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Clipping while recording - no more warning in L8?


PBenz

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There is nothing above 0dBfs in the digital domain. That is why it is called 'full scale'.

 

Your interface hardware may have LEDs or something to indicate that the incoming electrical level is greater than what it can encode digitally (this is a clip) but regardless of how much over that is, the overage is still never going to be encoded.

 

Whether your DAW and your interface software create a warning when full scale is reached is the issue here.

 

Not that it should matter.

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There is nothing above 0dBfs in the digital domain. That is why it is called 'full scale'.

 

Your interface hardware may have LEDs or something to indicate that the incoming electrical level is greater than what it can encode digitally (this is a clip) but regardless of how much over that is, the overage is still never going to be encoded.

Understood.

 

 

Whether your DAW and your interface software create a warning when full scale is reached is the issue here.

 

Not that it should matter.

But it does matter to anyone who tracks him or herself. One should not be expected to watch the meters while trying to deliver their best performance. I personally don't care if the interface or the DAW gives me the warning - I just want the warning. TotalMix (the mixer for the Fireface), gives you that warning, but it doesn't stick.

 

I am checking now to see if this can be enabled.

 

I learned a lot from this thread. I will say again - clipping will be much, much less of an issue for me going forward.

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Hey, what... You don't like my avatar? LOL

 

The little headroom number value in the Logic channel strip (above the bar meter) is continuous hold by default (it will reset only when you restart the transport). If at the end of a pass this read '0.0' you have almost always clipped the recording. The problem then becomes finding the clip instance(s) and deciding whether or not you can live with them, sonically. If you're running a tracking session, stopping to do this is a bit of a vibe-killer and is seriously unprofessional.

 

Hence I would recommend leaving yourself lots of room. If it's 10dB from clip, you still have, what, 130dB dynamic range at 24 bits? That's pretty damn good. And if it's too quiet, you can add a whole bunch of gain in the DAW before you start getting into a situation where the DAW is contributing significantly to the noise floor.

 

Hey, what is your input gain-staging like? Are you running compressors or EQs with a lot of make-up gain? Because the first place I hear overload is almost never at the A/D. My preamps, etc. will start crunching out first...

 

Which can be desirable sometimes, of course.

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The little headroom number value in the Logic channel strip (above the bar meter) is continuous hold by default (it will reset only when you restart the transport). If at the end of a pass this read '0.0' you have almost always clipped the recording.

I knew that! :) Duh! You're right! I don't need no stinkin' clip detector - just seeing 0.0 is enough. Just have to be on the lookout for it.

 

Hence I would recommend leaving yourself lots of room. If it's 10dB from clip, you still have, what, 130dB dynamic range at 24 bits? That's pretty damn good. And if it's too quiet, you can add a whole bunch of gain in the DAW before you start getting into a situation where the DAW is contributing significantly to the noise floor.

Yes, definitely. There's a ton of misinformation out there on this issue, and this thread opened my eyes to it.

 

Hey, what is your input gain-staging like? Are you running compressors or EQs with a lot of make-up gain? Because the first place I hear overload is almost never at the A/D. My preamps, etc. will start crunching out first...

4 mics feed into a Focusrite ISA 428, then direct into the Fireface. The other 4 feed into the preamps on the Fireface (I use those for my toms). Very simple.

 

I'm glad you asked, though, since this setup bugs me. The gain on the 428 is set so low - it barely moves the meters (on the 428). And now I'll be going even lower to open up that headroom as you and others have suggested. Is this "correct"? Could I be doing something different to improve the sound of my raw tracks? I've read other posts where people have put attenuators (like an ATTY) on the output of the preamp in order to drive the preamp harder, but I tried this during the brief time I had an API 3124+ and didn't really care for it. It just seems that my 428 is not doing much, but perhaps this is how it should be.

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The little headroom number value in the Logic channel strip (above the bar meter) is continuous hold by default (it will reset only when you restart the transport). If at the end of a pass this read '0.0' you have almost always clipped the recording.

I knew that! :) Duh! You're right! I don't need no stinkin' clip detector - just seeing 0.0 is enough. Just have to be on the lookout for it.

 

Hence I would recommend leaving yourself lots of room. If it's 10dB from clip, you still have, what, 130dB dynamic range at 24 bits? That's pretty damn good. And if it's too quiet, you can add a whole bunch of gain in the DAW before you start getting into a situation where the DAW is contributing significantly to the noise floor.

Yes, definitely. There's a ton of misinformation out there on this issue, and this thread opened my eyes to it.

 

Hey, what is your input gain-staging like? Are you running compressors or EQs with a lot of make-up gain? Because the first place I hear overload is almost never at the A/D. My preamps, etc. will start crunching out first...

4 mics feed into a Focusrite ISA 428, then direct into the Fireface. The other 4 feed into the preamps on the Fireface (I use those for my toms). Very simple.

 

I'm glad you asked, though, since this setup bugs me. The gain on the 428 is set so low - it barely moves the meters (on the 428). And now I'll be going even lower to open up that headroom as you and others have suggested. Is this "correct"? Could I be doing something different to improve the sound of my raw tracks? I've read other posts where people have put attenuators (like an ATTY) on the output of the preamp in order to drive the preamp harder, but I tried this during the brief time I had an API 3124+ and didn't really care for it. It just seems that my 428 is not doing much, but perhaps this is how it should be.

 

I don't think this is how it should be... The thing I was taught about gain-staging, which really applies with all signal path, was to try to get the right level as close as possible to the source (mic or instrument) and then to try to maintain this level with as few adjustments as possible throughout the chain.

 

Sounds to me like you have some mis-matched equipment. Are the outputs from the Focusrite thing balanced, nominal +4dB? And the channels that you are putting them into on the Fireface, do they have preamps? You need to take the ouputs of the Focusrite pre's into line level inputs in order not to overload the inputs of the next stage... And even at that, there may be a selection in the RME software that specifies the nominal input level (-10dBV or +4dB), which you should match to the specified outputs of the Focusrite.

 

Running one preamp, padding it, and then running it into another preamp will probably never sound that cool (transistor preamps don't distort like tube regardless, and should be overdriven with extreme discretion). IMHO those API 312s are some of the best amps money can buy. Clear, deep, reasonably fast... I love them. I'm not alone in this... Friends who work in big places in LA are saying that API is fast overtaking SSL as the preferred mix console for pop and rock sessions down there, and their consoles use the same 2520 design.

 

If I could only have one preamp, the 212/312/512 would be it.

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Thanks Marcel. Upon closer inspection, the Fireface inputs were set to mistakenly -10dBV when they should have been +4dB to match the Focusrite's outputs. Changing this brought the input level down and allowed me to turn up the gain a bit on the Focusrite, and now the meters are registering. (I'm by myself so I can't really see what the values are but at least they are clearly moving now).

 

My kid must have gotten in there and changed the setting. Oh wait - I have no kids. Can I rent a kid somewhere so I can blame this on him? LOL

 

I'm definitely not running preamp outputs into another preamp. I think I'd have to be really high to try something like that. 8)

 

Re: the 3124+, man I wanted to like that preamp. Especially on drums. But it just wasn't cutting it for me, especially on the kick. I can't help but thinking I had a bad unit or something - the damn Fireface preamps sounded better to me (for the kick). Recently, I picked up that Preamps in Paradise DVD from 3daudioinc.com, which I found to be very educational. It gives you the opportunity to hear 24 different preamps across drums, guitar, bass, and vocals. If I were shopping for a new preamp for my drums, that True Systems Precision 8 sounds pretty sweet.

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Yeah, a friend of mine bought a box of those Precision 8s and loves them. I haven't had a chance to use them yet, although I find it hard to imagine liking anything more than APIs.

 

To each his own, y'know?

 

Glad you got things working. BTW, a compressor or limiter in between the pre and the interface is also going to give you more opportunity to 'push' the preamp while not overloading outputs... And give the inline pad option another go, now that things are working right. May sound totally different. I've done that with tube pre's a lot in past, with good results.

 

Are you sure you just don't like the sound of overdriven transistor mic pre's as much as you think you should?

 

It's not a sound I enjoy in most situations...

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Hi!

This is why I still recommend learning and utilizing proper gain staging, despite the fact that you can ignore gain staging "rules" at many/most points with a DAW. The first step is to be sure that what is reading at the start is reflected properly at every sot in the chain that has metering.

 

As Marcel has said, once you hit full scale, there is no more. If you look at a waveform (not usually the most accurate way of appraising, but so be it...), you will see the squared off clips when you exceed full scale. The ADC conversion has received a signal that exceeds it's limit, and cannot do anything with it...except cap it, basically. If properly gain staged and calibrated, Logic will never see anything over 0.0dB. But, if you see it get close to 0.0dB, it is your cue that you are nearing full scale at the ADC. Back it off! Unless you are close mic'ing a 747, you have PLENTY of headroom. Even with 16 bits you still should have enough...but, 24 is more than enough, and is a nice figure to work with IMO.

 

PBenz- the book you have coming should help make things much clearer. It is a relatively deep read...it is an incredibly deep subject, and to grasp it beyond layman's terms and explanations would (in most cases) require ALOT of school time. The book allows "most of us" to understand the basics without the school part. I am VERY interested to hear what you think once you have finished reading- please write back? It has been very cool to see how you are learning right now. It is VERY refreshing to see someone who is genuinely so enthusiastic, and is showing a true caring for something they obviously love. Very refreshing thread...

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Nikki - wait - so I have to actually read that book once I get it? LOL Seriously, though, I am very much looking forward to it - it will be here tomorrow, and I'll definitely write back once I get into it. Hopefully the knowledge I gain will not only help me to get the best sound out of my little home studio, but also allow me to intelligently wade through all the opinions out there, form some solid ones of my own (based on facts - not other's opinions!), and maybe even be able to help out others down the road.

 

I'm sure it's obvious but I'm relatively new to the wonderful world of recording audio. I started off a few years ago on Cubase SX 2 recording nothing but MIDI (V-Drums). But ultimately I got sick of playing electronic drums and just had to return to the real deal, so I treated myself to a new DW kit last year and decided to take on all the challenges associated with recording them. I studied room acoustics, and still managed to screw up the low end (by basically ignoring it. :-) ). There is so much more to learn. So much more money to spend. :-)

 

I actually already knew about how clipped waveforms looked, and I recall the light bulb going off when I saw it for the first time. Makes total sense.

 

Anyway, thanks to you and Marcel and everyone for your help. Now I'm off to order my Pro Tools LE system. (ha! Just kidding. Sort of. ;-) I'm still in search of the "perfect" DAW - perfect for me, at least. Logic 8 is so close, I just get so frustrated with it at times. )

 

Paul

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  • 1 month later...

Nikki - since you asked me to write back after I've had a chance to go through that book, I thought I would do so by resurrecting this old thread.

 

I actually finished it in a few days, although I admit I didn't read every word. I initially jumped straight to the Myths section which totally blew my mind and really opened my eyes! I still go back and reference that section from time to time, and I've been hoping to some day create a summary document to share with my friends who won't have the time to read the book.

 

I could probably spend the next few hours writing about what I've learned, but I'll spare you the details. :-) I just wanted to come back and thank you for recommending that book. It's changed the way I approach recording, and it's given me a solid technical foundation that I can use to filter through the snake oil that's constantly being spread around the Internet regarding sampling rates, bit depth, analog vs digital, etc.

 

One thing that's been on my mind lately is Dan Lavry's "Sampling Theory" document which makes the case for 60kHz as the theoretical optimal sampling rate. I tried to read that document several times but I have to admit it's just too dense for me. Armed with the facts from Nika's book, I was hoping to find a good discussion about this (and sampling rates in general) on the ProSoundWeb site, and I just this evening I finally found a post that referenced this 42-page post on a different forum that started in 2001 by non other than Nika himself! Here's one of his quotes:

 

"And that silly little mathematical principle, my friends, is the foundation for part of the great marketing hoax of 96kHz."

 

This is going to be a juicy read! :-)

 

Cheers, and I wish everyone here a very Happy New Year.

 

(ps - I've yet to convince myself that I need to record in anything other than 44.1)

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  • 2 years later...

Guys, am i not getting this or can you not just play the track back at the EXACT same level you recorded at in Logic, all the way through and see does the clip meter go?? If your clip meter isn't going realtime then after recording, don't touch any channel strip settings and playback???

Somebody explain please why this wouldn't work as this would be just as convenient as the pop up message?

Joe,

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In a nutshell, as of Logic 8 the dev dispensed with the overload pop-up message that was present in L7 and previous version of Logic/Logic Audio. So yes, the levels will play back exactly as you record them. No change there. The only change is that the machine no longer reports to you when a recorded signal has clipped.
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  • 4 years later...

Thought I would revive this 5 year old thread with, I too miss this feature thats been gone for like 10 years. Was wondering where it went ;)

 

I am trying to bounce some stems and it looks like they are clipping just a hair and I would like to know for sure. Is there a utility or plugin that I can use to see if my audio is clipped? Im not clear on how to get this info out of the audio editor. Obviously the quicker the method the better.

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