The 1616m is also fine. It's the same 1010 as the main card, just with the 0606m dock instead of the 0202m daughter card (1010 + 0606m = 1616m). However, the 0404 (PCI/PCIe only) has a less powerful DSP.

Yeah, because it's hardware and not software.

Most likely not. I believe Logic doesn't have a native VST support anymore (you have to convert them to AU) and besides, do you have Mac drivers for your interface at all?

 

Problem is that even if EMU cards or similar could help a little with own effects, that wont be a lot, 'cause I don't use a lot of effects. I prefer tweek the sounds directly in sampler /synth. So if does not bring help for synth and samplers, i need an other solution.

 

...And there's no other solution for offloading VST and VSTis than Reamote, Oly. Really. Steinberg is the only one who devised a similar solution and it's called VST Link, but it works over SPDIF. Reamote is so much easier to set up since it uses the network cards. The only requirements are that you have functional TCP/IP protocol and network cards on both computers, Reaper on both computers, and same VST and VSTi plugins on both computers. It's so easy to make a second laptop or a computer you have an additional DSP resource that way. I don't think many people realise that... or know of Reamote. It's ingenious and so logical. Why other DAW software doesn't have it is beyond me. Most of them, like AVID/DigiDesign are just trying to make you buy more of everything, and making something like Reamote is not good for that. It makes it too simple to stay with the old software and OS and just connect them too easily to use the DSP resources so you can get away with a few very cheap computers that way...

There's also something called ReaStream that Cockos invented. It works similar to Rewire but sending Audio and MIDI over the network. That's one of the reasons I find Reaper so incredibly good since you can do pretty much everything with it. You can have like 16 computers in a network connected to a network hub [for less latency] calculating VST DSP for you all from one central computer... If you can invest in such an "audio rendering farm" and if you need it in the first place. It would be really too much. I find it hard to even clog one computer's CPU with VST and VSTis. Of course, it has to be a good computer with a good CPU like my trusty Phenom II 965. Or some trusty Intel i5 or i7.

btw. E-MU 1616M is a great card that I often recommend to the people on budget. You can't get better ADDA and usability than that for the price. And it also has a somewhat usable DSP FX that run on its DSP CPU. However, then we're not talking about the same ballpark as Konnekt 48 or Fireface 800, Apollo etc. It's just a cheap prosumer card that is really good and gives a lot for the price but contains compromises in the design to make it cheaper and bring more profit to the capitalist slavemasters.

I still think you're best off using some cheaper but great PCI or PCIe card with great drivers [even M-audio Audiophile 192 would do] and sending the digital audio out and in from and to some external audio card like TC Impact II or Focusrite Scarlett.

Cheers!

p.s. Scope is a good joke. Expensive one. Buggy drivers. I've worked with it. One guy I know have Scope in the studio. I say never fall for anything that costs more than a 1000 quid. That's my rule, except for the monitors. Speakers, I mean.

 

I hope I'm not talking gibberish to you, Olymoon? Do you completely understand what I'm saying in my posts? You can ask me to clarify something, don't be ashamed mate. As I said in some other post - audio engineering these days encompasses so much knowledge, it's a science on its own, every part of it, from computers to sending the finished CD to the replicating plant or uploading to the Soundcloud. *yes*

The easiest part of it is to be an artist, or an "artist". Well, that's debatable, too, if you're both as in my case... My brain gets an overload of info at times... so I have to use some means, at times illegal means, to ease the pain somewhat. It's hard to cope with so much input at times... Thankfully I also have my dearest wife to keep me sane, too. And I cook and make healthy food for her and for me as a hobby. That relaxes me. Not to mention kvass and sauerkraut give you so much health benefits for almost no effort, or money at all. Great shit. [errr literally. drinking kvass and consuming sauerkraut regularly does give you great shit and rids you of all the stomach problems like heartburn. ]

Thankfully [2], I got rid of the "social networking" BS. That thing is so painful. I rather go to a pub and have a pint or a dozen with the chaps. That's real social networking. But it doesn't happen as often as I'd like.

 

SineWave Thank you for all your explanation. Yes I understand perfectly, I know a bit about computers and networking.
And no I wouldn't feel ashamed to ask. I begun this thread to find answers so...

2 questions then,
The final audio interface should have 2 digital input, right?
Didn't you mention latency about this solution? Do you think vst instrument could be played at 128 samples buffer?

 

http://www.cpubenchmark.net/cpu.php?cpu=Intel+Core+i7+950+%40+3.07GHz
One more thing.. Your trying to find a better audio interface.
It seems more like you need a better cpu + 16gb ram or more.

That would take care of your problems.

Sure having a good sound card/audio interface is another reason.

I have the feeling that upgrading your cpu + decent interface (pci/firewire/usb) would do a better job. Than dropping 7k on a top end new audio interface.

P.S. You never post it "Ram information". Which i suppose is either 8gb or 16gb.


Also if you are making "film production". Working at 192khz would rarely do any good with vst/kontakt/synths

192khz is use for recording/working on high classical music. Where you really want to capture the extreme detail of a string instrument(s).


96Khz is use for film production.

48khz can be use for modern music/commercial music
44.1khz is the standard for commercial music

44.1khz and 48khz have almost no difference, other than 44.1khz being for cd.



90% of users/clients/customers won't notice any difference between all this Khz. Only way to notice a difference is having a 10 thousand set of speakers and a high end studio equipment to sample/work on it. (Specially at 192khz)



It was late! My bad. i miss it lul

 

xHitoKiri Thank you for your advices.

You say: P.S.

I stated that at the beginning of the post:
I know, some will say that I should freeze my tracks and so. But I want to fell free to change sounds and so while composing, so I'd like to find a sound card that allows this.
My computer is powerful enough for this: i7950 @ 3.1Ghz, 12Gb DDR3 corsair Ram on triple channel, ATI HD 5770 graphic card 1Gb of GDR5 ram.

Besides, meanwhile I have been looking for more infos, and I just discover that the reality is simpler and worth than what I thought. The problem is only a question of driver,(@&# Avid). and no it is not the same with asio4all.

I could try a very similar simple audio card as mine but from another brand with well made drivers and I could go down to 64 samples buffer at 48khz.

Why did I wanted to go up at 192khz? As I stated before, because higher sampling frequency = less latency.

So my conclusion is: now I only need to look at the AD/DA quality question, as I guess that any professional brand will provide good drivers.

Thanks to all participant here as each one has brought knowledge to this thread that surely will be helpful for many persons.

 

I'm on core 2 duo and 96 is no problem for me.

 

devilorcracker 96 What? Samples buffer or Khz sample rate?

What is your sound card?

DO you use a lot of Vst/vsti plugins together?

 

I've been struggling with the concept of high end AI's and my style of composition... I always was under the understanding that a high end AI was crucial when tracking only. But isn't this true of MIDI composing and MIDI latency as well? I don't understand how minimizing the buffer can cause audio to break up but there be no adverse effects when maxing it out.

 

Bump

Minimizing the buffer oblige your computer or pci card to calculate all audio at the same time, tht's why it causes problem resulting in glitch and noises, when the computer annot calculate all fast enough.

Maximizing the buffer let more time to your computer to calculate but then add latency, so when you play you dont hear the sound immediately, which can result in, no pleasure and out of tempo.