Originally Posted by DannyBuoy

So in J.River I am able to set the Audio Output to "Direct Sound" and then in the "output Settings" and I am able to set the USB Device to 24/96 and
I have tested it with Flac, MP3, and WMA and the sound is very stable, and to my ears, it seems more relaxed with less semblance as I was experiencing with the 16/48 native settings.

Originally Posted by bearmann

Here you can see that the output format in foobar2k is fixed to 24bit. So if your record is a normal cd rip (16bit) it'll be converted to 24 bit! But your native 24bit record (some live tape or hi-def DVD rip) won't be touched because it already has the correct/needed bit depth.
Long story short: because of this every 24bit record won't be touched (= bit perfect) and every 16bit record will be converted (NOT bit perfect).

Here you can see the resampler. Every record will be resampled to 96k. So again, if you have a hi-def 24/96 rip nothing will happen (read: bit perfect!).
But if your record has only been sampled with 44.1k it'll be resampled/upsampled to 96k! (NOT bit perfect!)

Again, long story short: With those two settings you'll ONLY get 24/96 records bit perfect to your Mav, because you force foobar2k to convert/resample every record to 24/96.

Originally Posted by sp70

24/96 Output in Ubuntu 10.04 to the Maverick D1

Forewarning: This is not meant for the linux illiterate, i've tried to make this guide as newbie friendly as possible, but some knowledge of the command line is expected.

Step one is to disable Pulseaudio from autospawning on login, we won't uninstall it, but it won't start up on login anymore, leaving you just ALSA.

To do this, run "sudo nano /etc/pulse/client.conf" in terminal and uncomment the line "; autospawn = yes" and change yes to no, leaving you with: " autospawn = no" then save that and exit.

Next, make sure "libasound2-plugins" in installed via synaptic package manager.

Then, you open your favorite editor again, gedit or nano for instance, and create a file named ".asoundrc" (minus quotations) and inside paste this code:

Save this file in your home folder. Now I recommend restarting, and if all went well, pulseaudio should no longer be running and you will have full 24/96 output with the highest quality upsampling algorithm for non-24/96 files so everything is output in 24/96. The only drawback is that it puts a decent load on your CPU, i've got a bottom of the line C2D and with 24/96 files it uses a solid 10%, and with non-24/96 files approx. 15-20%. Also, make sure your alsamixer levels are set appropriately after you restart, I usually keep mine around 40% for both master and PCM. To set them, run "alsamixer" in your terminal of choice. Lastly, since were using alsa, there can only be 1 output at a time, so you have to close out your music player if you want to watch a youtube video for instance as there can only be one open stream at a time. Its a pain, but its a necessary sacrifice for audio quality, especially if you use your machine's audio mainly for music library playback like I do.

Troubleshooting: If you get no sound on startup, make sure that your using the right PCM device. In this case, i've got my internal audio card, and my D1 through USB. Its the second sound card (1) and the first device number (0) so, "hw:1,0" in this case. To make sure that your Maverick wasn't assigned a different hw number, open a terminal and run "aplay -L" and you should see a listing for USB Audio, and the device number, you can simply replace that number with the ones in my above posted .asoundrc to get what you need. Also check your alsamixer levels to make sure nothing got muted.