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Please be aware that much of the software linked to or mentioned on this forum is niche and therefore infrequently downloaded. Lots of anti-virus scanners and so-called malware detectors like to flag infrequently downloaded software as bad until it is either downloaded enough times, or its developer actually bothers with getting each individual release allow listed by every single AV vendor. You can do many people a great favor when encountering such a "problem" example by submitting them to your AV vendor for examination. For almost everything on this forum, it is a false positive.
Recent Posts
2
Validated News / Re: TAK 2.3.1
Last post by d4k0 -
Who are those "some" and why they don't recommend foobar2000 for reencoding?
Many people here can confirm that using foobar2000 for re-encoding is safe.

I searched and found one post I remembered again on Hydrogenaudio:

https://hydrogenaud.io/index.php?topic=106974.msg879004#msg879004

To be more precise: There is no technical problem with Foobar2000 and reencoding, but Foobar2000 can't replace/update existing files, it always creates new one which creates extra work afterwards.

The "easiest" solution seems to add some characters to the new file name. When the encoding is done, you have to delete all files without the the extra characters and remove these characters from the new files:

I have a conversion preset set up in Foobar, it converts to Leve 8 and adds "~~" to the beginning of every file name. Once conversion is done, delete every file that doesn't have the prefix and use something like MP3tag to remove it from the new files.
I also like to bit-compare the new files before deleting the old ones, i might just be paranoid though.
https://hydrogenaud.io/index.php?topic=106974.msg878938#msg878938

You can still use the source folder option, but rename the output file to include something that differentiates it from the original. Of course, you'll need to have space available for the converted tracks plus the original until you delete the originals.

What I would do, use source folder, and and then use %filename%-16 for the output filenames, this will allow you to convert, then you can easily filter out all the tracks that don't have "-16" in the filename and remove them.

Depending on how much space you have available, you can convert an album or group of tracks at a time.
https://hydrogenaud.io/index.php?topic=98750.msg854514#msg854514

There also seems to be foo_run, but there doesn't seem to be much documentation:

You can't replace the source files with converter. The term "converter" may be a bit misleading as it doesn't convert the originals. It allows encoding a new copy.
If you absolutely must replace files in place it can be performed with foo_run.
https://hydrogenaud.io/index.php?topic=117340.msg968867#msg968867

I think I will probably use the extra character solution.



Just be sure not to enable DSP, Additional decoding and ReplayGain in Converter settings. And you can use Binary Comparator ( https://www.foobar2000.org/components/view/foo_bitcompare ) after conversion to compare resulted files with sources.

Thanks, I'm already using Binary Comparator, it's a really useful plugin  ;).
3
General - (fb2k) / Re: libretro resampler version.
Last post by kode54 -
Something you may add in the future, if you feel like it: Neon on aarch64 supports float32x8 and float64x4 too, I think. It also supports a single intrinsic or opcode to sum up all of the vectors of a single register into one output, instead of having to pick them one at a time and add them with conventional math, using the vaddv opcode or intrinsics.

For example, this simple resampler in libsidplayfp:

https://github.com/kode54/libsidplayfp/commit/61c9a88de942f1ec428177880cc87f39a957aa31

Doing it with vectors of ints, though. Wish I'd known that it uses such small ratios to begin with. (~99 samples to 1)
5
Audio Hardware / Re: Seeking advice on new active speakers
Last post by Warepire -
Unlike higher end home audio stores which are almost gone, over here, there are still Pro audio stores (2 in my city) that you can go to and see/listen/buy. The reason I'd suggest this vs buying online as most do today, is the bolded part above.
There's unfortunately nothing like that in my area. There used to be but they closed a few years ago. Which is why I am trying to gather information online, and reviews weren't helping. They either went over my head and/or had titles like "perfect for electronic music" which isn't the music I aim to play the most.

There is almost no way to know whether you will have a bit of hiss with active monitors using single ended (rca) at your listening location. If that's not what you meant by "quiet", then there's less risk buying online, then having to return.
A slight hiss/buzz would be fine with me as long as I can actually hear the music. I had a pair of earphones once that just couldn't reproduce it at all (just silent), same with the more extreme death metal (too static:y) (or what the technical term for those situations are), which is what I am trying to avoid. Or maybe such a worry is not a thing with speakers at all?
6
3rd Party Plugins - (fb2k) / Re: foo_uie_lyrics3
Last post by Bladru -
v0.5 leaks 3 GDI objects per track. Could someone please fix that?

You can show the number of GDI objects per process in the Task Manager. I had this lyrics panel embedded, and foobar got to 6k or 9k GDI objects, at which point UI in other programs began to slow down. A workaround is to restart foobar or to keep the panel in a separate window and to close it from time to time.
7
Lossless / Other Codecs / Re: How to convert DTS 5.1 (one file wav/dts+cue) to multi-ch Flac
Last post by Rollin -
Foobar, when I play dts files (as one big wav) identifies songs as 16 bit, 2ch, 44khz independently what is real bitrate etc.
This is because your files are packed into wav container.

Is there any "checker" what can check these files and inform me what is REAL bitrate, channels etc?.
ffmpeg

Is there really  easy soft, with input box and output folder.... And no questions asked.
ffmpeg can easily convert to flac, but cannot split. But you can split resulting flac with foobar2000
ffmpeg.exe -i inputfile.dts outputfile.flac
9
Lossless / Other Codecs / Re: How to convert DTS 5.1 (one file wav/dts+cue) to multi-ch Flac
Last post by Skie -
1. I really didn't know that it's soooo complicated. I just don't want to have big DTS files (where 1 file is 1 CD) because it's not possible to play it at my place with my equipment (whatever it is, it's not important).
All I wanted is to have clean multichannel FLAC, with original bitrate and as close to original sound as possible.

2. Foobar, when I play dts files (as one big wav) identifies songs as 16 bit, 2ch, 44khz independently what is real bitrate etc. Very irritating. Sometimes I know if wav is 24 or 16 bit but sometimes not. Is there any "checker" what can check these files and inform me what is REAL bitrate, channels etc?

3. Is there really  easy soft, with input box and output folder.... And no questions asked.

Thank you all for your opinions on this matter, which you wrote here and I hope you will write more
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