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Routing analog & digital audio between two DAWS

Hey all,

I'm doing the planning for my new system.
I have magix samplitude pro x, and I'm considering protools HD12.

Magix would be the main capture/compose/edit system due to high track count and clean coding.

PTHD would be primarily for mixing (mainly volumes and panning). Since it does 10 video tracks and 7.1 it's the unfortunate (expensive) choice.

Basically id like to pipe the edited audio from Sam into PTHD via the digital outs RME babyface -into- focusrite Scarlett 18i20.

I've been told in the past 'once it's digital, it's digital' but after learning I've seen there's room for coding and error rates.

I'm just curious if this is a 'safe way' to move essentially finished tracks into the mix daw. PTHD does 64 audio tracks/10 video tracks at 192k. This is where I'll combine the audio and video.

I alsk will have magix movie edit pro premium which handles 4 camera angles.

So I'll be piping audio and video from the magix to PTHD.

Eventually I'll be able to afford Sequoiawhich does many things particularly on the broadcasting side that I'd like. But I'm
About 3 years away from that.

Basically is there a better way to pipe audio over than re-recoding via the digital outs? Is simple drag and drop from my NAS drive better?

Is there a better software combo? A different method to do what I'm describing? I'm open to any ideas.

If PTHD isn't needed I'll get the regular version to open my old projects. It's only limited to 1 video track however.

Comments

Profile picture for user Brother Junk

Brother Junk Thu, 10/06/2016 - 08:10

Ok,

I have three (video) monitors if I want, 2 computers, and @audiokid turned me on to the two daw idea and said to research...so I've been doing it (this English guy in the last vid is a trip) I have a couple q's now.

I looked into VSL once bc of how fantastically amazing it is, but it was hella money. IIRC $2k for the full boat? Hot damn is it nice though. I can't imagine that's dropped to $.02 lol.

From what I've read the daws don't seem to matter. Eg. I have Logic x and PT10/11. Seems like those will work fine, or I can just use the same thing on both.

What I'm a little unclear on is what the separation gains you? The last vid on this page, I'm with him up until about 4:40 then he starts talking about it. So, is it simply that you you can use one computer to run all the VST's, plug-ins etc, but it tracks to a different daw, that has no plug-ins and therefore more power on tap? If that's the reason, I get it...I just want to make sure i'm not missing something. Or is it that too many plug-ins, cause errant 0/1's and it's how you avoid that?

Also, I haven't noticed this anywhere, are both computers are plugged into the DAC e.g. with firewire? Or does the slaving just make the slave tracks come through the monitors? It seems like if they were both plugged into the dac you wouldn't gain the full benefit?

I think I'm almost there, and I'm actually kind of already set up to do most of this. I never saw the virtual effects rack part of vsl. I've never heard of a virtual effects rack ever actually. That would be nice. Logic comes with some decent plug-ins. To be able to use them in PT (I prefer PT) would be nice.

I will go keep reading.

Oh, and I found this rack porn, maybe AK's, dunno, I found it under his name? Either way it's gonna take me 10 minutes to clean up the drool.

Can anyone tell me (ballpark) what that room right there would cost? I'm asking bc I'm totally ignorant about all that, but that looks $$$$!

P.S. I have read a fair amount about it, but a lot of it is just so far over my head. Like the thread by you @audiokid about hybrid daws. My vocabulary in this field is not as good as yours. I read that thread and vertigo kicks in lol. So, I may just ask a bunch of smaller q's

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Brother Junk Mon, 10/24/2016 - 07:41

kmetal, post: 442344, member: 37533 wrote: Logic has a great reputation . I've never had the opportunity to use it. I think a lot of people who are into more electronic forms of music, and using a lot of loops sway towards logic as their main platform.

It does excel with those sounds. Better than PT imo.

Just in general, for $200, for how smooth it is, rarely stalls on you, a ton of plug-ins...it's pretty damn good.

The new version has a virtual drummer which is pretty cool. I haven't messed around with it a lot but it's cool for setting a basic groove and writing the song. Then I can just copy the groove and play it plus all the fills etc on my Roland TD-11's.

I don't use it a ton bc I want to get faster with PT, but it's pretty solid. I'm not a huge fan of the work flow for editing but again, $200 - comes with a ton of sounds, and a ton of plug-ins, and it works flawlessly on the Mac (well, it only works on Mac) what I mean is it's fast and smooth, and it almost never pukes on you.

I'm becoming a bit of a daw junky though, so take it fwiw. I can't think of one that I truly dislike.

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dvdhawk Mon, 10/31/2016 - 12:12

DonnyThompson, post: 442754, member: 46114 wrote: At that point, do we perhaps need to look at the engineer's lack of talent in getting the best capture possible? That's not rhetorical, gang... I'm really asking here...
Personally, I don't know how else to look at it, because if you had all the aforementioned pro-level equipment, and you still feel the need to "fix" certain tracks, then I think you have to look at the possibility that the fault lies with the engineer, because it's certainly not the gear, right?

Again, I'm not being rhetorical... my question(s) really are sincere; I'd like to know what my peers think...

@DonnyThompson

This could probably be a separate thread too. But to give my answer to your question, I think you and I have a similar views on this.

If someone is getting good results, and having some success using a particular approach - I'm all for it, whatever works for you. The SOS guy probably acquired one widget at a time and applied them on top of what (one would hope) was a pretty quality recording to begin with -given the level of gear and expertise. Each new plug-in probably gives it something he finds .1% more pleasing to his ear. I would hope he doesn't need them for grand sweeping adjustments, or to compensate for poor tracking.

I try to use plug-ins very sparingly, but like a lot of you I usually have a pretty clear vision of where the mix is going to end up when I'm tracking - so I don't hesitate to print EQ, or even modest compression if I know that's going to stick. We all know that you can have your kick, snare, hi-hat, and bass guitar forming the absolute perfect pocket in the mix, but if you solo'ed any one of them they might (as @Kurt Foster would say) 'sound like ass'. For me, it's always better and more efficient in the end, to spend an hour trying different mics and find the sweet spot to aim them, versus fighting the mix every hour after that. Most of the tracks, I might not need any EQ on them unless it's for a specific effect in a specific song. Better signal in -> better signal out. Garbage in -> plug-ins -> filtered garbage out. (no matter how many times the folks on the ISS filter the water…. they're still drinking urine).

That being said I do routinely use plug-ins as needed, primarily for EQ, compression, delay, and reverb. I'm always mindful that there's going to be a trade-off when algorithms are involved. Computational error, even if it's usually not noticeable, is sure to leave a cumulative pile of artifacts if you overdo it.

As far as the plug-ins themselves, I'm under no illusion that a $50 - $300 plug-in can perfectly emulate every nuance of a $30,000 piece of hardware, but that doesn't mean they're of no value. And as it's been said before, no two pieces of hardware are truly identical either. I've never had my hands on a Fairchild or Pultec, so how would I know? All I know for sure is that I like what a BF LA-2A plug-in sounds like and use it more than the stock compressor. I like the Pultec EQ plug-in that I have, and I use it in certain situations, but less often than the stock parametric in StudioOne.

I've personally been doing a version of the decoupled DAW thing for a long time when a project merits it. I have a buddy with some upscale hardware, and I do the editing / mixing ITB, and we pass that stereo mix in realtime through his rack hardware and record the resulting 2-track on a separate DAW at 44.1kHz. The capture DAW will usually have a limiter on the inputs, but basically we're setting levels as if we were going to DAT, or any other 2-track recorder. Ideally, we won't need to nudge any levels once it's been captured into the second DAW.

The core piece of hardware in that process being my buddy's Avalon 747. I haven't found anything yet that doesn't sound noticeably better just by virtue of passing through it - even before you engage any of its functionality. If it's from a cold start, you do have to let it warm-up for 30 minutes or so, but then it's rock-steady after that. The tubes give the sound instant gravity, the tube compression circuitry is great for what we do. It's not overly dense or dark, but I can see where some might not like it for classical music. Luckily, we're not recording the Frogtown Philharmonic. If you haven't used a 747 you might not believe the icing on the cake is the 6-band graphic EQ. The center-frequencies, the Q, and the amount of cut/boost of each band have been carefully tailored individually (by someone with exquisite taste) so that each band is perfect and incredibly musical. You can sweeten the track, you can completely change the character of the track with radical settings, but you cannot ruin a track (even if you're trying to for sake experiment) with the stupidest comb-filtery looking 3-up / 3-down EQ settings you can think of. The character will change, but the mix will not come undone on anything we've tried.

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bouldersound Mon, 10/31/2016 - 12:35

Brother Junk, post: 442756, member: 49944 wrote: That's the first thing I thought of. But you guys are the ones who have toyed with the real thing. I've never messed with a real compressor in my life. But I've always thought of the plug-ins as the secondary, or lesser option. But that's an assumption...I've never actually gotten to compare.

A decent compressor plugin is way better than a run of the mill analog compressor. Really high end hardware compressors do things that can be hard to emulate digitally. Actually, all compressors to things that are hard to emulate, but what normal compressors do isn't worth emulating.

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audiokid Mon, 10/31/2016 - 12:56

bouldersound, post: 442817, member: 38959 wrote: A decent compressor plugin is way better than a run of the mill analog compressor. Really high end hardware compressors do things that can be hard to emulate digitally. Actually, all compressors to things that are hard to emulate, but what normal compressors do isn't worth emulating.

+1

To add: I have owned some of the best and I wouldn't waste a dime on any hardware compression when it comes to hybrid mixing or mastering now. They are all a complete waste of money and time.
ITB is 100% better in all respects including being able to side-chain better than any hardware comp can ever do. Which is where it all comes of age.
Tracking though, love them. Especially the down right dirty UA tube and tranny stuff.
Clean comps, ITB is once again, superior.

To my ears... The higher end the hardware, the more you realize digital is better.

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Sean G Thu, 10/06/2016 - 17:25

Brother Junk, post: 441917, member: 49944 wrote: What I'm a little unclear on is what the separation gains you?

You mix in one and capture in the other. Allows for mixing & processing at higher sample rates and capturing at say 44.1 /16 bit for CD without sample rate conversion issues.

Brother Junk, post: 441917, member: 49944 wrote: Also, I haven't noticed this anywhere, are both computers are plugged into the DAC e.g. with firewire?

Two sets of converters are required...one for coming out of DAW 1 and one going back into DAW2. Firewire is one option.

Brother Junk, post: 441917, member: 49944 wrote: Can anyone tell me (ballpark) what that room right there would cost? I'm asking bc I'm totally ignorant about all that, but that looks $$$$!

I can't speak for Chris (@audiokid ) who owns that set up, but for something along the lines of a set up like that I don't think you would get away with spending anything less than 6 figures.

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kmetal Thu, 10/06/2016 - 18:03

Brother Junk, post: 441917, member: 49944 wrote: I looked into VSL once bc of how fantastically amazing it is, but it was hella money. IIRC $2k for the full boat? Hot damn is it nice though. I can't imagine that's dropped to $.02 lol.

It's an expensive endevour to get the full kit. I purchased the VSL pro 5 player on sale for $180, plus $30 for the veinna key license dongle. The player, which is the part that hosts vsti and plugins, and a basic but great sounding set of samples comes with it. The audio fidelity imho cannot be touched by any other orchestral sample company. You get what you need and want and can afford.

Brother Junk, post: 441917, member: 49944 wrote: What I'm a little unclear on is what the separation gains you? The last vid on this page, I'm with him up until about 4:40 then he starts talking about it. So, is it simply that you you can use one computer to run all the VST's, plug-ins etc, but it tracks to a different daw, that has no plug-ins and therefore more power on tap? If that's the reason, I get it...I just want to make sure i'm not missing something. Or is it that too many plug-ins, cause errant 0/1's and it's how you avoid that?

Vsl player is different than a decoupled/mixdown 2 daw workflow.

The vsl player allows you to basically use an additional computer connected via Ethernet, and allows you to host vsti and use audio plugins on a seperate computer to take the load of the daw. This comes in handy w cpu intensive plugins an vsti's.

I'm going to use my vsti computer(s) to host vsl orchestra, and BFD, which are fairly intensive. This lets me keep buffers low on the main daw.

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Kurt Foster Mon, 10/31/2016 - 13:30

audiokid, post: 442824, member: 1 wrote:

To my ears... The higher end the hardware, the more you realize digital is better.

or the better the hardware the better anything sounds. that's a no brainer. i don't believe it can be attributed to solely digital however. i just don't agree. digital is fine. it works. but it's not better. just different. perhaps you like it more and that's fine. there's nothing digital has to offer that analog won't do as far as actual documentation of a performance. digital is easier to edit. that's the only advantage i can see. but then, i know how to pull down a fader at the end of a track. :ROFLMAO:

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audiokid Mon, 10/31/2016 - 15:38

Kurt Foster, post: 442832, member: 7836 wrote: or the better the hardware the better anything sounds. that's a no brainer. i don't believe it can be attributed to solely digital however. i just don't agree. digital is fine. it works. but it's not better. just different. perhaps you like it more and that's fine. there's nothing digital has to offer that analog won't do as far as actual documentation of a performance. digital is easier to edit. that's the only advantage i can see. but then, i know how to pull down a fader at the end of a track. :ROFLMAO:

+1 for tracking.
-1 for mixing or mastering.
:cool:

Once ITB, stay ITB. the love of analog compression lives in tracking and ends at mixing. You won't see money on analog mixing or mastering compressors ever again. For front end though, love them :love::love::love: so we are both going to be smiling on that.

Until you get up to great conversion, I can't say I blame you but once you have the best of both worlds it goes like this....

Digital compression is extremely accurate and very fluid.
Digital compressors do all sorts of functions from extremely subtle to broad stroke aggressive impact, brick-wall limiting to surgical de-essing and triggering other freq's in the most creative and dynamic, or delicate ways, no analog compressor could ever compete.

... and stereo digital compression rocks. :D

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Brother Junk Fri, 10/07/2016 - 08:01

Sean G, post: 441936, member: 49362 wrote: Two sets of converters are required...one for coming out of DAW 1 and one going back into DAW2. Firewire is one option.

TY...so the guy above (last vid posted in this thread), that is not a true dual daw system? Or at least not what audio kid and a couple others here are doing?

Sean G, post: 441936, member: 49362 wrote: I can't speak for Chris (@audiokid ) who owns that set up, but for something along the lines of a set up like that I don't think you would get away with spending anything less than 6 figures.

Oh....sorry @audiokid, I thought you were just posting a pic, I didn't realize it was your room. I wouldn't have been so rude to ask how much had I known. But wow...what a space man. It makes me teary just looking at it.

kmetal, post: 441940, member: 37533 wrote: I purchased the VSL pro 5 player on sale for $180, plus $30 for the veinna key license dongle. The player, which is the part that hosts vsti and plugins, and a basic but great sounding set of samples comes with it. The audio fidelity imho cannot be touched by any other orchestral sample company. You get what you need and want and can afford.

$210 isn't so bad.

kmetal, post: 441940, member: 37533 wrote: The vsl player allows you to basically use an additional computer connected via Ethernet, and allows you to host vsti and use audio plugins on a seperate computer to take the load of the daw. This comes in handy w cpu intensive plugins an vsti's.

So, for $210, would that get all the benefits of a dual daw system? Or you still need 2 contverers and the summing mixer?

I think what's confusing me is that the vid with the English guy demonstrating VSL seems to be doing, essentially, what a dual daw setup does?

I think maybe my brain just needs a lil break before I cycle back through the info as I'm ending up with more q's instead of fewer. I appreciate the patience. You guys are hard to keep up with!

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audiokid Fri, 10/07/2016 - 11:24

Brother Junk, post: 441961, member: 49944 wrote: Oh....sorry @audiokid, I thought you were just posting a pic, I didn't realize it was your room. I wouldn't have been so rude to ask how much had I known. But wow...what a space man. It makes me teary just looking at it.

No worries, its worth mentioning because it is an amazing system. Only because you asked, I put about $130,000 USD into it, had another $60,000 in various gear I didn't end up using because ITB was superior to my ears. I've sold about half of it and am going in slightly different direction. My focus is still 2 DAW's, passive summing as a character section with Sequoia for both DAW's.
2DAW's rocks and you do not need to invest in all that to achieve sonic excellence.
A lot of what you see is also for tracking and mastering. Most everything in that picture was used for tracking, mixing and mastering. Its setup to be used in any direction at a click of a mouse.

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Brother Junk Tue, 10/25/2016 - 07:14

audiokid, post: 442116, member: 1 wrote: That being said, if that option was waved, I may take on that challenge. My DAW system meets world class sonics which will not degrade the path. In fact, if I was provided 96k tracks, real time SRC (DAW1 > AD> DAW 2, would sound better to my ears over bouncing down. To my ears capturing at the destination SR still sounds better when its done in real time which takes two DAW's.

What do you mean by the above? Specifically this part (I think) "In fact, if I was provided 96k tracks, real time SRC (DAW1 > AD> DAW 2, would sound better to my ears over bouncing down."? I see your parenthesis flow chart, I'm wondering about someone not in your system...

I ask because I have an 08 Mac Pro, dual Xeon (I forget what speed, but fast) 12 g ram, 3 hard drives. The ram is not equal in the channels. E.g. There are 4 channels, but 12g of ram. Ideally you want 4g x4, or 8g x 4...same ram, same speed etc. But I don't think that is causing this.

Because of that conversation had here, (I would link it but I forget where it was) about sampling at 96k vs 48 (the artifacting effect below the nsf etc) I started running PT at 24/96. I have always done 24/48.

What is happening is that I've got 2 tracks going, and 2 plug ins, each in a separate track/bus (reverb, delay). I keep getting that message that says CPU overload (or whatever it is) and to increase the hardware buffer size. Well that's gone all the way up to 2048, and it's still puking on me. I have nothing notable on the Mac. Just PT, Logic, and whatever comes with Mavericks OS. That is literally, all that is on it.

At Surefire, one of the computers they have is the same as mine. Except they have spotify, pandora, endless programs on it. You bring up the apps, and it's pages of them, and I've seen them do all kinds of crazy routing, bussing, plug-ins, high track count etc. It's a pretty crazy amount of @$%$. Mine is so bare bones...

What am I doing wrong?

And @audiokid (or anyone)...I entered this site loudly with debate about something. I said at the time, that it just happened to be one topic that I know a lot about (driver design)...and I said at the time, that I'm probably going to flood this place with questions so stupid you will be in awe...and I feel like this is one of them lol, ready?

Assuming the uneven ram spacing/count isn't to blame, can I record in 16/44.1 for the track count. And I mean literally record with a mic. And then when I want to mix/master, change the project to 24/96? Or is that where the sampling errors come in...the errors that your setup avoids? If I record it at say, 16/44.1 so that I have a high track count...can that just be changed later on to 24/96? Or if it's recorded at 16/44.1, does it stay there forever? Essentially, is up-sampling (is that the term?) to 24/96 after the recording is done, just a gimmick that adds 0/1's to make it fit? Or will it be a genuine 24/96 (plus a few errors?)

I'm wondering if the problem I'm having at 24/96 with PT sounds normal to you? And fundamentally whether or not I misunderstood that conversation about 48khz vs 96. Once recorded, am I really changing the bit depth and sample rate if I change it?

p.s. I'm a mess (told you lol). If you can make out what I'm asking above, I'd appreciate it. I do understand bit depth and sample rate, but I guess not how it works in a daw after recording, e.g. changing from 16/44.1 to 24/96 after recording....does it work that way?

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DonnyThompson Tue, 11/01/2016 - 06:46

Brother Junk, post: 442870, member: 49944 wrote: What does this term mean?

In studio lingo, the word "tracking" is synonymous with "recording". ;)

"Poor tracking" would include things like using substandard mics, or bad mic placement, or a bad sounding room, or clipping the inputs of your audio capture device, or not having sufficient gain available for lower output mics to perform at their optimum...
It would also include electronic noise, ground problems, RFI, bad room reflection, etc.

It's basically anything that will degrade the quality of sonics of the tracking ( recording) process.

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Brother Junk Tue, 11/01/2016 - 08:25

DonnyThompson, post: 442872, member: 46114 wrote: In studio lingo, the word "tracking" is synonymous with "recording". ;)

"Poor tracking" would include things like using substandard mics, or bad mic placement, or a bad sounding room, or clipping the inputs of your audio capture device, or not having sufficient gain available for lower output mics to perform at their optimum...
It would also include electronic noise, ground problems, RFI, bad room reflection, etc.

It's basically anything that will degrade the quality of sonics of the tracking ( recording) process.

Thanks. That's what I thought but wanted to be certain.

Kurt Foster, post: 442873, member: 7836 wrote: just kill me.

I'm not sure if you are annoyed by the stupidity of my question, (maybe it's unrelated) but if it is my response, it has a reason. I've been doing a lot of back reading on the site and I'll see things like "poor tracking", "less than adequate tracking", "tracking problems", "tracking daw"etc.

I just wanted to make sure that I was understanding it correctly. The statement was something like, "don't use a plug-in to mitigate poor tracking" (paraphrase). I just wanted to make sure I understood the intention of that advice.

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kmetal Tue, 10/25/2016 - 20:08

Brother Junk, post: 442593, member: 49944 wrote: I ordered the Vienna Ensemble Pro 6 to do the above. Can't wait...

Thanks @kmetal , I never would have known about this feature otherwise, and it's a major selling point for me.

Congrats!! Vsl is a great company and quality product.

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kmetal Fri, 10/07/2016 - 21:05

Brother Junk, post: 441961, member: 49944 wrote: So, for $210, would that get all the benefits of a dual daw system? Or you still need 2 contverers and the summing mixer?

I think what's confusing me is that the vid with the English guy demonstrating VSL seems to be doing, essentially, what a dual daw setup does?

I think maybe my brain just needs a lil break before I cycle back through the info as I'm ending up with more q's instead of fewer. I appreciate the patience. You guys are hard to keep up with!

These are two seperate concepts BJ.

The video is showing how to have a dedicated computer for virtual instruments and pluggin processes, that is connected to your main daw computer and session.

Think of it like having a a Dsp card, or outboard effects processer, but it's a computer instead. The video illustrates a way to ease the load on the daw computer, for heavy hitting vsti like BFD and vsl.

This configuration is known as a master/slave computer. It's common to have more than one slave computer for running full orchestra software synths for movie soundtracks. Lol I think Hans Zimmerman runs 13 computers at once. Don't quote me on that but i seem to think I saw him say that in an interview.

The 2 daw system Audiokid employs is for completely different reasons.

It is decoupled, meaning the two computers are running seperately, as opposed to master/slave where there running as one big computer.

In the decoupled daw system the summing mixer is the link between them. The general idea is to make use of the summing power (headroom) of analog, instead of the daw master bus, and to avoid SRC in the box.

So you make a mix in daw 1 send it broken into stems thru the summing mixer, and record the mix as a new steroe track in daw 2 at whatever sample rate it's going to be delivered in.

Your basically re-recording your multirack mix in stereo, in a very broad sense.

Boz and audio kid are the pioneers of this, I'm just a student of it, and don't yet have my setup complete to do it in practice.

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audiokid Tue, 10/25/2016 - 20:31

Brother Junk, post: 442594, member: 49944 wrote: And @audiokid (or anyone)...I entered this site loudly with debate about something. I said at the time, that it just happened to be one topic that I know a lot about (driver design)...and I said at the time, that I'm probably going to flood this place with questions so stupid you will be in awe...and I feel like this is one of them lol, ready?

No worries, we are all learning. No question is a stupid question either.

Brother Junk, post: 442594, member: 49944 wrote: Essentially, is up-sampling (is that the term?) to 24/96 after the recording is done, just a gimmick that adds 0/1's to make it fit?

Upsampling would be a complete waste of time, imho, unless for some reason you do this to create a special effect.

On that note: If I was to Upsample, (we used to do that in the 80's thinking it was improving the older 8bit samples)

I would only do this now.... if I was mixing as session in example 44.1 , DA > analog mix gear to add analog "flavour or effect, > AD> capture the analog mix back on a second un-coupled DAW at example: 96k in order to preserve a higher SR analog capture. But even then I would most likely avoid the 96k capture and simply get it at 44.1 as well. But I'm also assuming I am summing at this stage of the mix too.
Sorry if this is confusing you.

To simply answer your question. Don't bother Upsampling. The less SRC (sample rate converting) Up or Down the better.

https://en.wikipedia.org/wiki/Sample_rate_conversion

https://en.wikipedia.org/wiki/Upsampling

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