If so, how are you using it and what do you like/dislike?
If so, how are you using it and what do you like/dislike?
I'll take that as a no.
I do have the native version. Got a nfr licence of the first version when I did a review about it. Never looked back at it though.
In a nutshell: Nice Gui and overall look (as always with Waves), easy to operate. Downside: Waves Plugins only, at least as yet! So for me, I would rather go with the LiveProfessor, if you need a VST Plugin host.-->
http://ifoundasound.com
Way cheaper, way more features than the Multirack, no restictions in the plug in choice.
Or just use SAC als Pluginhost! For me, SAC is the most stable one and you don't have to careful with plugin latency;-)
Maybe Bob should do a downgraded SAC version, a Multirack contender so to say. It seems, that everybody is working with plugins in a live enviroment, even if they don't have a clue, what they are doing-->
https://www.youtube.com/watch?v=CnYR6b7D3qc
One of the AMP testers is using it. Seems to like it. Basically unless you want to jump through a lot of hoops, you'll want to run it on a separate computer. In which case you just treat that system like you would any other hardware effects units. No big deal.
If you want to run it on the same machine, I would have separate sound cards for each software. Although some audio drivers like RME will let multiple pieces of software talk to the card, you would still have to be careful about which I/O ports are used by each software and at the very least play around with RME's total mix to bring software returns back to software inputs, etc..
I would just keep it simple and install two audio cards, use one for SAC and one for Multi-rack.
I also agree with Wurst. I've been using Live Professor. It's very affordable and works great. I run it on a separate PC and that handles my effects and audio playback and links in digitally to the AMP or SAC computer.
Richard B. Ingraham
RBI Sound
http://www.rbisound.com
Email Based User List: http://tech.groups.yahoo.com/group/sac_users/
I'm the Amp tester who's using it. I just wanted to get some input from the good folks over here. I really like some of the waves stuff, like the GEQ, IR Live, and the C4.The slight added latency doesn't affect me because the system is FOH only, no monitors. I haven't used it with SAC yet, soon though.
My name on this forum is Mojogil because I screwed up when I originally signed in. I thought it was just a login name, not a username.
Richard B. Ingraham
RBI Sound
http://www.rbisound.com
Email Based User List: http://tech.groups.yahoo.com/group/sac_users/
Since I have some attention to this thread, I'll ask another opinion question:
What audio interfacing hardware would you use, in a two computer configuration, to obtain the lowest possible latency between the two? My question assumes that cost is not a major concern. Of course in the real world, cost is always a concern, but I'm curious.
I'm looking hard at the Antelope Orion 32 as the main interface, and a madi card on both machines. I like this device because the audio connections are screwed in solid. That would be 64 channels of I/O at 48k. I am planning a trip to a local music store that has one in stock and spending some time with the Orion. Hopefully within the next couple of weeks. I'll post my findings.
Another variable to think about is remote preamps. I could use this scenario with a Midas DL251 for example (using the network bridge).
Your input is appreciated
Thanks,
Gil
The added audio latency will be based on the drivers for the particular audio interface plus whatever interconnection you use between them.
Most of the good sound cards seem to be able to run at latency low enough to make it all usable. RME, MOTU, etc... I don't really know anything about the Orion piece. MOTU seems to have a bit lower settings than RME at least in their PCI offerings. Of course those are discontinued now...
As for the interconnection between the two interfaces, most of the non network based digital audio interconnections will ussually give you the lowest numbers. So ADAT, AES, SPIDF, MADI, etc.. should all have relatively about the same added latency... very low. As soon as you go to analog and back you've added a conversion whcih will add more latency. Plus you've added noise (albeit very small amounts with a good interface). Using network based connections will ussually add more latency as well. I would avoid that if latency is a concern. All the networked based audio adds latency every time you hop on and off the network. So going from SAC to an effects machine is one hop and then back to SAC is another hop. If you're also doing networked based inputs to your system, that's another hop. And if you're amps are sitting on the network that's another hop as well. How much each hop add depends on the protocol used. Some are faster (less latency) that others. Read the specs carefully.
Richard B. Ingraham
RBI Sound
http://www.rbisound.com
Email Based User List: http://tech.groups.yahoo.com/group/sac_users/
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