Dont understand whats going on. The original is obviously clipped full. But no red lines show when opening it.
Audacity isn’t looking at the wave shape. It’s just checking the peak levels and warning of potential clipping. It’s possible to get false-positives and false-negatives. If you have clipping and you lower the level (by entering a negative value in the Amplify effect) you can “hide” the clipping from Audacity.
But if I use the declipping tool, there is infact clipping in the current file.
There are two inter-related settings. “Restoring” clipped peaks is obviously going to make the peaks higher. Try a lower number (a more negative number) for Reduce Amplitude… or run the Amplify or Normalize effect after Clip Fix to bring down the level before exporting. Audacity uses floating-point which has no upper or lower limits so the new restored peaks are going over 0dB, but not really clipped yet.
…When I tried Clip Fix on CD rip once or twice it made the waveform look better but it didn’t make it sound better. Or it still sounded bad.
BTW - If you work with MP3s, MP3 changes the wave shape making some peaks higher and some lower. So if you export as MP3 it might “show red” again when you re-import. MP3 can also go over 0dB so those new-higher peaks aren’t really clipped either (ii the file wasn’t clipped before MP3 compression) but you can clip your DAC if you play it a “full digital volume”. For that reason, many people normalize to -1dB or so if they are making MP3s. Personally, I don’t worry about it… MP3 is lossy anyway and I’m not convinced that the slight clipping is audible.
And back around to the point, why doesn’t it show clipping in the original? Is it hard limited?
At 16-bits you can have values between +32,767 and −32,768 and those values represent 0dB. Just one count shy (a tiny fraction of a dB lower) and It’s probably not going to show red even if the waveform is badly clipped.
Digital clipping happens at exactly 0dB so if you see red immediately after recording (before you change anything) you clipped the analog-to-digital converter during recording and Audacity is telling you “the truth”. Unless you were extremely-extremely lucky and your peaks happen to exactly hit 0dB. Once you (or somebody) starts changing things it’s just showing possible or potential clipping. And of course, you can overdrive and clip a preamp or power amp and Audacity will never know if there is analog clipping.
These are supposed to be quality FLAC files, it don’t look it to me.
FLAC is a lossless format but it may not have been made from a “high quality” recording or a high quality production. Or if it’s a [u]Loudness War[/u] production it’s intentional and the mastering engineer may consider it “high quality” even with some clipping.
Almost all commercial recordings have some limiting & compression.
If it sounds OK, try not to worry about it. Unless you made the recording yourself, and in that case leave some headroom next time. It really doesn’t look THAT bad zoomed-in. Usually we want to completely avoid clipping because it is distortion but a couple of dB of occasional clipping usually isn’t audible.
It’s not good spending a lot of time jammed right up against 100% or 0dB. It is recommended that the tips and peaks of your live voice recordings never go much louder than -6dB to -10dB.
The submitted standard for ACX Audiobook is -3dB peaks. If you go over that, ACX may reject your file. The Audacity mastering tools shoot for -3.5dB, slightly quieter, so there is a little slop room in the conversion from your WAV Edit Master to MP3.
Conversions aren’t perfect and if your goal is almost clipping, the conversion error may throw you over the edge. Over the edge distortion may be audible. Run away.
A note about Clip Fix. Its goal isn’t to repair the show. Its goal is to use the waves before and after the clip to guess what the original wave might have been if it didn’t get damaged. It has no idea what the content was. That’s also why it can’t work on shows with massive damage. If the waves before and after the clip are clipped, you’re stuck. That puts you in level two:
Lots to digest there, thank you much the both of you.
To clarify a little more. These are not my music files. If I was creating them, no way I’d leave it at near 0db full pegged. Its obvious data above 0db is lost.
So in a situation where you are taking 16 or 24 bit FLAC files and at minimum, just de-clipping them - what direction would you go?
Clip than normalize sound / export at best quality and call it good?
I’m looking at having to go through hundreds of existing FLAC files in my library. In my theature, it is night and day, vs just raw dogged right at near 0db for the entire track - as the originals came.