ronnie,
I personally feel that mics and pres are the most important thing (besides raw talent and a great song). Nothing IMO can make as much of a difference. Good converters can improve things but you need to be generating a quality signal before the converter. It's not like the converter will transform a crap signal into a good one .... there's no such thing as a "good-u-lator" ..

Get some nice mics and Class A pres ... I prefer those that are transformer balanced (like vintage Neves or Great River MP2NV) but it's good to have a couple of nice electronically balenced pres (like Millennias HV-3 or a Grace 201) for tracks that need to be squeaky clean. Tube pres can be very nice also .... keep an eye out I think I am getting a Revolution Redd pre (a clone of the pres the Beatles used) to review soon ... I will let you know how it fares.

A few channels of good compression is nice to have as is at least a couple channels of good EQ ... Good all around comps are the Manley ELOP and the UA La2a and 1176 ... for EQs the Speck ASC or the Neve / AMEK 9098 (also a nice electronically balanced / clean mic pre) are very good.

Hi Kurt, thank you for your prompt reply, i've been researching.

Would you recommend the following as a good overall 2-channel 'signal path' to my DAW?

MP-2NV
http://www.greatriverelectronics.com/mpnv2.html

Universal Audio 2-1176 Dual 1176LN
http://www.uaudio.com/products/analog/2-1176/index.html

Matched pairs of the Speck ASC equalizers
(dead link removed)

Sorry i have to ask, but, what's the best method to actually converting using my 002? Should i scrap the 002 (or just use it as a control surface)?. I'm a bit confused, because the 002 has its own pres. Should i bypass the 002 and uses the digital interfaces available?

Also, any recommendation on a few professional quality mics for vocals and guitars would be appreciated. I should add that I am happy to pay you or someone else for consulting if it is appropriate. My intention is not to abuse the 'free board'.

Thanks in advance.

Ronnie

ps/ Can you comment on the Focusrite ISA 428 or the new Focusrite Liquid Channel?

Ronnie,
All advice here is free and that's why we are here.. but thanks for being so considerate. It's refreshing to say the least.

The pres in the 002 are better (supposedly) than the ones in the 001 ... but still not as good as the pres you and I have been talking about.

The signal path you choose would be very nice indeed. The 1176 can be a bit edgy in some apps ... which is why I mentioned the Manley ELOP or the LA2a ... they are smoother but not as fast. They also do not have all the attack, release and ratio selections the 1176 has. In a perfect world, one would be able to choose between both.

In keeping with the quality of the pieces you have selected, I would recommend a Neumann U87ai, AKG 414's for large diaphragm condenser mics. .. as well as AKG 451's, 460's, 480's, Shure SM81's or Neumann KM184's for small diaphragm condenser mics.

You will also need some good dynamics for drums and guitar amps. Sennheiser 421's, (great on toms and amps) 441's and 604's are all good as is the venerable Shure SM57 (for snare, toms, amps), I like the Beyer N201 for snare too ..... For kick drums, floor toms and bass amps the AKG D112e is good as is the Audix D6 ... Also worthy of mention is the Shure SM7a, if you can find one used (not the SM7b) or the EV RE20 for VO work, kick drums, bass amps ...

A couple of each would be great but you could just do with a bunch of 57's if you needed to ... It all depends how far you want to take it .

Thanks Kurt,

Can you address this for me?

Sorry i have to ask, but, what's the best method to actually converting using my 002? Should i scrap the 002 (or just use it as a control surface)?. I'm a bit confused, because the 002 has its own pres. Should i bypass the 002 and uses the digital interfaces available?

thank you

You have to use the 002 ... PT le will not run unless it is connected.

I think the best thing is to use the ADAT lightpipe inputs ... however on a 001 / 002, this will limit you to only 8 tracks at a time. If you can do it, keep the optical cable length to a minimum (the shorter the better) to avoid adding jitter ... if you can find them, glass fiber ADAT cables are available. They add much less jitter to the signal.

The pres and the converters in all PT gear leaves something to be desired for many recordists. With the TDM systems you can use AES inputs to get as channels of digital info to the box as you need by adding additional DD interfaces but with PT le I believe you are limited to one interface and therefore the 8 analog ins the spdif in and out and the ADAT port for a total of 18 channels at a time.

Thanks, so how do you physically plug in 8 XLR cables into the ADAT optical cable?

Is there a breakout box or something?

Do you know of a manufacturer off hand that you would recommend for such a device?

Thanks,

rw

Is something like this what i need?

http://alesis.com/products/ai3/

thanks, rw

You would need to purchase a different interface converter box that had mic pres and/or line ins and lightpipe outs. Many companies make these ... Aardvark, Lynx, RME, Apogee and even Alesis just to mention a few off the top of my head ... although the specs vary on these different box's, they all most likely use the same chips from Alesis (they have a lock on the lightpipe protocol patents) so the performance will have more in common than not unless you are using the internal clock of the interface. It is more common to clock off the host program (in your case PT) within the computer so in practice, if this is how you run the system, they all will perform similarly as far as the digital conversion. There will be differences in the analog parts of the circuits ... but they are for the most part, negligible.

Kurt,

Unless I'm mistaken, all the Alesis chip does is interface with the lightpipe. It doesn't do the actual analog-to-digital conversion. The converter chips (which do vary widely between manufacturers and price points) are far more important to the sound, not to mention the analog components. The only thing the Alesis chip will affect quality-wise is jitter.
Also you cannot lock a clock off the "host program," you can only lock to the clock generated by the PCI interface card in your computer. Usually these interfaces use relatively cheap PLL's and it's best to clock any device internally.

Hi Dasbin,

Can you recommend a converter box for my scenario (above)?

As i am begining to understand Kurt's explanations and recommendations, using high quality hardware based mics/pres/eqs/comp and the 002 adat optical interface will result in the 'optimal results'.

thanks in advance.

rw

dasbin,
Thanks for the claification ... I use a Frontier card and AI3s and the AI3's are clocking off the PCI card. I notice a better, more stable sound (better bass and spatial imaging) when I clock this way instead of clocking off the AI3's ... and that's what I am relating to. Doesn't ADAT lightpipe carry the clock info within it, thereby negating the need for an external wordclock connection? This is how I have always understood it is.

As far as the chips, this is the first I have heard in this regard. All that I have learned in the past supports these manufacturers are getting their converter chips from Alesis but I would not be willing stress the point without more info, input, data ... I have every confidence in what you are saying ... you are usually correct in your comments.

So won't I interface the A3 with the 002? Or do i need a PCI interface card as well?

No you can go lightpipe out from the AI3 to the 002 ...

Great, any thoughts on the actual converter? Dasbin thought i should consider that as well? I assume that my current ones are integrated into the 002?

Kurt Foster wrote: dasbin,
Thanks for the claification ... I use a Frontier card and AI3s and the AI3's are clocking off the PCI card. I notice a better, more stable sound (better bass and spatial imaging) when I clock this way instead of clocking off the AI3's ... and that's what I am relating to. Doesn't ADAT lightpipe carry the clock info within it, thereby negating the need for an external wordclock connection? This is how I have always understood it is.

When I try and run PT off the "internal clock" with my RME interface, not only does it sound like crap, but it's completely unstable and jittery. I always let the RME clock do the work, in fact, I think that in PT, you have to run it that way or else you'll get static and pops and clicks.

Kurt Foster wrote: dasbin,
Thanks for the claification ... I use a Frontier card and AI3s and the AI3's are clocking off the PCI card. I notice a better, more stable sound (better bass and spatial imaging) when I clock this way instead of clocking off the AI3's ... and that's what I am relating to.

Clocking is really such a complicated thing that it varies wildly setup-to-setup. In your case, because you have multiple devices that need to share a single clock, you're better off choosing a physical "mid-point" - in this case the PCI card. This is because daisy-chaining wordclocks from one unit to the next can be a bad idea, as can making it go through the additional PLL inside the PCI card after being generated in one of your converters.

Doesn't ADAT lightpipe carry the clock info within it, thereby negating the need for an external wordclock connection? This is how I have always understood it is.

It does carry it... but it's far from ideal. There's so much other digital (and analog, actually) crap on the lightpipe that seperating the wordclock signal is a tough task on the hardware and very difficult to do without introducing jitter. I'm not certain if the Alesis units have wordclock ins/outs or not, but if they do you'd likely be better off using an actual dedicated 75 ohm wordclock run. In this case setting one of the converters to master is the best bet if you don't have an external generator.

As far as the chips, this is the first I have heard in this regard. All that I have learned in the past supports these manufacturers are getting their converter chips from Alesis

This is interesting and I'm geniunely intruiged as to what exactly "all that I have learned in the past" is. I spent the last 20 minutes or so trying to dig up some actual data but there is a distinct lack of it on the internet at least. From an electrical design standpoint, it simply doesn't make sense for a digital data converter to be doing audio conversion at the same time. Such integration is common in the consumer market but essentially nonexistant in prosumer/pro designs (except RME, which I believe actually uses a relatively cheap CODEC chip in its Multiface... which just goes to show just how much of a difference a good analog section and clock can make).
So far I haven't even found evidence that Alesis makes standalone DAC/ADC's, let alone ones that are integrated into a lightpipe chip. That's a big leap of technology. Most converters on the market are made with the chips from one of two (AKM and Cirrus Logic) manufacturers (three if you count Burr-Brown which doesn't appear in many pro designs at all).

EDIT: http://www.alesis.com/downloads/manuals/EC2_Manual.pdf Here I have found a manual for one of Alesis's own products with lightpipe capability. This is a manual for a converter upgrade. The interesting part is:
"The EC-2 upgrade uses premium AKM 5393 analog-to-digital (A/D) and AKM 4393 digital-to-analog(D/A) converters..."