2 interfaces, 2 jackd, couple them

[bluebell]

As I wrote Ihave 2 interfaces and need all of their outputs.

My first attempt with alsa_out on the 2nd interface worked but when it ran with 48000 then starting Audacity broke it.

So I am using another setup now:

jackd (on interface 1)
jackd -n monitor (on interface 2)

They can be coupled:

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jack-stdout -q -e float "Calf Studio Gear:Equalizer 12 Band Out #1" "Calf Studio Gear:Equalizer 12 Band Out #2" | JACK_DEFAULT_SERVER=monitor jack-stdin -q -e float system:playback_1 system:playback_2

Advantage: seems to be very stable
Disadvantage: additional latency on the signal that goes to the 2nd jackd

[bluebell]

I found zita-njbridge as well. Latency is much lower than using jack-stdout/in for coupling 2 jackd.

[khz]

• ALSA - MultipleCards >> http://alsa.opensrc.org/MultipleCards , "~/.asoundrc" >> https://www.alsa-project.org/main/index ... el_devices
• Jack - multiple soundcards >> http://jackaudio.org/faq/multiple_devices.html
• JACK - alsa_in and alsa_out >> https://github.com/jackaudio/jackaudio. ... _AlsaInOut
• Multiple USB Sound Cards >> viewtopic.php?p=113047#p113047
• Zita-njbridge >> https://kokkinizita.linuxaudio.org/linuxaudio/
• Pulseaudio >> https://www.freedesktop.org/wiki/Softwa ... #index31h3
• NetJackTrip >> viewtopic.php?p=95720#p95720
• audioadapter >> see below

I did it with the "audioadapter" (4.), all in one qjackctl (Jack2) instance:
The boot sequence of the (PCI/PCIe) sound cards (Audio/MIDI interfaces) is important so that the same prerequisite (sound card sequence) is always given when the computer is started.

1. Define first sound card. What's in the computer:

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cat /proc/asound/cards

edit:

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nano /etc/modprobe.d/alsa.conf

Example, take what "/proc/asound/cards" says!

- same drivers :

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    options snd-ymfpci index=0
    options snd-hda-intel index=1,2

- different drivers:

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    options snd-emu10k1 index=0
    options snd-via82xx index=1

Festlegen des Default-Device https://wiki.archlinux.de/title/Advance ... ult-Device

2. Save settings in alsamixer (as root), make settings for each sound card separately and save them:

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alsactl store

3. Programs take certain sound card, edit ~/.asoundrc:

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nano ~/.asoundrc

to (example):

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    defaults.ctl.card 0
    defaults.pcm.card 0
    defaults.timer.card 0

# https://wiki.linuxaudio.org/faq/start#q ... sa_drivers


4. Merge multiple sound cards into qjackctl:

A)
Setup > Advanced: "Server Prefix" select "jackdmp", "H/W Monitor", "H/W Meter", "Soft Mode", (RT) "Priority (0 - 99)" (https://linux.die.net/man/1/jackd)

In qjackctl setting: Setup >> Options >> [X] Run script at startup

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jack_load -i "-d hw:1 -i16 -o16" 9632 audioadapter && jack_bufsize 128 &

-d <-- you want to merge (sound card) (hw:1/hw:0/...)

-i <-- your number of sound card inputs
-o <-- your number of sound card outputs
9632 <-- as you want to call them (jackport name)
jack_bufsize <-- set the JACK buffer size
# https://github.com/jackaudio/jackaudio. ... r_NetJack2

• ==>
  • Synchronisation your cards: with SPDIF, word clock or an internal sync port.
    Is there anything about this in the manual of your used sound card?
    (Personal, unverified and possibly false assumption: "? I think you need to synchronize the soundcards over one of the possibilities mentioned above. If it doesn't offer your sound card it won't work. The audioadapter does not do this. ?") IMHO
  • https://en.wikipedia.org/wiki/Word_clock
  • https://www.rme-audio.de/english/techinfo/sync.htm (German: https://www.rme-audio.de/techinfo/sync.htm); (https://www.rme-audio.de/download/hdspeaio_e.pdf)
  • http://manual.ardour.org/synchronization/

B)
Alsa2jack (MIDI):
install:

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a2jmidid

Setup >> Options >> [X] Run script after startup

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sleep 5 && a2jmidid -e &

• • Code: Select all

    sleep 5 && a2j -e &

    Use 'a2j -e' instead of 'a2jmidid -e'.
    The 1st option enables dbus.

    viewtopic.php?p=74397#p74397
  # sleep - delay a specified time period

##

https://linux.die.net/man/1/jackd
==> Install "a2jmidid" and "pulseaudio-module-jack".

In qjackctl Setup:
Jack with a2jmidid and pulseaudio:

• Setup >> Settings > Parameters > MIDI Drivers [none]
• Setup >> Settings > Advanced > "Server Prefix" select "jackdmp" (multi-processor machines https://github.com/jackaudio/jack2/blob/master/README)
• Setup >> Settings > Advanced > [X] Soft Mode
• Setup >> Settings > Advanced > (RT) Priority (0 - 98)
• Setup >> Options > [X] Run script after startup: "sleep 5 && a2j -e &" (a2j enables dbus) or "sleep 5 && a2jmidid -e &" (without dbus)
• Setup >> Misc > [ ] ([none]) Provide support for ALSA sequencer
• Setup >> Misc > [X] D-Bus interface Activation
• Setup >> Misc > [X] Enable JACK D-Bus Interface

Optional:

• Setup >> Settings > Advanced > [X] H/W Monitor
• Setup >> Settings > Advanced > [X] H/W Meter

Jack without pulseaudio:

• Setup >> Options > [X] Execute script at startup: "systemctl --user stop pulseaudio.socket && systemctl --user stop pulseaudio.service"
• Setup >> Options > [X] Run script after shutdown: "killall -9 jackdbus && systemctl --user start pulseaudio.socket && systemctl --user start pulseaudio.service")

[bluebell]

>> http://jackaudio.org/faq/multiple_devices.html
I did it with the "audioadapter"

Interesting but I can't find the word "audioadapter" there. Where did you find docs about the audioadapter feature?

[khz]

No idea anymore. I was desperate and had searched the Internet for a solution. There were not much contributions/documentation and if then very different to it.
After several failed attempts of the different possibilities, the "audioadapter" worked. I documented it right away in the net because I and also other users maybe later want to have it easier.

1.) On the fast even that found: https://github.com/jackaudio/jackaudio. ... r_NetJack2
2.) jack with multiple cards?: viewtopic.php?t=16062
3.) That's my original post: https://www.sequencer.de/synthesizer/th ... st-1156186

[bluebell]

No idea anymore. I was desperate and had searched the Internet for a solution. There were not much contributions/documentation and if then very different to it.
After several failed attempts of the different possibilities, the "audioadapter" worked. I documented it right away in the net because I and also other users maybe later want to have it easier.

1.) On the fast even that found: https://github.com/jackaudio/jackaudio. ... r_NetJack2
2.) jack with multiple cards?: viewtopic.php?t=16062
3.) That's my original post: https://www.sequencer.de/synthesizer/th ... st-1156186

Very interesting. I'll play around with it.

The solution "coupling 2 jackd with zita-j2n and zita-n2j" has the disadvantage that zita-j2n crashes when I change the bufsize of the jackd where it's connected to. Else it would be nearly perfect (low latency).

[khz]

I'm not sure right now if the "audioadapter" command is important. "audioadapter" may be superfluous.
The "jack_load" command is important!

Once you have found the working setting, please document it comprehensibly for other users.
Good documentation is so helpful!

[bluebell]

I'm not sure right now if the "audioadapter" command is important. "audioadapter" may be superfluous.
The "jack_load" command is important!

Once you have found the working setting, please document it comprehensibly for other users.
Good documentation is so helpful!

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jack_load -i "-d hw:2 " onboard audioadapter

onboard: the name as seen in qjackctl, can be chosen freely
audioadapter: the resampling module, name has to be exactly like this

That does the job on one of my machines. Changing bufsize seems to work. This is the most promising attempt.

In a few days I can test it on the other machine with 2 usb-interfaces.

[bluebell]

The audioadapter internal client is the best solution for me. I can't believe that it is so badly documented, especially the internal clients audioadapter and netadapter.

Scarlett 2i4:
jackd -p512 -t2000 -dalsa -r48000 -p128 -n2 -Xseq -dhw:1

Add Scarlett 2i2:
jack_load -i "-dhw:2 -p64 -r48000" 2i2 audioadapter

[khz]

Now also included in the grandiose GNU/LINUX@AUDIO ~ /Wiki $ Howto.Info.


Wish/Request/: @qjackctl @Cadence/Cadence-JackSettings
Offer audio and net adapters under "Settings >> Parameters"?
Would make sense IMHO :freedom: and so

[didgewind]

Now also included in the grandiose GNU/LINUX@AUDIO ~ /Wiki $ Howto.Info.

Thanks, but, where is this?

[merlyn]

Thanks, but, where is this?

It's in @khz's signature.

[bluebell]

Sorry but after some time I wasn’t happy with any solution. Audioadapter increased its buffer without reason, alsa_out crashed when I started Audacity and so on.

I bought a MOTU UltraLite AVB. It has a good internal mixer (with EQ) that can be controlled with a web browser. Problem solved.

[merlyn]

Audioadapter increased its buffer without reason,

I see. Is that while it was playing back? Or do you mean one day the latency was OK, then the next day the latency was bad?

I've used audioadapter and it seems pretty good. I haven't done involved testing. I use it with jackdbus so I use jack_control to set parameters.

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$ jack_control ip audioadapter
--- get internal parameters (type:isset:default:value)
            capture: Provide capture ports.  Optionally set device (str:notset:none:none)
            playback: Provide playback ports.  Optionally set device (str:set:none:none)
            device: ALSA device name (str:set:hw:0:hw:GR55)
            rate: Sample rate (uint:set:48000:44100)
            periodsize: Period size (uint:set:512:64)
            nperiods: Number of periods of playback latency (uint:notset:2:2)
            duplex: Provide both capture and playback ports (bool:set:True:True)
            in-channels: Number of capture channels (defaults to hardware max) (sint:notset:0:0)
            out-channels: Number of playback channels (defaults to hardware max) (sint:set:0:2)
            quality: Resample algorithm quality (0 - 4) (uint:set:0:0)
            ring-buffer: Fixed ringbuffer size (uint:set:32768:512)
         

The ringbuffer seems to affect latency the most. The default value is crazy high and I didn't see a value set for that in the command you posted above.

[bluebell]

Audioadapter increased its buffer without reason,

I see. Is that while it was playing back? Or do you mean one day the latency was OK, then the next day the latency was bad?

I've used audioadapter and it seems pretty good. I haven't done involved testing. I use it with jackdbus so I use jack_control to set parameters.

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$ jack_control ip audioadapter
--- get internal parameters (type:isset:default:value)
            capture: Provide capture ports.  Optionally set device (str:notset:none:none)
            playback: Provide playback ports.  Optionally set device (str:set:none:none)
            device: ALSA device name (str:set:hw:0:hw:GR55)
            rate: Sample rate (uint:set:48000:44100)
            periodsize: Period size (uint:set:512:64)
            nperiods: Number of periods of playback latency (uint:notset:2:2)
            duplex: Provide both capture and playback ports (bool:set:True:True)
            in-channels: Number of capture channels (defaults to hardware max) (sint:notset:0:0)
            out-channels: Number of playback channels (defaults to hardware max) (sint:set:0:2)
            quality: Resample algorithm quality (0 - 4) (uint:set:0:0)
            ring-buffer: Fixed ringbuffer size (uint:set:32768:512)
         

The ringbuffer seems to affect latency the most. The default value is crazy high and I didn't see a value set for that in the command you posted above.

The ringbuffer increased after some minutes of operation to ridiculous large values.