Mixing and -DB

guitargirl7

New member
I was searching through the topics and posts in the forum when I came across a lot of different opinions about recording in -DB numbers. I've heard numbers from -6db to -24db. Being a "newb", I'm unsure where to set my numbers at. I'm currently using around -12.3db. What would you recommend?

Info:
I'm recording classic folk rock type music (mostly acoustic and female vocals), not anything heavy metal or scremo.

Gear:
Toshiba Satellite L775D laptop
Scarlett 2i2 interface
PT 10
SM57 mic

Thank you!
 
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Assuming you're recording 24 bit audio, keep peaks well below 0dBFS, preferably around -12dBFS. Average levels will end up around -18dBFS. There's no need to be super precise about it. If you do that you will be running your analog input path near its optimum level and you'll have plenty of headroom during mixdown.
 
Hi there

When you're recording in 24bits, you can safely leave quite a bit of headroom as you can boost the signal level post-tracking without raising the digital noise too much.
So you can quite comfortably have your average music level sitting at -12 to -24dBFS when recording with peaks hitting around the -12 to -8dBFS mark.

As long as the audio coming into your computer is roughly sitting halfway up the PPM meters you'll be fine. There's usually sufficient signal to noise in most good quality
converters to enable you to boost the signal afterwards without introducing too much digital noise, and when you're mixing everything together the combined audio will naturally increase the level of audio hitting the master buss.

Dags
 
Mixes just have to be peaking under 0dbfs. The only time you really have to worry about level is when you are recording (keep the levels averaging around -18dbfs while keeping the peaks below 0dbfs). Any other time, you just have to make sure it is below 0dbfs and you will be fine.
 
Mixes just have to be peaking under 0dbfs. The only time you really have to worry about level is when you are recording (keep the levels averaging around -18dbfs while keeping the peaks below 0dbfs). Any other time, you just have to make sure it is below 0dbfs and you will be fine.

Not quite, digital world models analog world. Digital dBFS which is Full Scale is totally diff from VU. Guitar Girl what your really asking is about proper Gain Staging this is the single most important thing you will ever learn as a young eng. Allowing sound to hit at 0 in the digital domain is not good seeing as VU metering dosent even go that high VU 0 = -20 on you daws dBFS meter. So that tells you your target level your should be at for mixing...(A post I just posted earlier on a similar post)

This is the key, I haven't heard but im sure "guitar zero" mixes sound like as if done on a analog console. I like that yellow is the new red. Just trying mixing with -18 rms and peaks @ -10 and then do a mix the conventional way (with instrument/track levels are pumping above -6 dBFS) and you will be SURPRISED of your results. Things that are digital must go analog at some point even while in digital so why not build/create/design the sound as if in a analog domain and not in a digital one. This gain staging that "guitar zero" just mentioned is proly the greatest advice any young or old eng could know. I even see pros that still pull up pro tools session and everything is pumping above -6 I just shake my head in disappointment. Analog modeled plugins even clip "internally" in a digital domain. No you dont hear it but the crunching of the harmonics "DESTROYS THE SOUND" if you dont believe me drive the plugin till the VU meters is slapping to the right like a rag doll and listen to the sound. Then gain stage the sound and re enter the sound into the plugin to where it hits right @ 0 VU tell me you dont hear the difference.
 
You are missing my point. the only time the average level is relevant is in the analog world. Once inside the box, the only thing that matters is not clipping. According to your instructions, ( mixes at -18rms and peaks at -10dbfs) the mix would only have a crest of 8db. That is a smaller crest than most crushed to death masters.

The only time you need to worry about average levels in the box is when you are using hardware emulating plugins. Other than that, there is no distortion until you hit -0dbfs. (until you bring it back into the analog world)
 
you will find, with time....
that bringing tracks into your daw, that are peaking much higher than about -10db, will eventually create a bit of a 'harsh' sound in the collective..

again, it goes back to proper gain staging.


if you are in a 24 bit environ, don't worry about not using available headroom.

it's not about level, so much is it is about 'tone' and gain staging.

meters in DAW's are notoriously weak...

you could very easily be peaking, and it not show up.

and this makes your sound 'crunchy'.

and will never register on a typical ITB meter...................
 
you will find, with time....
that bringing tracks into your daw, that are peaking much higher than about -10db, will eventually create a bit of a 'harsh' sound in the collective..
again, it goes back to proper gain staging.

if you are in a 24 bit environ, don't worry about not using available headroom.

it's not about level, so much is it is about 'tone' and gain staging.

meters in DAW's are notoriously weak...

you could very easily be peaking, and it not show up.

and this makes your sound 'crunchy'.

and will never register on a typical ITB meter...................

I didnt say it GONZO did, Fairview I know your an intelligent person trust me I thought the same way you did until I TRIED it. Then I researched to learn the difference and the techniques and the benefits. Just try it. Take an old mix and gain stage it properly Kick and snare to taste (round -6 & -8) the rest at analog VU Meter Level (-18 RMS & peak -10dbFS) that -8 crest you mention wont even matter at the Master Mix Bus when all the sounds are bussed or sent to one master track But ALLLLL that headroom will matter at that Master Mix Bus stage why? cuz all the levels have to add up to that Master Mix Bus (I laugh when I see Pro's Eng Master Mix Bus hitting 0) just give it a try believe me. Dose it really make sense to say that the only time you need to worry bout level is when you using analog equip? Are you listening to headphones or Monitors (analog). Your audio interface that hooked up to your computer (analog). Most dont even know the manuals to thier Digital/Analog interfaces even tell you the best level for digital. Since your levels are pumping above -6, and -3 are you "really" hearing the true representation of the sound and if you are is it at its best quality or worst? ALL PLUGINS are molded after hardware in some form or fashion (analog was first not Digital) even if the are quote on quote "digital" the characteristics are that of analog equipment no matter how you flip it. So the benefits of using analog techniques in a digital domain would render amazing results ( I mean isnt that what we are going for anyway in the Digital world a replica or some type parallel to the analog world. Some how to make our sounds REAL!as if done in the analog domain)


Reason why ITB mixes do not sound as good OTB, IMHO the main points
1.) Head Room - VU meter 0 equals -20 dbFS Yes! -20 dBFS not even -18. So mixing at these levels wont make a positive significant difference and benefit?? I beg to differ. You ever take a bad sounding mix or a mix that just seems like its just way to much and over driven and all on top of each other and you take the exact mix (no level changes) and pipe it into an SSL and its just instantly sounds better. Apart from the sound of the hardware and warmth its instant headroom thats the main reason why it sounds so good and has ability to "BREATH" all of a sudden the instruments all start to be heard clear and masking of freq instantly fade and your start to hear every single nuance of every instrument so clearly. Thas not the NS-10s buddy believe me its that SSL. The analog mixer gives instant gain staging. (You ever tried to over drive a SSL board? its almost impossible no matter how high the levels it still sounds good its an instant & constant headroom gain staging component. We dont have that in the digital world we have to create it. Hence the idea of gain staging is only associated with the DIGITAL DOMAIN).

2.) Analog - Harmonics & Sound & Distortion & Color - Believe it or not all these things change with different amounts of "headroom" thas available the more headroom you have the more of these characteristics take shape and take shape PERFECTLY. Less Headroom less "good" harmonic sound, with high level you start destroying the harmonics which inturn destroys the analog sound. (term by the other fellow. "yellow is the new red") A great saying! The "good" analog distortion becomes bad distortion which takes away from the color of an analog modeling domain. Grab any SSL modeled plugin Comp, Channel, EQ, I prefer the Waves NLS Channel (Mike SSL option, Mic input pressed) and send it a signal hitting 0 dBFS HORRIBLE, OMG TURN IT OFF!!! Then take the same signal and trim the sound before it goes into the plugin till you see the VU meter in the plugin barely grazing 0 AAAHHH!!! AMAZING DIFF!

3.) Analog Warmth - Analog warmth is another reason ITB sound less attractive. That warmth the full rich sound in analog world is unprecedented. Digital is sharp & brittle and weightless no rich sound at all. But with some pretty good analog modeled saturation plugins the warmth can be artificially added but same rules apply. Tape saturation is an analog characteristics try over driving a saturation plugins and it sounds terrible. You always find your self leveling the input to a saturation plugin so the plugin wont distort the heck out of your sound and change it so much to where it sounds as if your entire session is a metallic record. The same rules would apply for the entire DAW digital domain for every track, every mix every time.
 
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You are missing my point. the only time the average level is relevant is in the analog world. Once inside the box, the only thing that matters is not clipping. According to your instructions, ( mixes at -18rms and peaks at -10dbfs) the mix would only have a crest of 8db. That is a smaller crest than most crushed to death masters.

The only time you need to worry about average levels in the box is when you are using hardware emulating plugins. Other than that, there is no distortion until you hit -0dbfs. (until you bring it back into the analog world)

I think we need to stop thinking in terms of separation between the digital and analogue domains because, in any modern system, there are elements of both and they have to co-exist in order for optimal system operation to occur.

For instance.

A DAW and an audio interface form a complete audio system with inputs and outputs. Signal gets picked up, VIA THE ANALOGUE STAGE, on input and is SAMPLED AND ENCODED in the conversion stage, ready for storage and playback. When playback occurs, there is a reversal of the process: The RECONSTRUCTION PHASE, and an ANALOGUE OUTPUT STAGE.

Throughout the system there is a relative translation of signal amplitude be it represented in bits or voltage, and I assure you, it's a very good idea to keep the levels as consistent as possible and within the proper operating ranges of particularly the analogue components because they - unlike the magical 32-bit floating point architecture of the DAW - do have limits to how much voltage they can handle whilst providing clean and clear signal. I still contest that just because any given audio circuit is rated to, say, +20 dBu, it does not mean that that it only produces distortion when it clips. I have heard with my very own ears subtle distortion start to occur much lower than clip point in many modern pieces of gear when they are pushed too hard. I believe there is a reason why high quality studio hardware (i.e. outboard) has much more headroom than many of the entry level and prosumer devices we see currently.

So what's my point? Basically that the analogue components in a modern recording chain - and there are many - are not linear at all points of amplitude. Ever heard of "colour" when driving a master bus too hard? Ever heard of "analogue mojo"? Yeah, they're related.

Of course, there are other reasons for keeping your levels in the optimal range (generally considered 0dBu to +4dBu or around +/- 0VU/-18dBfs for peaks) and that includes intersample peaks occuring at the reconstruction phase. There are more.

There are just more reasons than not to do it, so I say forget about internet wisdom and rather do what is reasonable by staying mindful of and knowing the limitations of your equipment well. And of course, listen. Your ears will tell you eventually.

Cheers :)
 
I didnt say it GONZO did, Fairview I know your an intelligent person trust me I thought the same way you did until I TRIED it. Then I researched to learn the difference and the techniques and the benefits. Just try it. Take an old mix and gain stage it properly Kick and snare to taste (round -6 & -8) the rest at analog VU Meter Level (-18 RMS & peak -10dbFS) that -8 crest you mention wont even matter at the Master Mix Bus when all the sounds are bussed or sent to one master track But ALLLLL that headroom will matter at that Master Mix Bus stage why? cuz all the levels have to add up to that Master Mix Bus (I laugh when I see Pro's Eng Master Mix Bus hitting 0) just give it a try believe me. Dose it really make sense to say that the only time you need to worry bout level is when you using analog equip? Are you listening to headphones or Monitors (analog). Your audio interface that hooked up to your computer (analog). Most dont even know the manuals to thier Digital/Analog interfaces even tell you the best level for digital. Since your levels are pumping above -6, and -3 are you "really" hearing the true representation of the sound and if you are is it at its best quality or worst? ALL PLUGINS are molded after hardware in some form or fashion (analog was first not Digital) even if the are quote on quote "digital" the characteristics are that of analog equipment no matter how you flip it. So the benefits of using analog techniques in a digital domain would render amazing results ( I mean isnt that what we are going for anyway in the Digital world a replica or some type parallel to the analog world. Some how to make our sounds REAL!as if done in the analog domain)
First of all, I've been doing this for almost 30 years, I have tried it. You are simply misunderstanding what I'm saying. The harshness from running your levels too hot happens on the analog side, not the digital side. Yes, you are listening through analog monitors, but if you have the level turned down (on the analog side, not the digital side) the problem goes away.


Reason why ITB mixes do not sound as good OTB, IMHO the main points
1.) Head Room - VU meter 0 equals -20 dbFS Yes! -20 dBFS not even -18.
It depends how your converters are calibrated. Motu calibrates so line level = -15dbfs. This varies from manufacturer to manufacturer. Some converters can be user calibrated.
Thas not the NS-10s buddy believe me its that SSL. The analog mixer gives instant gain staging. (You ever tried to over drive a SSL board? its almost impossible no matter how high the levels it still sounds good its an instant & constant headroom gain staging component. We dont have that in the digital world we have to create it. Hence the idea of gain staging is only associated with the DIGITAL DOMAIN).
The clarity of NS-10's?????!!!! Anyway, I have overdriven an SSL, it doesn't sound as good as overdriving a Neve, but the point of an SSL is clarity and slickness, while the point of a Neve is 'character' and 'mojo'.

2.) Analog - Harmonics & Sound & Distortion & Color - Believe it or not all these things change with different amounts of "headroom" thas available the more headroom you have the more of these characteristics take shape and take shape PERFECTLY. Less Headroom less "good" harmonic sound, with high level you start destroying the harmonics which inturn destroys the analog sound. (term by the other fellow. "yellow is the new red") A great saying! The "good" analog distortion becomes bad distortion which takes away from the color of an analog modeling domain. Grab any SSL modeled plugin Comp, Channel, EQ, I prefer the Waves NLS Channel (Mike SSL option, Mic input pressed) and send it a signal hitting 0 dBFS HORRIBLE, OMG TURN IT OFF!!! Then take the same signal and trim the sound before it goes into the plugin till you see the VU meter in the plugin barely grazing 0 AAAHHH!!! AMAZING DIFF!
Yes, but the peak level isn't what creates this distortion and coloration, it's the average level. If the average level is too high in the analog domain, everything starts sounding pinched and harsh. The peak level doesn't matter as much because most analog circuitry is capable of handling high volume, short term transients without a problem.

3.) Analog Warmth - Analog warmth is another reason ITB sound less attractive. That warmth the full rich sound in analog world is unprecedented. Digital is sharp & brittle and weightless no rich sound at all. But with some pretty good analog modeled saturation plugins the warmth can be artificially added but same rules apply. Tape saturation is an analog characteristics try over driving a saturation plugins and it sounds terrible. You always find your self leveling the input to a saturation plugin so the plugin wont distort the heck out of your sound and change it so much to where it sounds as if your entire session is a metallic record. The same rules would apply for the entire DAW digital domain for every track, every mix every time.
This is just silly. Digital doesn't sound bad or characterless, cheap digital recorded poorly does. Can you tell the difference between an album that was recorded analog vs. one that was recorded digitally by listening to the CD? If you can, than digital isn't the problem, since you are listening to a digital recording of an analog one and the 'analog goodness' wasn't completely stripped by the digitization process.

I've typed this a thousand times before, so I will give the readers digest version. If you want the full version, search this forum or google one of my gain staging articles.

When recording, record at line level. That is the level that all the analog equipment in the chain is designed to perform best. Line level is 0dbVU. VU meters are slow and do not pick up transients, because (within reason) they are not important in the analog gain staging process.

Unfortunately, a lot of equipment doesn't come with analog VU meters, so you are stuck with the ones in the DAW. These are peaking meters and really only measure the peak power with any accuracy. This is because the peak level is what is important in the digital world. As long as you are in the box, all you have to worry about is being sufficiently above the noise floor and below -0dbfs.

All the trouble and confusion comes in when you try to go between the two worlds.

Say your converters are calibrated to -18dbfs = line level. Things like distorted guitars and violins recorded at 0dbVU on the preamp meters would have a level bouncing around -18dbfs on the DAW meters. Instruments like piano would be up around -10dbfs, even though the preamp meters would read 0dbVU. Instruments like drums that hit 0dbVU would have a good chance of clipping the meters in the DAW. This is because the crest factor of all these instruments is different. A violin has no percussive transient, so it's average and peak level are pretty close to each other. A snare drum has a huge initial transient and a very short, much quieter decay, so it's peak is way above it's average level.

Remember, on the analog side, average level is what is important and on the digital side, peak level is what is important.

So, there is the delema. How do you know what level on what meter to record anything at? The simple answer is, if you have both analog and digital meters, to record everything at 0dbVU while watching that you don't go over 0dbFS.

If you only have digital meters, then set the recording levels for a sustained note at -18dbfs (or what ever your converters are calibrated to) and just keep an eye out for any transients that might be getting too close to 0dbfs. For drums, I tend to set the level so it peaks at -6dbfs on the hardest hits,
not because recording hotter that that for drums would sound bad, but just so there is a buffer zone for when the drummer gets a little too happy.

I've got a session staring up in a few minutes, so I've got to stop for now. But just one more thing about your advice that the mix should be averaging about -18dbfs rms (I agree with that) and peaking at -10dbfs. Most commercial CDs have more of a difference between the peak and RMS than you are describing. (if you just brought that mix up to peak at 0dbfs, the average would be a -8dbfs) Having peaks up dancing around -4dbfs or so would be more like it for a dense mix. If it were something like acoustic guitar, the rms level would have to drop a lot to keep the peaks at -10dbfs because of the nature of the instrument.

That is the biggest point of all: The peak to average recording level changes with the nature of each instrument you are trying to track or mix. Which is why most of the answers to questions about audio are "it depends".
 
Ah, thank you farview. I now understand what you were saying and am therefore in agreement with you.

I still think my points are valid, however.

Cheers :)
 
First of all, I've been doing this for almost 30 years, I have tried it. You are simply misunderstanding what I'm saying. The harshness from running your levels too hot happens on the analog side, not the digital side. Yes, you are listening through analog monitors, but if you have the level turned down (on the analog side, not the digital side) the problem goes away.

On what planet?... goes away? it never goes away maybe less in loudness but dosent go away. Wait! So if I run a level HOT!!! on the digital side it wont be harsh and distort?? It only distorts and is harsh in the anaglog world not in the digital world???????????????...Come on Fairview (I know your smarter than that)

It depends how your converters are calibrated. Motu calibrates so line level = -15dbfs. This varies from manufacturer to manufacturer. Some converters can be user calibrated.

True

The clarity of NS-10's?????!!!! Anyway, I have overdriven an SSL, it doesn't sound as good as overdriving a Neve, but the point of an SSL is clarity and slickness, while the point of a Neve is 'character' and 'mojo'.

My point still stands rather Neve or SSL they are analog period analog headroom still applies

Yes, but the peak level isn't what creates this distortion and coloration, it's the average level. If the average level is too high in the analog domain, everything starts sounding pinched and harsh. The peak level doesn't matter as much because most analog circuitry is capable of handling high volume, short term transients without a problem.

Wrong! "Peaks dont change the distortaion or cloration?" lol Im sorry sir your sadly mistaken. Now your right RMS effects the distortion and coloration "More" so but peaks do still cause distortion and change of color if that wasn't the case your daw wouldn't have a "clip" indicator"

This is just silly. Digital doesn't sound bad or characterless, cheap digital recorded poorly does. Can you tell the difference between an album that was recorded analog vs. one that was recorded digitally by listening to the CD? If you can, than digital isn't the problem, since you are listening to a digital recording of an analog one and the 'analog goodness' wasn't completely stripped by the digitization process.

Your missing the point of my argument I never stated that Digital could never sound as good as analog I just listed the short comings of digital verses analog. When digital tools are put in the right hands its can very well sound like an analog mix. I am hybrid. I am for both, I am not against digital. (you see fatso chimed in "THE FORVER DIGITAL DEFENDER WARRIOR" lol) I am completely for digital when it falls short of analog there are tools to overcome these short coming I am well aware and in support of. We were discussing gain staging or did we switch?

All the writing of the digest I agree with..from the beginning I and GONZO didn't agree with was this statement by you
Once inside the box, the only thing that matters is not clipping. According to your instructions, ( mixes at -18rms and peaks at -10dbfs) the mix would only have a crest of 8db. That is a smaller crest than most crushed to death masters.

This is what we both felt you were incorrect about , a -8 crest on a individual track doesn't even apply in proper gain staging! If I had a crest of some symphony sample of -12 crest it wouldnt matter on a individual track because that is the nature of the track.


Say your converters are calibrated to -18dbfs = line level. Things like distorted guitars and violins recorded at 0dbVU on the preamp meters would have a level bouncing around -18dbfs on the DAW meters. Instruments like piano would be up around -10dbfs, even though the preamp meters would read 0dbVU. Instruments like drums that hit 0dbVU would have a good chance of clipping the meters in the DAW. This is because the crest factor of all these instruments is different. A violin has no percussive transient, so it's average and peak level are pretty close to each other. A snare drum has a huge initial transient and a very short, much quieter decay, so it's peak is way above it's average level.

Not be rude this statement is irrelevant I am not talking about recording level not once did I bring up recording level. I am speaking of once every thing is in the digital domain and gain staging as if in an analog domain, that is my argument not recording level.

But just one more thing about your advice that the mix should be averaging about -18dbfs rms (I agree with that) and peaking at -10dbfs.

Not the entire Mix at -18RMS & -10PEAK meaning not the Master Mix Bus. Im referring to each individual instrument would be -18 RMS Peal -10 dBFS, TOO TASTE AND CHARACTER OF THAT INSTRUMENT.

Most commercial CDs have more of a difference between the peak and RMS than you are describing.

Again I must disagree, IM SO NOT TRYING TO BE AN ASS LOL. But maybe you mix classical music lol. Me most Commercial CDs, Pop, Hip Hop, R&B. The RMS & Peak level is not a major difference. Are you aware of the Loudness Wars?? and umm... We LOST!!! and the commercial public won! they wanted Louder, LOUder and LOUDER! I must disagree in todays Commercial CD's there is NOT much difference between Peak & RMS. We are loosing more and more dynamic range by the second. Music today is one big wall of sound there is no such things as dynamic range anymore, as you mentioned.

(if you just brought that mix up to peak at 0dbfs, the average would be a -8dbfs) Having peaks up dancing around -4dbfs or so would be more like it for a dense mix.

I would never bring an entire mix to 0 thats my whole point. If your talkin instruments same rules apply. If the mix was at 0 would it matter where the RMS in the digital domain?? its still clipping because its hitting 0. your statement "peak bouncing -6"... is irrelevant to the argument.

If it were something like acoustic guitar, the rms level would have to drop a lot to keep the peaks at -10dbfs because of the nature of the instrument.

Now this is what I was waiting for YES! YOU ARE CORRECT! If you had to get an acoustic guitar to -10dBFS PEAK that is alot of level to drop you sir are correct! It would be almost to quite (you would think) As in reference to what tho "Fairiew"?... How would you know that?? How would you know that, that Acoustic guitar would have to drop SO LOW it would almost be inaudible? How would you know that? You would know that because of ALL OF THE OTHER INSTRUMENTS IN YOUR MIX ARE LOUDER! DING!!! DING!! DING!!!. If you pulled up a mix with an acoustic guitar or say even a flute, something really quite and you instantly brought it down to -18 RMS & -10 Peak dBFS you would have to RE-Level or Re-Mix all of your other instruments to balance around that acoustic gtr or that flute to make it audible again and that my friend "Fairview" IS GAIN STAGING! (See I told you, you were smart) :thumbs up:....WAIT! Now am i saying every track should be balanced off of the acoustic gtr? NOOO!!! Am i saying that gain staging should start with the most quiets instrument? NO!... I am saying you should start the gain staging with the LOUDEST instrument in the mix! The Kick & Snare hence why my first statement I said Kick and snare -6 to -8 dbFS PEAK! WHY? cause its the LOUDEST instrument in the mix!
 
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My general rule is to not have a channel going more than -12/-10 db peak (and I will not die if there 1 or 2 peaks around -6 db). In doing so, you still use enough bits (and I mean enough) and don't cram the circuit of your converters or outboard gears, being that they are calibrate at -18dbfs.

But I think most importantly, it's a lot about how we deal with the transient/envelope ratio on each track (going to the stereo buss). That's why I so much love tape, saturation or and the slate console plugs in the first inserts. Stavrou have really good advice regarding this subject in his book.
 
That is the biggest point of all: The peak to average recording level changes with the nature of each instrument you are trying to track or mix. Which is why most of the answers to questions about audio are "it depends".

So now you see why I cant agree with your last statement "Fairview? is not cuz I dont like ya lol,or cuz I think Im smarter or better Im sure I could learn alot from you. But the answer to most audio LEVEL questions are not "it depends"? And even the pro's think so

Mix Eng Hand Book: The 6th Element of Mixing, Balance - level relationships between instruments

All Engineers Asked the question of sound level at the START of their gain staging for ALMOST EVERY SONG!

Benny Faccone (Sting, Pattie Label) " I usually start with bass at about -5 and kick at about -6. The combination of both (meaning the added volume on a Bus/Aux) should be a total of -3. By the time the whole song gets put together I used the computer to adjust levels. I WOULD HAVE TRIMMED EVERYTHING BACK (hmm wonder he did that coulda been just the computer he was using..cheap dells) The bass would be hitting at around -7 when Im done."

Don Smith (Rolling Stones, Need I say More). I start my kick and bass in the area of -7 "VU" by the time we put everything together its at +3 VU"...Now Don was competely on a analog mixer that +3 VU is still = to -15dbFS Point THERE IS EVEN GAIN STAGING IN THE ANALOG DOMAIN WAT!! lol.

Ed Sealy (MULTI Grammy Award Winning Mix Eng, Martina McBride, Trick Pony, Clay Walker, Blake Shelton, Dolly Parton )
"Gain Staging, ughh usally a great place to start would be with the kick drum at -6 or -7. Then I'll try to get a bass level comparable to that. IF ITS NOT EXACTLY COMPARABLE ON THE METERS, because one is peaking and one is sustaining then I at least get them to SOUND COMPARABLE... (you see even if he doesnt get them to match in level he wont be to far off because he gain staged the kick correctly which EVERYTHING is going to be in reference to that. See you cant go wrong in accurate gain staging even when the character of the music dosent allow you to match levelsl) "I get them to at least "sound" comparable because later, in mastering if you affect one you will affect the other. So as long as the ratio is pretty close between the two, so then if you go to adjust the kick at least its not going to unhack the bass way out as long as they relate to each other. Thats kind of a good starting spot for me.

Lee DeCarlo (Kenny Loggin, John Lennon Aerosmith) "Ill get the snare drum constantly hitting the back beat of the tune at around -5, and everything gets "BUILT" around it...

Again why my statement said use Kick and Snare to Balance or build off of. Gain staging is not new I just follow the pro's on this one but 70-80% of eng (YES EVEN PRO"S) do not perform this technique I just dont know why. Does it our make mixing skills better than theres? Not really. But could it make a mix a little better. I think it could and thas all Im saying!

I was just like you Fairview had the same views, thas why I asked BEFORE you criticize just try it .Recall an old mix and try it and play the before & after mix. (might have to turn your monitors up a lil for the "better" balanced mix so mark it and calibrate it.) Know your gonna say, "But the overall level is lower than my pervious mix :facepalm: ok send it off to the mastering eng and let him get the same loudness level as the previous mix duh! lol. Listen to before & after mix and Before & after Master on Both and tell me what you think.. I DARE YOU lol!
 
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My general rule is to not have a channel going more than -12/-10 db peak (and I will not die if there 1 or 2 peaks around -6 db). In doing so, you still use enough bits (and I mean enough) and don't cram the circuit of your converters or outboard gears, being that they are calibrate at -18dbfs.

But I think most importantly, it's a lot about how we deal with the transient/envelope ratio on each track (going to the stereo buss). That's why I so much love tape, saturation or and the slate console plugs in the first inserts. Stavrou have really good advice regarding this subject in his book.

I agree with you 100% Mindsound on every point targets -12/-10 and it WILL NOT kill the mix if a few are around -6. Yes you certanily have enough head room and bit depth at these levels I agree. Transients on each track is SO VERY IMPORTANT! You right many forget that detail Mindound. I may self apply Waves NLS Channel plugins first on every track and sometimes when needed the trim plugin before the NLS as to trim my audio.

-My process of gain staging of individual tracks/channels. IF audio is from a 3rd party source its most likely HOT! I apply trim tool I trim -18. (Yes every time no matter the source or instrument, wouldn't it be great if NLS had a input trim parameter built in like the Samson and VCC but anyway.)

-Then I apply my NLS Channel plugin I pull the fader all the way down on the plugin. (mean while leaving my daw fader at unity gain 0.). Then I initiate the "Mic Input" to enable my analog headroom, gain and analog molded saturation.

-Then I slowly bring up te fader of the plugin (now actual;y bring the sound level of the track INTO the Analog modeled plugin in channel, all the while watching my VU meter at the top of the plugin at that point I know there is no way I will clip in the analog domain or digital domain and I start this with the kick. In most cases this leaves my kick anyway where from -8 to -6 dbFS. That is my process. Just like Mindsound...(If more than one are on the same cord of correction you can then assume that the majority is correct - Judith Butler)
 
Goota Love Mo Facto, I respect his views alot lol. If we ever had a DIY Hero he would be it lol. In a room full of haughty analog heads bring Facto to set'em straight.
 
Just a little FYI for ya.

VU meters are not strictly calibrated to dB because it is, for the most part, a loudness meter so it's sometimes difficult to know what you're looking at. Another interesting fact is that 50% OF THE SCALE REPRESENTS THE TOP 6dB OF RANGE. Once you're at +3VU (the top of the VU scale) you should actually be somewhere near full scale.

Remember the VU meter has a rise and fall time of 300 ms, which is too slow to catch peaks so it's reading should be taken with consideration.

Cheers :)
 
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Poor Guitargirl7. She asked what she thought was a simple question and suddenly found herself in....

do DO do do....do DO do do....

THE TWILIGHT ZONE.

As you may gather, Guitargirl7, you've brought up a topic that's a constant source of debate and one that engenders all sorts of strong opinions.

To try and answer your original question, when you're tracking, try to set levels so the average of your track is around -18 on your DAW with the peaks going higher than that. How much higher? It depends on what you're recording. Some instruments and voices have a wider dynamic range than others. The main thing is to leave enough head room that even a brief peak doesn't go above 0dBFS since that's where clipping starts.

What level to shoot for when mixing? Well, if you mix "in the box" it doesn't matter much as long as you don't hit that magic 0dBFS level where there's clipping. The exceptions to this are if you have external rack gear or use some plug ins that emulate analogue performance.

Once you've finished your mix (at whatever level) then you'll probably want to tweak the levels right up to around the 0dBFS point (using compression and/or hard limiting) otherwise your mixes will sound quiet on a CD or MP3 player compared to commercial recordings.

Hope this helps.
 
On what planet?... goes away? it never goes away maybe less in loudness but dosent go away. Wait! So if I run a level HOT!!! on the digital side it wont be harsh and distort?? It only distorts and is harsh in the anaglog world not in the digital world???????????????...Come on Fairview (I know your smarter than that)
The hardness and distortion doesn't happen on the digital side as long as you stay under clipping. It will not change with level (other than it's relationship to the noise floor) until 0dbfs is exceeded. All the harshness happens on the analog side from running the levels too hot. BTW, it's farview, not fairview.





Wrong! "Peaks dont change the distortaion or cloration?" lol Im sorry sir your sadly mistaken. Now your right RMS effects the distortion and coloration "More" so but peaks do still cause distortion and change of color if that wasn't the case your daw wouldn't have a "clip" indicator"
Pay attention, I was talking about the analog side of things. The color and distortion in the analog chain is sensitive to the rms level, but not the peak level (within reason). This is because the power supply (and the rest of the circuitry) can deal with short busts of high energy, but cannot sustain that energy level. That's what causes the distortion and harshness.




All the writing of the digest I agree with..from the beginning I and GONZO didn't agree with was this statement by you

This is what we both felt you were incorrect about , a -8 crest on a individual track doesn't even apply in proper gain staging! If I had a crest of some symphony sample of -12 crest it wouldnt matter on a individual track because that is the nature of the track.




Not be rude this statement is irrelevant I am not talking about recording level not once did I bring up recording level. I am speaking of once every thing is in the digital domain and gain staging as if in an analog domain, that is my argument not recording level.
This statement was referring to your earlier advice from this post...
I like that yellow is the new red. Just trying mixing with -18 rms and peaks @ -10 and then do a mix the conventional way (with instrument/track levels are pumping above -6 dBFS) and you will be SURPRISED of your results.
Isn't this saying to mix at -18dbfs rms with peaks of -10dbfs?



Not the entire Mix at -18RMS & -10PEAK meaning not the Master Mix Bus. Im referring to each individual instrument would be -18 RMS Peal -10 dBFS, TOO TASTE AND CHARACTER OF THAT INSTRUMENT.
Again, it would depend on the crest of the instrument. There is no way to get a snare drum to have that little dynamic range, or a violin to have that much.



Again I must disagree, IM SO NOT TRYING TO BE AN ASS LOL. But maybe you mix classical music lol. Me most Commercial CDs, Pop, Hip Hop, R&B. The RMS & Peak level is not a major difference. Are you aware of the Loudness Wars?? and umm... We LOST!!! and the commercial public won! they wanted Louder, LOUder and LOUDER! I must disagree in todays Commercial CD's there is NOT much difference between Peak & RMS. We are loosing more and more dynamic range by the second. Music today is one big wall of sound there is no such things as dynamic range anymore, as you mentioned.
I don't know how you are measuring the crest of commercial CDs, but even the loudest ones I've ever seen have a crest of about 7 or 8db. I do mostly metal, but I also work with a symphony orchestra, musical theater, jazz, blues etc...

A sine wave has a crest of 3db, so music having a crest of 8db is really, really, really, smashed.



I would never bring an entire mix to 0 thats my whole point. If your talkin instruments same rules apply. If the mix was at 0 would it matter where the RMS in the digital domain?? its still clipping because its hitting 0. your statement "peak bouncing -6"... is irrelevant to the argument.
Technically, it doesn't distort until it tries to go over 0dbfs, but -.01dbfs is not distorting at all (digitally)



Now this is what I was waiting for YES! YOU ARE CORRECT! If you had to get an acoustic guitar to -10dBFS PEAK that is alot of level to drop you sir are correct! It would be almost to quite (you would think) As in reference to what tho "Fairiew"?... How would you know that?? How would you know that, that Acoustic guitar would have to drop SO LOW it would almost be inaudible? How would you know that? You would know that because of ALL OF THE OTHER INSTRUMENTS IN YOUR MIX ARE LOUDER! DING!!! DING!! DING!!!. If you pulled up a mix with an acoustic guitar or say even a flute, something really quite and you instantly brought it down to -18 RMS & -10 Peak dBFS you would have to RE-Level or Re-Mix all of your other instruments to balance around that acoustic gtr or that flute to make it audible again and that my friend "Fairview" IS GAIN STAGING! (See I told you, you were smart) :thumbs up:....WAIT! Now am i saying every track should be balanced off of the acoustic gtr? NOOO!!! Am i saying that gain staging should start with the most quiets instrument? NO!... I am saying you should start the gain staging with the LOUDEST instrument in the mix! The Kick & Snare hence why my first statement I said Kick and snare -6 to -8 dbFS PEAK! WHY? cause its the LOUDEST instrument in the mix!
I was talking about a mix with only acoustic guitar, we are talking about mixing and that statement of mine was directed toward your statement that a mix should be -18dbfs rms with peaks at -10dbfs. Again, a mix with a crest of 8db is way too crushed. It can be argued that a master with 8db crest is too crushed, but like you said, we lost the loudness war.
 
So now you see why I cant agree with your last statement "Fairview? is not cuz I dont like ya lol,or cuz I think Im smarter or better Im sure I could learn alot from you. But the answer to most audio LEVEL questions are not "it depends"? And even the pro's think so

Mix Eng Hand Book: The 6th Element of Mixing, Balance - level relationships between instruments

All Engineers Asked the question of sound level at the START of their gain staging for ALMOST EVERY SONG!

Benny Faccone (Sting, Pattie Label) " I usually start with bass at about -5 and kick at about -6. The combination of both (meaning the added volume on a Bus/Aux) should be a total of -3. By the time the whole song gets put together I used the computer to adjust levels. I WOULD HAVE TRIMMED EVERYTHING BACK (hmm wonder he did that coulda been just the computer he was using..cheap dells) The bass would be hitting at around -7 when Im done."
What year was this interview? In the early days of digital (up to the late 90's) most converters were calibrated to line level = -10dbfs or -12dbfs, depending on the converters. You had to deal with things much differently back then. In fact, back then, the big problem was not being able to run the preamps hot enough.

Don Smith (Rolling Stones, Need I say More). I start my kick and bass in the area of -7 "VU" by the time we put everything together its at +3 VU"...Now Don was competely on a analog mixer that +3 VU is still = to -15dbFS Point THERE IS EVEN GAIN STAGING IN THE ANALOG DOMAIN WAT!! lol.
+3vu is -15dbfs rms, however the peaks will be much higher, so this is irrelevent.

Ed Sealy (MULTI Grammy Award Winning Mix Eng, Martina McBride, Trick Pony, Clay Walker, Blake Shelton, Dolly Parton )
"Gain Staging, ughh usally a great place to start would be with the kick drum at -6 or -7. Then I'll try to get a bass level comparable to that. IF ITS NOT EXACTLY COMPARABLE ON THE METERS, because one is peaking and one is sustaining then I at least get them to SOUND COMPARABLE... (you see even if he doesnt get them to match in level he wont be to far off because he gain staged the kick correctly which EVERYTHING is going to be in reference to that. See you cant go wrong in accurate gain staging even when the character of the music dosent allow you to match levelsl) "I get them to at least "sound" comparable because later, in mastering if you affect one you will affect the other. So as long as the ratio is pretty close between the two, so then if you go to adjust the kick at least its not going to unhack the bass way out as long as they relate to each other. Thats kind of a good starting spot for me.
I think you need to go back and re-read my posts, you are doing nothing but proving my point for me.

Lee DeCarlo (Kenny Loggin, John Lennon Aerosmith) "Ill get the snare drum constantly hitting the back beat of the tune at around -5, and everything gets "BUILT" around it...

Again why my statement said use Kick and Snare to Balance or build off of. Gain staging is not new I just follow the pro's on this one but 70-80% of eng (YES EVEN PRO"S) do not perform this technique I just dont know why. Does it our make mixing skills better than theres? Not really. But could it make a mix a little better. I think it could and thas all Im saying!

I was just like you Fairview had the same views, thas why I asked BEFORE you criticize just try it .Recall an old mix and try it and play the before & after mix. (might have to turn your monitors up a lil for the "better" balanced mix so mark it and calibrate it.) Know your gonna say, "But the overall level is lower than my pervious mix :facepalm: ok send it off to the mastering eng and let him get the same loudness level as the previous mix duh! lol. Listen to before & after mix and Before & after Master on Both and tell me what you think.. I DARE YOU lol!
You have so completely misunderstood what I have written. And again, I've been doing this professionally for almost 30 years. I started with all analog working in big studios doing commercial stuff, movies etc... I've been around for the switch to digital, I am very aware of proper gain staging, conversion theory, the differences between fixed point and floating point mix busses, etc... I've had to deal with analog tape, the early Sony DASH machines, DAT, ADAT, DTRS, UMATIC, protools, Cubase, etc... and I've made my living doing this for most of that time.

Re-read my statements, you have misunderstood.
 
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