Great post skip, i found this my self a while back i am referencing 0VU = -18 DBFS, proper gain stagging is indeed a key to good sounding mixes in the digital domain, i also agree 100% with DrBill about using LPF when mixing in digital, that will lead to smoother top end and warmer tracks, tried and true
but some pieces of the puzzel are still missing, analog gear has color. harmonic distorsion, tube saturation ... etc
Just to add my 2 cents, I often see outside "hot" tracks coming in needing 6-10 dB of gain reduction to be in a comfortable range for my board.
On my own sessions the tracking is monitored on analog meters so I am setting levels that my system likes without much reliance on the DAW meters. I do use the PT meters for displaying "overs".
You folks will think I am smoking crack but here is what I will normally do. I put the trim plug in first followed by either the SSL channel or the URS strip Pro. Then with the trim plugin up on the encoders Of My Icon D-dontrol. I put the fader at around 0. Then I adjust the trim plugin until the track fits into the mix fairly well. Now just EQ and compress to taste and do fine adjusts with the fader. This always works well and I never have a headroom issue. Plus all my fader are at the sweet spot. I know I am smoking Crack.
Does anyone else feel like the sound changes based on how hard you hit the converters? Sometimes I prefer the sound of a hot signal and sometimes not, but to me there is a difference. I have no scientific proof. I also tend to prefer to have very slight compression going into digital as opposed to analog. I think partly becuase I prefer the less dynamic range you hve with tape.
You folks will think I am smoking crack but here is what I will normally do. I put the trim plug in first followed by either the SSL channel or the URS strip Pro. Then with the trim plugin up on the encoders Of My Icon D-dontrol. I put the fader at around 0. Then I adjust the trim plugin until the track fits into the mix fairly well. Now just EQ and compress to taste and do fine adjusts with the fader. This always works well and I never have a headroom issue. Plus all my fader are at the sweet spot. I know I am smoking Crack.
Skip, just out of curiousity, do you find yourself using the E-Channel or G-Channel more for various tasks?
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Skip, just out of curiousity, do you find yourself using the E-Channel or G-Channel more for various tasks?
E channel for drums, G channel for BGV/Keys/ Lead. Mostly I have been using the URS channel Strip Pro. When ITB. That always gives me what I need for just about anything. I do the same thing when I am Mixing on my G+ in the the Grey Room. E for Drums G for Vox. I still Love Mixing on the Big SSL. Its very Very Big Sounding. However Not all Clients can work with the restrictions of an Analog Desk. Its very nice to have all types of room In the Facility.
The Reason Most ITB mixes don’t Sound as good as Analog mixes. This is a repost from another thread. Hope you find it usefull.
Ok, I'm going to try and give you An ITB education, as my over 24 years has taught me. Here is what I try and teach to students. I'll try and keep the math to a minimum.
First, I own a high end analog setup's Via an SSL 4K with 1/2' 2 Track YADA YADA, ICON with Killer OB FX And classic Compressors, YADA YADA/ Hybrid Setup Via AWS 900w/ 24 Channels Of Xlogic Killer OTB FX and Comps YADA YADA. Point is not to impress, or brag in any way, but to let you know everyday I work on a verity of systems. This has led me to The Following conclusion.
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.
So in dumbass terms, you insert a plugin that will trim down 20 db or if its running at -20 db leave it there?
And video would be great.
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"Mixing in the box requires thinking outside of it - mixing analog requires inspiration..."
Everything in the original post is true, but it isn't why itb mixes don't sound as good (warm/deep/wide/clear/smooth/alive) as analog mixes.
Gregory Scott - ubk
.
Gregory, I was simply trying to point out what Most ITB mixes sound horrible. I am attempting (perhaps Not Very Well) to show that analog sorta forces to work with in the described voltage range. Analog VS digital was not what I wanted to start here however I based on Decades of experience I believe That ITB can sound as good as Analog Mixes. Perhaps Analog has a slight edge over All Dig, however With the right person can be very Very Good.
You folks will think I am smoking crack but here is what I will normally do. I put the trim plug in first followed by either the SSL channel or the URS strip Pro. Then with the trim plugin up on the encoders Of My Icon D-dontrol. I put the fader at around 0. Then I adjust the trim plugin until the track fits into the mix fairly well. Now just EQ and compress to taste and do fine adjusts with the fader. This always works well and I never have a headroom issue. Plus all my fader are at the sweet spot. I know I am smoking Crack.
Crack or no crack, that makes a lot of sense to me!
"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."
actually 0vu or +4 is 1.228 volts...since we are all being so technical about it....stike
On my board's old style moving needle meters that difference in voltage would amount to a deflection of less than one RCH (to be technical) so it would be impossible to discern. Also technically known as "good enough for rock and roll"
One of the amazing things in thinking about these threads is that tracking to DAW's has pretty much been accepted as a professional standard way of working. And some of the top mastering engineers & mastering houses are now doing mastering totally ITB and i hear no one complaining about those masters.
So its basically the part in between ie; the mixing where it seems people have the most issues. But what surprises me is that before DAW's became an option for mixing people were still complaining about their mixes lacking something per say even in the days of an all analog studio environment. First it was that they had to move the mix from an all tube console to a board with transformers/transisitors. Then it was that the board with the transformers colored & slowed the signal too much so moving it to a console with no transformers was the ticket. When that wore off and transformer based consoles made a come back, it was that they had to move the mix from a Neve or API to an SSL, so the SSL must be lacking. After that moving the mix from an SSL to an even more cleaner surface for instant recall like a Euphonix or the many digital mixers after took away the "mojo" .
And now we've come full circle.
So basically has anything really changed? The only thing is that their are more people complaining because there are more people trying to engineer and with the internet now we get to hear about it every second of the day.
But through all the bitching great sounding records have been mixed on every format.
So bottom line folks...maybe the problem is not what's in front of the eyes but more likely what's behind them.
On my board's old style moving needle meters that difference in voltage would amount to a deflection of less than one RCH (to be technical) so it would be impossible to discern. Also technically known as "good enough for rock and roll"
I know, that it's only Rock n' Roll........but.....I like it.....
If I've recorded everything at around 0 dBVu - which corresponds to 16 dBFS (with my converter, according to Lynx support), then any plug in on any given track would still be seeing a good level, right? If all those tracks combine together and peak the mix buss up near zero, that's OK, right? In their manual Steinberg has stated that causing overs on any given channel (except, obviously, the mix buss) do not impact quality whatsoever - I believe this may have something to do with floating point computations. However, if I were to say put the waves MB compressor across a guitar buss and it was showing a fairly constant level of -10, you're basically saying it would sound better to trim the group down and make up the gain with the output of the plug-in or the fader? I have always been careful not to clip plug-in inputs, but it seems pretty odd that plug-ins would be designed to function like analog equipment in this respect, unless they were trying to emulate analog distortion, e.g the UAD Fatso - otherwise what would be the point of designing software that would gradually degrade your sound as you pushed further past 0dBVU? Or does this have something more to do with intersample peaks?
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Quote:
Originally Posted by u b k
.....Along with a link to one or three of their own mixes that demonstrate what the poster is claiming. Otherwise, they're just blowin' smoke out their @ss and asking me to breathe deep.
If I've recorded everything at around 0 dBVu - which corresponds to 16 dBFS (with my converter, according to Lynx support), then any plug in on any given track would still be seeing a good level, right? If all those tracks combine together and peak the mix buss up near zero, that's OK, right? In their manual Steinberg has stated that causing overs on any given channel (except, obviously, the mix buss) do not impact quality whatsoever - I believe this may have something to do with floating point computations. However, if I were to say put the waves MB compressor across a guitar buss and it was showing a fairly constant level of -10, you're basically saying it would sound better to trim the group down and make up the gain with the output of the plug-in or the fader? I have always been careful not to clip plug-in inputs, but it seems pretty odd that plug-ins would be designed to function like analog equipment in this respect, unless they were trying to emulate analog distortion, e.g the UAD Fatso - otherwise what would be the point of designing software that would gradually degrade your sound as you pushed further past 0dBVU? Or does this have something more to do with intersample peaks?
-20 through -16 dbfs seems to be the range that most companies say should equal 0vu....me personally....I step up on my converters going in and then pad coming out....so that -6dbfs = 0vu....and then on the in 0vu = -6 dbfs....seems to have solved this problem of scale, at least for me....been doing this for quite a few years now....and the more I use this method....the more i like....keeps things hot in the DAW and not too hot on the desk....
I am not saying that input "gain" to a plugin does or does not change the sound of the plugin (although I just don't see how that would be possible)....as you mentioned the only thing I could see this affecting would be a plugin that intentionally has a gain triggered algorithm....i.e. more "gain" equals some sort of harmonic generation....
I am with you on this one....I can't see why trimming gain in the digital world would make something sound more "analog"....
One of the amazing things in thinking about these threads is that tracking to DAW's has pretty much been accepted as a professional standard way of working. And some of the top mastering engineers & mastering houses are now doing mastering totally ITB and i hear no one complaining about those masters.
So its basically the part in between ie; the mixing where it seems people have the most issues. But what surprises me is that before DAW's became an option for mixing people were still complaining about their mixes lacking something per say even in the days of an all analog studio environment. First it was that they had to move the mix from an all tube console to a board with transformers/transisitors. Then it was that the board with the transformers colored & slowed the signal too much so moving it to a console with no transformers was the ticket. When that wore off and transformer based consoles made a come back, it was that they had to move the mix from a Neve or API to an SSL, so the SSL must be lacking. After that moving the mix from an SSL to an even more cleaner surface for instant recall like a Euphonix or the many digital mixers after took away the "mojo" .
And now we've come full circle.
So basically has anything really changed? The only thing is that their are more people complaining because there are more people trying to engineer and with the internet now we get to hear about it every second of the day.
But through all the bitching great sounding records have been mixed on every format.
So bottom line folks...maybe the problem is not what's in front of the eyes but more likely what's behind them.
You folks will think I am smoking crack but here is what I will normally do. I put the trim plug in first followed by either the SSL channel or the URS strip Pro. Then with the trim plugin up on the encoders Of My Icon D-dontrol. I put the fader at around 0. Then I adjust the trim plugin until the track fits into the mix fairly well. Now just EQ and compress to taste and do fine adjusts with the fader. This always works well and I never have a headroom issue. Plus all my fader are at the sweet spot. I know I am smoking Crack.
Heh...you're not smokin' crack.
I used to use the McDSP AC-1 plug to trim my levels to my desk. Nowadays, I just track levels to DAW using 0VU as a reference. For mix work I get recorded by other engineers, I usually mix the tracks ITB then stem stuff out to the board. The levels are manageable, if not just a *bit* hot, though I hardly use my desk's channel EQ these days (lotsa tastey outboard, with lotsa headroom!), so my ass is covered.
Everything in the original post is true, but it isn't why itb mixes don't sound as good (warm/deep/wide/clear/smooth/alive) as analog mixes.
In my experience, hardware EQ, compressors, reverbs at this time still outclass software, and tape certainly has its own mojo (an interesting discussion and an alternative way of looking at this whole controversy, for those of you who haven't read it, can be found here: http://www.gearslutz.com/board/q-butch-vig/398471-recording-tape-vs-pro-tools.html ), but, I am not at all convinced that a console is necessary to produce the best result. Granted, I am very much invested in the DAW (with 14 hardware inserts) way of doing things, so it's healthy for me to think this way.
You can actually test yourself to see if you can reliably hear which mix of the same song was done on a console vs. in protools.
I'd like to see everyone who has lately demoralized many of the ITB'ers among us by repeatedly claiming that great mixes are impossible without a console, take this test and have the unfalsifyable results made public.
If you have a stereo file, it's under the "multiple mono" folder.
fantastic! should this be done with all sessions, or is tracking less aggressively just as effective? If for example, I tracked keeping the meters below 50-60%?
Sorry for the confusion, just trying to understand the technique.
fantastic! should this be done with all sessions, or is tracking less aggressively just as effective? If for example, I tracked keeping the meters below 50-60%?
Sorry for the confusion, just trying to understand the technique.
Tracking with this is in mind is the same at the end. Remember Plugins that us modeling will conform the the same voltage levels I have outlined. O DBFS is insanely hot in terms of voltage reference. The analog analogy would technically be called "Ass Melting Hot"
Many years ago when talking with an engineer (the slide rule kind, not the faderjocky kind) who designs the stuff we use, he asked me if I would ever do a 2" analog mix on a Neve without using HPF's. I said "no, they are essential to cleaning out the mud and getting a tight and open sounding mix". He said "Good. In the same respect, you shouldn't be doing an ITB mix without using LPF's either". Same problem, opposite end of the spectrum. 180 degree problem. The buildup on a super accurate HF's, what you put in is what you get out, digital playback system is equally destructive to getting a smooth top end as LF mud is to getting a clean mix on the bottom.
As to how to do that, what to use where and where to set your LPF's, I'll leave that to your own experimenting. I can't deliver it all in a nice neat package now, can I???
This is pretty intriguing, never heard of this before. Anything published on this subject by the digital gurus? Anyone else have any experience with this?
JSL
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