It's possible to do limiting, or avoid clippings, before sound hits HD? (no hardware)

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underp

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I don't have any hardware compressor or limiter. I'm going directly from the preamp to soundcard.

Is there any way to limit the sound, and prevent any clipping from going to the computer, and prevent any distortion from being recorded.

Thanks.
 
I would say probably not...

what software are you using to record?

I also think that hardware would have an overload limit? Could anyone expand on that?

I'm not familiar with stand alone preamps cos I go through a mixer preamp. With my mixer preamp I have to set the trim level so that it does not peak. I then find that it generally does not peak within my recording software.
 
If you set your preamp levels where they're supposed to be, you aren't going to be close enough to clipping to worry about it.
 
John has it right.

That said, there can be situations where it might not be feasable to monitor the levels and ride the fader. This can especially happen in live recording situations where you are recording yourself or where for logistical reasons there is not an engineer at the board at all times. In those instances, if you are recording direct to the computer, many recording software packages will let you put a plug-in signal chain on the recording input signal. In those cases you can put a limiter plug-in on the input signal just as a safety net in case the levels do start creeping up when no one is looking.

G.
 
Yeah man adjust your preamp gain. You want to get the most signal you can without clipping. Then I'd use your software for comp, eq. etc... I use outboard comps and eq's on some things and cubase VST for other instruments. Just remember if you do use outboard stuff before you go into your computer or HD don't over do it, cause you cant go back without re-recording it over again.
 
SouthSIDE Glen said:
...there can be situations where it might not be feasable to monitor the levels and ride the fader. This can especially happen in live recording situations where you are recording yourself or where for logistical reasons there is not an engineer at the board at all times. In those instances, if you are recording direct to the computer, many recording software packages will let you put a plug-in signal chain on the recording input signal. In those cases you can put a limiter plug-in on the input signal just as a safety net in case the levels do start creeping up when no one is looking.
Unfortunately, that doesn't really accomplish anything. If you clip the mic pre, or worse yet if you clip the A/D converter, it's too late to fix that by the time your signal reaches some plug-in. And once the signal is digitized, it is (by definition) already limited, so a limiter plug-in would have no work to do.

So basically, the answer is no, it's not possible to do this without hardware. The right answer is to set the levels to provide sufficient headroom so that no clipping occurs. Some people like to place a limiter between the mic pre and the A/D converter, but if you set up your gain staging correctly, it won't be necessary. In any event, once the signal has been converted to digital, it's too late to fix any such problem.
 
The right answer is to set the levels to provide sufficient headroom so that no clipping occurs.

...is the right answer.

Are you recording 24 bit?

If so, you have more headroom than you will ever need to record any single track. If you set the output of your mic pre (DI, etc.) to peak at -12 dB, or even -18 or -20 below -0 dBFS.

-0 dB Full Scale might be indicated differently in different meters (hardware or software) - 0 dB VU might be indicated as -18 dB, for example, so you might set the output of your hardware to peak there. The low level details will not get lost in noise as they would have if you tracked analog this low, but distortion of the loudest parts is generally much worse in digital than in analog.
 
I'm recording in 24-bits.

But i'm doing some live recordings where i need the gain a little bit high, cause the vocalist tends to whisper in some sections.

But also, there's some sections where he lose control and could cause i little bit of distortion in my PC.


So... i want to maintain enough gain to the whispering sections and also, prevent any distorion from reach the hard drive.


It is possible to do limiting with some kind of plug-in or something in real time?
( maybe a stupid question, because the sound hits the soundcard before any post processing ) but just in case, i want to know.


thanks.

**********Edit**********

I have a delta 1010lt

Soft:
Pro Tools
Audition 1.5 / 2.0
Ableton Live
Krystal
Vegas
 
You want to get the most signal you can without clipping.
No, that's totally what you DON'T want to do. You want to keep the signal where the *hardware* wants to be. 0dBVU is still 0dBVU. It never went anywhere or disappeared when digital came up on the block. Getting the "most signal without clipping" is going to add somewhere around *18dB* of gain (and noise, and fuzziness, and loss of focus, and distortion) to the signal before it even gets to the converter (which is ALSO praying for a 0dBVU signal on the input).

It is possible to do limiting with some kind of plug-in or something in real time?
Absolutely not - The signal is already digital (and as mentioned, WAY too hot in the first place). By then, you've already clipped the converter.

We've got more headroom now than at any time since audio recording was conceived people... A 24-bit signal that PEAKS at -47dBFS still has higher resolution than a compact disc. HEADroom is GOOD room. Let's use some of it. Recordings sound SOOOOOO much better when we do...

You're probably going to use up all that headroom in the mastering phase anyway - Do yourself a favor and use it up ONCE instead of at every chance you get.
 
Pugins happen after the signal is past the AD converters, so that won't help as you had already surmised.

Your stuck with recording it low, and boosting during mix, or I guess buying a compressor. But really dooing it in the mix would most likely be better then massive gain reduction by a compressor.

EDIT: John beat me to it....
 
intersting stuff here...

I've only recently moved to recording 24 bit. I took the rather false and foolish aproach that I only needed to record in 16 bit cos that's all you rip the CD to.

And the headroom thing... I've known for some time what headroom is but never realised how that works in the digital domain. So, yet again, I have learnt something:)

And I'd like to learn more... about the differences between 16 bit and 24 bit (apart from 24 being bigger that 16) and the relationships between bits and dBs and such. I've read elswhere that when you are recording in 16 bit and you have a signal at 0Db, only then is the recording truly 16 bit. For each 6dB below 0dB you lose a bit. -6dB = 15 bit, -12dB = 14 bit and so on. So, is it the same when you are recording in 24 bit? i.e. do you still only lose 1 bit per -6dB? In which case, if I do the math(s) correctly then you can go down to -48dB and still be attaining a 16 bit digital capture. My reasoning appears to be born out by this statement:-

Massive Master said:
We've got more headroom now than at any time since audio recording was conceived people... A 24-bit signal that PEAKS at -47dBFS still has higher resolution than a compact disc.

I feel a google coming on;)
 
underp said:
But i'm doing some live recordings where i need the gain a little bit high, cause the vocalist tends to whisper in some sections.

I would imagine that the bigger problem here, that can't be fixed by anything other than exeptional isolation, is bleed.

At least that's the way I see it... and I generally don't know what I'm talking about...
 
underp said:
But i'm doing some live recordings where i need the gain a little bit high, cause the vocalist tends to whisper in some sections.

But also, there's some sections where he lose control and could cause i little bit of distortion in my PC.
Sounds like what you need most is training for your vocalist in two areas:

- basic singing
- mic technique

You may not be able to get him to change his singing style, but if you can teach him a bit of mic technique, it may make your job quite a bit easier. A good vocalist needs to know how to work the mic to compensate for the variations in levels caused by his singing.
 
Massive Master said:
No, that's totally what you DON'T want to do. You want to keep the signal where the *hardware* wants to be. 0dBVU is still 0dBVU. It never went anywhere or disappeared when digital came up on the block. Getting the "most signal without clipping" is going to add somewhere around *18dB* of gain (and noise, and fuzziness, and loss of focus, and distortion) to the signal before it even gets to the converter (which is ALSO praying for a 0dBVU signal on the input).


Absolutely not - The signal is already digital (and as mentioned, WAY too hot in the first place). By then, you've already clipped the converter.

We've got more headroom now than at any time since audio recording was conceived people... A 24-bit signal that PEAKS at -47dBFS still has higher resolution than a compact disc. HEADroom is GOOD room. Let's use some of it. Recordings sound SOOOOOO much better when we do...

You're probably going to use up all that headroom in the mastering phase anyway - Do yourself a favor and use it up ONCE instead of at every chance you get.
I stand corrected.

That makes totally good sense. I am a guitar player before recording guy. I would always here guys in the studio tell me you want the guitar tracks hot. I guess that's why I have it stuck in my head.
 
Gilliland said:
Sounds like what you need most is training for your vocalist in two areas:

- basic singing
- mic technique

You may not be able to get him to change his singing style, but if you can teach him a bit of mic technique, it may make your job quite a bit easier. A good vocalist needs to know how to work the mic to compensate for the variations in levels caused by his singing.

Indeed, and I've posed the same point elsewhere when discussing diversity in dynamics and coping with them during recording.


BUT

It may be that, artistically speaking, that's what they want to capture... extremes in dynamics.
 
For what it's worth I recently started thinking of headroom in Digital audio in rough refrence to analog - since 0dBFS is the limit in digital, and there is no way that I will record or mix way up there, I genereally keep a mental reference of -24dB digital as my "zero" point. 10-15db up from that is the quality 'safe zone' with my hardware - leave the creative loudness maximizing to the mastering engineer who hopefully works in a room that has pristine acoustics and equipment.

Has anyone read 'mastering audio" by Bob Katz. I thought it was well worth the investment, despite the fact that it is now slightly outdated in respect to technological assessments (2002-2003) - I don't think the concepts have changed though.

at any rate .. That's why there are Jedi masters like John around to decode the mysteries ;-)
 
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