Solid State Logic .. SSL Big SiX

I would call that flat out distortion. Not saturation. hehe. But i think that is what seperates the big expensive vintage consoles vs the cheaper and more modern mixers. The modern and/or cheaper stuff are “clean” all the way to distorition, but the old stuff started adding more and more harmonics the more you gained.

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Hahaha. Definitely. It gets pretty crispy

I read gearslutz and other websites about this BigSix. Do I understand correctly that people are a little afraid with this because of the soundcard (probably more latency than i.e. universal audio) in combination with the fact there is no direct sound-mode for recording instruments?

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Overall latency is a bit of a concern for me, but not having a direct sound mode isn’t. Reason being, on a mixer like this, your latency is always cut in half because the mixer is the half way point. You can monitor from the computer at the same time you are recording something externally in, and the mixer is the halfway point.

Maybe I’m thinking of it the wrong way but couldn’t you just monitor out from the mixer and not listen back through your daw?
That’s what I do w/ my old mackie onyx FireWire mixer/interface

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That makes me curious. I didn’t really work like this with my xone 96 but it’s the same: I wonder when you listen to some tracks receiving your instruments directly, and d some tracks coming from the computer in other tracks, which has some latency for sure.

Right. Unless you are tracking live instruments like guitar or drums i would imagine it would be pretty manageable

I’m wondering still, will try with my Xone some day: how a sequence played directly from say and Elektron machine in some inputs, and am audio clip played by the computer on another track align, while computer and instrument are synced…

Maybe only a matter of offsetting the clock.

But I feel I misunderstood the concern about latency. What do people mean?

half because the mixer is analog, and if digital, it creates latency to be sent through the digital pathway of eq, compressor, multiband, etc? Since this part is in the mixer and not in the computer, it decreases latency?

that’s what I thought. Just play standalone, plug in your computer and daw, and record in it, but listen from the mixer directly. I have a lack of experience with this kind of equipment, and 3k is a bit too much to just hope it’s fits me

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latency: delay in sound input and sound you hear from the speakers.
delayed sound is annoying when making music

when playing guitar and feed it into a soundcard, many soundcards have these ‘direct audio’ function, where the sound you hear is not sent into the daw (with its latency), but directly sent out of the soundcard, to minimize latency to close to none.

quality of soundcard with specific specs influence latency. That’s one of the reasons people buy very expensive universal audio apollo soundcards. If you have an amazing mixer, created by a famous mixer-company with an amazing track record for mixers, that has a soundcard causing too much latency may spoil the party.

Right. Some interfaces have direct monitoring, which cuts down on latency, and this is something you can achieve with the Big Six. You are hearing what you are playing in directly, before it hits the converters, and then you can monitor other tracks from from the DAW at the same time. In a a mixer like the big six, this is easy to accomplish with the monitor controls. Also, you can do input monitoring if you’d like, which is hearing your signal hit the converters and coming back out with the rest of the DAW. This would increase latency, but some people may like that as an option.

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thanks, really clear :slight_smile:

As makers of high end analog consoles, it is the first line on their job description to do just that.

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@wouzer yes, I know about this and thanks for articulating it here. As @HoldMyBeer says, I expect the direct signal to take the Analog path in the mixer directly, I mean, it has to have this direct monitoring, I can’t imagine suchh a mixer without a direct path from input to speakers without making a round-trip of conversion.

Is that what people worry about when they talk of fear of latency? If so I believe there’s nothing to worry about.

What I was trying to hint at is the case when I’d hear something plugged in the inputs directly AND something coming from the computer via a track set to USB input, which is an expectable use case. There, the computer audio is subject to output latency, potentially additional latency caused by effects (e.g. a convolution rêver).

Thinking as I’m typing, I guess it’s no different from using a soundcard that has direct monitoring and playing with both live sound and recorded sound. One needs a solid sync option that accounts for sending clock to the live sequenced instruments while playing back what’s recorded. That’s the reason why I bought that expensive solution, E-RM multiclock…

It showed up, no tracking or shipping email or anything… but I guess I did only pay $2.5k… WTF?

Initial observations… sounds a shit ton better than my 1642. It DOES work with AUM on my iPhone. Will try and test with the MPC Live later and update y’all!

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mpc latency update when you get a chance thank you thank you thank you

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You bet! Should I just use audio tracks and plug an out to an in and record it to see what the delay is from the original, or were you thing something else?

Also, assuming it works at all I guess…

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More than anything I’m interested in your general feel as in how much you think you can work with the latency or not but whatever kind of test you feel most confident in would be great

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