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Thread: Metering in digital domain

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    levels in digital domain

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    I have been reading a lot of info on where to keep levels when recording and mixing strictly ITB. Almost every thread i have been reading ends up in a debate and the main reason for the thread gets lost to an argument. When I first started recording digitally i would keep the input levels as close to unity as possible and my mixes sounded like crap. Then I started to crank my monitors and record and mix with the levels significantly lower. The mixes started to sound better but i still think that my gain staging thru plugins are wrong. I know this has been discussed a lot but could someone give a little advice or point me to a good thread that has good info and less arguing? Thanks in advance. Id also like to add that ive been using plugins like ssl comp and v-comp, from what i understand there are specific gain staging recommendations for these plugs to respond as intended. I also have been using the Sonalksis free g to check the levels of every channel.
    Last edited by offcenter2005; 06-24-2012 at 20:04.
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    This has come in handy a few (hundred thousand) times:

    Proper Audio Recording Levels | Rants, Articles

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    This is an old chestnut that even experienced engineers have trouble with. I'll try my best.

    I think the first thing to get your mind around is decibels, their suffixes, and the difference between them. For instance, most analogue gear is metered on the dBu (decibels unloaded) scale and this can be confusing when trying to relate it to the dBfs (decibels fullscale = 0dBfs). The first thing to notice is that 0dBfs does not equal 0dBu. On an analogue console you'll most likely see 0dBu somewhere in the middle of the meter with a positive scale all the way up to +22dBu or more. This scale was devised to indicate headroom above nominal operating level, which is supposed to be around 0dBu to +4dBu. Conveniently, a 1kHz sine tone at +4dBu will reflect as 0VU on a VU meter (Volume Units), the slow-moving, loudness-type meter of old. So, how do we translate this over to the digital system?

    Well, the first thing to know is that all digital gear that converts an analogue signal to digital will have it's own internal calibration. This calibration will translate to 0VU/+4dBu/dBfs. Thus, this is a typical calibration equation:

    0VU = +4dBu = -18dBfs

    This equation tells us that in this specific system, it's a good idea to keep your peaks around -18dBfs to leave sufficient headroom for erroneous levels. Now, every system is different and every interface has it's own calibration so if I were you I would check online for the calibration of your particular interface. The dBfs figure will give you an indication of where to keep your highest peak levels.

    Of course, there are no hard and fast rules, so use your discretion. It's OK if a few peaks go over -18dBfs. That's what the headroom is there for. As always, however, conservative levels are a better idea than slamming anything.

    Now, when it comes to plugins, the basic rule is to keep perceived loudness equal at the input and output of the processor. This usually renders peak level useless to reference. You have to use your ears. It's not that difficult, however, so make good use of the bypass button! That's pretty much it.

    Hope that helps.

    Cheers

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    Thanks for the replies. From what i have read , in between all the BS thats what I assumed was the underlying message. I started noticing a fuller clearer mix when keeping my levels lower. If i turned up my monitors and turned down the channels everything seemed to come into focus. Thanks again.
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    Quote Originally Posted by offcenter2005 View Post
    I started noticing a fuller clearer mix when keeping my levels lower. If i turned up my monitors and turned down the channels everything seemed to come into focus.
    Most modern DAW software uses 32-bit (or 64-bit) floating point math, so signal levels within the software have very little effect on sound quality. For proof, see the article I wrote for GC Pro's Audio Solutions magazine:

    Audio Solutions magazine

    Click Archives, then select the Summer 2011 issue. It explains DAW internal calculations in detail. I also prove the same points with nulls tests in my hour-long video in the section starting at 53:39. Note that some distortion and tape-sim type plug-ins are level sensitive, but most "normal" plug-ins are not.

    Now, getting audio in and out of a converter is another matter, though again any competent converter should be able to handle a wide range of input levels without compromising the sound quality. For example, look at the specs on the last page of this Lavry converter manual:

    http://lavryengineering.com/pdfs/lavry-da924-manual.pdf

    Even when sending a loud signal at 1 dB below clipping, the distortion is only .0009 percent. Other converters may not be this clean, but most are still very clean right up to the point of gross distortion. I believe that when people reduce levels and notice the sound improved, the problem was occurring somewhere else in the signal chain.

    --Ethan

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    Quote Originally Posted by Ethan Winer View Post
    Most modern DAW software uses 32-bit (or 64-bit) floating point math, so signal levels within the software have very little effect on sound quality. For proof, see the article I wrote for GC Pro's Audio Solutions magazine:

    Audio Solutions magazine

    Click Archives, then select the Summer 2011 issue. It explains DAW internal calculations in detail. I also prove the same points with nulls tests in my hour-long video in the section starting at 53:39. Note that some distortion and tape-sim type plug-ins are level sensitive, but most "normal" plug-ins are not.

    Now, getting audio in and out of a converter is another matter, though again any competent converter should be able to handle a wide range of input levels without compromising the sound quality. For example, look at the specs on the last page of this Lavry converter manual:

    http://lavryengineering.com/pdfs/lavry-da924-manual.pdf

    Even when sending a loud signal at 1 dB below clipping, the distortion is only .0009 percent. Other converters may not be this clean, but most are still very clean right up to the point of gross distortion. I believe that when people reduce levels and notice the sound improved, the problem was occurring somewhere else in the signal chain.

    --Ethan
    I think a big part of the sound changing and sounding more focused was the fact that instead of changing the levels with the fader I changed the trim level. After that i readjusted my plugin chain settings. The real problem probably came from the gain staging through the plugins. Maybe im wrong but if I were hitting the plugins too hard with the signal it was causing some distortion.

    Are you saying that all signals coming into the converters should be showing between -18 to -12 for optimal sound and trimming will do nothing but lower the already recorded signals volume?
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    No, what I'm saying is most plug-ins sound exactly the same no matter what level you throw at them. Now, this is not true for all plug-ins, but it is for most of them. This is very easy to test for yourself, and I urge you to do that. It won't take long, and you'll learn something very important about how your audio software works. See my article and video linked above for more info on how to test this. I promise it's well worth your time. At least if you want to know the how and why aspects of audio.

    --Ethan

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    Quote Originally Posted by offcenter2005 View Post
    When I first started recording digitally i would keep the input levels as close to unity as possible
    What do you mean by "levels as close to unity as possible"? Unity applies to gain, not level. Using terms incorrectly like this could contribute to misunderstandings.

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    Quote Originally Posted by bouldersoundguy View Post
    What do you mean by "levels as close to unity as possible"? Unity applies to gain, not level. Using terms incorrectly like this could contribute to misunderstandings.
    What i meant was when setting the gain of the input and output signal to the mixer-converter-daw, i would keep the meter reading at near unity gain = 0db. Is this not the right terminology? What i meant was input gain.
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    Quote Originally Posted by offcenter2005 View Post
    What i meant was when setting the gain of the input and output signal to the mixer-converter-daw, i would keep the meter reading at near unity gain = 0db. Is this not the right terminology? What i meant was input gain.
    What the meter displays, for example 0dBVU, is the level, not gain. The input preamp adds gain as needed to make the signal the desired level. Where you get a unity gain setting is on the faders where it is the 0dB point. It just means the fader isn't adding or subtracting any gain and the signal passes with its level unchanged.

    I would say you're setting the input gain for a level of 0dBVU on your meter. That probably comes out as +4dBu at the outputs to the converters. Depending on the converters that may end up in the vicinity of -18 to -12dBFS.

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