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General - (fb2k) / Re: wav 32bit fixed-point to FLAC
Last post by Peter -

With the increased pressure to make a 64-bit version of foobar2000, I will be eventually releasing a new version that's not ABI-compatible with the current components.
At that point, I can also address the above, by using 64-bit floating-point for internal data format. (Yes, changing the data format means breaking all components.)
However, I don't find the current arguments convincing, sounds like whatever software writes the offending files should be configured to use already-well-supported 32-bit floating-point instead.
General - (fb2k) / Re: wav 32bit fixed-point to FLAC
Last post by Peter -
Let me clarify.

foobar2000 uses 32-bit floating-point as its internal data format.
All audio is converted to 32-bit floating-point when decoding.
24-bit fixed-point and 32-bit floating-point can be transferred losslessly.
Unfortunately, least significant bits of 32-bit fixed-point get lost when converting to 32-bit floating-point and back.
I don't consider this much of an issue because theoretical dynamic range of 24-bit is already overkill. However, to stop people from complaining about unsignaled data loss, warnings have been introduced when converting 32-bit fixed-point.
foobar2000 can PLAY 32-bit fixed-point just fine, unless you consider lossless delivery of full 32bit stream (rather than most significant 24bit part of it) to your audio device to be critical.
Conversion of 32-bit floating-point IS LOSSLESS, foobar2000 is more than happy to read and write it, not having to convert to an internal format and back.

I don't understand why pro audio software isn't using 32-bit floating-point as an intermediate format, it has practically infinite dynamic range and doesn't suffer from clipping.

Using fixed-point instead of floating-point gives no performance advantage, quite the opposite (no need for clipping when applying gain).
General Audio / Split file with PCM float samples by cuesheet
Last post by 9p -
I have a vinyl album rip in form of WAV audio file together with a matching cuesheet. The audio file is mono 96 KHz with samples in PCM 32-bit floating point format. I would like to split it into individual tracks. Unfortunately my usual tool of choice shntool refuses to work with float samples:
Code: [Select]
shnsplit: warning: unsupported format 0x0003 (IEEE Float) while processing file: [audio.wav]
shnsplit: warning: none of the builtin format modules handle input file: [audio.wav]
shnsplit: error: cannot continue due to error(s) shown above
I know I could change the sample format to integer with something like ffmpeg but I'd prefer something that can work with float samples natively if possible. I also noticed that Opus encoder is able to take this file as input without any problems. Do you know other good splitters I can use for this task? (preferably ones supporting Linux)
General - (fb2k) / Re: wav 32bit fixed-point to FLAC
Last post by Porcus -
@regor, are 32-bit integer widespread at all?
32-bit float is at least float, so they have a mission in life.

(Also, from what I understand, 32-bit float - like DSD - should be converted using WavPack standalone rather than foobar2000.)
General - (fb2k) / Re: wav 32bit fixed-point to FLAC
Last post by regor -
... it's not about playback. It has been told a million times and people seems to not want to listen to it.

All DAWs in the world work with 32 bit files, whether people like it or not. And it's not about magical listening differences, but reducing artifacts if further editing the files on the future. So 32 bits is used for archival of production files. Many people here use foobar too, and the ability to batch convert those files to a reasonable format for playback would be useful. The same for tagging, preview listening, etc. Not having to use a DAW for that is QoL change.

That's why wavpack allows 32 bits (it makes no sense to archive file in .wav when you can compress them). It's not about raising the bar to 256 bits or 1024 bits... It's about giving support to bit depths used in the real world and making our lifes easier. If you don't want to support it is your choice (and it's ok), but let's stop the -anti-audiophile- (*) discussion of those bit depths not being "useful" in the real world... because they are. Real world is not only about listening (and I think we are at a point where foobar has gone farther than being just a player: tagger, converter, file management, etc.).

Flac has chosen the same path and it has been clearly a poor choice. 2021 and you still have to save production files as wav (or wav-pack) because the reference encoder/decoder of the most extended lossless format on the world doesn't allow 32 bit files. A joke. But all DAWs in the world save the files in 32 bits by default. The joke becomes even bigger. We are wasting disk space just for the joy of wasting it.

(*) Because that's the only real reason we still have this discussion. People continue arguing about those bit depths being useful or not for listening, and that's not the point at all. But some "scientific" users are so focused on that question that totally forgot audio is not only the listening experience of the 'final user'. There are tons of users who also record, produce music, etc.
Site Related Discussion / Re: Hydrogenaudio IRC channels
Last post by Peter -
I removed the nick registration requirement for now.

Shortly after setting Libera channels up there was a flood of spam bots joining and yelling "#channelname was moved to freenode", blocking unregistered users was the only way to stop them.

Also, various IRC clients automate the process of identifying and joining channels on startup, without the need to write scripts or alike.
General - (fb2k) / Re: wav 32bit fixed-point to FLAC
Last post by Peter -
Please provide one scientifically valid reason why 32bit fixedpoint PCM should be used or supported and I will investigate how to get it working properly for fb2k 2.0.
It seems to me like once I raise the bar (heh), soon people will want 64bit and 128bit, which are just as pointless.
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