The Digi 002 allows for a max of 18 inputs at once.

8 Analog
8 ADAT
2 SPDIF

If you want to bypass the 002's converters you have to use either ADAT or SPDIF. Both at the same time will give you 10 channels of audio that bypasses the 002's converters.

I'm not familar with the alesis but I don't think it supports SPDIF, and I highly doubt it's converters are any better than the 002's.

As for the AD-16 it only supports ADAT, and the 002 only has one ADAT in port so you can only get a max of 8 channels from the Apogee.

If you use the 002 with the AD-16 you can take 16 all together. 8 though the adat port, and 8 from the 002's analog ins. This will give you what you need. Also if you lock the 002 to the AD-16's clock it will improve the 002's conversion. So if you run it like this you will get 8 channels of excellent conversion, and 8 channels of decent conversion.

But then again you can only go up to 48K, but that's not that big of a deal.

Re: 16 channel Digi002 recording w/ Apogee AD16x and Alesis

mikecornett wrote: Hello,

I currently run an 8 channel Digi002r setup. I'm in desperate need of being able to run 16 channels, and am heavily interested in the Alesis Ai3 to add the other 8 channels. I'm not too please with the A/D converters on the Digi002. As of now, I mix digitally, and don't run all of the channels back to the board except for the stereo monitoring, so I'm not in desperate need for other D/A converters quite yet. I know that the Ai3 has a/d/a converters in the unit.

I'm really interested in the Apogee AD-16x which has the steller A/D conversion which includes the Big Ben word clocking hardware. I'd like to be able to use this unit but am confused on how I can effectively use this with my Digi002 and Alesis Ai3. I understand that I'm able to bypass the A/D converters in the Digi002, but don't know if it's possible to do so with the Alesis unit. For a true 16 channel recording WITH the Apogee AD16x, am I forced to buy another Digi002 just to be able to bypass the A/D converters and get another 8 channels, or can I do this with the Alesis?

Thank you kindly

Heya

What John said is true, ADAT on ALL Digi's hardware only goes to 48Khz, which I can't blame them for. ADAT is an OK media for audio.
Two ways to improve the ADAT recording--1) clock via word clock and not via optical. 2) Use glass fiber Optical cables!

Since it's a 002 it doesn't lock to word clock since it has no word clock connections. I think that is a silly design if you ask me!

The Big Ben Clock aka CC776 is a software based DSP Clock. No hardware in a sense...just some eprom's and the usual caps and resistors :P

Also the fact that having an AD-16 X would be nice but theres no way to truly use all 16 channels at once. So here's what you do...

Rosetta 800- you have 8 in and 8 out. 8 out to the 002 via optical. 8 out to the Rosetta 800 via optical. This way your DA is taken care of! Yes, AD is the most crucial stage of recording but the DA can be a crucial stage as well. If your DA is garbage, no matter what is clocking it, it can make your mixes sound bland....

The clock in the R800 isn't Big Ben but it's the original Apogee clock which I'm sure a lot of people will agree that it's still rock solid! Good enough for the job you need and the gear you have.

This way with the Apogee clock you can improve the sound of the digi's converters and have a good DA to use!

How's that!

Opus :lol:

I still need the other 8 inputs/outputs to utilize 16 channels....even if I were to use the alesis unit for the added outs, i couldn't use the apogee a/d? I want to make sure I have 16 ins that all use the same apogee a/d converter. Is this impossible with the digi 002?

Yes it is impossible ... 8 in/out on the ADAT lightpipe, 2 digital in/out on the spdif ... and 8 analog on the mic pre / line ins on the 002 ...

would I have to get another digi002 unit to do this? I know that digidesign's hardware reacts differently than most interfaces being that the software works only if the hardware is connected, so would the computer be able to work with 2 digi002 units at once?

There's got to be a more affordable way to get 16 channels of audio and using the ad16x while working with Pro Tools, and without having to go with the HD system.

Unfortunately LE systems you can only use one hardware device!
It's not like TDM systems where you can add more PCI cards to attach more interfaces.

That's the limitations of LE systems...track count and hardware interfaces.

Opus

Mike,

You shouldn't stress that much over the 002's converters. There are alot of things that will improve your recordings, converters being the least of which.

The best way to improve your sound?
(1)A good sound source
(2)Knowing how to Mix
(3)High End Pre-Amps, Mics, Compressors, Ect.
(4)Then maybe converters

This is just my opinion however.

If I were you and I needed to take in 16 channels at once with the 002, I would record all the critical stuff (Guitar, Voc, OH's, ect.) with the Apogee's converters and send it to Pro Tools via the Adat input. Then I would use the 002's 8 analog ins to record the less critical stuff. Use the 8 analog ins for the kick, snare, and toms. I'm not saying that these aren't important, but you can easily make them sound good using drum samples to either reinforce them or totally replace them.

Trust me, you can get a very decent sound this way. It comes down to the engineer, not they equipment.

But if all else fails get rid of Pro Tools! I myself am a Digi 002 owner and I can't wait to get rid of that peice of crap! I'm goin with Nuendo/Cubase, Digital Performer, or Logic. That way I can have some flexibility and won't be stuck using Digi's lame hardware.

Johnjm, thanks for the reply, but I've already boiled it down. I know how to mix, I have high end preamps and compressors, I have a great sound source, and I'm down to converters.

The point is I need 16 LIVE tracks....Not 8 now and 8 later...I already do split up sessions that way, but need to be able to do 16 live.

Also regarding the AD16....If it ONLY uses ADAT, and ADAT can only go up to 48k, then what's the point in the AD16 mentioning that it goes up to 192k?

OPUS would know the answer better than I but I believe it also uses an AES/EBU protocol as well ... and with a soundcard set up that is 96k capible, you can record at these higher sample rates.

This may not be the path to better audio however ... 96k through a MOTU interface will not sound as good as 44.1 or 48k with an Apogee ... it's not just the sample rate, it's how good the conversion is in the first place. I have heard a lot of people say that 16 bit done well, is better than 24 bit done poorly.

All that being said, my Alesis AI3's seem to do the trick just fine for me ... and a lot of other people seem to agree. I get compliments all the time on the quality of my recordings. I think stable clocking has more to do with the convereters quality than anything else.

Great mics, pres and compression on the way into the box, as well as (dispite what many think) reaching for as hot of level as you can get without overs, will make a much bigger difference than anything else. It has for me.

Originally postred by mikecornett:

The point is I need 16 LIVE tracks....Not 8 now and 8 later...I already do split up sessions that way, but need to be able to do 16 live.

It's been said at least three times (this is going to be the fourth): with a Digidesign LE interface you only get 18 inputs that can be used simultaneously:

8 using the analog I/O and thus the Digi's stock converters

8 using ADAT I/O and thus adding an 8 channel converter of your choice

2 using SPdif I/O and thus adding another 2 channel AD converter THAT DOES SPDif

Digidesign interfaces like the 002 are not daisy chainable, you're stuck with one, and that's clearly written on the manual.

The quality of Digi's stock converters is obviously not at the same level as Apogee or other stand alone converters (but I wouldn't go for the Alesis which is almost equal to those if you really want to see a difference that's not only in price), but if the analog I/Os are fed with a good quality signal from good mic, pres and compressors and the 002 is clocked to a good source (via ADAT or Spdif) they definitely can be used.

So, in the end doing the math will bring you to :

10 channels of good quality ( AD conversion of your choice dependant)

8 channels of OK quality (from the Digi's converters).

You can skip to another software but still you'll be limited to this same solution as long as you use Digidesign harware.
If this isn't enough for your needs then you should consider getting a system based around different software and hardware like Nuendo or other, and go for an hardwarwe solution that allows multiple ADAT ports with S/MUX capability for 96K (like Nuendo with RME or Steinberg hardware, Linx, Motu etc...) so as to be able to use the Apogee AD16, or go for a Digidesign Pro Tools HD system.

I hope that I made myself clear, if not, the problem is somewhere else.

L.G.

The reason being that you can't do anything higher than 48Khz on the Digidesign products(Any of their products...LE..Mix or HD Systems) is that they themselves will not implement S/MUX into their hardware. I don't think they ever plan to do so.

ADAT on the Apogee products can and will go to 96Khz...anything higher than that using optical is just plain silly. Besides...does anyone here know people that are even recording higher than 96Khz?!!!

Opus

lorenzo gerace wrote:
I hope that I made myself clear, if not, the problem is somewhere else.

L.G.

L.G. Thanks, and I apollogise, I thought I had mentioned that anything outside the ProTools realm would be fine for consideration as well, but I take it I didn't post that in this thread. My bad.

As for the AD16x....I just came to the realisization that it has a connector in the back labelled 1-8, 9-16. Someone brought to my attention that this is a DSUB connection, and that they make XLR-DSUB snakes.....Does this mean I can use the AD16x as my i/o device??? In this situation I wouldn't need any other i/o box to try and figure out how to work them together.

Thank you all very kindly.

The AD16x. is an analog to digital converter only. Inputs but no outputs.

The AD-16 X has three DB 25 connectors on it...
Two are the 16 channels of analog input...
the third is the AES input.
Then there is the four optical connectors...if you are using
44 or 48Khz then you will only use the first two optical connectors..if you are using 88.2 or 96khz..then it's all four optical connectors.

Yes, the AD-16 X is only an Analog to Digital converter..it does not have any DA built in.

Opus

There's the companion DA -16 X for outputs. Together, they are a very elegant solution. I'm-uh savin' mah pennies!

What makes the Ai3 an interesting partner to the 001 and I think it would follow, to the 002 ... is that it has an input lightpipe and an output lightpipe. This and the fact that is is very well behaved when the data is bad.

The PT LE can be in internal clock with the Ai3 set to slave. The Ai3 is listening to the 001 and so will chase. When the Analog inputs are then sent to the digital converters and then sent to the 001 card all things are in continuous sync ... audio word that is.

Given the data from 001 to Ai3 is good and continous , the audio sent from Ai3 to PT LE should stay ion good condition during the record.

You say you don't like the converters of the 002. :(
At this stage I can't suggest any way or getting around these. This does include lifting the lid and heating up the soldering iron. I don't suggest you try any of the secret mods unless you are very experience with TECH stuff. If you were then you would be here asking ... no offence meant.

I expect the 002 and 001 to work in partentership with the Ai3 in very similar ways. I have been doing 16 input recordings live with the 001/Ai3 combo since the Ai3 was released.
Not perfect but a cheap solution to my problem.

I have had trouble when slaving PT to externals through the lightpipe and even more trouble when trying to get the SPDF going at the same time. I think you would need an SPDF unit with similar slave abilities as the Ai3. That is both the Ai3 and the SPDF unit would chase the MOA hardware by virtue of the outgoing audio data.

The MOA (Digidesign) harware doesn't have the word clock input and so this does close some doors and equipment choices.

I hope that made some sense.

Kev wrote:
I have had trouble when slaving PT to externals through the lightpipe and even more trouble when trying to get the SPDF going at the same time. I think you would need an SPDF unit with similar slave abilities as the Ai3. That is both the Ai3 and the SPDF unit would chase the MOA hardware by virtue of the outgoing audio data.

What kind of problems have you had trying to sync PT via SPDIF?

I'm getting a Rosetta and planning on slaving my 002 using SPDIF. Do you think I will have any problems?

First, I don't yet have a 002 to I have to guess a little. I do expect it to work much like the 001 and it is still ProTools LE we have to deal with.

Althought PT can chase either the LightPipe or the SPDF input I don't feel that PT does a particularly good job of it. Hardware devices like the Ai3 do seem to chase rather well by comparison.

Second, I don't have a Rossetta and you didn't specify which one you will use. Looking above at the units Kurt has show us the Rossetta 200 has SPDF in and out. The front panel shows an input source select choice of SPDF and also seems to have a sync LED. It also has a Sync selector.

This does seem to have all that is required to chase a 002 accurately. The R200 can be set for SPDF sync and the External Light (EXT) will probably indicate when things are OK. The sample rate will probably display the current PT session rate and automatically change if you change the rate in PT.

I do expect that the system would function, if you did chose to set PT to chase the R200. It is up to you to chose the method that works best for you.
The addition of a BNC word clock to a 002 would have made many people much happier.

It is always interesting when the a company like Apogee make an openning statement like :

Connect the ROSETTA 200 directly to your Pro Tools HD or Mix Systems or to Logic, Nuendo or other DAW via the
X-FireWire card.

Why did they deliberately leave out PT LE ???

These things always require solid testing before you commit to important recording sessions.

Does this back pannel differ from the one Kurt showed us above ??
look at the SPDF and word clock possitions ... and it has expansion cards.
Always difficult to make comments on equipment when we may not be talking about the same things. :roll:

Application: Any digital device connection that supports S/PDIF, DOLBY AC-3 and DTS on a 75 ohm coaxial connection.

That's straight from the Apogee site in reference to the SDPF on this unit. My tech mind is racing with questions on rates and bit depth and number of channels in the AC3 and DTS streams. :shock: ???
These added formats will not be useful through the 002. :cry:

I've been drooling myself over the Ad16x but please iluminate something for me: How do my 16 pres connect to the AD16x in the first place? I don't see 16 xlr ins... in my 2 ch rosetta there are 2 xlr ins from my pres, then I go out via spdif into my 828 mkii- :!:

Kev wrote: First, I don't yet have a 002 to I have to guess a little. I do expect it to work much like the 001 and it is still ProTools LE we have to deal with.

Althought PT can chase either the LightPipe or the SPDF input I don't feel that PT does a particularly good job of it. Hardware devices like the Ai3 do seem to chase rather well by comparison.

Second, I don't have a Rossetta and you didn't specify which one you will use. Looking above at the units Kurt has show us the Rossetta 200 has SPDF in and out. The front panel shows an input source select choice of SPDF and also seems to have a sync LED. It also has a Sync selector.

This does seem to have all that is required to chase a 002 accurately. The R200 can be set for SPDF sync and the External Light (EXT) will probably indicate when things are OK. The sample rate will probably display the current PT session rate and automatically change if you change the rate in PT.

I do expect that the system would function, if you did chose to set PT to chase the R200. It is up to you to chose the method that works best for you.
The addition of a BNC word clock to a 002 would have made many people much happier.

It is always interesting when the a company like Apogee make an openning statement like :

Connect the ROSETTA 200 directly to your Pro Tools HD or Mix Systems or to Logic, Nuendo or other DAW via the
X-FireWire card.

Why did they deliberately leave out PT LE ???

These things always require solid testing before you commit to important recording sessions.

Does this back pannel differ from the one Kurt showed us above ??
look at the SPDF and word clock possitions ... and it has expansion cards.
Always difficult to make comments on equipment when we may not be talking about the same things. :roll:

Application: Any digital device connection that supports S/PDIF, DOLBY AC-3 and DTS on a 75 ohm coaxial connection.

That's straight from the Apogee site in reference to the SDPF on this unit. My tech mind is racing with questions on rates and bit depth and number of channels in the AC3 and DTS streams. :shock: ???
These added formats will not be useful through the 002. :cry:

Well I have the Original 2 Channel Rosetta AD. I'm worried because I don't think the Rosetta has a provision allowing it to be slaved. So the 002 has to chase the Rosetta and I'll be using SPDIF. What exactly do you mean by PTLE doesn't chase particulary well? Can you be more Specific?

This is what I have:

http://www.apogeedigital.com /products/rosettaad.php

Johnjm22 wrote: What exactly do you mean by PTLE doesn't chase particulary well? Can you be more Specific?

quite simply I think it did a crap job on the LE version with a 001.
I'm a tech .. and I get pedantic about things.

I did a great number of long winded tests to see if the combination would support 30 min to 1 hr recordings with 16 inputs and simulated lighting rigs and bumpy power etc etc ....
A live show in a club can't be stopped just because i lost word clock sync blah blah ....

I found that the audio files recorded did have some faults. Not so noticable on playback in a mix situation BUT it does leave you wondering what else is going wrong during the conversion.

With the Ai3 chasing PT the results where better. This judgement is only from a Tech's standpoint and not about better sound.
Better sound is important BUT only after the system is verified to work or operate correctly first. It could sound fantastic but if it has a fault on occasion then it is not a reasonable option for live recording.

I can go around in circles here trying to explain many situations in broadcast audio were the better unit was rejected in favour of a unit with better fault recovery. Mpeg decoders can freeze and this is worse than a unit that with auto re-start.

sorry for the cryptic info
.... and I hope some of this makes sense.

mmmm ... :(

I don't see two SPDF RCA connections. Adat=SPDF ???? for the second connection.
I think your only option is to set PT to chase the Rossetta
2-channel, 24/96k A/D Converter.
As the name suggests - it only converts Analog to Digital.
.... and no word clock input. :?

I probably need to have this unit explained to me cos ... it all looks silly to me. :roll:

"How do my 16 pres connect to the AD16x in the first place? I don't see 16 xlr ins... "

It uses a DB-25 multipin connector for the analog inputs and outputs. You cab get snakes that go from DB-25 to TRS or XLRs

The Rosettea Kev is talking about is the Rosetta ... The newer version which I posted pic to is the Rossetta 200 and the rear panel is different ...

Kev wrote: [quote=Johnjm22] What exactly do you mean by PTLE doesn't chase particulary well? Can you be more Specific?

quite simply I think it did a crap job on the LE version with a 001.
I'm a tech .. and I get pedantic about things.

I did a great number of long winded tests to see if the combination would support 30 min to 1 hr recordings with 16 inputs and simulated lighting rigs and bumpy power etc etc ....
A live show in a club can't be stopped just because i lost word clock sync blah blah ....

I found that the audio files recorded did have some faults. Not so noticable on playback in a mix situation BUT it does leave you wondering what else is going wrong during the conversion.

With the Ai3 chasing PT the results where better. This judgement is only from a Tech's standpoint and not about better sound.
Better sound is important BUT only after the system is verified to work or operate correctly first. It could sound fantastic but if it has a fault on occasion then it is not a reasonable option for live recording.

I can go around in circles here trying to explain many situations in broadcast audio were the better unit was rejected in favour of a unit with better fault recovery. Mpeg decoders can freeze and this is worse than a unit that with auto re-start.

sorry for the cryptic info
.... and I hope some of this makes sense.

mmmm ... :(

I don't see two SPDF RCA connections. Adat=SPDF ???? for the second connection.
I think your only option is to set PT to chase the Rossetta
2-channel, 24/96k A/D Converter.
As the name suggests - it only converts Analog to Digital.
.... and no word clock input. :?

I probably need to have this unit explained to me cos ... it all looks silly to me. :roll:

I hate my Digi 002. PTLE sucks!!

Johnjm22 wrote: I hate my Digi 002. PTLE sucks!!

:(
that's a shame ...
cos I love my 001 + Ai3 + XP on an el cheapo PC squashed into a rack box thingy.

Once again, tonight, one of my systems will get rolled out to do a panic live record session. Guitarist is booked into hospital next Tuesday to have an operation on their hand and will leave her out of action for some time. Band is desperate.

The scum bag Kev system all racked up into a road case with Kev-Mic-pres will be used to log a band rehearsal.

I usually grap a copy of the HardDrive and later at my leasure I'll pick one of their second string songs , transfer it all into the TDM system and do a bit of edit and mix magic ... all pretty standard stuff ... and more times than not this song will appear in their next album. The Band never knows it at the time and I generally pick a song that they are not SO attached to ... which lets me get away with some edits and trick they might not normally let me do. Later during album time and when things look bad for an extra track I'll slip the song in during a break and just let it play.
Just love to watch the faces as they try to work out who and when ???

smug mode :D

sorry ... all part of the little fun things I do during those very boring sessions and yet another young rock-god band is pounding away thinking they are breaking new ground.