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Thank you so much!  That worked perfectly.  I should RTFM more closely.   :B
The function $meta() bypasses all mappings. Use $meta(album artist).
Hey all,
I use multiple players (both hardware and software) in different situations with a large library and have spent a great deal of time conforming the tags in my entire collection to the several possible requirements of those players.

Along the way, I realised that %album artist% (not %albumartist%) falls back to displaying %artist% if the tag is not present.

The inability to display an empty %album artist% makes it impossible to tell if the tag is actually present or not except under specific circumstances (like when %artist% and %album artist% both exist and are actually different from each other).

Does anyone know of a way to force %album artist% NOT to fallback to %artist%? 

are you planning to expand your script like wrote in post #18 ?
Scientific Discussion / Re: Audibility of phase shifts and time delays
Last post by danadam -
And while I revived that old thread... the formula from the first page:
Code: [Select]
1/ ( 2 pi bandwidth number_of_levels)
I wanted to make sure I got it right. It is a bandwidth, not a sampling rate? And a number of levels of concrete signal, meaning that timing precision of a signal depends on its level? So for full scale 44/16 signal it is:
Code: [Select]
1 / ( 2 * pi * 22050 * 2^16)
Is that correct?

Also, is there any "citable" online source for this? (not that I don't believe it :-) )
3rd Party Plugins - (fb2k) / Re: foo_discogs
Last post by zoomorph -
a script is needed so that if there are 2 or more styles indicated in the release (on the release page), then they would be written in one line, separated by a comma (or via /), and not a semicolon in different fields of the tag.
Try adding the $join() function to the formatting string for that tag.
Scientific Discussion / Re: Audibility of phase shifts and time delays
Last post by danadam -
I downloaded those files already a year ago, so I can't promise I'll try 3 samples delay any time soon. Probably in another year ;-)
Code: [Select]
foo_abx 2.0.2 report
foobar2000 v1.3.10
2017-11-22 02:35:21

File A: Impulses shift 0 samples 2klp norm 4416 .flac
SHA1: 8fc00a4bb6a1bb0a66ec5c83cfaa36f9d8fddd13
File B: Impulses shift 4 samples 2klp norm 4416 .flac
SHA1: 6133aaa124c97a3f768f3d9216af2eb07b7c0bf3

Crossfading: NO

02:35:21 : Test started.
02:38:48 : 01/01
02:39:48 : 02/02
02:40:31 : 03/03
02:42:31 : 04/04
02:44:00 : 05/05
02:44:54 : 06/06
02:47:26 : 07/07
02:49:35 : 08/08
02:56:36 : 09/09
02:57:49 : 10/10
02:59:46 : 11/11
03:01:23 : 12/12
03:03:16 : 13/13
03:04:04 : 14/14
03:07:30 : 15/15
03:07:30 : Test finished.

Total: 15/15
Probability that you were guessing: 0.0%

 -- signature --
General Audio / Re: Merge two audio files (ADX extension)
Last post by kode54 -
ADX is a lossy ADPCM format, and decodes to 16 bits per sample. No need to use dithering when mixing them, but I would advise using whatever lossless format you want as an intermediate format, and possibly use something like VGAudio to encode it to either ADX again, or HCA.

If you turn off looping in VGMStream before converting them to WAV, mix them, then encode the resulting mix to ADX or HCA with the correct loop offsets, you can have a looping track with minimal fuss. Or you could look up on how to tag an Ogg Vorbis track with loop info, and rename it to .logg for VGMStream.
General Audio / Re: Merge two audio files (ADX extension)
Last post by DVDdoug -
What format do you usually listen to?

FLAC is often the best format.  It's lossless, compressed to almost half the size as WAV, and tagging (metadata) is better supported than WAV.   The only downside is that not every computer will play it without installing a CODEC.   You may want to save a FLAC archive even if you want to listen to an MP3 version.

MP3 (or AAC) can often be transparent (sound identical to the original) so there's nothing wrong with choosing it.

Should I use Dither as well?
Dither is for downsampling the bit depth.   But, dither (or the effects of dither) are not audible at 16-bits or more under any reasonable conditions so in reality it doesn't matter one way or the other.

What you should look out for is clipping.   Mixing is done by addition (summation) so you can get clipping.    If you get clipping, or if you're not sure, run Amplify or Normalize after mixing and before exporting.   Audacity uses floating-point internally/temporarily so it can go over 0dB, but normal WAV or FLAC files are hard-limited to 0dB and you can get clipping when you export.    If you have peaks over 0dB, Amplify  will default to whatever negative dB gain (attenuation) that's required to bring the peaks down to 0dB.

General - (fb2k) / Re: New HDD and Mass lossless conversion
Last post by wcs13 -
OK, here's the Bit Compare result : "4 out of 34290 track pairs could not be compared."

How can I find those 4 track pairs in that huge text log ?... What kind of text string am I supposed to search for ?
And what is wrong with them ? Why couldn't they be compared ? @Peter , are you listening ?
foo_bitcompare's log needs some work in order to be really useful, IMHO.

Please help. Thank you.