What does Zero dBu actually mean? A comprehensive explanation.
HTML-код
- Опубликовано: 20 мар 2021
- Correction:
05:05 The "U" more commonly stands for UN-LOADED, but is also occasionally written as UN-REFERENCED
What does Zero dB actually mean? A somewhat comprehensive explanation of what Zero dBu actually means and how it relates to the terms "Unity" and "Nominal" gain. I also quickly cover some of the differences in different db Scales like dBu, dBV, dBFS, and dBSPL.
Below are some affiliate links to gear that I used to make this video. These affiliate links really help me out in being able to keep creating content for this channel. Enjoy!
Filmed with the Panasonic GH5 with 12mm 1.4 lens.
GH5: amzn.to/2Q7DcC1
Lens used, Panasonic 12mm f1.4 for micro four thirds: amzn.to/3mAnc84
Awesome Mic that I used for this video and for all my video calls: amzn.to/39SCkbB 
Хобби
I’ve been doing audio for 25 (not full time) years and I still learned a lot from your video. Excellent work thank you so much.
Glad it was helpful!
Finally an explanation of dB that I actually understand! Thank you.
No problem!
I hesitate to comment here because this video is so well done. However, in the interest of accuracy I feel I have to point out a number of inaccuracies. Basically there were a lot of terms thrown around, blurred together, and inaccurately defined that cause me to warn anyone watching this video and reading this comment to Further Research the topic.
0dBu is 0.775 volts as stated. However…
+4dBu is 1.23 volts and the actual reference point we use in Pro Audio.
The 0dB that he is using as a baseline is not 0dBu, but 0dBVU.
0dBVU is what we calibrate VU Meters to when receiving a +4dBu signal, again 1.23volts not 0.775volts, specifically with a 1kHz sine wave test signal, for 1K Tone.
The 0dB on a fader is not referenced to anything, and is Unity Gain meaning that the signal coming in is exactly what’s coming out. If you push that fader up past 0dB, the decibel markings on the board should describe exactly how many decibels of gain the signal is increased by. The Faders on Mixers are often calibrated with say “10dB of Gain in Hand” meaning you can boost the signal 10 decibels above Unity.
Nominal Levels and Unity Gain are not synonyms. The Nominal level is the internal operating level a system was designed to operate at. Often a Console have Balanced connections to the outside world, and is calibrated externally to +4dBu as discussed above. But the first stage is often Unbalancing that signal, dropping the signal level by 6dB, and internally many consoles operate Unbalanced at a -2dBu Nominal Level.
Along with this discussion is also Signal to Noise Ratio and Headroom.
SNR is measured between the Nominal Level and the Noise Floor (specifically defined and measured).
Headroom is the difference between the Nominal Level and Distortion or “the Ceiling” (again specifically defined and measured).
Lastly, the important calibration numbers or “Standards” to remember are as follows:
1kHz Tone @ 1.23volts = +4dBu aka “Line Level”
+4dBu = 0dBVU
0dBVU = -18dBFS calibrated A/D & D/A Converters in the digital realm (EBU Standard. SMPTE is -20dBFS, but the Europeans won this fight)
In a perfectly calibrated system, that 1K Line Level tone should be able to make it through every piece of Analog and Digital equipment you own completely unmolested.
Again, I’m sorry for chiming in. And possibly nobody will ever read this. But I felt responsible to the audio universe to prevent someone seeking information from learning lessons that they’ll later have to unlearn, the hard way.
Thank you so much for this.
Hi there G. I’m glad you didn’t hesitate, thank you for the comment and all of the very useful information.
To preface things before my reply…….the reason I made this video is that out in the live-audio, working world, which I’ve been a part of for over 18 years, I’ve run across countless “audio engineers” that simply don’t know the basics, they don’t know what they’re doing, and they’ve learned by either just turning knobs and luckily making something work, or they learned from someone else, who seemingly did the same thing……..turned knobs until it worked, or learned from someone else. And bad habits have gotten passed down, around, and through the local audio industry (and broader), and most of the audio people I’ve worked with simply can’t answer basic questions about the audio signal or their equipment, or they have it bassackwards, and that has gotten countless audio “engineers” in a spot they couldn’t get out of in terms of feedback, signal flow, and gain structure.
And when it comes down to basics, 99% of people that I’ve run into, simply have no idea the difference in different db’s (dbu, dbV, dbSPL, and dfFS). It frustrates me so much, I wanted to make a video to just start at the base level.
So, that’s where I started from, and when I said around the 2:25 minute mark that “people most often refer to unity or nominal gain…..”……it’s these inexperienced people that are ACTUALLY hooking up audio equipment and attempting to run it that are giving me answers like this, because they just don’t know the difference, they don’t know how to gain structure their equipment to get everything up to line level appropriately, and so they’re throwing out terms they have heard like “unity” and “nominal” and they just don’t know.
And honestly, some things I did not know that you pointed out is that 0dbVU is a reference to +4dbu, so, thank you for that information, as I have now done a bit more research. But in my 18-20 years doing live-event production, and previous schooling before that, I don’t believe that bit of information ever came up, we always only learned dbu or dbV, mainly as it related to pro/consumer line levels……and we left VU up to VU-metering, etc.…….but that does NOT negate the fact that zero dbu is STILL referencing .775 volts of electricity, it still is that. Whereas +4dbu is 1.2228 volts. Obviously more dbu’s = more voltage on the same scale. But most audio consoles that we use out in the live-event production world don’t, or rarely have VU meters, so, all of the LED Meters on the board are typically in dbu scales, so gaining inputs from mics up to 0dbu is typically (*should be) the starting point of input-gain for a console…..get the input gain up to measuring Zero dbu on the input meter…..then the fader, if at zero, won’t boost or cut that signal through the console. So, inputs/faders, still referencing dbu most of the time…..not *really using a dbVU scale.
However I now notice at 7:23 when I put that image of my Mackie Big knob (lower right hand corner) that it literally says “0=+4dbu”…………ha!!! I didn’t even see that on my own damn meters!!! I am, occasionally, only human and still make mistakes…but yeah, wow, didn’t even see that.
And your are correct of course that Nominal Levels and Unity Gain are not synonyms……..again, I am aiming this video towards the countless “engineers” out here on the street that are put in charge of being “A1,” and not knowing what they’re doing. So, yes, you are correct, they are NOT synonyms…….I never said they were, but when I ask young “audio engineers” questions like “what does zero 0bu mean?” they start throwing out answers like .......”um…….unity gain right? no wait, nominal, that’s what it means.” But they aren’t actually giving me an answer, because they don’t know…..they think 0db, nominal, unity are all the same thing, and, like you said, they are not.
I love all of your info because yes, it’s absolutely all correct, and I certainly want to put out the correct info, I hate to misspeak, and I do not want to be someone propagating incorrect information as well, so…….again, I greatly appreciate your comments, it makes me need and want to keep criticizing my own work to a higher level so I can also get better.
Cheers, and thank you again G.
Also, I think my main point to a lot of people comes at 10:42 - 11:30......because the audio operators out there in live-event production-land are just not paying attention to proper gain structure.
or, or.........of course, there is the very distinct, obvious, possibility that I am wrong in one sense: If my Mackie Big Knob (and therefore possibly all of Mackie) have decided that "0" on their meters actually refers to a voltage of 1.228 (+4dbu), then, it stands to reason that other audio professional audio equipment may be doing the same.......so, gaining an input to 0 on their meters, may actually mean +4dbu......because they've simply shifted the scale. Which, kind of irks me because sometimes those faders are still listed as dbu, and if they are, one should assume zero dbu on that fader/scale should/would equal .775 volts. However, I guess I need to read reference materials more closely to ensure the numbers on the LED scales, etc, are or are not shifted to reference +4dbu when they're "at zero."
Thanx G, you've given my brain more to think about, study, learn, and get better at!
The Irony...
In 1996 I bought both a Mackie Big Knob, and a Mackie 1604VLZ Mixer (along with a Pro Tools Project system on a PowerMac 9500!).
I still have, and occasionally use them both, and know them quite well.
Your comments reminded me of something, and I just found it to my and hopefully your amusement.
In those days (and still now for some of us) you "RTFM"! On page 3! of the 1604 Manual you get this:
(1) LEVEL-SETTING PROCEDURE
Message to seasoned pros: do NOT set levels
using the old "Turn the trim up until the clip
light comes on, then back off a hair"' trick.
When a Mackie Designs mixer clip light comes on, you really are about to clip.
This procedure really works-it assures
low noise and high headroom. Please read on.
It's not even necessary to hear what you're do
ing to set optimal levels. But if you'd like to: Plug headphones into the PHONES output jack, then set the C-R PHONES knob about one-quarter of the way up.
The following steps must be performed one
channel at a time:
1. Turn the TRIM, AUX send and fader
controls fully down.
2. Be sure the 1-2, 3-4 and L-R channel
assignment switches are all disengaged.
3. Set the EQ knobs at the center detents.
4. Connect the signal source to the MIC or
LINE channel input.
5. Engage (push in) the channel's SOLO
switch.
6. Push in the MODE switch in the output
section (LEVEL SET (PFL) mode)-the
LEVEL SET LED will light.
7. Play something into the selected input, at
real-world levels.
8. Adjust the TRIM control so that the
display on the meter stays around "0."
(Only the left meter is active in the
Level-Setting Procedure.)
9. If you'd like to apply some EQ, do so
now and return to the previous step.
10. Disengage that channel's SOLO switch.
11. Repeat for each of channels 1-16.
...Later in the manual, in the section on metering you get this:
Meters vs. Reality
You may already be an expert at the world of "+4" (+4dBu=1.23V) and "-10" (-10dBV=0.32V) operating levels. Basically, what makes a mixer one or the other is the relative 0dB VU (or OVU) cho-sen for the meter display. A "+4" mixer, with a +4dBu signal pouring out the back will actu-ally read OVU on its meter display. A "-10" mixer, with a -10dBV signal trickling out, will read, you guessed it, OVU on its meter display. So when is OVU actually OdBu? Right now!
At the risk of creating another standard,
Mackie's compact mixers address the need of
both crowds by calling things as they are: OdBu (0.775V) at the output shows as OVU on the meter display. What could be easier? By the way, the most wonderful thing about standards is that there are so many to choose from.
Thanks to the 1604-VLZ PRO's wide dy-
namic range, you can get a good mix with
peaks flashing anywhere between -20 and
+10dB on the meter display. Most amplifiers
clip at about +10dB, and some recorders aren't
so forgiving either. For best real-world results,
try to keep your peaks between "0" and "+7."
Please remember: Audio meter displays are
just tools to help assure you that your levels
are "in the ballpark." You don't have to stare at them (unless you want to).
Thank you sir! After 20 years of dj-ing on Xone mixers I finally have a technical understanding of what I’ve been intuitively doing! Excellent explanations👍
Thank you! I’m glad the video helped :)
Maaaaaaaaaaaaaan, you explained in 11+ minutes what I’ve been trying to understand for the past 11+ months!!! Great work!!! I’ve subscribed to your channel.
damn man... you gave so much information its amazing. (found you by searching things on the car audio side of RUclips). its like none of the super famous car audio guys dive deep into this sort of stuff but its shocking how it sorta ties into things. you really "wowed" me with the DB being different from the db we use in reference to hearing thing; definitely learned something new here. thanks very much for the new-found knowledge!!
Hi there Davonte, thank you so much! I'm glad the video helped and I'm glad I was able to provide some new and interesting information for your brain. cheers!
ExcelLent job sir! This video deserves 90 likes per viewer! If it had more graphic/animated explanation, it would have deserved 100!
Glad it helped!
Incredibly informative. Thank you for this!
You’re welcome. So glad it helped.
Best explanation I’ve seen or read in 10 years of trying to understand this. Thanks!
Hi there Buzz. Thank you so much, and you are very welcome. I was a bit confused for years when I started out as well. And then I simply referred back to my very first audio book from audio school, and there it was, plain as day. The thing is........NONE of my instructors over the years EVER mentioned this fact. It's in the literature, but nobody ever talked about it in audio school, nor did anyone talk about it when on show-site, building a concert, building monitor-world or a FOH rack, setting up a stage, etc. And, to be honest, after so many years, I finally got the feeling that most "audio people" I ran into simply DID NOT KNOW! Anyway, I'm very glad this helped you, cheers!
Dude, hands down the best description ever. I finally get it. Thanks a ton.
You're welcome, I'm so glad this video helped :)
REALLY good, thorough explanation. Even so, I need to watch again!
Yeah, I realize now I need another/follow-up video to this......I've just been thinking about MORE information we audio-folk need to know and just been figuring out how to say it succinctly.
Just remember the main thing I'm referring to in this video specifically is 0dbu. That "u" after the db is very important, it's a specific reference point.
Whereas dbV is different, dbFS is different, and dbSPL is different.
But MOST of the time, when we're talking audio signals, we're either talking dbu (analog audio), or dbFS (digital)........but even some digital consoles still have some metering in dbu.......
Finally got it!! I always used 12 o’clock on my gain and turn up the fader label … now I know that I have to use the gain to reach 0db then use my fader.
Exactly!
Very well said, perfectly explained. Thank you so much. This was way easier to follow than years of classes. yeah... I went to school for audio, and it was never explained this well.
Thank you EvanW, so glad this was helpful. Yeah, looking back in my audio-notes and books from 20 years ago, I don't recall ever being taught this simple fact, and in that same 20 years I never encountered any "audio engineer" that could tell me the answer either. :)
Great explanation! Thank You.
You’re welcome. And thank you as well :)
Awesome, simple, effective explanation. Thank you.
HI there. Thank you so much.
Excellent presentation, thank you.
Thank you! and you are welcome :)
Thanks for this very informative video about dB
Glad you liked it! You're welcome :)
Dang, best and most comprehensive but succinct explanation I've seen!
Thank you Don :)
Totally agree!
very excellent ---- articulate and intelligent - nice production values
Thank you so much Thomas!
Thank you for your wonderful explanation⭐️⭐️⭐️⭐️⭐️!
You’re welcome Cindy :)
Very useful information
Thank u. I’m glad it helped :)
This is a great video
Thank you Cuby. I hope the information was helpful. :)
really good. Thank you!
Thank you. And you’re welcome :)
I am rapper that finally took up producing my own stuff and dBu was the next thing I needed to understand. I will be subscribing
Hi there MSSpeak. Awesome, I'm glad my video helped. Always remember that audio signals are just electricity and mostly relate to lower voltage (mic level) or higher voltage (line level), AND, to be able to know where your audio levels sit at every point in your audio signal chain.....always starting at the beginning, the the mic.....then moving the preamp and ensuring you use the Preamp to properly gain your mic-level signal from your mic up to line level for your console or DAW (pro tools, etc).
Strong and proper audio signals at the beginning (and especially at/with the preamp) will ensure you keep the amount of system noise or "hiss" to a minimum (signal-to-noise-ratio).
Happy recording and good luck!
@@MarcusHutsell in theory I understood what you said, I tried applying it in a recording I did for social media. It sounded way better and I wasn’t as worried as much about clipping. It’s so simple I was overlooking it, mainly setting the gain on preamp (which I imagine is jargon for audio interface?). On this topic, when you say “line level,” do you mean xlr vs usb? Does “higher voltage” mean better quality cables (lines) and also higher voltage equating to higher quality sound? Thanks!!!
,
I know this is an old video. And I have no real interest in audio engineering, I am an amateur radio guy. But this video was so clear and useful as background info. Thanks.
Wow!! Thank you so much!!!!
You’re welcome :)
Good video 👍
Thank you Sponge Knock :)
Thank you i learnt so much...❤
You're welcome 😊
Thank you so much 🌷😇🌷
You are welcome :)
Thank you very much :)
You're welcome! However, I will note one mistake (apparently a typo) I made at 3:42:
+4 dBu actually equals 1.228 Volts, not 1.778 volts!!! I can't believe I didn't catch that, I must have just mistyped or copied-pasted something, I'm not sure. But if you need to know exact voltage levels and conversions, I use this Extron calculator: www.extron.com/calculators/db-to-volts/?tab=tools
thank you
You're welcome!
Is there an exception that can be made to go above 0db for a documentary soundtrack that's just not quite loud enough @ -0.01 db??? Or should I totally avoid this to be on the safe side?? Much appreciated
I have a question. What is happening when the fader moves from 0 (nominal) to a higher value? If the fader at 0 is passing the signal (voltage) it got from the preamp where does the added voltage (signal) come from when the fader moves above 0?
Hi again Buzz. Very good question, and, since I am NOT any sort of electrical engineer, I can't speak to that answer with 100% certainty, but, from what I do gather, there is an additional line-amplifier built in to the signal chain after all the effects/sends, and before/in-line with the fader, so that when you need to "push up past zero" that the line-amplifier is literally, then, amplifying your original signal to pass MORE than than your gain voltage past/after the fader. Something is amplifying it inside the console if you gain up to zero dbu at the head-amp, and then are able to push +10db, etc, additional at the fader.
So... Question... My church has an X32 compact mixer, and we do have our pastor/guest wireless microphones at or around the 0Db mark, and our main speaker slider(fader?) Is basically constantly at 0Db, yet the worship mic's and instruments (keyboard and guitar) have their individual channel sliders(faders?) Just willy nilly wherever they need to be, without regard to needing to be stuck at a certain Db level. The question then, is, are we specifically doing it wrong, or is there a reason to do it the way we are...
For background on my situation, I'm the (volunteer) audio tech guy for my church and all I have to do is change something if I'm asked to, else it's basically all preset before I get there, and I'm looking for information on how I can improve the audio, although it sounds ok to me... The other thing is, what's ok to me might not be ok to someone else, as I don't have very good hearing in one ear...
Oh, also, even though I've been doing this off and on for a few years like maybe 5 or 10, I would still consider myself a noob in this particular field so I hope I have my information correct on my question
Hi there. I'm assuming when you say your other faders (worship mics, keyboard, guitar) are just "willy nilly," that generally they are (probably) lower than zero db on your fader? AKA, they are lower in "volume" coming out of your PA than your main-speaker mic?
You're not necessarily doing things incorrectly, because that situation (your other mics/faders being lower), would therefore be an accurate VISUAL representation of your mix; AKA, your faders are set at the volume you want to amplify in the room more-or-less. So it makes sense your band and other instrument's faders sit lower than your main speaker.......IF you are gaining most of your inputs to the same level on the INPUT meter. (Which I still think is the better way to do it, since it will produce less noise at your destination).
For Example...... drums make WAY more acoustic volume (SPL) in your space even without putting a microphone on them, They're loud to begin with!! Therefore you don't need to amplify them AS much as someone's voice because the human voice is just a lot quieter than a drum. And, IF you are gaining your inputs up to around zero dbu on your input meter of the x32 (which is probably equivalent to somewhere between -12 and -9 on the X32's DBFS input meters), then of course your fader position will most likely end up with the drums sitting with lower faders than your talking vocals.
Some live engineers start by setting their faders at zero and then "gaining" their inputs.....and, while that method works, it has its detriments, mainly in terms of adding potential system noise if your input gain levels are very low (regardless of final fader position). I don't like this method because it teaches people the wrong way to approach gain structure, it means your other inserts, EQ's, buss-sends, compressors, etc, won't activate properly if your INPUT gain to that channel is too low: Everything DOWNSTREAM of the preamp RELIES on the input level AT the preamp......less input gain means less ability to send that input to a buss, less input on the gain means you'll have to turn your compressor threshold WAY down just to grab and compress your signal.......and it means your automixeers *MAY not activate properly (we use automixers a lot in corporate audio with multiple people on stage speaking at a time), and automixers can really help with feedback control and acoustic noise.....but if input-gain-structure is low or bad, the automixer MAY actually not open at all when a person is speaking, acting more like a gate, and that can ruin a live show..........all because the engineer didn't pay attention to proper INPUT gain on the individual channels.
I made a couple of videos just recently showing how improper gain structure leads to more noise, I'll link them below.
Audio 101 - gain and noise - Part 1: ruclips.net/video/6KbR-hb4HS8/видео.html
Audio 101 - gain and noise - Part 2: ruclips.net/video/d3hVYH0xVNg/видео.html
Also remember that gain structure is a balance between all of your different types of inputs. Dave Rat has a great video on this: ruclips.net/video/-RXlnVDVLrc/видео.html
MOST IMPORTANTLY: As long as you understand every aspect of your gain structure, inputs, outputs, fader position and understand what CHANGING any/all of these things will do to your sound/signal (good or bad), then you can make informed choices about how you want to structure things and the results of those choices. There's nothing wrong with running some inputs a little lower.....as long as you know the effects of doing that.
I'll also put some links here to some other channels (and specifically one guy who does probably exactly what you are doing in a church environment).
This guy is great!! www.youtube.com/@AttawayAudio
ruclips.net/video/R7zKiFErHs4/видео.html
Gain Structure from the pros: ruclips.net/video/1OIbpbJ0Zv4/видео.html
Good luck!
@@MarcusHutsell I think I understand most of that... Yes, when I say Willy nilly I mean that guitar, piano and vocals are not at unity gain for their individual volume sliders, so the piano might be at -30, the guitar might be at -20 and the voices of the singers might be anywhere between -40 and -20, whereas the pastor/uest speaking mics are usually +/-5Db from unity...
Remember, I am just using this as a demonstration, as I don't have the mixer right in front of me or anything(so my numbers won't likely be 100% accurate), and also remember that our church didn't do the original setup for the mixer, we just took it after it was set up and pre-tuned (the installer/company owner used to go to our church and he's always been rather intimately involved with our system and it's upkeep from basically as far back as I can remember) and all we have done usually unless we need to change something on the monitor(s) (like individual monitor instrument volume), is to change the individual volume sliders. I honestly don't mind that we do it the way we do it now, mainly for the reason of it does work I'm just making sure that I am doing it correctly as opposed to not doing it correctly because of how we've always done in the past and how we've done since the installer pre-setup from memory what all the mics and instruments should be for gain and whatnot...(he might have had the worship team there for a quick sound test/tuning, but if he did I wasn't there.) The only thing I know for sure that was tested, was the bass in relation to the computer input on the mixer so that we could have somewhat proper sounding audio through the computer's aux/headphone port, although I don't know if the settings that the installer changed were just for the aux ports in question or the stuff on stage as well.
Again, thank you for the information you gave me, I'm definitely going to look at a lot more closely when I have more time to, and see what sense it makes. It's definitely way different than our old analog board, and some aspects are easier, while some still feel harder
Also, another question, sorry, I'm so used to seeing either on TV or in person or online or wherever when I see a digital VUmeter, the LEDs or whatever it is that is on that particular meter style, I always in usually the top of the green or into the yellow whereas hours doesn't generally do that or at least it might now but hasn't in the past at least on our old board, and I'm wondering if it has to be way up there on the visual audio representation of the vertical meters? (or horizontal depending on setup)
Great explanation.... So, how does this all relate to VU ?
Hi there. So......VU still measures voltage, it's just a different scale and staring point.
0VU = +4 dBu.
So, they're close enough in voltage levels..... 0 VU, 0dbV, and 0dbU are all very close to 1 volt, but still they are not quite the same measurement of voltage when at their respective "zero" origin points.
@@MarcusHutsell Hi thanks for that explanation.. I split my time between non broadcast audio and Broadcast audio and alway wondered about the difference in levels between those different worlds. Broadcast seems much "hotter" than non-broadcast, to the point were comming off the broadcast board i need to pad down the lines to keep the non-broadcast board for pegging the needles...
@@sambaker7255 Hey Sam. So,.....I'm mostly in the "non-broadcast" world of live-audio.......and I can say this after 20 years in the business........the "live-audio" engineers (at least in my city) are mostly gain structuring COMPLETELY WRONG, which is WHY you see non-broadcast "not as hot." It's a very big problem that I'm trying to combat on a daily basis. I have actually been sitting at my laptop the past few hours typing up a script for yet another gain structure video.
The problem is that most "live" engineers have learned on the street from other live-engineers, and a lot of bad habits got started and keep getting passed down....and that mostly involves gain structuring backwards.......AKA, turning up the amps to full, and reducing the "volume" with their input gains. Drives me crazy and they don't even know what they're doing to their audio because in large rooms and PA's, you *may not hear that noise......but it's there, and they wonder why they have problems and can't fix them.
So........stick with your broadcast standards....get those audio levels up to that nice, strong, line-level 0dBu electrical audio levels and just do it the right way......speakers and amps come AFTER gain structuring mics and consoles.
Anyway, rant-over, cheers! :)
Granted… I ranted a bit, probably odd topic as I may have misunderstood your comment at first, you may have only been talking about how the level coming out of a broadcast console seems to be quite hot for the input of a non-broadcast console….Which makes sense, but it’s all about voltage levels… So as long as you know what the DB values represent in voltage to then you can always match inputs, outputs, and vice versa. I have many XLR pads in my gear packs to compensate for such things.
Very very useful information. How do i figure out what 0 dBu level is on my Pro Tools dBFS meter? Is it -6dBFS --12dBFS -10dBFS or -18dBFS? Is this where i should set my preamp gain to be above the noise floor to get a clean vocal recording and below 0dBFS to avoid digital distortion? Thank you Marcus.
Hi there Shaun. Thank you, and your question is really on-point, and that's where it gets tricky because none of the digial companies agreed on "what level of DBFS refers to 0dbu," so, it varies across manufacturers. I think Yamaha Digital consoles are around -12 DBFS, but some of my equipment it's closer to -14 DBFS, and some it's -18 DBFS. So, yeah, that's the tough part, the companies didn't agree on a "relative" level. But, it's generally in the "teens" of dBFS.
When I'm mixing in DBFS (Pro tools, Audition, Premiere, Garageband, Logic), I always just mix to get my signal hitting around -6dBFS for the final mix after all compression, amplification, EQ, etc. Granted, that has nothing to do with INPUT gain of the mic/preamp/interface, but I generally think of input gain and related DBFS meters as "as soon as the first yellow/orange meter starts lighting up over the green meters, then that's roughly a 0 dBu 'equivalent.'" That's not exactly a scientific example, and, I haven't used Pro Tools in a while, I'm typically using Audition or mixing directly in Adobe Premiere. but on most of my digital recording eqiupment like my Zoom F8N, or an AJA KiPro, or other field recorder......you can set all your "faders" at "zero" so they're not amplifying or cutting the signal....put a multimeter on pins 2 & 3 of the XLR outputs, and start sending a signal into the preamp and watch your multimeter. The Zoom F8n has +4dbU outputs, which should mean that, when sending everything at "unity" through the recorder, that the output should read +4dbu which is equivalent to 1.23 volts. And, again, ensuring the faders are all at zero (including the master fader......so, no faders are boosting or cutting the incoming signal), then on my Zoom f8N I get a reading of about -14 DBFS on the input (1k Sine wave) when my multimeter on the output reads 1.23 volts. Telling me that -14 DBFS on my F8N equates to about 0dbu.
Why 1.23 volts for +4 dbu vs .775 volts for 0 dbu?? Because again, the output of the Zoom F8n states that it's a "+4 dBu output," which means when the signal through any console is all at 0 dbu (think a Mackie 1604, etc), then the output will actually boost it up to +4 dBu, RIGHT at the output stage, even if the master fader is at 0. Same goes for my Zoom F8n. ..........make sense?
Still hard to tell in a system when you have more components, but, it's a good place to start.....multimeter on the outputs and watch your voltage :)
@@MarcusHutsell Over and over again I would like to say thank you. This really help.
@@shaunblack3142 You're welcome Shaun :)
You seem to be conflating OdB fader level with OdBu. In some consoles this may be the case, but in general pro consoles and the Zoom devices you mention are references to +4dBu. That only means that if you inject a +4dBu signal to a channel ,with its fader at 0dB and the master at 0dB, you will get +4dBu at the output. If you put in a 😊0dBu signal it will come out at 0dBu because 0dB on the fader is unity gain for the channel. So 0dBu will not get boosted to +4dBu in a +4 device.
very nice learning here ! so can above 0 db u/v/fs/spl damage a speaker when mastering a song? coz they always advice to put a limiter at the end of the signal chain. tnx
Hi there Rets. Great question....so, yes, when your audio goes OUT of your computer and back into the analog world via XLR, 1/4" or other audio cables, it then goes back to a dbU reference voltage because it's standard AC voltage within those audio cables. Remember that dbU can go as high as +24dbU (maximum) on some professional audio equipment, it just depends on the equipment AND how much you turn up the output volume of that device that feeds INTO your speakers. The next step would be determining how much dbu or voltage your speakers can actually handle. If they're rated for on +18 dbu maximum, then if you feed +24dbu into the speakers if could damage them. However, +18 and +24 dbu is a lot of voltage and will equate to a LOT of SPL coming out of the speaker and you'd probably damage your hearing if you were feeding that loud of an output (voltage) through a speaker and amplifier.
Most equipment states that it's output is either -10dbV (.316 volts) or +4 dbu (1.23 volts)....and most of the time in my home studio, when my equipment is at maximum output for a loud mixing environment, I'm pushing probably +4 or +8dbu....but at normal volumes I'm pushing about -8 dbu.
So, that's a lot of info to really say........if you're talking about dbu and analog audio and amplification.....you'll be just fine with most equipment and yes you CAN go way above 0dbU on analog audio equipment, and I don't have any analog audio compressors or limiters on my analog equipment at home......however, I HAVE had compressors and limiters on analog audio equipment in a live-music environment to protect the amplifiers and speakers within the theater where I worked.
If you're talking about the audio INSIDE a computer like mixing in Pro Tools, Audition, Apple Logic, etc.....then we're talking about dbFS or db Full Scale, and that has a maximum ceiling of 0 dbFS....so no, you NEVER want to go over zero dbFS inside a computer, because it will automatically sound distorted. When I'm mixing audio within the computer in Audition, and even Adobe Premiere, yes, I almost always put a limiter on the master track output of my software so that none of the audio ever hits 0 dbFS. That's a good, simple way to keep your audio from clipping.
@@MarcusHutsell ok ! i get it thanks for the detailed knowledge ! but there are another info about headrooms for 16-bit and 24-bit audios, in which case the 24-bit have a higher headroom before clipping/distortion occurs. is 0-dbFS absolute for both 16 and 24 bits audio ? or you can go beyond 0 with 24-bit ?
@@retsmej Hi there Rets. Good question. 24 bit and 16 bit both have the SAME maximum limit of 0 dbFS, so, no, 24 bit can't go "higher" than 16 bit. The difference is that with 24 bit, there are more individual steps or values that can be assigned to the "volume" level of your waveform. To simplify it, 16 bit has "16 levels" it can choose from to determine the proper amplitude to store that information as ones-and-zeroes, and 24 bit has "24 levels" it can choose from to store that amplitude as ones-and-zeroes. But, they both still have the same MAXIMUM limit. So, if someone were to state that 24 bit has more headroom, that's not exactly correct, because it will still clip at 0dbFS, HOWEVER......if you're pushing your volume/amplitude levels high, then 24 bit will have more values it can assign.......and....if a high peak of amplitude reaches "95 %" for example......24 bit may be able to assign "95" to that value, whereas 16 bit may round it up to "100" and you're clipped. So......same maximum limit, but more values can be assigned within 24 bit. Although it's still not a great idea to push your volume/amplitude levels THAT high anyway, because you'd always be close to clipping. 24 bit is technically more dynamic because it can store more values that relate to the amplitude of your waveform.......can we really tell the difference in our ears? Probably not, but more information for the computer is usually a good thing, to an extent.
I hope that helps!
@@MarcusHutsell wow ! i got it now, you explained so well . thank you very much !
dBu means unloaded, not unreferenced, there always has to be some reference as dB is a relative quantity . When an audio fader with +10dB at the top is at 0dB it is really at 10dB attenuation and then followed by a 10dB gain stage. This is not the best gain structure although nearly all mixers do it. VCA faders are different as there is no gain until it is invoked with a CV, with audio faders the gain is there all the time.
Hi there. This is weird, because when I made this video (3 years ago)......I had to do a little research myself to actually find out what the "u" in dBu meant because #1, I don't recall ever learning it in the (somewhat) limited audio-school I had, and #2, I want to put out accurate information.
So, 3 years later, after your comment, and doing some more searching, I now find that most articles do, indeed, "refer" to the "u" as unloaded. Which leads me back to questioning WHY I would have said "unreferenced" in the first place. Either I just completely mis-remembered the term when writing my script, or I just had a Mandela-effect moment. However, I do still find some articles that list it as "unloaded OR unreferenced,".....and to me, "unreferenced" does make sense because there is no reference to the audio-lines having any specific impedance on them.....then again, I am certainly no electrical engineer, and I do not claim to be very good at knowing all about impedance and ohms in those electrical circuits.
So, I really appreciate this, I now wonder if I should either edit-out that part, or just re-record the video, I really don't like false info.
@@MarcusHutsell Hello, you are still confused and confusing. dBu cannot be unreferenced, it has exactly the same reference as dBm: one milliwatt in 600 ohms. The dB is a unit of relative power and power in a resistor is V^2/R. For dBm the R is 600ohms for both values and it cancels out. If you actually were interested in power there is a correction factor of 10log(R2/R1), but in most audio we are more interested in voltage ratios and ignore the resistance (within reason). It is still a power ratio and so the squared voltage values have to be used, that's where the 20 in the formula comes from. Most meters measure the voltage and convert it to theoretical power, on a VU meter the needle is deflected by a voltage and the maths is done implicitly on the scale markings (that's why they are uneven), on a multimeter or computer the maths is either computed or looked up in a table.
0dBu is not a good working signal level, most modern circuitry works better at +4dBu to +14dBu and will still have headroom up to over +20dBu.
why does my soundcraft desks gain pots read: 0 , U, -20, -30, -40, -60 (clockwise) ? ......surly turning the gain pot up increases the dB, does anyone know ?
That's interesting, what model Soundcraft mixer are you using?
Spirit Folio SX @@MarcusHutsell
0 = +4dBu, then what will the led meter reading when I feed +24dBu signal ?
Hi there XO. So, the weird thing is that the visual example I showed was on my Mackie Big Knob where it says "0 = +4dbu," was a bad visual example, because MACKIE specifically decided to change the visual reference point and what their "zero" means on their own equipment (so, be aware of that with some Mackie Equipment. Despite me owning that Big Knob for many years, I never noticed, even though it was there staring me in the face!!. So, a lot of Mackie equipment will actually be at +4dbu (1.228 Volts) when that specific piece of Mackie equipment is at "zero" on its meters. I didn't take my OWN advice of reading the labels and the user manual to note that.
I think this is/was pretty bad of Mackie to do that because it kind of changes what people are looking for or looking AT on their analog meters. But, at least Mackie did stamp the reference right on the front so that we know what's goin gon. Anyway.......that's a little beside the point. But it is important to note that ONLY on specific Mackie equipment does "0 = +4dbu," unless otherwise noted.
But almost no other manufacturer that I've seen does this.......but again, always important just to check the scale of the meters on the equipment to see their reference point.
To answer your question, just use any online dbu to volts calculator. I like the one over at Extron. www.extron.com/calculators/db-to-volts/?tab=tools#:~:text=dBV%20refers%20to%20a%20voltage,is%20referenced%20to%200.775%20volts.
So, if you type in "0" into the dbu field and click "calculate," you'll get .775 volts. If you type in +4 into the dbu field, you'll get 1.228 volts.
So, if you type in 24 into the dbu field, you should get 12.277 volts. So, if you're sending +24 dbu in/out of an audio point in your chain, it should be sending 12.277 volts of AC current. So yeah, 12 volts at +24 dbu!!!
So, as long as our meters on our equipment state what their reference is, then we can know, for example, that if our meters go up to +24 dbu, then at that level, we'd have 12.277 volts going in-or-out of that point.
Most consumer audio equipment isn't going to be able to output +24 dbu, but other pro equipment and audio mixers can. Take the Allen and Heath QU32, it can push out a maximum of +22dbu, which equates to 9.752 volts.
Use that Extron calculator, it's great!
Cheers.
@@MarcusHutsell Thank you!
As an electronics guy its weird to me that all this info is so hidden. Like in electronic stuff Im used to datasheets saying those are the input levels in voltage, and the outputs levels and lets add a couple of graphs if you want more details. Mixer datasheet is like yeah so this is an headphome output and this is the stereo output, same plug but 0 zero details about the difference between the 2 lol.
The odd thing is that, I never remember ANY of my teachers talking to us about voltage, db, dbu and/or how the voltages relate to mic level/line-level, etc. We did a lot of things and worked with a lot of gear in my relatively short time in audio-school.....but I just don't remember ANYONE explaining mic level vs line level.....just that they're "different." (then again, I could have just forgotten that part of the curriculum, but if it was taught out-loud.....then I forgot somewhere along the way).......until a few years into audio which was about 20 years ago, when I noticed a lot of newbies just didn't understand what they were doing, and I wasn't 100% sure of a few things, so I grabbed my audio recording book from circa 1999-or-so......and there it was on one of the pages:
0dbu = .775 volts.
So, the info was there, but it certainly wasn't emphasized in school.
I also DO find it in many user manuals, but it's just not talked about in the real world which REALLY makes a lot of people operating audio just NOT fully understand their gear or their gain structure.
So, I've been trying to educate not only here on RUclips but also in my daily work in live-event-production. It seems to be an uphill battle!! But I'm trying!
Cheers :)
i'm sorry mate, its not AC (ALL audio circuitry) is DC. it would be physically imposible for such circuitry to run on AC.
Hi there Ahsim. What I mean in the video and what I think I stated well enough, is that the audio SIGNAL is AC, all audio signals must be AC (alternating) or there isn't any sound. *Especially before it enters any audio console, it's all analog (mics, XLR cables, etc). And, yes, modern digital consoles, being basically a huge computer, and being powered by an AC to DC converter power supply, are probably "keeping the power on" inside and performing their operations running on DC power, because computers (laptops, etc) need that constant voltage for their circuitry to work. Granted, I'm no electrical engineer either, so I don't actually know the specifics of how the mixers are processing the incoming AC Voltage from our XLR lines........either a DC offset where the AC voltage of the audio signal rides on TOP of the DC voltage needed to power the console, or whether it gets digitized immediately, in which case it's still represented as an alternating waveform by sampling. Once the audio mixer is on, powered, and running, the audio signal or voltage that it ACTS upon must be in an alternating form, as that alternating form represents the actual sine wave(s) of the audio signal. So, even if the circuitry is running on DC, it must have an AC voltage of some type to do something with....AKA, the audio signal itself. Again, this can be either the AC voltage riding ON TOP of a DC voltage, a DC offset, and the AC voltage riding on top of that DC voltage is filtered out later to produce the original sine wave(s).......Leslie speakers operate like this.
If we're talking purely digital, then the analog, AC voltage coming in from the microphones may get sampled and turned in to a digital signal as soon as it passes the preamp and it all becomes ones-and-zeroes inside the console..........but that digital representation of the audio signal is STILL represented in the computer as an alternating waveform.
So, I guess what I mean in the video, and I don't mean to mislead people, is to understand that audio signals are an alternating waveform and that when talking about zero dbu, we're referring to .775 v of alternating current electricity. Like the signal flowing through an XLR cable, if it's being sent out of an XLR output from a mixer over to a speaker.........it's most definitely AC Voltage.
Thank you for the comment. If you have any other specific electrical engineering (as related to audio consoles, etc) to send my way, I'm always up for more knowledge.
Thank you Ahsim.