Skip to main content
Recent Posts
3rd Party Plugins - (fb2k) / Re: Columns UI
Last post by gob -
Just wondering if there's any known performance issues between CUI's spectrum analyzer and WASAPI playback. The audio is fine, but the spectrum analyser is extremely choppy when outputting to any WASAPI device. I seem to recall there being a fix for this.
3rd Party Plugins - (fb2k) / Re: [Not my release] foo_uie_panel_splitter
Last post by fabiospark -
Is there a way to know the vertical space used by an auto wrapped text?

I have a "performer" field that sometimes gets quite populated and
I would liket to check if it can be shown into four lines otherwise
I would add a "... and others" at the end.

To do so I need to know how many lines the wrapped text is going to use
but the wrapping is done by the panel itself (thankfully) so I don't have
a way to know that by myself.

Audio Hardware / Re: Subwoofer advice needed
Last post by Fairy -
Just to let you know. I've been to a good hifi store where I have bought my CM10's and NAD and I ordered the B&W DB4S subwoofer. Quite a new model. A bit (a lot!) above my initial budget, but it's so damn advanced compared to the average subwoofer. No dials and switches, everything happens inside the amplifier via a bluetooth connection. You can set a lot more parameters.

Only problem left... I have to wait till wednesday or thursday before my sub arrives  :-[  :-[
Scientific Discussion / Re: Audio Summing Algorithm
Last post by Nikaki -
You just add the sample values. Nothing special needed. If you exceed 1.0, you can just clip when you convert the final result to integer samples, or divide every sample in the final result so that the highest peak is 1.0.

If you want to guarantee that there won't be any clipping during the mixing, you would need to divide the samples by the amount of streams to be mixed.

There are results for this on the net. You might want to search for "mixing audio samples clipping."
Scientific Discussion / Re: Wrote some scripts that partially undo early Exciter type devices
Last post by jsdyson -
Here is the current (functioning) source for the 'unexciter' code.  It does work, and does remove some of the nasal quality of some of the older recordings.  It runs exceptionally slowly and is not algorithmically optimized.  This is more of a sound effect (or undoing the sound effect) rather than a recovery/decoding program.  That is -- there is a slight change in the sound of a recording, and the results are middling at best.  The only places where it seems to be of substantial benefit are for ABBA recordings (pure DolbyA ones -- not the CDs from Polar,etc) and also it seems like the old Herp Alpert/Brasil'66 really benefits.  Otherwise, it is more of a 'tone control.'

Is this worth it?  Probably not, but I had advertised that I was working on this, and trying to be honest/transparent about the semi-bust for this development (the DolbyA decoder and compressor/expander on the other hand are fantastic.)

The program takes standard input/output redirection for .wav files (16 bit, 24 bit or 32bit floating point, any sample rate between 44.1k through 192k or higher -- but 88k through 192k are best.)   Higher sample rates are slower (lots more fine grained calculations.)
The best options for cleaning up/correcting old recordings are the following:
--width=1.414 --hilb=-1.0 --dr=0.156
Since I am not really supporting this, and there is likely very little interest (has to be used on Linux -- didn't do a Windows port) -- the explaination of the args are thus:
--width is the M/S Side expansion before subtracting the hilbert of the signal.
--hilbert is the proportion of the hilbert added/subtracted, -1.0 means subtract 100% of the hilbert of the M/S expanded version.
--dr is the proportion of the weirdly phase shifted signal that is effectively subtracted from the signal.

Bugs:  not necessarily very useful, considering the dreadfully slow runtime, the results might be  inverted (by mistake.)
Benefit: forced me to rewrite the filter infrastructure so that it can adjust to any sample rate -- now the new filter code benefits the psuedo-DolbyA decoder and soon will benefit my super nice expander and useful compressor.  (BTW, an example of the results of my expander is the almost total uncompressing of 'Shake it Off' -- my expander can do incredibly high expansion ratios without jerking...)  The site of the examples is:

The ABBA example on the site (SOS) was un-(Aphex)-Excited.

I try to follow through with implications that I make -- and attached is the current (working and functioning) source code for the unexciter.

Scientific Discussion / Audio Summing Algorithm
Last post by antony96 -

I'm a newbie in digital audio programming. Can anyone point me to an article/formula on how to mix two or more audio streams together? It's probably very simple, but I haven't found anything via Google/Bing.

I'm operating on 32-bit floats on a Mac and managing my own audio callback loop (not using Core Audio mixer objects)

Support - (fb2k) / iOS app: "Playback error: Length of object is unknown"
Last post by perdido34 -
When I try to play audio files from media servers on my home PC, sometimes I get a message, "Playback error: length of object is unknown," followed by a long http link to my PC where the server and files are located. Then no sound will play. The screen shows the filetype incorrectly as m4a.

A few minutes later, there is no issue and the same files play without a problem (FLAC, WAV, mp3, DSD64etc.). I have an iPhone 8+ running iOS 11.3, but I had the same intermittent problem with earlier phones and iOS versions. It doesn't matter what filetype I'm using or which media server I use (Asset uPnP or foobar2000 on my PC).

Any suggestions?
General Audio / Re: Massive Attack encode Mezzanine into DNA using Opus
Last post by jaybeee -
yeah I thought that. I suppose 5000 tiny beads could turn into 25,000.
More like 2 million, or more. I would imagine they would use the lowest possible bitrate for this. It's not like anyone is going to complain about audio quality here  :P
hehe that could very well be true also. I was thinking they'd use >200kbps, hence the ~25,000.
3rd Party Plugins - (fb2k) / Re: foo_uie_lyrics3
Last post by gac123 -
hi everyone, i have a quick question regarding the saving of lyrics to the tag

it seems that the way it is beeing saved by Lyric Show 3 is it concatinates the time for lines that are repeated

so the way the synced lyrics are layed out in foobar with the lyric show 3 plugin is as follows:

[ti:The Contrarian]
[ar:A Perfect Circle]
[al:Eat the Elephant]

[00:30.98]Hello, he lied
[00:35.32]Like velvet this magician's
[00:42.25]Sleight of tongue and hand
[00:46.99]Hello, he lied
[00:50.10]Beware, belie his smile
[00:54.11]As warm and calculated
[00:58.78]As heroin

but when actually looking at the tag (with mp3tag for example), it seems the times are joined:

[ti:The Contrarian]
[ar:A Perfect Circle]
[al:Eat the Elephant]

[00:30.98][00:46.99][02:47.02][03:03.03]Hello, he lied
[00:35.32][02:51.27]Like velvet this magician's
[00:42.25][02:58.28]Sleight of tongue and hand
[00:50.10][01:54.09][03:06.15]Beware, belie his smile
[00:54.11][03:10.10]As warm and calculated
[00:58.78][03:14.97]As heroin

this actually creates a problem for other players that support lyrics from tags in that instead of presenting them neatly like Lyric Show 3:

Hello, he lied
Like velvet this magician's
Sleight of tongue and hand

Hello, he lied
Beware, belie his smile
As warm and calculated
As heroin

it looks messed up:

Hello, he lied
Hello, he lied
Hello, he lied
Hello, he lied
Like velvet this magician's
Like velvet this magician's
Sleight of tongue and hand
Sleight of tongue and hand
Beware, belie his smile
Beware, belie his smile
Beware, belie his smile
As warm and calculated
As warm and calculated
As warm and calculated
As heroin
As heroin

so my question is, is there a way to force Lyric Show 3 to just save it to the tag as it is entered when editting them directly from the panel?
SimplePortal 1.0.0 RC1 © 2008-2018