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Routing analog & digital audio between two DAWS

Member for

12 years 2 months
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

Member for

19 years 2 months

Kurt Foster Tue, 11/01/2016 - 15:00
kmetal, post: 442892, member: 37533 wrote: 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 names made me a rockstar engineer.

+1(y)

Member for

12 years 2 months

kmetal Sat, 09/10/2016 - 01:24
audiokid, post: 441179, member: 1 wrote: Am I understanding you correctly, that you are very interested in reampling. Thus why you like the idea of analog mixing?

I'm 'over' analog mixing unless I've got the whole she bang. I'm super into digital these days particularly w recall, and the plethora of delivery formats. I want the ability to cater the mix to different formats like sound cloud screw tube guy tunes Ect. cd quality has somehow become obselete. I'm interested also in super high sample rate as a final delivery, since computers and devices can just play them for the most part, although I have to verify that.

Re amping in the analog realm is gonna be done by a radial jd8 I belive is the model. It's a purpose built re amp box that splits out to 6 amps and has two instrument inputs.

That said I'm just as interested in the amp sim plugins some of which sound fantastic. That's why I'm using a dedicated computer/interface for it so I can run in standalone mode and have low latency reguardless of what the tracking daw buffers are at.

Thanks to vsl pro player I can use a whole other computer to run my vsti's and effects. It essentially turns an additional computer into a huge dsp processor/instrument. With the option to add no additional latency or buffers. This helps keep the tracking daw running within a responsive buffer and reliably. Vsl pro player was the biggest pleasent surprise so far this year, I thought I was just buying a cool sample library. Turns out it allows you to runs plugins and pipe audio via Ethernet. Eliminating things like propreiteru pluggin formats (ehem AAX)

I just 'thought' I needed some sort of analog mixer between my tracking/mix daw and my capture daw. Ideally the thing has no knobs or anything. I want a 'straight wire' approach to the summing thing.

I won't have budget for summing/master outboard for at least 3 years.

Being in surround I'll probably avoid this all together due to high channel counts (7.1)

I'm actually frustrated with lack of recall of old sessions and plugins so I'm going the opposite of analog mixing. Essentially a fully printed unedited final region. I want to be able to bring the tracks/stems into any mixer or daw and set the faders at unity and have my mix.

I am just missing a piece between the 8ch interface and the 2ch

No monitor controller yet either, another area where 7.1 is killing me. I basically need two, or just get used to 7.1 and the monitor controllers 2ch sum.

Right now I'm gonna just switch via the interfaces remote app and plug two diff steroe sets into each.


audiokid, post: 441180, member: 1 wrote: 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.

The Orion I was looking at is thunderbolt. It's latency spec is killer, I am concerned w overall stability both sonically and tech wise.


Again w such high channel counts via multiple 7.1 references, and chamber mics/re amp type things for natural ambiance and re amping vsti and all that the Orion is the only one that came close to real world for conversion as far as price vs quality. Burly would be well over 10k and aurora around 8k.

Tracking is mostly for fun or add ons as far as actually micing things up. Most will be virtual. My main concern is the da for mixing.

audiokid, post: 441181, member: 1 wrote: Internal clocking is the only way to go.

Never done it any other way, not planning to. I'm moving towards a 3-5 PC setup by the time I have dedicated orchestra and drum pcs, so clocking to the tracking daw is how it's done in that case.

Member for

5 years 1 month

Brother Junk Wed, 10/26/2016 - 07:30
audiokid, post: 442612, member: 1 wrote: 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

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.

Member for

8 years 9 months

DonnyThompson Sat, 09/10/2016 - 05:56
kmetal, post: 441185, member: 37533 wrote: Thanks to vsl pro player I can use a whole other computer to run my vsti's and effects. It essentially turns an additional computer into a huge dsp processor/instrument. With the option to add no additional latency or buffers. This helps keep the tracking daw running within a responsive buffer and reliably. Vsl pro player was the biggest pleasent surprise so far this year, I thought I was just buying a cool sample library. Turns out it allows you to runs plugins and pipe audio via Ethernet. Eliminating things like propreiteru pluggin formats (ehem AAX)
Wow.... That's a cool workflow Kyle. So... if I've got this straight, you're saying that with this VSL program you mentioned, and using an Ethernet connection between 2 PC's, you can use one PC as your production platform and the other PC as the DSP host?

Member for

12 years 2 months

kmetal Sat, 09/10/2016 - 13:09
DonnyThompson, post: 441191, member: 46114 wrote: Wow.... That's a cool workflow Kyle. So... if I've got this straight, you're saying that with this VSL program you mentioned, and using an Ethernet connection between 2 PC's, you can use one PC as your production platform and the other PC as the DSP host?

Yeah if I understand it correctly that's how it can work. Vsl player is basically like a host for instrument tracks. Where you can load vsti and use the effects inserts. The huge part is they allow you to set busses and interface as the I/o for the track. So you can send audio through that vsl channels inserts. I could be wrong but that's the way it seems in the descriptions and videos.

This really is super amazing! They push they're 'Mir' reverb pretty heavily on their site, and I pretty much ignored it as extra until recently. I think it might end up my go to reverb. It's a pretty advanced convolution reverb where they took IR from various locations in real places and basically let you move your instruments around on a virtual soundstage. Seems like the best way into high sample rate surround reverb as far as cost vs performance.

@1:45 it gets relevant. Around 2:30 he states you can use veinna pro server to host all your plugins as a virtual effects rack.




@4:40 it kinda quickly shows how to do it. Stating you can route audio to and from your host daw/sequencer



The first couple minutes explains the concept. It even allows you to have both Mac and PC running together! The rest just demonstrates how to run it.

Member for

5 years 1 month

Brother Junk Tue, 10/18/2016 - 05:29
kmetal, post: 442109, member: 37533 wrote: Lol this was $10 at the store the other day. I was an hour away from so, I gambled, figuring I was in reality purchasing a cheeseball effect.

Suprisingly it's pretty cool, to my ears has some decent sound to it. It's a small software company, but I'm happy w the purchase. Not center stage quality, but I think useful enough to merit. I'm finding w amp sims it like one has a good marshall, the other a good whatever, the other a good 5150. It seems like there's only a couple good solid amps from each company's bundles. Anyway worth checking out if your bored.



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.

Member for

21 years

audiokid Wed, 10/26/2016 - 10:20
I'll chime in on a few of your questions.

Brother Junk, post: 442632, member: 49944 wrote: Or do you do all sessions in 24/48 and just bounce down? (Wouldn't that be SRC?)
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.

Brother Junk, post: 442632, member: 49944 wrote: 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?
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.

Brother Junk, post: 442632, member: 49944 wrote: 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?
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.

Member for

4 years 8 months

JayTee4303 Mon, 12/26/2016 - 07:18
What a great thread! Every once in a while, other forums will touch on some of what's here, but the response is mostly "huh wut?", so there's not much back and forth on these topics. Four room facility here, 11 or 12 PCs last time I counted, we use up to five in "DAW-farm" configurations. Usually not all at once, it just depends on what's going on. Control room hosts a pair of I7 4790s or 4970s, memory is the first thing to go. One video, one audio.

Audio utilizes a MOTU PCIe 424, with three 2408s and an HD-192. Not exactly Myteks, but the massive ADAT routing capability serves some very useful purposes, and we get 120 dB dynamic range, in and out, from 10 channels on the HD-192, plus AES which is near permanently populated with the Bricasti M7. Video has an M-Audio C600, just for monitoring, we pull a SPDIF stereo feed out to the Audio box and monitor from there. We will move into X.1 audio for video, if and when revenue supports it, but haven't looked into better audio so far. Instead audio arrives via file transfer, after mixdown and mastering.

On networking, in addition to QOS concerns with Dante and AVB, make sure you create a non-blocking backplane architecture with your routers and switches. 8 or 24 Gigabit ports seems great, but if your 24 port Gigabit switch or router's backplane can't handle 48 (full-duplex) GB datastreams at once, you might as well be running Fast Ethernet. Or worse...somewhere around 55-60% saturation, ACKs and RESENDs will bury your network anyway. Ethernet, like H20, is an incompressible medium!

We use a flexible networking schema, to keep core computers unexposed to the net, while maintaining update capability when necessary, with internetworking always available. Basically, frequent net connected boxes (we call them DMZ machines. not strictly accurate, but illustrative)r un DHCP, from a router/server, with IP addys limited to the x.y.z.3-100 range. Dot one is the gateway of last resort, obviously, with dot-2 and dot-3 reserved for wireless access points. Core machines, mostly isolated from the net, run hard coded IPs, on the exact same x.y.z. schema, but limited to dot 101 thru dot 255. Anytime a single cat-5 cable connects the two networks, the IP schemas seamlessly merge, or, well....they COULD if we put in the necessary hours on the router table codings, hey...it's on the list, we'll get to it! In practice we usually have to do some pinging, from a shell window, to get everybody on the same page, after merging the Core and DMZ networks.

For digital routing, we use a Z-Systems Optipatch Plus. 30 Toslink ports fully leverage the 2408's and 424's significant ADAT I/O. One sample thruput latency, and it reclocks the signal, extending Toslink's inherent 6 meter limitation w ADAT, to 12 meters, over plastic fiber. We are beginning to find the limits of the device as all 30 ports are now populated, but each of the 2408's triple ADAT ports , offer "sidechain" ADAT routings for dedicated processing pipes, where we don't need the Optipatch's routing flexibility.

The Tracking Room DAW, originally the sole computer, is an older I7 2600, with a Profire 2626 gateway to 8 analog and dual ADAT ports. The whole room is set up to be operated by an engineer/artist, standing up, wearing a guitar or bass. Good for when a couple other players are over and there's no designated engineer. For the most part these days, it handles VSTi hosting for guitar or keys, or both, once we get options pared down to choices.

The Live Room DAW is an older Core Duo, under very light duty, simply passing audio thru it's pair of 2408s. There's a subsystem for each major instrument, which we streamlined for "right now" composing work, with patchbay options to simply swap out our gear for client gear, while keeping the facility routing setups intact. Our V-Drum kit comes in via a "sidechain" (not thru the Optipatch) ADAT pipe, from an Octapre Dynamic. Acoustic drums use the patchbay to access the Octapre. The Core Duo is also where we capture MIDI from the V-Drums or triggers, one of the the usual times we'll sync up multiple DAWS. We've long seen the utility in being able to do drum replacement right here, or even drum VSTi hosting, and since the Core Duo doesn't begin to cut it for either app, it is next on the block for replacement. Maybe a mid grade i7, direct swap, but we are beginning to need more expansion slots on the video box in the CR, so we might go dual Xeon/Titan there, and move the current vid box to the LR, we'll see.

There's a Core Duo in the vox booth, w ADAT connectivity, but we house our better pres in there too, so we backed that up with an XLR snake into the CR, for operations with zero fan noise. The idea is that songwriters can write in iso there, then pipe offerings out system-wide, but it hasn't found much use to date. The rest of the PCs are single cores, or i5 laptops, the singles are support, (internet, playback, realtime outboard control via MIDI and MIDI synth programming), and the i5 laptops are used with the live rig.

Long enough for an initial "hey!' in this great thread, you can check out pix and more info on our webpage: www.e4mm.com .

Ok, one more thing, a question, sort of... The Bricasti lives on AES in the CR. We are firm believers in the un-synced dual DAW approach, stemming back to the legendary SOS article "Does Your System Need A Master Clock?" Master clock is Master Clock, while Chase Sync is always chasing. Reaching into a gray area here, bear with me. Using the M7 as a hardware insert, it loses a LOT of its brilliance. Brilliance heard while monitoring. We don't get that brilliance back unless we tap the M7s AES outs, as mono inputs, re-recorded live into Sonar. One more time, because I know I'm not explaining well.

Vox, in the CR, cans on, I monitor the mic>pre>HD-192>M7 via DSP. My cans hear the AES input from the M7, and whatev I'm playing back from Sonar. Sounds Bricasti Brilliant.

Recorded vox, run thru AES to the Bri and back into Sonar as a hardware insert, NOT rendered, sounds dull and lifeless.

Recorded vox, out thru AES into the Bri, re-recorded onto two mono tracks, hardpanned L and R, the brilliance is back.

What gives? Is this a latency comp issue? Or is it Chase Sync chasing? The Bri processes at 96, according to Casey, and being a video houise we run 24/48.

Or is is a combination of these, or something else entirely?

Hard facts, AND wild speculation appreciated.

:-)

Member for

12 years 2 months

kmetal Tue, 10/18/2016 - 13:20
Brother Junk, post: 442327, member: 49944 wrote: 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.

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.

Member for

12 years 2 months

kmetal Sun, 10/09/2016 - 21:10
audiokid, post: 442004, member: 1 wrote: Passing audio between two DAW's is an advanced workflow. If you think one DAW is cool, two is dope. :love:

Lol I think I'll be up to 3 daws and 3-6 computers.

Samplitude has super high track counts at 192 so, and amazing editing/processing, I'm gonna write/edit in that, then pipe or drag and drop into protools which plays nice w avid media composer for mixing AV, and decoupled summing into Sam or audacity. Gonna end up w the eleven rack for virtual guitar, and a few vsti computers.

Really tho the summing stays the same. The rest is just Dsp computers.

I scored THE MOTHERLOAD of exceptionally good sounding vintage/modernish/classic synth vsti from UVI.

They samples all the originals and then mastered them at sterling sound, at least some of them list sterling, but they have that sterling polish for the most part. Sterling masters have grabbed my ears without me knowing at least 10 times in my life. There's something special about that mastering house.

Anyway GC had this for $100. I thought it was 200. And it sells for 500 list. Really deal aside I think it's very good. Particularly the 80's style synths are spot on record quality. I've been wanting to compose a few 80's jams lately.

I highly reccomend anyone who's interested in synths check this company out, and right now if yor near a GC grab this bundle on clearence.

Stoked.! My software set is about 3 programs away from being 'complete' enough for a starting foundation w no missing links and all impressive imho sounding stuff..... It's a large collection(s) but I feel like 70% of it is usable. There's doesn't seem to be a whole lot of filler.

Have a listen to all the audio samples on the page while your browsing.

http://www.uvi.net/en/vintage-corner/vintage-vault.html

Vintage vault- $500, $100 on clearence 'best of' playlist

[MEDIA=soundcloud]uvi-official/sets/vintage-vault-bestof

Lol 4 hours of sample tracks from the vintage vault.

[MEDIA=soundcloud]uvi-official/sets/vintage-vault

Legacy synths- $50, $20 on clearence. This has some of the more modern stuff on it. It's a GC exclusive so when it's gone it's gone, UVI still supports it, but it's been removed from there active products page.


[MEDIA=soundcloud]uvi-official/sets/synth-legacy


Those are just demo songs!!! I think this collection smokes, but I'm kinda new to the synth world.... So feel free to correct me and point me elsewhere.

Member for

5 years 1 month

Brother Junk Thu, 10/27/2016 - 06:22
audiokid, post: 442646, member: 1 wrote: If you are using prosumer gear.... I would suspect you are better off tracking at its optimal SR.
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.

audiokid, post: 442646, member: 1 wrote: 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.
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 $$$$$$$

audiokid, post: 442646, member: 1 wrote: Good converters should sound excellent at 44.1. The cheaper ones will not.
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.

Member for

5 years 1 month

Brother Junk Mon, 10/10/2016 - 11:42
kmetal, post: 442043, member: 37533 wrote: Lol I think I'll be up to 3 daws and 3-6 computers.
When you guys say that, do you mean (for example) PT x3? (fwiw, I don't think that's what guys mean)

So, if not...what is the benefit of using 3 different daws? You noted Samplitude has a high track count at 192khz...so why not just use Samplitude x3?

I'm not saying you should use the same daws...I'm wondering why one chooses not to?

Member for

21 years

audiokid Mon, 10/10/2016 - 12:29
Brother Junk, post: 442053, member: 49944 wrote: When you guys say that, do you mean (for example) PT x3? (fwiw, I don't think that's what guys mean)

So, if not...what is the benefit of using 3 different daws? You noted Samplitude has a high track count at 192khz...so why not just use Samplitude x3?

I'm not saying you should use the same daws...I'm wondering why one chooses not to?
3 DAW's seem a bit over the top but I will never say never! Being said though, the 3rd DAW could without doubt be used for VSTi support. I've though about this but in my particular workflow I don't use VSTi as much. (currently) I prefer to go outboard via keyboards using a Kronos X, Nord Lead4 and the MPC Renaissance for the drum programming.

I choose to use Sequoia for both DAW's, because I have that luxury and take this to the max. But, I also have AbletonLive in both DAW's as well. Why? Because I also like it and may want to do particular tasks that Ableton excels in.

I also may upgrade a basic license for Pro Tools but for nothing more than being able to import the exact tracks as what clients give me. It would be business reason only.
Pro Tools has nothing on Sequoia but because it is accepted as the industry standard DAW for professional studios, it can be an asset to say, just send me the PT files.

The common approach for me is to dedicate my mastering applications to DAW2. DAW2 replaces the 2 bus section of DAW 1. Remember that.
DAW 1, (2 bus > mastering section) is completely disabled. DAW1 is all about tracking and mixing down.
DAW2 is all about capturing the session and preparing it for upload etc. It is also what I use to compare a more detailed cause and effect of everything I do when it comes to learning.

The two together should ideally be uncoupled and have the ability to become an advanced learning platform where you can network files between each other.
Also note: for my particular approach, a dedicated monitoring system should also be able to switch independently from post to pre of both DAW's including online monitoring.

Multiple ways to monitor is a vital part of my workflow. Without that, I would be back to guessing and may well forget about the 2DAW approach. Relying on a basic monitor section that can only view DAW 1 for example, is like parallel parking a school bus without windows or side mirrors.
The more advanced monitor controller is like having a camera at the door way, inside, back of the bus and a drone outside.

Member for

11 years 7 months

bouldersound Thu, 10/27/2016 - 10:50
audiokid, post: 442099, member: 1 wrote: I can't believe how cheap this stuff is now. When I was looking at Avid Composer, I think the whole thing was like $20,000.00!
I must be missing something here.
This stuff is under $100.00 :unsure:

@bouldersound Keep in mind, Magix bought Sony's software yes? Boulder, you know much about the video side of this conversation?

I know a little about the video side. I've been using Sony's software since before Sony bought it from Sonic Foundry, starting with CD Architect (version 2, I think) and Sound Forge 4.5, then getting into Video Factory, Vegas Video 3, Vegas 6 and now Vegas Pro 13. I've been recording and mixing multitrack audio since Vegas 3. I think this all started for me around 2000, 2001. I'm waiting to hear what Magix does with Vegas.

Member for

12 years 2 months

kmetal Thu, 10/27/2016 - 10:52
audiokid, post: 442612, member: 1 wrote: 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

Strangely the engineer did this on the new Coldplay record according to his 'inside track' interview in SOS. Maybe last marchs issue I think. I recently read it and was kinda surprised. Both that he had a capture rig, and more so he used it to capture at a higher rate. lol I'm like you got it backwards bro... but that's why he gets the big bucks. He addressed it in he article saying he'd used a higher sample rate to track/mix but the computer can't handle that along w all the plugins.

He captured thru Burl and cranesong DA not sure about the capture ad.

It's difficult for me to cryitisize someone working on that level, but really, between the backwards capture and the million and a half plugins, his method seemed kinda lame. Especially considering all the high end gear he had to track and mix with.

All that other stuff seems like bullocks to me..

Brother Junk, post: 442632, member: 49944 wrote: 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?

Buffer underuns are a common plague in PT, particularly the older versions like 7,8 and 9 to a lesser extent. They re wrote the newer versions to better support native (non dsp) computers. It's still the most common error messsage in PT and often un justified. PT doesn't like any sort of changes to its settings for some reason. It's a good idea to restart the computer when you change any settings with buffers or sample rates particularly in PT.

24/96 shouldn't overload that level of computer even tho it's getting on in years. I would throw a shot in the dark and say you should get at least 24trks at that sample rate without error, especially at that high a buffer.

I'd try it in one of your other DAW programs and see if you have a similar result. PT has its good merits but rock solid relability, and CPU efficiency are it.

If your other daw(a) react the same way then it's something to do w the system.

Your running two HDD right? One for you programs and one just to record audio too?

Brother Junk, post: 442632, member: 49944 wrote: Or do you do all sessions in 24/48 and just bounce down? (Wouldn't that be SRC?)

It's being in 24 bit that makes the most audible differnce.

If the destination is cd then many will track at 24/44.1. I've done most of my work at 24/44.1 for various reasons. I've done a little bit at 24/96.

It's basically a good idea to use 24 bit word length, that affects some important things like gain staging and headroom.

Once your in 24 bit. The sample rate makes far less of an audible differnce.

Like Audiokid said not all converters sound better at higher sample rates. Most comverters in general were designed w 96k in mind (even if they go higher) and that's usually what their specs are published on. Prosumer stuff is undoubtably designed w 44.1/96 in mind even if they go higher. It's unlikely someone using a budget interface would have the other gear necessary to take advantage of ultra high sample rates.

As long as your 24 bit you should be doing just fine at any given sample rate like 44.1.

Member for

12 years 2 months

kmetal Thu, 10/27/2016 - 10:56
Brother Junk, post: 442674, member: 49944 wrote: 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'd replace that mbox with a focusrite Scarlett or preferably and RME babyface PRO. RME is affolradle professional walking the prosumer line on the side of pro. MOTU sits in between focusrite Scarlett and RME. MOTU is prosumer at its best, and is suitable for modest professional work.

Member for

12 years 2 months

kmetal Tue, 10/11/2016 - 15:46
Brother Junk, post: 442053, member: 49944 wrote: When you guys say that, do you mean (for example) PT x3? (fwiw, I don't think that's what guys mean)

So, if not...what is the benefit of using 3 different daws? You noted Samplitude has a high track count at 192khz...so why not just use Samplitude x3?

I'm not saying you should use the same daws...I'm wondering why one chooses not to?

Again we are talking seperate things, Chris explained his setup.

I want to use both mainly becuase PTHD does 7.1 and plays nice w avid media composer. Sam only does 5.1, and my license is for pro x until I grab the upgrade. It's reasonably priced but I've spent my alloted money on my software set for now, and just want to get going learning Sam before I upgrade.

My reason for Sam is its coding is clean, and it's editing is amazing, it's track count is high.

So I'll be composing in Sam w tons of tracks and vsti loaded, editing it all, then piping into PT for mixing/video integration, then once my mixes are set I'll be capturing those back to Samplitude.

If I find the PT thing not worth while I'll stick w Sam the whole way. But it does play nice w avid media composer so that's a big plus.

What I decide depends on audio quality and how each daw handles the computer resources I'm able to give it.

The reason I'm using a dedicated recording daw/session (not necessarily cpu at least from the get go) is becuase I want an extensive amount of vsti, amp sim, and effex readily available in standby. This is heavy on the cpu. Also having a lot of plugins and stuff active degrades the audio you hear back from the machine. So I want to compose and edit without load times and menu shifting, then I'll export the audio/midi (possibly) to a nice clean session that's got a much more simplified template setup for mixing. This eases cpu usage and doesn't clog the audio busses and reduces that phasey degradation from tons of plugins active.

I'm basically gonna compose and pre mix/edit in one session/daw, mix in the next clean one, and master in the last. This could be done on one or two machines, or more.

The main reason for multiple computers besides the nessary 2 for decoupled mix capture, is strictly for realtime performance of soft synths and amp sims.

The networked computers (vsl player / Ethernet) are at the mercy of the the host buffer sizes, so I want the host cpu to have as few instrument tracks and plugins as possible. Leaving all the processing to the vsti slaves computers, which are at low buffers, and at the mercy of its hardware.

The decoupled guitar amp sim computer is so I can run amp sims in realtime reguardless of how my buffers are in the daw. I'll pipe Audio Out of the amp sim cpu interface, into the daw, and monitor the amp sim in realtime via its dedicated interface. I'll play along to the daw and backing tracks via its own interface and speakers. That way everything is in sync, w zero latency, reguardless of buffer sizes in the daw. Also since I'll have a rack unit amp simulator, and various amp sims, only Guitar Rig runs at 192k, amplion, and gtr3, and amplitube, run at 96k. So I need to pipe them out analog and convert at the daw interface. Becuase my sessions will be 192.

Becuase the amp sim has its own decoupled interface and PC, I can leave the buffers at realtime, and hear my guitar in realtime w minimal latency. My backing tracks can have whatever buffers they need via the daw, it won't matter because I'll be playing to the daw tracks, to things will sync up. I just don't want latency thru my guitar, which is what I'd get if I simply loaded an amp sim pluggin in the daw. It would be effected by the buffer sizes which vary at different points in the project. I mute the guitar tracks I'm recording into the daw, and listen only thru the amp sim PC/speakers, this avoids latency from the daw. I hear my backing tracks via the daw PC/speakers at the same time. Latency isn't an issue, my and tracks sync, becuase I'm playing to the buffered tracks. So while stop and play buttons may have lag, they'll be no lining things up later due to bad sync, and good feel due to low latency.

I want to have free reign for last minute overdubs deep into a mix. Provided my daw can handle recording an audio track at even the largest buffer, I'll be good.

This all took a ton of thought and planning, and I'm really just testing the notion with the new setup. In theory it should work as described (unless I missed something).

The dedicated amp sim computer is no different than having your amp in the room, and playing along/recording to your daw, w the guitar track muted in the daw. So your guitar is coming thru the amp, the rest is coming thru your studio monitors.

It's when you get into higher buffer sizes like 512 that monitoring amp sims thru the daw mixer/speakers looses proper feel for me. I'm just taking it to the next level, by dedicating a PC and interface to virtual guitars. I sold all but one of my amps, and I love how good the new amp sims are sounding. I'll grab a couple real amps over time. But w instant recall and realtime performance in standalone modes amp sims have their positives.

Obviously there's still some latency and that depends on the interface and connetion type. But the RME interface I'm looking at is somewhere around 3ms if I recall so that's quick. Like I said I don't find latency unbearable till buffered of 512, which was somewhere around what 15ms or more on my old FireWire interface?

audiokid, post: 442054, member: 1 wrote: 3 DAW's seem a bit over the top but I will never say never

I don't operate any other way. Lol. I think in extremes. The dedicated vsti computers I think will be worthwhile, and having a dedicated video editing computer seems like a must if I plan on top notch (4/8k) video and audio.

The reason I subscribe to divide and conquer w the computers is becuase I think you get more done, vs one super computer. Like 4K for a new Mac or 4 i7 pcs w 32-64gb of ram cost about the same money give or take. I reckon I can do more w 4 pcs than one Mac Pro.

Also it allows me to stagger them so as they become obselete it's not a complete replacement. Every couple years I can move demote one to lesser duties. Or have it lets me keep one for longer that's doing something super basic like hosting a standalone guitar amp sim. I should get about 8 years out of even a dual core cpu in that role.

This multi cpu system is really an experiment to see what's best, and where the points of diminishing returns are.

So far I've only got two. A laptop for graphic design and office stuff, and a desktop for audio/video.

Next is a humble one for realtime guitar, or a i7 10 core or Xeon level for AV daw. At which point my current i5 gets moved to vsti.

Since everything is soft synths, I'd like to keep buffers low at all times.

I also want to follow the prince mentality and have the bulk of my stuf loaded and a simple 'track enable' away. I don't wanna be menu fishing.

This new setup isn't the ultimate in fidelity, but based around instant creativity. The fidelity will come in time.

audiokid, post: 442054, member: 1 wrote: Pro Tools has nothing on Sequoia but because it is accepted as the industry standard DAW for professional studios, it can be an asset to say, just send me the PT files.

The only feature PTHD has on sequioa is the ability to load 64 video tracks vs 1. It also can sync directly to avid media composer (via avid sync hardware :( )

Other than that sequoia absolutely smokes the others. My whole new rig was designed around 192k capability and Sam/seq are already supporting 384. Don't you know that drives my obsessiveness wild lol!

If I didn't spend(ing)so much money on my basic instruments/vsti/interfacing at this point, I would have just purchased sequioa even tho I got tremendous deals I'm still about 900$ in on drumagog and my synths and BFD.

PT is included in my (future) amp sim rack unit along w the ilok, the whole kit is 500. From there I'll use pt and Sam, and then decide on whether the PTHD license makes sense 1500, from certified retailer (vs 2500 from avid), or if I'll save and grab sequoia. There's a good chance I may end up w both, depending on how much video work I seem to be doing. Media composer is an amazing program.

Like you Chris the PT thing for me is partly to be up w 'industry standard'. The studios Disney have pt for a long time it it was annoying to say 'well we use DP...'

Frankly I've used almost all the DaW's out there and they're all pretty good.

I have old files from pt I need to move, so I'm gonna grab it while I can get the deal. If it's still obnoxious/unreliable like the old 7x version watever. If not I'll likely keep on with it. Partly.

audiokid, post: 442054, member: 1 wrote: Multiple ways to monitor is a vital part of my workflow. Without that, I would be back to guessing and may well forget about the 2DAW approach.

I think this is THE most overlooked part of you/boz's decoupled system.

Your keyboard collection is cool man! I'm going the software route to save space and have a larger variety of sounds available than I could afford otherwise.

My buddy has like 30 classic synths, and the hardware does sound richer.... But there's always compromises and vsti is it for me for now.