I use M-audio audiophile 24/96. I don't see any configuration options to use the 24 bit sounds, every player plays up to 16 bit.
Can you help me use the full potential.

Regards

Hi, as far as I am aware, and reading this conversation...

http://lists.freedesktop.org/archive...ay/006915.html

Pulse should be able to handle it.

The 24bit @ 96kHz is for recording audio (you won't notice it playing an mp3 file).

This higher than CD resolution is nice when editing sound tracks,

allowing finer adjustments to the cut and paste editing and post-processing (adding effects, etc).

My Edirol UA-25EX (roland) will only do 96kHz when recording, no realtime monitoring of the previous track/signal.

But it does do a few other things I like and use.

Bottom line, if the M-audio 2496 is being used as playback of mastered material,

like movie sound and music tracks, it will operate as a normal sound card and deliver the program material.

If you want to record high quality sound tracks (for multitracking) the config lies with your software applications.

Hope this helps, Glenn

Hello, GlennsPref

When I'm listening to 24 bit flac files, there is a notisable difference in the quality. Human ear can't notise the difference between 41khz and 96khz but that is not relevant
My player plays only on 16 bit on Ubuntu. I need it to play 24bit so i can listen to my 24bit flac music.

Regards

I manage to fix the problem. Thank you for your help.

Hi, how did you fix it?

cheers, Glenn

I know that the problem has already been solved, but nobody posted the solution, and this post on LQ was the first thing that showed up on DuckDuckGo. Would have been nice if the information had been right there for me when I needed it, so I'm posting what I've found.

Here it is:
We're going to need to edit /etc/pulse/daemon.conf as root.
For Ubuntu (or really any distribution using Gnome), open up the terminal and paste (ctrl+shift+V, if using your keyboard):

*use kwrite for KDE, leafpad for LXDE, etc.

Gedit (or your text editor, if not gedit) should pop up with the file.
Find the lines: (hint: they're at the end of the file | use ctrl+F if you're lazy :P)...And change their values. I would personally change only one at a time, starting with the sample rate. Accepted ones are generally: 44100, 48000, 96000, and [rarely] 192000. Decomment the line when you are done. For example: Restart at your leisure.

...You restarted, right? Make sure the audio is working. If not, try a different sample rate or recomment out the line. If it is, indeed working, it's time to change the sample-format. There are actually a crapload of values for this; unfortunately, the only way I know of to find the best value is trial and error. Keep going to a higher value (restarting each time) until it no longer works. Here are the 24 and 32 bit values: As the default is s16le, it's probably a good idea to start with the *le ones if Pulseaudio was working with the default value.
Example: Assuming everything went well, your audio should now be amazing, just like it would be in Windows.

Source: http://lists.freedesktop.org/archive...ay/006917.html

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Thank you for your efforts.

(Quite Interesting.)

Cheers Glenn

I found out what le and be mean:
le = "little endian"
be = "big endian"
Presumably, you would also be able to use:
alaw, mulaw or imaadpcm, float or int, unsigned or signed, in addition to le or be
(I leanred that here: http://web.njit.edu/all_topics/Prog_...yer/sound.html)
But I'm not sure how. Float would, of course, increase audio quality even further if it isn't on by default, so it would be cool to get that in there, if possible. Also, big endian is undesirable for the average computer under most circumstances, so for those reading my mini-tutorial above, you shouldn't use big endian unless your audio is completely broken or you have a good reason to do so.

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