Gigperformer +Motu Ultralite mk3 hybrid + some plug-in efx + MacBook Pro 2012 via FireWire as a digital mixer for live gig
Does anybody use? Or know if it will work fine?
I just need 6ch 
Thanks
Lots of people use GP for mixing – and GP will work on MacBooks all the way back to 10.9
That said, make sure the plugins you want to use for effects can be handled by your MacBook in terms of CPU needs.
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Hi, thanks for your reply, I’m testing and I found some latency… is this normal ?
In the Audio Options, you need to have your buffer as low as possible (without introducing clicks/pops).
Individual plugins can also add latency. In Wiring view, hover your mouse over the plugin block, and it will show what its latency is via the tooltip.
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Hi, I tested the audio buffer settings yet but I still hear some delay…
Even if I only create 1 ch with 1 rev efx
I will try with another audio interface I have ( Steinberg)
Thank you for your reply
Care to share the settings?
what is your sample rate and buffer?
IMHO… 44.1 KHz is sufficent for live VST and this will allow you to lower the buffer further (vs. 48 KHz) to reduce added latency.
48 KHz is more important when you have low end frequencies that potentilaly will distort but the difference to your audienes ears would be most likely impreceivable even if you’re mic’ing floor toms.
Also, don’t forget to high-pass filter everything before the mix (except VST instruments).
bk
And 48kHz is mandatory when you use AES
Actually, if you raise the sample rate, that will lower the latency.
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48 KHz reduces latency compared to 44.1 KHz but in my experience it increases the likelyhood of cracks/pops at an equal buffer rate.
So if dropping to 44.1 KHz allows for a lower buffer increment, that must be overall less latency than 44.8 KHz at a higher buffer, no?
ie: 44.1 KHz @ 64 samples is less latent than 48.8 KHz @ 128 samples
?
I do not understand that. I thought that the Nyquist theory states that to reconstruct a signal, the sample rate should be twice the highest frequency, so 44.1KHz or 48KHz should both be more than enough for handling low frequencies like those from floor toms (although there will be some overtones, i.e. the click from the tip hitting the tom). When it is about the amplitude (which tends to be higher for low frequencies, because there needs to be moved more air), then the bit depth is more a thing, except that 24 bit is quite a dynamic range. Or am I misunderstanding you? 
The difference as I have undertood it in practical terms is in the anti-aliasing and the slope of how it deals with frequencies that exist out side of the reproducable range of 44.1 KHz so recording at 48 KHz is of benefit. It might be that benefit only exists in the upper frequencies but I assumed that anything sub 20 Hz would be affected in a simlar way, perhaps not.
Again, negligable and unlikley you’d hear any differences, espechally live. AES/EBU and SPDIF both require 48KHz I do believe, probably most MADI or DANTE is clocked the same way. Digital audio for video is always 48 KHz too… but if, per my other post, dropping sample rate alows you to lower the sample buffer and results in lower latency, that would be a reason to use 44.1 KHz live.
Sub 20Hz is almost always filtered away, because there is almost no home equipment that is able to reproduce it anyway. There are church organs that can do 5 Hz, but that’s more like a small earthquake. Maybe Dolby Atmos in a movie theatre?
Apart from that: The sample factor between 100 Hz and 44.1 KHz is 441. If that is not enough for rather faithful reproduction of the signal, then the reproduction of 10 KHz signals (which have only 4.41 samples) would result in a heavily mutilated output. So I guess you will not have to bother about the sample rate for low frequencies.
One of the reasons for higher samples rates is because of the low pass filter needed for A->D conversion. To guard the converter from frequencies below 22050 Hz (Nyquist), the input low pass filter will have to be very steep, which introduces all other sorts of problems. With i.e. 96 KHz the input filter can be more relaxed: 96 KHz is more than 2 octaves higher than 20 KHz. That results in a much simpler filter with less side-effects. Same kind of discussion for D-A conversion to filter out the frequency of the sample rate.
Feel free to correct me, but be gentle: I had a rough day 
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For the same buffer size
Depends on your computer
Latency is defined as
sample buffer size/sample rate
In other words, increasing buffer increases latency. Increasing sample rate decreases latency
and per that equation, instead of changing buffer size, increasing sample rate decreases latency