Hello all!

I had a question for exclusively dance producers, mixers, master folk who could possibly be mastering for tracks that sit on Traxsource. What is a good RMS for a dance track?


I've done much reading and research and I find I like to top off at -6rms and I always leave -.02 headroom. Reason being, I've heard that many dj's will have plugins and such on their software when they are mixing that could interfere with the intended volume of the original track (limiter, compressor etc.).

Any feedback on this would be appreciated!

Cheers,
-T

 

RMS does not matter. When peaking at 0DB FS in your daw, -10 LUFS for beatport/trax track, -16 to -14 LUFS when delivering the mix for online streaming (iTunes, Spotify, Soundcloud).

You can target your original mix towards -16 LUFS, is fairly easy. Then two separate masters, one for download one for streaming.

PS: For some audio codecs peaking at -0.2 dB FS is not low enough to prevent clipping. You may need to go as low as -0.7 or even -1.5 dB FS

 

I usually peak at -2 LUFS

 

see https://audiosex.pro/threads/loudness-on-different-online-platforms.25671/#post-198087

 

Yea RMS is pretty much dead, get used to the loudness units.

 

Last edited by a moderator: Jun 14, 2017

Like they said, RMS isn't telling the whole story and you need more measurements to get your loudness right. LuFS is the most important, but some less used ones are even more useful and reliable for streaming mastering : PLR measurement for example.

In addition to your RMS meter, you should use an LUFS, LKFS meter, and maybe a PLR or similar one too. (Nugen MasterCheck)

 

Last edited: Jun 15, 2017

I noticed that for mp3 I need to peak no more than -1.5 or something. It always gets distorted otherwise. I usually do that and then normalize the mp3. Am I doing it wrong?
EDIT I'm talking about peak level, not loudness

 

I don't see how you can normalize an mp3 file since is compressed - any process in any software applied to a compressed format does a decompression to wav and a recompression to mp3/aac/etc

Normalization is the process of finding the highest peak in your song (let's say for the entire song, there is one peak at minute 2 which is the loudest and is - 0.8 dB) and adjusting the overall gain of the song so that peak is at zero (in my example will result in 0.8 dB boost).

So if you have a wave that you exported at -1.5 dB FS then you load that wave into a convertor and convert it to mp3 with normalization box checked, the convertor will basically normalize your wave to 0 dB FS then convert it to mp3. Kind of defeats the purpose of using a smaller peak output in the first place. I would recommend to output from your daw a song peaking at -3 dB FS, and the same song at 0, then convert both in mp3 with and without normalization and see how it goes.

 

If I convert a wav file that has peaks over -1.5 dB (generally), the resulting mp3 clips a lot. That's why I export directly an .mp3 that has the loudest peak at around -1.5, and then normalize it.

 

How do you do that? name the steps, the program(s) you use?

 

When track is almost finished, I put limiter/maximizer as last. I export with a ceiling of -0.2 dB usually, but it depends on the track I guess, but since we're talking about dance music, it's usually limited hard and it's pretty loud. (I'm no expert at mix/mastering btw). Then I save it: this is the .wav file. For the mp3, I just lower the ceiling and export as mp3. Now I have an mp3 that doesn't go above -1.5 dB circa. I import the mp3 and since there's still some headroom I normalize. In FL Studio it's just right click on the sample -> normalize. After this I export the mp3 too and I have both .wav and an .mp3 that doesn't distort or clip anything

 

Most of you are way off!
EBU R128 was agreed on by all Radio and TV stations. The internet is a bit behind, but still I would recommend to get used to EBU R128. It tells us that we have to aim for -23 LUFS (+/- 1). Also, we are not allowed to exceed -1 dBTP (dB relative to True Peak). Additionally some stations will have other requests, like a stronger dbTP limit, or certain values in M and S.

If you're sure you will never be featured on any professional media, you can ignore this. Else, better get used to it. (I'm having a hard time to get used to it, but hey, it helps the music so much!)

EDIT: Just in case, someone is looking for a professional, yet free, LUFS meter, this is by far the best you get for free:
http://youlean.co/youlean-loudness-meter/

 

Last edited: Jun 14, 2017

Why your EDM will sound better with more dynamics – and 6 places to find them:
http://productionadvice.co.uk/edm-dynamics/

 

Now that's strange (the fact that you have differences). As soon as you bring the mp3 into your daw, it gets converted to a temporary wav, then that wav gets normalized, then gets recompressed to mp3.

 

Well, as you can see, the streming services are aligning to more edgy -16 to -14 LUFS. And in Youtube's case, a song gets the platform loudness normalization only after a certain number of plays. The loudness for CD is still -10 to -8 LUFS so it makes sense to work with what we have in practice, not what we have in theory.

http://productionadvice.co.uk/spotify-reduced-loudness/

 

IM A FUCKING IDIOT !!
I should export the WAV at -1.5, and then convert it to mp3 and normalize it...
how did i miss this
thanks

(and sorry for off topic all)

 

-- WAVE > Normalization > MP3 conversion
If the obove solution results in intersample peaks
-- WAVE > max peak level -1,5 dB FS > no normalization > MP3 conversion

 

Ive been using this since I read about it on this site.

About what the codecs add as far as levels? Try the codec preview in ozone 7. You can preview what the streaming codecs will add, and even solo the artifacts. Then you can adjust your levels before hand, so they dont re-encode your music with clipping or overs.

 

Last edited: Jun 15, 2017

Maybe I don't understand but this doesn't make much sense to me...
1. You should keep your peaks at -1.5 in your DAW using your (possibly true peak) limiter
2. Simply export your music in mp3 (or other lossy) format - done
3. I don't know of any program that natively normalizes mp3s

Edit: I would avoid normalization - it is a destructive process, it will change your file and it is completely unnecessary in your case

 

Last edited: Jun 15, 2017

You can theoretically do this losslessly with MP3 since MP3 has per-frame gain. (see mp3gain)