The "Secrets" of Mixing ITB

[mchimes]

As a home recordist who has resigned himself to working with the benefits and limits of the DAW alone (i.e. no console, external summing or bussing), I am always looking for ways to maximize the sound quality of the music I make.

I have noticed a common notion among the experienced, that very good sounds can be made and (from there) maintained in the digital domain alone. But there always seems to be caveats attached to such statements like:

. . . if you understand how to mix ITB
. . . if you gain stage properly in the plug-ins
. . . if you know the differences between analog and digital
. . . if you know where the "VERY small sweetspot is in PT"
. . . if you know how the digital mixing bus works
. . . etc.

So I'm wondering, specifically, how YOU approach mixing ITB? (or would approach it if you weren't hindered by that $600,000 Neve console

How is it different in practice/theory for you than analog mixing?

What do you pay more/less attention to ITB?

"How much harder do you have to work?"

What are your "secrets?"

Looking forward to your thoughts,
Mike

[Blast9]

Excellent post...

personally, i'm sceptical about the so-called digital "sweet spot"

I'd like to know ALL your secrets, so I can tell everyone else, and pretend they're mine

******************************************************

Joking aside, I think we've got the start of a very useful thread...

[Jules]

There was (is) a problem with PT Mix + systems when track counts got over 30 or so.. the sound would (does) get thin.

This seems to have been fixed in HD now.....

I would say that using the latest & greatest plugins is key.

[mchimes]

Originally posted by Jules
T

I would say that using the latest & greatest plugins is key.

But, specifically, how do you use them?

For instance . . . look at how Dave Pensado uses the Waves C1 Compressor:

“To me, of all the equalizers you could use on a vocal, I like this best. But you’re saying, ‘that’s not an equalizer, that’s a compressor.’ The way I set it up, it’s set up as a dynamic equalizer. In other words, what I’m doing is I’m telling the compressor to only compress the frequency that I’ve set it for. I’m telling the compressor that whenever the frequency gets above a certain threshold, knock it down. But you’re saying, ‘why can’t you do that with an equalizer’? Well you can, but it’s just not as smooth for some reason. If you listen to Christina Aguilera, she sings her verses smooth and powerful and round, and the compressor—if you could actually watch it—is actually doing nothing on those verses. But when she gets to the high notes, she gets a lot of 2 kHz in her vocal, and the compressor automatically pulls that down, and makes the vocals sound less harsh and less piercing. And you can control the amount you want to leave in. Christina actually likes a little bit of that in her vocal, so I can’t pull too much out, but the compressor does it automatically. This setting is useful on just about everything. If you move the frequency to 8 kHz, or 10 kHz, then you’ve automatically got a de-esser. The only thing you’ve got to be aware of is that it will automatically add about a 340-sample delay to your sound. So just highlight the waveform and shift it to the left, 340 samples, and you’ll be okay. Of all the plug-ins that I use, it’s the most utilitarian. The setting shown here is my setting. Of all the secrets I’m giving away, that could potentially be the most useful.”

I'm interested stuff like this . . . possible chain combinations of plug-ins and using tools in new or unprescribed ways that would yeild an unforseen benefit.

Keep 'em coming

[DrC:Drive]

I have PT HD and mixing ITB yields excellent results, i used to have Mix+, and its a no brainer when it comes to comparison. One trick is to use analog inserts, but mix ITB. I buss my individual drums to an Aux bus and strap a Pendulum Audio ES8 across this bus and mix it back with the original. It gives a high end sheeny punch to the sound, and glues the drums together. Each drum track will also have Phoenix and Sony Dynamics. I will also mult the snare to my Avalon 737sp EQ, to get the upper frequencies right in a way that plugins cannot. i will mult the BD 3 times and add different amounts of Sony EQ, sometimes an SPL transient designer and mix all to taste. Get the phase right, and the drums are cooking!! This hybrid approach but still mixing ITB works well for me

Cheers

James

[Knastratt]

I don't really think there's that much difference. It's - as always - still in the ear of the engineer. There are, however, a few traps to fall into.

•Don't over saturate. Hitting the meters hard is stupid. With 24 bits resolution there's plenty of room in "the last" 8 bits to let the dynamics develop. tutt

•Having gazillions of possibilities in plugs doesn't mean you need to use them all. Still, "less is more" applies along with treating the tracks carefully not to bash'em.

•Having powerful tools for corrections doesn't mean that basic recording theory is surpassed. Good room acoustics, proper mic placing, right choice of mic pre, ample amount of compression/limiting during tracking are issues that never can be corrected for later in the mix. The same goes for performance.

•It is still not a loudness race. Master with care.

Obvious points? Absolutely. And thats exactly where lots of mistakes are done.

[mchimes]

Great advice! Somtimes we can't see the forest for the GUI

[Lagerfeldt]

· Yes, be very aware of what's going on in each plugin, it's quite easy to overload without it actually showing.

· Be aware of potential very high freq build-up (16kHz+), that will give a very tiresome sound. Perhaps use gentle sloped eq's to rolloff the very top on some processed tracks.

· Work with 24 bits and a higher samplerate, e.g. 96kHz to avoid unnecessary digital artifacts, especially when equalizing

· When mixing in a DAW you will normally need a bit more compression than in the analogue domain, since you don't have the luxury of tape compression or any other kind of automatic compression (tube stuff, mixer summing etc.). Obviously you can compress the individual tracks a bit more, or perhaps some light bus compression is the answer (use sensible A/R settings). We're talking compression both as in "gain reduction" and as in "glue", i.e. melting stuff together (drums+bass for instance).

· Another big issue is the low quality reverbs available for DAWs. Get something with a decent ER algorithm, that's magic for getting stuff to sound real or melt together in the mix.

· Use digital to your advantage. Brickwall limiting is a powerful tool when it's not overused (Waves L-series etc.). Use sidechain compression to melt stuff like kick+bass, and at the same time controlling your levels. Try the Waves Renaissance Comp for this (TDM).

· And the best advice is probably: Get just 1 excellent outboard compressor (a tube one would be a good choice so you can use it for warming up stuff too). My choice is the unbelievable Gyraf Gyratec X. Obviously when going D/A A/D with a DAW you need something good like PT HD, RME or Apogee etc.

[Charlie-O]

Having the right tools is key,

Without a good clock, things just never come together. Theres really very little seperation, like mixing with pillows on your monitors.

The right plugins, there so many companys that say they have the key to analog emulation, and say they emulate the greatest of vintage gear. DO NOT FALL INTO THIS TRAP! It can cost you thousnds in software and clients! Just becuase the plug makes it sound different, does not mean it makes it sound better. Choose wisely and use them sparingly, the more calculations a computer does on your audio, the more it will return "different" from the netherworld of digital. SOmetimes truncation is involved in some of this math, so USE YOUR EARS! Oh yeah, URS plugs are the shit, and compressor bank and mcdsp line are the chronic!

Heres a trick Ive posted befor( its has good results for me)

Mix your song ass you would normally but when bouncing to disk, set the bass and drums for a set of outputs(1&2), the hooks for outputs(3&4), The leasd vocals for outputs(5&6), The fx for outputs(7&8), and the Accomp. tracks for outputs (9&10). Then BOunce these 2 track stems out, and immport them back in on stereo tracks, the app. is now sumiing only 5 stereo tracks instead of who knows how many. If you dont have thes outputs you can always use the busses, or do one set at a time.

Im curiouse to see you results with this method....

[Lagerfeldt]

I have a comment on that last suggestion:

Each time you're bouncing, you're discarding bits since internally the DAW may be using e.g. 48 bits, and maximum for a bounce could be 24 bits.

Since dithering is not something you use until the final stage of mastering, you are infact creating 5-6 stems of individually truncated audio.

Adding these again later introduces a new process of audio manipulation, and another truncation process.

So whether this is a better option than simply bouncing everything together in the first place, I'm not so certain.

I think it would make for an interesting experiment though.

[Bob Olhsson]

Dithering needs to be done every time you reduce the number of bits stored or sent out of a port. The little bit of additinal noise will be less audible than the distortion caused by not dithering and will accumulate at half the rate the distortion would during subsuquent passes. On better workstations, dithering to 24 bits is handled automatically so submixes are not a problem.

[Charlie-O]

"On better workstations, dithering to 24 bits is handled automatically so submixes are not a problem."

A short list of these would be nice

[mchimes]

Originally posted by Lagerfeldt
I have a comment on that last suggestion:

Each time you're bouncing, you're discarding bits since internally the DAW may be using e.g. 48 bits, and maximum for a bounce could be 24 bits.

Since dithering is not something you use until the final stage of mastering, you are infact creating 5-6 stems of individually truncated audio.

Adding these again later introduces a new process of audio manipulation, and another truncation process.

So whether this is a better option than simply bouncing everything together in the first place, I'm not so certain.

I think it would make for an interesting experiment though.

There is a massive thread somehwre (Unicornation maybe) about this process. There are alot of people who are "hearing" a beneficial sonic difference between an internally bounced mix and that same mix stemmed out the digital I/O and back in (no conversion or analog summing mind you). Apparently those that are hearing a difference are attributing it to the "inferior" internal bounce mechanism.

When I tried it on a very small mix (3 guitar tracks and vocal) it was a dead null either way.

It would be nice to have some double blind data to back up or dismiss this current belief.

Bob . . . are you saying that even running stems out and back in the digital I/O, while not involving conversion, involves dithering?

I guess it would if you are processing tracks with 32 or 64 bit floating point (I'm using DP) math. It would have to be reduced to the throughput of the digital I/O, which is 24/44.1 in many cases. Is that correct?

So, depending on whether your DAW automatically dithers tracks assigned to a hardware output, this (the notion of improving sound quality by stemming out/in digital I/O) may or may not be true.

Hmmmm . . . . I see some experiments in my future.

Mike

[steins]

Originally posted by mchimes

. . . if you know where the "VERY small sweetspot is in PT"

Excuse me for being an idiot, but can someone explain what the ProTools "sweet spot" is? Is it about recording levels? Mix Bus levels? Monitoring? And why is it small? And most important: where is it?

BTW, Excellent post!

Stein Tore

[Lagerfeldt]

Originally posted by Bob Olhsson
Dithering needs to be done every time you reduce the number of bits stored or sent out of a port. The little bit of additinal noise will be less audible than the distortion caused by not dithering and will accumulate at half the rate the distortion would during subsuquent passes. On better workstations, dithering to 24 bits is handled automatically so submixes are not a problem.

If you apply dithering to all your files, you'll most likely end up with some massive veiling.

If we're talking POW-r or IDR, you'll get an obvious veil, and with UV22 it would be downright audio suicide.

Of course the veiling would be less on 24 bit files than 16 bits files (accumulative), but dithering is a process best reserved for the final master, even if you're using 24 bit files IMO.

-
Absolutely there's a big difference between summing in the DAW (depending on the DAW) and an external digital mixer (again depending on the mixer). I've tried and tested this myself, also tesing various DAWs. Both the old ProTools and Logic engines were basically doing horrible stuff to the mix. However, I'm yet to be convinced by this stemming process having an advantage today in e.g. Logic Pro. I shall look into it soon.

[mchimes]

Originally posted by charlieo3601
Having the right tools is key,

Without a good clock, things just never come together. Theres really very little seperation, like mixing with pillows on your monitors.

OK . . . maybe someone can help me here. I am a bit confused about what it means when someone says I "clocked" unit A to unit B.

"Without a good clock . . ."

I have a MOTU 828 mkII. It is a very feature laden $750 product that obviously can't have a great clock at that price point. Yet I read pretty consistently (from people like Dan Lavry) that any digital device (at least in theory) should perform better when clocked to it's own crystal (internally). Standing in the face of this "white paper" data is scores of people who have bought master clocks to "tighten" up their audio (buying them for mere sonic improvement . . . not necessarily needing them as a functional master clock for multiple devices.)

I guess what I'm confused about is this: Does a "better" clock influence just the monitoring (i.e. sounds smoother, deeper, detailed, etc.) or does it actaully enhance the recording/mix process "under the hood" of the computer/hardware relationship? In other words, is it making the data in the computer sound better or is it making better sounding data from start to finish?

So lets get specific. Lets say I buy an Apogee Big Ben and slave the MOTU 828 to it via word clock. I then begin to track and mix with it. At this point I would either hear a diifference or I wouldn't, right? Would this be the answer to "without a good clock, things just never come together?" Should things then "come together" now that a "good" clock is controlling things?

Now in that case, I am still using the D/As of the MOTU 828, but they apparently sound better because they are being stabalized buy a superior clock.

But Let's talk about another scenario. I want a Lavry Blue D/A. Let's say I interface a Lavry Blue to the S/PDIF output of my 828. So now I would have "better" conversion happening, right?

This is where I'm confused. The Lavry box just takes the digital S/PDIF signal and converts it to analog . . . but it utilizes it's own clock to do this. It doesn't even have Word Clock so it would be working off it's own crystal. But in this case the 828 wouldn't be Slaved to the Lavry would it? It would just pump out it's signal and the Lavry would make sense of it, convert it and spit it out to my amp. So in a sense, if I bought the Lavry, I am not really altering the clocking of the 828 am I? I am no doubt getting a better sound when monitoring, but would this apply to the actual audio data while tracking and mixing? or just to what I'm hearing on output. In this scenario, would the Lavry would be making the data ITB sound better, but would it make better sounding data on tracking/mixing?

Sorry if this is elementary . . . I just am just confused about this whole clocking thing.

Whew . . . don't break a tooth on any of that..

Mike

[stedel]

Originally posted by Lagerfeldt
· Yes, be very aware of what's going on in each plugin, it's quite easy to overload without it actually showing.

· Be aware of potential very high freq build-up (16kHz+), that will give a very tiresome sound. Perhaps use gentle sloped eq's to rolloff the very top on some processed tracks.

· Work with 24 bits and a higher samplerate, e.g. 96kHz to avoid unnecessary digital artifacts, especially when equalizing

· When mixing in a DAW you will normally need a bit more compression than in the analogue domain, since you don't have the luxury of tape compression or any other kind of automatic compression (tube stuff, mixer summing etc.). Obviously you can compress the individual tracks a bit more, or perhaps some light bus compression is the answer (use sensible A/R settings). We're talking compression both as in "gain reduction" and as in "glue", i.e. melting stuff together (drums+bass for instance).

· Another big issue is the low quality reverbs available for DAWs. Get something with a decent ER algorithm, that's magic for getting stuff to sound real or melt together in the mix.



· And the best advice is probably: Get just 1 excellent outboard compressor (a tube one would be a good choice so you can use it for warming up stuff too). My choice is the unbelievable Gyraf Gyratec X. Obviously when going D/A A/D with a DAW you need something good like PT HD, RME or Apogee etc.

Yep some very cool advice given in the posts so far.
In my humble experience, you can usually get an OK sounding product, but one that really cuts it for you is something else - although as always what's fine for some people doesn't work for others, so as a way of being a caveat please don't get pissed off at some of the more expensive conclusions and strategies I've used - a lot is my own prejudices.

A couple of pointers given here by lagerfeldt I would really like to second namely (erm IMHO):

1. Beg, borrow, steal - basically do what you have to to get the best outboard reverb you possibly can. I tend to agree re reverb plugins. Ideally if I was you I'd try and seek out a second-hand Reverb 6000 (sorry yeah I know not cheap) but you get amazing sounds, amazing flexibility and great digital patching capabilities (Analog to didgital to analog back to digital then back to analog and digital and then possibly one more final analog for mastering worries me). Might sound over the top advice but inferior reverb can be like bad internal summing and AD/DA conversion, namely it can really "collapse" the sense of space in your sound. Apparently TC Electronic are bringing some of their 6000 stuff out on their Powercore platform - might be worth sussing it out.
2.Yep (at least) one really good stereo Outboard compression unit. Avalon is cool, I use a Manley SLAM which also covers another point namely:
3. The best DA/AD converters for getting your stereo mix out. I know there's a lot of "Travelling Medicine Show Hair Tonic" talk about converters but the final stage is critical (IMHO).

4. (less expensive) Plug Ins - Sony EQ Bundle, Sony Inflator and/or and Cranesongs Phoenix can really help things along. Can help with the high frequency build up that Lagerfeldt mentions - something I've noticed myself, although this could be inherent in the AD/DA unit I use (a 192 i/o).

5. If you do use any external gear, (eg reverb) then a good Master Clock is necessary eg. Apogee Big Ben.

6. The point Knastratt makes namely "Having powerful tools for corrections doesn't mean that basic recording theory is surpassed" is also a key point. Yep take your time mic selection, placement, matching up with a suitable mic pre etc etc etc. (But you know this right?).

Anyway all this is only MHO, but coming from an analog background I spent a lot of time setting up my (still incomplete but ran out of money for now) system!

Cool thread
Kind regards

[mchimes]

Thanks for a thoughtful reply Stedel! Good stuff.

[C.Lambrechts]

great stuff allready said in here.

for me, the secret of a great sound starts @ a great source. I keep running into too many people who are trying to 'fix it in the mix' and then blame it on the mix bus or the plugins or the this or the that. At least in 7/10 of those cases I run into it went wrong right @ the start.

Once it's in there :

- right choise of plugins / outboard combination.

- When I mix I tend to go by a 'less is more' rule for myself. Often you hear things like 'the mix doesn't breath' .... 'it sounds squashed' etc etc ...

Let me put it this way : if a certain track doesn't sound loud enough you can do 2 things : turn that track up or turn the rest down. Especially in a digital world ... the first choice will eventually get you into trouble ... causing some of the previously mentioned expressions.

oh , ... and go by what you hear , not what you see on the screen. A great mix or sound doesn't necessarily look good.

[Leo Goler]

Originally posted by Bob Olhsson
Dithering needs to be done every time you reduce the number of bits stored or sent out of a port. The little bit of additinal noise will be less audible than the distortion caused by not dithering and will accumulate at half the rate the distortion would during subsuquent passes. On better workstations, dithering to 24 bits is handled automatically so submixes are not a problem.

Better workstations should employ higher bit depth for those calculations. 24bit is not enough for very complex sound design work as trunkate artifacts seem to come too loud at times.

[Blast9]

Both the old ProTools and Logic engines were basically doing horrible stuff to the mix. However, I'm yet to be convinced by this stemming process having an advantage today in e.g. Logic Pro. I shall look into it soon.

Lagerfeldt...

What specifically did you find was happening, and was it a consistent problem?

[Charlie-O]

"Without a good clock . . ."



Perhaps an explanation:


My humble studio was based around a Digi002 (wich I love). I would convert everything with this box. I then gathered the balls to buy a Apogee trak2. From then on everything in the studio was clocked with this, including my digi002. ALll my tracking if possible was A/D through the trak 2. Now when listening to stuff with now mix befor the trak 2, it all sounds smeared, with very little seperation. Everything later, has a seration, width when panning is applied I dont have to go to far out of the center to "hear" the panning. All together its made things alot easier.

On a side note, befor a nice set of converters I would recomend a good pre!

A mistake I have made so many times is with compression, and plug-ins. ALot of "great"mixes started with the performance and very little processing. Just becuase the plug list is a mile long, doesn't mean that every track should be processed.

[rainsinvelvet]

Originally posted by charlieo3601
"Without a good clock . . ."

A mistake I have made so many times is with compression, and plug-ins. ALot of "great"mixes started with the performance and very little processing. Just becuase the plug list is a mile long, doesn't mean that every track should be processed.

I'm finding this true on a daily basis.. Good point!


ERic

[Lagerfeldt]

Originally posted by Blast9
Lagerfeldt...

What specifically did you find was happening, and was it a consistent problem?

I'd like to answer this, although it is totally irrelevant now, since virtually nobody uses the old Logic (pre 5.x) or ProTools anymore. On a side note: apart from the 888's I also had the 882 interfaces from digidesign, and they sounded horribly dull.

Back to the problem: It was 100% consistent, and an audio engine (software) problem: You would experience a noticeable reduction in the stereo perception and some high frequency loss when bouncing (maybe some kind of phase problem, sounded to me like a 14kHz hi cut).

= more mono and dull

It is both audible and measurable, and something that's been addressed in the newer versions, so it's no longer an issue.

At that point in time I actually preferred going to DAT (!) instead of bouncing my final mixes, later I switched to summing in a digital mixer. Now I'm back to bouncing.

[RKrizman]

Originally posted by djui5
[B]Personally I don't think PT's has a sweet spot. I think each mix has a sweet spot..and once you find it...well..that's great.

Good point Randy, I never thought of it that way.

-R

[BrianBrian]

1.) Ditch ProFoolz
2.) Buy Samplitude Pro
3.) Buy all the Voxengo plugins
4.) Buy 2 or more UAD-1 cards w/ plugins
5.) Investigate TriTone Digital plugins

Ok with all this stuff you should be able to make mixes that compete with analog ones as long as you have skills.

[syra]

I dont really think there are secrets...I used to think like that but the longer I work on my mixes and develop my ear the better the results. I haven't come to any conclusion ITB which is extremely different from mixing on analog besides you can't push to red

It comes down to one thing for me. Being able to hear in your head exactly what you want something to sound like (from your kick to the whole mix...)...and then knowing your tools/toys well enough to get it.

[max cooper]

I often use a hallway in my building to set up a reverb send. I find that it grounds the ITB mix in the real world.

Also, minding the source, as has been mentioned here seems to be all important. I think there's not substitute for being meticulous.

My choice is the unbelievable Gyraf Gyratec X

Haven't heard this unit, but it sure does look nice!

[Lagerfeldt]

Haven't heard this unit, but it sure does look nice! [/B]

Yes, it's the one piece of gear I'd take with me on a desert island.

A/B tested it for a month against the Manley Vari-Mu, and while they're not exactly the same, the Gyratec X Vari-Mu did have the edge by a decent margin.

Also it's one of those very rare pieces of equipment that is very forgiving, i.e. it works great with a variety of settings on the same material.

The tubes in the input stage add some amazing harmonics, and really is 21st century magic. Also you'll be able to churn out some amazing level with this thing, and the way it handles peaks is much more reliable than the Manley Vari-Mu, so I can go at least 2dB hotter into A/D than the Manley.

Because of the way the input and output stages are designed, it can be used to color the sound in various ways. Gaining heavily on the input (on a normal mix, anything over 1 o' clock) will lift harmonics and give that true tube sparkle sound, and gaining on the output above 2 o' clock will slowly begin to saturate the output via (a different kind) of tubes and the trafo, giving a more squashing limiter type sound. This last feature is useful if you need something extra hot and you're willing to sacrifice a bit of fidelity.

Oh, and it doesn't hurt it looks sexy too!

[mchimes]

Originally posted by syra

It comes down to one thing for me. Being able to hear in your head exactly what you want something to sound like (from your kick to the whole mix...)...and then knowing your tools/toys well enough to get it.

Very, very true! In "Behind The Glass," most of the top producers interviewed there said the same thing in one way or another.

But . . . and for others who are responding to the idea of "secrets" . . . it was just a thread title to get traffic.

What I'm really asking (and was elaborated on in the original post) is:

"What do YOU do to get your ITB mixes sounding "good?"

Of course there's no magic bullet . . . so do you have any specific treatment or discovery for addressing an audio issue ITB?

Mike