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[percy]

Only digital noise on output of in 4 stereo channels separated Asus Xonar AE sound card configured in ALSA

I am building a new JRiver music server for my multi-room setup as replacement for my current 10 year old server which is running Ubuntu 22.04 and Mediacenter 31.0.83 and has an ALC1200 as well as a Asus Xonar Essence ST+H6.

The new server is also installed with Ubuntu 22.04 and Mediacenter 31.0.83.

As this new server must provide audio for 4 stereo channels and 4 mono channels just as the old one did, I installed an extra Asus Xonar AE sound card as replacement for the Asus Xonar Essence ST+H6 with was a PCI card.

The onboard audio which is provided with an Realtek ALC897 chips provides the mono channels and works as expected.

But I cannot make the Asus Xonar AE sound card work as I wish (providing 4 stereo channels from the available 8 channels.

The configuration works perfectly when I try to output with “aplay” command (whatever PCM format). But using the same device in Media Center only outputs noise (digital garbage)

I attached a set of my configuration and settings and output of several Linux commands and the corresponding Media Center logfile which I hope can clarify my problem.

I would be extremely grateful for any help or suggestion. Many thanks in advance.

[mwillems]

Thoughts:

1. The first thing I would try: have you tried using the default alsa "plug" or "plughw" device output in JRiver?  I ask because aplay appears to be routing through the plug device, and the plug will sometimes help with minor compatibility issues.  For example, I had a Steinberg interface that would output digital static on any device other than the ALSA plug device, but the plug worked perfectly.  It was truly a mystery. 

2. Have you ensured that no other software is locking the sample rate of the interface?  I've had similar issues with JRiver when, say, the system audio via pulse has locked the interface.  For testing, make sure to confirm that, say, system audio is set to an entirely different interface (like the MoBo interface) to confirm that's not the issue.

3. Your asound.conf is very peculiar.  Have you tried testing with the default or even an empty asound.conf to see if you can get sensible playback?

[percy]

Thanks already for your useful info. The default device in the OS settings is the HDMI audio of the display port.
I made some extra tests. Changing the default card in the sound settings of the OS from the actual Xonar to HDMI of the display port does not make any difference.
“plughw:CARD=CARD,DEV=0 XONAR SOUND CARD, USB Audio Hardware device with all software conversions” This device selection in MC31 works flawlessly.
"hw:CARD=CARD,DEV=0 XONAR SOUND CARD, USB Audio Direct hardware device without any conversions", "front:CARD=CARD,DEV=0 XONAR SOUND CARD, USB Audio Front output / input"
Hiss just as my asound.conf defined ones.
Some questions: is there a way to discover how plughw device that works is configured? Is there an OS configuration file or a command that can dump those configuration settings?
What would be the pcm name to be used with the aplay command for the “plughw:CARD=CARD,DEV=0 XONAR SOUND CARD” device, that way I could see what asound.conf parameters would be appropriate for that device?
Using the card without asound.conf file would not give me the possibility in MC31 to use the rear, side, center channels for separate rooms playing different music (Or at least I do not know how that could be done).

Thanks for any extra suggestion that i can try out

[mwillems]

Glad the plughw is working for you.  As I said, I had an interface that only worked with that interface and no others, and I never quite figured out what it was doing differently.  My understanding is that the plughw device will do conversions as necessary.  For example, my problematic Steinberg interface had more logical channels than I was trying to use (it showed up as something like a 17-channel interface despite only having 8 analog outputs, of which I only needed 6), and I would get either errors or static with all other output options (apart from pulseaudio for some reason, but I was trying to avoid that for several reasons).  I always assumed the plughw was just more tolerant of channel counts that weren't exact matches for the interface's exposed number of channels, as sample rate switching still seemed to work as expected.  However, after much experimentation with settings, I couldn't figure out how to get working output any other way than using plughw with that particular interface, so I eventually just gave up and embraced a working solution as the plughw seemed to be good enough for me.  Years later, I eventually replaced the interface with one that has better Linux support (A topping DM-7).

But, if you can figure out what the plughw is doing differently, I'd be interested to know!

Here's a link to my troubleshooting saga with my notes at the end.  Not sure if it will help, but part of the issue was that the device was reporting incorrect format support which led to the static in my case (i.e. jriver was trusting the device's self-report for compatibility, which lead to static output).  https://yabb.jriver.com/interact/index.php/topic,131885.msg914906.html#msg914906

[percy]

Thank you, I planned to do the same thing and buy a couple of TOPPING D10S and dump the Asus Xonar AE. And it is correct that if the number of channels in the asound configuration doesn’t match the hardware reality then MC will not work and neither there popup nor there logfile gives useful information. Probaly the should make the logging more configurable (like log4j form the java language) so that one can select log info just for the playback chain or the import or whatever but anyway something is better then nothing at all
Thanks again for jour feedbacks

[percy]

For those interested I kind of solved my problem (at least it works)

Forcing in asound.conf the rate of the slave.pcm to 44100 and the format to S16_LE and then configuring the JRiver Mediacenter “Tools -> Options -> Audio” Device settings of all the separate zones (pcm type  plug) to auto and forcing in “DSP & Output” format the “Sample rate” of all input types to 44100 (The channels settings where left on 2 channels (stereo) with extra channels on 6).
Now all zones work on all types of sample rate and I can play different tracks to different zones simultaneously. The downside is of course that everything is down converted to 44100 Hz/16 Bits which is kind of a bummer but all these zones are for background music only.
Many thanks again to Mr. "mwillems" for his kind support.