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Phil O wrote:I may be slightly off topic, but I think this ties into this discussion.

One thing I try to keep in mind when tracking is what I call bloom. You set everything up, your musician/s are warmed up ready to go and you've checked levels. Now you find that by the end of the first take everything has bloomed (increased by 3 to 4 dB). Whatever recording level you subscribe to, be sure to give yourself 3 to 4 dB of bloom room.

Incidentally, it almost NEVER goes down or stays the same, it usually goes up, even with seasoned pro musicians.

Phil

That's the idea behind -18db rms. You have more than enough room normally for when that happens.

I've played with the -15dBfs to -20dBfs thing for several years now. I have noticed that newer mixes from tracks recorded in this range (RMS in this range) sound much more open. Though I have not read through this entire thread (it may have been mentioned), one thing that you need to consider ITB with hot levels is plug-ins. Sure, DP has 1500dB of dynamic range at 24-bit, so you know that tracking hot, once past the preamps/DIs/converters, is going to be a non-issue when relying solely on DP. But we aren't relying solely on DP. We're all using 3rd party plug-ins.

I have found that IK Multimedia's TRacks and Waves RenMax bundles end up clipping from hot tracks run through them. I've spent 2-3 years working with this issue and finding solutions for this. I'm talking digital clipping. My solution is to add a TRIM plug-in 1st int he chain for a hot track and drop the gain down into the -18dBfs range before hitting 3rd party plug-ins. Clears it up and prevents distortion across your mixes. When you add up multiple tracks with this issue it really effects your mixes in a negative way.

And also, analog gear is made to perform at specific levels. It's cool to hit transformers hard for some nice warming on the way in, but if those levels are hitting your converters too hard, you need to pad them down with something like an ATTY after the analog chain and before the converter.

With digital, we're no longer needing to escape a noise floor, however, we need to keep bad distortion and digital clipping out of the signal chain before DP, as well as prevent this same issue from creeping up in DP with 3rd party plugs.

Also, if you have a 16 track + mix where all of the levels are at -18dBfs when each track fader is at "0" in DP's mixer, you are going to see that added together you may have a mix on the Master Fader that's hovering around +3dB (-3dBfs). Most mastering engineers want mixes with headroom between -3dBfs and -6dBfs in order to have space to work their magic.

Experiment and have fun and try it both ways. Also, if you want volume, turn up your studio monitors!

Jeff

This is a very good thread because (IMO), it asks some important questions about general analog/digital recording technique.

The first round of questions had to do with recording level. That question pertained specifically to the measurement within DP. The recording level of DP is a separate measurement from the actual amount of analog signal that is going into the audio interface.

The amount of signal going into the audio interface as it relates to record level in the software has to do with the maximum gain range of the inputs of the audio interface.

Last time I checked, the suggested AES spec for maximum input level of an A/D converter is 20dbu. That means that if the analog signal is 16db above the +4dbu reference, it will hit digital zero on the input of the interface (and that will also be the audio record level).

MOTU interfaces have changed analog I/O gain ranges over time. The original 2408 interface had unbalanced RCA input jacks that were referenced to -12db. I don't remember the to talk gain range of that model, but it was nowhere near 20dbu. With the 2408mk3, the total input gain range is 18dbu. On the 8Pre, you can actually send in 26dbu full scale signal!

We do not put analog calibration of I/O levels on the interface because that extra stage in the signal chain will cut down on available dynamic range.

So yes, if you have mismatched audio interfaces, you could see level differences between the various models. In a perfect world, this would never happen. In the real world, it happens. Gear specs change, and that is certainly not exclusive to MOTU products.

So when we are talking about levels, it's important to differentiate between analog levels going in and out, and digital levels within the DAW.

In my first post, I mentioned that I generally take my recording peaks up to within 3db or so of peak input. I don't use compressors or limiters on input. I would much rather capture the full dynamic range of the signal, and compress it later if I have to.
Therefore the average signal when I record is way below the -3db peak I see as top end for transients. The average signal strength has to do with the content of the signal. I don't even concern myself with average signal on input. I trust my input meters. If one sample clips, the clip light in DP goes on. As long as I don't see a clip light, I know I'm capturing the full signal with no problems. Therefore it is entirely possible that my average recording level is -20 or so. Could be lower if I'm recording a very dynamic signal.

I'm an old school analog guy. I was brought up calibrating gear with test tones. I do trust the technology inside the DAW, but I still do it the old fashioned way outside the DAW. I calibrate my signals all the way through to get best S/N ratios.

Proper level matching from component to component is one of the most important aspects of engineering. Without proper matching at each stage, all the rest of the audio specs of the signal chain go out the window.

A few other bits and pieces.
I've done several of my own mastering projects recently. The way I do it is to make a first generation 24 bit mix. I watch the mastering limiter very carefully. My peaks are -1db and I don't let the mastering limiter do more than 1-2db of limiting. In my mixes, the RMS at that point is typically around -16 to -20db (or more).
I then take the 24 bit mix and slam it for the final output. I may use a combination of multiband compressors, optical limiters, EQ, and the final master limiter. For modern hard rock I try to get an RMS of around -12 to -14db. Depending on the music, I may let it breath a bit more with -16db RMS.

So although I don't have the qualifications or experience of Paul Frindel or Bob Katz, I think we're pretty much on the same page in terms of levels in a DAW.

Dave

My ears tell me I get better signal when I record hot at 24 bits

Was your response to the initial question. The point of most of this discussion was that printing "hot" causes more problems than it solves. There is simply NO advantage to recording hot, you gain nothing and immediately rely on software to adjust your signal to a useable level. The less you manipulate the signal the more true to the input it remains.

I would love to read the AES spec on converter levels, do you have the paper number so I can grab it? Digi uses -18dbFS=0VU as their stock calibration on their trimmable converters, giving +22 as the max input level (18 over +4). I'm surprised that they wouldn't follow the AES spec. What's the max input level on the HD192?

To some degree the end user can calibrate the a/d side of their MOTU interfaces in software after the a/d converter in CueMix, so long as the converter can handle +24 at the input you can scale it any way you want after the converter. Worst case might be going for -12dBFS which would lose 2 bits of dynamic range, as you say, but would allow the end user to integrate the converter into their personal workflow. Obviously it costs more to put analog trim pots on the converters, and for most of the middle market it is not worth your time or effort. It would have been worthwhile on the 1296 and HD192, though.

I intrigued by your point that one of the MOTU interfaces can accept input of +26, I'm assuming that's with the 20dB pad engaged? I can't imagine that you're referencing 0dBFS to +26! I also can't imagine that the box is capable of putting out +26 on the d/a side, so there's no chance of input=output, right?

The only thing MOTU stands to gain by suggesting a standard operating level (+4dBm=0VU=-20dBFS, for example) is better sounding tracks and better end product from DP users.

I know the question was directed to you, personally, but here on "MOTUNATION" (ha!) you're the voice of MOTU... so people take your word as "official."

As someone who came from the analog recording world, too, it amazes me that you have no interest in average level... I would find it impossible to use any of the gear on either side of my converters if I wasn't pretty much ONLY concerned with average level.

tom

Seems like a lot of these answers might have something to do with musical styles and types of input (live horns vs. synth electro-beat or techno-pop). The one is much less predictable in peaks and averages than the other, for example. I don't mean to horn in on a discussion that's basically out of my league or range of interests, but as a musician who evolved into engineering, that seems to be what I notice when I'm setting levels: the predictability of the overages and averages determines how high you can set the levels. Some types of music seem to start out more "compressed" than others.

Shooshie

Shooshie wrote: the predictability of the overages and averages determines how high you can set the levels. Some types of music seem to start out more "compressed" than others.

Shooshie

Absolutely. Your #1 rule in regards to a DAW's meters while tracking should just be:

Don't clip them, ever.

If you are worried about inter-sample peaks clipping your file then get a meter that displays that kind of information, and don't clip that.


Having a rule like "I always go for -3dBfs peaks" or "-20dB rms is the only way imo" is silly out of context. When it comes to gain staging, there is a "sweet spot" for everything. If you have a highly compressed signal going into your ADC, like an electric guitar, playing distorted rhythm through a noisy amp, it'd be silly to track at -20db, if you in turn need to use digital tools to make it louder come mix time. Just crank the amp and use your headroom, that's what it's there for. Likewise, no sense in tracking background vocals near 0dbfs...they will need to get turned down...just do what makes sense and what sounds good and you'll be fine. There is plenty of speculation and discussion out there about this or that, but a lot of people are making good sounding mixes in the meantime. Too bad the songwriting is usually vomitous.

Well, that's not helpful at all. This is not speculation. The fact is that the analog gear that is in front of your converter, and after it when you monitor (or if you mix analog) was designed to operate at a target level. That's what 0VU is. The fact that you don't care does not make it worthless or useless.

If anyone cares, the link to the PSW forum that I posted earlier is full of useful info, as is the Sonnox limiter manual which I also linked earlier. Useful info for engineers and people who want to improve their engineering chops.

If you want to talk songwriting, I agree... far too many people obsess about gear as a distraction from the hard work of writing better songs.

Personally though, it's my job and profession to make the best sounding records that I can... so a technical discussion about standards is pretty much exactly what's important to me.

I'm lucky enough to do almost all the tracking on the records I do, but I absolutely try to encourage good recording habits when dealing with clients that record at home.

Here's a worthwhile "technical" thing to see, if anyone cares...

http://www.turnmeup.org/

Please watch the movie if you can spare two minutes.

tom

Thanks for the link and great info Tom !! Keeps the perspective in check .

T

tomeaton wrote:Here's a worthwhile "technical" thing to see, if anyone cares...

http://www.turnmeup.org/

Please watch the movie if you can spare two minutes.

tom

I'm not sure whether you were talking to Old Timey or to me, but the stuff in the video -- which is absolutely right, of course -- is kind of basic. That's what we've been raging against for the past decade. I guess what I'm saying is that you're preaching to the choir. From the days of analog to the present I've had to work my way through the entire signal chain to find my standards for a given setup. But it started with knowing the ranges and limits of the audio sources. Some sources are more predictable than others. You find the level that seems to be the maximum that any instrument or group reaches, back off a bit, and that becomes the "hottest" you can record without compression. There may be other ways of achieving the same thing. Knowing the "centerline" and max/min for every part of the signal chain is helpful, but you still have to check levels before you begin recording. I'm not sure what where we're in disagreement; in fact, I don't see where you've said anything that is in conflict with my thinking at all. But the difference between Magic Dave's levels and your levels may be a result of the predictability of the peaks of what you record, which may be entirely different kinds of music. That's all I'm saying.

Shooshie

No, Shoosh... it was the "speculation and discussion" thing that set me off.

Anyone who has ears can hear the results of mixing ITB at lower levels.

In a 24 bit world there is no reason ever to print "hot."

tom

PS

I show that "TurnItUp" video to every client. It's the best presentation of the issue I've ever seen... and it takes no time to "get it."

tomeaton wrote:...Anyone who has ears can hear the results of mixing ITB at lower levels...
...In a 24 bit world there is no reason ever to print "hot."...
...I show that "TurnItUp" video to every client. It's the best presentation of the issue I've ever seen... and it takes no time to "get it."...

Tom, could you clarify please. The TurItUp video has to do with compression, but when you say "lower levels" and "print hot" are you talking strictly about levels or the average level shift resulting from compression?...Or both?

Phil

I echo my past post. We are talking about about multiple things simultaneously. At first I thought the question was for Magic D - our reliable and highly respected MOTU friend who I'm sure takes his personal time to hang with us and help us out - about recording levels into DP. Now... we are talking about over compressing and dynamic squishing.

Originally it was my understanding that some hot engineer said record at lower levels because you will avoid intersample distortion and your mixes will sound better (paraphrase). Now we are talking about the tendency to squish the snot out of recordings. In my opinion - two very different topics. You can always take a mix recorded at lower levels (in 24 bit) and ruin it with bad mastering.

Having said all of that, I do like a slamming album. There are mastering engineers that can make things hot and dynamic. I think of Stephen Stephen Marcussen and most of the guys at Sterling Sound. I have to say I often like it. Then there's guys like Bob Ludwig who keep things dynamic but it still sounds loud and slamming. However, I know other of other (more hacks) mastering engineers that make it all sound like crap.

So... yup

CC

conleycd wrote:I echo my past post. We are talking about about multiple things simultaneously.

Yeah, that's what I was trying to ascertain with my question to tomeaton. I wasn't sure if he was talking about the loudness wars as a tangent or if he was addressing the original poster's question. In any case, the "what's the best level to record digital signals" question remains controversial, not to mention confusing at times. I think ultimately, "use your ears" is good advice.

Phil

Loudness wars was a tangent, but related in the "loud is best" kind of thinking that is so prevalent.

I felt rather taken to task for having a technical discussion, so I was just pointing out another technical discussion worth having, for those that are interested.

Printing hot is absolutely a different animal than hyper-compression, but the two things stem from the same mistaken belief.

We have to understand that dynamics are essential to the enjoyment of music, and that maintaining "natural" dynamics (whatever that means in your particular scenario... pinned electric guitars obviously have less dynamic range than a piano) brings the listener closer to the source, which is a relatively sure way to help them enjoy the end product.

I am so tired of putting on a cd and hearing tune after tune scream at me... mashed crash cymbals tearing at my ears on every pop song chorus...

If you want your ITB mixes to have an "easy" feeling dynamic, it really helps to bring your levels down. If you can do that without applying processing to your basic tracks so much the better. So printing at reasonable levels helps the performance of your analog gear, doesn't tax your converters, allows you to apply less dynamic manipulation in the mix stage and allows your plugs and mix buss to breathe.

Part of the problem of "printing hot" is people comparing their tracks to finished, mashed cds. So, for example, I hear people compress the crap out of their vocals ON THE WAY IN to get signal level... not to get the sound of a particular compressor... and it's just misguided.

People recording at home have not always been privy to the process of production that made their favorite records sound the way they do. The goal should be to make the signal sound as much like you believe it should at each step, but with the understanding that tracking, mixing, and mastering are separate processes with individual goals.

Blah, blah, blah. (or, for the Seinfeld fans among us, Yadda, Yadda, Yadda)

tom

tomeaton wrote:
Part of the problem of "printing hot" is people comparing their tracks to finished, mashed cds. So, for example, I hear people compress the crap out of their vocals ON THE WAY IN to get signal level... not to get the sound of a particular compressor... and it's just misguided.

People recording at home have not always been privy to the process of production that made their favorite records sound the way they do. The goal should be to make the signal sound as much like you believe it should at each step, but with the understanding that tracking, mixing, and mastering are separate processes with individual goals.

Blah, blah, blah. (or, for the Seinfeld fans among us, Yadda, Yadda, Yadda)

tom

I agree with you Tom largely. But it is very annoying when I have accidentally recorded an instrument or vocal too quiet and then have to go back and turn it up in the mix because the rest of the instruments are decent and the faders are all over the place. What... do I keep every fader on the floor except for the quiet instrument. I find in that situation I start - accidentally - mixing with compressors to compensate for a week signal.

I also agree newbees tend to over compress when they find out about compression. I found Russ Long's video (Guide to Nashville Recording) helpful for understanding tracking compression. Essentially, he tracks with fast attack and fast release no more than 1-3db on hot spots going into tape/DAW.

CC