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Originally posted by wooden

since 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?



Originally posted by wooden

since 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?



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.

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.

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.

So the purpose of lowering your gain structure and not hitting the a/d so hard is simply to avoid unwanted analog distortion.

Consider that many people find that DAW recordings are waay too clean, and apply saturation plugins and all sorts of tricks to dirty up the sound and make it more analog ...

So forget your meters and trust your ears. Maybe you will like the analog distortion of your converters (not talking about clipping at all).

Also, your preamps and compressor (if used) have similar analog distortion characteristics. Don't blame your converters for distortion that might be occuring before it gets to them.

Ultimately - there is a wide range of taste in sounds. Some people buy certain gear to get clinical, sterile, accurate sounds. Others buy certain gear to get warm, distorted, character sounds.

I think it's very useful to know exactly what each piece of gear you have is capable of. Abuse it, make it distort, back off, see how clean it gets. Then you will know it's "sweet spot", or at least know it's weaknesses.

Understanding each link in the chain let's you make decisions that help you get the sound you want.

Frankly, I think VU meters are fairly useless. Sometimes to get the sound I like, I'm pegging a VU meter and I feel sorry for it. Othertimes, to get the clean sounds I want, the VU meter is hardly moving. Fairly irrelevant in my opinion.

It's a new digital world - peak meters are more important. That probably offends the old school, but let it be.

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Originally posted by wooden

since 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,

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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,



OK - I give up. You are willfully ignorant about what I am saying - and I can't say it any clearer so I give up. You are intent on misunderstanding and misquoting me, so I just give up.

There is absolutely no way that I am "insisting that you can go beyond 0dBFS" within the converter. I made this point very clear by saying "i'm not talking about digital clipping, running out of numbers."

I have agreed all along that you should not record too hot in the analog realm - at least, not unless you like the distorted sound. And this distortion is waaaay before 0dBFS clipping.

Why are you so intent on misunderstanding, and presenting bogus information - such as saying that you can clip a digital summing bus by exceeding 0dBFS inside the computer? This is false, misleading information.

Nobody is arguing about tracking too hot. I was taking issue with your bogus information concerning digital summing. Summing involves "summing" many digital tracks together. So even if you record 24 tracks at nominal 0dBVU - when you summ them together, they can exceed 0dBFS internally.

Using your advice - which I disagree with - you would say that you should lower the track faders, so the don't sum to anything higher than 0dBFS. You are saying it's possible to overload the digital summing bus - and I am disagreeing because this is not true.

Have you ever done the simple sine wave test I described above? I'm think you haven't, because it would become clear that you have egg on your face.

Anyway - I give up. It will still annoy me when I see you telling newbies this misleading information, but I guess they can work it out for themselves.

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Originally posted by Kiwiburger



OK - I give up. You are willfully ignorant




he he he. where strikes yet another chord.

Sorry you ran into this. You are 100% correct in what you are presenting here, at least in a very theoretical manner - you can't clip 32bit floating math.

Don't let where get under your skin too far.

-Todd A.

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You're right - I won't labour the point. It's fairly academic anyway - but I like to understand 'why' I have do something. I'm never satisfied I know what i'm doing until I know 'why' i'm doing.

Where is 100% correct about aiming for the sweet spot of a converter, which is typically around 18dB below full scale. This is absolutely spot on advice, compared to the general recommendation to track as hot as you can.

It was just his recurring statement that you do this to avoid clipping the digital summing bus that irked me. In my view you do this to avoid unwanted coloration in the analog circuit of the converter. And maybe unintentional coloration in the preamp and any other analog circuitry in the chain.

Analog generally gets dirtier the hotter it gets.

Pure digital audio (numbers, not voltage) get cleaner the hotter it gets - right up until the point it suddenly clips. But floating point math never clips.

The same principles apply when coming out of the box - but in practice, because of the loudness wars, this whole thing gets thrown out the window.

Sorry to be so pedantic - no hard feelings.

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Hey y'all...
I talked to Yamaha today about my 4416. They said I should be going in at-12dbfs.
They confirmed everything you guys were saying about input levels.
They didn't agree with me that the guy who wrote the manual should get a shot in the nuts for not putting this kind of info in the manual.
LOL
About the 32bit floating thingy...isn't that whole argument pointless if you track correctly, or am I missing something here?:D

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Protools HD uses 48 bits fixed (newer native DAWs use 64 bits, either fixed or floating).

But from the Protools website I got this fact:

48 bits fixed allows enough headroom to sum 128 tracks of phase aligned sinewaves all peaking at +12dbFS!

For practical purposes, you just aren't going to clip a summing buss - not even in Protools HD, which is arguably the worst.

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About the 32bit floating thingy...isn't that whole argument pointless if you track correctly, or am I missing something here?

Yes - if you track correctly, it would be hard to screw up.

I wasn't ever suggesting tracking too hot. I was simply trying to explain the reason why you don't want to track to hot.

It's not because you can clip a digital summing bus. It's for reasons in the analog realm.

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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.

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Originally posted by Anna Log

Hey y'all...

I talked to Yamaha today about my 4416. They said I should be going in at-12dbfs.

They confirmed everything you guys were saying about input levels.

They didn't agree with me that the guy who wrote the manual should get a shot in the nuts for not putting this kind of info in the manual.

LOL



It does seem weird that no manufacturer states that in their manual... I have a 400F and the manual says that I should adjust the gain so that the orange light lights up frequently but without lighting the red (clip) light. The orange light is -6dB, it's quite far into the alleged 22dB headroom (I wasn't able to understand that whole 22dB headroom thing until I read this thread).

I can see how recording hot can affect sound quality: since amplifiers have a maximum slew rate, the stronger the signal, the more difficulty they have carrying a high-frequency signal (if you send a signal that is twice as strong, it means the voltage swing is twice as fast, or something like that...).
I was reading on preamp design and how more gain usually means less bandwidth, very interesting.

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. I guess this can't be universal, but it does work! When I started doing that, it kinda scared me to see my guitar tracks so weak on the screen, but you have to realize that as you pile up track, the volume goes up.

As for the whole where/kiwi argument. I understand Kiwi's point and to me he his right (from my own experience). However, I noticed that if I run a plugin on the master bus, it can cause problem if the pre-fader signal is too hot. I guess not all plugins like to receive a signal hotter than 0dB.

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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.

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However, I noticed that if I run a plugin on the master bus, it can cause problem if the pre-fader signal is too hot. I guess not all plugins like to receive a signal hotter than 0dB.

Absolutely true. Many plugins are designed to model analog, and get crunchier the hotter you go, and may totally clip at 0dBFS. That is purely by design - it's not a limitation of the internal digital stream.

However - I personally believe when Mixing (as opposed to Pre-Mastering) you shouldn't have any plugins on the master bus. Otherwise you have limited choices when mastering. And because you should be mixing with sufficient headroom for mastering, especially if you intend to output via D/A converters. So you don't even need a limiter, because your master bus output is going to be well under 0dBFS.

I'm definately not recommending that anyone " slam the {censored} out of your {censored}". That is Where's willful ignorance on display.

Slamming into the masters will not sound as good as keeping proper fader levels ... the sonic difference is immediately obvious.

The results of abusing analog converters are immediately obvious - yes. Nothing to do with the "masters". That is analog thinking - it doesn't apply to digital.

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

That's a lazy mans method, and guaranteed to ensure under-recording, with resulting noise and loss of bit depth.

The only area of contention in my mind is concerning the misconception that you can't use the master fader to lower your mix to the appropriate output level. I see no justified reason to believe that - my ears can't hear any reason to be scared of doing this.

This thinking affects what Where calls "keeping proper fader levels ". I believe he is suggesting that to alter the output level of your mix, you must alter the level of all individual channels. That seems an incredibly restricted way of working.

Personally - I think analog VU meter thinking doesn't work that well with digital. Its fine for outside the box, but inside the box it's peak dBFS that matters most (which is why DAW makers give us peak meters). Possibly the exceptions would be certain plugins, but they often have their own meters for the specific task.

My experience of recent years is mainly with Cubase SX. I haven't used Protools for a while, and can't comment on how it sounds - but Digidesign literature suggests the same principles apply.

Seems to me that people with major issues with digital summing quality tend to be Protools users for some reason - there might well be something to it that I don't know about.

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Have been reading this thread and really don't understand really much of what is being said. I record directly into a MOTU 896HD and i usually keep my levels as hot as they can get before they clip, is this wrong?

I don't use any analog mixer, all my keyboardsmics go right to my MOTU.

please help
:(

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Originally posted by Sinesis

Have been reading this thread and really don't understand really much of what is being said. I record directly into a MOTU 896HD and i usually keep my levels as hot as they can get before they clip, is this wrong?


I don't use any analog mixer, all my keyboardsmics go right to my MOTU.


please help

:(



Hmmm.

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That's a lazy mans method, and guaranteed to ensure under-recording, with resulting noise and loss of bit depth.



I understand the noise issue here.
I thought we were recording at 24 bit no matter what the input level?
:confused::(

So is everything that comes in under -18dbfs... under 24 bits?
Or is it under -12dbfs.

Where o where do I start losing bits?
Yamaha said I should be at -12dbfs.

Maybe I should make up better songs, then this kinda stuff wouldn't matter so much.

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Originally posted by Anna Log



I understand the noise issue here.

I thought we were recording at 24 bit no matter what the input level?

:confused::(

So is everything that comes in under -18dbfs... under 24 bits?

Or is it under -12dbfs.


Where o where do I start losing bits?

Yamaha said I should be at -12dbfs.


Maybe I should make up better songs, then this kinda stuff wouldn't matter so much.




Now you know ahy I said what I said at the beginning of the thread ;)

This conversation can be daunting.

Don't say losing bits- you're not.

Say you're losing resolution- the converter isn't as accurate the lowel your level. That's a separate consideration outside your Mix buss, which has to sum all those signals together.


For now, just record healthy, but not hot. No peaks to 0. That will get you going.


Just remember to listen- If you hear distortion or saturation you'll know it. Watch your clip Lights. None should be on.

Now MAKE SOME MUSIC!!!!!!

he he.


-Todd A.

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Originally posted by The Chinese


Nothing really to flame in your post.... however, It's a bit advanced for the original poster, which is why I kinda just alluded to it in my earlier post.


Unfortunately, the truth is often complicated. Kiwi has it right, and his post(s) should have ended the thread. Too bad people who don't understand often have the most strident opinions....


I personally think that 32bit floating math is one of the bigest reasons that our engineering skills are falling by the wayside. I mean, you can keep tossing track after track after track with almost no thought. No longer do we have to meticulously record our tracks etc.


This is exactly right, you don't have to worry about this sort of thing anymore once you have 32 bit floating or even 48 bit fixed summing busses. This is straight computer science, there's no (rational) alternative argument possible.


Todd, why do you think this is a bad thing? Gain staging was an important engineering skill in analog boards, which we all mixed on for years. It's still important when using analog outboard gear. But for this application, it can be safely set aside - it's actually quite convenient, no shame in that.


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?


Like Kiwi said, the game is simple. On the way in, use lots of bits for more resolution, don't exceed 0dBFS on the peaks. If you want to run it a little less hot that's OK too. 24 bits gives you a good result in a pretty wide band.


On the way out, if you have 32 bit float or 48 bit fixed, don't sweat the summing. Of course if you're not staying digital all the way to the CD (printing to tape or something), you have to watch the level on the way out.


Honestly, this thread is much (impassioned) ado about nothing. However, there is some good info in here for people who don't understand the internal operation of digital gear, provided they can separate the wheat from the chaff.




Terry D.

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Originally posted by MrKnobs



Terry D.




Terry-

I'll tell you why I feel the way i do.

Often times, since I work on a large dubbing stage, with a more "traditional" setup, I often see andd pay the price for poor technique in Engineering. Not that Anna is going to send her mixes to a large stage, but in preparation for doing so. It's engineering, and the basic rules, IMO, should be somewhat followed so that when one travels to a {censored}kicker room with a more traditional setup they aren't fighting that kind of stuff. Where is spot on about that.

I recognize that what Kiwi said is 100% accurate, and also agree in theory with you. I don't believe in Where's absolutism on this at all, rather I feel that it's a "Just because its there doesn't mean you should use it" type situation.


another reason is plugin headroom- fine a mix buss is capable of infinite headroom, but may plugins are not. for example, I feel that Waves plugins have a definite limited amount of headroom, even on PTHD (this is the platform I use everyday, and is also my personal system). I find myslef with a much better sound by recording at 0VU ref to -20dBfs.

Bottom line is that I see young engineers all the time coming into the Post world (this is my professional business) that have a difficult time recording a VO, and then wonder why it sounds so saturated.

YMMV.

Hope i explained myslef somewhat clearly.

-Todd

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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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Originally posted by where02190



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?



I don't know about your method of working, but I don't have any converters in the signal path after my master fader. Once I'm digital, I stay digital all the way to the CD.

But I guess you're talking how it sounds in the monitors. I haven't noticed any difference, but to be honest I most often automatically work with the master at unity gain from many years of working with analog boards. I use groups to tie faders together (drums, vox, etc) so I don't have to pull a lot of faders down individually, and I use trim automation a lot also.

I'll try your experiment.

Terry D.

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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.

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