samples. Well, at least Fireface 800 has 64 samples of latency that adds up to the usual ASIO latency. it's not much, but PCI and PCIe cards don't have any additional latency.

The new RME card that supports 192kHz and is a replacement for HDSP 9652 is called RayDAT, if you're interested. It comes as PCI or PCIe card, your choice. PCIe is better if you have an Intel computer with a newer chipset that doesn't support PCI any more. I have AMD computer so I'm good with a PCI card. However, if you don't need so many outputs and inputs I'd suggest you to have a look at new HDSP 9632 PCIe. it's about half the price of RayDAT and it's got analog I/O, headphones out with direct monitoring, good converters, and digital I/O that you could use with an external DA or AD.

 

m2314

This is exactly what I am afraid of. I don't want to buy an audio interface only to discover that it is made to handle expansive proprietary good hardware imitation for sound engineers.
And this is why I ask if someone have experience with them.

 

I don't understand. Of course the DSP on Konnekt 48 and Impact II TC interfaces supports only TC plugins. What's the deal with that? Same as Powercore supports only Powercore plugins and UAD supports only UAD plugins. You can't offload normal VST plugins to any outboard DSP. Although I must admit I almost never use them when I'm not recording on location. I find VST plugins better quality. DSP plugins are only really useful for recording IMO. But I mean REALLY USEFUL. Because there's no latency when you use a DSP reverb. It goes directly through the interface to headphones out, and limiter/EQ without any latency is really useful for recording so you don't get any digital overs in the recording.

Unless you have Reaper and two computers like I do. Then you can offload some VST processing to another computer via Reamote. It works rather well. ;)

Since you're not recording anything, Oly, I agree that DSP FX would not be useful for you at all. Maybe reverb, but you can have only one reverb and natively you can have as much as your CPU supports... this one reverb is quite good, though. Limiter is good also. I don't use its EQ at all.. well, just as HP filter, actually, when needed.

 

"40 Ozones in one project, great job mate, good luck... 192kHz probably recorded at 64 bits for purity? But made through a cheap preamp/cheap audio interface? Yeah, it makes so much difference with such an interface. Good luck to you, too... try 384kHz I know just the right interface"

You didn't spot sarcasm did you?

except for usb noise part, if you have a sound interface running on same bus as a usb powered harddrive or a fan, it does bring noise.

 

I would recommend that you try an access virus TI2, it has USB audio that works really well and will take a lot of the stain of your processor when using plugins. It appears that the better quality Vsti's you use, the more CPU they take up and the virus TI2 has plenty of high end free sample libraries, it is also 16 part multitimbral and is good for layering sounds. Although you are limited to three audio outs via USB you can use the other six mono outputs on the back with a mixer or audio interface giving you six stereo outs in total. It also has good a on-board Fx processor and can display each timbre as a Vsti in your DAW making it an Ideal replacement for high quality plugins.

For a DAW I would recommend Ableton, because it has a downsampling function that allows you to master a mix using 96khz to a 48/44khz format without losing much, if any quality. This means that while you are mixing your track, you can set the rate to 48/44kkhz and not use to much CPU, but just increase the rate to 96khz when rendering your final mix.

 

twinny123 This is a complete different angle, like old time when I had a 24 track soundcraft and synths , racks, samplers and so.
I hope I can solve another way.

SineWave
If expansive audio interface don't relieve the computer of vsti /vst calculation, then they are not the solution for me.

Not a bad idea.

 

Olymoon, I don't know of any audio interfaces that you can offload VST/VSTi processing to. There's no such thing, no matter what the cost, and understandably so because VST and VSTi plugins are programmed for x86 and x64 CPUs and only Intel and AMD CPUs under Windows, Linux or OSX can run them. Or a thingie like Muse Receptor, but Receptor is a computer also that runs custom version of Linux, and it's an expensive one.

Nowadays you can assemble little but powerful computers for peanuts... and if you can offload just a few synths to it, it can be a great help. But Reamote is most useful during mixing as it introduces latency to/from, so it's not very useful while you're just playing and arranging. Essentially, both latencies of both ASIO drivers on both computers add up and plus the latency of plugins and network cards can give you "nice" latency in the end. If you use 128 [x2+LAN] samples of latency on both computers, you can actually play synths normally, though. At 64 [x2+LAN] samples the latency would be pretty much bearable as you probably know it.

I'm actually looking forward to replace the motherboard in this small computer this year or the next for something more powerful because I want to be able to offload more VST and VSTis to it. For recording only it is absolutely fine. Atom is really slow, but it doesn't put out much heat... and I like to cool the CPU either passively like this Atom, or with a really quiet fan. Both AMD and Intel will soon release interesting 10-20W CPUs that are like 10 times more powerful than Atom. I'm especially eyeballing this new AMD that will consume only 15W and have a decent VGA onboard, too. We have to wait and see. If you're not bothered by noise some nice Intel i3 or AMD A8 will do just OK. Something cheap but not too cheap CPU, and cheap but not too cheap mATX mobo. ;) But I think these new low power CPUs are worth the wait.

 

@Qlymoon, like many other said, no audio interface would give you better VST(i) performance. A rather cheap option would be a PCI/PCIe E-MU card based on the 1010 (e.g. 1212m). These won't improve the VST performance either, but they have a dedicated DSP for the included hardware-accelerated effects, so you could use some EQs, Comps, Delays etc. without using your CPU.

Br,
Olaf

 

If you are not going to record you should find a motherboard that supports 192k.......I forgot to add in my post that my onboard realtek soundchip supports 192k.Perhaps you should look into a gaming laptop like I did.

 

Ok, Thanks to every one who's participate here, so it seems that we are step by step coming to some kind of answer.

So far, we can say that Audio interface wont relieve the calculation of vst/i plugins.

We can say that some audio interface, like pci/e ones comes with their own plugins, and using these would relieve the computer, there for allow less latency.

Olaf In EMU website, I see the 1616M, which seems closer to my needs. (Some times I have to plug a guitar or a mic)
That claims : "Hardware-accelerated effects - over 600 standalone and E-MU Power FX VST plug-in effects with no CPU overhead" so this could help.

Then looking further in the same direction I found this: Sonic Core SCOPE Xite-1

Seems very interesting as it does exactly this, offer synth and effect plugins which power consumption is assumed by the card.

Seems the same approach than ProTools HD Native

Both with their own format, so not very flexible, to try software before you buy (If you know what I mean)

But both are very expansive and they are not enough opened.

So what could be the other options, If it is a calculation question alternative approach would be to change OS, may be with Mac os and Logic, I could have a better result.

At least, this I can try for free.