SAC (Software Audio Console)

[BrentEvans]

Yep, Lightbridge. I couldn't remember off the top of my head. When it works, it's not too bad. Latency could be better, but I'm not sure my expectations are realistic. I would sweat bullets before a serious gig if I had to depend on it though.

128 buffer size is respectable on that, 1x128 in SAC is reasonable. Latency is a strange thing though... sometimes you have way more acoustic latency than digital latency, and it makes it a non-issue. Sometimes, the latency is actually too low for something like in-ear monitors, where you either have to be in perfect phase and time alignment or else delayed by 6-10ms (same as hearing wedge monitors).

It's a shame to hear the Presonus hasn't fared any better, though I heard the lack of Lion drivers are a problem anyway. I do have a 424 PCI (need to upgrade that to a PCIe), and I thought about getting an 828 for the ADAT ins, except I'd like to do S-MUX on 16 channels, and there aren't enough ports for that. I use the analog outs on the Digimaxes for monitors while recording or when I'm not running a computer at all. I'd hate to lose that versatility. My setup tends to be somewhat dynamic, so I'm always looking for flexibility.

Can't comment on the Mac drivers... I don't feed the reality distortion field that is Apple.

There's really no need to upgrade to PCIe unless you have to have it to fit a slot. I use the PCI cards, they work just fine. I even have an old PCI 324 card in one SAC installation (those are finicky though). You would use a 2408 with the 424, not an 828. There's also really no need for anything more than 48k 24 bit with a SAC system. I run 44.1/24 bit, and it's the cleanest best sounding console I've ever heard or worked on. I frequently use LS9s, M7s, and I've had enough opportunity with iLive and the new Soundcraft digital to say that they aren't any better than SAC in sound quality. Those two are probably equivalent.

Another gotcha... my primary workstation is a 1U rackmount i5 with a single PCIe slot for the 424 (well, there's also a mini-PCIe with a Firewire adapter in it) so using another PCI-based interface is somewhat out of the question. I was hoping NAMM would provide a covenient solution. Maybe next year.

The real solution for a SAC rig is to spend a few hundred dollars and build a dedicated SAC PC. You don't want it on a shared machine, as it there are some rather specific recommendations to make it run smoothly. Anything at all running in the background can stomp on the realtime memory access and glitch the live audio.

Besides... 1U pcs typically run hot and loud. Who wants that in a live environment.

[SirNickity]

Great stuff.. thanks for the comments. You're right, 2408.. not 828. The 1U box is a little noisy. It's entirely a custom build. I picked each part specifically, and the low-profile blower (CPU fan) is the only part I was forced to compromise on. It's thermally controlled, and certainly quiet for being intended for servers, but once it's under load the RPMs go up and thus the noise. When I use it for DJing or live sound processing, the ambient noise level is high enough that it doesn't matter too much.

I understand the Apple sentiment. It's difficult trying to use OS X on non-Apple hardware, so I definitely wish they were less draconian. But, the architectural differences of the underlying OS makes Mac OS far superior to Windows for performance applications, and that makes it worth it to me to try. I have no doubt SAC is fast, stable and capable. It can be done, but as someone that works in IT and has done some software development, it is my opinion that Win32/64 applications are at an automatic disadvantage. If there were a purpose-built real-time audio Linux distro, I would be there in a heartbeat.

[BrentEvans]

Great stuff.. thanks for the comments. You're right, 2408.. not 828. The 1U box is a little noisy. It's entirely a custom build. I picked each part specifically, and the low-profile blower (CPU fan) is the only part I was forced to compromise on. It's thermally controlled, and certainly quiet for being intended for servers, but once it's under load the RPMs go up and thus the noise. When I use it for DJing or live sound processing, the ambient noise level is high enough that it doesn't matter too much.

If it's off in a room, not much of a problem, but something to be aware of. In any case, SAC systems really work best as dedicated systems, so this should be a non issue. SAC doesn't need much... a basic Core2Duo or i3 system with no bells and whistles will do just fine. In fact, the fewer bells and whistles, the better.

I understand the Apple sentiment. It's difficult trying to use OS X on non-Apple hardware, so I definitely wish they were less draconian. But, the architectural differences of the underlying OS makes Mac OS far superior to Windows for performance applications, and that makes it worth it to me to try. I have no doubt SAC is fast, stable and capable. It can be done, but as someone that works in IT and has done some software development, it is my opinion that Win32/64 applications are at an automatic disadvantage. If there were a purpose-built real-time audio Linux distro, I would be there in a heartbeat.

True, Mac may be "better," but what is "better?" If you measure it by software availability, it loses. If you measure it by stability, it wins. What good is stability if you can't do what you need to do on it?

IIRC there is such a linux distro, but drivers are a problem. Until enough users demand good Linux drivers, manufacturers have little incentive to code them. Until they exist, there's not much you can do on Linux. It's Catch 22.

[BoostFab]

It sucks that some of the large proprietary companies uses Linux as the platform for the engine of their digital mixers, but they don't contribute back to the community. They all want to protect their intellectual property.

I'm a big fan of Mac OSX for everything design and graphics, Adobe stuff runs much better on OSX IMO. But like it's been said on here before, use the right tool for the job. SAC runs on Windows, so it's best to build a dedicated Windows machine for SAC, don't shoestring your rig for production/mission critical tasks.

As far as MOTU goes, I like having one PCI card and breakout to the 2408 units for easy expandability. I'm still using the 324 card and XP. It's very stable, of course XP is so stripped and light, it's almost like the embedded version.

[SirNickity]

I don't really know what the embedded Linux guys could contribute. The hardware is likely to be proprietary, so drivers wouldn't benefit anyone else much. At that point, they're pretty much using Linux because you can literally run the kernel, and your app, and nothing else. The app is their product -- so that's not going open. The kernel is already actively developed -- nothing to contribute. What we really need is MOTU and M-Audio and friends to break the chicken-egg cycle and throw some drivers out there. Echo Audio did that for a bit, but then it got to where updates would be released "soon" (and by soon I mean eventually scrapped). The drivers that were made available are still in the mainline kernel today though.

For the record, I wasn't suggesting running SAC on Mac OS. I checked, not available. That's what dual-booting is for. Traktor Pro, Studio One, and my plugins however.. those work fine. In my case, another dedicated PC for one app would just be a waste -- and if it were the only thing I use Windows for, it's practically dedicated anyway. I don't do sound professionally, it's just a very expensive hobby that I jump into when I have the time and interest.

[Rick Lee]

In the spirit of this discussion... someone gave me an EMU 1212m PCIe card a while back but it physically wouldn't fit on the MB. I'm upgrading my computer in the next few weeks- is EMU worth using with an ADAT preamp for a small SAC system?

[BrentEvans]

In the spirit of this discussion... someone gave me an EMU 1212m PCIe card a while back but it physically wouldn't fit on the MB. I'm upgrading my computer in the next few weeks- is EMU worth using with an ADAT preamp for a small SAC system?

I don't know if anyone is using that card or not, but it would probably work at some level. The only way to know is to try it. The demo is downloadable from www.softwareaudioconsole.com and should interface with the ASIO drivers enough to tell you if its going to work or not.

[Rick Lee]

I'll put it on my to do list when I upgrade my computer- I've been using the Tascam 1641 USB 2.0 for a couple of years and it works great- my monitoring latency is ~3.6ms but the EMU card has just been gathering dust because a heatsink is in the way on the MB.

Looking forward to when USB 3.0 devices come out- at the speed it runs there shouldn't be any latency issues at all.

[BrentEvans]

Looking forward to when USB 3.0 devices come out- at the speed it runs there shouldn't be any latency issues at all.

That's only part of the story. Even USB 2.0 is capable of very low latency transmission, but the drivers are usually coded poorly. Most hardware has an in-the-box solution for monitoring, so the OEMS don't put as much emphasis on low-latency drivers.

Another issue that affects latency is processor speed. The processor has to be able to complete all the calculations in a certain amount of time, or you slip buffers. Increasing the amount of time to the processor increases the size of the data loop, increasing latency.

Truthfully, latency is overthought. There is more acoustic latency from a wedge monitor to the ears of the performer than an entire digital system typically adds, and certainly more acoustic latency in FOH. DAWs can compensate for it during recording. The only time it becomes a problem is with in-ear monitors. Some performers are sensitive to certain amounts of latency in their monitors, but adding a bit of delay (thus increasing latency) often solves the comb filtering that is percieved (similar to pulling a speaker back from 2 ft to 6-8ft, same as a wedge). Other tricks can be applied, such as adding reverb or leaving an ambient mic open, to fool the brain into accepting the comb filtering as normal.

[Doug Hart]

good to know Brent... even a 3ms latency drives me crazy with in-ears. Never made sense to me because I knew acoustic latency was more than that. Never considered that it was a phase problem, not a delay problem

[Doug Hart]

I'll put it on my to do list when I upgrade my computer- I've been using the Tascam 1641 USB 2.0 for a couple of years and it works great

You're using this with SAC? That would be good news for me as I have been considering this unit for a small SAC system.

[Rick Lee]

You're using this with SAC? That would be good news for me as I have been considering this unit for a small SAC system.

No, but I've been wondering the same thing for a while. I've had it for over 3 years and it has never messed up or went BSOD on me. I've had to manually compensate for delay in recording (loop a measure of click track and then manually line up the samples) which is 1307 samples in Reaper. I'm running a 5 year old dual core AMD at 2 ghz so surely with something newer it would do better. I can record 9 tracks at once without any dropouts at 3.9 ms - I've never used all the inputs at once so 9 is all I enable.

Using a soft synth it'll glitch sometimes but I think that's because I only have a gig of ram.

[SirNickity]

I think it could also be memory bandwidth. I had some glitchy problems go away when upgrading from a Core 2 to an i5 setup. I never even came close to 100% CPU usage, but I think all the various memcpy calls in the digital signal chain were taking too long. The i5 system has both a faster CPU clock and less latency from the CPU to RAM. With realtime DSP, that matters. I think. Kinda hard to prove without ridiculously expensive test equipment, but I have a strong suspicion.