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Thread: What DB should I be peaking at before mixing?

  1. #11
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    All great stuff, however no-one seems to have mentioned one very important (IME) point; frequencies below ~500 hz are going to push peak and lufs numbers up with the result that the whole mix and individual tracks will seem quiet to the human ear even when meters show red.

    So try this: put a low shelf on one track. Set it with the breakpoint in the 200-500 hz and watch the meter as you slowly bring down the low end. You should see the peak and/or lufs drop significantly. Don't leave it like that, it's just a tool to find out how much dynamic range is being taken up by the low frequency content of that track(the difference between eq'd reading and non eq'd reading). You will find you can increase the volume of the track now without creating overs on the main mix.

    This is one of the reasons that a commercial release with a lot of dynamic range can sound both louder and clearer. The low frequencies can both add "mud" or loss of clarity and reduce the total apparent volume by taking up much of the dynamic range with unneeded low frequency content.

    So start just as you have been doing-get a relative mix. Then start getting rid of un needed low end and compensating accordingly with the relative volumes as you go. You will end up having more volume and clarity on the main mix.
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  3. #12
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    Quote Originally Posted by Gtoboy View Post
    All great stuff, however no-one seems to have mentioned one very important (IME) point; frequencies below ~500 hz are going to push peak and lufs numbers up with the result that the whole mix and individual tracks will seem quiet to the human ear even when meters show red.

    So try this: put a low shelf on one track. Set it with the breakpoint in the 200-500 hz and watch the meter as you slowly bring down the low end. You should see the peak and/or lufs drop significantly. Don't leave it like that, it's just a tool to find out how much dynamic range is being taken up by the low frequency content of that track(the difference between eq'd reading and non eq'd reading). You will find you can increase the volume of the track now without creating overs on the main mix.

    This is one of the reasons that a commercial release with a lot of dynamic range can sound both louder and clearer. The low frequencies can both add "mud" or loss of clarity and reduce the total apparent volume by taking up much of the dynamic range with unneeded low frequency content.

    So start just as you have been doing-get a relative mix. Then start getting rid of un needed low end and compensating accordingly with the relative volumes as you go. You will end up having more volume and clarity on the main mix.
    Exactly, My point was that if the OP is not doing any subtractive EQing in the low end and low mids the meters are going to be booming. Putting a limiter on the mix buss is not the solution IMHO. I can't speak for other engineers I only put a console simulator on the mix buss.

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    You are confusing 'volume' with 'headroom'. Volume is a subjective effect of human hearing, and is produced in audio equipment by selecting higher or lower voltage levels, typically through (mostly) passive resistive channel 'faders'. Just to confuse matters still further 'volume' is not 'gain': the latter usually refers to the first stage of your mixer or chosen recording interface, it uses 'active' electronics and is the amount of electronic 'boost' you have to dial in to bring your mic (or whatever) into the voltage range your interface needs to operate properly. 'Headroom' is a somewhat ambiguous reference either to the 'blank space' you have in hand above your maximum recorded level before your mixer / gear hits distortion OR (and IMO more usefully) the difference between your wanted recorded sound and the unwanted background noise that is always present in any electronic equipment, including audio kit. There shouldn't be a problem when you get around to applying compression, EQ, etc. With analogue kit you just turn down the level ('volume') of the mix or discrete track you're sending to your compressor (or whatever), and with digital equipment you just 'de-amplify' i.e. fade that source as much as is necessary so that it comes back from your fx processor without hitting 'clip'. Generally it is better to start with a 'hot' signal at the first stage of the recording process, i.e. what you're sending to he mixer. You can always turn down the volume (voltage) of a 'hot' source so as not to 'bust' the next stage of processing. But if you record with an unnecessarily 'cold' i.e. weak signal, you are going to get proportionately more electronic or natural background noise in your finished product, which you will never be able to get rid of. True, all decent recording software now includes remarkably good 'noise reduction' options - as long as the noise is an unvarying constant, unlike human chit-chat or passing traffic - but it is still better to have 'swamped' the noise in the first place. Finally, in most analogue mixers at least, audible 'clipping' will not occur if your red light winks occasionally; like car speedometers, the manufacturer is 'playing safe'. In professional equipment, the nominal 'full deflection' signal is +4dBu, that's about 1.2 volts, (.775 volts for consumer devices) but in practice can peak much higher before anything nasty happens; in fact in the days of reel to reel tape we deliberately ran as 'hot' as we could get to feed the tape: the result sounded 'juicier', although you'd probably NOT want that in recording Classical music

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