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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.

Comments

dvdhawk Wed, 10/12/2016 - 08:29
Kyle, you may not need to print DVDs, I do though.

My video clientele needs physical DVDs (and musicians still want CDs) often in quantity, so I have a Bravo on-disc inkjet printer and a simple 1:5 duplicator tower (again, they've paid for themselves over and over again). Short run duplication still makes sense at the local level. Anything above about 500 units is better sent off for full replication, but there is some money to be made doing smaller quantities. 300 units is about the tipping point where you need to weigh your options. My customers often want DVDs they can sell, or give away at trade shows, etc. and only need 50-200. If I'm packaging DVDs or CDs, I use Photoshop / InDesign to layout the artwork, and then take them to the local printing/litho house for printing and cutting, and then assemble everything here. Letting a professional do the printing is the only way it's practical for me. Not only can he print them for less than I'd spend on ink; it's a superior print, on superior heavy paper stock, and he's got the machine that can cut the whole stack at once to the 1/10,000 of an inch accuracy faster than I could hack one out with a standard paper cutter.

audiokid Wed, 10/12/2016 - 09:26
Boswell, post: 442113, member: 29034 wrote: Where I differ from Chris is that I deliberately do not perform pro-level mastering, since I am a firm believer in having another set of ears employed at the mastering level.

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)

Boswell, post: 442113, member: 29034 wrote: Chris has got the right broad picture, but I don't have a fixed mix setup - how I set it up depends on the material I am mixing
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 :)

audiokid Wed, 10/12/2016 - 10:46
(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.
class="xf-ul"> 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.

bouldersound Sat, 11/05/2016 - 08:35
kmetal, post: 442892, member: 37533 wrote: 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.

I admit to having a few channels of mundane dbx compression on my front end. But they do get bypassed fairly often. There's also a Pro VLA and a Drawmer 1960 to track through.

kmetal Wed, 10/12/2016 - 17:52
Boswell, post: 442113, member: 29034 wrote: The capture box can be another DAW, a stand-alone 2-track recorder or multiple tracks on another HD24XR, the stereo pairs differing, for example, in compression threshold.

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

Boswell, post: 442113, member: 29034 wrote: Flexibility of monitoring is key, as Chris mentioned, and I have the monitoring routing set up so I can tap into any of the points in the journey of the signal from raw tracks to finished mix

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.

Boswell, post: 442113, member: 29034 wrote: Where I differ from Chris is that I deliberately do not perform pro-level mastering, since I am a firm believer in having another set of ears employed at the mastering level.

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?

dvdhawk, post: 442114, member: 36047 wrote: Kyle, you may not need to print DVDs, I do though.

My video clientele needs physical DVDs (and musicians still want CDs) often in quantity, so I have a Bravo on-disc inkjet printer and a simple 1:5 duplicator tower (again, they've paid for themselves over and over again). Short run duplication still makes sense at the local level. Anything above about 500 units is better sent off for full replication, but there is some money to be made doing smaller quantities. 300 units is about the tipping point where you need to weigh your options. My customers often want DVDs they can sell, or give away at trade shows, etc. and only need 50-200. If I'm packaging DVDs or CDs, I use Photoshop / InDesign to layout the artwork, and then take them to the local printing/litho house for printing and cutting, and then assemble everything here. Letting a professional do the printing is the only way it's practical for me. Not only can he print them for less than I'd spend on ink; it's a superior print, on superior heavy paper stock, and he's got the machine that can cut the whole stack at once to the 1/10,000 of an inch accuracy faster than I could hack one out with a standard paper cutter.

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.

audiokid, post: 442116, member: 1 wrote: 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.

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.

audiokid, post: 442117, member: 1 wrote: (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.
class="xf-ul"> 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.

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?

audiokid Wed, 10/12/2016 - 18:36
kmetal, post: 442135, member: 37533 wrote: 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.
This is precisely what I am talking about.

kmetal, post: 442135, member: 37533 wrote: The whole digital patchbay really changes the game on how analog gear gets incorporated.
Indeed.

kmetal, post: 442135, member: 37533 wrote: 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.
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.

kmetal, post: 442135, member: 37533 wrote: Does the digital patchbay have adda? If now how is the analog gear connected? Right to I/o on your interface?
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.

DonnyThompson Sun, 10/30/2016 - 04:47
@audiokid, @Sean G, @Boswell, @kmetal , @pcrecord @bouldersound , @Brother Junk , @dvdhawk , @Davedog

kmetal, post: 442682, member: 37533 wrote: 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.

Like Kyle, this is where I also get confused.

I'm certainly not against using plugs; I think most here would consider me to be an advocate for their use - if they are needed - and in that regard, if they accent your mixing style in a positive way, and make your mixes sound better than what you'd get without them, then they can be useful tools. After all, not all of us have the luxury budgets available to have Neve/SSL/API/Dangerous/ pre's, with Cranesong, Apogee or Antelope conversion... and personal indulgences aside, I'm not sure it's even financially worth it any more; to invest in that stuff as a business move; we know that more than just a few of the big, pro (some even famous) studios - have been shutting their lights off and locking their doors in the past few years, ( some would even say at an alarming rate) and that many of these places have fallen directly as a result of smaller studios popping up nearly everywhere... and "budget" recording rigs being what they are, I think that this is where the plug-in market flourished. And, I do think that plugs can be useful; whether it's for a particular "modeled" sound, or for more forensic-type frequency or gain tuning.

But... for those who do still have and use the "creme de'la creme" of audio capture devices, with top-notch gain chains... well, I guess I just don't see it.

If you have very nice OB pieces available - say, real LA2's, 1176's, Pultecs, Focusrite Red compressors... and you had fantastic-sounding front-load capturing through the nicest, top of the line mics and preamps available, then why should you need to "fix" anything? I'm not saying that tweaks shouldn't be applied, that's what mixing is all about; getting multiple tracks and takes to sound good all together - but why would you reach for a $20 LA2A plug-in if you have an actual LA2a in your rack?

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

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

Sean G Sun, 10/30/2016 - 05:48
I cannot understand why you would use a plug-in if you had the real thing present...but then again there are those who hold the view that once you are in the box, stay in the box.
You only have to look at guys like Andrew Scheps who have gone down the totally ITB road now, and its not like these guys who are at the pointy end don't have access to the very best outboard gear available at their disposal...if they didn't already own it in the first place.

Another perspective is that it may just be a workflow thing too...everything has to be done at 10 times the speed now compared to even a decade ago...the budgets are gone when you could spend half a day just dialling in an audio chain or auditioning hardware...who needs that when you can just drop a plug-in onto a track and audition it in 30 seconds and if you don't like it you can remove it and just reach for another. You can set a whole template up in a DAW as we all know that is there from the first second...theres' no patching in like the analog days.

Then there's the whole recallability factor...next time you open your session its all there as you left it...nobody leaves the faders on a console untouched for 3 days waiting for a revision to come back to them..that is if you still have or are using a large format console today.

With the amount of things like plug-in latency, phasing issues, smearing and digital distortion that come hand-in-hand with plug-ins and the chug-alug effect in chaining in multiple plug-in after multiple plug-in on track after track after track, you need to eliminate as much of that as possible starting with the capture. Fix it in the mic, not in the mix I say.
That way, if it sounds like ass (thanks to Kurt I now love that saying) then it can only come down to two things...shitty equipment or shitty technique.

Get both of those things right and you are on the home straight.

kmetal Wed, 10/12/2016 - 18:43
Awsome, that reply clears up a lot. I think I finally understand how each component plays into the system as a whole.

audiokid, post: 442136, member: 1 wrote: 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.

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

Brother Junk Sun, 10/30/2016 - 06:39
Boswell, post: 442719, member: 29034 wrote: In the case of a HEDD in the SOS article, it's acting as an effects box on the main mix. If I understood the article correctly, he was using it to add subtle colouration to his 2-bus mix, so there was no sense of its output being an accurate capture of the input. If the subtle additions had significant components in the top octave of the incoming sampling rate (10 - 20KHz), then it's fair enough to capture the result at a higher rate. An exact doubling of the rate is indeed easier in processing terms than (say) going from 44.1KHz to 96KHz, but it still involves significant processing rather than simply filling each of the missing samples with the average of the two samples either side. There is also the point that if the box will go both up and down in rates (in -> out), then it probably uses a general SRC algorithm for all the possible rates, and will only go transparent if the input and output rates happen to be the from the same clock. The faults in this way of thinking are exposed when you realise that the captured output probably has to come down to 44.1KHz at the end.
Ty, that's about what I assumed. Colouration is a better word than gimmick, but you basically understood what I was asking

kmetal, post: 442721, member: 37533 wrote: Surprising to me was his use of parelell master/mixdowns, where he blends various 2trk mixdowns to taste. Also the use of izotope ozone was a bit surprising too.
I use Ozone! lol. I love Ozone actually. And they have a vocal one that I like but I can't think of it atm. Not Alloy, it will come to me. I think that's actually my favorite Izotope plug-in...and I can't think of the name of it.

I don't know what Parallel master/mixdowns means but I'll look it up. I'm assuming it's not like a wet/dry track mix.
kmetal, post: 442722, member: 37533 wrote: lol no shortage of storage. If you've got an internal HDD then it'll be better than using an external HDD for audio. This due to faster transfer rates of sata 3 vs USB or FireWire.
Yes, what has me wondering though isn't the Sata/USB/firewire speeds. It's that I don't know if the ethernet (often called gigabit ethernet, so 1,000,000 bytes per/s) applies the same speed to local connections. In other words, I'm not sure if the 1gb ethernet connection is still limited to that speed on a local network. I'm not saying it does or it doesn't. I'm saying I genuinely don't know. But if it is Sata 3 (that's around the right time, I'll look today) it's hard to beat that speed. 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.

kmetal, post: 442722, member: 37533 wrote: I've found that interface to be the best compromise of quality / features / price, available right now. If you don't require high I/o. It's expandable I/o his the adat.
Yeah, I will def pick that up in the future. I don't require many more i/o than I have, nm 12 (or 24, whatever it was).

Sean G, post: 442755, member: 49362 wrote: I cannot understand why you would use a plug-in if you had the real thing present...
That's the first thing I thought of. But you guys are the ones who have toyed with the real thing. I've never messed with a real compressor in my life. But I've always thought of the plug-ins as the secondary, or lesser option. But that's an assumption...I've never actually gotten to compare.

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.

Just out of curiosity, have any of you ever compared hw to the plug-in that imitates it? If so, what did you find?

Sean G, post: 442755, member: 49362 wrote: chug-alug effect it chaining in multiple plug-in after multiple plug-in on track after track after track
What is the chugalug effect?

**Edit, the pi I was thinking of is Nectar/2. I'm a hack, unlike the rest of you, so, I don't know if it would make your cut quality wise, but I love that plug-in. That, Melodyne and PT and I can do vocal tracks pretty quick.

audiokid Wed, 10/12/2016 - 18:51
kmetal, post: 442139, member: 37533 wrote: I think it's a mentality in cooking where you don't use more than 3 seasonings in a plate.
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.

Sean G Sun, 10/30/2016 - 06:58
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?
Yep...I have a Warm Audio EQP-WA...a Pultec clone, now whilst it doesn't sound exactly like a Pultec EQP-1A, its pretty damn close if you read the reviews, and a plug-in of the same just cannot replicate what running through Cinemag transformers and real tubes can do IMO. I like it so much I have another on back order from a month ago which is finally arriving tomorrow.

Here is an SOS review where they put the Warm Audio EQP-WA up against an actual Pultec EQP-1A in a studio comparison, and compare it to plug-in versions of Pultecs.
http://www.soundonsound.com/reviews/warm-audio-eqp-wa

Brother Junk, post: 442756, member: 49944 wrote: What is the chugalug effect?

Lol...thats' what I equate multiple plug-ins doing on a track ...chug-alugging down the track like a train...chug-alug....chug-alug....chug-alug....dragging down your cpu performance and adding latency to a mix....add to that the multiplier effect by the number of tracks loaded with plug-ins as well:D

Sean G Sun, 10/30/2016 - 07:07
Don't get me wrong, there are some really good plug-ins out there, and some are pretty close to the hardware they emulate. There is a trade-off with using anaolg hardware as well when you are coming out of the box and then going back in again...thats why IMO you want to have your hardware going in as part of your chain or in the middle between 2 DAWs...no going back into the same box as guys here like @audiokid & @Boswell will tell you.

There will always be a degree of degredation of the audio signal...thats a given and impossible to avoid. Its how you manage and minimise it as much as possible that which makes the difference.

kmetal Wed, 10/12/2016 - 21:32
audiokid, post: 442140, member: 1 wrote: I mix back and forth religiously in mono more than I ever have.

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?

Boswell Sun, 10/30/2016 - 07:38
Brother Junk, post: 442756, member: 49944 wrote: Yes, what has me wondering though isn't the Sata/USB/firewire speeds. It's that I don't know if the ethernet (often called gigabit ethernet, so 1,000,000 bytes per/s) applies the same speed to local connections. In other words, I'm not sure if the 1gb ethernet connection is still limited to that speed on a local network. I'm not saying it does or it doesn't. I'm saying I genuinely don't know. But if it is Sata 3 (that's around the right time, I'll look today) it's hard to beat that speed. 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.
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.

Brother Junk Sun, 10/30/2016 - 09:31
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.
I had totally forgotten about this. And the capital B thing. I never set up networks, nor has ethernet ever been an option that I was concerned about.

So, is this why the studios I've seen use the externals? It would be considered point to point I think...

**Edit, I'm an idiot. They are doing it for the portability.

The question arose bc externals are expensive, and they have older Mac Towers like mine in some rooms. You could buy a standard drive for half the cost of the externals they are using...maybe even less than half.

So why the external route? I asked the RE just to see if he knew and he said the owner takes them home a lot.

Question asked and answered...hoorah! That one has been bugging me for a long time.

kmetal Sat, 11/05/2016 - 19:38
Dbx has a compressor w just one slider on it. No other controls. That thing has a ton of sack on bass guitar. Love that little thing.

The cool thing about Dbx is it does have a sound. For better or worse. The 160 is moderately priced and you see them all the time in commercial studio racks. Some cheaper gear has a certain charm. Again for some things it works others it doesn't.

Insold my 166xl recently in my gear purge. I gave that one w the one slider to the studio. I think I sold then166 for as much as I paid and the other one was given to me so no loss.

Boswell Thu, 10/13/2016 - 04:43
kmetal, post: 442135, member: 37533 wrote: So are you saying you print different 2trk mixdowns with different settings in the same pass?
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.

kmetal, post: 442135, member: 37533 wrote: 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?
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.

kmetal, post: 442135, member: 37533 wrote: 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.
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.

kmetal, post: 442135, member: 37533 wrote: 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?
Yes, depending of the material and the purpose of the mix.

kmetal Fri, 10/14/2016 - 16:45
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!!

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

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

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

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

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

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

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

I will go keep reading.

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

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




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

Brother Junk Mon, 10/24/2016 - 07:41
kmetal, post: 442344, member: 37533 wrote: Logic has a great reputation . I've never had the opportunity to use it. I think a lot of people who are into more electronic forms of music, and using a lot of loops sway towards logic as their main platform.
It does excel with those sounds. Better than PT imo.

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

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

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

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

dvdhawk Mon, 10/31/2016 - 12:12
DonnyThompson, post: 442754, member: 46114 wrote: At that point, do we perhaps need to look at the engineer's lack of talent in getting the best capture possible? That's not rhetorical, gang... I'm really asking here...
Personally, I don't know how else to look at it, because if you had all the aforementioned pro-level equipment, and you still feel the need to "fix" certain tracks, then I think you have to look at the possibility that the fault lies with the engineer, because it's certainly not the gear, right?

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

@DonnyThompson

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

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

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

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

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

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

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

bouldersound Mon, 10/31/2016 - 12:35
Brother Junk, post: 442756, member: 49944 wrote: That's the first thing I thought of. But you guys are the ones who have toyed with the real thing. I've never messed with a real compressor in my life. But I've always thought of the plug-ins as the secondary, or lesser option. But that's an assumption...I've never actually gotten to compare.

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

audiokid Mon, 10/31/2016 - 12:56
bouldersound, post: 442817, member: 38959 wrote: A decent compressor plugin is way better than a run of the mill analog compressor. Really high end hardware compressors do things that can be hard to emulate digitally. Actually, all compressors to things that are hard to emulate, but what normal compressors do isn't worth emulating.
+1


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

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

Sean G Thu, 10/06/2016 - 17:25
Brother Junk, post: 441917, member: 49944 wrote: What I'm a little unclear on is what the separation gains you?
You mix in one and capture in the other. Allows for mixing & processing at higher sample rates and capturing at say 44.1 /16 bit for CD without sample rate conversion issues.
Brother Junk, post: 441917, member: 49944 wrote: Also, I haven't noticed this anywhere, are both computers are plugged into the DAC e.g. with firewire?
Two sets of converters are required...one for coming out of DAW 1 and one going back into DAW2. Firewire is one option.
Brother Junk, post: 441917, member: 49944 wrote: Can anyone tell me (ballpark) what that room right there would cost? I'm asking bc I'm totally ignorant about all that, but that looks $$$$!
I can't speak for Chris (@audiokid ) who owns that set up, but for something along the lines of a set up like that I don't think you would get away with spending anything less than 6 figures.

kmetal Thu, 10/06/2016 - 18:03
Brother Junk, post: 441917, member: 49944 wrote: I looked into VSL once bc of how fantastically amazing it is, but it was hella money. IIRC $2k for the full boat? Hot damn is it nice though. I can't imagine that's dropped to $.02 lol.

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


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

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

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

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

Kurt Foster Mon, 10/31/2016 - 13:30
audiokid, post: 442824, member: 1 wrote:

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

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

audiokid Mon, 10/31/2016 - 15:38
Kurt Foster, post: 442832, member: 7836 wrote: or the better the hardware the better anything sounds. that's a no brainer. i don't believe it can be attributed to solely digital however. i just don't agree. digital is fine. it works. but it's not better. just different. perhaps you like it more and that's fine. there's nothing digital has to offer that analog won't do as far as actual documentation of a performance. digital is easier to edit. that's the only advantage i can see. but then, i know how to pull down a fader at the end of a track. :ROFLMAO:

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

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

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

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

... and stereo digital compression rocks. :D

Brother Junk Fri, 10/07/2016 - 08:01
Sean G, post: 441936, member: 49362 wrote: Two sets of converters are required...one for coming out of DAW 1 and one going back into DAW2. Firewire is one option.
TY...so the guy above (last vid posted in this thread), that is not a true dual daw system? Or at least not what audio kid and a couple others here are doing?
Sean G, post: 441936, member: 49362 wrote: I can't speak for Chris (@audiokid ) who owns that set up, but for something along the lines of a set up like that I don't think you would get away with spending anything less than 6 figures.
Oh....sorry @audiokid, I thought you were just posting a pic, I didn't realize it was your room. I wouldn't have been so rude to ask how much had I known. But wow...what a space man. It makes me teary just looking at it.
kmetal, post: 441940, member: 37533 wrote: I purchased the VSL pro 5 player on sale for $180, plus $30 for the veinna key license dongle. The player, which is the part that hosts vsti and plugins, and a basic but great sounding set of samples comes with it. The audio fidelity imho cannot be touched by any other orchestral sample company. You get what you need and want and can afford.
$210 isn't so bad.
kmetal, post: 441940, member: 37533 wrote: The vsl player allows you to basically use an additional computer connected via Ethernet, and allows you to host vsti and use audio plugins on a seperate computer to take the load of the daw. This comes in handy w cpu intensive plugins an vsti's.
So, for $210, would that get all the benefits of a dual daw system? Or you still need 2 contverers and the summing mixer?

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

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

audiokid Fri, 10/07/2016 - 11:24
Brother Junk, post: 441961, member: 49944 wrote: Oh....sorry @audiokid, I thought you were just posting a pic, I didn't realize it was your room. I wouldn't have been so rude to ask how much had I known. But wow...what a space man. It makes me teary just looking at it.
No worries, its worth mentioning because it is an amazing system. Only because you asked, I put about $130,000 USD into it, had another $60,000 in various gear I didn't end up using because ITB was superior to my ears. I've sold about half of it and am going in slightly different direction. My focus is still 2 DAW's, passive summing as a character section with Sequoia for both DAW's.
2DAW's rocks and you do not need to invest in all that to achieve sonic excellence.
A lot of what you see is also for tracking and mastering. Most everything in that picture was used for tracking, mixing and mastering. Its setup to be used in any direction at a click of a mouse.

Brother Junk Tue, 10/25/2016 - 07:14
audiokid, post: 442116, member: 1 wrote: That being said, if that option was waved, I may take on that challenge. My DAW system meets world class sonics which will not degrade the path. In fact, if I was provided 96k tracks, real time SRC (DAW1 > AD> DAW 2, would sound better to my ears over bouncing down. To my ears capturing at the destination SR still sounds better when its done in real time which takes two DAW's.

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

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

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

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

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

What am I doing wrong?

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

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

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

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

DonnyThompson Tue, 11/01/2016 - 06:46
Brother Junk, post: 442870, member: 49944 wrote: What does this term mean?

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

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

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

Brother Junk Tue, 11/01/2016 - 08:25
DonnyThompson, post: 442872, member: 46114 wrote: In studio lingo, the word "tracking" is synonymous with "recording". ;)

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

It's basically anything that will degrade the quality of sonics of the tracking ( recording) process.
Thanks. That's what I thought but wanted to be certain.
Kurt Foster, post: 442873, member: 7836 wrote: just kill me.
I'm not sure if you are annoyed by the stupidity of my question, (maybe it's unrelated) but if it is my response, it has a reason. I've been doing a lot of back reading on the site and I'll see things like "poor tracking", "less than adequate tracking", "tracking problems", "tracking daw"etc.

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

kmetal Fri, 10/07/2016 - 21:05
Brother Junk, post: 441961, member: 49944 wrote: So, for $210, would that get all the benefits of a dual daw system? Or you still need 2 contverers and the summing mixer?

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

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

These are two seperate concepts BJ.

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

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

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


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

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

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

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

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

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

audiokid Tue, 10/25/2016 - 20:31
Brother Junk, post: 442594, member: 49944 wrote: And @audiokid (or anyone)...I entered this site loudly with debate about something. I said at the time, that it just happened to be one topic that I know a lot about (driver design)...and I said at the time, that I'm probably going to flood this place with questions so stupid you will be in awe...and I feel like this is one of them lol, ready?
No worries, we are all learning. No question is a stupid question either.

Brother Junk, post: 442594, member: 49944 wrote: Essentially, is up-sampling (is that the term?) to 24/96 after the recording is done, just a gimmick that adds 0/1's to make it fit?
Upsampling would be a complete waste of time, imho, unless for some reason you do this to create a special effect.

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

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

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

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

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