# Bussing/summing/ endlessly submixing in a DAW - Problems



## madbulk (Jan 14, 2007)

Got this brand new for 2007 daw template of mine (Logic) -- a fairly elaborate pile of submixing sections that I thought would make it easy for me to do... anything. Give a fella 30 inch monitors and watch him create channel strips, baby!
What I can't get over is the evident signal loss everytime something moves to a new stage.
Should I be looking to simplify my mixer or should I be looking for something amiss in my path?
Or in other words, do these DAWs do a really lousy job of bussing and summing, or must I have done something stupid in designing my mixer? (Nobody say "both.")


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## Bob L (Jan 14, 2007)

Hi... normally I stay put in my own forum... but Peter has invited me to look in over here and offer some input on discussions such as this.

I am the developer of the SAW product line, currently SAWStudio. It exists because of issues exactly like this. I have been a recording engineer for over 30 years and have been involved in the growth of digital audio since its infancy. I was initially dissatisfied with the audio quality results of the early digital systems as compared to the analog sound I was used to and decided to jump in and experiment with some alternative engine and interface design concepts. The result has turned into a 14 year career and the development of the SAW (Software Audio WorkShop) approach to digital audio.

You may find it interesting if you are wrestling with quality and performance issues such as mentioned in this thread.

The engine is quite different than the industry standard approach to digital audio and is created using hand coded assembly language instead of C++ and object oriented code design. The results are quite different than the rest... it offers an alternative for those looking for something more from their current digital audio experience.

The design is focused on recreating the audio experience from the point of view of the typical high-end analog recording studio. The interface follows the flow of the large mixing console and tapedeck method of recording, and then expands on that to add in all of the exciting digital based editing functionality that does not exist with an analog based setup.

The engine uses integer math instead of floating point and this, along with the huge headroom of the internal summing bus design, eliminates these signal degradations you are reffering to when looping signals back around on multiple passes.

The end results can stay very silky and smooth and sound very analog, even with many submix stems being used in the finished mix.

The efficiency of the code also eliminates the need to do this even in the most dense of sessions, for the most part, so it also offers other options of getting complex mixing projects done in different ways.

The virtual console can completely replace, if desired, very high-end expensive recording large mixing desks and give you the ability to mix in your environment as if you were in the most expensive studio control rooms anywhere in the world.

The console eq and gates and compressors are designed to match very high-end gear and offer very warm audio results... this allows you to do most of the needed processing directly within the internal hand coded math based design and only use DX or VST floating point plugins for special effects. The results can be very impressive.

I have stayed completely in the virtual environment now for years, with no need to use external mixing hardware, and am happier than ever with my project results.

Sorry to have this sound like a commercial, but Peter asked me to introduce myself and make myself available to offer alternative solutions here on the forum for those interested in taking their audio results to the highest levels.

My expertise is not in orchestration or music theory, but in technical audio and recording, and I am happy to offer my ideas and observations to the group in this regard.

I have been scanning through the various message threads and am very impressed with the level of dedication I see here and the depth of discussions concerning the constant evolution of the virtual sampling world.

I have no desire to sales-pitch anyone into switching DAWs... but as you begin to dig deeper into the idea of a better finished audio product and/or a more efficient manner of getting there within the DAW experience, you may find that SAWStudio offers you quite a different ride... some love it... others hate it... but I leave that for each to decide for themselves.

Even though my life revolves around SAWStudio, and most of my methods will pertain to how its done in that environment, many of these concepts and techniques will apply to mixing audio, in general, on a multitude of DAWs and hardware.

I also invite everyone who is interested over to my website and forum link at www.SAWStudio.com for more info. Feel free to watch the online demo videos to get a quick introduction to my approach to digital audio.

A special thanks to Peter for inviting me in to his world.

Bob L


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## José Herring (Jan 14, 2007)

Welcome Bob,

I was wondering if you could post an introduction to SAW studio in the commercial section of the forum. I'd love to discuss it more and should do so without Hijacking Madbulk's thread.


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## Bob L (Jan 14, 2007)

Jose,

It seemed to me that the commercial forum thread was setup for sample libraries and VST synths... it seemed that this thread was a good jumping in point for me because the idea of signal degradation was one of the main reasons I started designing the software.

But, if you care to start a thread with some of your questions about SAWStudio or mixing in general, I would certainly be happy to get a discussion started.

I do not really feel comfortable starting such a thread myself, as I am concerned about causing bad feelings with the other forum members concerning what might be construed as spam. :smile: 

Bob L


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## Frederick Russ (Jan 14, 2007)

madbulk @ Sun Jan 14 said:


> Got this brand new for 2007 daw template of mine (Logic) -- a fairly elaborate pile of submixing sections that I thought would make it easy for me to do... anything. Give a fella 30 inch monitors and watch him create channel strips, baby!
> What I can't get over is the evident signal loss everytime something moves to a new stage.
> 
> Should I be looking to simplify my mixer or should I be looking for something amiss in my path? Or in other words, do these DAWs do a really lousy job of bussing and summing, or must I have done something stupid in designing my mixer? (Nobody say "both.")



I've seen this debate before. Its in regards to comparing digital sum bus vs analog sum bus. For most purposes the 2-channel sum bus found in Logic should be sufficient, but I have heard an expensive proposition of creating 4-8 stems of your work (for example, one brass stem, one strings stem, etc), route thoseò§   N‚Å§   N‚Æ§   N‚Ç§   N‚È§   N‚É§   N‚Ê§   N‚Ë§   N‚Ì§   N‚Í§   N‚Î§   N‚Ï§   N‚Ð§   N‚Ñ§   N‚Ò§   N‚Ó§   N‚Ô§   N‚Õ§   N‚Ö§   N‚×§   N‚Ø§   N‚Ù§   N‚Ú§   N‚Û§   N‚Ü§   N‚Ý§   N‚Þ§   N‚ß§   N‚à§   N‚á§   N‚â§   N‚ã§   N‚ä§   N‚å§   N‚æ§   N‚ç§   N‚è§   N‚é§   N‚ê§   N‚ë§   N‚ì§   N‚í§   N‚î§   N‚ï§   N‚ð§   N‚ñ§   N‚ò§   N‚ó§   N‚ô§   N‚õ§   N‚ö§   N‚÷§   N‚ø§   N‚ù§   N‚ú§   N‚û§   N‚ü§   N‚ý§   N‚þ§   N‚ÿ§   Nƒ §   Nƒ§   Nƒ§   Nƒ§   Nƒ§   Nƒ§   Nƒ§   Nƒ§   Nƒ§   Nƒ	§   Nƒ
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## Bob L (Jan 14, 2007)

Does the current Logic allow virtual summing mixes to be created just in the software directly, or does it force you to mix back through the audio card devices?

If you must go through the hardware conversion process, then you will definitely suffer some signal degradation... no different than bouncing tracks on an analog 24 track deck. If you can do software only bounces, and if the math is handled properly, you should have no resulting signal degradation.

One test you can try is to setup a basic mix and create a mix file of it... then place that result back onto another track with all its settings set flat and the fader at zero db making no theoretical level adjustments... then phase reverse that new track and see what happens. If the bussing and stem mix process is really the source of your perceived quality loss, you will hear some residual audio because the process itself caused math errors and the results do not pahse cancel perfectly.

You can do this process in SAWStudio with perfect phase cancellation, so you don't have to worry about quality loss if you feel the need to group and create stem mixes on a very complex project.

Bob L


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## ComposerDude (Jan 15, 2007)

"32-bit" floating point -- which by the way is the default native audio format for OS X Core Audio -- gives you about 24 bits of actual audio data plus an 8-bit exponent that shifts those bits in magnitude. Supposedly it can represent low-level signals with greater precision than fixed point but I'm convinced that part of a "transparent" sound is maintaining low level resolution/precision _in the midst of_ high level signals, and there's some question how effective 32-bit float may be in this regard. Certainly enough people have complained about "in the box" summing compared to the same summing done via outboard mixers that this effect deserves more scientific scrutiny to identify precisely what these people are hearing.

64-bit floating point may provide a solution but in practice won't be as efficient on commonplace 32-bit native machines. With future adoption of Vista and 64-bit machines that may change.

What floating point does buy you is more forgiving gain-staging in your mix -- the theoretical ability to have your faders down really low, 0.00 0.00 0.00 0.00 0.00 0.00
08:20:01 eth0 0.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00
08:30:01 lo 0.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00
08:30:01 eth0 0.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00
08:40:01 lo 0.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00
08:40:01 eth0 0.00 0.00 0.00 0.00 0.00 0.


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## Nick Batzdorf (Jan 15, 2007)

Just treating this as a Logic question and leaving aside SAW and tiny epsilons for a second, there's absolutely no reason you should be losing quality at every stage when you use bus objects in the Logic mixer, Brian.

Unless you're talking about subtle differences - which are real, but not the kind of differences "you can't get over" - my hunch is it's something silly like losing level because of the pan law setting.

What specifically are you hearing?

Sorry to hijack this thread.  I'm actually interested in SAW - I haven't seen it for years.


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## madbulk (Jan 15, 2007)

Hi Nick. thx. 
The very worthwhile crux of all this notwithstanding, I'm sure in my case it's something 'silly' indeed, some basic pre/post fader bus error in my architecture. It's not the pan law cuz it's not unique to any hard panned tracks or such. And what I'm hearing and seeing is not degradation, it's just a 50% level drop if I send to a bus/aux fader versus going right to the output stage. If I really can just go +6db at a pre-master and make it all up, maybe that's fine, but it's certainly counter-intuitive.
Simple example.... Stylus, whether stereo or multi-ch, is outputting to a bus called "stylus submix." Stylus master A is clipping. Stylus submix is peaking at -10db. Where is all that signal going? I hate to revert to a "what the hell am I screwing up here" question even after this expanded into such a thoughtful discussion, but with everyone's permission, they can co-exist.


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## JohnnyMarks (Jan 15, 2007)

Reminding of the oft-heard advice to avoid summing tracks showing peaks great than -6dB, to protect from "intersample overs." Paul Frindle (of Sony Oxford and SSL design fame) has discussed this at length online (search "Paul Frindle" and "intersample") and, as I understand it, believes this issue is a primary cause of sound degradation on DAW mix busses. This suggests to me that boosting your faders to +6dB wouldn't be the answer...

Cheers.


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## madbulk (Jan 15, 2007)

Ashermusic @ Mon Jan 15 said:


> There is a lot about Logic that is not intuitive and that you are not understanding.


This, I don't doubt. But maybe you could just describe the particular thing that explains why the level is so much weaker when output to a bus post fader and then brought up on an aux.


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## Ashermusic (Jan 15, 2007)

madbulk @ Mon Jan 15 said:


> Ashermusic @ Mon Jan 15 said:
> 
> 
> > There is a lot about Logic that is not intuitive and that you are not understanding.
> ...



It shouldn't be, which is why I say there is something wrong with your setup. I can't diagnose it without seeing it. You are welcome to send me a Logic song and I will look at it.

But trust me, you would be wise to call Steve or someone like him.


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## madbulk (Jan 15, 2007)

I'm sure Steve is good at what he does. Thanks for the suggestion. And it's super kind of you to offer to look at the file. But if Logic shouldn't do what it's doing, I'm happy rooting out the problem. Really, that's what I wanted to know -- that it shouldn't be.


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## Nick Batzdorf (Jan 15, 2007)

That's what I thought. You're talking about a 50% level drop - which in reality is more than 3dB, I'm willing to bet - and we're talking about summing busses. I knew this had absolutely nothing to do with any of that.

Okay, let's think this through. You have Instrument objects, right? Their outputs are assigned to busses, ja? The busses are at unity gain, I assume?

First question: are they stereo busses?


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## madbulk (Jan 15, 2007)

3db it is. And yes, to answer your question, stereo buss.

Experiment to get us on the same page, perhaps:
New Logic Project/Shared Template/Basic Production; Stereo RMX on Inst 1; RMX Factory Multi the first (Adrenaline Rush/80_Chaos Theory).
Re-route output of inst 1 to a bus. Output that bus to another bus. That bus to a third bus. Call that third bus on Aux 1. Unity gain all around.

Inst 1 peaks at 2.5.
Bus 1 peaks at -0.4
Bus 2 peaks at -2.7
Bus 3 peaks at -5.7
Aux 1 peaks at -8.7
And Out 1-2 peaks at -1.1

So, that's why my elaborate series of submix sections was showing so little signal toward the end. But what's the logic? Or is it really not supposed to do this?


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## Nick Batzdorf (Jan 15, 2007)

That's bizarre!

I'll try it when I get a chance. It's obviously a pattern, though - 3dB each state isn't random.


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## Nick Batzdorf (Jan 15, 2007)

Actually, can you email me your session? [email protected]


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## madbulk (Jan 15, 2007)

Be super happy to when I fire everything up again tomorrow. So sleepy are we on this coast -- those of us on this coast with 1 and 3 year olds anyway.


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## david robinson (Jan 15, 2007)

g'day, yep logic's summing is questionable..
i used to use the auxes but found an amount of degradation that was not acceptable, levels not withstanding.
so, now, i'm reduced to using groups as a work around. not good at all.
sounds a lot better tho.


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## Nick Batzdorf (Jan 15, 2007)

There's nothing wrong with anything's summing as far as I'm concerned, but that's another subject.

Madbulk, I just put an empty EXS24 on an Instrument track and looped a bar with a sustained note (empty EXS24s play sine waves, which makes it easy to check the level).

I assigned the Instrument track's output to bus 1-2, assigned bus 1-2's output to 3-4, 3-4's output to 5-6, 5-6's output to 7-8, and 7-8's output to my main output pair. The level is identical to the tenth of a decibel in every channel strip, according to the meters.

It's quite possible you just have a corrupted session. Please try to recreate the same thing in another one and see if it does the same thing.

Either way, there's absolutely nothing wrong with the way Logic passes levels.


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## madbulk (Jan 16, 2007)

yeah, well, that's fine for you but what am I supposed to do?!
I ran your experiment from the default autoload and had the same results as you.
Great.
But when I make a new project based on "shared template/basic production" I get my mess.
So what's the difference? Where's the -3db checkbox I need to uncheck?! 
Here's that file. It's not my autoload, but it's doing the same thing and its pared down to the one inst and 8 busses.


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## Ashermusic (Jan 16, 2007)

madbulk @ Tue Jan 16 said:


> yeah, well, that's fine for you but what am I supposed to do?!
> I ran your experiment from the default autoload and had the same results as you.
> Great.
> But when I make a new project based on "shared template/basic production" I get my mess.
> ...



I tool a look at it and found the issue. This problem seems to occur in Logic with this kind of routing when the pan law is set to -3 db instead of -3 db compensated.

Go under File to Song Settings>Audio>and change the Pan Law to either -3 db compensated or 0 and the problem goes away.


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## madbulk (Jan 16, 2007)

Rats, Nick's instinct was correct? That SUX! Like the guy isn't doing enough right around here! And holy cow... how did I not even look at it when you mentioned it? Did I mention I'm sleepy a lot? Sorry about that.
I hadn't yet quantified it as 3db, had only glibly said 'half.' Thought I was exaggerating. It really was half.
And thank you so much, Jay. Feel free to add this page to your upcoming Logic book. "Don't let this happen to you."
-3 compensated it is.
So, do all the prefab templates that ship with Logic already have this? Or must I have picked it up somewhere along the line and it was adopted at a higher level than template?
How did it happen? Any idea?


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## Ashermusic (Jan 16, 2007)

madbulk @ Tue Jan 16 said:


> Rats, Nick's instinct was correct? That SUX! Like the guy isn't doing enough right around here!
> And thank you so much. Feel free to add this page to your upcoming Logic book.
> So, do all the prefab templates that ship with Logic already have this? Or must I have picked it up somewhere along the line and it was adopted at a higher level?
> How did it happen? Any idea?



There are quite enough Logic books and DVDs out there without mine, thank you

It seems that some if not all of the templates are set this way I only opened a couple and they both were.


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## madbulk (Jan 16, 2007)

good catch.. but that was probably just a little metering variation because I was sending it a pile of drum loops and not the much more sensible sine wave that Nick later suggested. When I put a sine through it you got the halving right down the line. And Jay obviously explained why. What I don't know is why "out 1-2" in that model looked like it was making up gain at the end. Something else must've been amiss there.


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## Nick Batzdorf (Jan 16, 2007)

Good.

Now let's have some fun. Wanna argue about summing? 

Better yet, someone go shout "STEREO MIC TECHNIQUE" into a room full of mastering engineers and run like hell.


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## kid-surf (Jan 17, 2007)

summing --

I know my summing may not be the be all end all. Yet...

Yet... my stuff sounds good enough to me. Case closed. :D 

In other words, I've got way more important things to worry about. But I hope guys like Bob L continue to make this stuff _their_ priority. 

PS... i know you were joking, Nick.


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