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analogue transfers between uncoupled DAW's

Discussion in 'Recording' started by DonnyThompson, Dec 17, 2014.

  1. DonnyThompson

    DonnyThompson Distinguished Member

    Okay... so, I've got a decent second audio I/O for several days... a friend of mine is letting me use his Focusrite I/O while he's on vacation, so I thought I'd try mixing to a separate DAW.

    Questions...

    There will be no external analog summing involved, hi voltage or otherwise. So, without this step, should I even bother with mixing to a secondary DAW?

    If the answer is yes, should I be using the analog outs of my main I/O (The Presonus VSL) to the analog ins of the other I/O (the Focusrite), or ...should I be doing this all via SPDIF, which both units have?

    The second PC is a laptop, on which SoundForge 8 is loaded. I planned to mix directly to this prog, and then, instead of rendering it, or "bouncing to", or "exporting as", simply just save it as a 44/24 bit .wav file. Is there any reason I shouldn't be doing it this way?

    Or, once again, without the benefit of summing in between, should I even bother with this?

    Thoughts?
     
  2. Boswell

    Boswell Moderator Distinguished Member

    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?
     
  3. DonnyThompson

    DonnyThompson Distinguished Member

    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. :)
     
  4. Boswell

    Boswell Moderator Distinguished Member

    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.
     
  5. DonnyThompson

    DonnyThompson Distinguished Member


    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?
     
  6. audiokid

    audiokid Staff

    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.
     
  7. Boswell

    Boswell Moderator Distinguished Member

    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.
     
    TheJackAttack likes this.
  8. pan60

    pan60 Active Member

    :cool:
    watching.
     
  9. DonnyThompson

    DonnyThompson Distinguished Member

    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. ;)



    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!
     
  10. Kurt Foster

    Kurt Foster Distinguished Member

    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 ....
     
  11. audiokid

    audiokid Staff

    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.
     
  12. audiokid

    audiokid Staff

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

    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.
     
  13. DonnyThompson

    DonnyThompson Distinguished Member

    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?
     
  14. audiokid

    audiokid Staff

    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 ;)
     
  15. audiokid

    audiokid Staff

    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.
     
  16. DonnyThompson

    DonnyThompson Distinguished Member

    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.
     
  17. Boswell

    Boswell Moderator Distinguished Member

    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.
     
  18. DonnyThompson

    DonnyThompson Distinguished Member

    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.

    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. ;)
     
  19. DonnyThompson

    DonnyThompson Distinguished Member

    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? ;)
     
  20. audiokid

    audiokid Staff

    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. :)
     

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