I would say that with these conditions it's probably not worth while trying analog transfers between the two DAWs. My experience is that it works best when mixing many tracks recorded at a high sampling rate down to a 44.1KHz stereo pair. Paring down the number of source tracks and sampling rate simply reduces the benefit, and the benefit may be negative going from stereo to stereo at the same sampling rate with no added analog effects, unless it's just sampling rate change that you want to achieve.

If I were in your position, I think the experiment I would do is set the two interfaces up on the two different DAWs and track two separate sets of takes using the same mics into the two interfaces. I would then have fun listening to the sets of recordings played through the same replay gear and making notes about what I liked, what I didn't like and what's just different.

Sounds like you have a fun weekend coming up!

Which Focusrite unit is it, by the way?

Boswell, post: 422468, member: 29034 wrote: Which Focusrite unit is it, by the way?

Scarlett 6i6. 2 channel pre, w/ SPDIF.

I was planning on A/B'ing them as you suggested anyway... I just thought I'd try the "mix to separate uncoupled DAW" path while I had it.

But, since I don't have an analog summing device to put in between them, nor do I have any special OB gear I want to use to add character, ( with the exception of one 1176/ black face,/ Rev F series/, circa 1977 (?) but one won't do me much good for a stereo transfer. ;) )

So, based on what you've said, I guess there's no real benefit to doing it.

Thanks, Bos. :)

Sorry, I forgot to respond to the S/PDIF bit: this is a digital bit pattern, so should not be affected by routing through different interfaces.

With the Focusrite, I believe that the S/PDIF output can only replay from the computer and that there's no routing to send the digitized analog inputs out of the S/PDIF, so you can't just use it as a converting pre-amp to the Presonus. However, it may work the other way round by selecting "Main to S/PDIF out" in the Presonus after creating a stereo mix of input channels. I still think a separate pass using standard DAW recording would be better as it would avoid any signal manipulation inside the Presonus.

Boswell, post: 422475, member: 29034 wrote: I still think a separate pass using standard DAW recording would be better as it would avoid any signal manipulation inside the Presonus.

I was hanging right there with you right up until the above. LOL.

I don't follow you here. Are you saying that even if the Presonus can send a stereo mix out of its SPDIF, and into the SPDIF IN on the Focusrite, it's still better to not do this, and either:

A. use standard analog ins and outs on both ?

or..

B. Don't bother with the secondary mix-down DAW method at all, and just stick to the standard ITB rendering on the Multi-track DAW?

Other than the already mentioned benefits we get from the uncoupled process, I'm convinced a large percentage of my results is directly related to the capture DAW Master Bus and DA monitor perspective. Being able to hear what your uncoupled mix sounds like on the capture side, while mixing into the master is the most critical step to all this. You are at a huge disadvantage if this step is missed. This is why DAW1 and a console monitor section is little value now.
Think about a game of golf and where the cup is. Is the cup on DAW1, the analog console or after the limiter between the export to the web. If you can't see the hole, you are guessing. This is also why I use a mastering program on DAW2. I mix into the master bus just as we do ITB, but now moved to another computer via the hybrid uncoupled process.
Contrary to what I used to believe, I'm convinced this is a lot simpler. Simply uncoupling two DAW's is all you need to do. No console, no fancy analog gear.
DAW1 >DA uncouple AD> DAW2> DA > monitors. Should you want to use optional hardware like a console, summing box etc, it goes between the two converters "DA AD". Simple.

Having an independent monitor controller that will connect all three locations instead of having to unplug makes it flow better, but the main idea is to monitor off the DA of DAW2.

DonnyThompson, post: 422476, member: 46114 wrote: I was hanging right there with you right up until the above. LOL.

I don't follow you here. Are you saying that even if the Presonus can send a stereo mix out of its SPDIF, and into the SPDIF IN on the Focusrite, it's still better to not do this, and either:

A. use standard analog ins and outs on both ?
or..
B. Don't bother with the secondary mix-down DAW method at all, and just stick to the standard ITB rendering on the Multi-track DAW?

It depends what you are trying to test. Having said that I doubted that you would get a meaningful improvement by using two DAWs, I took it that you were then interested in whether you could hear a difference between the Presonus and the Focusrite pre-amps under your conditions (same mics, same recording acoustic, same monitoring etc). I didn't want using the mix process inside the Presonus to muddy the waters.

Chris has come in and said that his greatest improvement is from clock uncoupling between his two DAWs, which I fully believe. To go a bit further than that, the tests that I did with what I called the "two-box" method all show that my best improvement came from (a) using a high sampling rate for the source tracks, (b) not using a digital SRC, (c) performing an analog mixdown from box 1 to box 2, and (d) not locking the sampling clocks of the two boxes (if they happen to be at the same nominal rate).

All these conditions interact in different ways, so it's hard to say whether you would get a better result with your proposed trial, but don't let me stop you trying it out!

Incidentally, I even got some improvement by using 48KHz source rates rather than going all the way to 96KHz for box 1 when box 2 was at the CD standard of 44.1KHz. This seemed to show that there is something going on right at the top of the audio band which when multiplied by the number of source tracks causes unpleasantness in the mix if digitized at the same rate as the source.

One set of experiments I have not done is using the two-box mixdown with both boxes clocked at the same nominal rate, and then seeing if I could detect any difference between having the two boxes running with asynchronous clocks or the same (external) clock. A test like that may be a way of teasing out the contribution of the different components.

:cool:
watching.

I do have the ability to monitor separately off of the 2nd DAW.

And with some choices, too. I have a Hafler Trans-nova 1500, as well as an old (60's) Crown 20/20 ( still works and sounds great, btw), and as far as monitors, the following:

Alesis Monitor Ones (6" passives)
JBL 4408's (8" passives)
Tannoy Reveals ( 6" passives).
Auratone 5C's (passives - and old, like late 70's - but they still sound just like Auratones are supposed to sound)
Yammie NS10's (passives, late 80's)

No worries about inserting a clock in the chain, as even if I wanted to ( which I don't, I trust you guys saying that I shouldn't) I don't have one, anyway, so that decision is made for me. ;)

Boswell, post: 422479, member: 29034 wrote: Incidentally, I even got some improvement by using 48KHz source rates rather than going all the way to 96KHz for box 1 when box 2 was at the CD standard of 44.1KHz.

All of my Source DAW projects/tracks have all been done at 44.1 - or 48 - so, from what I gather you saying, in order to really hear a difference, my best bet would be to do a new project at 96k/24 (or 32 float) and then mix it down to the separate, uncoupled DAW, where I would use the analog ins and capture at 48k (or, 44.1 if I was planning on it eventually seeing a CD) ... is this a correct assumption?

Or, as Chris has said... might I hear a difference in quality, even with the source DAW at 44, and then, using an analog routing, capture on the second DAW (while monitoring off the second DAW)?

Head spinning... Hep Me, Jeebus!

i notice a difference at 44.1 ...or 48 ..... i mean with analog summing. even a Mackie will sound better than an itb mix imo ....

Boswell, post: 422479, member: 29034 wrote: One set of experiments I have not done is using the two-box mixdown with both boxes clocked at the same nominal rate, and then seeing if I could detect any difference between having two the two boxes running with asynchronous clocks or the same (external) clock. A test like that may be a way of teasing out the contribution of the different components.

We are indeed onto something here, Bos.

Q: Is there something weird with how one DAW sums?

Never the less, I am convinced world class level is achieved when you are mixing into mastering software on the capture DAW. Sequoia is proof of this for me.
The ability to mix into a digital mastering matrix is sonically comparable and in some area's, superior to analog mixing and mastering hardware. The bonus, this step saves you thousands of $ buying gear you do not need.

Uncoupling adds an interesting change along with being able to avoid SRC.

As an added choice, a transformerless console offers clear advantage because you will want to keep both analog and digital options available to you, without either side effecting the other. There will be times tubes or trannies add the extra mojo digital can't emulate, Or there will be times you do not want analog colour in the most pristine session or channel.
Being able to hard bypass or audition hardware for critical evaluations is also how you learn to emulate the analog matrix. Each year digital technology improves and when it happens, you will know how to emulate one more step that one was once, only achieved via analog. The M/S matrix in Sequoia is a great example of this. $70,000 of my analog chain is devoted to the hardware M/S section. Sequoia can replace most of it now.

DonnyThompson, post: 422487, member: 46114 wrote: No worries about inserting a clock in the chain, as even if I wanted to ( which I don't, I trust you guys saying that I shouldn't) I don't have one, anyway, so that decision is made for me. ;)

I used a $6000 10M super clock for 6 months and it was 100% useless. It improved nothing because my system is clocked.

DonnyThompson, post: 422487, member: 46114 wrote: All of my Source DAW projects/tracks have all been done at 44.1 - or 48 - so, from what I gather you saying, in order to really hear a difference, my best bet would be to do a new project at 96k/24 (or 32 float) and then mix it down to the separate, uncoupled DAW, where I would use the analog ins and capture at 48k (or, 44.1 if I was planning on it eventually seeing a CD) ... is this a correct assumption?

Or, as Chris has said... might I hear a difference in quality, even with the source DAW at 44, and then, using an analog routing, capture on the second DAW (while monitoring off the second DAW)?

I track at 44.1 all the time and still get better results like this. In fact, quite a few members have given me there sessions and master at 44.1 and I have shown improvement just by uncoupling to the master DAW.

Its a win win from how I'm hearing it.

What is the best way to insure that a mix through Am-munition's M-S processing matrix will translate with integrity to another, separate, uncouple DAW, without using an analog summing device, or M-S encode/decode matrix in-between?

Okay. Let's say I have Samp as my multi track platform. SR is 48k, resolution is 24 bit. I've strapped ammunition to the 2-bus, and am working with it in M-S mode. I then come out of the analog outs of one pre, into the analog ins of another, to a separate PC, with Sound Forge as the capture software, set for 44.1/16 bit. I hit record on one and hit play on the other.

What am I missing to insure that the M-S, or, for that matter, a straight forward stereo 2 mix will translate honestly? Is a high voltage summing device needed? Is any summing device required?

Or, is the above scenario - without any summing or matrix encode/decode device involved - all that needs to happen?

audiokid, post: 422477, member: 1 wrote: Contrary to what I used to believe, I'm convinced this is a lot simpler. Simply uncoupling two DAW's is all you need to do. No console, no fancy analog gear.
DAW1 >DA uncouple AD> DAW2> DA > monitors. Should you want to use optional hardware like a console, summing box etc, it goes between the two converters "DA AD". Simple.

Having an independent monitor controller that will connect all three locations instead of having to unplug makes it flow better, but the main idea is to monitor off the DA of DAW2.

This is it right here: If you had 2 versions of Sam, it would be choice. You are better using another DAW and having Sam on the capture because it is the DAW with the best M/S and summing / mastering code.

Process:
DAW1 (Sonar) >DA uncouple line in> AD> DAW2 Samplitude> DA > monitors

with the exception of gear and converters I use, I believe Samplitude on DAW2 will get you the same results as me. Mix into Samplitude on DAW2 and capture the mix at -10 or test to taste. I use Prism AD so this is all subjective to mine vs what you have. Normalize the channel, EQ to taste. Add AM-Munition on the master bus and start experimenting. Optional choices are mono the bass 200 hz full or paralleled to choice. Maybe not at all. All your choices on DAW2 will be based on how you process the mix on DAW1. Follow? If you hear something happening on the master DAW2, look for the suspects on DAW1. You are mixing into a master. As you are using AM Munition and the limiter (maybe L2, ProL etc), you can keep tweaking the mix on DAW one. The more you do this, the more you will learn about the mix. As you develop your skills, you will want two versions of Samplitude because iy is the best tracking, mixing and MASTERING DAW for us.

Use your ears. Export both MP3 and Wave and compare. Go back to the mix and keep learning.

let me know how you smile ;)

To add:

M/S is where you improve the space of the mix. I also may use a "TINY" amount of Reverb on master bus to glue a common space to the whole mix. Remember, you are doing this to open it up. If you need "channels with effects", you do that in the mix on DAW1.

I have an Master template because once you get a mix close, you will start comparing mixes between each captured track with a common master section. This is also an excellent method to compare dozens of mixes. This is when you really start learning about less is more.

audiokid, post: 422494, member: 1 wrote: Optional choices are mono the bass 200 hz full or paralleled to choice.

I've never mono'd that high before... usually I set it for 100hz and down, so this intrigues me. I'm gonna try that.

For me, the capture box is just that: it captures the 2-mix at the target rate and makes a 24-bit file preserving exactly the samples as read from its ADC. Once that track is a captured file I can do what I like with it, and this is where digital storage really comes into its own - the data is exactly as captured. That's really why I prefer a simple capture method, even using Audacity, where the open-source nature of it meant that I have been able to trace through the C code to satisfy myself that it makes no changes to the incoming samples on their way to storage as long as you do not use anything in the Effect menu.

If you are prepared to carry out the extra pass of a 2-mix raw capture, it doesn't matter that you don't have a copy of a top-class DAW on your capture box, as you can send the file to any computer that has your DAW of choice for your mastering or whatever process you need to perform on the 2-mix. If I'm mixing a pro song, it's at this point that the captured tracks go off to the mastering house. If I'm doing a demo or it's for charity or other freebee work, I bring it up to an adequate level with a bit of compression and maybe some small EQ adjustment ready for CD burning. I don't call that mastering.

Incidentally, thanks to Kurt's flagging, I've just bought a copy of the Harrison Mixbus software. I'll get it installed over the Christmas period and experiment with where best it can slot into my workflow.

I really don't have much interest in mastering - because I'm not a mastering engineer. I know there are guys who can wear all the hats - as recording, mixing and mastering engineers - but I'm not one of them. ;)

What really interests me is insuring that I can get the highest possible quality/integrity in the final output on my end, so that when it does go off to a real M.E., I've given them the best possible pre-mastered final mix that I can deliver.

I'm going to set up this workflow tomorrow ( i have a mix session today) and I'm hoping that I will be able to hear the difference that I've read about here - in using a second capture DAW and mixing/monitoring off of that end, instead of mixing and rendering it all ITB.

Boswell, post: 422517, member: 29034 wrote: That's really why I prefer a simple capture method, even using Audacity,

I'm pretty sure that SoundForge will suffice as the capture platform. It may not be as good as Sequoia, but I don't believe I would ever come close to using everything that Sequoia offers as a mastering platform, because again, I really have no interest in mastering. ;)

I own and really like MixBus. I don't use it for everything I do, but it serves me well when I am working on a project that calls for it. And heck, at $39, what's the risk, right? ;)

DonnyThompson, post: 422516, member: 46114 wrote: I've never mono'd that high before... usually I set it for 100hz and down, so this intrigues me. I'm gonna try that.

I'll play with that setting. I don't believe it goes below 150. I'm not saying even use it! Its all dependent on what you want and what you are doing on DAW1. I'm not even saying use AmMunition or a limiter. Its all optional. :)

DonnyThompson, post: 422525, member: 46114 wrote: I really don't have much interest in mastering - because I'm not a mastering engineer. I know there are guys who can wear all the hats - as recording, mixing and mastering engineers - but I'm not one of them. ;)

What really interests me is insuring that I can get the highest possible quality/integrity in the final output on my end, so that when it does go off to a real M.E., I've given them the best possible pre-mastered final mix that I can deliver.

I'm not a mastering engineer either but I think and study like one. What we are doing is mastering backwards. Hybrid and the 2 DAW system is a Mastering approach. The advantage to the entire step is we are now able to mix into the capture making better mixes than before. Not seeing this one is selling yourself short.
To my experience there is huge advantage mixing into the master DAW. In fact, I wouldn't even bother with this, if I didn't do exactly what I suggest. ;) Your loss at this point.

PS. The capture is still unscathed. You don't need to change a thing, but, why wouldn't you. That's where this all becomes really obvious. :D

audiokid, post: 422535, member: 1 wrote: Your loss at this point.

I'm not against this process at all - in fact, there's like at least 10 posts above with my name on them, asking all the questions I can to understand the process. If I wasn't very interested in doing it, I wouldn't have barraged this thread with all the queries that have. ;)

What I meant was that I have no interest in being a mastering engineer. :)

DonnyThompson, post: 422539, member: 46114 wrote: I'm not against this process at all - in fact, there's like at least 10 posts above with my name on them, asking all the questions I can to understand the process. If I wasn't very interested in doing it, I wouldn't have barraged this thread with all the queries that have. ;)

What I meant was that I have no interest in being a mastering engineer. :)

I know :love:
I'm just saying to whomever is listening ;)
Mixing into the capture is the bomb here. Including, monitoring off that section. At the end of a year, you will be wondering if you should even be sharing half this stuff so loudly. Its why I remove a few of my posts, occasionally. We're seeding for the harvest.
Just saying...

My next step is probably a stereo summing device. I've been looking at models by Dangerous, SSL, TK Audio, Coleman... I'm nowhere near being able to afford it yet, but I can see it as part of my future workflow.

Coleman RED48 Mmmmmm.....once you go red you won't go back

ric3xrt, post: 422578, member: 47701 wrote: Coleman RED48 Mmmmmm.....once you go red you won't go back

I love hearing about all this stuff, please tell us more? ! :D

Back in nov I was asked to run some Protools files(22 tracks in to Black face adats,I have a few different summing boxes but I made it clear that the only summing box I had of the quality he would expect was the Neve, 8616,and that we would have to bus some of the tracks.....he wasn't willing to bus tracks at this point, just wanted to hear if the ADATs were the answer to the sound his client was looking for.
He showed up a week later with a red 48, I have to admit I was disappointed at 1st, the Red 48 there are no insert points for the channels only for the main bus, each of the channels are panned hard left or right, so there is still a lot of ITB work. but the sound was clean and clear, and the Mix Cue came in handy. I've heard few people say the red 48 has a big Console sound to I disagree It didn't color the sound at all. in the end I traded a Mackie SR 40.8 mixer for the RED 48
I love the Coleman gear I do have and its all been reliable and clean, in the edit room I have a Coleman MS2, and its fantastic for comparing source tracks , no color added what goes in comes back out. In the Control room the MS2 is great for comparing what a band has recorded to their favorite CD. I can't say enough about their Quality and sonic clarity.