LinuxMusicians

@Kernel Configuration A lowlatency kernel has been built with "1000Hz" and "Preemptible Kernel".
A realtime kernel is a kernel built with "1000Hz" and "Complete Preemption".
?

If so, I would definitely install a lowlatency and/or realtime kernel.
You can install/uninstall different kernels and select the one you want to boot and test it yourself.

The Preempt / 1000Hz combo works pretty well for audio & general desktop computing. It's a good all-rounder.
RT Kernel's can be tricky & problematic and I didn't notice much if any benefit over a Preempt kernel with regards to xruns.

Maybe we're going off topic a bit. All in all: Linux support for the Behringer 404HD is still good.

Nevertheless I couldn't access the 4 recording input channels on the unit. Just 2. Tried 2 Ubuntu studio versions (16.04, 18.04) plus kxstudio 14.04 and of course kubuntu 16.04. Anybody achieved to do it?

ajoliveira wrote:Nevertheless I couldn't access the 4 recording input channels on the unit. Just 2. Tried 2 Ubuntu studio versions (16.04, 18.04) plus kxstudio 14.04 and of course kubuntu 16.04. Anybody achieved to do it?

1. Determine the numbers of your soundcards:

   Code: Select all

   cat /proc/asound/cards
   

   Output will look something like this:

   Code: Select all

    0 [CMI8738        ]: CMI8738-MC6 - C-Media CMI8738
                         C-Media CMI8738 (model 55) at 0xd000, irq 20
    1 [PCH            ]: HDA-Intel - HDA Intel PCH
                         HDA Intel PCH at 0xf7c10000 irq 45
    2 [U192k          ]: USB-Audio - UMC404HD 192k
                         BEHRINGER UMC404HD 192k at usb-0000:00:1a.0-1.1, high speed
   

   
   So in my case the UMC404 is card number 2.
2. Start Jack and make sure Jack uses the UMC404, not one of your other soundcards:

   Code: Select all

   jackd -d alsa --device hw:2
   

   Qjackctl is a great little tool, but I do not support starting Jack with it. Start Jack manually by using command above.
3. Start Audacity (to make a test recording, I don't support anything else, and most definitely not commercial software). Choose Jack as so-called audio host like in the screenshot below. If everything's right then you can choose "4" under "Input channels" (like in the screenshot below) and every channel should produce output/a signal.

Troubleshooting:

• Kill all other (old) instances of Jack before (re)starting Jack:

  Code: Select all

  pkill -9 jackdbus
  killall jackd
  

• Kill PulseAudio:

  Code: Select all

  pulseaudio --kill
  

• See if pulseaudio and/or jack is running:

  Code: Select all

  ps -e|grep 'pulseaudio\|jackd'
  

Greetings

Endless search...

cat /proc/asound/cards
0 [PCH ]: HDA-Intel - HDA Intel PCH
HDA Intel PCH at 0xb3610000 irq 29
1 [U192k ]: USB-Audio - UMC404HD 192k
BEHRINGER UMC404HD 192k at usb-0000:00:14.0-3.2, full speed

arecord -l
**** List of CAPTURE Hardware Devices ****
card 0: PCH [HDA Intel PCH], device 0: ALC282 Analog [ALC282 Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: U192k [UMC404HD 192k], device 0: USB Audio [USB Audio]
Subdevices: 1/1
Subdevice #0: subdevice #0

jackd -R -P89 -dalsa -dhw:1 -r192000 -p128 -n3 -Xraw

jackd -R -P89 -dalsa -dhw:1 -r192000 -p128 -n3 -Xraw
jackdmp 1.9.11
Copyright 2001-2005 Paul Davis and others.
Copyright 2004-2014 Grame.
jackdmp comes with ABSOLUTELY NO WARRANTY
This is free software, and you are welcome to redistribute it
under certain conditions; see the file COPYING for details
no message buffer overruns
no message buffer overruns
no message buffer overruns
JACK server starting in realtime mode with priority 89
self-connect-mode is "Don't restrict self connect requests"
audio_reservation_init
Acquire audio card Audio1
creating alsa driver ... hw:1|hw:1|128|3|192000|0|0|nomon|swmeter|-|32bit
configuring for 192000Hz, period = 128 frames (0.7 ms), buffer = 3 periods
ALSA: final selected sample format for capture: 24bit little-endian in 3bytes format
ALSA: use 3 periods for capture
ALSA: final selected sample format for playback: 24bit little-endian in 3bytes format
ALSA: use 3 periods for playback
scan: added port hw:1,0,0 in-hw-1-0-0-UMC404HD-192k-MIDI-1
scan: added port hw:1,0,0 out-hw-1-0-0-UMC404HD-192k-MIDI-1
scan: opened port hw:1,0,0 in-hw-1-0-0-UMC404HD-192k-MIDI-1
scan: opened port hw:1,0,0 out-hw-1-0-0-UMC404HD-192k-MIDI-1

Thanks for your patience, just 2 channels are shown and audacity starts in 192k (, ardour just in 48k. Will conduct further tests. Pulse is restrained to 192k 24-bit, didn't kill it yet, ./asound.conf was wiped out, just in case.

After starting Audacity:

JackEngine::XRun: client = PortAudio was not finished, state = Triggered
JackAudioDriver::ProcessGraphAsyncMaster: Process error

Will continue digging...

Linuxmusician01 wrote:

Code: Select all

cat /proc/asound/cards

Output will look something like this:

Code: Select all

 0 [CMI8738        ]: CMI8738-MC6 - C-Media CMI8738
                      C-Media CMI8738 (model 55) at 0xd000, irq 20
 1 [PCH            ]: HDA-Intel - HDA Intel PCH
                      HDA Intel PCH at 0xf7c10000 irq 45
 2 [U192k          ]: USB-Audio - UMC404HD 192k
                      BEHRINGER UMC404HD 192k at usb-0000:00:1a.0-1.1, high speed

So in my case the UMC404 is card number 2.


[*]Start Jack and make sure Jack uses the UMC404, not one of your other soundcards:

Code: Select all

jackd -d alsa --device hw:2

I'd use the name instead of the index: hw:U192k, as the index can change between reboots.

Puseaudio constantly restarted in my system so:

sudo nano /etc/pulse/client.conf

and introducing the line:

autospawn = no

rebooted, ceased to have pulseaudio

started jack with:

jackd -d alsa --device hw:U192k

started Audacity with jack and system and only two channels are there.

If I start jackd with:

jackd -d alsa --device hw:U192k -i 4

jackdmp 1.9.11
Copyright 2001-2005 Paul Davis and others.
Copyright 2004-2014 Grame.
jackdmp comes with ABSOLUTELY NO WARRANTY
This is free software, and you are welcome to redistribute it
under certain conditions; see the file COPYING for details
no message buffer overruns
no message buffer overruns
no message buffer overruns
JACK server starting in realtime mode with priority 10
self-connect-mode is "Don't restrict self connect requests"
control open "hw:U192k-i" (No such device)
control open "hw:U192k-i" (No such device)
creating alsa driver ... hw:U192k-i|hw:U192k-i|1024|2|48000|0|0|nomon|swmeter|-|32bit
control open "hw:U192k-i" (No such device)
ALSA lib pcm_hw.c(_snd_pcm_hw_open) Invalid value for card
ALSA lib pcm_hw.c(_snd_pcm_hw_open) Invalid value for card
ALSA: Cannot open PCM device alsa_pcm for playback. Falling back to capture-only mode
Cannot initialize driver
JackServer::Open failed with -1
Failed to open server

Conclusion: alsa reports 2 channels to jack in spite alsamixer shows both front and rear mics plus mic1.

Please use the code function for better reading. Thank you. [/code]

Thanks khz.

So I killed jackd and went directly to arecord
Conclusion:

1. The alsa driver does not support 4 channel:
2. The alsa driver does not natively support rates higher than 48k. For higher rates you have to resort to plughw (not hw). It will even accept 4 channel but it records beat tones.

I seem to be stuck to 2 channels at 48kHz. Btw, my alsa version is: and:
It seems I have to open up the device to get the chipset name so as to go to:

ftp://ftp.iitb.ac.in/LDP/en/Alsa-sound/ ... ingle.html

and check if it is supported.

Greetings again

As I am essentially a stubborn mule, I checked hardware and took a vicious hub out of the way, now everything is working, either in USB2 or USB3. Including direct Alsa recording, Audacity, either in direct Alsa or jack mode, and Ardour, even at 192k rate.

Conclusion: Always connect your device first of all directly to a USB port before trying hubs...

Thank you for all your effort.

ajoliveira wrote:Greetings again

As I am essentially a stubborn mule, I checked hardware and took a vicious hub out of the way, now everything is working, either in USB2 or USB3. Including direct Alsa recording, Audacity, either in direct Alsa or jack mode, and Ardour, even at 192k rate.

Conclusion: Always connect your device first of all directly to a USB port before trying hubs...

Thank you for all your effort.

Wait a minute. What you're saying is that you had these troubles because you were using a USB hub? Never thought that might cause problems w/ USB audio devices, man! Thanks for the tip! Would have never asked if you were using a hub because I did not know that this can cause troubles.

Yes, Linuxmusician01, hubs are little devils. Now I have pulseaudio again just because of bloody firefox 60 which discontinued alsa support and everything still works.
Moving on to the measurements using an Audio Precision P1Plus. That will compensate you just a little for the effort.

I work around the Firefox ALSA issues by using PaleMoon browser. Unless they've recently dropped ALSA it should still work.
Chrome may also support it but since I don't use it I cannot be sure.

It also used to be possible to compile FF yourself with ALSA, but at some point they were going to remove it from the codebase. Dunno if that's happened yet.

Behringer soundcards have worked pretty well for me on Linux, a UCA202 & a UMC404. Typically the problems people experience with them are more to do with their particular hardware & software setups than due to problems with the actual soundcard. Many of the "premium" brands costing a lot more don't offer any real or imagined improvements. Obviously I'm not referring to pro-audio gear like MOTU, rather to some of these average quality soundcards which sell for premium prices and get hyped on forums by hipsters.

Unfortunately the Linux Audio learning curve is very steep and the pre-baked audio distros cannot satisfy everyone's particular requirements, so the experience quickly turns into a nightmare when weird issues are encountered and there isn't enough knowledge & experience to fix the problem.

khz wrote:Everything that was recognized with "not good" click on the following links "http://wiki.linuxaudio.org/wiki/system...." and read and change accordingly.
Then perform the quickscan again for checking.

Realtime kernel is optional, good to have but not mandatory. IMHO

Q: How to optimize my system for audio / midi ?

A: For some uses, it can be helpful to run a realtime (RT) kernel, but generally this isn't necessary anymore. See the following JACK FAQ entry: Do I need a realtime kernel to use realtime scheduling?

In Linux kernels 3.0 and later, many of the additionally required realtime patches have been incorporated as standard. For those who are recording Audio, a standard non-realtime kernel may be sufficient for your needs, and running Jack with a non realtime kernel will work fine.

You need to configure your system, however, to allow JACK to use realtime scheduling.

This wiki has some more articles and general information about real time operation.

Those who produce a lot of MIDI I/O, may want to use a Linux kernel with non-standard timer interrupt frequency (aka “tick rate”). The standard frequency is set at 250 Hz, and realtime kernels are generally set at 1000 Hz. You may still have to use a realtime kernel to obtain accurate MIDI playback using the higher timing setting.

https://wiki.linuxaudio.org/faq/start

If there are problems/questions to the individual points (not good >> "http://wiki.linuxaudio.org/wiki/system....") please ask here.

Hello all!!

Despite I've been out for awhile (involved in performing/teaching projects), this week I had the chance to record two nights of concerts during a trombone festival in Paraíba I've promoted.

Back to the topic of "QUICK SCAN" I've finally had the time to read and investigate most of the "NO GOOD" indications and here is my last scan done tonight:

== GUI-enabled checks ==
MAY 27 2018 @ 22:29

Checking if you are root... no - good
Checking filesystem 'noatime' parameter... 4.13.0 kernel - good
(relatime is default since 2.6.30)
Checking CPU Governors... CPU 0: 'performance' CPU 1: 'performance' CPU 2: 'performance' CPU 3: 'performance' - good
Checking swappiness... 10 - good
Checking for resource-intensive background processes... none found - good
Checking checking sysctl inotify max_user_watches... >= 524288 - good
Checking access to the high precision event timer... readable - good
Checking access to the real-time clock... readable - good
Checking whether you're in the 'audio' group... yes - good
Checking for multiple 'audio' groups... no - good
Checking the ability to prioritize processes with chrt... yes - good
Checking kernel support for high resolution timers... found - good

Kernel with Real-Time Preemption... not found - not good
Kernel without real-time capabilities found

For more information, see
http://wiki.linuxaudio.org/wiki/system_ ... ime_kernel

Checking if kernel system timer is high-resolution... found - good
Checking kernel support for tickless timer... found – good

== Other checks ==
Checking filesystem types... ok.
not found.
** Warning: no tmpfs partition mounted on /tmp
For more information, see:
- http://wiki.linuxaudio.org/wiki/system_ ... tion#tmpfs
- http://lowlatency.linuxaudio.org
** Set $SOUND_CARD_IRQ to the IRQ of your soundcard to enable more checks.
Find your sound card's IRQ by looking at '/proc/interrupts' and lspci.


I did them mostly tonight, but even with all the "no good" warnings, I've been using my Laptop for recording and doing mixing using both Ardour and MixBus (after installing the KX repos AMAZING stuff thank you!!) I'll try to fix the remaining warnings but the REAL TIME is something that I really have doubts. I also installed a low latency Kernel and did not notice a huge difference in terms of performance.
I'll do some test with latency/buffer/compensation and let you all know ok?

Again thank you for your support

I was also able to record a live recording with a very old mini pc (2008) Intel dual core running Mint with Ardour and Jack running (after installing the KX repos). I've recorded with all 4 channels @ 24Bits 96K