does max volume in mastering have more to do with compression and limiting in mixing?

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I hope this doesn't sound like im not understanding but the better the mix is to begin with then there will be less work to be done when mastering? A great mix and more attention to individual tracking will essentially lead to a louder overall sound when mastered? And that has to do with knowing where and when to compress, eq, limit and instrument placement in the stereo field? Is there any videos on the web i could check out to get a basic understanding of these factors? Please excuse my ignorance but i have a good ear when it comes to tracking and getting decent sounds but its just getting things to meld and sit where they should.

It all start with what's in front of the mic and capturing a great performance.

From there, everything builds on what's before it.
performance/production -> recording -> mixing -> mastering

Loudness should not be the goal. The goal is to connect.

Here are a bunch of video's to check out: ThisWeekIn |  Pensado’s Place
..and mastering podcast: Square Cad: The Mastering Podcast - Latest Episode
ThisWeekIn |  Pensado’s Place
 
A great mix and more attention to individual tracking will essentially lead to a louder overall sound when mastered?
Piggybacking on Tom's post -- A monkey with a limiter can make a mix "loud" -- How that mix holds up is up to everything that came before. Loudness during the mastering phase is more or less an afterthought (although a lot of people seem to think it's the entire purpose). Don't get me wrong --- Taking advantage of a mix that's capable of that sort of volume is certainly considered. We don't rack up gear that has absolutely obscene amounts of usable headroom just to look at it. But the vast (VAST) majority of mixes out there don't have the potential to handle that sort of volume without some fairly nasty damage. Concentrating on volume is just putting the cart in front of the horse.
 
Look, I understand the whole anyone can smash a track to achieve loudness ! I'm not feeding into the volume wars issue. I'm just trying to understand what helps to achieve a good sounding master. I want clarity and volume. You can't show up to build a house and show someone all the awesome sheds you have built. I'm on the right track I think the best thing to do now is keep practicing . Thanks for the input I really appreciate it.
 
Hi Guys,

On the subject; more a technical question for final output:

What dB should you limit to?
Is -0.1dB too high? I've read conflicting things, somewhere said as low as -3.0dB, to counter the various boostings that playback hardware/software has built in?

I've opened up professional reference tracks and they seem to go to 0.1dB or even 0.0. Did a test output at -0.1dB and it's fine at home, but slightly distorted at work (possible soundcard boost or something?).

Confused!
Cheers
Judd
 
There will be places in the signal where a negative peak of one frequency lines up with a positive peak of another frequency so that the actual signal peak level is lowered at that spot. If you filter out one of the frequencies the other will peak higher.

Is there a way to know ahead which +/- peaks will "line up", so that when cutting back one frequency, you can anticipate which other frequencies will be bumped up...or is that info tied to the inner design of the EQ and not easy to learn/know looking the front panel of the EQ?
I understand it will be a nearby frequency...but is there a more specific way of know that if say, rolling off at 175Hz...there will be a bump up at XXXHz?

I'm just considering if it would be more advantageous to then just lower that "bumped up" frequency to compensate for the level rise, rather than lowering the overall level...or does that become a domino effect and with each lowering of one frequency, anther bumps up?
Seems like the LF is where it's most noticeable (the level rise effect), I guess because of the greater energy in the LF, whereas higher up, even if it's happening, you don't see/hear that level bump....???
 
Is there a way to know ahead which +/- peaks will "line up", so that when cutting back one frequency, you can anticipate which other frequencies will be bumped up...or is that info tied to the inner design of the EQ and not easy to learn/know looking the front panel of the EQ?
I understand it will be a nearby frequency...but is there a more specific way of know that if say, rolling off at 175Hz...there will be a bump up at XXXHz?

I'm just considering if it would be more advantageous to then just lower that "bumped up" frequency to compensate for the level rise, rather than lowering the overall level...or does that become a domino effect and with each lowering of one frequency, anther bumps up?
Seems like the LF is where it's most noticeable (the level rise effect), I guess because of the greater energy in the LF, whereas higher up, even if it's happening, you don't see/hear that level bump....???

I don't think there's any way to know for sure. Filtering out one band always has the potential to raise peaks slightly on a complex signal. This is separate from the eq bump mentioned, so all the frequencies not filtered are potential candidates for increased peaks and there's not much remedial filtering you can do. The good news is that it's most likely and most noticeable on signals than have had a limiter applied or been clipped. The usual headroom recommendations leave plenty of space for these peaks, but in mastering you have to be careful that an eq after a limiter doesn't push peaks over 0dBFS.

[Edit] It's also worth pointing out that the increased peaks don't represent audibly increased levels at any frequency. There's no need to compensate with more eq. It's simply a headroom issue.



---------- Post added at 11:05 ---------- Previous post was at 10:50 ----------

Hi Guys,

On the subject; more a technical question for final output:

What dB should you limit to?
Is -0.1dB too high? I've read conflicting things, somewhere said as low as -3.0dB, to counter the various boostings that playback hardware/software has built in?

I've opened up professional reference tracks and they seem to go to 0.1dB or even 0.0. Did a test output at -0.1dB and it's fine at home, but slightly distorted at work (possible soundcard boost or something?).

Confused!
Cheers
Judd

The two figures I hear are -0.1dBFS and -0.3dBFS. The idea is that with a processed signal it's possible that a consecutive pair of samples at 0dBFS can encode a waveform that actually goes above 0dBFS. As I understand it the Redbook standard specifies a DAC that doesn't have any headroom above 0dBFS, so players that meet that standard precisely can clip on some digital signals. Leaving .1dB or .3dB of headroom reduces the chances of that happening. I use -0.3dBFS out of caution and ignorance. Going as low as -3dB seems excessively conservative to me.

The distortion you hear on different systems may be the result of how the volumes are set in your OS or soundcard software. They aren't very precisely marked. It may be that pushing them past a certain level, perhaps 7/10 of the way up, clips the signal. If your speakers have volume control in analog try turning that up and the digital volume down in the OS.
 
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Thanks for the reply Boulder. I did a bit more reading myself; inter-sample clipping, dithering, etc. etc.
Learning a lot! This PDF report is quite interesting level_paper_aes109.pdf, discussing 0dBFS+ in digital systems.

Exported a few tests, between -0.1 and -1.0. The -0.1 is perfect on all systems, except one. On that one, even the -1.0 is slightly distorted when volume is 100%. So I figure there's some filthy boosting going on somewhere!

I wonder how many people are listening on those type of systems? Ah well, that's their problem - everything seems to be -0.1 (or -0.3 as you said).

Interesting stuff!
Cheers
Judd
 
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