How can I render an wav to mp3 in reaper without clipping?

[Audio~Geek]

also inter-sample clipping.

[manducator]

also inter-sample clipping.

Yeah, but can you do something about it?

[faun2500]

Inter sample peaks. Some limiters have a setting to detect inter sample peaks. SSL have a free little plugin to monitor them as does still well iirc.


I also have these red lights from imported mp3.

[mfb]

Hello,

I have a wav file with loduest peaks @ -0.3 dBFS.

When I try to convert it to mp3 (with EZ CD Audio converter), I got clippings up to +0.7 dBFS.

Rednering in reaper with a limiter doesn't help eiterh. How can I render a wav to mp3 without any clippings?

The wav and the mp3 are based on the same digital samples (assuming wav is 44100 Hz, 16-bit). So I don't think this can happen converting a wav to an mp3. If you set Reaper to *output* a wav, ie. bounce your original wav to a new wav file (as opposed to an mp3) then the output wav will likely also be clipping.

I think what is likely happening is this. You have some original wav file, with peaks at -0.3 dBFS. You drop it into a track on Reaper. Reaper, being a DAW, now lets you modify this wav in all kinds of ways. In particular, you will have a track fader and a master fader, both of which will likely go above 0db. If all the relevant track & bus faders
in Reaper are set to 0db (and you're not adding any effects), then Reaper's mixdown should also only go to -0.3 dBFS (I think numerical rounding errors should be <0.1 dBFS).

It might just be easier to use some audio editor e.g., Audacity or Audition to just convert the original wav to an mp3 rather than running it through the complexity of Reaper.

I suspect that its not actually possible for an mp3 to encode values that, when properly decoded, translate to >0 dBFS. In other words, the problem is almost certainly not in the mp3 encoding but rather in the signal you are *feeding* the mp3 encoder, which is likely some form of bus output from Reaper.

[manducator]

mfb,

Maybe you'r right, I used Poikosoft's EZ CD audio converter and it also gives clipping but less severe; only +0.1 dBFS.

Thanks for the explanation.

[wesleyt]

I regularly see inter-sample peaks when going from 24-bit aiff or CAF to 16-bit 96 kbps m4a (I'm usually exporting a podcast) when my peaks are -0.3 to -0.1. I have successfully gotten rid of them by using Ozone to dither down to 16-bit with ISP protection before exporting to m4a.

I sometimes use the "Mastered for iTunes" script to go from aiff to m4a, which also seems to minimize ISPs. It's fine for the podcast.

If the ISP peaks are just +0.3, I sometimes just re-normalize to -0.4 and re-export. I won't do that for +0.7; that affects RMS too much for me.

I recently read you can use a limiter with 8x over sampling to effectively squash ISPs. I've not tried this, but it makes sense to me.

I see lots of options.