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1
3rd Party Plugins - (fb2k) / Re: Biography Discussion
Last post by pIv -
I only comment this line:
// window.DlgCode = 0x004;
Script work.
In version 2.2.0 of foo_jscript_panel marc2k3 remove window.DlgCode.

P.S. marc2k3 propose a solution:
window.DlgCode = 0x004;
replace to
if ('DlgCode' in window) { window.DlgCode = 0x004; }
I check it - script work.
2
General Audio / Re: AM distortion from soundcard at frequencies above 12kHz
Last post by Radio Pushka -
I figured out a way to solve this problem. My FM radiostation gets very distorted when an instrument plays above 12 khz on a couple of songs. I was running it through the sound blaster 3, just fresh from the store. I have linux, and I use the qmmp player. After playing with it for while, I installed all of the ladspa plugins I could find. so I found this Mag’s Notch Filter, i enabled it and set it to 12 khz and It fixed the problem for me. check out the screen shots. I hope this helps.
4
Validated News / Re: Chromecast Audio
Last post by andy o -
There are a couple of car adapters that have been announced at CES, which if pretty much the same electronics were put in a home-friendly form factor, could replace the Chromecast Audio and add Asssistant functionality to boot. Amazon already sells a device like that, but you've gotta be into the Alexa ecosystem, and I don't think you can Google Cast to it. Also I'm gonna go on a limb here and predict that Google is not going to give these car Assistant adapters the ability to multi-room, so hopefully instead of having to modify one of these for home use, Google comes up with a proper Home product with an aux output.
5
3rd Party Plugins - (fb2k) / Re: foo_sid
Last post by Whosondephone -
May I make a feature request?  An option to double the song length value specified by the database text file.  Silence detection and fade outs after the loops would be nice too, but you've done enough for us already.  Thanks again for the great plugin!
6
3rd Party Plugins - (fb2k) / Re: foo_discogs
Last post by zoomorph -
alec.tron,

There's no way to reorder the tag list but you can delete a tag and add it at the bottom.

If you want to clear a "temp" tag, you can define the same tag again later on with an empty tag formatting string. That will delete it.

I accomplished what you were asking for with the following tag formatting strings:

TEMP=$pput(feat,$filter($flatten($multi_if($any($multi_strcmp($sextend(%<<TRACK_CREDITS_SHORT_ROLES>>%,%<<RELEASE_CREDITS_SHORT_ROLES>>%),'Featuring')),$multi_if($put(aj,$sextend(%<<TRACK_CREDITS_ARTISTS_JOIN>>%,%<<RELEASE_CREDITS_ARTISTS_JOIN>>%)),$joinnames($put(an,$sextend(%<<TRACK_CREDITS_ARTISTS_NAME>>%,%<<RELEASE_CREDITS_ARTISTS_NAME>>%)),$get(aj)),$get(an)),)),))

and

TITLE=%TRACK_TITLE%$if($any($pget(feat)),' (Ft. '$join($pget(feat))')',)

I encountered a couple bugs on the way which were fixed in 2.19!
7
3rd Party Plugins - (fb2k) / Re: foo_discogs
Last post by zoomorph -
Version 2.19:
- Fix $filter() function not removing empty strings from array.
- Fix $pput() and $pget() to operate per track as expected, rather than per release.
- Fix filtering on expanding master release on find release dialog.
8
Support - (fb2k) / Re: NO benefits from using ASIO ?
Last post by Teddy_the_barber -
Regarding direct sound and "god knows what's going on in its inner workings".
Well, not much and definitely nothing top secret. The only things that can alter your audio stream are: 1. Resampler (does nothing when the rate is matched and if not, is transparent anyway), 2. Limiter ( does nothing if the signal is below -0.13dbFS) - has a pretty fast release time and is of high quality. Can be audible for really "hot" stuff, but is easily bypassed by simply reducing volume in foobar. 3. Dither (irrelevant, as it is easily disabled by choosing 24 bit output in audio settings). That's it, god knows and now you know.)))
10
Support - (fb2k) / Re: NO benefits from using ASIO ?
Last post by Teddy_the_barber -
In Xonar family sound cards all audio output via WASAPI(regardless of the mode used) or DS is passed through Asus driver panel, which includes its own resampler. However, if the rate of the audio stream matches the rate chosen in asus panel, you'll get bit-perfect playback. If your library contains files with different sample rates and resampling everything to a one fixed rate with smth like SoX is not an option because of the unbearable noise at ~ -170 dbFS, then using asio makes perfect sense. It's the only output mode in which asus panel and thus the built in resampler are completely bypassed.
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