32 bits Flac splitting

Good morning

I split a 32 bits / 192.000 Hz FLAC using Audacity

It is really easy to do using label track and export multiple

But, when I choose FLAC format, the bits level is limited to 24 bits.

A pity, why no 32 bits option ?

Thx for your help

See: FLAC - Format

Sample size in bits:

  • 000 : get from STREAMINFO metadata block
  • 001 : 8 bits per sample
  • 010 : 12 bits per sample
  • 011 : reserved
  • 100 : 16 bits per sample
  • 101 : 20 bits per sample
  • 110 : 24 bits per sample
  • 111 : reserved

FYI - By default, Audacity works in 32-bit floating-point so if Audacity shows “32-bit float” that’s not necessarily the original format.

Thx Steve

Ok, for audacity it is 32 bits float.

Real 32 bits didn’t exist with flac, have I unberstund correctly ?

It is 24 + 8 bits, but what is the advantage to get a float 5 bits ?

Yes it show 32-bit float

I check the file with Fakin’ the Funk and it find :

566 Kbps / 192 000 Hz / 24 bits …

So it is a 24 bits for FtF !!!

what is the advantage to get a float 5 bits ?

1. There is a lot of “math” in digital signal processing and certain things (like filtering) require summation. The programming is easier in floating-point and big numbers (from summing) are not a problem. Virtually all audio editors/DAWs work in floating point.

2. The “maximum count” in integer audio is defined as 0dB and it can’t go higher.* If you have a typical 0dB normalized/maximized file and you boost the bass (or anything else that boosts the volume) you’ll go over 0dB. Integer audio would be hard-clipped at 0dB and permanently distorted. With floating-point there is essentially no upper (or lower) limit so you can go over 0dB temporarily and then reduce the volume to a “safe level” before exporting.

Note that Audacity “shows red” for potential clipping. It doesn’t really know if the waveform is actually clipped.

Also, digital-to-analog converters (playback) and analog-to-digital converters (recording) are integer-based and they will clip if you play a floating-point file that goes over 0dB at “full digital volume” so your finished files shouldn’t go over 0dB even if you use floating-point WAV.


  • Everything is automatically scaled during playback (and recording) so although a 24-bit file has bigger numbers than an 8-bit file, they play at the same volume.

FLAC files are nearly always either 16-bit or 24-bit.
8-bit and 12-bit FLAC files are also possible, but hardly ever used.

Audacity supports importing any valid FLAC file (8 to 24 bits per sample).
Audacity supports exporting 16-bit and 24-bit FLAC.

On import, Audacity decodes FLAC files to 32-bit float PCM. This conversion is lossless.

If you mean, what’s the advantage of the additional 8 bits (32-bit float vs 24-bit integer), the main benefit is that 32-bit float can go over 0 dB. This can be a major advantage during production because clipping at 0 dB (which all “integer formats” do) can cause irreparable damage.

There may also be a very slight performance advantage in using 32-bit float during production as 32-bit float is a “native” number format in computers.

Thank you for both of you

So after splitting, I get real 24 bits.

As I ear they with my DAC NAD D1050 connected to NAD amplifier and KEF boxes, I thing that there is no major difference compare to 32 bits float.

With this material the 8 bits datas isn’t usefull

Am I right ?

If I understand this vidéo : https://www.youtube.com/watch?v=LBaaml1QXSQ

32 bits float are used to keep informations during mixing, and to correct if needed the signal. Thing not posible with fix 24 bits.

But for earing a good 24 bits is clearly correct.

I am right ?

Audio CDs are 16-bit and are considered to be capable of excellent audio quality.

32-bit float is better during mixing and processing audio, primarily because 32-bit float eliminates the risk of accidental overload (“clipping”).


Yes, for 32 bit float it is what I find and finish to understund

For the quality of 16 bits CD, compared of LP ripped in 24 bits, I didn’t agree :wink:

When recording at 16-bit, you need to be more careful than when recording at 24-bit. In the latter case you can safely allow a lot more headroom, but with 16-bit the headroom should be no more than 6 dB for optimal quality.

with 16-bit the headroom should be no more than 6 dB for optimal quality.

And for 24 bits ?

With 24-bit recording you can probably allow a lot more than 6 dB headroom, if you want to, with no audible loss of sound quality, though it does depend on the quality of the equipment.

Somewhat oversimplifying, but as a general idea:
The theoretical maximum dynamic range for 24-bit audio is 144 dB. If, for example, you have a microphone / pre-amp combination that have a dynamic range of 100 dB, then theoretically you could peak at -44 dB and still retain the full dynamic range of the analog hardware.