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No, I don't want to modifiy the files.
Opus / Opus killer @128 kbps
Last post by Steve Forte Rio -
3 years ago one of the users from my forum published the ABX log (russian) which shows that he hear difference between lossless and opus -b 128. As I understand that was opus 1.1 (October 2014)

Now I tried that sample with libopus 1.2.1 (also -b 128) and here's the result:

Spoiler (click to show/hide)

The difference is almost at hearing threshold but is perceptible and sounds like some additional high-frequency noise (Opus sounds with more sharpness than original sample).

The sample: 02_the_riddler_hide_and_seek.flac
Encoded sample: 02_the_riddler_hide_and_seek.opus

As I read in Wiki, Opus recommended bitrate for music is 96—128 kbps, but why it's restricted there to maximum of 128? Maybe it should be raised to ~160 kbps?
mp3gain doesn't write replaygain tags as such.  It modifies the internal gain settings for the audio.  It also, or just, writes tags indicating what changes it made so that they can be undone at a later date.  The tags are not the same as the replaygain tags that have come to be a standard now, and are not likely to be recognised by any decoder you are using.  The (reversibly) modified audio will be played back at the adjusted gain by any decoder.  It is worth noting that the gains from mp3gain are not very accurate compared to modern replaygain tags.
Must you use mp3gain? If not, try foobar2000 (if you're on Windows) as it'll do it for you
Ok, thanks everyone. I think I'll just leave it.

Sometimes I overthink things and seeing this pilot signal I thought that maybe I could get rid of it for the lossless archival version, but as it's almost certain that I cannot hear it then why bother. I'll use the lowpass when making the mp3 though.

@EekWit: you're right, it does, cheers.
General - (fb2k) / stop after queue
Last post by vladkornea -
Under "Playback / Order" there is "Default", "Repeat (playlist)", "Repeat (song)", "Random", "Shuffle (tracks)" , "Shuffle (albums)", "Shuffle (folders)". However, the most useful option to me is missing: "Queue (then stop)".

In the system tray, there are these options: "Stop", "Pause", "Play", "Previous", "Next", "Random", "Preference", "Exit". However, the most useful option to me is missing: "Stop after current".

Foobar2000 has an excellent interface. I hope you will take the opportunity to improve it further. I think there's no denying that these two features would significantly improve foobar2000 for many users. And it takes like what, a day to implement?
Just the tag yes, you would have to go back to that menu after you calculated the RG to apply it if you'd like to edit the file, which by memory now I don't remember if you can apply it to the lossless files, I know for sure you can on lossy files.
AAC - General / Re: Which encoder to use today?
Last post by eahm -
ThaCrip, usually wait until they ask to provide info they may not need because they may confuse instead of helping.

foobar2000 looks automatically for the \encoders folder, why would you made a C:\TEMP or any other folder outside where foobar2000 is aleady pointing at? Just copy the QTFiles/QTFiles64 inside the foobar2000's configuration's \encoders folder or inside the foobar2000's folder directly if it's in portable mode. Also you excluded the 64bit option which is faster, not by much but I prefer it for example and other may too, again, I'm telling you by experience, wait until they ask for the extra step so you can provide detailed info for exactly what they need in that particular case and you eliminate the risk of providing unneeded or misplaced information in advance.
Thanks a lot! That's what I was looking for.

The correct use for it is Right click on the album -> ReplayGain -> Scan selection (Plus DSP) and add a custom Equalizer preset right?
It affects only a tag right? or the FLAC itself?
Since it's inaudible,  I'd just leave it.    Unless you wanted to try some general noise reduction for the tape FM/tape hiss, which should kill that 19kHz tone too, since it would be included in your noise fingerprint.

A notch filter would generally give you more reduction, but at -87dB and 16-bits, if you can knock it down by 10 dB you'll be down to dead-digital silence.    So a low-pass filter is probably equally as good.

but thought that using a resampler like sox...
Down-sampling requires   low-pass filtering so there's no advantage to resampling and  downsamping (except for a smaller file).    If you could downsample without low-pass filtering (which no resampler will do) the 19kHz "signal" would simply be "aliased-down"  to a lower frequency.