When working with DAW we all know that when we exceed the 0db we clip. But in Analog/Hardware Mixer and Effects Processors (eq,etc.) the clipping is way above 0db and it seams +16db is fine when we are in live concert situations (in the master buss channel) ..
What does this mean? Computer = 0db = Clip and Hardware = 0db = No CLIP .. i mean after all its the same unit (decibels) Help!!

 

Diginal?

 

@ Mild pump milk,

*no* I think that it must be : 'Vaginal'...

 

See here:

http://en.wikipedia.org/wiki/Decibel#Suffixes_and_reference_values
and
http://en.wikipedia.org/wiki/DBFS

In analog professional audio applications dBu is used, whilst in digital audio uses dBFS

 

when it comes to analog 0 is 0db
but 0 in digital is -18

 

Thanks for asking that!

Yes, what rickbarratt said is mostly true. Except that not all analogue equipment is calibrated to 0 dB [which is also called "unity gain"] being at -18 dBFS, but thereabouts. Some is calibrated even to -14 dBFS, but mostly to -18 dBFS or lower. That difference between unity gain and absolute maximum is called "the headroom". Better quality analogue equipment is calibrated to even lower levels than -18 dBFS as unity gain and has more headroom. Yes, dBFS means "decibels full-scale", meaning the digital scale. You can imagine why is that so? Analogue equipment has a "sweet spot" which is usually calibrated to -18 dBFS when the audio has the most optimal sound to noise ratio and sound to distortion ratio. People had to invent this kind of metering for analogue audio because in analogue audio you have to consider this sweet spot when mixing and also because you have to have some point of measure for the specs.

That's why you can read around the web about "gain staging" as a good mixing practice, even when mixing in the box [DAW]. What gain staging is is mixing at analogue levels in digital. It gets you better mixes for various reasons, but mostly because most, and especially cheap, digital converters, can start to clip much earlier that 0 dBFS, so you don't actually hear the audio at its best. So it's always a good idea to use VU meters on every channel that are calibrated to -18 dBFS while mixing. An interesting after-effect of that is that you will never get clipping on the master channel and you just don't have to worry about it. In most cases you will get peaks from -6 to -3 dBFS at the most which is great when you need to send your mix to a mastering engineer later. They'll thank you for sending a good sounding mix they can work with and get better results from! Also, if you like some of your channels clipped or distorted you can do it, but keep their level at -18 dBFS, or better yet keep them at the level that suits the mix best.

I hope this helped! Cheers!

 

I think it is time for a quote of the German Enginieering Genius and Professor of everything Audio named Sengpiehl here...

"There is no such standardized reference. x dBFS is a digital voltage level
(peak) and y dBVU or dBu is an analog voltage level (RMS).

Digital and analogue are two totally different realms.

That's why there is no relation between dBFS and dBVU or dBu, whatsoever.

Analog meter (ppm): attack time 10 to 300 ms − reading rms values.
Digital meter: attack time < 1 ms − reading peak values. That is really some
difference.

Advice: Watch only your digital meters and go up to 0 dBFS, but never go
over this mark. We use "headroom" in the analog domain. That is OK, but
we don't need digital "headroom" as an always "unused" forbidden
zone.
You are free to choose your private headroom, if you like that, but there is no
standard that you have to do that. The demand for a high modulation level, stand
in the contrary to the claim, to avoid overloading.

Never take the following funny guessing game for granted. Use it only as a rough guide:

European & UK calibration for Post & Film is −18 dBFS = 0 VU = +4 dBu"

Source: http://www.sengpielaudio.com/calculator-db-volt.htm

PS: Make sure to bookmark his Page. It might look shitty, but it's a treasure of solid information.

 

thanks you all

 

Some says analog modeled plugs works best when feed with a calibrated -18db ... cause they are build from an analog model.
So it seems better not to over charge the level of recordings ... Yes ok ...
But, numeric recording is linear ... it means that full 16 bit definition is only reached at 0dBfs ... where are we at -12db ...
This one of the reason why audiophiles quickly said that CD sound was ... awful. The were right before some tricks like dithering
So you must use good analog to digital converters, and use 24 bit ... bounce in 32bit before mastering, keeping the most of audio définition

 

I don't understand what you said about the "Numeric recording". "Full 16 Bit definition is only reached at 0 dBFS"

That assumption is wrong! Because the bit depth itself DEFINES the 0 dBFS Limit and not the other way around. See, 1 Bit is equivalent to a dynamic range of 6 dB. So 16 Bit equals the overall dynamic range of 16 x 6 dB, giving you a potential headroom of 96 dB (from -96 to 0 dB FS). The interesting thing is, not what happens at the upper Limit, but at the lower limit near the noise Floor. -96 dB is not that low a level. We can still hear what's going on there. And at 16 bit the Converter starts to introduce digital fragmentation. And dithering is need to mask these fragments by "wrapping" them into noise. When using 24 Bit the noise floor much lower, -144 dB and -192 dB @ 32 Bit, so digital fragments are not an issue.

Btw, you only benefit from the headroom of the 32 Bit Float Format in the digital realm when processing inside your DAW. Once the signal is send to the DAC it will be 24 Bit, because there are no 32 Bit D/A Converts available yet.

All in all, Bit depth is a bit overrated. Sample Rate is more important. Especially in Recording