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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 Pro Tools 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-recording 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.


Brother Junk Sat, 10/08/2016 - 06:45
kmetal, post: 441987, member: 37533 wrote: 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,

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.
Wow, I think you answered just about every question I had/could have about it.

kmetal, post: 441987, member: 37533 wrote: It is decoupled, meaning the two computers are running seperately, as opposed to master/slave where there running as one big computer.
I understand it now (at least to a point). That's why AK's setup needs two converters and the summing board. It's two, totally separated systems that meet at board. And the VSL master/slave thing is obvious now.
kmetal, post: 441987, member: 37533 wrote: These are two seperate concepts BJ.
I never thought about what my shorthand name would be when I chose it lol.
kmetal, post: 441987, member: 37533 wrote: and to avoid SRC in the box.
I don't remember reading why that is a bad thing, but I will search SRC ITB. I know it's probably due to errant conversion, or at least inferior, but not sure why yet. It doesn't really matter bc the price is way too much for me atm.

Thanks K, AK and everyone, I know it wasn't my thread, but I feel like I have a rudimentary understanding of the hybrid systems, and I feel like I understand the VSL setup.

The VSL setup I could afford and benefit from very soon. I find it remarkable that you can use a windows machine and a mac together for it. Although I haven't kept up with the new OS's and their abilities so maybe it's old hat now.

audiokid Fri, 09/09/2016 - 18:10
You are going to have so much fun once you get this all sorted and tuned up for what you need.
I would say, go either way.
I wouldn't bother with Pro Tools at all but if you are getting tracks with plugs already part of their mix, stems etc... then I would indeed have PT ready to use. That being the way it evolves, I would then use Samplitude as your mastering DAW. But, if you ever build a system that functions like mine, you can move audio all over the place and use either DAW to do what the other just did. Neither are one or the other. They are both fully capable to be a tracker, summer, master system. And that's when it all gets amazing.

kmetal Fri, 09/09/2016 - 18:25
Yea man I'm very excited. PT is in consideration becuase of its video functionality. That's the absolute main reason. It's a big expense for the functionality. The alternative would be media composer for the same price but with Hollywood level video stuff.

Having 2 high class audio programs kinda seems overkill or a lot of investment for those 10 video tracks.

Not a matter of saving money, rather not buying redundant stuff for no reason.

What are the key(s) to your rig that allow such ease of moving audio around? I decided as soon as a heard your rig years ago you were on to something. It's just finally time for me to take the plunge...

audiokid Fri, 09/09/2016 - 18:35
kmetal, post: 441174, member: 37533 wrote: I decided as soon as a heard your rig years ago you were on to something. It's just finally time for me to take the plunge...
Thanks. It is cutting edge, and so much fun too. But so is ITB. And Samplitude is amazing just on its own.

kmetal, post: 441174, member: 37533 wrote: What are the key(s) to your rig that allow such ease of moving audio around?
Being uncoupled and a Cat5 cable lol. Having a 3 way monitor system and 2 independent converters. Its so simple, yet so fluid and logical.
The monitor controller is the key component because it allows me to listen to each or all area's.
Having digital patch-bays allows me to switch gear on the fly as well. I never need to unplug things. That being said, I now do not need gear anyway. Samplitude has everything but a Bricasti. I sold all my consoles now. They are all dated technology imho. Two DAW's makes the study and manipulation of audio, easy. Being able to hear what you do in all sections, make it possible to work this way. Uncoupling un ties your hands.

kmetal Sat, 10/08/2016 - 13:53
as far as the SRC I Don't have an awsome answer. I think the key w the two daw system is you sum and SRC at the same time. Basically anytime you eliminate a step of processing in audio it leads to less degradation.

I scooped vsl a few months ago, I haven't worked w it yet since my new setup is coming together gradually. They just came out w version 6 so now is a good time to buy. I'm gonna wait to upgrade to get some money's worth out out my version 5.

The differnce is some navigation improvements to the gui, and they streamlined things more so you should see even better performance as far as cpu resources.

I can't belive more people haven't caught on to the power of the vsl player as an effects/vsti host.

I didn't even know it did that until after I purchased it purely for its sounds.

In addition to the mac/win bridging, you can also run Vst plugins in protools which is cool. It really opens a lot of doors.

audiokid Fri, 09/09/2016 - 18:47
kmetal, post: 441177, member: 37533 wrote: Is it becuase it's all MADI?
I choose PCIe MADI interfacing because it was the fastest and most stable interfacing for the Orion 32. Before that, I used AES EBU with RME PCIe AES32 . That is excellent but it wasn't an option for the Orion. I used better converters than the Orion but I needed 24 channels and to get another 8 channels it would have cost me another $10,000 so I dumped my 16 channel AES system for the Orion 32. Its good enough.

I am very serious about composing and sound design too. I am also a two channel at a time composer so my interfacing and clock has to be dead locked. I require an interface that makes overdubbing flawless. PCIe MADI and the Orion works. USB sucks. Plain and simple.
FW also sucks. so Madi or AES is really the only option for PC based DAW systems.

audiokid Fri, 09/09/2016 - 18:53
PCIe interfacing make mixing 32 AD DA tracks go flawless. PCIe with either a Madi or AES EBU interface has been the only way I could get there without compromising my workflow.
Under 8 channels, FW or USB seems fine.

Internal clocking is the only way to go. External clocks like the $6000 10M ending up looking like a patch for the unaware guys using rats nests. I always have better results without external clocks and a PCIe interface. Plain and simple.

kmetal Tue, 11/01/2016 - 14:43
DonnyThompson, post: 442754, member: 46114 wrote: Again, I'm not being rhetorical... my question(s) really are sincere; I'd like to know what my peers think... ;)

I don't blame the engineer as much as the bands. Bands just aren't as committed or tour seasoned many times. Look at the SOS article. The engineer was cobbling songs together from the singers iPhone voice memo recordings.

The singer lived in California the band mates and studio in England.

Now granted it's Coldplay and they're talented like em or not, but still.

I blame the bands for much of it. Laziness selfishness whatever.

All I know is I engineered dozens of bands in the same room w the same gear and house drums, similar setups. When the pros were playing I was a better engineer. The results were faster, easier, and better sounding. In short the better manes made me a rockstar engineer.

Literally same drum kit. All the eq and compression in the world couldn't do what a good drummer did to the kit.

Brother Junk, post: 442756, member: 49944 wrote: I wonder why some studio's only use externals? Or maybe they are connecting it with Sata 3....I never examined them from the back.

If they're 'only' using externals it's either a mistake on their part, they're using it for backup, or possibly samples.

Internal drives are better for the purpose of audio. W SSD now, Sata 3 is 'adequate, or slow' depending on things. That's both by the specs and in reality.

Brother Junk, post: 442756, member: 49944 wrote: To be completely honest, I still don't fully comprehend how to use compressors. I mean, I can make it work for me, but I've seen people who hear a track, take a second, and then set knee, ratio, threshold etc....just bam, bam, bam.

Compression is arguably the most difficult thing in audio engineering to understand. Took me over ten years to really know what I was doing.

Part of it is because compression isn't something in general that's obvious to hear. Most compression is done to be fairly transparent. Eq you hear, 3db of gain reduction you probably won't. In other words more often than not good compression technique is synonymous with subtly.

I highly reccomend you grab the 'mixing engineers handbook' asap. It's an excellent comprehensive, easy to follow book. It has step by step techniques for eq and compression and tips for helping your ears hear thes ebtings better. Plus great tips for mixing common instruments. Again w some quick steps.

It's a great read and I kept it next to the console for years when I started working at a pro studio.

And the 'recording engineers handbook' if your tracking live instruments in any way.

Brother Junk, post: 442756, member: 49944 wrote: Just out of curiosity, have any of you ever compared hw to the plug-in that imitates it? If so, what did you find?

I've used a hardware urie 1176 and many software versions. There not really close. There similar in a sense that jeans or jeans or t shirts are t shirts not sweatshirts.

But with compression in particular there is an element of live interaction w the performance that the hardware has. Also the way analog over drives is much more pleasing/different than the pluggin versions. They're similar in basic tendency like say 'punchy' compressor, but tone and everything else there not really that close.

I've also compared the API eq w the waves version, and again not really similar in sound. Digital artifacts aside, the pluggin was far more exhaggerated than the hardware. Much more of an audible effect. I think a lot of emulations do this, they exhaggerate the tendency of the piece they're emulating.

It's not fair to expect the same setting to sound the same when comparing hardware to software, but trying my best to match Similar levels subjectivly of boost and cut both the pluggin and hardware sounded unique to themselves.

Boswell, post: 442759, member: 29034 wrote: Gigabit ethernet gets its name from the propagation rate of the measured unit. "Giga" = 10^9 and "bit" = bit, not byte. So the rate on the ethernet cable is 10^9 bits/sec or 1,000,000,000 b/s. This corresponds to 125,000,000 bytes/sec or 125MB/s. Note the capital B when referring to bytes and the lower case b when referring to bits.

This is the rate that the bits within a packet of information would travel. Given that there will be multiple layers of wrappers round each packet and also gaps between packets, the end-to-end data rate of the payload could well be less than half the maximum bit rate of the transmission medium.

One of the difficulties in using ethernet as a digital audio transmission medium in a multipoint network is that the underlying hardware offers no guarantee (a) of the end-to-end transmission time, (b) packets will arrive in the order in which they were sent, due to being routed on a per-packet basis, (c) a packet will arrive at all and (d) a packet will arrive uncorrupted. Because of issues (b) - (d), one of the higher protocol layers takes care that a long message can be assembled correctly from shorter packets, often involving re-transmission of lost or corrupted packets. All this bodes badly for real-time audio, but is fine for transmission of audio data files. These problems do not apply to point-to-point ethernet links where there is no other traffic.

Can I have an autographed copy of torn book!? Excellent break down.

dvdhawk, post: 442816, member: 36047 wrote: @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.

Well said. Judicious use of effects analog or pluggins is so essential.

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.

Agree for mixing. For tracking I've found dbx in particular to be quite good on some things. Either in the instrument/amp chain, or mic signal chain.

The press is eureka channel's compressor is also be very useful and is very transparent.

audiokid Sat, 10/08/2016 - 21:48
kmetal, post: 441997, member: 37533 wrote: as far as the SRC I Don't have an awesome answer. I think the key w the two daw system is you sum and SRC at the same time. Basically anytime you eliminate a step of processing in audio it leads to less degradation.
exactly, and possibly correct or enhance any loss or exaggerations that might need tweaking as you mix into the master DAW going from example 96 to 44.1 at the same time.
So in other words, mix into a final mix or master that is going from 96k to 44.1 while taking advantage of outboard gear while a conversion pass is happening, live. There is no guessing because you hear it all right then and there.

There are a lot of advantages to the two DAW system. To put it blunt, it enables broader approach to learning, sound designing and experimenting everything from cause and effect to better control of what happens between the capture, mix and the online results.

Passing audio between two DAW's is an advanced workflow. If you think one DAW is cool, two is dope. :love: