rsinger

Voda La Void
 
In my case, the requirements boil down to permanently setting up recording chains with the least hassle for the best quality, for a broke a** musician working solo.   Live drums, electric guitar, acoustic guitar, bass, vocals and eventually cello.  I want it set up, ready to record when I feel like it, any instrument, at any time.  
 

Do you need multiple inputs for that? I play guitar synth so I have both a mixer and sub-mixer. I have various synths and guitar synths going into my mixer. Since I'm only recording myself I don't need multiple inputs - I just have 4 inputs in my audio IF and the mixer goes into that (and a line in from a guitar amp). Everything is setup - I just move faders on the mixer up and down depending on what synth or processor I want to record. 

Oh, absolutely.  10 mics for drums.  2 mics for acoustic guitar.  1 mic for vocals.  1 input each for bass and electric guitar from DI boxes that use mic level inputs (like the Palmer).  Each of them will have their own gain levels, of course, set.  I do this right now with my 1620 but it's messy and uses a lot of outboard mic pre's to feed line level inputs since I only have 8 mic pre's on it.  I just un-mute the channels associated with whatever instrument I'm recording.
 
What I'm trying to do now is clean all this up and take a step into the modern world because my DAW and mixer are old and wearing out.  
 
Another thing that makes permanently setup recording chains a little more necessary is that I write more in sections now, as opposed to whole songs at once.  Used to...I would write the whole song (guitar and vocals) and *then* go to record, basically shoehorning bass and percussion at that point - and record all of an instrument's parts at one go.  I did about 6 album's worth of crap like that.  
 
Nowadays...I'm way more into the instrumentation, not just vocals over backing music, and I want more out of each instrument, so I write in sections, doing each instrument to complete a section, if you will.  So I need the sounds of these instruments to be very consistent so it doesn't change throughout the song. 
 
Plus, it's awesome to just sit down, push a couple buttons, and you're set. Man...it's almost like being *just* a musician...for a time.  

Studio Vheissu
I recently started to use a Behringer X-32 and it works fine as an audio card so long as you don't need more than 44.1 kHz. The mic pres are Midas so no problems there.

 
My X32 worked fine at 44.1k and 48k.  Because of video work, I always used it at 48k.  What have you encountered that made 44.1k your preference?   Also, there was a firestorm of discussion and vitriol about the difference between the 'real' Midas preamps of the M-series, and the 'Midas-designed' preamps of the X-series.  That's why I did my series of articles on the issues and measured/documented the differences.  

Voda La Void
 
In my case, ....
 
The least hassle means *not* playing with ADAT expansion interfaces.  I really wish that could work, but I can't seem to find any information on how latency is handled with two devices with different processing times.  I've scoured the web for articles, and they get really close, but they never really touch on the point I'm after. ... I can see me inventing new cuss words working with a wireless tablet control solution...

Yes, you'd probably be happier with physical controls or with using the PC-based control functions (also supported).
 
I'm not sure what information you need about "how latency is handled with two devices with different processing times".  If you mean you're not sure about how ADAT delay affects the overall latency picture of the interface into which it is connected, then the answer is that it adds 15-to-50 samples of delay on the way in, depending on the product, which means between 1/3 of a millisecond to a full millisecond.  The remaining ASIO latency of the audio interface works as normal, and the ADAT device is not independently visible; it's just an expansion of the number of available inputs (and sometimes outputs) as seen by the DAW.
 
The 1/3-millisecond to 1-millisecond conversion delay translates into the same effect as moving a microphone 4-to-11 inches from the source. For your 10-channel drum setup, if 8 channels of the drum set come in via ADAT and 2 channels of the drum set come in via the native microphone ports of the interface, then it's as if the ADAT-connected microphones are slightly further away. If you choose (for example) floor-toms or the room pair as the 2 "oddball" inputs, then there's nothing that should create a concern regarding the coherence of the drum recording. 
 
Does that answer the point you were after?

GaryMedia

Voda La Void
 
In my case, ....
 
The least hassle means *not* playing with ADAT expansion interfaces.  I really wish that could work, but I can't seem to find any information on how latency is handled with two devices with different processing times.  I've scoured the web for articles, and they get really close, but they never really touch on the point I'm after. ... I can see me inventing new cuss words working with a wireless tablet control solution...

Yes, you'd probably be happier with physical controls or with using the PC-based control functions (also supported).
 
I'm not sure what information you need about "how latency is handled with two devices with different processing times".  If you mean you're not sure about how ADAT delay affects the overall latency picture of the interface into which it is connected, then the answer is that it adds 15-to-50 samples of delay on the way in, depending on the product, which means between 1/3 of a millisecond to a full millisecond.  The remaining ASIO latency of the audio interface works as normal, and the ADAT device is not independently visible; it's just an expansion of the number of available inputs (and sometimes outputs) as seen by the DAW.
 
The 1/3-millisecond to 1-millisecond conversion delay translates into the same effect as moving a microphone 4-to-11 inches from the source. For your 10-channel drum setup, if 8 channels of the drum set come in via ADAT and 2 channels of the drum set come in via the native microphone ports of the interface, then it's as if the ADAT-connected microphones are slightly further away. If you choose (for example) floor-toms or the room pair as the 2 "oddball" inputs, then there's nothing that should create a concern regarding the coherence of the drum recording. 
 
Does that answer the point you were after?

Yes, that's precisely the point I was after, thanks!  
 
It sounds like it's important to be sure the latency of the expansion is the same as the native interface - so if they are the same make and model, paired up via ADAT then it doesn't sound like there would be any issues, really. That's really good to know. But if the expansion is some other brand and model, therefore different circuitry and processing times, than the native interface, then I could run into issues, I'm assuming.  
 
I guess I was hoping the DAW might calculate latency by input and compensate, rather than a global calculation for the interface as a whole.  I guess not.  
 
I know that's all kind of down in the weeds, but I get hung up on things like this...thanks for the information.  I feel better about going ADAT if I have to, now.  

Voda La Void
 
Yes, that's precisely the point I was after, thanks!  
 
It sounds like it's important to be sure the latency of the expansion is the same as the native interface -....
 

Nope! 
 
The ADAT expansion only has to conform to normal ADAT protocol. It's not model or brand-specific at all!  You can freely mix and match ADAT devices with any hardware.  For example, I have attached a Mackie BlackBird to an Apogee Quartet, an Audient ASP800 to a Behringer X-ADAT, an ART TubeOpto to an Echo Audio Pre8, the Audient ASP800 to the Echo Audio Pre8, and the ART TubeOpto to the Behringer X-ADAT.  Everything works fine. 
 
While the ASP800, the BlackBird and the TubeOpto have different conversion latency values, these values are small enough to be of no consequence, especially considering the relatively higher latency of the audio interface to which they are connected.  For example, the Echo Pre8 was usually run in live recording situations with a 512 sample buffer, yielding an input latency in excess of 10-milliseconds.  The additional 1/2 millisecond of the TubeOpto was of no concern.  

Nowadays...I'm way more into the instrumentation, not just vocals over backing music, and I want more out of each instrument, so I write in sections, doing each instrument to complete a section, if you will.  So I need the sounds of these instruments to be very consistent so it doesn't change throughout the song. 
 
That is an important and very solid reason to go for a system that has enough inputs. 
I don't record live drums in my studio, For that I use my laptop and record elsewhere. 
But I'll use my Yamaha 01v which I have saved scenes for each band I've ever recorded or played in. 
 
My studio set up I have only 6 channels but that's all I need and I don't have to swap out cables. Vocal mike, Guitar amp mike, Bass and acoustic DI's.
I also have the Yamaha 01v and use the SPDIF and that has weird things like my Korg 05/rw,
Roland 505 , DTX drum analog out and I can use it for miking my acoustic guitar or a DI from my Blackstar amp and odd's and ends. So those 16 into 2 SPDIF channels come in handy. 
 I use a little mixer to monitor everything. 
 
My 01v is real old and living on borrowed time therefore my quest for a replacement. I would also use the new mixer live so I need effects. 
 
So I'm a big fan of leaving everything set up and ready to go.
I then have track templates for each of these inputs.
I can set up a project in less than a minute and be recording an idea. 

GaryMedia

Voda La Void
 
Yes, that's precisely the point I was after, thanks!  
 
It sounds like it's important to be sure the latency of the expansion is the same as the native interface -....
 

Nope! 
 
The ADAT expansion only has to conform to normal ADAT protocol. It's not model or brand-specific at all!  You can freely mix and match ADAT devices with any hardware.  For example, I have attached a Mackie BlackBird to an Apogee Quartet, an Audient ASP800 to a Behringer X-ADAT, an ART TubeOpto to an Echo Audio Pre8, the Audient ASP800 to the Echo Audio Pre8, and the ART TubeOpto to the Behringer X-ADAT.  Everything works fine. 
 
While the ASP800, the BlackBird and the TubeOpto have different conversion latency values, these values are small enough to be of no consequence, especially considering the relatively higher latency of the audio interface to which they are connected.  For example, the Echo Pre8 was usually run in live recording situations with a 512 sample buffer, yielding an input latency in excess of 10-milliseconds.  The additional 1/2 millisecond of the TubeOpto was of no concern.  

So, are all of these boxes that close in latency?  
 
If expansion device B takes 16ms to process input audio and send it to the ADAT link to native device A, with a 10ms interface latency such as yours above, then wouldn't that produce a 6ms delay from any input in device B? 
 
Maybe that's not a real world scenario.  And that's great, if so.  I just assumed all these boxes were wildly different.  
 
 
 
 

The real-world numbers are all under 1-millisecond latency.
 
As I mentioned in previous posts about the two boxes for which I had firm numbers, the 'slowest' box I have is the Audient ASP800 at 39 samples.  At 48-khz, that's 0.8125-milliseconds or 13/16ths of a millisecond.  The ART TubeOpto 8 is 32-samples delay in A/D conversion which is 0.666-milliseconds or 2/3rds of a millisecond.
 
Re-read post my post #14 in this thread for additional details.  

Cactus Music
I was shopping for the same thing a year ago and chose the Soundcraft Boards because of both features  and reports from Sonar users that the drivers worked!! That's important. 
I was looking at the 12 channel but they make all sizes
And many are only 2 channel USB. So pay attention
https://www.sweetwater.com/store/detail/Sig22MT 
 

I forgot to ask you about zero latency monitoring on this board.  I see it has a headphones jack, but does the driver software let you mute the recording playback so you can monitor at the board with zero latency?  
 
So far, I haven't really been too worried about latency since I figured that would always be resolved by monitoring at the interface.  And, I generally never record tracks with plug-ins engaged and all that.  Just raw audio mostly.  

The term Zero latency is a term applied to all audio interfaces that allow you to monitor the input directly before it passes through the A/D converters. 
It is misused as people see it and think wow that audio interface must be good because it has zero latency! 
Of course it's zero, it's the same analog output that's been there since they first built an audio mixer. 
 
This is then mixed with the playback from your DAW. This is where a good ASIO driver is needed to adjust the timing offset to match perfectly the latency in the system. System latency or  Round trip latency,  is the figure that varies from interface to interface. 
I didn't purchase the Soundcraft board yet, I just picked it as the best of the lot for my needs to replace my old 01V. 
It's not a high priority as the 01v still works but the day will come I would think, that it will die an ugly digital death. 
 
So even without using it I can safely say you'll have proper monitoring and the drivers are reported to be working with Sonar and many other DAW's without issue. I would guess it will perform at least as good as the 2 interfaces I use right now. If one wants better performance then we are talking much more $$ per channel.