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General - (fb2k) / Re: foobar2000 v2.0 bugs
Last post by 0x00000ed -
Beta 29 x64, portable.
Upon opening FB2K window is not maximized. If I maximize the window, exit, restart, the window is not maximized again. Does not happen in Beta 27 (haven't tested Beta 28).
29 too.

F2K v2.0 beta 29
Two playlist annoyances:
-it does still have those abrupt jumps to anywhere when changing something that can alter the name of the group.
-it's very slow switch from tab to tab that it has a lot of content (more than 500 items and up), and the situation gets worse the more number of items.
Tested on 100k tracks in multiple tabs, everything works for me.
General - (fb2k) / Re: foobar2000 v2.0 bugs
Last post by Fabcore -
F2K v2.0 beta 29
Two playlist annoyances:
-it does still have those abrupt jumps to anywhere when changing something that can alter the name of the group.
-it's very slow switch from tab to tab that it has a lot of content (more than 500 items and up), and the situation gets worse the more number of items.
General - (fb2k) / Re: foobar2000 v2.0 bugs
Last post by Chibisteven -
New bug in Beta 29: foobar2000 no longer remembers if it's main window was maximized.

Easy to reproduce:
1. Run foobar2000 and maximize window.
2. Close foobar2000.
3. Run foobar2000 again.
General - (fb2k) / Re: What's the best way to increase loudness while preventing clipping?
Last post by Kyddo -
Got it. Thank you. I finally have a much better understanding of the whole issue. I'll probably just leave it the way it is then and keep the files I originally got them. I'll keep in mind to search for headphones with better loudness specs (currently have the Sony wf-1000xm4 earbuds. Fantastic quality but just want it to be a little louder. Perhaps the issue is the phone I play my music from haha.)
General - (fb2k) / FFmpeg Decoder Wrapper (foo_input_ffmpeg) passthrough output
Last post by gorman -
First of all, I'm no coder. Looking at the console, it seems to me that the ffmpeg plugin assumes a certain use for it. Namely that of decoding all kind of audio that users pass to it. That's understandable. Object-based surround codecs, though, can't be decoded this way and, in this case, would need a true passthrough option. FFMPEG has that capability, I think through its spdif_muxer (I guess the name has historical reasons, considering hi-res formats are beyond spdif capabilities).

Is there anything that could be done to make this accessible to end users? One could use only .mka files for Atmos content and pass those through to an AVR capable of decoding them.

It's the same that happens if you play WAV encapsulated AC3 or DTS content, without having a decoder installed for that kind of content. If you use WASAPI and output bitperfect content (fixed volume at 100%), the AVR decodes it if connected through HDMI (or spdif in that case, I suppose).

Currently the hardcoded instruction looks like this:

ffmpeg.exe" -i "INPUTFILE" -map 0:0 -f w64 -acodec pcm_f32le -

Easiest solution could be that of having a "tick" one could activate, per file type, that would turn that into a passthrough working one. Or maybe having an option to completely disable the hardcoded portion, leaving only -i "INPUTFILE" and users to find out for themselves how to activate passthrough (I'm not sure it's possibile through just command line arguments, haven't investigated due to not being possible to currently eliminate the hardcoded arguments).
FLAC / Re: Retune of FLAC compression levels
Last post by Porcus -
CDDA results, likely confirming that the change in LPC order selection does matter.  Which is not to say it is a bad thing, given the speed-up.
Also -r8 seems to not be worth it.
Corpus: 38 CDs from my signature. Not reliably timed, for that I have to leave the computer untouched and run repeats.

Baseline is current -8.  Run with 1.4.2 (x64), takes ten minutes and a half.  About the same time is the first of these:

* -8r6 -A subdivide_tukey(4).  Not the first that I tested, but I put it here because it takes pretty much the same time as 1.4.2 at -8. 
And compresses 0.02% worse.

* retune -8r6  -A subdivide_tukey(3). That is, the same parameters as 1.4.2 -8, so the changes are in the LPC order selection algorithm.
Considerably faster: eight minutes.
Every file is bigger than 1.4.2 -8, but only slightly so: none hit the 0.1 percent difference mark. The classical music increases by 0.042 percent, the heavier rock by 0.026 percent, the "other" in between.

* old -7
Maybe half a minute faster than retune -8r6  -A subdivide_tukey(3).  Bigger files, except some classical music.  The classical section is 5 parts per million bigger.
That means it is about the speed of retune -8r5 -A subdivide_tukey(3), see the final comparison.

Then the impact of "-r":
Above I did the retune -8r6 -A subdivide_tukey(3) (= old -8 options).  Changing the "r" to 7 or 5 costs/saves half a minute (atop eight minutes).  Impact:
* The "r" to 7: ten parts per million.  One album as high as 0.011 percent. Eight CDs (six classical and two metal) ended up with exactly the same number of bytes.
* The "r" to 5:  62 parts per million.  One album (a different one!) up by 0.080 percent.

One final comparison:
Since retune -8r5 -A subdivide_tukey(3) is about the same time as 1.4.2 at -7, what is the difference?
retune -8r5 -A subdivide_tukey(3) produces slightly smaller files: 0.019 percent.
The difference is about zero for classical music; those files increase by 4 ppm (the impact of the -r is about 10 ppm, and the rest makes for -6).
Driving the difference in favor of the retune, are Kraftwerk and Arman Van Helden, they are electronic driven and are the ones that benefit most from increasing the "-r". You'd expect then that they would lose from the -r5 etc. setting? No, they benefit even more from the subdivide_tukey(3) and whatever other changes you made.
General - (fb2k) / Playlist Search UI element?
Last post by Scoox -
I discovered foobar2000 two days ago and I think it's absolutely incredible. I'm truly grateful for such a fantastic piece of software. Whilst configuring the UI, the need arose for a way to search within playlists and it turns out foobar2000 already has this feature built in, although it is somewhat hidden (hold shift while clicking the View toolbar menu, then click Playlist Search). I notice Playlist Search cannot be added as a UI element, and I was wondering if this is something that might be implemented in the future? Luckily, it can be mapped to a keyboard shortcut, which I have done.

A component named SimPlaylist includes a playlist search UI element but the styling is a bit different from the default foobar2000 UI and there are a couple of UI graphic bugs. Seeing as foobar2000 already has a built-in Playlist Search facility that works as expected, I'd much rather use that than a component.
General - (fb2k) / Re: What's the best way to increase loudness while preventing clipping?
Last post by fooball -
Would increasing the replaygain db to like 92 work then (is that even possible or is it forced on 89 db)? Because unlike pre-amping, replaygain doesn't alter the data of the file. So whenever I need to I can just put the song in foobar and take out the replaygain info, thus putting it back to normal, whenever I get a better speaker.
It doesn't matter how you do it, maximum is maximum.  The best you can do whether by pre-amping or ReplayGain is scale the data so that the absolute peak level for a track is the maximum.  Any more than that and the peaks will get clipped.  The purpose of pre-amping or ReplayGain is to restore tracks which have been recorded "quiet" to a normality.  The target dB for ReplayGain is set to allow a reasonable dynamic range before the onset of clipping (but very dynamic tracks might still clip).

Compressors (eg VST) reduce the dynamic range of a track, which means the softer parts are made selectively louder without boosting the peaks into clipping.  This will give an overall impression that the track is louder, even though the loud peaks are not.  It's intended to overcome noisy environments where the soft bits get lost below the noise floor.

If your problem is background noise, then a compressor will help.  If the fundamental problem is that your audio system isn't powerful enough, there is only one solution.