Limiters and oversampling

[vortexsupernova]

dBTP = decibel true peak
ISP = another name for true peak

So I'm one of the self taught laymen out here :p I never took a course in making music digitally or watched extremely comprehensive tutorials... Can someone explain this to me in some simple words,

• Digital limiters without oversampling (or other ways of detect ISP) will react to the digital waveform, not the true waveform, and thus won’t react as a limiter should react. This does not mean it will sound bad in any way, just that it doesn’t react as intended.
• -Tip: If you want to work with limiters that lack oversampling; up-sample your 44.1 or 48KHz projects to a higher sample rate and by that have the limiter react more accurately. <- is this possible in LMMS?
• We’ll most probably have a loudness standard for streaming media within a couple of years and that standard will be ISP aware and set at -1dBTP. This means that if your audio peak at +2dBTP it will be turned down by at least -3dB before reaching the end user. All that extra loudness you gained by letting random equipment create a positive peak (that might distort) will be lost.
• Encoding to lossy formats; ISP will make it harder for encoders to do a good job. If you don’t have the tools or knowledge to check how your audio will perform post-encoding I would recommend to stay away from positive True Peaks.
• It’s not recommended to use any of the limiters with a positive score in 44.1KHz projects. They should handle ISP better as your sample rate goes up. (The list is on the website article I linked below)

The article -> https://www.saintpid.se/en/isp-true-peak-limiters-test/

*I have extreme frustration in math

Why does it have to be oversampled when using a limiter? (is there a setting to change the audio settings when editing the project file (not the exported file) in 44.1 kHz to other rates?)
And the bit depth, what does it do? I always see 24 bit for music production (is there a setting in lmms tho?)

I only see the setting in changing the bit depth and the sample rate upon export and not on while editing the song

I do not know what is Nyquist rate, the other maths in music production and what do they do or why they are important, might probably quit this hobby lmao bruh

[musikbear]

Can someone explain this to me in some simple words,

Maybe LostRobot. Try to reach out to him on Discord

[Jaknife]

If you're still wondering, Put the Hard Limiter effect and grab a amplifier and put it at -1db.

When you convert your file to mp3 128kbps (like most websites do) the threshold kind of disappears, filling the rest of the waveform.

If you're going to export it as a WAV (default) then leave it at 44.1 Khz because 48 Khz changes the quality slighlty (in a bad way), bit depth at the highest (24 bit).

I could go on and on but I'm oversimplifying this to make it not hard on yourself.

[vortexsupernova]

If you're still wondering, Put the Hard Limiter effect and grab a amplifier and put it at -1db.

When you convert your file to mp3 128kbps (like most websites do) the threshold kind of disappears, filling the rest of the waveform.

If you're going to export it as a WAV (default) then leave it at 44.1 Khz because 48 Khz changes the quality slighlty (in a bad way), bit depth at the highest (24 bit).

I could go on and on but I'm oversimplifying this to make it not hard on yourself.

Thank you, very helpful, I did not know its hard limiter is good

[brandystarbrite]

If you're still wondering, Put the Hard Limiter effect and grab a amplifier and put it at -1db.

When you convert your file to mp3 128kbps (like most websites do) the threshold kind of disappears, filling the rest of the waveform.

If you're going to export it as a WAV (default) then leave it at 44.1 Khz because 48 Khz changes the quality slighlty (in a bad way), bit depth at the highest (24 bit).

I could go on and on but I'm oversimplifying this to make it not hard on yourself.

Nice info Jaknife.