LMAO! Guffaw LOL. That was precious! And here... I thought by your posting... that there had been an actual security breach at Presonus LOL? Yeah doggy!

I guess there's something new in the works? And only a couple more days until we find out! Cool!

Do you know what it is Chris? I sure don't? I haven't been keeping up of course LOL. And you know why LMAO. I'm an old FART. Steadfast in my old school ways. Because it works for me. The same way as it does for Bob Clearmountain, Chris Lord Algae and others, who are older, as old and younger than I. Or is that younger than me? Well I flunked English so... I'm stupid, an underachiever. And proud of it.

Bart Simpson is my favorite actor.
Mx. Remy Ann David

No, but it looked like it was a rackmount inside that blanket. Man, that company has good marketing idea's and great sense of humor.

Agreed.

The only other thing I'd add to that, would be that they also have a quality product line.

Between Presonus and Focusrite - the quality you get with either of these two companies in terms of build, sound, compatibility, and even expandability, are fantastic.

Okay... so yeah... it's obvious that they aren't Neve, Millennia, Great River, or any of the other high-dollar pristine pro studio-caliber preamps...( although some of the upper-level stuff from both of them is really nice) but then again, for the prices and models in the same basic caliber level to each other, it's hard for me to understand why anyone on the beginner or novice level (or even a few pro's like me with a limited budget) would choose any other manufacturers beyond Focusrite or Presonus, when it comes to Pre's/Audio I-O's/Converters.

The best equipment purchase I've made in the last 6 years was the Presonus VSL1818.

The difference between it and my old pre/I-0 (Tascam 1641) was like night and day. There was nothing subtle about it.
There was no "well, I guess it's a little bit better"... no way. Much to the contrary, the difference I heard was huge.

Coupled with Samplitude Pro X, both lifted my game, big time.

Now I'm wondering what this new thing is? It's a great promo video LOL. Wish they'd give us a hint. :)

IMHO of course.

d/

This must be it:

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http://www.presonus.com/products/StudioLive-RM-Series

Seems cool. But those XMAX preamps are what I have that a certain engineer around here *cough* didn't care for. ;)

Hehe, converters and mics make a big difference too! It's all subjective to the chain and ability to keep your chain running within the sweet spots. If you go beyond that happy spot, pres and converters tend to reveille their weaknesses.

I might be old fashioned, but would anyone else be nervous to rely on a computer/tablet to run FoH in a live situation? These look intriguing, but I'd have to test the crap out of one before giving the toss to a stand-alone mixer. Even assuming the device works flawlessly, I haven't known a laptop that does.

I can see having an identical backup laptop with all the software/settings synced. You know the big acts would do that anyway.

Reverend Lucas, post: 419180, member: 48050 wrote: I might be old fashioned, but would anyone else be nervous to rely on a computer/tablet to run FoH in a live situation? These look intriguing, but I'd have to test the crap out of one before giving the toss to a stand-alone mixer. Even assuming the device works flawlessly, I haven't known a laptop that does.

I was very nervous about it too the first time I tried using the SL24.4.2 remotely with an iPad, but figured I would always have the hardware console to fall back on. After 12 - 18 months, with more and more jobs relying solely on the iPad - I have to say it's been very reliable and made some jobs possible that would have been nearly impossible otherwise. I learned the very first time out, that no matter how good the in-house WiFi appears to be at soundcheck - set up your own router and create your own private network. That's the only glitch I've ever had. When the place filled up with people with their own bandwidth hogging devices, I got knocked off the seemingly stable public network a couple times, and momentarily lost contact with the StudioLive. It stays at the last settings until you reconnect, or go touch the mixer. I got disconnected twice and ended up connecting my own router anyway half way through the first act.

I rented out my bigger system last month for two nationally touring artists passing through a local high school auditorium along with a local opener. The headliner was traveling with a tour manager / sound man, the two openers (one local, one national) were both my responsibility. I'd seen the tech-rider months ahead of time and I talked to the headliner's sound guy earlier in the week prior to the show asking his preference of an analog 32-ch Soundcraft board, or the digital 24-ch StudioLive - wanting to make him happy. He said he'd be happy with either, but would prefer digital. I asked if he had used a StudioLive. He said he hadn't, but he was somewhat familiar with them, and willing to give it a try. The day of the show, I had just finished sound-checking the two openers when the headliner and his sound man arrived. I'd brought several racks of hardware EQs, compressors, reverbs, etc. in case he was more comfortable with knobs. I connected up his rack of IEMs, while I talked him through what was routed where, and then showed him the iPad running SL Remote --- and he was GONE --- that was the last I saw of him, (or my iPad) until the end of the night. He didn't leave a single fingerprint on my console. He did his soundcheck (very quickly and efficiently adjusting FOH, IEMs, and wedges) from various places around the room. And at showtime he mixed his part of the show from a couple different spots in the back of the auditorium - he even ran out to the lobby for a minute to deal with some merch issue, my iPad in hand. Not bad for his first time getting his hands-on one.

Had I known he was going to take to it so quickly, I could have saved at least 20 minutes of set-up time required to set up shop at the back of the room. I wouldn't have needed to commandeer a table, build a flight-case fortress, pull the snake back the aisle, pull power, patch in several racks that never got used, tape down cables, etc.

If I were touring, I'd buy the new rack mount RM-Series and use it as the band's monitor mixer for IEM. With control of my personal IEM mix from iPad, iPod, or iPhone. A LOT more control and A LOT less expense than a comparable Aviom system, or similar.

Wireless mixing is especially great for outdoor shows. You can leave the SL24.4.2 on stage, pull one 30ft. drop-snake to the far side of the stage, and connect everything else directly to the mixer. No more table, canopy, tarps, snake and power running through the grass for people to step on, trip over, drive over. If it rains, tuck the iPad under your shirt and find a dry spot - the rest of the FOH gear is all on-stage under a roof behind the band. And you don't have to anchor yourself to one spot. Want to see what it sounds like further back on the opposite side? Grab your lawn-chair and go.

It's a brave new world.

Nice post Dave, thanks! I just ordered a 24AI , 2 x SL328AIs and 2 x 18 subs. What a cool system this is, Wow!

This is such an excellent tutorial, check it out!

http://dotsub.com/view/a3822672-bf5e-4b5c-be4e-a1aa712260fa

JohnTodd, post: 419176, member: 39208 wrote: Seems cool. But those XMAX preamps are what I have that a certain engineer around here *cough* didn't care for. ;)

Who doesn't like the XMax pre's? I mean, yeah, okay, maybe they aren't Neve's, but they also aren't a grand per channel, - and, they aren't meant to be colored like those other types of preamps are, either...
They are meant to be neutral, transparent, and give plenty of headroom, and they do what they say, so what's the problem?

I've yet to talk with anyone who owns - or has owned - a StudioLive AI that didn't like the preamps... Even Chris said he thought they were great when he had his original Studio16 AI some months ago.

Oh wait.. I think I just figured out who you are referring to...it must be her holiness, who will go to her grave proclaiming that an SM58 and a Neve preamp is the absolute requirement to make good recordings...

Sorry about that, I'm a little slow... I'm still in the process of coming back to consciousness since I returned from my vacation. ;)

No, it's not her.

it me, :cool:
but, here's the deal... the reference to pre's was mentioned by me in a mix discussion relating to John's quest for "good, better, best" ways to improve his sound. It relates to his modern workflow, not Remy's idea of old school recording.

John's talent and quest for better sound is gear related and the "less is more" concept. Any one of the three (mic, pre or converter) will improve his sound immensely. The question on a limited budget is which one first?

Since he is tracking 2 channels at a time, tight conversion is paramount. This is what a Lavry AD11 or Prism Lyra offer. This would be the perfect fit, but its not cheap either.
The pre in either of those, combined with a clock is hands down better than what he is using.
Contrary to "Remy, who is naive to John's workflow, chimes in over and over with really dated advice. When you are tracking and using VSTi, your conversion and clocking is paramount. You are also competing with great sound libraries so to sound glued, you need to capture your own tracks just as good or better than the VSTis, not worse.

Remy, = NO VSTi, has no idea about composing or producing music like a "modern musician" , is clueless when it comes to the pros and cons, what we need to improve sound 2 tracks at a time. Johns mixes, like most are swirly, lack a tighter focus. What I hear is 100% related to plug-ins and clocking but a better mic and pre would also be a good step to improve his sound.

He likes acoustic guitars and has great vocals so, I would be saving my pennies and get something that gets me a sound that is as good as the sample libraries.

Back to the reference on that pre's.
This pre is most likely as good or better than most in its price range but if you are comparing that to top end, which is what I am always about when I am mixing or producing someone I feel is deserving ( or asking), its not what I would call stellar.
To my ears then, the best way around that is don't push the signal beyond the sweet spots. This is where pro's are able to make even low end gear sound pretty damn good. We hear it starting before its accumulating too late in a mix.

Check Out Dan's mixes on his StudioLive 24.4.2 . Amazing sound!

PS

Check Out Dan's mixes on his StudioLive 24.4.2. Amazing sound! But, if my memory serves me right, his band is tracking it all? They are not doing the overdub 2 track at a time method, which is always problematic to accumulative aliasing phase distortions.

Chris, can you explain to me how you get the cumulative phase issues from overdubbing isolated signals? I'm not sure what you mean.

Real simple!

If you are using a poor clock, punching in and out, that alone is problematic

If you are tracking to a track all the way through, using those to cue off of, which bleed in a mic, a bad clock won't line up and will create phase via bleed. This a big issue to me.

Same with reverbs that are part of original takes.

Plugins all create accumulative aliasing distortions

There are so many issues that accumulate From bad clocking and overdubs it's where I make my money

I'm not a tech guy so I'm not going to explain the math behind this but, if you look up the top mastering engineers, you will discover what I have and I garden tee you will immediately step up your game.

It's also why I don't track back to the same daw. It's all part of accumulative aliasing

It's the smear sound

Does this help?

Posted using my iPhone

Posting on my phone erk

The basic rule I follow, try and not effect any other track or part of a song with something that has already been quantized once

When in analog, I mix in analog, when in did oral, I stay there

When I over dub, I make sure cross fade don't overlap anything that will flip the transients that I hear on stereo bus's

The first time I heard the term accumulative aliasing distortions was from Bob Olson mastering

Before that, I heard this accumulating but didn't know what it was called.

IMHO, it's the ugly ingredient of digital audio. Most people don't even hear it happening because it accumulates through processing. At the end if a mix, it's a smearing sound that lacks space, clarity, tight bottom end and width that sums well.

My phone it correcting words, I'm at work, sorry

Maybe Bos will explain this beautifully

To add, I'm pretty confident to say, Samplitude is the king at avoiding this

Daws are different. Sonar as an example has accumulative smearing to my ears

sequoia in comparison, sounds so tight. Everything stays right no matter how many tracks I add

But, if I add third party plugs, some will instantly shift the transients and I hear an aliasing phase occurring.

There are plugins that people rave about , that I hear effect the transients just activated on a track

I never heard this in pro tools though. I mean, pro tools sounded smeary all the time compared. I believe this to be true because the code of third party plugins and their poor clocking is problematic, all accumulative from close that isn't remaining true as say, Samplitude or sequoia

Third party plugins and poor clocking is what we are all going to be focusing on next.

Less is more

It's important to stress that clock stability has no effect on DAW performance or on plug-ins; it only comes into play at the boundaries: A-D and D-A conversion. For a plug-in, the sort of things that show up as inferior performance are poor coding, incorrect rounding, non-constant path times and lack of attention to multi-threading opportunities, and this is on top of any suspect quality in the algorithms employed.

When running a hybrid mix, you have to be quite careful about signal routes that do not take the direct path through the mix. These include routes like effect sends, effect returns and digitally-implemented compression. Unless you have all-analog effects (such as plate reverbs) and analog compressors, it's wise to take S/PDIF or ADAT outputs from your source DAW into the effect units and then perform the D-A conversion after the effect. This removes the multiple D-A, A-D, D-A paths, but does mean that you have to think your effects through as parallel effects and not serial effects. Parallel compression, on the other hand, is a whole different art and is not so widely applicable. It's one reason that I prefer old-style analog compressors when hybrid mixing, as they can be inserted in the analog paths without conversions.

The time-alignment of these multiple paths into the mix is crucially important, so you need a DAW that can handle the very small temporal adjustment shifts necessary to preserve the phase integrity up to high audio frequencies. Again, source recording at 88.2 or 96KHz eases this chore, since for audio captured at these rates, irregularities in the top octave are less audible in the final mix. I'm not being funny here; frequencies in the 20-30KHz band generated by transients can be experienced by people (such as me) whose hearing measured in the steady-state falls off at around 15KHz. Look back at some of the threads we have had in RO concerning perception of high frequencies and how acoustic sources such as tingsha bells that have significant energy at high frequencies can really show up differences in top-octave behaviour of audio systems and processing. Get it right for a simple source such as a bell and then a symphony orchestra is suddenly a whole lot clearer.