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Post by johneppstein on Apr 1, 2020 22:55:44 GMT -6
You have way more flexiblity in the analog world with gain staging than the digital, at least when it comes to pushing devices. In analog, depending on the unit, you can over drive certain circuits/tubes/transformers/ect to get different colors and things. Which I think most who use analog gear are pretty familiar with, it's one of the best reasons to use analog gear. In digital, if you hit 0 or go over, you're screwed. For me, even though I record at 32bits in software(all my converters are only 24bit, not a lot of 32 bit converters out there) I push the levels as high as I can. I do a lot of classical recording, so getting the S/N ratio of the space and performance is everything. Changing the mic pre gain changes how I place the microphone and how the baseline sound is to start with. Which again for me is everything. I don't do tons of heavy EQ, certainly no compressor, rarely anything in post honestly. It's all about mic placement and gain into the recorder. So i push the gain, I try and peak my Grace micpre's at -3dBfs at the loudest part of the performance. I've learned Grace pre's very very well(m802 particularly, m108 since last year as well) and know how to set them usually after hearing a rehearsal once while also anticipating that it's going to be louder during the live performance because it always is. I do this because in my case in post if you just "raise the gain" of the clip or even the volume too much the noise floor of the room becomes VERY obvious during the quiet sections. This isn't acceptable in the classical world it needs to be clean as possible and natural. Now, most the music I think people here are working on do not have that as an issue I think it is safe to say. The noise floor is nothing most the time, you can hide things in a big production and filter the crap out of things. That said, even when I record other stuff, I still shoot for peaking a -6dBFS or -8. How you get there with whatever analog chain you decide before you, doesn't really matter. You can get lots of cool color's by pushing and pulling the sound with gear. It's not like you aren't doing that in post anyways with plugins or if you use outboard. That said, VU meters are great in the analog world to use. But if you're tracking and don't have a true Peak meter showing you DBFS, that's a big gamble in my book. Wouldn't do that personally.
However, the thing that a lot of digital guys don't get is that since your noise floor is so ridiculously far down on any remotely modern converter rig, and since digital doesn't vary tone with level like some analog stuff does, you can really track way far down (by analog standards) with no problems. In fact most mastering engineers love it when a project comes in just ticking -15dBfs. There is really no sane reason to deliver a project that's any hotter than that.
And that definitely gives you better odds in your "big gamble" because you can be off by 10 dB, give or take, and still not get yourself into trouble, at least not if you're working with an ME who knows what he's doing.
BTW, most of the guys who really know will tell you that you really shouldn't track channels digitally at levels higher than -15dBfs. The argument gets fairly esoteric, but what it comes down to is that during mixdown you can get intersample peaks that produce nasty digital clipping even when you think you're good. And the problem compounds with the number of channels in the mix. The technical explanations get a bit beyomd me at this point but Bob O might be able to do better.
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Post by chessparov on Apr 1, 2020 23:19:06 GMT -6
I figured you might know just a l-e-e-tle bit more than me.  I just know enough to be mildly dangerous, in a Studio...Apartment! Chris No - didn’t mean it to sound catty. I just was trying to explain what I was trying to ask. Jamie definitely had some good info there. Gonna do that in my next vocal. Thanks John, for being so thoughtful (I did "get" the spirit of your original intent BTW). In fact, I really appreciate how ALL you guys are patient with me-including my attempts at humor!  It reminds me of when I was a pre-teen/teenager, hanging out with the strongest chess players, in the L.A. area. Except I was a lot shyer then, with the Big Dogs. Besides John, when I feel catty, I simply stay ITB.  Chris P.S. Oops, just saw John E's post. I have recorded at 24 bit, and gone -4. "Forgive me Father for I have pinned" |
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Post by Deleted on Apr 1, 2020 23:53:16 GMT -6
I carefully gainstage everything when setting up to track to not clip and then pull the faders down at least twice in the box 20db or so. All digital clipping makes the sound more 2d and the clipped transients never sound right to me on monitors. That is just more work to do to round the clipped transients and resulting IMD. It's better for the mix to not have to do that.
Step 1 is use a real pro ad converter when tracking. Greater than +20 dbu on the input so you can get everything gainstaged right. The less you can fuck up, the less you will fuck up. +24 dbu max imput is always useful imo. Apogee soft limit can still save your ass despite Apogee making so much prosumer stuff now. 24 bit digital means nothing. You can set what the digital levels are going to be when it hits the computer on a lot of converters but you can't fix clipping the analog circuitry or actual converter chip. No AD converter can come close to real 24 bit headroom. It's all about the analog headroom going in. It's not even about SNR or THD+N. This is about the beefy the power supply, analog parts, and voltage rails are. Things that chase specs at the expense of sounding neither good nor accurate always do a worse job in the end. What's lost here is gone forever. Luckily good multichannel converters are fairly commonplace now (Apogee, Lynx, MOTU, SPL, even Focusrite Rednet is okay) but there are some pretty bad sounding or low headroom multichannel converters that are commonly recommended on gearslutz. If you're not monitoring post AD at all off a DA loopback, then you should be monitoring post AD because if it doesn't get into the computer, then it will never be there so only monitoring right off a console without a loopback is not telling you what's getting captured and what's being lost.
Not everything is Burl, Lavry, or the Dangerous AD+ (with transformer on) and takes overloads without getting nasty. You don't want to intentionally clip in tracking. Those converters mostly don't have a way to overdrive the analog circuity slightly and then hit the actual converter chip without flatlining. They are not tape. Quite frankly, the good ones that clip nasty like Lynx and MOTU AVB tend to actually be cleaner than those for getting the tracks into the box even if you're just using the computer as a tape machine. If you're doing anything that has any sort of vibe already (from the performance, instruments, room, or analog gear) you don't want it to clip even if you have a converter that can clip without the analog part getting nasty or collapsing. This is not printing the mix yet. I mainly do death and black metal with actual drums and guitars. Nothing that sounds like it could come out on Century Media or Nuclear Blast in 2020. I don't need any more crap on my plate with this shit. The moment anything is digitally clipped, it becomes more 2D and requires more work to sound acceptable.
Basically just don't clip anything going in. Don't actually clip solid state pres or use them in the noisy range at the very very top of their gain. Lots of stuff starts to get noisy there. Use everything in the "starts to get vibey and get bigger right before it clips or audibly distorts" range if you can and have time to carefully gainstage everything. Otherwise, just convert safely and cleanly, even with colorful gear. It doesn't matter how hot it is as long as nothing is digitally clipped.
Once you get the tracks on the drive and not clipped, you're safe. Pull down the faders -20 or so. Digital is not an analog console. You want even more headroom than a console. You have 24 real bits now because it's digital. You don't want to digitally clip or truly overload anything ever. No tracks, plugins, or buses. Do your thing itb or otb but don't clip it. Rough mix it and eq and process everything. Anything that is recorded afterwards as overdubs like vocals or guitar leads, pull the faders down too. If you have a converter that clips without it getting super nasty, and you really want to do it, (I'm not a fan of digital clipping), do it slightly as a kind of monitoring loop as a bus limiter. Whatever is your two bus prior to the clipping, send it otb in a loop back into the converter, clipping, that and then monitoring that input. And do that since the rough mix like it's a 2 bus compressor because it will affect everything. I hate clipping the AD anyway. When it comes time for automation, pull all the faders down again to make sure you don't clip anything when riding or drawing faders to try not to clip anything. If the mix is good on the monitors, the mono grotbox, headphones, and in the car, then I'm good.
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Post by Ward on Apr 2, 2020 6:38:53 GMT -6
Now that we have established all the rules . . .
Feel free to break them.
To smithereens.
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Post by cowboycoalminer on Apr 2, 2020 7:17:10 GMT -6
If you have a preamp, like a 1073, with input trim and an output attenuator, its good practice to leave the output atten at unity (all the way up), set the input trim till you reach your desired level into the DAW, and use the output atten to dial or fine tune the level if you still need to. BUT, if you want to push the preamp into blishful distortion, you keep pushing the input trim up till you hear the preamp breakup in a pleasing way, and keep pulling the output trim down to compensate. I usually keep my peaks between -12 and -6. At 24bit the only danger in recording too hot is clipping your converters, usually with fast transients (like drums) or really dynamic sources (like some vocalists). Even then, just because the red light hits intermittently doesn't necessarily mean you've yet clipped your converters. This ^^^^ |
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ericn
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Balance Engineer
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Post by ericn on Apr 2, 2020 7:58:02 GMT -6
It’s the Zen of audio balancing noise, overload, and tone. You do what you have to reach your own sonic nirvana.
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Post by svart on Apr 2, 2020 8:07:54 GMT -6
Just for clarity...I understand how input and output works on a mic pre (I’ve been doing this for a little while)...maybe I should have asked how hot you’re going into the next stage...in order not to achieve distortion or clipping. I guess using my ears has served me well for several decades. No audio gear that I know of has specs for compression(saturation or clipping) point. They never tell us at what level their gear will start to clip. IF they did that, then you might start to be able to devise a mathematical way to determine optimal levels, but until then, it's just finding it for yourself through the "listen for the crackle" method. |
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Post by svart on Apr 2, 2020 8:19:23 GMT -6
If you have a preamp, like a 1073, with input trim and an output attenuator, its good practice to leave the output atten at unity (all the way up), set the input trim till you reach your desired level into the DAW, and use the output atten to dial or fine tune the level if you still need to. BUT, if you want to push the preamp into blishful distortion, you keep pushing the input trim up till you hear the preamp breakup in a pleasing way, and keep pulling the output trim down to compensate. I usually keep my peaks between -12 and -6. At 24bit the only danger in recording too hot is clipping your converters, usually with fast transients (like drums) or really dynamic sources (like some vocalists). Even then, just because the red light hits intermittently doesn't necessarily mean you've yet clipped your converters. If I'm not mistaken, the original 1073 did not have an output attenuator. |
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Post by EmRR on Apr 2, 2020 8:48:29 GMT -6
No audio gear that I know of has specs for compression(saturation or clipping) point. They never tell us at what level their gear will start to clip. IF they did that, then you might start to be able to devise a mathematical way to determine optimal levels, but until then, it's just finding it for yourself through the "listen for the crackle" method. Yes, because it's a moving target depending on the type of circuit, the kind of distortion it makes, and what the knee of that distortion is, depending on the amount of negative feedback applied. You can spec 1% distortion but that tells you nothing about the harmonic content, or where 3% is, or 5%, or 10%. Some things it's another 0.5dB, others it's another 10dB. |
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Post by Guitar on Apr 2, 2020 8:48:51 GMT -6
If you have a preamp, like a 1073, with input trim and an output attenuator, its good practice to leave the output atten at unity (all the way up), set the input trim till you reach your desired level into the DAW, and use the output atten to dial or fine tune the level if you still need to. BUT, if you want to push the preamp into blishful distortion, you keep pushing the input trim up till you hear the preamp breakup in a pleasing way, and keep pulling the output trim down to compensate. I usually keep my peaks between -12 and -6. At 24bit the only danger in recording too hot is clipping your converters, usually with fast transients (like drums) or really dynamic sources (like some vocalists). Even then, just because the red light hits intermittently doesn't necessarily mean you've yet clipped your converters. If I'm not mistaken, the original 1073 did not have an output attenuator. True, but there was also the channel fader, and the fact that hitting tape hot is not the same as hitting ADC hot. |
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Post by svart on Apr 2, 2020 9:05:02 GMT -6
No audio gear that I know of has specs for compression(saturation or clipping) point. They never tell us at what level their gear will start to clip. IF they did that, then you might start to be able to devise a mathematical way to determine optimal levels, but until then, it's just finding it for yourself through the "listen for the crackle" method. Yes, because it's a moving target depending on the type of circuit, the kind of distortion it makes, and what the knee of that distortion is, depending on the amount of negative feedback applied. You can spec 1% distortion but that tells you nothing about the harmonic content, or where 3% is, or 5%, or 10%. Some things it's another 0.5dB, others it's another 10dB. I know this. That's why I'm saying it. In RF, there's a point called P1dB. It's the point of incident signal compression equal to 1dB. In layman's terms, you run out of voltage headroom in the amplifier, so the signal no longer gets louder. If you were to increase the input signal beyond the saturation point so that the output can no longer increase by 1:1 ratio, there will be an amount of signal on the input that equates to the output not being able to increase by 1dB. P1dB doesn't care about the types of harmonics created by the saturation, because as you mentioned, it's highly dependent on the types of circuits, but it's a measurement that could easily be used for all amplifiers because it defines the point of nonlinearity that can be used to determine the linear range for any amplifier. I think the problem is that most audio devices fudge their headroom numbers a little, and using compression point would probably dispel some of the marketing fluff they rely on. |
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Post by chessparov on Apr 2, 2020 9:26:22 GMT -6
"Set the Attack" then..."Release the Crackle!" (love them old stop motion Animation Movies!  Chris |
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Post by Blackdawg on Apr 2, 2020 9:27:12 GMT -6
You have way more flexiblity in the analog world with gain staging than the digital, at least when it comes to pushing devices. In analog, depending on the unit, you can over drive certain circuits/tubes/transformers/ect to get different colors and things. Which I think most who use analog gear are pretty familiar with, it's one of the best reasons to use analog gear. In digital, if you hit 0 or go over, you're screwed. For me, even though I record at 32bits in software(all my converters are only 24bit, not a lot of 32 bit converters out there) I push the levels as high as I can. I do a lot of classical recording, so getting the S/N ratio of the space and performance is everything. Changing the mic pre gain changes how I place the microphone and how the baseline sound is to start with. Which again for me is everything. I don't do tons of heavy EQ, certainly no compressor, rarely anything in post honestly. It's all about mic placement and gain into the recorder. So i push the gain, I try and peak my Grace micpre's at -3dBfs at the loudest part of the performance. I've learned Grace pre's very very well(m802 particularly, m108 since last year as well) and know how to set them usually after hearing a rehearsal once while also anticipating that it's going to be louder during the live performance because it always is. I do this because in my case in post if you just "raise the gain" of the clip or even the volume too much the noise floor of the room becomes VERY obvious during the quiet sections. This isn't acceptable in the classical world it needs to be clean as possible and natural. Now, most the music I think people here are working on do not have that as an issue I think it is safe to say. The noise floor is nothing most the time, you can hide things in a big production and filter the crap out of things. That said, even when I record other stuff, I still shoot for peaking a -6dBFS or -8. How you get there with whatever analog chain you decide before you, doesn't really matter. You can get lots of cool color's by pushing and pulling the sound with gear. It's not like you aren't doing that in post anyways with plugins or if you use outboard. That said, VU meters are great in the analog world to use. But if you're tracking and don't have a true Peak meter showing you DBFS, that's a big gamble in my book. Wouldn't do that personally.
However, the thing that a lot of digital guys don't get is that since your noise floor is so ridiculously far down on any remotely modern converter rig, and since digital does vary tone with level like some analog stuff does, you can really track way far down (by analog standards) with no problems. In fact most mastering engineers love it when a project comes in just ticking -15dBfs. There is really no sane reason to deliver a project that's any hotter than that.
And that definitely gives you better odds in your "big gamble" because you can be off by 10 dB, give or take, and still not get yourself into trouble, at least not if you're working with an ME who knows what he's doing.
BTW, most of the guys who really know will tell you that you really shouldn't track channels digitally at levels higher than -15dBfs. The argument gets fairly esoteric, but what it comes down to is that during mixdown you can get intersample peaks that produce nasty digital clipping even when you think you're good. And the problem compounds with the number of channels in the mix. The technical explanations get a bit beyomd me at this point but Bob O might be able to do better.
Eh I've heard that "argument" about never going above -15dB and I don't buy it. While I do agree converters have sounds and tones(everything does). But for instance a piano piece I'm working on right now had a dynamic range of 60-65dB. If I was peaking at -15dB during tracking then the pianissimo section would be at -80dB or so. Which is well into noise floor territory and not acceptable for classical. As so as you boost that up the noise floor comes with it. By Noise floor I mean everything btw. The HVAC of the room, microphone, preamp, it all adds up and is obvious when it's louder all the sudden. And I'm spoiled with an insanely quite space in the middle of no where with no traffic driving by and we can tweak the HVAC to get it quieter(have to be careful not to piss of the piano or it goes out of tune if too dry, too humid, too hot or too cold) I can however always turn things down in the mix in a nice transparent way with riding faders. Like I said though, this is a classical thing. For your standard bass drums guitars vocals recording session and mix down, you're definitely not dealing with that kind of dynamic range and even then it is not just accepted but common place to squash a lot of the dynamics in the name of tone and color. So noise floor isn't an issue in the same way. I bet most ME's out there rarely ever getting -15dB from anyone anymore. Most are happy to get -3dB haha which is sad in it's own way. But it is also just as accepted and common place for mix engineers to put stuff on the mix bus to get it louder and better shape. I do that. You do that. I don't think there is a person on this site that doesn't mix with some sort of chain on the mix bus. |
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Post by ragan on Apr 2, 2020 9:36:57 GMT -6
For a mic pre, I'll start with the output attenuation all the way open (no attenuation) then bring up the input until I get a workable level (I just try to keep peaks not hitting higher than -6). If I want some saturation out of the preamp I'll inch the input gain up and the output attenuation down until I like the sound. If I'm going through other analog stuff on the way in (EQ, comp, whatever) I'll have that stuff in the chain the whole time and just do a little output/input dance with each piece until I like the sonics. Whatever I'm spitting out into the ADC I just keep it peaking at -6 or less.
There's no noise penalty for trimming the signal in the DAW so no real reason to try to squeeze every dB out of the very top end of your tracking signal.
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Post by EmRR on Apr 2, 2020 12:28:25 GMT -6
Yes, because it's a moving target depending on the type of circuit, the kind of distortion it makes, and what the knee of that distortion is, depending on the amount of negative feedback applied. You can spec 1% distortion but that tells you nothing about the harmonic content, or where 3% is, or 5%, or 10%. Some things it's another 0.5dB, others it's another 10dB. I know this. That's why I'm saying it. Yes, I'm agreeing with you and building it out..... |
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Post by notneeson on Apr 2, 2020 13:13:15 GMT -6
One Bob O recommendation that has stuck with me is that there's no real reason to drive the line stages of your AD hard.
This excepts things like the Burls, and clipping the AD for effect, but has served me well and might be part of why I think convertor differences are way over hyped on the forums.
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Post by johneppstein on Apr 2, 2020 13:21:30 GMT -6
Eh I've heard that "argument" about never going above -15dB and I don't buy it. While I do agree converters have sounds and tones(everything does). But for instance a piano piece I'm working on right now had a dynamic range of 60-65dB. If I was peaking at -15dB during tracking then the pianissimo section would be at -80dB or so. Which is well into noise floor territory and not acceptable for classical. As so as you boost that up the noise floor comes with it. Doesn't really matter if you "buy" it or not, it's still true. You need to bone up on the importance of intersample peaks in digital audio. Here's one reference, selected at randon out of a whole page of Google search results.
You also need to familiarize yourself with the noise floor to bit depth relationship in digital recording. If your converters run at 24 bit fixed point your digital noise floor is -144.49 dBfs. So your pianiossimo piano is nowhere remotely close.
That's a BIG fallacy right there. What you're talking about here is background noise in the incoming signal. That has ABSOLUTELY NOTHING to do with the noise floor of your audio conversion. Jacking up the level of your digital signal is gonna jack up that input noise - your converter doesn't know the difference between incoming noise and incoming music.
Sure, but what does that have to do with what we're talking about?
I'm not even considering that sort of ruined audio in this conversation. The stuff that I've been doing the past few years is essentially traditionally oriented country - we don't "smash" anything. And although I do have around 16 channels of analog compression, we generally just tickle the comp meters on most stuff. I like music that actually has some dynamic range to it. Not as much as a full blown orchestra, but a lot more than is common in modern pop and rock.
Most people who claim to be MEs these days aren't, as far as I'm concerned. I'm talking about people like Sterling sound, Paul Stubblebine, Chuck Zwicky, and our own Bob O. - REAL Mastering engineers.
Not going to comment on the current crop of mix engineers - I don't want to offend people unnecessarily.
But to get back to the point - Classical music requires a dynamic range of around 80 dB, give or take. A 24 bit, fixed point converter, like an old fashioned Protools box, has a noise floor of -144.49dBfs. 144.49 - 80 = 64.49 dB. If you allow 15 dB for mix headroom that still gives you 49.49 dB above noise floor for your pianissimo parts. So 49 dB isn't enough headroom above noise for you? Really?
And modern converters give you 32 bit float. Even ignoring the float, a 32 bit converter gives you a -192.66 dBfs noise floor. Do the math.
There is absolutely no reason to dfeliver something that peaks over -15dBfs to your mastering engineer.
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Post by Deleted on Apr 2, 2020 14:32:21 GMT -6
However, the thing that a lot of digital guys don't get is that since your noise floor is so ridiculously far down on any remotely modern converter rig, and since digital does vary tone with level like some analog stuff does, you can really track way far down (by analog standards) with no problems. In fact most mastering engineers love it when a project comes in just ticking -15dBfs. There is really no sane reason to deliver a project that's any hotter than that.
And that definitely gives you better odds in your "big gamble" because you can be off by 10 dB, give or take, and still not get yourself into trouble, at least not if you're working with an ME who knows what he's doing.
BTW, most of the guys who really know will tell you that you really shouldn't track channels digitally at levels higher than -15dBfs. The argument gets fairly esoteric, but what it comes down to is that during mixdown you can get intersample peaks that produce nasty digital clipping even when you think you're good. And the problem compounds with the number of channels in the mix. The technical explanations get a bit beyomd me at this point but Bob O might be able to do better.
Eh I've heard that "argument" about never going above -15dB and I don't buy it. While I do agree converters have sounds and tones(everything does). But for instance a piano piece I'm working on right now had a dynamic range of 60-65dB. If I was peaking at -15dB during tracking then the pianissimo section would be at -80dB or so. Which is well into noise floor territory and not acceptable for classical. As so as you boost that up the noise floor comes with it. By Noise floor I mean everything btw. The HVAC of the room, microphone, preamp, it all adds up and is obvious when it's louder all the sudden. And I'm spoiled with an insanely quite space in the middle of no where with no traffic driving by and we can tweak the HVAC to get it quieter(have to be careful not to piss of the piano or it goes out of tune if too dry, too humid, too hot or too cold) I can however always turn things down in the mix in a nice transparent way with riding faders. Like I said though, this is a classical thing. For your standard bass drums guitars vocals recording session and mix down, you're definitely not dealing with that kind of dynamic range and even then it is not just accepted but common place to squash a lot of the dynamics in the name of tone and color. So noise floor isn't an issue in the same way. I bet most ME's out there rarely ever getting -15dB from anyone anymore. Most are happy to get -3dB haha which is sad in it's own way. But it is also just as accepted and common place for mix engineers to put stuff on the mix bus to get it louder and better shape. I do that. You do that. I don't think there is a person on this site that doesn't mix with some sort of chain on the mix bus. Not just classical. I’ve had this problem with very dynamic vocalists that need to be heavily eqed, compressed, and processed. The same with clean guitar leads and clean basslines. Trying desperately not to clip or overdrive the converter leads to too quiet digital tracks that can’t rise out of the noisefloor of the analog gear and analog portions of the AD converters. Raising the gain raises the noise floor. These can’t hold up to processing or being summed together as well as recordings with a lower noise floor. Classical has another problem in that the stack up of tracks compounds all equipment issues, a lot of the engineers are gear deaf, too hardheaded to be told what the problem is, and are using crap like a bunch of Octamics into an older RME unit with audible and measurable IMD. The distortion and digititus can be massive then when everything is summed together itb and what should sound huge and probably did sound huge live, often sounds small despite the huge dynamics. |
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Post by Deleted on Apr 2, 2020 14:55:28 GMT -6
Intersample peaks are a very big deal John but no converter has a noisefloor of -144 dB. The analog parts of it don’t. The switchers in the chip don’t. The mics and mic pres don’t. You’re hitting resistor and switcher electrical noise. And all of that stack up of external and gear noise is just being converted and saved to the file in the computer. The converter outputs it as a 24-bit file but it itself has a lesser dynamic range. A cheap, crappy converter like a Focusrite Scarlet, Presonus, or Steinberg not the AXR4 thing, can’t even hit 16-bit THD+N. A workman like Babyface Pro will measure slightly cleaner than redbook standard. Now that’s just with a 1 kHz sine wave. When used with material that isn’t fixed sine waves, all bets are off. A Babyface Pro sure as hell isn’t that clean in the high end. The RME treble (not as bad as before but still bad) won’t allow that. Stack a bunch of tracks through it and the noise and distortion compound. Hence the constant search for cleaner and more accurate conversion or the acquiescence to conversion that distorts in a pleasing way like the Lavry Saturation and Burl transformers.
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Post by chessparov on Apr 2, 2020 15:01:51 GMT -6
I did notice Tom, what you're talking about in my Prosumer interfaces (Presonus Audiobox/MBox 2 Mini).
Not so much in my Mackie Blackjack. But that has Cirrus converters.
Chris
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Post by stormymondays on Apr 2, 2020 15:08:07 GMT -6
Here's a pretty thorough overview: www.soundonsound.com/techniques/gain-staging-your-daw-softwareI follow that same approach. I gain stage everything as per the most usual analog specs, with 0VU = +4dBU. All our precious analog gear was designed around these levels, and as Al Schmitt said about the VU meter, "it's red for a f****n' reason". So the analog part is easy if you have some reliable VU meters in the gear. Now, in a hybrid studio most of us have the problem of semi-pro converters where +4dBU can equal MANY different levels. The most "analog-like" level conversion would mean that a +4dBU analog signal would create a -20dBFS digital reading at the DAW. Most of the converters we RGO members use don't do that. I'm pretty sure that Avid Pro Tools gear does, though. My RME Fireface 802, on the correct setting, has +4dBU=-15dBFS, which is not too bad. I've chosen -18dBFS=0VU myself in the daw. Thus, I tend to record with levels from -18 to -10 and don't sweat it. Then for mixdown I put a VU meter first (Klanghelm), do my trims analog style, pull all the faders down -5dB and I essentially have a correct starting point. Since my monitoring is also calibrated, I'm never worrying about level from that point on, mixing or overdubbing. I'll also point out that as several posters have said, there is no reason to record hot with 24-bit converters, and it's easy enough to do the experiment yourself if you don't believe it. There is no noise penalty in adding 10 dB of digital gain in the box, at all. Hope that helps! |
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Post by EmRR on Apr 2, 2020 15:54:29 GMT -6
Intersample peaks are a very big deal John but no converter has a noisefloor of -144 dB. Semi-random addition. When I tested my MOTU 16A, I saw -137 dBFS as the worst noise performance at 16kHz. Spec says +20dBu max output, so equates to about -117dBu on noise at that frequency, other freq better, but that's not the cumulative number for all freq. They quote distortion plus noise at -110dB. My previous converters from a decade earlier were much noisier. |
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Post by christopher on Apr 2, 2020 16:48:09 GMT -6
After re-reading the first post.. here’s what I’ve been doing mostly for mixing outboard:I downloaded some freeware analysis like SPAN and made sure I can see the DAW noise floor on the meters either in the DAW or SPAN. Once I have that working, I’ll see where the AD inputs noise floor is actually peaking when nothing is plugged in.
Then I plug the analog stage I want to use... without anything on the input yet..and adjust the output “fader/attenuators” so that the noise floor of analog piece is a dB or 2 near the digital input noise floor, either above or under it, whatever I think is better. The idea being that when I later need to turn up the track in the DAW, the noise floor that is raised will be the less ugly sounding noise floor. A short recording of nothing, with analog noise floor on top and then digital noise floor on top.. normalize them and you’ll hear if both are fine or one is uglier.
Once I have the output noise floor aligned with digital, I get to work on the analog input stage.. pushing that higher and higher to get my recorded headroom as high as I can in analog. At a certain point, saturation will start to happen and it will sound thicker and awesome; at this point decide what I want .. back it off? Or push it more? In the end I figure digital gain is the cleanest option, I’ll probably get the sound I like, record and normalize if it’s too low in dBFS. This is all a pretty new method for me, so far I like it.
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Post by mrholmes on Apr 2, 2020 17:04:21 GMT -6
It’s the Zen of audio balancing noise, overload, and tone. You do what you have to reach your own sonic nirvana.
Some noise can be nice.... With the years passing by....I have my problems with all those gain staging theories....
Pushing the input like crazy in context A can sound wonderful, abusing the same gear in context B sounds awful. A wise man on this MB told me "Use your ears" ( svart)
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Post by johneppstein on Apr 2, 2020 20:37:01 GMT -6
Just for clarity...I understand how input and output works on a mic pre (I’ve been doing this for a little while)...maybe I should have asked how hot you’re going into the next stage...in order not to achieve distortion or clipping. I guess using my ears has served me well for several decades. No audio gear that I know of has specs for compression(saturation or clipping) point. They never tell us at what level their gear will start to clip. IF they did that, then you might start to be able to devise a mathematical way to determine optimal levels, but until then, it's just finding it for yourself through the "listen for the crackle" method. That's why god invented oscilloscopes. |
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