Best Recording Levels

[sowby]

I think everyone here needs to read Bob Katz's "Mastering Audio" book. All this is explained quite clearly and thoroughly. I'm not gonna try to show off my super skills and pretend to know all, but I will give an excellent piece of advice: USE YOUR F***IN EARS! If it sounds good, it is good. Period! Get back to making music.

[digilogin]

Originally Posted by O.G. Killa

LOL... I feel like Luke Wilson's character in "Idiocracy" here. Do you want me to tell you that I can talk to DAWs, and that they like lower levels better???

hello,

with all due respect, i don't want you to tell me anything.

you seem hyper-concerned with measurements, and you appear more interested in noise than in the accuracy of the signal of interest.

dither does not improve accuracy, it merely masks quantization noise. accuracy is improved at higher bit depths.

virtually all complex sounds contain low amplitude information. thus they benefit from greater bit depth.

again, you can pull back the channel faders, after the audio is converted. you can attenuate signal at or before the input of your plugins. you can calibrate your interfaces' input and output trims. there are ways to avoid the gainstaging problems you are concerned with, while still using more bit depth at the converter than you appear comfortable with.

digilom

[DrFord]

I like Cheese. And my beeetz are fat. This is becoming a pissing contest where I believe both of your are correct. I believe that the voltage statement is true, and I believe that the 24 bit sampling is true. So how can you both be right...!

Well It's probably somewhere in the middle not either side's extremity. As much as we'd like to keep everything at the perfect voltage, when you are in the studio and the drummer is drumming and the strummers are strumming and the hummers are humming and the... well you get the point, you do what you have to regardless of voltage and perfect sampling and you don't think about intersample peaks or bit depth, you hit record and pray to God you recording captured the magic moment.

That is the very reason I am a better producer than engineer, though honestly... I really do respect both of your minds and I feel I have learned some intersting trivia reading / being a part of this thread.

I stand by my original post that getting a hotter signal is better in the long run, however I have been trying to not go soooo hot as I am attempting to keep my voltage grounded (when I have the time to worry about it.)

D

[O.G. Killa]

Originally Posted by digilogin

dither does not improve accuracy, it merely masks quantization noise. accuracy is improved at higher bit depths.

Ugh! I give up. Spread all the misinformation you want. You really have a problem admitting you are wrong.

While yes I agree that accuracy CAN be improved with higher bit depth... DITHER improves accuracy of whatever bit depth you are currently using or if you are converting from a higher bit depth to a lower one. Just do a google search for "Does dither improve sampling accuracy?" You'll find dozens of articles proving your statement wrong. Maybe you should read up on dither a little more?

[digilogin]

Originally Posted by O.G. Killa

Ugh! I give up. Spread all the misinformation you want. You really have a problem admitting you are wrong.

While yes I agree that accuracy CAN be improved with higher bit depth... DITHER improves accuracy of whatever bit depth you are currently using or if you are converting from a higher bit depth to a lower one. Just do a google search for "Does dither improve sampling accuracy?" You'll find dozens of articles proving your statement wrong. Maybe you should read up on dither a little more?

hello,

i don't have a problem admitting anything. its just that i am not wrong. and, to be honest with you, i do not care if you want to record at -90. have fun. if anyone is spreading misinformation, its you.

aside from some theoretical, and experimental garbage, the relevant and reliable sources appear to agree that accuracy [in the sense of the number of discrete, representable divisions of amplitude within a specified overall range of amplitude in the low amplitude area] does not improve with dither. nor linearity, if that is what you are considering. dither's effectiveness lies in reducing quantization noise.

"Note that dither can only increase the resolution of a sampler, it cannot improve the linearity, and thus accuracy does not necessarily improve." - Wikepedia: "Analog to Digital Converter"

if it were possible to truly meaningfully increase accuracy with dither [not just give the illusion of it], using some enormous amount of some exotic noise shaping, it would be irrelevant, because that is not a viable mechanism. and, assuming, arguendo that such a thing were possible, it would never be as effective a way of improving accuracy as simply recording at a greater bit depth. overuse of dither is problematic also. dither is not intended as a substitute for proper recording levels.

i understand your premise. you are essentially arguing in favor of recording at a low enough level so that the adc does not record any analog noise [i.e. in your model the noise is out of range of the adc]. apparently you feel that one should only record the top 90dB or so, thereby "leaving the noise behind". however, unless you literally record at a low enough level so that the noise is completely out of range of the adc, then you are always going to be presented with a situation where the noise you have "pushed down" is simply brought back up during mixing and processing. now you have both the noise, as you have brought it back up, and a less refined [less accurate] signal of interest.

you also seem to be ignoring the fact that we can discern musical content well down into the analog noise floor. so by "chopping off the noise" you are also chopping off part of the signal of interest. you are proposing a technique that involves "throwing the baby out with the bathwater", so to speak.

these things have been thought through and duly considered by the designers of the product. i think the manual recommends tracking with the software meters typically hitting / peaking at -6 to -12 or so. that works fine, conservatively, and you don't have to go crazy watching for overs or worrying about intersample overs and so forth. if you have a source that has a real wide dynamic range you may have to be even more conservative.

also, if you have good gear, it does not start to crap out the minute it gets a little bit "out of spec". and, in any event, as explained there are ways to address all of the gainstaging issues you were concerned about.

digilom

[MoritzRock]

Originally Posted by O.G. Killa

What noise? Your mic preamp that you are cranking up is going to have more noise in it than your protools system. Most mic preamps have a noise floor of about -90dB, protools is around -118dB to -120dB. Your mic and mic preamp will have AT least 30dB MORE noise in it than your DAW. So, if you are trying to record at louder levels the only thing you are doing is putting MORE NOISE into your tracks because you are turning your mic preamps up to get the signal as hot as possible without clipping.

If you are recording into a DAW, you are NOT USING TAPE. There is no need to record as hot as possible since there is no tape noise/hiss to compete with. You are only going to make your recordings noisier by turning up the preamps.

Calibrate your studio and everything will fall into place. The 192IO is set from the factory to -18dBfs (on the PT meter) = +4dBu = 1.228 Volts = 0 VU (on an Analog VU meter).

This means, the old analog technique of "keep the needle right around 0 on the VU meter" translates to "Keep the signal right around -18dB on the Protools meter (which is just under halfway up the meter)".

Some of the best tracking engineers I've seen, record everything with the faders set to "0" and change the mic preamp gain to place things proper ly within the mix WHILE TRACKING. Most people today don't really do this because pulling the fader down in PTHD doesn't really effect the sound of the track (until you get down around -90dB on the fader). Whereas on an analog console as soon as you start pulling the fader down you are changing the sound (since the fader is a voltage controlled amplifier/variable resistor).

So, to answer the original poster's question... if you are going to record and mix completely in the box you are better off keeping the levels lower for better Signal to noise ratio and to keep intersample peaks from clipping plugins and such.

Hi!
Pretty much all well said,
However yeah there are engineers that try to do recordings and mix the signals in with the faders at zero as some tried to do this also in the analog domain in the past... put the faders at zero and you got the mix type of thing right??
In digital is almost kind of possible, in analog well there is the noise to deal with, however in my opinion anyway, this barely works in term of the right gain structure an recording levels (at least if you don't have a channel fader to dictate your level to tape that is, if you do then yes that is well possible..

Now, as i always known it (am i wronG here??)

0dBU=0.775V. +4dBU= 1.233... Volts Right??? (0VU at +4dBu)

And as Mr Bob katz says in his wonderful book when recording at 24bit, a signal level that reaches -48dBFS is a full scale 16bit recording (144-96=48)
Lots of good records where made on Adats 16bits... so take it from here at -18dBFS you are well up there... headroom is your friend, well.... at least is some peoples friend...

Good thread but, Still amazes me how some people are just argumentative and don't like to learn stuff from people with more experience...

Pass it on, i will be happy to listen...

[JFreak]

Guys,

Remember that one track of 24 bit audio is a different story than mixing 100 tracks of 24 bit audio to produce one final track of 24 bit audio.

If you're only recording that one 24 bit track of audio and are going to master that; sure, just try to record it with as many bits as possible. Bottleneck is either the A/D converter (having approx 120dB dynamics) or something before that. Either way you still have to decide how much dynamics your FINAL track should have, for example going to 16 bit CD you'll have to compress it to 96dB anyway so it does not matter much whether you succeed to record 120dB dynamics. Yes there are better formats but you get the picture; and to think one step further: how is the end product going to be enjoyed? With a large scale PA that can actually put out all those 120 decibels? If not, when for example enjoyed in a common living room "reasonably loud" (say, 85 decibels) it's still not going to deliver everything that is possible IN THEORY.

That said, more dynamic music will sound more dynamic but what's the point in trying to capture +100dB dynamics when virtually nobody ever hears it?

And back to the another point; mixing tracks together. Digital audio mixing is pure math, like it or not, IOW summing zeros and ones together. And summing more tracks will always requre longer word length for the result track! So if you take a hundred tracks of 120dB audio, how many bits will you need if the master buss cannot be allowed to clip? How many bits will you have to "bring down" the final track to make it fit into 24 bits? Does it sound better or worse compared to "bringing down" the tracks before mixing them together?

Do you really need/want to mix together tens of tracks of +100dB dynamics? Really never use compressors? Ever thought about the usable "real world dynamics" that you want your track to feed the mix buss? Isn't it more common to be wanting "lead vocal to float on top of the mix" rather than have crazy dynamics going from whispering that is almost non-audible to shouting that loud the listener turns down the volume?

Really, come on, think about this. Complain about dynamics when you think that you're going to record a track that is too flat. It's like using 60's gear with today's artists. Not going to happen very often. If you need more dynamics, you'll know it before you hit record.

But if you feel you're recording nice sounding track that feels dynamic enough, then everything is all right and you can move on. No need to tweak where the benefits are minimal.

[Searcher77]

Well I just hooked up my Mbox2 le 8 to my Imac OSX10.5.6 and it working but I need an external hard drive to complete the basic system. I am an old analog guy and I started jamming the meters but got them down in the green. The midi instruments that are set up are way hot into the -5 -6 zone and Its confusing since the volume slider for its track doesn't effect its level. How are midi track levels set I'm wondering. I am way green midi guy.

[O.G. Killa]

Originally Posted by Searcher77

Well I just hooked up my Mbox2 le 8 to my Imac OSX10.5.6 and it working but I need an external hard drive to complete the basic system. I am an old analog guy and I started jamming the meters but got them down in the green. The midi instruments that are set up are way hot into the -5 -6 zone and Its confusing since the volume slider for its track doesn't effect its level. How are midi track levels set I'm wondering. I am way green midi guy.

Midi track meters I believe show midi velocity. I could be wrong about that. So, a velocity of 127 would peak all the way at the top of the meter.

If you are using an Instrument track, that is a different story since the meter on that track is showing the audio output of the plugin.

Some VI and sample manufacturers "master" their audio for their plugins, which makes them very loud as you've been finding out.

In my experiences, this usually hurts the sound quality of the final mix unfortunately... There is really nothing else you can do except to turn them down. But even then they are usually slammed with compression and tweaked with EQ, which makes it harder to get the sounds to fit within the context of all the instruments you've recorded.

[RickD16v]

Originally Posted by DrFord

OG, I hate to say it but I absolutely disagree. Digital noise is just as prevelant as it was on analog tape. It's different noise but it exists all the same. I have beautiful mic pre's (avalons and focusrite ISAs) and they don't make tons of noise, but the reason is because I give them hot signals to begin with. Yes, if you whisper a track off a synth into a mic pre and then boost it to hell you are going to get mic pre noise.

The other thing I am talking about is RF noise, noise from unbalanced cables, bad power noise, cell phone noise, cross talk, there are so many different ways noise can become part of your mix, and like I said, as soon as you compress and boost the output gain, you are going to boost the noise that always exists in your recordings.

Now headroom is another thing entirely, and so is dynamic range, another reason to record hot so that you can fully benefit from the dynamic range of sounds - example is a synth sound with a cool reverb or delay built into the sound that slowly fades away into nothing.

Another reason is that I paid good money for my preamps and the sound of a good tube really helps any mix as far as I am concerned. Sterile digital recordings need all the harmonic help they can get.

I totally agree with you Dr Ford. The idea that just because the second means of recording is better than the first doesn't mean you should not optimize things and just sacrify -XXdB of dynamics.
I think it's quite absurd: you have here people saying "oh, we have here so much more dynamics than CD.....let's continue as if we had only 16 bits to play with & sacrify the rest..."

Really quite amazing how many people don't get it: it's not just ONE SNR we're playing with here, it's the first PLUS the second...the digital support may be better than the analogue in terms of noise it still DOES have its own noise etc and that is being ADDED to the analogue source. Whether the analogue source had less dynamic range or not is completely irrelevant. When you record at -18dBFS you better have a bloody good justification for it and so far i haven't heard of anything measured, just impressions & possibly mere myths.