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.
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- 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 
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.