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1
Unfortunately it was reported earlier this year and wasn't fixed in 17.10. Hopefully it will be moved to main or fixed in another way before 18.04 LTS.
2
foo_la.dll doesn't work in new version of Foobar.

Foobar cannot play lossless audio.

Foobar says that listeners must create the component with newer Foobar SDK.
Spoiler (click to show/hide)
If you're happen to have Visual Studio installed, please make the dll for me!
3
I believe it's sample rate.
"The time resolution of a 16 bit, 44.1khz PCM channel is not limited to the 22.7┬Ás time difference between samples. The actual minimum time resolution is equivalent to 1/(2pi * quantization levels * sample rate). For 16/44.1, that is 1/(2pi * 65536 * 44100), which is about 55 picoseconds. To put that in perspective, light travels less than an inch in that time. "
4
General Audio / Re: Merge two audio files (ADX extension)
Last post by notbugme -
You mean, there is a vocal track and a "karaoke" track, and you want to mix them into a song?

This is definitely the case.

I just remembered something...   Audacity doesn't mix 'till you export.   If you re-import the file and it's hitting 0dB* it's probably clipped.

There are a couple of ways to handle this.   You can export to 32-bit floating point, re-import and normalize.   Or, reduce the volume of both tracks by -6dB before mixing/exporting.   (The -6dB may leave you with some headroom.)

* You can check the peak level in Audacity by running the Amplify effect (and you can cancel before applying the effect if you wish).     Amplify will pre-scan the file and default to whatever gain change is needed for normalized/maximized 0dB peaks.   For example, if it defaults to +2dB, your peaks are now -2dB (you have 2dB of headroom).   If it defaults to -2dB, your peaks are +2dB and a regular (integer) WAV file will clip.   If it defaults to 0dB (no change) then your peaks are exactly 0dB which means the file was previously normalized or clipped

Audacity showed me the mixed file was clipping, but using Amplify with a -0,001 db value was enough, so I guess that's why I couldn't here any distortion, and both tracks were probably already normalized. To save the file, I disabled dither (since Audacity uses it by default with FLAC) and put 16-bit as default sample format in Preferences -> Quality.

Would it be more correct to convert the two ADX files to FLAC in advance before working them on Audacity?
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bandpass: Thank you for your reply. I sometimes use Ubuntu but mostly Void and OpenMandriva when I listen to music and I haven't checked them yet. I didn't know that Ubuntu doesn't include soxr. Strange.
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3rd Party Plugins - (fb2k) / Re: foo_scrobble
Last post by elia_is_me -
I've not tested this yet but given it has dialogs for artist/title title formatting, it should support not submitting based on tags like the old component.

Just make sure the value is blank if a certain condition is met and then the component should either not submit or last.fm will refuse the submission because of bad parameters. For example...

Code: [Select]
$ifequal(%rating%,1,,%artist%)
+1
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thanx, so i need the "obsolete" foo_run 0.3.7 or how do i use your lines?
8
Thanks, I'll give that a try.
If we forget about playing through the Chromecast which phone apps can I use to just play my media server files on my phone?
The BubbleUPnP app I linked above.
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Thanks a lot.
10
@EpicForever you can add -loglevel 0 to Additional arguments. This should fix disappearing of waveform for files from your archive when decoding with foo_input_ffmpeg.
Westbam... is really broken/malformed. ffmpeg, mpc-hc/be+LAV, VLC, mediainfo - all report wrong length.