LMMS • Forums

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

vortexsupernova wrote: ↑

Mon Apr 13, 2020 7:03 am

Can someone explain this to me in some simple words,

Maybe LostRobot. Try to reach out to him on Discord

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.

Jaknife wrote: ↑

Wed Apr 29, 2020 1:02 am

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

Jaknife wrote: ↑

Wed Apr 29, 2020 1:02 am

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.