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Why You Want to Leave Headroom in Your Mix


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One "professional" mastering engineer ripped me apart for suggesting that people leave some headroom when they mix, because "that doesn't matter any more with digital," and how stupid can I be.

 

Right.

 

Here's why it matters...

 

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Even though the master out (outlined in yellow) is showing signal levels below zero, the True Peak reading (outlined in red) is showing about +4 dB over 0, which is the signal level after reconstruction through the D/A converter. This phenomenon is referred to as intersample distortion. Granted, the signal going on a CD or whatever will be below 0. However, whoever is mixing will be hearing 4 dB of peak clipping, which will give an inaccurate view of the mix (put a clipping plug-in on your master bus, hard-clip the top 4 dB, and then tell me you don't hear a difference).

 

This is why I recommend people leave 6 dB of headroom when mixing, and that's what you give to the mastering engineer. If you want to follow the TP meter readings like a hawk to see if intersample distortion is occurring, by my guest...but it's a helluva easier just to mix with the peaks hitting -6 dB, and let the mastering engineer do the final tweaks. Well, a good mastering engineer, that is.

 

 

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Whooooaaaaaaa.....I've never once heard anyone argue AGAINST leaving headroom for a mastering engineer. SMH.
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He also said that I was wrong about advising artists not to compress/limit the master bus, and leave that kind of processing to the mastering engineer...he said I was stifling their artistic vision.

 

Mastering engineers seem to be binary - either super-kind, helpful, and willing to share their knowledge in any way they can, or prissy, uptight, and with a sense of entitlement. I haven't run into a lot of middle ground :)

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Thankfully, I have only worked with the former, not the latter. This sort of stuff doesn't make any sense. When I hear someone making "absolute" statements ("you should never" or "you should always"), my red lights immediately light up.
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Thankfully, I have only worked with the former, not the latter. This sort of stuff doesn't make any sense. When I hear someone making "absolute" statements ("you should never" or "you should always"), my red lights immediately light up.

 

If I was to summarize the #1 rule for all the true professionals I've worked with regarding audio, it would be "there are no rules."

 

I'm often mystified by people who come into forums and ask "what about doing XYZ" and then wait for people to weigh in...when they could just try it for themselves.

 

 

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Yeah. If there's any rules that audio professionals should abide by, they are likely rules such as: "Have really great taste", "Try and keep the noise level down", and "Get really great performances....or failing that, get really great at editing." :D
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Yeah. If there's any rules that audio professionals should abide by, they are likely rules such as: "Have really great taste", "Try and keep the noise level down", and "Get really great performances....or failing that, get really great at editing." :D

 

That's not a rule, it's just good practice - something that we should try to do as best we can all the time. However, understanding how technology can impose certain limits makes for more predictable results. Take the subject of this discussion, for example. When people figured out that inter-sample peaks could exceed what's represented by 0 dBFS, they figured out how to fix the problem - by re-calculating the peaks before sending them out from the D/A converter. That's not too difficult to do when you have a 24-bit standard as DVDs do.

 

Some newer DVD players incorporate this technology, and others don't. If you really want to add this kind of distortion, you can't be sure that everyone listening won't have it fixed in playback. But then, you don't know what a listener will have between the D/A converter and his ears, so you still can't be sure how he'll hear your carefully crafted mix - so you carefully craft it so that, while it may not sound the same through every playback chain, at least it won't include elements that emphasize the bad features of a listening system.

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Good points. My issue was more that if the person doing the mixing didn't realize that they were hearing signals that were clipping, and basically acting as if a limiter had been put in the signal chain, they wouldn't realize they weren't hearing the mix properly - and might make bad mixing decisions as a result.
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Good points. My issue was more that if the person doing the mixing didn't realize that they were hearing signals that were clipping, and basically acting as if a limiter had been put in the signal chain, they wouldn't realize they weren't hearing the mix properly - and might make bad mixing decisions as a result.

 

The way we typically work these days, the problem is more likely to crop up in the mastering process rather than tracking or mixing. When you're mixing in the box, you're always listening to the output of a D/A converter, so if you've pushed the level up too far and you can recognize the sound of inter-sample overload you can avoid it. But in mastering, sometimes the last thing in the chain is an analog limiter, and it's that signal that gets digitized to create the digital master. If you monitor on the analog side of the A/D converter, assuming your analog monitoring chain has plenty of headroom, you won't hear anything wrong. The time to catch that is when listening to the playback of the digital master, but by then you're probably tired of hearing the songs. ;(

 

There are "true peak" metering plug-ins now that will indicate inter-sample peaks that can exceed the headroom of a D/A converter and warn you to drop the level a bit.

 

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