Mixing Question - Levels

[ernstinen]

Hello Ernstein....not sure if you took offense to what I said....?? I was just answering the original question, which asked if anyone had a similar experience mixing... with no reference to any post except the original question......Hope you have a great weekend.... Peace out brother....!!!

No offense taken! I was just joking about the tone that has evolved in this thread! Peace shall prevail on this earth.

Dwayne, just for my own knowledge: You said analog is -18 vs. digital. --- Is that on a "pro" +4 system, or a "consumer" -10 system? This is a real question --- no joke here! Because my analog board is +4. --- So, when I used to mix analog to DAT out of a +4 board and averaged a conservative -6db on the DAT, um --- HELP! [My brain hurts a lot (David Bowie)]

Thanks is advance,

Ern

[mazz]

Check this out:

http://www.digido.com/level-practices-p ... ystem.html

My understanding of this from my limited perspective compared to others, is that with digital, you pretty much have to choose your reference level and have that correspond to what 0VU would be on an analog system. In this article, he promotes several reference levels. It's important to also have an actual volume reference (in db SPL). Basically the recordist chooses their reference level, for instance -18 dbfs (a digital reference) and makes that their 0dvVU reference (analog reference) and sets the speakers at this point so they output pinknoise at around 80db SPL (actual volume of the sound coming out of the speakers). Basically now you have a baseline electrically, digitally and sonically that you can return to and trust. With -18 you now have 18db of headroom.

This should be able to be done on any system whether it's -10 consumer or +4 pro, because those are just electrical specifications. Of course, the impedances and levels all must match, etc., but the bottom line here is establishing a reference level that is consistent and repeatable, no matter what system one is using.

What I've been learning from studying this stuff is that there are several different "flavors" of the unit db, so we have to understand what units we are referring to, because db doesn't mean anything unless it's qualified with a reference. For instance, 0 dbfs (decibel full scale) is a digital reference that means (all bits turned on), whereas 0dbVU is an electrical spec on an analog system. So making digital respond in an analog fashion (at least conceptually) requires one to establish these baselines.

Obviously folks are generating great sounding music without knowing about this stuff, but knowing it can enable one to put out product that has a consistent level. A side benefit is that it encourages making music with dynamics, which most of the time seems, to my ears at least, to be more emotionally satisfying.

Again, just sharing what I know from my limited perspective and what I've been studying. I'm sure other more experienced folks will fill in the gaps (hopefully), but the article I linked is pretty comprehensive and by studying it my understanding of this subject has grown tremendously.

Cheers!

Mazz

[gtrmann]

............ By 1986 I was an assistant for Full Sail in Orlando.........

Hey Dwayne,
I live in Tampa, I toured that facility around 1987 or 1988..... when they had the synclavier units there....... Where you working there then...??????

[Len911]

Mainly because of this thread, and also because I needed to establish a point of reference, I used Wavelab, which also includes the K-sytem meters by the way, and analyzed various cd's I have. I didn't use an spl meter to mark my monitor gain pots, I used the cd's instead. From all that I basically decided I was going to use 0 db for the peaks, and about -14db for the rms average levels. Then I remastered the songs I have on my reverbnation page that were so inconsistently mastered. Now I do a global analysis on the songs after I master them to see where I'm at. From the cd's I have I had a couple, zz top great.hits and tom petty greatest hits, that were in the -12, -11 db, and they sound worse than those -13 thru -19. Most cd's were similar, Bob Dylan's Nashville Skyline was the most conservative, -6db for peak, and -19db rms average. If I only turned up the gain a couple clicks, I was volume wise up there with Tom Petty the most compressed, however, the difference in "quality", and coincidentally as they both are in a similar vein vocally, Bob's was so much better. The only classical cd I had was opera, and it was similar in compression to the others. The best mastered imo was an Aaron Neville cd, that was a little different, in that it was -2 or -3db on the peaks. George Massenburg mastered that one. Although the exercise wasn't scientific, I've heard enough to determine my own settings. Now if I can just leave my hands off the damn gain control knob, I should be a little more consistent anyway,lol!!

[cardell]

Great information...thanks Len!

Stuart

[mojobone]

I took a class in digital recording at UCLA, which was waaay over my head, but one thing I believe I remember: When recording tracks in digital, it's similar to signal-to-noise ratio in analog. The idea is to fill up all the "bits" (?) to get the beef on the tracks, just like getting into the red in analog on each track.

If I'm wrong about this, clue me in! But if that is correct, that means that the tracks are recorded to digital capacity, so lowering levels should not degrade the sound, just lower the volume of the sound.

Is this right? If not, I wasted my $600 because I didn't learn anything !

Thanks in advance,

Ern

You might want to get your money back

There is no such thing in digital..........because it's digital. Bits have nothing to do with saturation. The "beef" in digital has more to do with sample and hold fidelity, power and low jitter, not at what level your input is to your DAW.

What happens in volume after you record has to do with the program algorithm, not the record process.

Bits in digital refers to how often a snap shot of the wave is taken. 24 bits takes twice as many snap shots of the wave there for it is more a true picture of the wave. But how high you drive an input for recording will not change the fidelity whether it be 16 or 24 bit rates.

Ern is correct regarding 16 bit systems; it's less true at 24 bits, but likely true enough, back when he took the course, heh. Dwayne appears to be referring to sample rate, here, rather than bit depth. The term "bit rate" comes from data compression, and is meaningless in PCM digital sampling, but used to be interchangeable with sample rate, before we had such a thing as data compression, hence the confusion. Mazz is correct about the term "decibel", it has to refer to something else, because it's a relative term. Dwayne is correct about analog signals being "hotter" (also a relative term, cf. the ratio of Angelina Jolie:Jennifer Aniston) than digital signals. Bob Katz calls this "the analog advantage" and it's all fully and completely explained in his book Mastering Audio, which I highly, highly recommend.


Regarding bit depth/sample rate, let me use a quick analogy which should be helpful in understanding PCM digital recording: think of how a motion picture camera works; the sample rate is like the frame rate of the camera, and bit depth (or resolution) is like the number of pixels per frame. HTH





Terminology can become very confusing when we're not all talking about the same thing, and in audio engineering, accuracy counts. (it really is an art and a science)

[gtrmann]

Hi everybody,
I am sure I am going to catch a lot of flack for this, but.... the way I see it is....

The original question was about how if you mixed many tracks together that were all peaking at around 0dBFS, the final mix would be over 0dBFS and distorted. The person had to bring the levels of his tracks down in order to keep the final two mix under 0dBFS......

I am not sure what that has do do with the sampling rate, the amount of bits used per sample, or the 0dBFS standard of -18DBu.....

No wonder why we are not getting more forwards...... we are all off target...........

I am here to try and make friends....so if I go to the rally this year, I will have some people to hang out with.....

Ok ....you can shoot me now....!!!!!!

[mojobone]

I think we dealt with the original question pretty well; the upshot is, it harms nothing to turn your signals down when mixing in a 24-bit digital system. At 16 bits, less detail is captured in low-level signals, so best practices involved keeping the levels hot. It's no longer necessary, and even then, was only necessary during capture, not mixing.

[Cruciform]

Thanks Mojo. Simple, to the point, useful.

[gtrmann]

Thanks Mojo. Simple, to the point, useful.

YES, great answer Mr. Mojobone.....simple, to the point, and accurate.... I applaud you....!!!!!!!