Converting lossless M4A to FLAC degrades resolution from 24-bit to 16-bit

[Arcee72]

When I convert M4A files that contain 24/96 or 24/192 lossless audio streams to FLAC with FREAC, the FLAC output files end up being 16-bit resolution instead of 24-bit. File size is dramatically reduced. I lose a lot of information, not what I want.

I do get a warning that I am converting lossy audio to lossless, and I see that the Generic MPEG-4 Audio Decoder is used.

In settings, under processing I turn off 'Enable signal processing' as I just want to make a bitperfect conversion.
Still, 24/192 converts to 16/192.

I've tried both Fre:ac versions, on my Silicon iMac and on my Windows 11 PC, both with the same result.

[enzo1982]

I just tried converting a 24/96 ALAC/M4A file to FLAC and it came out fine. I'm also seeing Apple Lossless Decoder in the decoder field during conversion.

Are you using an up to date version of fre:ac? The native Apple Lossless decoder was added in version 1.1.6. Before that, fre:ac used FFmpeg to decode ALAC files, but that defaults to 16 bit output even if the file is 24 bit.

It's also possible that fre:ac does not recognize your ALAC files for some reason and falls back to FFmpeg as a last resort. In that case you would get 16 bit output.

[Arcee72]

Thank you @enzo1982 - I do have the latest version from the official fre:ac website and using it on an iMac with a Silicon processor, in the latest MacOS. Are you using Windows or MacOS and did you install the Apple Lossless codec separately or did it come automatically with the initial installation of fre:ac?

[enzo1982]

It's actually the same behavior on both, Windows and macOS. The Apple Lossless decoder comes with fre:ac; no need to install it separately.

If possible, please make one of the problematic M4A files available on a cloud drive and send a link to support@freac.org. I will then have a look and figure out what's the issue.

[xkater]

I was about to ask a different question, but i try to answer this instead -
In Options->General Settings->Components->DSP (scroll all the way down) you can set both 'Sample Format' and 'Sample Rate' converters. I just tried this and i was able to set Sample format to 'Integer 24bit' and Sampling rate to "96000". Using these i was able to 'upsample' 16bit/44kHz to 24bit/96kHz, i also upsampled to 32bit/96kHz.
One anomaly: mediainfo reported both files as "32bit", although the file set to convert at 24bit file (reported as 32bit) was almost 20% smaller - this could be MediaInfo error ?!?
One anomaly : in MediaInfo

[xkater]

just tried it again and got the same MediaInfo - both files declared as "32bit" even though one is 10K smaller.