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2
Support - (fb2k) / Custom tag values separators
Last post by 0x4D657373696168 -
When i try to separate custom tag values by using standard separators (& , and ;) - they don't work. I get one tag with all this symbols.
For example, i'm tagging one file with 'mood' tag and set values 'Calm / Relax; Wild Nature'. And i get that very _one_ tag - 'Calm / Relax; Wild Nature'.



When i set these values in 'genre', for example - all work.
4
General Audio / Re: Why wasn't there ever a VHS based consumer audio format?
Last post by Nikaki -
"Conventional wisdom": even among those with a type IV switch, many couldn't really record to metal tape.
Question: was that just audiophoolery? (I stuck to chrome myself.)
No, type IV was actually superior. There's a demo of noise/hiss levels in the video I posted (starting at 15:25.) The sound is extremely clean. Back then it sounded very near to the CDs I recorded from. And that's without dolby noise reduction. With dolby, there's virtually zero noise. But a poor deck will always produce crappy results though. A decent deck is needed to get good recording and playback quality.
5
Support - (fb2k) / Play command issue
Last post by Nicolayo -
Hi everyone.

I use foobar to play tracks for theatre plays. Usually I select the track from the playlist with keyboard arrows and once the cursor is over the track, I press <Enter> to play it via shortcuts.

A month ago I changed my laptop and downloaded latest version of FB2K. The problem is 'Play' command has a subtle but important change:

  • When status is 'playing':
    IF cursor is over played track: Play command plays the track from beginning.
    ELSE Play command plays the track where the cursor is
  • When status is 'stopped': Play command plays the track where the cursor is
  • When status is 'paused': Play command resume the playback of paused track, no matter where the cursor is. This behavior should be for Play/pause command instead, just like older versions of FB2K.

I've tried different configurations of "Playback follows cursor" and "Cursor follows playback" with no results.

Anyone knows a way to reverse this?
Is this a bug or just a change?
Where can I get an older version?

Im using foobar version 1.3.17 on Windows 10.
Shortcuts involved:
Spacebar: Play/pause
Enter: Play

Thanks!

7
Scientific Discussion / Re: Audio Summing Algorithm
Last post by jsdyson -
I believe @ziemek.z misunderstanding of FFT (i.e. the fact that FFT returns a complex function from a real variable) comes from how audio editors often display FFTs. Many - including Audacity - display it as a sequence of buckets, where frequencies of each bucket are painted in a voiceprint. The tools often quitely ignore the Imaginary part of the resulting function, and simply paint the magnitude of the complex value, and ignore the argument, which in that case is phase.
Okay -- that makes sense.  From what I know (and from my previous practical experience), the phase is all important.  When I wrote my prototype compressor/expander, the math operations were done on complex numbers.  Using magnitudes only was just a waste of time and just produced garbage.

In my recent work on compressors, expanders, Aphex Exciter removers, DolbyA decoders, etc -- when trying to clean up the sound of some old recordings -- I found that they were sometimes screwing with the phase in bad ways and summing/subtracting phase shifted versions of the signal to/from itself -- the results might have given an effect of making middle freqs more intense (in the case of the parameters used in certain devices), but with the super high quality equipment of today -- the ugliness becomes more apparent.  At least, they weren't zeroing the phase, but even playing with the quadrature (the stuff 90deg out of phase) should not be done lightly.   Even though I didn't like the results -- they weren't going to crazy with the phase -- it is just that phase can mess things up simiilarly to messed up freq response.  Doing some phase things (fancy for the time that 4ch matrix was common) is part of how matrix quad could work reasonably well.   IMO, f possible, unless done absolutely carefully -- not a good idea to play with the phase of any aspect of the signal.  There are right ways/good reasons for doing it -- but if questions need to be asked on forums like this, then it is a good idea to avoid doing it for now :-).

John
8
Scientific Discussion / Re: Changing filter's coefficients to compensate sampling rate changes
Last post by jsdyson -
By MathWorks: ( https://www.mathworks.com/help/audio/ref/weightingfilter-class.html?s_tid=gn_loc_drop )

"These coefficients are recomputed for nonstandard sample rates using the algorithm
described in Mansbridge, Stuart, Saoirse Finn, and Joshua D. Reiss. "Implementation
and Evaluation of Autonomous Multi-track Fader Control
." Paper presented at the
132nd Audio Engineering Society Convention, Budapest, Hungary, 2012.
"
(AES Convention Paper 8588)

Looks like the original coefficients are calculated using:

Code: [Select]
% ITU-R BS1770-4 --------------------------------------
fs = 48000;

% HSF
db = 3.999843853973347;
f0 = 1681.974450955533;
Q  = 0.7071752369554196;
K  = tan(pi * f0 / fs);
Vh = power(10.0, db / 20.0);
Vb = power(Vh, 0.4996667741545416);
pa0 = 1.0;
a0 =      1.0 + K / Q + K * K
pb0 =     (Vh + Vb * K / Q + K * K) / a0
pb1 =           2.0 * (K * K -  Vh) / a0
pb2 =     (Vh - Vb * K / Q + K * K) / a0
pa1 =           2.0 * (K * K - 1.0) / a0
pa2 =         (1.0 - K / Q + K * K) / a0

% HPF
f0 = 38.13547087602444;
Q  =  0.5003270373238773;
K  = tan(pi * f0 / fs);
rb0 = 1.0
rb1 = -2.0
rb2 = 1.0
ra0 = 1.0
ra1 = 2.0 * (K * K - 1.0) / (1.0 + K / Q + K * K)
ra2 = (1.0 - K / Q + K * K) / (1.0 + K / Q + K * K)
% ------------------------------------------------------

Could this same code be used to calculate coefficients for other samplerates by just changing the vlue of parameter fs[/]?


I use a similar subroutine for similar purpose in my software -- that is, I just change the fs value -- the filter specs are defined by the  'f0, q values.  I don't use that precise function -- I start with something like this:  H(s) = (b0*s^2 + b1*s + b2) / (a0*s^2 + a1*s+a2);, where i have precalculated the b0-b2 and a0-a2 values for the kind of second order filter that I want.  Then I have a function that accepts the b values, a values, fs, and the frequency for the warping parameter.  So -- no matter the fs frequency -- I have a 2nd order IIR filter that matches the specified characteristics.  Of course, the filter must be reasonable for the fs values that it is used for.   Also, I have a set of functions that build FIR filters are runtime also (given cutoff, filter type, #taps, etc.)

I didn't carefully review your specific function, but it does look somewhat similar to the one that I wrote.  You are going in the right direction.
Joh
9
Scientific Discussion / Re: Audio Summing Algorithm
Last post by polemon -
I believe @ziemek.z misunderstanding of FFT (i.e. the fact that FFT returns a complex function from a real variable) comes from how audio editors often display FFTs. Many - including Audacity - display it as a sequence of buckets, where frequencies of each bucket are painted in a voiceprint. The tools often quitely ignore the Imaginary part of the resulting function, and simply paint the magnitude of the complex value, and ignore the argument, which in that case is phase.
10
General Audio / Re: Why wasn't there ever a VHS based consumer audio format?
Last post by Porcus -
Compact cassettes ... ahwell ...

And almost no one I knew was buying metal grade cassettes anyway, only normal and chrome.
Many tape recorders didn't have chrome/metal settings. According to the Wikipedia article, commercial releases on BASF chrome tape were treated as Type I (for compatibility, I guess?).

"Conventional wisdom": even among those with a type IV switch, many couldn't really record to metal tape.
Question: was that just audiophoolery? (I stuck to chrome myself.)

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