where02190
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Posts posted by where02190
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Originally posted by JohnnyXand every mic has a proximity effect.
Inbcorrect, omnis do not.
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Originally posted by gsHarmonyI agree that 10" is probably not too far.Originally posted by gsHarmonyI almost always tracks vocals at 8-12".
OK so make up my mind then.....as where I come from 10" is exactly in the middle of 8" and 12", making 10" your average distance.
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10" from an LDC in the studio for vocals is definitely NOT too far away, unless your vocal room totally sucks, inwhich case you should use another room or modify the one you're in.
Don't vary the distance unless you are singing much louder in one section than the other. Use the volume automation in your DAW to control the levels between verse and chorus, and maybe a bit of light compression if you have a good comp and know how to use it.
Volume automation is, while more time consuming, infinitely better than compression IMHO, and combining the two makes creating a killer vocal track, assuming the track is decent to begin with, pretty simple.
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10" from an LDC in the studio for vocals is definitely NOT too far away, unless your vocal room totally sucks, inwhich case you should use another room or modify the one you're in.
Don't vary the distance unless you are singing much louder in one section than the other. Use the volume automation in your DAW to control the levels between verse and chorus, and maybe a bit of light compression if you have a good comp and know how to use it.
Volume automation is, while more time consuming, infinitely better than compression IMHO, and combining the two makes creating a killer vocal track, assuming the track is decent to begin with, pretty simple.
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IMHO those things are fine for the appartment dweller who can only play on headphones, but for a real electric guitar sound nothing sounds like the real thing.
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If you're going to stuff the guitar to lessen the feedback, just play a solidbody.
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Turning down comes to mind......
Ease up on the compression when trying to achieve these tones, as when you don't play, the compressor eases up and you get all this gain, which causes feedback.
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But I guess you're talking how it sounds in the monitors.
I'm not sure if you're being a wise ass here or not, so I'll give you the benefit of the doubt.
Many top engineers have commented here and in various articles regarding mixing and techniques about the difference between keeping the master fader at 0 or not, and I've not read one that recommended pulling the master fader down over keeping proper gain staging.
There is a very obvious audible difference. -
bUt it all comes back to that converter....0dBFS is a real ceiling. Why not just record correctly and not pull your master down so far?Hmm.... because it doesn't make any difference which way you do it?
Then why, no matter what system, no matter what converters, does it sound better if you leave that master fader at 0 and adjust the track faders? -
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Anyway, I read somewhere that you should track with the level just high enough that with every fader at unity, you'd have your desired mix
All software and hardware are different, but I find -10 to be more the norm. In order to be able to put all your faders at 0 and have a nominal 0dbu reference level mix, your tracked levels would IMHO be too low, and your s/n ratio well below what it could be. -
This has been discussed a multitude of times here. There is a clear and obvious difference in sound when proper gain staging, whether tracking or mixing, is observed. Slamming into the masters will not sound as good as keeping proper fader levels so, with the master fader at 0(the only way in most daws to meter accuratesly, since most all DAW's master fader meters are post fader), the nominal level is referenced to the 0dbu reference level.
There are posts in this thread that state this very thing, the sonic difference is immediately obvious.
However if you wish to remain ignorant of this and continue to slam the {censored} out of your {censored}, by all means go for it. -
Todd definitely as an arranger/producer, that old Utopia stuff is still incredible....
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Originally posted by woodensince i am going directrly into my 1616m i dont have good metering options. Can i insert this voxengo's free virtual VU meter in the track to measure it? is there any other better way?
Provided you are metering pre fader/eq and any other plugins, yes.where...I would like to get out of your hair with this topic and on to another...thanks for your patience.My Yamaha AW4416 has a rinky dink little meter that reads 0to-60db in normal mode, and from 0to-26db in "fine" mode.You guys are saying to stay close to -18dbfs. On this unit, when I am around -18 db on my meter, I am still one tick below orange. It just seems soooo low. But I have to say I think it sounds better down there. Is -18db the same as -18dbfs
You'd have to check with Yamaha as to whether the meters are indicating dbu or dbfs, but I suspect they are probably reading dbfs, and yamaha simply titles them db.No - a digital meter is obviously post-a/d converter. It's just your analog stuff that needs to be set up around nominal 0dBVU. Up to, and including, your A/D converter.
Nope, you've missed the point entirely still. Set the converters dbfs input level to it's 0dbu reference.Some of this advice is more academic than useful. As long as you don't clip your a/d converters, or seriously under-record, you can't go too wrong.
Again this is IMHO not good advise. If all your tracks are peaking at odbfs(the maximum allowable digital signal) then your track faders will end up at the bottom of the throw when mixing to prevent summing overload/mix buss odbfs overages, resulting in very little room to play in terms of fader throw and level control.If you understand that a/d converters are an analog device - the more you push them, the more distortion you get (within it's analog electronic circuit - not talking about digital clipping, running our of numbers.
AD are not just analog. AD stands for analog to digital, the input is analog, the output(into your daw) is digital. That digital end has a finite limit, 0dbfs. If you want to keep insisting that you can go beyond that and still have audio that is musical, please post an example. Otherwise stop giving this bogus advise. It's well known fact that over0dbfs is digital cliping, unlike analog, it is anything but musical, -
In the studio, I wish I could sound more like Tim McGraw and Byron Gallimore. I think their stuff sounds amazing. Produced, but not overly produced.
Live, I'd kill to mix like Robert Scovill. I go see bands I don't even like (TP) just to hear him mix.
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Originally posted by gsHarmonyBy average level, do you mean RMS, or do you just mean that on average the signal should be around -18 dbfs?
Your average/nominal signal should target your converters 0dbu reference, which, if you don't know it, -18dbfs is a good place. This reference varies depending on the interface, from between -20 t0 -12dbfs. Check with the manufacturer of your ADDA for specifics.
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Within the digital domain, you can safely exceed 0dB Full Scale, with zero damage to your audio wave.
If you like the sound of nails across the chalkboard I guess.
I suggest you post some clips of your "music" that was recorded exceeding 0dbfs. Square waves are so pretty afterall, so nice and, well square.....
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There is no reason to be afraid of using the master fader as required.
There is if you're turning down the master fader to compensate for overshooting 0dbfs because track faders are too hot, slamming the summing. It has been proven over and over that, while there is considerable more headroom into the summing than in an analog console, it is not infinite, and there is clear sonic and dynamic difference when paying close attention to what's going into (PFL) the mix buss.
There is no reason to worry about digital channels exeeding 0dB, not until you reach the D/A converter.0db no, 0dbfs most definitely. excedding 0dbfs makes for a noise that is absolutely not musical in any sense.
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I mix through a digi board anyway, so I can see what's going on at every stage.
And here is the big misunderstanding. What you don't see on most digital interfaces is what is going INTO the mix buss, the metering on the mix bus is POST FADER.
Leave that master fader at 0, and see what your meters say. That's what's going into the mix buss.
Pulling down the master fader 10db to compensate for poor fader positions which result because of recording too hot does NOT sound the same as recording proper nominal 0dbu inputs, leaving the master fader at 0, and adjusting the individuail track faders to prevent overshooting 0dbfs on the mix buss.
While on the mix buss subject, again that same 0dbu reference should be observed for nominal levels going out of your DAW as well. Running your mix buss at or near 0dbfs leaves little to no room for the mastering engineer. (I'm sure Massive will have a comment on that as well.)
I would love to see DAW and console manufacturers that have abandoned PFL on the mix buss to return that feature. SO many times I see engineers with all the faders at 0 and the master pulled way back, and the mix is just painful to listen to. Reverse that, keep the same output level on the mix buss, but lower track faders with the master fader at 0, and the mix opens up and sounds IMHO far better sonically.
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Originally posted by TBushYeah- what I meant was that thepeakswould be "just under 0 (dBFS)"- digital has great headroom and no need to blow it there...
NO, that's not correct. Again, slamming into the converters as close to 0dbfs is NOT the way to record into digital. Target your nominal, that means average input level at your converters 0dbu reference. that will leave more than sufficient headroom, typically around -6dbfs. Think about it, 24+ tracks, all peaking at -1dbfs, slamming into the summing buss....+18dbu x 24+ tracks....can you say distortion?
There is absolutely no need to be peaking at or even near 0dbfs. all you're doing is eating up your headroom, meaning your faders actual or virtual) will be at the bottom of their throw to prevent summing smashing, leaving you little range of movement.
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All the bits are used a 0dbfs. You lose one bit for every 6dbfs, so -18dbfs is using 21 bits.
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And the standards for Digi interfaces vary also. For instance, the 001 was -18dbfs=0dbu, whereas the 002 is -14dbfs.
Also consider this: 0dbu+0dbu=+6dbu.
While digital summing is much more forgiving, it too has a finite limit. Imagine 30+ tracks, recorded at .5dbfs peaks, with all the faders at 0. Your master fader ends up around -50 in order to keep from overshooting 0dbfs. You're pounding the crap out of the summing amps.
This is probably the #1 mistake new engineers make.
Forget everything you ever heard or knew about analog tape, none of those rules apply. By simply keeping your nominal levels within the parameters discussed (ie targeting the converters 0dbu reference to dbfs), all other things equal, your end result will sound far, far better in audio quality. It's simple, and it is very, very effective.
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One can (as I have suggested many times) DIY room tuning for very very short money. There is a wealth of info available free online about the subject.
Getting new monitors is not the answer to a bad room. The DIYer can tune a room for less than the price of a modest set of monitors, and achieve superior results without replacing what may be perfectly good monitoring, only to find that the room acoustics are the issue. Learning about the acoustics of recording is IMHO essential basics that the beginner can greatly benefit from before exploring better quality equipment. As we all know, the best gear in the world won't make a bad room sound good.
However this has nothing to do with the posters question about levels.
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You may be misleading people in that a pristine recording is required to be a GOOD recording.
Actually what I said was "...proper levels are the most important factor in digital recording."
As engineers our role, whether it is for our own music or someone elses, it sot capture it in the best manner possible given the equiment and limitations therein. Recording at the properly levels, in this case not slamming the input levels nominally over the 0dbu reference, is IMHO a crutial part of the process.
What people want to listen to, and how it's recorded are two entirely different subjects. There is some amazing music that sounds like ass, but spiritually moves me. The other side of the coin, there is a whole heap of music that sounds great but wouldn't ever make its way to my ears for more time than it takes to turn it off.
Music will move you no matter how it is recorded, however IMHO ignoring the basic fundamentals of recording because of that is not ignorance, it's stupidity, and great music will suffer.
amplifying an accordion
in Craig Vecchione's Live Sound & Production
Posted
Accordians can be a bitch. A clip on close to the open end of the bellows (where the bulk of the sound comes out) works best, and a pair won't be a bad thing. I find the countryman isomax works great, they are very small and can easily be placed close to the source without impeding the player, and they sound great with minimal eq.