0 dB VU is WHAT level in the DAW meter?

[1_dup16]

0 dB VU is WHAT level in the DAW meter?

ADDA converters have analog components too, and are tuned to work with other analog gears. So it would be useful to work at 0 dB VU to ensure that the analog components of the ADDA interface are not overloaded

 

But where is this 0 dB VU from full scale? -12, -16, -18, -26????

 

Thanks

[1_dup16]

a very interesting answer:

 

http://members.fortunecity.com/andrewvee2/pages/audio_levels.htm

 

"How Much Headroom is Enough?

 

Have you noticed that solid-state equipment starts to sound pretty nasty when used near its clip point? All other things being equal, the amplifier with the higher clipping point sounds better, in my opinion. Perhaps that's why tube equipment (with their 300 volt B+ supplies and headroom 30 dB or greater) often has a "good" name and solid state equipment with inadequate power supplies or headroom has a bad name. Traditionally, the difference between average level and clip point has been called the headroom, but in order to emphasize the need for even more than the traditional amount of headroom, I'll call the space between the peak level of the music and the amplifier clip point a cushion. In the days of analog tape, a 0 VU reference of +4 dBv with a clipping point of +20 dBv provided reasonable amplifier headroom, because musical peak-to-average ratios were reduced to the compression point of the tape, which maxes out at around 14 dB over 0 VU. Instead of clipping, analog tape's gradual saturation curve produces 3rd and 2nd harmonics, much gentler on the ear than the higher order distortions of solid state amplifier clipping.

 

But it's a different story today, where the peak-to-average ratio of raw, unprocessed digital audio tracks can be 20 dB. Adding 20 dB to a reference of +4 dBv results in +24 dBv, which is beyond the clipping point of many so-called professional pieces of gear, and doesn't leave any room for a cushion. If you adapt an active balanced output to an unbalanced input, the clipping point reduces by 6 dB, so the situation becomes proportionally worse (all those headroom specs have to be reduced by 6 dB if you unbalance an amplifier's output). Be particularly suspicious of consoles that are designed to work at either professional or semi-pro levels. To meet price goals, manufacturers often compromise on headroom in professional mode, making the so-called semi-pro mode sound cleaner! You'll be unpleasantly surprised to discover that many consoles clip at +20 dBv, meaning they should never be using a professional reference level of +4 dBv (headroom of only 16 dB and no cushion). Even if the console clips at +30 dBv (the minimum clipping point I recommend), that only leaves a 6 dB cushion when reproducing music with 20 dB peak-to-average ratio. That's why more and more high-end professional equipment have clipping points as high as +37 dBv (55 volts!). To obtain that specification, an amplifier must use very high output devices and high-voltage power supplies. Translationbetter sound. One of the most common mistakes made by digital equipment manufacturers is to assume that, if the digital signal "clips" at 0 dBFS, then it's OK to install a (cheap) analog output stage that would clip at a voltage equivalent to, say, 1 dB higher. This almost guarantees a nasty-sounding DAT recorder, because of the lack of cushion in its analog output section.

 

To summarize, make sure the clip point of all your analog amplifiers is at least 6 dB (preferably 12 or more dB) above the peak level of analog material that will run in the system. I call this additional headroom the cushion.

 

How can you increase the cushion in your system, short of junking all your distribution amplifiers and consoles for new ones? One way to solve the problem is to recalibrate all your VU meters. You will not lose significant signal-to-noise ratio if you set 0 VU= 0 dBv or even -4 dBv (not an international standard, but a decent compromise if you don't want to throw out your equipment, and you have the expertise to make this standard stick throughout your studio). Try it and let me know if things sound cleaner in your studio. Once you've decided on a standard analog reference level, calibrate all your analog-driven VU meters to this level. Here's a diagram describing the concept of cushion.

 

Translating between analog and digital points in the system

 

Let's discuss the interfacing of analog devices equipped with VU meters and digital devices equipped with digital (peak) meters. When you calibrate a system with sine wave tone, what translation level should you use? There are several de facto standards. Common choices have been -20 dBFS, -18 dBFS, and -14 dBFS translating to 0 VU. That's why some DAT machines have marks at -18 dB or 14 dB. I'd like to see accurate calibration marks on digital recorders at -12, -14, 18, and -20 dB, which covers most bases. Most of the external digital meters provide means to accurately calibrate at any of these levels.

 

How do you decide which standard to use? Is it possible to have only one standard? What are the compromises of each?

 

To make an educated decision, ask yourself: What is my system philosophy? Am I interested in maintaining headroom and avoiding peak clipping or do I want the highest possible signal-to-noise ratio at all times? Do I need to simplify dubbing practices or am I willing to require constant supervision during dubbing (operator checks levels before each dub, finds the peaks, and so on)? Consider your typical musical sources. Are your sources totally digital (DDD)? Did they pass through extreme processing (compression) or through analog tape stages? Pure, unprocessed digital sources, particularly individual tracks on a multitrack, will have peak levels 18 to 20 dB above 0 VU. Whereas processed mixdowns will have peak-to-average ratios of up to 18 dB (rarely up to 20). Analog tapes will have peak levels up to 14 dB, almost never greater. And that's how the three most common choices of translation numbers (-18, -20, and -14) were derived. That's also why each manufacturer's DAT recorder has a different analog output level. It used to be easy to match a recorder to a console. Only one major manufacturer of DAT machines provides user calibration trims for analog inputs and outputs. My least favorite DAT machines have fixed output levels, and I've installed custom trimpots in many of them."

[where02190]

It depends on how the converter is calibrated. for instance, the Digi 001 and M-box are claibrated to -18dbfs=0dbu, whereas the 002 is calibrated to -14dbfs=0dbu.

 

Check manufacturers specs for your specific unit.

[philbo_Tangent]

Originally posted by 1:

0 dB VU is WHAT level in the DAW meter?

ADDA converters have analog components too, and are tuned to work with other analog gears. So it would be useful to work at 0 dB VU to ensure that the analog components of the ADDA interface are not overloaded

 

But where is this 0 dB VU from full scale? -12, -16, -18, -26????

 

Thanks

A lot depends on the audio and the reference level of the A/D or D/A - - if the audio is a steady 1KHz or 400 Hz test tone, and the reference is 0 dBV, then 1 Vrms will give you 0 dBFS.

 

If the reference is 0 dBu, then 0.775Vrms will give ou 0 dBFS.

 

If the audio is NOT a steady test tone, then all bets are off - it will depend on the crest factor of the audio signal (crest factor = peak volts / rms volts) and the integration period of the VU meter.

 

All VU meters are set up to integrate (or, loosely speaking, average) the sound level over a certain time frame. The time frame can vary somewhat.

 

But, because of this, a VU meter will read a pretty low level (perhaps -17 dBVU) on a short, large peak (such as a snare hit) that might well reach a peak level of +6 dBV.

 

An often-used guideline for translating (music) VU levels to DAW record levels are: -24 to -16 dBVU should provide -6 to -3 dB peak FS (when recording into a DAW). The exact numbers depend on the crest factor of the music - I generally set the DAW so the peaks of the music hit -6 dBFS if recording at 24 bits, or -3 if recording at 16 bits.

 

Hope you find this helpful...