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ITB digital mixing - best practise?
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Author ITB digital mixing - best practise?
nickbaba
I've been reading a lot of threads on analog summing vs mixing ITB, and hearing people talk about 'rookie errors' etc that make ITB-mixes sound bad. No details were given , but I am assuming they mean things like mistakes in gain staging, over-compression, over-EQing etc.

Couldn't find a thread on ITB mixing best practises, and thought it might be useful, as so many (most) people mix ITB these days. But I'm asking for your wisdom, not preaching.

Anyone care to share their hard-won knowledge on best practise for a successful ITB mix?

For instance, where do you have your individual channel/master levels peaking? What's your plug-in chain on the Stereo Output bus? What obvious mistakes should be avoided? etc
craigie77
probably best to start of with researching the K- metering system
and having your monitor's calibrated to this as well

i use k-16.. gives plenty headroom and enough dynamic range for me...


tho production is heavily dependant on the style of music you want to do

i start of from my Kicks at - 12 db and work up from there

and shelving the sub freqs really helps... othere than that just cut loads of unneccesarry frequencies as well....even if you think theres nothing there i'd still cut it

with regards to master chain...- nothing. ( except maybe a little compression for mixing.( TDR- Free and amazingly transparent compressor---- http://www.tokyodawn.net/tdr-kotelnikov/ ). but take that off when i render down,,,, then do the mastering in something else)

i know folks are gonna disagree... but this is how i work
stk
Everyone has their own methods. Honestly it is a lot harder to screw up ITB than OTB, with regards to gain staging, noise floor and the like.
"Rookie errors" are far less damaging and meaningful when you have essentially infinite headroom and infinitesimal noise floor.

I've been releasing music for almost 20 years and I still do dumb things like reducing (or boosting) peak levels by 12db across the board because I've pegged (or starved) everything on individual channels (because I tend to fly off the cuff and experiment with extremes signal chains as I go) and can't be bothered going back to address individual gains.
Never had any complaints.

It's all art, rules are for suckers grin

Of course there are some common sense things that you learn as you go along, regarding polarity ("phase"), mono compatibility, restrictions of some physical reproduction media (vinyl). But none of these are specifically ITB related.
I think when some people speak of "rookie mistakes" they mean that due to the low barrier of entry for sound production these days, people do not know things that they would have otherwise been required to learn.
But never underestimate the creative power of naivety.

Cheers
Refund




https://soundcloud.com/refund/refund-prismatic-1
[s]https://soundcloud.com/refund/refund-prismatic-1[/s]

My production style currently is to completely fucking brickwall limit everything until the song is a weakened emancipated version of itself and has absolutely no balls left.

There are some really good guidelines about balancing the mono and stereo content of a mix to keep everything in check, but I fuck off those rules and make everything stereo all the time (making the eq and sidechaining portion of the job about 10x harder in the process)

I record multitrack out of my modular in multiple runs, you can do this with a multi input soundcard, but in my case I sequence from my trigger riot / modcan touch sequencer, which has rock solid timing. so I can record multiple takes 2 tracks at a time (aligning stuff takes fucking forever) I strongly suggest a multi input audio interface if you're not broke/backwards like I am.

This is a live track so I've played it a hundred times already and can perform the song even when all the other parts are muted.

as far as mixing is concerned, since I'm working in floating point land (I use reaper) my only concern is to keep the volume sliders in a 'useable area' i.e. not too low on the virtual fader and not too high that I'll hit the 'limit' that i can boost something to.

this mix has 8 elements

kick
snare
hat
hat2
bass (fmvdo)
acid (doepfer sync through wasp filter)
lead (the bone crushing cyclebox sound)
atari (ataraxic translatron)

you'll see I have multiple tracks of everything, so to keep these in check, each of those are summed to a 'group'

drums
bass
acid
lead
atari

there is some parallel distortion processing of the acid and drums, I have a send that goes out to a seperate channel for extra processing. there is also a glitch plugin on the drums for the breakdown.

I send the kickdrum output to the compressor for the 'lead' and 'bass' sound for some very subtle (but crucial) sidechaining, my kicks are already very weak, so I don't want it to get lost in the bass of the other sounds.

In the channel strip you can see that I EQ everything, then I add effects.

I try to remove all the bass components of anything that isn't a kickdrum or a bassline. the high end I don't really give a fuck about since you ear is good at hearing that stuff without tonnes of help.

I'm a big beleiver in desctructive eq, a lot of videos online well tell you to be super careful when eq'ing because you'll destroy the guitar tone that someone has worked so hard to create. my advice is to fuck off that advice when dealing with modular and do what you want.

first: I usually search for frequencies I don't want by making a thin bandpass filter, boosting it, then scanning the frequency range for all the 'horrible' sounding frequencies, once I have found them, I cut them by a small amount.

then I make some super wide parametric filters and create an 'eq curve' that accentuates what I like about the sound as destructively as I want. keeping in mind what may make it sit better in a mix.

If I have some elements that conflict in the same audio frequency spectrum, I'll try to use the eq to seperate them and give them thier own space.

Most of the in-channel effects are really subtle and only there to add stereo content.

you'll probably notice a modest amount of envelope automation data too, I like to 'virtually' ride the levels so that everything sits where I want it, sometimes the 'phrasing' of the material changes the perceived loudness even at exactly the same volume, my editing is to fix that.

Once I have the mix how I want it, I make further groups for any 'meta' effects I want to do, in this example there is a 'fade group' with the atari and lead channels so that I can do a filter/volume/glitch fadeout.

and lastly, I 'master' everything in the master track, but that isn't very modular specific and is pretty well known stuff (eq, verb, compress, limit)

and if you listen to my track and think it sounds horrible, at least you know what NOT to do.
melodydad
There are several Focal Press books such as Mixing Secrets For The Small Studio. I have not read that particular book but if you look at the Kindle version on Amazon, you can browse the first chapter. Also the book has links to audio examples - even multitrack files for honing your skills.

Also UK's Sound On Sound mag have some useful publications.
nickbaba
This is cut and pasted from a thread on GS posted by Skip Burrows - hope that's OK. it has some relevant info which maybe useful to someone browsing this thread for tips:

Quote:
To learn to mix ITB coming from an analog world you must revisit what Voltage reference Analog consoles work at, and make appropriate adjustments to translate this to work ITB.

The first thing we must ask is simply what is 0VU. What does it mean to us. Lets use an SSL G+ as our point of reference mainly because I work on those every day. If we put a signal into the line input of the SSL so the channel meter reads 0vu, that also, is referenced as +4 or 1.23 volts. A kick ass SSL will go out to about +24DB, so we have approximately 20 DB of headroom above the 0 VU point on the meter before the signal goes to crap.

Now let take a common situation. A Client hands you a Protools session and you spread it out over the SSL console. Like most people today every track is recorded as hot as hell. Most pro Eng's will use proper gain staging and get the now slammed meters reading around 0VU or 1.23 volts. By lowering the line trim we now have a good level into the desk so we can Compress/Gate/EQ the Signal without it overloading the processing. Sounds simple right? Remember that all outboard equipment was designed to work around the 0VU/+4/ 1.23 Volt reference. So by putting the incoming signal at around this reference, your rack equipment will work better as well.

Why use a +4 reference? Well remember that the 1.23 volt reference came from the tube days where 1.23 volts was enough voltage over the plate noise that you still had a good signal to noise ratio, but still left room above 1.23 volts to allow for normal audio operations.

Now to ITB. Lets pretend we have the same setup as we did on the SSL. Client hands you a session that’s recorded hot as hell. Now most folks mixing ITB don't understand reference levels when relating it to Digital. To have the same amount of "headroom" as we do on the SSL we must create a reference of 0VU or 1.23 volts at -20 from 0DBFS or the top of the Digital scale.

So if you simply place the good old trim plugin as the very first plugin, you now have the ability to adjust your tracks to our Mixing (+4/1.23 volt) reference IE -20. Just like you did on the SSL. You have have the same amount of headroom. Now with your tracks properly gain staged, you can add EQ/dynamic plugins and not run out of headroom. You can also insert hardware and they will operate much better as they are operating at the level they were designed to operate at.

Plugins use the same reference at real equipment. Never try and drive them to the top of the Digital scale. Don't try and make your mix look like a master. You don't do that on an analog console, so why do we do it ITB?

The answer is simple. DAW meters suck Butt. There should be a meter mode in all DAW's that makes the meter at 3/4 scale equal -20 at 1.23 volts. Just like the old VU. This way, novices will quit corn-holeing their levels.

Something to think about. The noise floor of an analog desk is about -75 DB from our +4 reference. Our equivalent "problem level" below our -20 reference in digital is well over 100 DB. So please don't let people tell you analog has more "headroom" than digital. This is simply not true. Headroom is only relative to your noise floor below your reference. Remember if the volume is to low, turn up the darn speaker volume.

Running a Digital mix right to the top of the scale is like running your SSL mix buss where the VU meters are slammed all the way to the right and you are constantly hitting it at +25. No one will get a good sounding running the desk like that. You won’t get a good sounding mix in digital either.


So what does all this mean? Put simply, proper gain staging is essential to both analog and digital mixing. You just need to correlate the references between the two. Once you figure this out, I'll Guarantee your mixes will start to sound open and wide, just like the good old analog days.
glitched01
The main thing is to use your ears.
Concentrate on the content!

A crap tune will always be crap.

The basics? Keep the individual channels low and make sure the master mix isn't clipping. There are no magical mixer settings.

Extensive calibration is usually not necessary to get a decent result.

Those are the basics.

I have a friend who is obsessed with meters, levels, and "sweet-spots," but he rarely records anything. Don't be one of "those guys."
stk
nickbaba wrote:
This is cut and pasted from a thread on GS posted by Skip Burrows - hope that's OK. it has some relevant info which maybe useful to someone browsing this thread for tips:


My guess is that that advice dates from the days when Pro Tools had shit internal summing (or so I'm told).
In the modern reality of floating point processing, IMO it's pretty obsolete (correct me if I'm wrong, by all means).
Black Medicine
Not so long ago, I attended a workshop given by Robert Henke, and part of it was focused on mixing ITB. He gave a demonstration, using multiple Utility plugins, taking 80db of gain from a signal in a channel, then boosting the same signal back in the Master with 80db of utility plugins, and there was no signal degradation or difference. Then he did the opposite adding 80db of gain on the channel, and reducing it at the Master, and as I recall, that didn't work so well. I think we discovered that the maximum you can throw at the Master is about +60bd, before it starts farting out. Still, 60db of headroom is decent, and you can do more than that inside most of the stock plugins without breaking up. Not that you'd want to, but you have that kind of dynamic range at your disposal.


There are certain simple tricks that are a good idea to use whenever possible, depending on what you are trying to mix. Personally, I mostly make hard banging, fucked up techno, and say what you will, if a mix isn't good on that stuff, it's pretty obvious. There's just not a lot to hide behind. Dynamics matter, tonality matters, and it's one of the front lines of the loudness wars. Playing live is a whole different animal, but for making sellable tracks, you have to compete if you don't want to get left behind. If you're not trying to do that, and just going for art, or shits and giggles, or whatever, then maybe this advice will be of less interest.

Noise floor is certainly an issue with most sounds not synthesized in the box, and certainly with most samples, so I tend to chop anything below 180Hz on any sound that isn't a kick or a bass tone.

I used to just start with all of my levels around -12, and fiddle about until things jelled, but I got turned onto a trick using pink noise, setting levels at -20, and going from there. It's interesting.

What I sometimes will do in my composition sessions, is what is considered the entire wrong way, i.e. putting a mastering chain on the master WHILE I'm creating the tune, and adjusting my mix as I go so that when it hits the processing, it does what I want it to, maintaining a decent amount of predictability. When things are kind of ready, I'll do a couple of quick and dirty renderings, with a few things adjusted on the Master channel, then I'll turn that stuff off, back all of my levels off a bit, and render everything to audio, for my mixdown session.

Obviously, there are lots of things to consider, like compression on the channels, and such. Generally, I'll turn all compressors off before rendering stems for mixdown, and compress them then. Some sounds, I'll flatten with the compression on them before mixdown. There's no hard and fast rule.

The mixing session happens all in audio, with nothing warped, and in at least 24/48, if not 24/96. I get my levels right so that it sounds good to me, carve out whatever I can to let the sounds make room for each other, without losing character, and get my final mix to peak at about -6db.

Sometimes, I'll set up a mastering chain in this session, too, to help me adjust what needs adjusting, because sometimes, it's just not going to come out like you think it will. Sure, the old philosophy of not using mastering to fix a mix is a good rule for most types of music, like rock, or Jazz, but in the electronic world, you're not necessarily trying to capture all of the sweetness of any single instrument's character. Sometimes, you're just trying to beat the sounds into utter submission, and make the whole piece do what you want.

As far as things like EQ go, the rules are there to be learned, understood, then broken. I've watched serious badasses use the stock EQ's in Logic and Live to do outrageous twistings of sound that end up working and sounding amazing. I've also seen most folks crash and burn because they didn't learn to use EQ gently before they started going apeshit with them. Also, many of the ones that crashed and burned were using the EQs to shape sounds that ended up fighting each other for space in the mix. That's the last thing you want to do.

I'm not going to get into discussing side-chain compression or EQ, but it suffices to say they they can be incredibly useful tools for getting things out of each other's ways when you just can't give up certain tonal characteristics of two (or more) competing sounds in a mix without sacrificing the overall feel of the piece.

This piece was done almost entirely in the box, with a couple of sounds recorded in from my Mopho, and a couple slaps on a P-bass. I did it using the techniques described above, and it worked. The thing I did most is follow some of the most sage advice I've ever been given:


"Learn what a good mix sounds like, then trust your ears."

Enjoy.

https://hearthis.at/ghwpxHZQ/lizard-on-a-rock-mix-1/
Tropic Al
Good technique I use is to setup buses / groups for groups of instruments : i.e. drums, bass, guitar, keys etc...
And also setup a 'master' bus that all of these buses run through.
This way it makes it easy to process these 'groups' i.e compress all of the drums, eq all of the guitars, filter all of the keys etc...
And as everything is going through the master bus (where all of the master processing is) then it makes it easy to A/B any reference tracks that you have pulled in, without having to bypass any processing that would be on the master channel of the DAW.
cretaceousear
I liked your piece a lot Refund.. both the tune and the mix. thumbs up

There's still lots of stuff on the Reaper forum about setting master at 0 and never changing it. I think some limiters assume master is at 0 but apart from that it's all relative.

I think so many of the mix and mastering comments on t'net are to do with creating rock and pop sounds, but that's rarely the aim for users here - and in fact is something to be avoided.
All I really know is that though I've read and watched lots of tutorials, it's a process that can only be learnt through repeated practice.
chamomileshark
It's something I sometimes worry about and then I just let it all go.

My first recording were to either a ReVox or Portastudio and other than the tape compression that naturally came with that technology nothing else. And it sounded ok.
oisin
just a quick note re: ITB vs mixers etc.;

i have a pretty decent Allen + heath mixer in my studio now (made in england). it sounds quite good and can get some nice warmth and distortion. ive spent some time routing mixes through it. generally it sounded better, but in the end ive kind of given it up, opting to stay ITB: faster, easier to recall and not THAT much of a noticeable difference to justify it all.
Phil999
Harrison Mixbus is a good DAW for ITB mixing.
1977
Technical: It's all about gain staging but it's not as forgiving as 'good' analog can be. In other words: never clip it = don't go to the red at any point! [You can clip the mix afterwards during the mastering process if you have good enough tools in hand but that's something slightly different from 'ITB' mixing]. Other than that it's up to you and you can happily do it in the name of art. Be creative!
MoogProDG
I use a combination. Some effects ITB, 16 channel summing mixer, outboard on the summed groups, then into a stereo buss with API 2500 and stereo pultec EQs.

I will say this- If you record your source audio how you want it to sound on the record and don't say to yourself "That's close, I can always edit it later".. your mixes will be 100x rougher.
hyena
stk wrote:
Everyone has their own methods. Honestly it is a lot harder to screw up ITB than OTB, with regards to gain staging, noise floor and the like.
"Rookie errors" are far less damaging and meaningful when you have essentially infinite headroom and infinitesimal noise floor.

I've been releasing music for almost 20 years and I still do dumb things like reducing (or boosting) peak levels by 12db across the board because I've pegged (or starved) everything on individual channels (because I tend to fly off the cuff and experiment with extremes signal chains as I go) and can't be bothered going back to address individual gains.
Never had any complaints.

It's all art, rules are for suckers grin

Of course there are some common sense things that you learn as you go along, regarding polarity ("phase"), mono compatibility, restrictions of some physical reproduction media (vinyl). But none of these are specifically ITB related.
I think when some people speak of "rookie mistakes" they mean that due to the low barrier of entry for sound production these days, people do not know things that they would have otherwise been required to learn.
But never underestimate the creative power of naivety.

Cheers


off topic but mate, i just discovered your music as terminal sound system and must say i'm totally blown away. i'll start by grabbing the "a sun spinning backwards" 2x12" (or dust songs, still a bit undecided)
kites
like mentioned, it seems like a tuned room, monitors and headphones a crucial point to address first.

Unless your making tunes just for yourself.

How many of you use acoustic treatments and how little of it can you get away with? Of course there are varying factors like room dimensions, but I wonder how much mixing should be done before sending tracks to be mastered.

Also, does anyone use measuring mics and calibration software? Like this:
http://designingsound.org/2015/10/review-sonarworks-reference-3-measur ement-microphone/
stk
hyena wrote:
stk wrote:
Everyone has their own methods. Honestly it is a lot harder to screw up ITB than OTB, with regards to gain staging, noise floor and the like.
"Rookie errors" are far less damaging and meaningful when you have essentially infinite headroom and infinitesimal noise floor.

I've been releasing music for almost 20 years and I still do dumb things like reducing (or boosting) peak levels by 12db across the board because I've pegged (or starved) everything on individual channels (because I tend to fly off the cuff and experiment with extremes signal chains as I go) and can't be bothered going back to address individual gains.
Never had any complaints.

It's all art, rules are for suckers grin

Of course there are some common sense things that you learn as you go along, regarding polarity ("phase"), mono compatibility, restrictions of some physical reproduction media (vinyl). But none of these are specifically ITB related.
I think when some people speak of "rookie mistakes" they mean that due to the low barrier of entry for sound production these days, people do not know things that they would have otherwise been required to learn.
But never underestimate the creative power of naivety.

Cheers


off topic but mate, i just discovered your music as terminal sound system and must say i'm totally blown away. i'll start by grabbing the "a sun spinning backwards" 2x12" (or dust songs, still a bit undecided)


Thank you for the kind words, what a lovely thing to read before trudging off to another day at the day job smile

Speaking of how "not" to mix, Dust Songs is full of deliberate misuse of processing and (technically bad) gain staging in order to achieve a particular aesthetic.
But yes, off topic, but appreciated thumbs up
stk
I will add to my previous gung ho comments re: it's all valid/art. I do 100% believe and stand by this, but if you think there is ever a possibility of you using an old fashioned analogue mixer, by all means learn the basics of working with such a device. Knowledge is power, and you might surprised as to how unforgiving towards bad gain staging some analogue mixers are.

Cheers
hyena
you are very welcome.
i ended up ordering a sun spinning backwards smile
and also found out you are part of Halo, which i listened a lot to about 8-10 years ago smile
Yeggman
bonylam wrote:

i want to know the brand for the digital scale.


The "digital scale" that poster is referring to isn't a product or an item, and thus doesn't have a brand.

The post is comparing the different ways a given signal is measured / displayed in an analogue system vs in a digital system.

In an analogue system, the point where the signal clips in the system is, in the example given in the post, "+24db". In a digital system, the point where the signal clips is 0db.

In this example, the real signal level is the same, but the way the signal is measured/displayed in each system is different.

Analogue and digital have different scales for measuring the same signal. It's like the difference between measuring temperature using the Fahrenheit temperature scale or using the Celsius temperature scale - "32 degrees fahrenheit" and "0 degress celsius" are the same temperature - the freezing point of water - measured on different scales.
memes_33
biggest rookie mistake in ableton live: not disabling warp mode on loops during mixdown.
mrflannery
nickbaba wrote:
This is cut and pasted from a thread on GS posted by Skip Burrows - hope that's OK. it has some relevant info which maybe useful to someone browsing this thread for tips:

Quote:


Plugins use the same reference at real equipment. Never try and drive them to the top of the Digital scale. - so true. Especially with analog modelling plugins.

Don't try and make your mix look like a master. You don't do that on an analog console, so why do we do it ITB? - disagree with this for modern music. Mixing through a master buss helps to make the mix interact well with your processing. It's one of the great benefits of mixing ITB


Structuring my mixes through auxes assigned to instrument groups before going to the master buss was a real game changer for me!
coffeeshopped
As someone who was doing everything ITB and is just now regularly starting to use a hardware mixer and more external gear, my biggest takeaway is the immediacy of control you have with hardware: If something doesn't sound right, you reach out and grab a knob and change it, which to me is so much easier than guiding a mouse to the appropriate area of a screen and clicking and dragging a virtual knob or slider.

So, my two cents would be: a big key to better ITB mixes is to have external controllers that make it quick and easy to adjust your levels and your EQ, so that you can play with the mix easily and constantly. In the end it's a feedback process between the speakers and your ears and hands. Have as little friction as possible in the middle.
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