2buss comparison: Fatso vs. Oxford Dynamics

[minister]

Originally Posted by Paul Frindle

This is particularly important with todays 'pro-sumer' converters that very often have compromised analogue front end circuits..

this is a good point. we often forget that the sound of a converter also has to do with the analog portion of the A/D. the best convertors have the best analog components. this is also true of D/A.

the beauty of working with digital this way is that the mastering engineer and make it LOUDER and CLEANER without all that harshness.

[Paul Frindle]

Originally Posted by minister

the beauty of working with digital this way is that the mastering engineer and make it LOUDER and CLEANER without all that harshness.

Err - hopefully :-) But of course however good you make your mix, the mastering engineer has the power to mess it up - but that's another discussion..

My suggestion from experience is that the following is a workable gain staging for a mix :

Record peaking at -6dB (this reads -6dBFS when played back on the DAW).

Lose another 12dB at the head of the mix channel on playback using a trim plug or by turning down the input gain of the first plug-in (so the level is now -18dBFS).

Do all your EQing, processing whatever, tryng to make sure nothing gets hotter than around -6dBFS between plugs.

Mix the whole thing aiming to keep the output mix at -6dB or below before the final buss processing.

Then get the level back on the final mix process (i.e. busscomp or limiter) to aim for around -3dBFS.

If you have a final busscomp/limiter that shows you reconstructed levels (i.e. those that will come out of the DAC) you can push the level to just below the red light on the meter on the plug itself. This may allow you to go higher than -3dBFS depending on the programme.
If you have a final limiter that will correct for intersample peaks you can just go for it without worrying, knowing that the max legitimate level will be reached dynamically. But please bear in mind that by aiming at full modulation you now are effectively level mastering it yourself - as you are removing some options for the mastering engineer downline..


O.k - now this is a personal view from experience and I don't want to start controversy. But if it were me I would try my damnest to get all my tracks sounding correct together as a whole for the album - myself - and present a properly dithered full intended CD level 16 bit file to the mastering engineer as well as the actual mix master (at -3dBFS). This is to avoid the mastering engineer having to mess with it for continuity. Messing with EQ and compression after a maximised mix risks re-creating intersample overs and distortion you have so carefully avoided!! It is possible that some mastering engineers may not be as savvy in these matters as you think, (almost every CD I have heard in the last couple of years has been maximised to error either in the mix or mastering phases), so the aim is to get your legitimately optimised 16 bit file onto the CD if at all possible...

If possible, I would then be present at the mastering session to make sure that the programme was not subjected to further 'tweeking and messing' that I and the producer did not want.. If you get there and the mixes do not gel together, or sound too hot, toppy, or whatever in the mastering suite, the best thing technically speaking is for you to go back and tweek them yourself - i.e. remembering that any additional EQ should be done in the signal chain BEFORE your final busscomp/limiter.

I know that this is not always possible (or even allowed) but you can see where I am coming from :-)

[vtone]

Originally Posted by Paul Frindle

Err - hopefully :-) But of course however good you make your mix, the mastering engineer has the power to mess it up - but that's another discussion..

My suggestion from experience is that the following is a workable gain staging for a mix :

Record peaking at -6dB (this reads -6dBFS when played back on the DAW).

Lose another 12dB at the head of the mix channel on playback using a trim plug or by turning down the input gain of the first plug-in (so the level is now -18dBFS).

Do all your EQing, processing whatever, tryng to make sure nothing gets hotter than around -6dBFS between plugs.

Mix the whole thing aiming to keep the output mix at -6dB or below before the final buss processing.

Then get the level back on the final mix process (i.e. busscomp or limiter) to aim for around -3dBFS.

If you have a final busscomp/limiter that shows you reconstructed levels (i.e. those that will come out of the DAC) you can push the level to just below the red light on the meter on the plug itself. This may allow you to go higher than -3dBFS depending on the programme.
If you have a final limiter that will correct for intersample peaks you can just go for it without worrying, knowing that the max legitimate level will be reached dynamically. But please bear in mind that by aiming at full modulation you now are effectively level mastering it yourself - as you are removing some options for the mastering engineer downline..


O.k - now this is a personal view from experience and I don't want to start controversy. But if it were me I would try my damnest to get all my tracks sounding correct together as a whole for the album - myself - and present a properly dithered full intended CD level 16 bit file to the mastering engineer as well as the actual mix master (at -3dBFS). This is to avoid the mastering engineer having to mess with it for continuity. Messing with EQ and compression after a maximised mix risks re-creating intersample overs and distortion you have so carefully avoided!! It is possible that some mastering engineers may not be as savvy in these matters as you think, (almost every CD I have heard in the last couple of years has been maximised to error either in the mix or mastering phases), so the aim is to get your legitimately optimised 16 bit file onto the CD if at all possible...

If possible, I would then be present at the mastering session to make sure that the programme was not subjected to further 'tweeking and messing' that I and the producer did not want.. If you get there and the mixes do not gel together, or sound too hot, toppy, or whatever in the mastering suite, the best thing technically speaking is for you to go back and tweek them yourself - i.e. remembering that any additional EQ should be done in the signal chain BEFORE your final busscomp/limiter.

I know that this is not always possible (or even allowed) but you can see where I am coming from :-)

I have been following this thread with great interest lately and it seems to have stopped. I especially appreciate your insight, Paul.

I think the key words here are: INTERSAMPLE PEAKS

it's worth doing a search if you aren't sure what they are...

[Lexicondonn]

I have two distressors on order. Are these cool to use on 2 buss master mix, or do I need something else? I have one 1176. I would really just use it to stop OVERS, not so much to really compress. I was thinking about a pair of the UA LA3A.s as a second.

We record all kinds of music. but mostly pop rock

[minister]

as excellent as the Distressor is -- and it does work on the 2-Bus -- i would not waste it as a box that merely 'prevents over'. Nuke is brickwall and will prevent signals from passing your ceiling. but it is really for room mics and efects. if you don't know what you are doing, you can really eff up your signal. brickwall limiting is best left for the Mastering Engineer. compressing on the 2-Bus is fine if that is what you need to achieve your sound.

if you are having problems with overs, maybe try tuning down your signals.

my opinion.

[Lexicondonn]

Dam! Sorry, I posted on the wrong thread. I was reading this whole thing, and falling asleep. Not that it was boring or anything.

Well my question was really asking if the Distressors were good on 2 buss master mix as well as the other stuff their great on?

I have no troubles with my levels and I can't afford to go to a mastering house at leasst for my songs anyway, so I kind of do my own pre mastering if you will. Which sounds fine to me. I really don't care about getting that in your face sound, but I do try to stay at around 0.

If it sounds great, isn't it?

[minister]

hey, if it SOUNDS good, it IS good.




...sometimes i have to master or finish my stuff too...

[u b k]

Originally Posted by Lexicondonn

If it sounds great, isn't it?

depends on whose ears we're using.


gregoire
del
ubk
.

[Paul Frindle]

Originally Posted by Lexicondonn

Dam! Sorry, I posted on the wrong thread. I was reading this whole thing, and falling asleep. Not that it was boring or anything.

Well my question was really asking if the Distressors were good on 2 buss master mix as well as the other stuff their great on?

I have no troubles with my levels and I can't afford to go to a mastering house at leasst for my songs anyway, so I kind of do my own pre mastering if you will. Which sounds fine to me. I really don't care about getting that in your face sound, but I do try to stay at around 0.

If it sounds great, isn't it?

I understand this scenario well, it's a very powerful tool to be able to do some mastering on demo and proof of concept stuff, where the cost of professional mastering would be inappropriately expensive. The other potential plus point is that you can control the sound yourself and optimise your mix for the final result.

This is a complex one because in many ways comp/limiters fall into several categories and some straddle many categories, i.e. those that produce a 'sound style' and those that attempt to be accurate and unintrusive, those intended for instrument and vocal and those intended for complex output programme - and some that can do all these, provided you are proficient in setting them up properly.

But for simplicity and immediacy's sake, I think you would be better off opting for something that was designed with the accuracy and finesse intended for programme mastering, than potentially wasting an effect like the distressor on a task it was not specifically designed to do?

If you are running in the digital domain a purpose designed limiter plug-in is more suitable and less costly.

[jdjustice]

Originally Posted by Paul Frindle

But for simplicity and immediacy's sake, I think you would be better off opting for something that was designed with the accuracy and finesse intended for programme mastering, than potentially wasting an effect like the distressor on a task it was not specifically designed to do?

Originally Posted by Paul Frindle

If you are running in the digital domain a purpose designed limiter plug-in is more suitable and less costly.

...and there is not a better limiter plugin, IMHO, than the Oxford!!


~j.d.

[Paul Frindle]

Originally Posted by jdjustice

...and there is not a better limiter plugin, IMHO, than the Oxford!!


~j.d.

I'm honoured that you use and like the limiter, it was designed for this specific purpose. It is more powerful than many people expect. However it is by no means the only one and this kind of app is definitely worth looking into.

[84K]

I like "B" a lot better and I guess it is the fatso. Now I will go back and read to see if it was revealed yet. I though "A" sounded way less.... fat. So B must be the fatso. B is much better.

[Paul Frindle]

Originally Posted by WildCowboys

Slutz!
I just set up a little comparison I wanted to do for a long time and thought if someone here is interested I can post the resulting files!

I wanted to check out if it pays to go back to analog to do 2buss compression and then back in the DAW for limiting, or if I can get equally good sounding masters with just using plugins...

so here is the setup:
a rock´n´roll song mixed ITB then going through Sony Oxford Dynamics (compression 2:1 attack around 30ms, release around 80ms, GR 3-6dB) and alternatively into my Fatso (BussComp GR 3-6dB, no tranny, warmth set to 3 resulting in warmth-GR of about 2-4 on the meter, hitting the saturation pretty good so that both "0VU" and the "Pinned" were lit up most of the time)

then both mixes were level-matched and went through the exact same mastering eq and limiter setting.

here are the files, encoded as 320kbps MP3 files:

http://www.wildcowboys.com/rr_A.mp3
http://www.wildcowboys.com/rr_B.mp3

Check them out and see if you hear a difference and which one you like better!

My personal findings:
The difference is subtle and similar to my experience with OTB summing. The fatso version sounds a little bit more pleasing to my ear, a little more "like a record" (tm charles dye, haha), a little more "together" and "forward" - just the right thing for rock. I guess on R&B and hiphop I would not go the fatso route, it is not what those styles call for, but for rock it is worth the extra effort.


Some words about the song, it is called "Rockin´and Rollin´In The USA" by the group Tunesmith, a bit freaky and slightly retro, album will be released this year. ..

So tell me what you guys think about the differences or just tell me what you think of the song, production, mix, whatever!
Rock on,
Pat

To go back to the start of this thread; what about for a bit of fun (when I can find time at home) I try to do a mastering job on this with both the Oxford Dyn and the limiter? I can do one that as closely as possible matches your Fatso track and another which is as I would do it if it were my track?

The differences between Oxford and the Fatso tracks are quite dramatic. The Fatso track has a load of stuff going on at the top end which is splashing away - maybe because of the limiter that comes after(?) the comps - maybe because of the extra DAC cycle before the comp?

I've had fun doing this kind of thing on here before :-)

You would need to post the uncompressed track somewhere though.

[WunderBro Flo]

yeah I can post the uncompressed one. just gimme some time i am treally busy these days...gotta dig through the folders...

btw I never tried the Oxford limiter as a brickwall-type for mastering. is it intended for that? I always had the feeling it is more similar to an analog limiter...which sounds quite different than todays brickwall plugs.

Rock on!
Pat

[Paul Frindle]

Originally Posted by WildCowboys

yeah I can post the uncompressed one. just gimme some time i am treally busy these days...gotta dig through the folders...

btw I never tried the Oxford limiter as a brickwall-type for mastering. is it intended for that? I always had the feeling it is more similar to an analog limiter...which sounds quite different than todays brickwall plugs.

Rock on!
Pat

O.k. - if and when you may get time :-)

The oxford really is designed to be a one stop 'finishing tool', complete with intersample peak correction and dithering. You can do other stuff with it too, but that was the intention.

It can sound different from other limiters as it has novel processing after the gain limiter that acts to allow you to hear peaks and attacks without causing higher peak levels. This means that you can produce more punch, power and volume than a conventional app - or you can produce cleaner results at the same volume levels than with other apps.

The enhance (post processing) can be turned down to minimum so it acts like a conventional high quality limiter - if that's what is preferred.

[WunderBro Flo]

thanks paul, interesting!

i quickly tested it and found that it works almost like my favourite brickwall plugin but it still has some behaviour of a "normal" limiter. When making the attack short enough to prevent overs (below 0,010) and the release very short as well, I can get just 1dB less loud before I dislike it than with my usual brickwall. the differences are that with my old limiter I get more clarity and impact in the transients while with the Oxford Limiter I get more "compression-like" sound. It would totally depend on the source to say which one I prefer.

Would it be possible to use my old limiter up to the desired level and then run through the Oxford limiter to correct eventual intersample-peaks that the other plugin caused? Or is that not possible, meaning when a limiter caused the intersample-peaks then they are there and that´s it?

Thanks!
Pat

[Paul Frindle]

Originally Posted by WildCowboys

thanks paul, interesting!

i quickly tested it and found that it works almost like my favourite brickwall plugin but it still has some behaviour of a "normal" limiter. When making the attack short enough to prevent overs (below 0,010) and the release very short as well, I can get just 1dB less loud before I dislike it than with my usual brickwall. the differences are that with my old limiter I get more clarity and impact in the transients while with the Oxford Limiter I get more "compression-like" sound. It would totally depend on the source to say which one I prefer.

Would it be possible to use my old limiter up to the desired level and then run through the Oxford limiter to correct eventual intersample-peaks that the other plugin caused? Or is that not possible, meaning when a limiter caused the intersample-peaks then they are there and that´s it?

Thanks!
Pat

You need to set a slower attack and release so the input limiter overshoots (so you get some dynamics and punch) - then let the enhancer handle the peak levels for you. If you hit safe mode it will do this automatically and work on all peaks.

Wind up the input gain to get more compression sound.

If you get the occasional red light, wind the output down around -0.1dB

If this sounds ok to you, you can now wind the enhance up and produce programme that is around 3dB louder than a fully limited signal..

Give it a try :-)

[WunderBro Flo]

do you mean the warmth-control when you say enhancer section?

I have the PowerCore version, does it also have the "safe" mode? I cannot remember seeing that button ever...

I usually use the Inflator to get around 3dB of saturation so my brickwall limiter does not have to pump as much....guess that´s also your approach here, right?

thanks,
P.

[Paul Frindle]

Originally Posted by WildCowboys

do you mean the warmth-control when you say enhancer section?

I have the PowerCore version, does it also have the "safe" mode? I cannot remember seeing that button ever...

I usually use the Inflator to get around 3dB of saturation so my brickwall limiter does not have to pump as much....guess that´s also your approach here, right?

thanks,
P.

I'm sorry - I'm getting totally confused here - because we ended up talking about the limiter I figured you were talking about that, rather than the limiter section of the dynamics :-(


Completely ignor what I said in the last post :-(

I'm so sorry...

-----------------------------------------------------------------------

O.k - to talk about the dynamics now;

The thing to do here is to compress using a relatively gentle attack and release in the comp section to get the 'feel' of the sound you want. A ratio of about 4:1 or above and a 5dB soft knee might be appropriate, with threshold set around -3dB. This will create overshoots so keep the gain makeup down just a bit at first to avoid hearing clipping.

Then go to the limiter and set a threshold of around 0dBFS (flat out) with the moderate attack and release times initially around mid way on the controls (default). Decrease the release to get more 'grit' and loudness if needed.

Wind the warmth control to flat out - this will prevent any overloads however hard you hit it.

Then go back to the comp section and wind up the gain makeup control to get the required amount of compression and limiting..

[Paul Frindle]

Originally Posted by WildCowboys

thanks paul, interesting!

Would it be possible to use my old limiter up to the desired level and then run through the Oxford limiter to correct eventual intersample-peaks that the other plugin caused? Or is that not possible, meaning when a limiter caused the intersample-peaks then they are there and that´s it?

Thanks!
Pat

Sorry missed this question.. This time we are talking about the Oxford Limiter and not the Dynamics...?

Yes that would work. It does not matter how the intersample peaks are generated - because they only appear in the DAC reconstruction.

To avoid getting any of the Oxford limiter sound and just get the intersample peak stuff, you would need to:

- turn off the Safe Mode,
- set Enhance for minimum (0%)
- set Soft Knee to min
- set Attack to max and Release to min

- and set the Input Gain to 0dB. (making sure that no action occurs on the Gain Reduction meter).

Then select Auto Comp as usual - and optionally switch the meter to Recon if you want to see the peaks it is dealing with etc..

It will then compensate for any intersample peaks in the signal - however they were generated.

I hope this is helpful.. :-)

[beyarecords]

Hi Paul,

Originally Posted by Paul Frindle

Record peaking at -6dB (this reads -6dBFS when played back on the DAW).

Track with a peak meter reading of -6dB. Check.

Originally Posted by Paul Frindle

Lose another 12dB at the head of the mix channel on playback using a trim plug or by turning down the input gain of the first plug-in (so the level is now -18dBFS).

Ensure that -18dB hits the first plugin on the 2buss and remains at such a level between each subsequent plug on the 2buss until it hits the last plug, comp/limiter, at which point ensuring that a peak meter reading of -3dB is attained. Check.

I presume that loosing a further 12dB on the 2buss would not apply should Sony Oxford plugs, only, be incorporated in the project workflow and a decent intersample meter incorporated on the 2buss?

[peeder]

Someone else resurrected this thread so don't point at me...

Originally Posted by Paul Frindle

Err - hopefully :-) But of course however good you make your mix, the mastering engineer has the power to mess it up - but that's another discussion..

My suggestion from experience is that the following is a workable gain staging for a mix :

Record peaking at -6dB (this reads -6dBFS when played back on the DAW).

Lose another 12dB at the head of the mix channel on playback using a trim plug or by turning down the input gain of the first plug-in (so the level is now -18dBFS).

Do all your EQing, processing whatever, tryng to make sure nothing gets hotter than around -6dBFS between plugs.

Mix the whole thing aiming to keep the output mix at -6dB or below before the final buss processing.

Then get the level back on the final mix process (i.e. busscomp or limiter) to aim for around -3dBFS.

If you have a final busscomp/limiter that shows you reconstructed levels (i.e. those that will come out of the DAC) you can push the level to just below the red light on the meter on the plug itself. This may allow you to go higher than -3dBFS depending on the programme.
If you have a final limiter that will correct for intersample peaks you can just go for it without worrying, knowing that the max legitimate level will be reached dynamically. But please bear in mind that by aiming at full modulation you now are effectively level mastering it yourself - as you are removing some options for the mastering engineer downline..

If this is such a good idea, why isn't it built-in to the architecture of the DAWs and converters? In fact my understanding is that it often is. But I don't know for sure. Perhaps they are being slaves to maximizing published specs rather than usability.

So iow the 24-bit DAW should have headroom to cover intersample peaks and perhaps 12 DB of gain for plugs etc. I think Pro Tools has this, doesn't it? LE is 32-bit float, and there's no reason that format couldn't have extra headroom, is there? It can just be shifted back up transparently going out.

Similarly a DAC should handle 0dbfs output fine, even if it has to use the same automatic adjustment technique. Don't they? With so much commercial music being run out at 0dbfs these days, I have to imagine the DAC vendors handle intersample peaks just fine.

Originally Posted by Paul Frindle

O.k - now this is a personal view from experience and I don't want to start controversy. But if it were me I would try my damnest to get all my tracks sounding correct together as a whole for the album - myself - and present a properly dithered full intended CD level 16 bit file to the mastering engineer as well as the actual mix master (at -3dBFS). This is to avoid the mastering engineer having to mess with it for continuity. Messing with EQ and compression after a maximised mix risks re-creating intersample overs and distortion you have so carefully avoided!! It is possible that some mastering engineers may not be as savvy in these matters as you think, (almost every CD I have heard in the last couple of years has been maximised to error either in the mix or mastering phases), so the aim is to get your legitimately optimised 16 bit file onto the CD if at all possible...

If possible, I would then be present at the mastering session to make sure that the programme was not subjected to further 'tweeking and messing' that I and the producer did not want.. If you get there and the mixes do not gel together, or sound too hot, toppy, or whatever in the mastering suite, the best thing technically speaking is for you to go back and tweek them yourself - i.e. remembering that any additional EQ should be done in the signal chain BEFORE your final busscomp/limiter.

I know that this is not always possible (or even allowed) but you can see where I am coming from :-)

Now I think your general aim is correct, to reduce what you consider "error." But what I understand from the actual pop mastering people is that, just like in prior eras where distortion and saturation went from being "error" to being "mojo", the sound of ADC "overs" dazzling transients with tons of harmonic distortion (via squared tops or similar) is considered a good thing! "If it sounds good, it is good" or something like that.

So a mastering engineer on the front lines would probably cross her eyes at this suggestion...on one hand, you're calling her "mojo" "error". On the other, you're just using her as an audio consultant and listening room. That isn't the way it works...of course you acknowledged that plus the fact the suggestion was controversial...just pointing it out.

Very much appreicate the ongoing clarifications about digital. May I suggest if you are weary of these discussions that you post a FAQ written in the clearest language you can (your rundown earlier of ADC DSP and DAC was a good start on one) and just post a link to it when the issue crops up again. These threads are great but have lots of tangents and an odd flow, and books are great but expensive and time consuming...so referring to those isn't as useful.