(patch bays and routing hardware)

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

Example:

• I can choose from a list of analog comps, EQ's processors in my rack for that task.

• I am able to digitally move gear to be in front, middle or behind (ABC, BCA,ACB etc) in a bus lane or channel strip, mono or stereo process.
• I am able to do seamless changes of this hardware on the fly, which makes comparison learning very powerful.

I use two DAW's to capture, organize and make notes. The mass of this workflow can be saved so I can refer back to stored setting for extended learning and/or to repeat what worked for a particular project. This workflow bridges analog and digital together better.

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

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.

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?

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.

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.

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.

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?

kmetal, post: 442145, member: 37533 wrote: Are you using a stereo set w a mono button? Or do you have a single speaker mono reference?

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

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.

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!!

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.

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.

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.

I ordered the Vienna Ensemble Pro 6 to do the above. Can't wait...

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

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?

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

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

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

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

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.

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.

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.

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.

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.

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.

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

Was that was the article about Rik Simpson? I remember reading that and thinking Uh - huh, not the best solution to that particular problem. However, what he was getting from the HEDD was harmonic addition, and since that added detailed things to the mix, it's not a problem to have the output generated at a higher SR. I guess he was using digital in (at 48K) and digital out (at 96K) of the HEDD, so no DA - AD process involved there.

For many reasons, it doesn't fit my concept of a two-box uncoupled system.

kmetal, post: 442682, member: 37533 wrote: Your running two HDD right? One for you programs and one just to record audio too?

I think so...I use externals. PT is installed on the OS X drive though, is that wrong?

Maybe just for clarification, how would you accomplish that...just by putting the whole PT file (.ptx, audio files, etc) on the externals? I have internals as well, that o8 Mac Pro has 8tb in it lol (best $150 computer ever!)

The reason I do the external thing is because that's what I've seen done at studios. My external is connected with ethernet though instead of USB...maybe that's the problem? The only reason it's setup that way is because I need to find another one of that style USB.

I ask for the clarification bc I recall seeing this thing setting, buried deep somewhere in the PT menu, where PT will try to split the writing between two drives. At least, that's how I recall it...almost like a raid format. But I was told that this isn't necessary past PT9. It was more for the days when scsi drives etc were in use.

kmetal, post: 442683, member: 37533 wrote: 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.

The RME isn't that much $...gracias. This? (It looks sweet!) http://www.markertek.com/product/rme-babyface-pro/rme-babyface-pro-24-channel-multi-format-mobile-usb-2-0-high-speed-audio-interface

And it's frickin tiny. I had to look at all the views and zoom in to make sure it had all the i/o I need.

kmetal, post: 442682, member: 37533 wrote: Both that he had a capture rig, and more so he used it to capture at a higher rate.

So, this whole decoupled dual daw thing, I find super interesting. Feel free to correct me where I'm wrong. But it seems to combine the best of everything. And I felt like, I pretty much achieved a rudimentary understanding of it. Until I read what you wrote and Boswell wrote...lol Although, it sounds like this doesn't apply to a pure, uncoupled system, but I am curious as to...

What would be the point of a 48k in, and a 96k out? I've devoted a solid 1o minutes to that question and I can't think of why one would do it? E.g. Why would he not be 96k in/out? Or 96k in, 48k out? Bos, are you saying his particular equipment setup led to some sort of pleasing harmonic anomaly? E.g. like a Tube amp vs solid state? Basically, he took advantage of a useful gimmick he found?

Boswell, post: 442716, member: 29034 wrote: I guess he was using digital in (at 48K) and digital out (at 96K) of the HEDD, so no DA - AD process involved there.

So would the "the less SRC, the better" rule of thumb still apply?

Brother Junk, post: 442718, member: 49944 wrote: So, this whole decoupled dual daw thing, I find super interesting. Feel free to correct me where I'm wrong. But it seems to combine the best of everything. And I felt like, I pretty much achieved a rudimentary understanding of it. Until I read what you wrote and Boswell wrote...lol Although, it sounds like this doesn't apply to a pure, uncoupled system, but I am curious as to...

What would be the point of a 48k in, and a 96k out? I've devoted a solid 1o minutes to that question and I can't think of why one would do it? E.g. Why would he not be 96k in/out? Or 96k in, 48k out? Bos, are you saying his particular equipment setup led to some sort of pleasing harmonic anomaly? E.g. like a Tube amp vs solid state? Basically, he took advantage of a useful gimmick he found?

So would the "the less SRC, the better" rule of thumb still apply?

Usually the point of the uncoupled two-box method is that you capture the mix as accurately as possible at the destination sampling rate. As I've suggested several times in the past, keep in your mind the model of a stereo microphone being recorded by box 2. Everthing up to that point is concentrated on generating the virtual stereo microphone output.

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.

Boswell, post: 442716, member: 29034 wrote: Was that was the article about Rik Simpson?

Yup! I love 'inside track' articles becuase it's like being a fly on the wall on some pretty awsome high end sessions. I love seeing what those guys use and don't use.

Here's a link to the article (for anyone interested) , and a snapshot of the particular summing section of it. I subscribe digitally, but I'm not sure how much non-subscribers will be able to read.

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.

http://www.soundonsound.com/techniques/inside-track-coldplay-hymn-weekend

Brother Junk, post: 442718, member: 49944 wrote: Maybe just for clarification, how would you accomplish that...just by putting the whole PT file (.ptx, audio files, etc) on the externals? I have internals as well, that o8 Mac Pro has 8tb in it lol (best $150 computer ever!)

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.

So you've got your system drive which has all your programs and applications installed, and an audio drive where you save all your PT/daw sessions. This will save your recorded audio there by default. You also would have your samples on the audio drive, or on a third dedicated sample drive.

Brother Junk, post: 442718, member: 49944 wrote: The RME isn't that much $...gracias. This? (It looks sweet!) http://www.markertek.com/product/rm...mat-mobile-usb-2-0-high-speed-audio-interface

Yes that's the interface I was referring to. B&H sells it for 700$ even, 50 cheaper than most everywhere else.

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. The closest direct comoeteitor would be apogee duet. Which is similar in quality and channel count, but Mac only, no dsp, and a bit more money.

The others I've compared to offer higher channel and lower quality for a little cheaper, or higher channel count and similar quality for double the price.

On the cheaper side it's MOTU traveler, and focusrite scarelett. On the higher channel count, double price, and similar quality it's the apogee quartet, and the antelope zen tour.

Once you get above those price points you get into the agogee symphony, antelope Orion, and mytek, and other high end devices.

Brother Junk, post: 442718, member: 49944 wrote: So would the "the less SRC, the better" rule of thumb still apply?

Yes as far as I know it holds true across the board for digital.

Boswell, post: 442719, member: 29034 wrote: 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.

I'm confused about this, could you explain it a bit more?

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... ;)

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.

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.

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

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.

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.

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.

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.

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.

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.

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:

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

dvdhawk, post: 442816, member: 36047 wrote: or to compensate for poor tracking.

What does this term mean?