Skip to main content

Recent Posts

3rd Party Plugins - (fb2k) / Re: OpenMPT Module Decoder
Last post by mudlord -
How accurate is this compared to say bubsy's shenanigans when it comes to XM/S3M/MOD?

Scroll down that page and change the language from "Deutsch [Du]" to "English (US)".

Regards, ...

Thanks for the suggestions.
I tried to register at but my lack of German, and Google's poor translation made it impossible.

I couldn't work out what answer was expected from the CAPTCHA, and now it tells me I have to wait.
CD Hardware/Software / Re: Cuesheet Processor (CueProc)
Last post by dpr -
where can i get cueproc 1.10 from? is not working.


Looks like a command line tool that has completely disappeared off the web.  The Wayback Machine has no archived copies of this but the site itself is archived.

There's plenty of applications that can parse and do what cueproc does all with a nicer, easier to use GUI.

Thanks. Can food are process a directory tree of cuefiles?

I meant foo as in foobar2000
Sorry for triple-posting: Any plans to add UTF-8 log writing?

Thanks I love this program. I use it a lot to process many rips.
Opus / Re: Looking at Opus for MP3 replacement and have questions
Last post by Klimis -

It's not like mp3 the last 20 years was a dead format that received no quality improvements. We cannot compare the 2000s mp3 encoders at 128kbps with Soundcloud's Lame-based 128kbps encoder that is up to date.

Lame 3.93 came out in 2002, and while there was some improvements over the next 4 or 5 years, 2000s mp3 encoders were not that different from the ones we have today.
Still they is no comparison with any generic mp3 encoders of that time of most applications that cared little about quality at lower bitrates (like most music library programs that had a rip to mp3 function). I clearly remember struggling to encode something below 192kbps without the sound starting to get distracting on my tiny storage space phone. Most generic encoders of that time had a nose dive in sound quality below 192kbps. Lame offered unparalleled audio quality and still does for what mp3 is capable of and it still managed to pull the boundaries atleast for a slight bit the last decade or so. I wish I knew about it then, but then I wish I knew that my phone would play AAC-LC and I knew the technical advantages, but even more I wish there was opus around, it would have saved me hours of exchanging tracks on my phone's storage depending on my mood for music for the day.
Got a somewhat good reception on the ABBA chat group on these examples.  If you listen to the files whose names end in 'Clean', there is some very good processing going on -- very clear sound.   The Dreamworld example is amazing -- apparently it got compressed several times in succession, practically destroyed.  The result is quite listenable, but does show damage.
I would be doing more 'scientific' comparisons, but the differences & improvements (& problems) are so extremely obvious, that fine grained audio measurements aren't really needed.  Also, the gross level noise reduction can be seen with a spectrogram, even though the exact amount of NR isn't really measurable there.
These examples are disappearing in a few days, but the audio processor (the 'restoration processor') is now frozen, and I am repackaging the software moving the software into its own binary.   The psuedo-DolbyA has been frozen for a few weeks.  Binaries will be available in a week or so, and source code in a couple of weeks after that.  The VERY GOOD news is that the source code/function is now frozen!!!

Here are where the examples are -- again, the filenames with the 'clarification' have 'Clean' in them.  Some normally processed stuff is in there also.  Nothing in there is 'boring.'    There is a Carpenters example that is super clean also:
Support - (fb2k) / Re: Sample rate switching causes audio cutoffs...
Last post by Rollin -
I first thank it comes from my soundcard (Focusrite Clarett 8Pre), but if I use Reaper or Audacity instead (in ASIO mode as well), these cutoffs do not occur
Are you sure that samplerate switching really happens in your sound card when you use Reaper or Audacity? Can you see real current samplerate of soundcard somewhere?

You can try foo_dsp_pregap to mitigate problem.
Support - (fb2k) / Re: Sample rate switching causes audio cutoffs...
Last post by kode54 -
It may be possible to switch sample rates and formats with ASIO quickly, but I think those apps are simply picking one high rate and resampling to it. Try adding a resampler to your DSP chain, picking a target rate like 48000Hz, or 96000Hz if you really want to please the neighborhood bats and possibly destroy your tweeters.
General - (fb2k) / Re: Anyone Have a Layout To Share?
Last post by kode54 -
@MinorKey, post updated, please refresh the page.
Thank you, I grabbed the bin files, but I still can't see your user component list. :-(

I have attached the components list, since it is apparent that you cannot see imgur links.