Routing analog & digital audio between two DAWS

Discussion in 'Computing' started by kmetal, Sep 9, 2016.

  1. audiokid

    audiokid Chris Well-Known Member

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    Thanks for chiming in and sharing, Bos. To clarify, we are on the same page here as well.
    I share the same principles in regards to pro-level mastering. I am a firm believer in having another set of ears employed at the mastering level too.(y) When tracking or mixing, if a client has the budget, I would always recommend a pro-level Mastering Engineer to finish up their work.

    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. Even better if both have mastering capability specs.

    It would be fun to do a (SRC) sample rate conversion shoot here one day. Maybe when I finish up this next DAW build, we could challenge a pro-level ME to participate in that. (y)

    Same. Which is why the dedicated monitoring controller is so vital. Two uncoupled DAW's with an independent monitoring controller removes the shackles of having anything fixed :)
     
  2. audiokid

    audiokid Chris Well-Known Member

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    (patch bays and routing hardware)

    Here is where things get exciting. A system such as I describe can capture or exchange analog or digital information freely between both DAW's. I incorporate digitally controlled analog routers that connect analog mastering or tracking hardware to be used on either DAW to track, mix or master as well.
    Tracking, mixing or mastering hardware can switch gear positions to before or after in a grouped or up-grouped chain via digital switching routers.
    Switching routers can organize hardware in a matrix that connect to DAW1 or DAW2 or both.
    A matrix can have manual buttons such as the Dangerous Liaison and Dangerous Master or move gear via digital commands through something like the SSL X-Patch. The X-Patch will execute switching via a mouse or midi command. Wow.

    Example:
    • I can choose from a list of analog comps, EQ's processors in my rack for that task.
    • I am able to digitally move gear to be in front, middle or behind (ABC, BCA,ACB etc) in a bus lane or channel strip, mono or stereo process.
    • I am able to do seamless changes of this hardware on the fly, which makes comparison learning very powerful.
    I use two DAW's to capture, organize and make notes. The mass of this workflow can be saved so I can refer back to stored setting for extended learning and/or to repeat what worked for a particular project. This workflow bridges analog and digital together better.

    I can only imagine the power digital audio is going to reach in the years to come.
    If I was 17 again, preparing to make a name for myself in this industry, this is where I would be looking. I most likely would also add a 2" Tape machine just for the buzz of it all.
     
  3. kmetal

    kmetal Kyle P. Gushue Well-Known Member

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    So are you saying you print different 2trk mixdowns with different settings in the same pass?

    Do you find yourself making adjustments to the multitrack based on what your hearing when you've got the summing box/capture daw active?

    If so, at what point do you introduce the capture side? Do you have it active the whole time?

    Are you running any processing in the capture daw like a limiter, or is the capture daw the final snapshot, i.e. Once it's captured that's the final.

    When you give mixes to cleints do you have any bus style compression or limiting? If for no other reason just so the clients get a feel for the finished mix, or so they don't have to adjust the volume?

    Interesting hawk. It's something we should all consider. Lol I've got stacks and stacks of CD-Rs w the black sharpie writing on them. Definatly something I'll keep in mind, particular the on disk printer.

    That would be great! It would also be a useful reference to hear an ITB bounce vs the realtime capture a of the same mix. Especially if the decoupled sum/capture didn't include any additional processing.

    The whole digital patchbay really changes the game on how analog gear gets incorporated.

    Do you find yourself adjusting the analog unit knobs/settings? Or are you doing the CLA thing where the settings on the boz pretty much stay the same.

    Does the digital patchbay have adda? If now how is the analog gear connected? Right to I/o on your interface?
     
  4. audiokid

    audiokid Chris Well-Known Member

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    This is precisely what I am talking about.

    Indeed.

    Very good question. After extensive testing with Sequoia as the DAW, all but for a few specialty products do I feel outboard gear beats plug-ins. This would be Pultec MEQ-5 and Processors like the Bricasti.
    I use those manually and adjust "knobs" to suit the mix. If I need to recall those, I write down the settings.

    I hear a simple analog pass sounding better than a pass full of analog of all sorts of flavors. The more analog introduced to a mix, more it turns the mix into a mongrel effect, where it looses the wow factor to me.

    In my new 2DAW build, I am going to a Folcrom and one pre-amp to flavor it. Eliminating the console and most of the hardware now.
    A Bricasti has way more weight in how a mix ends up to all the thousands of dollars in analog gear ever did. Sequoia software has replaced the hybrid hardware bloat.

    So if you can understand what I'm getting at, I love my analog gear but I like it best for tracking. I don't foresee big leaps in analog useful for mixing anymore. The big rail preamp, the Bricasti's, MEQ-5 Pultec in a passive summing pass sounds like all I need. The rest is ITB.

    A digital patchbay does not have digital I/O or conversion. The only thing digital about it are the triggers that switch the relays. However, their obviously is a digital matrix program within it that connects all the options for the routers to switch. Does that make sense?
    They are amazing and a must for me. If configured properly, you can plug all your gear into them and never have to pull a cable. The down side to them at present, you need a lot of them if you use a lot af analog gear, and they can be very confusing to set-up. I have my analog gear downs to just a few specialized products now so one X-Patch is plenty for me now.
     
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  5. kmetal

    kmetal Kyle P. Gushue Well-Known Member

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    Awsome, that reply clears up a lot. I think I finally understand how each component plays into the system as a whole.

    I think it's a mentality in cooking where you don't use more than 3 seasonings in a plate.
     
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  6. audiokid

    audiokid Chris Well-Known Member

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    This is precisely how I hear it. There is a fine line between wow and mush. Much like how a duplicate track created for the stereo effect or fatness can go from fat to phasy. My last mix taught me a lot. Less analog in the pass, how awesome the Bricasti is and I mix back and forth religiously in mono more than I ever have.
     
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  7. kmetal

    kmetal Kyle P. Gushue Well-Known Member

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    I've forgotten I was in mono on more than one occasion. If you subconcosuly start bobbing your head you know you've got a good mix. If your mix is good in mono, it's almost always gonna be good I stereo.

    I find mono helps w vocal levels, and making room frequency wise for each instrument.

    Are you using a stereo set w a mono button? Or do you have a single speaker mono reference?
     
  8. audiokid

    audiokid Chris Well-Known Member

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    The Dangerous Monitor ST has a mono button. Works excellent.
     
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  9. Boswell

    Boswell Moderator Distinguished Member

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    Not very often, but the point is that the method of working allows this approach. I sometimes get jittery clients sitting with me at final mix stage, and they are very anxious to take away tracks that they think will cover all their options. In these circumstances, I can put the 2-track analogue result through two or more compressors on different settings and capture the compressor outputs in parallel. The downside is that it reduces the chargeable hours, but the upside is that it saves having to sit through their (often tedious) tracks several more times.

    In the sense that do I listen to the captured 2-track, yes. There are often subtleties between the analogue mix and the capture that are best adjusted in the mix and not in a post-capture process. However, I get the balance of the raw mix roughly right first, and then pay more attention to how it sounds on capture.

    The captured analogue output is the final capture, but this is usually going off to an ME who will naturally make changes or perform some processing on the tracks. But it is why sometimes I can use a simple 2-track capture device rather than a DAW for the second box.

    Occasionally, I will do a limiting step on the digital 2-track capture, but it would only be (a) if the mix has large excursions that would result in a low mean level after normalisation and (b) it's not the result that goes to the ME.

    In one of the past threads on the 2-box process, I explained that one of the starting points for me in developing the 2-box method was to try to re-create in modern terms the old "direct-to-disc" recording of the '60s and '70s, where I was convinced I had an enhanced listening experience over studio-processed recordings. Think of the output of box 1 followed by analogue mix as the output of a stereo microphone pair, and it's the job of box 2 to capture it as though it were "direct-to-disc". This is also the reason that I try to run box 1 at 96KHz to avoid top-octave phase effects, and present an analogue 2-track mix for capture that is otherwise indistinguishable from the output of a sophisticated stereo microphone.

    Yes, depending of the material and the purpose of the mix.
     
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  10. kmetal

    kmetal Kyle P. Gushue Well-Known Member

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    Thanks boz! Lol about jittery cleints and final mix anxiety! I'm familiar with that!

    It's an interesting idea to print the mix to a a few different tracks in parelell, especially w a passive summer like the rolls device, which relies on external pre amps for makeup gain. It gives an opportity to use a transparent, subtle, and heavy handed set of pres, all at once. Lol anything to put off a commitment is good. Just kidding. But i could see some advantages even if the choice was left to the ME.

    I wish I was lucky to enough to work on projects that got mastered by a true ME!! Cheers to you for that sir!!
     
  11. Brother Junk

    Brother Junk Active Member

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    I don't know if they are any good or not, but Logic has a ton of different amps, mics, mic placements, number of mics etc. Ribbons, condensers,

    I actually like Logic for composition. I find editing with it to be slightly painful. For $200, it comes with a decent amount of stuff. I find sometimes I compose with one, and edit with the other.
     
  12. kmetal

    kmetal Kyle P. Gushue Well-Known Member

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    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.

    In general my new set up is focused around cross platform compatibility, and full sample rate support. This is to allow me to move around from place to place, and allow me to easily transport files. I think as Remote recording and producing takes it's foothold compatibility is going to be of utmost importance for keeping things smooth and effortless.
     
  13. Brother Junk

    Brother Junk Active Member

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    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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  14. Brother Junk

    Brother Junk Active Member

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    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.
     
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  15. Brother Junk

    Brother Junk Active Member

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    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?
     
  16. kmetal

    kmetal Kyle P. Gushue Well-Known Member

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

    audiokid Chris Well-Known Member

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

    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
     
  18. Brother Junk

    Brother Junk Active Member

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    Thanks! I've actually read both of those pages already. And nope, not confused (with regards to your reply at least). I basically understand how your system works now (after a lot of reading) and the differences of yours vs mine.

    So, I'm with you on the less SRC the better. I can see the benefit of not changing it all the time, or how this could introduce problems, in either system.

    Does the PT problem I'm talking about make sense to you? Is 24/96 so labor intensive that PT on that computer should be giving me that CPU overload speech? Telling me to remove plug-ins when there are only two, very tiny ones, being used, or increase the hardware buffer size when it's already at 2048?

    Or maybe, what would help me is if you guys could tell me what you typically work in...are you doing most stuff in 16/44.1 for the track count? What is common practice?

    Or do you do all sessions in 24/48 and just bounce down? (Wouldn't that be SRC?)

    Whether or not it's true (it's not pivotal to my question) I was told once that all mastering is done in 24/48...so I assumed I would want all sessions recorded in 24/48. And I think most of you are in agreement, that ideally, if I was going to master a project in 24/48, I would want the whole session to be in 24/48 right from the start, no?

    Then I read that discussion about the artifacting that can occur at 48 vs 96. So I tried a session at 24/96 and it won't run. I'm wondering if that sounds about right for an 08 Mac Pro?

    Or is that just an unnecessarily high bit depth/sr?

    If operating under the assumption that any SRC beyond the initial capture is bad (which I think is the crux of what you are saying) should I not be setting all sessions up as 16/44.1? Since that is what they will be bounced to anyway?

    Last q (for the moment lol)...I've read that at 16 bd, there is no point to running a higher sample rate than 44.1 (that doesn't make sense to me, but whatever) Do you guys ever set up sessions as 16/48? Or 16/96? What I'm wondering is this: If my computer can't handle 24/96...and perhaps I can only get 6 tracks in at 24/48 before getting the CPU overload speech, is it the bit depth causing the problem? Or the sample rate? Or is it an even 50/50?

    I realize that was a lot of questions and not very well articulated. Essentially, I'm wondering how to solve my track count problem in a way that preserves the highest fidelity possible. I can try setting up a bunch of sessions in different variations to see what works...but I would like to know the "on paper" answer as well.

    For you guys running these Mac towers (like the 08/09 era), how much ram are u running? I'm off to see if I can find the answer to that last one now.

    TIA

    **Edit, it appears people load up with 32gb of ram when possible. But many are just using 16gb.
     
  19. audiokid

    audiokid Chris Well-Known Member

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    I'll chime in on a few of your questions.

    If I am working a project, using my full workflow (2-DAW system) I would track at 96 and mixdown t0 44.1 or whatever the final destination mix was calling for or whatever a client asked for.

    Basically, if sonic's is the goal, I would track at 96k and go from there. I can go up to 192 but better converters sound beautiful at the comfortable compromise which is 96k. My Lavry Blacks are a beautiful sounding converter that doesn't go above 96k anyway. Read up on Dan Lavry.

    Being said, I suppose I would use the highest SR (DSD) if I was developing a library or under extreme sonic archiving. I have had DSD here as well. I owned 2 DSD Korg's here and although they sounded pure as gold... at the end of the day, my 2-DAW workflow produced sonically better mix's and it was much faster to get there.

    We all do our thing.
    Some Mastering Engineers like to get the fullest bandwidth to start with. To my understanding most of them do not sum into an uncoupled DAW like what I describe but there are some that do and those would be my choice to hang with (n).

    There are no rules though, but I do think music sounds better with less bouncing and capturing your SR in real time as apposed to bouncing down.

    To add... If the converters aren't great, my method of the 2-DAW approach is less favourable. Everything is subjective and there are no rules.

    Not to confuse you but here is another "subjective" way of putting this.

    Good converters should sound excellent at 44.1. The cheaper ones will not. Good converters not only sound better at lower SR, they also save CPU load, thus allowing smoother work ITB with less CPU related issues and hard drive consumption.

    If you are using prosumer gear.... I would suspect you are better off tracking at its optimal SR.
     
  20. Brother Junk

    Brother Junk Active Member

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    Gotcha brother. I think what you are saying is, that with the quality of gear YOU use, you could capture at the optimal bdsr, and one bounce down may not kill the cat. But because of the quality of your gear you could capture at 16/44.1 and it will sound good anyway, so that's what YOU would probably do, to avoid the src. I think I'm going to have to set up a bunch of sessions and find my best compromise.

    I found out that the studio I like, that uses the same Mac Pro tower I have, was 16/44.1, hence the number of tracks I think? I'm such an idiot, I never thought to open one of the projects that was started there and I sent to myself to finish at home. So that idea came to me, and they were 16/44.1

    They got a trash can Mac so they can run the higher bit depth and sample rate. So, for what I'm doing, I think 16/44.1 will work fine. Every place I market to has that limitation anyway...so I'm bouncing down no matter what.

    I get ya. I feel like I have a pretty good handle on how you are set up vs me now. I would imagine if the design includes one crappy converter, changing the design to incorporate 2 crappy converters isn't going to help. And since you have the analog portion, that piece, has got to be top tier. And good analog stuff is $$$$$$$

    My converter is the Avid Mbox Pro (3rd Generation) fwiw. Honestly, I think I liked the Scarlett unit I had prior to the Mbox, better.

    I've never actually tried listening for sonic differences between 16/44.1 and 24/48 on any of them. (I had another converter but I can't remember what it was, focus rite I think).

    Just for the info, if I were going to stay at the level I'm at (I'll list it quick) In other words, if you guys were in my situation, and you were only going to replace the converter, what would you replace the Mbox Pro with?

    08 Mac Pro Xeon x 2 (3.0's w/no oc!) The ram is a hack job I just don't have the funds to fix it yet, but it's 12gb now. Vid card 2, but I don't remember model #'s from 8 years ago (actually, I think it's a Radeon double up)
    2011 Macbook Pro (maxed out)
    Mbox Pro 3rd gen
    Roland TD-11's
    Tacoma DB-20
    Yamaha HS-8's
    A Bluebird mic.
    A scratch TT...with no mixer yet lol.

    I ask because, while I would love to play in a room like yours Ak, it's not realistic for me (illness issues). It's highly unlikely that I'm ever going to own a setup with so much hardware. So the hardware I DO have, I'm wondering what you guys would replace it with?Not out of necessity, it's fine for now. I'm curious what you guys would hypothetically replace it with.

    I should have time to setup the VSL today, hopefully.
     
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