RME Digiface USB and PipeWire

My work laptop provided by SURF, a Dell Pro 14 Premium, runs a custom Dell Ubuntu 24.04 ISO since otherwise half of the peripherals don’t work. Ubuntu 24.04 comes with PipeWire so I was curious if the Digiface would work with such a setup. Well, it does, albeit with some quirks.

First of all PipeWire also doesn’t get the outputs right. With the help of WirePlumber I could map channels 33 and 34 of the phones output to Front Left and Front Right. But then the next issue popped up: weird garbled, phasing, ringing audio. When using ALSA directly with Ardour this happened too so it’s not a PipeWire issue. Tried tweaking things first and also tried a recent Liquorix kernel. To no avail. Then I started trying out different buffer sizes using pw-metadata and noticed the phasing, ringing sound would go away when changing the buffer size. And I also noticed that the garbled sound would come back if I used a buffer size higher than 64. So with the use of WirePlumber I also set that now. Another owner reported that setting a higher sample rate also alleviates the issue. I still like to figure out though why I can’t use higher buffer sizes. I think it’s related to the USB topology of this laptop as I don’t have this issue on my own laptops.

The WirePlumber configuration file to remap the phones outputs can be found here: https://codeberg.org/autostatic/gists/src/branch/main/99-rme-digiface-usb.lua. Ubuntu 24.04 uses WirePlumber < 0.5.0 so it’s still in Lua format. You can place this file in ~/.config/wireplumber/main.lua.d/. And it starts with 99- for a reason, the interwebs suggest a lower number but then it gets overwritten by global configuration files.

Below the WirePlumber configuration file that I use on the Dell Pro 14 Premium with an extra property to work around the garbled audio issue.

rule = {
  matches = {
    {
      { "node.name", "matches", "alsa_output.usb-RME_Digiface_USB__23800125__7C01A52EA9692C8-00.*" },
    },
  },
  apply_properties = {
    ["audio.position"] = "AUX0,AUX1,AUX2,AUX3,AUX4,AUX5,AUX6,AUX7,AUX8,AUX9,AUX10,AUX11,AUX12,AUX13,AUX14,AUX15,AUX16,AUX17,AUX18,AUX19,AUX20,AUX21,AUX22,AUX23,AUX24,AUX25,AUX26,AUX27,AUX28,AUX29,AUX30,AUX31,FL,FR",
    ["api.alsa.period-size"] = 64,
  },
}

table.insert(alsa_monitor.rules,rule)
RME Digiface USB and PipeWire

Now what does rtcirqus actually do?

To answer that question let’s assume the following:

  • You want to do real-time audio, for instance you would like to do an overdub in a song project by using a MIDI keyboard and a soft synth.
  • You’re using a USB audio interface.
  • You’re experiencing xruns when using lower buffer sizes (64 or even lower) that result in crackling audio.

Basically all components of a desktop computer or a laptop have an IRQ. Let’s say IRQs are some sort of communication lines between the brains of your computer (the CPU) and those components. All these communication lines are more or less the same as in, none of these communication lines has priority over the other. But if you suffer from bad audio quality caused by xruns then giving the communication line that is connected to your USB device more priority than the rest could alleviate the audio issues.

This is where rtcirqus could be helpful. What rtcirqus does is detecting your USB audio interface when it gets connected, then it figures out what the IRQ of the USB bus is that your audio interface is connected to and raises the priority of the IRQ attached to that USB bus. In other words, rtcirqus gives the communication line between the USB bus that your audio interface is connected to and the CPU of your computer a higher priority. This higher priority translates into other communication lines being throttled and giving the communication line of our USB audio interface more room to transfer and receive its audio data.

rtcirqus needs to be able to prioritize that communication line though and this is where the threadirqs kernel option is for, it creates so-called threads of all available IRQs on a computer. Threads are more or less like processes running on your computer, with threadirqs enabled you can list them as such with a command like ps -eLo cmd | grep '^[irq'. So make sure this kernel option has been added to the kernel command line. In case you’re using a real-time kernel this isn’t needed as a real-time kernel uses threaded IRQs by default.

So if you find yourself in such a situation, head over to https://codeberg.org/autostatic/rtcirqus and give rtcirqus a try, maybe it helps you achieving lower latencies with your USB audio interface while retaining a clean audio output signal.

Now what does rtcirqus actually do?

RME Digiface USB on Linux

With the release of the 6.12 kernel the RME Digiface USB is supported under Linux. Since this is a very portable device with lots of IO (32 in and 34 out) it got on my radar as a possible candidate for making recordings at my rehearsal space. It took a while before I stumbled upon a second hand unit but in the end I managed to find one nearby. Hooked it up to my notebook with an up-to-date Liquorix kernel and it worked out of the box.

A lot of thanks go out to Asahi Lina for adding support for the RME Digiface USB to the 6.12 kernel. She even added the possibility to control the output format and clock source and to monitor sample rates, input formats, input sync statuses and the current sync source.

Screenshot of AlsaMixer showing all the controls available for the RME Digiface USB
Screenshot of AlsaMixer showing all the controls available for the RME Digiface USB

Have yet to test the Digiface USB with more than one ADAT device but with one device (a Behringer ADA8200) everything seems to work really well. Did some round-trip latency tests and with a buffer size of 16 samples and putting TotalMix in DAW mode (which had to be done on a different OS unfortunately but saved another 0.2 ms) I got it down to approximately 4.2 ms. Systemic latency with a buffer size of 16 samples and a sample rate of 48 KHz is 2 ms so somewhere a bit more than 2 ms of additional latency gets added. I think about 1.4 ms gets added by that converters if I did my maths right. According to the specs of the converters they add 22 samples (10 in and 12 out at 48 KHz). That leaves about 0.8 ms unaccounted for. Probably the USB bus buffer or something in the driver stack. Well, no real deal-breaker for me.

This image has an empty alt attribute; its file name is image-2.png
Ardour round-trip latency measurement with a buffer size of 32 and a 48KHz sample rate
Ardour round-trip latency measurement with a buffer size of 32 and a 48KHz sample rate

Because I’m still on Debian 12 I couldn’t fully use the device with PulseAudio though. PulseAudio only supports devices with up to 32 channels in or out and since the Digiface has 34 output channels PulseAudio refused to output any audio through it. As a fix I added the following to the ~/.asoundrc file in my home directory:

pcm.snd_rme_digiface_usb {
    type multi
    slaves.rme.pcm "snd_output"
    slaves.rme.channels 34
    bindings.0.slave rme
    bindings.0.channel 32
    bindings.1.slave rme
    bindings.1.channel 33
}

pcm.snd_output {
    type hw
    card USB23800125
}

To make PulseAudio pick up this extra snd_rme_digiface_usb device I added the following to ~/.config/pulse/default.pa:

# RME Digiface USB
load-module module-alsa-sink device=snd_rme_digiface_usb

And now I can select the Digiface as an output source and use the phones output in a desktop session.

Screenshot of pavucontrol with the RME Digiface USB right in the middle
Screenshot of pavucontrol with the RME Digiface USB right in the middle

Another big bonus is the custom pouch my daughter made for it. I can now toss it in my bike bag without having to worry about dings or scratches.

RME Digiface USB tucked in a custom cotton pouch
RME Digiface USB tucked in a custom cotton pouch
RME Digiface USB on top of its custom cotton pouch
RME Digiface USB on top of its custom cotton pouch

RME Digiface USB on Linux

Turntables II

Ran into a SL1210MK2 for about the same price as a new Super OEM on a Dutch second hand market place so went for that. And it was only a 30km drive. It’s quite a battered specimen but the core parts work properly. I did make an appointment with a repair service to take a good look at it. The tone arm connector is worn out and has too much slack, the target light doesn’t work and I’d like to have the RCA cables replaced with better quality ones. And it has some kind of quick start cable that I would like to have removed. And the dust cover is pretty tatty and missing its hinges.

It came with no headshell, the seller offered the original but I declined since I already have an unused original headshell lying around and I was planning on putting a premounted Ortofon 2M Red on it anyway. Got that one in already, together with a nice Tonar Cork ‘n Rubber mat. Also ordered a Rega Fono Mini A2D phono preamp. This model comes with a builtin ADC with a Texas Instruments PCM2900C chipset, so 48kHz/16-bit. This can be very handy for spinning with time coded vinyl, no need to add an extra audio interface to the chain. And yes, already played around with that using Mixxx which works amazingly well.

Turntables II

Turntables

What are the options these days when it comes to buying a new turntable that comes closest to what I already have without having to pay over 1000 Euros? So direct-drive, sturdy, reliable and decent sounding? Enter the Hanpin Super OEMs as they are called. These are the higher level Technics SL-1200 copies that are being branded under a myriad of different names:

  • Pioneer PLX-1000
  • SYNQ XTRM1
  • Reloop RP-7000 MK2
  • Reloop RP-6000 MK5 S
  • Omnitronic 5220/5250
  • Audio-Technica AT-LP1240
  • Mixars STA
  • Stanton ST-150

These are apparently all the same Hanpin DJ-5500 model with slightly varying features. Out of the above list only the Pioneer, SYNQ and Reloop RP-7000 MK2 models are currently available. But which one to choose? Inching toward the Reloop since that turntable has a height adjustable tone arm and personally I think it looks the best of all three contenders. But then this DJ-5500 design is like over 15 years old and apparently has some issues (hum with certain elements, anti-skating not working correctly) so maybe I should just go for another second hand SL-1200?

Argh, why didn’t I buy both SL-1200s back when I bought mine, together they were 700 Euros…

Turntables

Bye cable, hello glass

Bit the bullet a few months ago and decided to go for a glass fiber connection. So after 25 years of cable internet from Casema/Ziggo we’re now hooked up to the optic fiber universe. Let’s see if it turned out to be the right decision.

Last year our whole region got a glass fiber network and getting your household hooked up was free of charge initially. But that changed now and just before getting charged for a connection I applied to get it done. Also the new provider has cheaper subscriptions, multiple TV channels with Ziggo were not working properly and with a glass fiber connection we will be ready for the future. I did look up to it though because for getting connected some work had to be done in our front yard. And even though it turned out ok-ish in the end I wasn’t really happy with the unannounced ventures in our front yard. Quite a part of the garden had to be opened up twice and everything isn’t really put back the way it was. Definitely no gardeners but that’s completely understandable.

Today the last things were taken care of, the media converter has been installed together with the router of the provider. The employee of the provider made sure everything worked properly and was done in less than an hour. After he left I only had to pull the network cable of my home network out the new router, stick it in the media converter, configure IPTV for VLAN 300 and tada, working internet with my own router setup (two Asus RT-AC68U’s in an AiMesh configuration). Took out the new router of the meter cupboard and loaded my stash of beer back in.

And is it faster? Partially. I already had a 1Gbit/s connection with Ziggo, but that was an asynchronous connection so upload is way faster now, about 8 times. Other pro is that the media converter is way smaller and probably draws less power. Other than that nothing really changed after the media converter which I like, fortunately there was no need to overhaul my whole home setup.

Bye cable, hello glass

New notebook – The verdict

Well, Phoronix did a review of a similar machine and apparently it’s far from being a slouch and also has the fastest integrated GPU currently available. More on that GPU later. So no regret when it comes to those performance benchmarks. Actually no regret at all, so far the notebook performs really well.

There are some more things worth mentioning that add up to the positive verdict besides all the pros I already mentioned in my earlier posts. There is the battery life which is still pretty good given the performant and power greedy CPU. It can run for hours when idling. When running Ardour it’s done in about two hours though but then I work with all the sluices wide open. But it charges pretty fast. Another thing that struck me is that the notebook is so much quieter than the old one. And the keyboard is just really nice now that I got a bit more used to it. And I managed to map the last media key, the stop one, to something useful with my old friend xdotool. Mapped this media key to the “stop/cancel” keycode using udev and added a keyboard shortcut in XFCE that gets triggered by this keycode and that executes a small script that looks like this:

#!/bin/bash

xdotool search --classname ardour_ardour key --delay 100 h space h

Now if Ardour is running and I press the Stop media key, xdotool sends the key sequence “h space h” to Ardour with a delay of 100ms between the key strokes. “h” sets the playhead to start, “space” stops the transport and another “h” to be sure the playhead is at the start position. Only thing that I’d like to add is some kind of detection if transport is running or not.

Also managed to achieve an even lower latency with my USB interface by adding the option implicit_fb=1 to the snd-usb-audio kernel module. This not only gets rid of the kernel ring buffer getting flooded with warnings but it also results in clean audio at 32*3/48, so 2ms of nominal latency. So it’s on par with my old notebook, albeit with some more headroom. Lower doesn’t seem to be possible, it results in slowed down, distorted audio.

So would I advise everyone doing Linux audio to get this notebook or a similar spec’d one? Well, there’s this GPU that still seems to be a bit too new, too shiny and too fast for the kernel I’m currently running (6.7.2) so I’m getting reliable crashes with software like OwlPlug and occasional crashes when connected via HDMI to a second screen. But it’s tolerable and it will probably get sorted out sooner or later. Other than that, this thing flies and hopefully I can do another decade with this machine.

Edit: worked around the GPU crashing by copying /usr/share/X11/xorg.conf.d/10-amdgpu.conf to /etc/X11/xorg.conf.d/10-amdgpu.conf and disabling hardware acceleration by adding the line Option "Accel" "off" to it. Content looks like this:

Section "OutputClass"
        Identifier "AMDgpu"
        MatchDriver "amdgpu"
        Driver "amdgpu"
        Option "HotplugDriver" "amdgpu"
        Option "Accel" "off"
EndSection

After restarting Xorg I can now open OwlPlug without issues.

Edit: GPU crashes have been resolved by installing the firmware-amd-graphics from the bookworm-backports repository (version 20241210-1~bpo12+1) and by setting /sys/class/drm/card0/device/power_dpm_force_performance_level to high instead of auto.

Edit: Looks like USB audio performance has been boosted with more recent kernels (6.16.x), I can now use a buffer size of 16 without audio slowing down.

New notebook – The verdict

New notebook – Doing real-time audio

Configuring this new machine took some effort but it can now run reliably with a few milliseconds of latency. First thing I did was installing a liquorix kernel as I have a good experience with those. Added the threadirqs kernel option to /etc/default/grub and updated the Grub configuration with sudo update-grub. After a reboot I was greeted with threaded IRQ’s.

Next step was to prioritize the desired IRQ’s, highest prio for the USB bus to which my audio interface is connected and also a high prio for the onboard audio. I decided to go the custom route as the tool I normally use for this, rtirq, prioritizes all USB threads and I only want the USB threads prioritized that do the audio work. Additional challenge was that these IRQ’s change on every boot. Concocted the following script snippet.

#!/bin/bash

# Prioritize USB port with sound card attached
# Since IRQ's change on every boot figure out IRQ dynamically

# Set maximum priority of the IRQ thread
prio=90

# Next IRQ thread found will get a priority
# decreased with the value set below
prio_step=5

# System paths to look for information
proc_path=/proc/asound
sys_pci_bus_path=/sys/class/pci_bus

# The logic - A for loop that does the following:
# * Iterates through all cards that are set by ALSA
# * Determines if it's an USB card
# * Sets priority on IRQ thread if this is the case,
    lowest card number gets the highest priority
for card_number in $(awk '/\[.*\]/ {print $1}' /proc/asound/cards); do

  if [ -e "$proc_path/card$card_number/usbid" -a -e "$proc_path/card$card_number/stream0" ]; then
    snd_dev_card=card$card_number
    snd_dev_pci_bus_ref=$(grep -Eo "usb-[^[:space:],-]+" $proc_path/$snd_dev_card/stream0 | sed "s/usb-\(.*\)/\1/")
    snd_dev_pci_bus_ref_short=$(awk -F ':' '{print $1":"$2}' <<<$snd_dev_pci_bus_ref)
    snd_dev_irq=$(cat $sys_pci_bus_path/$snd_dev_pci_bus_ref_short/device/$snd_dev_pci_bus_ref/irq)
    snd_dev_irq_pid=$(pgrep $snd_dev_irq-xhci)

    chrt -f -p $prio $snd_dev_irq_pid

    prio=$((prio-prio_step))
  fi

done

This snippet assumes the card numbers are set properly by assigning each card their own index value through the snd-usb-audio kernel module. This can be done with a file in /etc/modprobe.d/, i.e. /etc/modprobe.d/audio.conf. For my USB devices the relevant line in this file looks like this:

# RME Babyface, Edirol UA-25, Akai MPK Mini, Arturia BeatStep Pro, Behringer BCR2000
options snd-usb-audio index=0,1,5,6,7 vid=0x0424,0x0582,0x09e8,0x1c75,0x1397 pid=0x3fb7,0x0074,0x007c,0x0287,0x00bc

So my RME Babyface gets the lowest index (card number) and thus the highest real-time priority.

For onboard audio the situation was a bit trickier. The Lenovo comes with three different audio devices:

  • Onboard audio, speakers and TRRS jack
  • Digital audio, HDMI
  • Onboard mic

I wanted to index all properly so they don’t get in the way of my USB devices. For the onboard audio and HDMI this was no problem, adding the following line to /etc/modprobe.d/audio.conf was enough:

# Onboard audio
options snd-hda-intel index=10,11

Unfortunately you can’t discern between multiple devices with the snd-hda-intel driver and in my case both cards also have no model name so they show up as HD-Audio Generic cards with ID names Generic and Generic_1. Not very helpful. Luckily you can assign ID names dynamically after boot so I used that to give each card a proper ID name:

# Assign proper ID's to onboard devices
# Device with vendor ID 1002 is HDMI
# Device with vendor ID 1022 is onboard audio
for card in card{10,11}; do

  if grep -q 1002 /sys/class/sound/$card/device/vendor; then
    echo -n HDMI > /sys/class/sound/$card/id

  elif grep -q 1022 /sys/class/sound/$card/device/vendor; then
    echo -n ALC257 > /sys/class/sound/$card/id

    # Prioritize IRQ of onboard audio
    snd_dev_irq=$(cat /sys/class/sound/$card/device/irq)
    snd_dev_irq_pid=$(pgrep $snd_dev_irq-snd_hda_intel)

    chrt -f -p $prio $snd_dev_irq_pid
  fi

done

Now both cards can be used with their ID names (so hw:ALC257 for instance) and the onboard audio device gets prioritized with the $prio value set earlier on. Now the only culprit remaining was the onboard mic.

The onboard mic, which gets ID name acp63, is driven by a kernel module with the name of snd-soc-ps-mach. Now this module doesn’t take the index parameter so enter the slots parameter for the top level snd kernel module. With this parameter you can set which slot gets assigned to a specific driver. So added the following to /etc/modprobe.d/audio.conf:

# Onboard mic
options snd slots=,,,,,,,,,,,,snd-soc-ps-mach

And voilĂ , this is what arecord -l now thinks of it:

$ arecord -l
**** List of CAPTURE Hardware Devices ****
card 0: Babyface2359686 [Babyface (23596862)], device 0: USB Audio [USB Audio]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 11: ALC257 [HD-Audio Generic], device 0: ALC257 Analog [ALC257 Analog]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 12: acp63 [acp63], device 0: DMIC capture dmic-hifi-0 []
  Subdevices: 1/1
  Subdevice #0: subdevice #0

Awesome, full control again! Everything properly named and prioritized. But how does this perform?

With my RME Babyface I can go down to 64 frames/period and 3 periods at 48kHz, so that’s a nominal latency of 4ms. And it’s rock solid at this setting:

$ ./xruncounter -m

******************** SYSTEM CHECK *********************

    Sound Playback: USB-Audio - Babyface (23596862)
     Sound Capture: USB-Audio - Babyface (23596862)
      Graphic Card: Advanced Micro Devices, Inc. [AMD/ATI] Phoenix1 (rev c7)
Operating System: Debian GNU/Linux 12 (bookworm)
          Kernel: Linux 6.6.11-1-liquorix-amd64
    Architecture: x86-64
               CPU: AMD Ryzen 7 7840HS with Radeon 780M Graphics

***************** jackd start parameter ****************

    /usr/bin/jackd -P80 -S -dalsa -dhw:Babyface2359686 -r48000 -p64 -n3 -Xseq

********************** Pulseaudio **********************

    pulse is not active

********************** Test 8 Core *********************

Samplerate is 48000Hz 
Buffersize is 64 
Buffer/Periods  3
jack running with realtime priority 80
Xrun 1 at DSP load 83.76% use 3.56ms from 1.33ms jack cycle time
Xrun 2 at DSP load 92.76% use 3.32ms from 1.33ms jack cycle time
Xrun 3 at DSP load 87.17% use 2.48ms from 1.33ms jack cycle time
Xrun 4 at DSP load 92.11% use 1.15ms from 1.34ms jack cycle time
Xrun 5 at DSP load 95.91% use 3.29ms from 1.33ms jack cycle time
Xrun 6 at DSP load 97.95% use 1.85ms from 1.33ms jack cycle time
in complete 6 Xruns in 16809 cycles                                  
first Xrun happen at DSP load 83.76% in cycle 16112
process takes 3.56ms from total 1.34ms jack cycle time

On my old BTO I could go lower though, it would run at 32 frames/period and 3 periods at 48kHz with clean audio but the Lenovo is limited to 64 frames/period. If you try to go lower you will get distorted audio and the kernel ring buffer will fill up with messages like below:

[11883.067551] retire_capture_urb: 1338 callbacks suppressed
[11883.083397] xhci_hcd 0000:66:00.4: WARN Event TRB for slot 1 ep 5 with no TDs queued?

Eventually I’ll dive deeper into this but for now I’m OK with running at 64 frames/period. The onboard audio runs too at very low settings but provides way less room to do anything useful. But just about enough to run some soft synths and MIDI input. As a comparison, here’s the output of xruncounter for the onboard audio. In my case onboard audio likes period sizes higher than 3 best. And no full duplex at this setting, playback only.

$ ./xruncounter -m

******************** SYSTEM CHECK *********************

      Graphic Card: Advanced Micro Devices, Inc. [AMD/ATI] Phoenix1 (rev c7)
Operating System: Debian GNU/Linux 12 (bookworm)
          Kernel: Linux 6.6.11-1-liquorix-amd64
    Architecture: x86-64
               CPU: AMD Ryzen 7 7840HS with Radeon 780M Graphics

***************** jackd start parameter ****************

    /usr/bin/jackd -P80 -S -dalsa -dhw:ALC257 -r48000 -p32 -n6 -Xseq -P

********************** Pulseaudio **********************

    pulse is not active

********************** Test 8 Core *********************

Samplerate is 48000Hz 
Buffersize is 32 
Buffer/Periods  6
jack running with realtime priority 80
Xrun 1 at DSP load 97.32% use 0.40ms from 0.67ms jack cycle time
in complete 1 Xruns in 5781 cycles                                  
first Xrun happen at DSP load 97.32% in cycle 5527
process takes 0.40ms from total 0.67ms jack cycle time

So in my opinion the Lenovo performs pretty well. I do need to run these tests on my old BTO too to find out how much performance I’ve gained. And if I could find a way to work around or find a solution for those xhci_hcd warnings so that I can go even lower then that would be terrific. It could very well be a limitation of the USB implementation of this notebook but I can live with that as it runs really stable at sub 5ms latencies.

New notebook – Doing real-time audio

New notebook – The last bits

Sorted out the last bits that didn’t work smoothly yet.

WiFi

WiFi connection was intermittent and slow. lspci had this to say about it:

01:00.0 Network controller: MEDIATEK Corp. MT7922 802.11ax PCI Express Wireless Network Adapter

There are numerous reports about this controller not working, almost all firmware related. The Liquorix kernel I’m running has the latest firmware so that was not the issue. Then I stumbled on a report related to power saving. Disabled that through NetworkManager by adding an extra configuration file in /etc/NetworkManager/conf.d with the following lines in it:

[connection]
wifi.powersave = 2

WiFi is now stable and fast.

Keyboard

Backlight

This can be controlled through sysfs or D-Bus but the system thinks there are only three settings: off (0), dimmed (1) or bright (2). But my keyboard thinks there are four settings, the fourth being what I would call “responsive bright”. This setting disables the backlight after a minute of inactivity and responds to key strokes by enabling backlight again to the bright setting. You can toggle this setting on the keyboard itself with Fn+Space but unfortunately the system does not know about this. I’ve worked around this by using a tiny daemon that listens for key strokes and enables the backlight and disabling it again when there’s no activity for a minute. There are several solutions for this, I settled for kbd_backlight_ctrl because it works and it’s just a few lines of C. Can’t set the brightness though but since it’s just a few lines of code I see this as an opportunity to improve my non-existent C skills.

Media keys

The keyboard has four dedicated media keys, play/pause, stop, previous and next. These work perfectly with a media player like VLC but Ardour does not recognize them. By creating supplementary udev hardware database entries I remapped the media keys so they’re now also functioning in Ardour which is quite neat. Only the stop key has no purpose yet as Ardour has no shortcut for what I want to map it to, stopping and going back to the start marker. Maybe I could create something myself through Ardour’s Lua scripting engine.

The hardware database entries look like this:

evdev:input:b0011v0001p0001*
 KEYBOARD_KEY_a2=space
 KEYBOARD_KEY_90=left
 KEYBOARD_KEY_99=right

These went into /etc/udev/hwdb.d/99-media-keys.hwdb. After running systemd-hwdb update && udevadm trigger as root Ardour now sees the play/pause button as space, previous as left and next as right. I used Remapping Keyboard Keys in Ubuntu with udev / evdev as a reference.

Audio devices

There are three audio devices on this machine:

$ lspci | grep -i audio
64:00.1 Audio device: Advanced Micro Devices, Inc. [AMD/ATI] Rembrandt Radeon High Definition Audio Controller
64:00.5 Multimedia controller: Advanced Micro Devices, Inc. [AMD] ACP/ACP3X/ACP6x Audio Coprocessor (rev 63)
64:00.6 Audio device: Advanced Micro Devices, Inc. [AMD] Family 17h/19h HD Audio Controller
$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: Babyface2359686 [Babyface (23596862)], device 0: USB Audio [USB Audio]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 10: Generic [HD-Audio Generic], device 3: HDMI 0 [HDMI 0]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 10: Generic [HD-Audio Generic], device 7: HDMI 1 [HDMI 1]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 10: Generic [HD-Audio Generic], device 8: HDMI 2 [HDMI 2]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 11: Generic_1 [HD-Audio Generic], device 0: ALC257 Analog [ALC257 Analog]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
$ arecord -l
**** List of CAPTURE Hardware Devices ****
card 0: Babyface2359686 [Babyface (23596862)], device 0: USB Audio [USB Audio]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 1: acp63 [acp63], device 0: DMIC capture dmic-hifi-0 []
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 11: Generic_1 [HD-Audio Generic], device 0: ALC257 Analog [ALC257 Analog]
  Subdevices: 1/1
  Subdevice #0: subdevice #0

So onboard audio, HDMI and a capture device acp63. As you can see I’ve managed to get onboard and HDMI out of the way by indexing them but unfortunately the kernel module for the capture device, snd-soc-ps-mach, does not seem to support indexing. I’d like to index it as card 12 because now that acp63 device claims 0 when my Babyface is not attached. Something for later investigation.

I’ll dedicate a separate post to doing real-time audio on this machine. I need to do some more stress testing but Ardour runs without complaining at 48kHz, 64 frames/period and 3 periods/buffer, so 4ms system latency. No weird things there.

New notebook – The last bits

New notebook – The day after

Everything seems to work, network, audio, external monitor, display brightness. And this is the first notebook I’ve ever come across to that has a touch pad with an actual real middle click! Love it! Keyboard is nice too, backlight works fine and it has a nice array of function keys that are almost all mapped now. Overall feel of this machine is really nice, aluminum housing and the chiclet keyboard feels stable and solid. Startup time is twice as fast as the BTO. Compared to my new notebook the old one feels strangely completely outdated.

Now what is in there? CPU is an AMD Ryzen 7 7840HS, an 8-core CPU with SMT. Disabled SMT though, Ardour seems to run slightly better on 8 real cores. 32GB of RAM which should be more than enough, my old notebook had 16GB and never had any issues with that. Storage is a 1TB SK hynix BC901 NVMe drive with more than decent throughput. Almost all USB ports sit on their own bus which is quite nice. The only bus that shares its IRQ with something else (the WiFi module) is bus 001. USB ports that use this bus are both the USB-C and USB Type-A connectors closest to the screen. So better not use those.

The only thing that is a bit flaky is the WiFi connection but came across more mentions about that. The connection sometimes randomly reconnects. Can live with it, when making music I often completely disable WiFi. Did have to use systemd automount instead of an fstab entry to mount my NAS as the WiFi connection apparently takes a bit too long to come up.

Didn’t have to tweak a lot so far. As I already mentioned I did disable SMT (Simultaneous Multi-Threading) and I’ve prioritized the IRQ of the USB bus my audio interface sits on. Not with rtirq or udev-rtirq (which doesn’t work properly on this machine, the wrong IRQ seems to get prioritized) but with a small script to fetch the IRQ number that seems to change on every boot and feeding the outcome to chrt. Installed a liquorix kernel on it and enabled threaded IRQ’s, disabled mitigations and USB autosuspend. Also allowed my user to set CPU DMA latency.

BTO vs Lenovo

It’s running Debian 12 with XFCE again. Simple, fast and looks good enough to me with the Greybird theme and elementary icons. And no more notebook that takes off into orbit, the new one is way more quiet.

On with making music!

Edit: the script I threw together could be useful for others too with USB audio interfaces connected to a machine that uses MSI (Message-Signaled Interrupts) enabled USB controllers. You can find the snd_dev_id of your interface with aplay -l.

#!/bin/bash

prio=90
prio_step=5
proc_path=/proc/asound
sys_pci_bus_path=/sys/class/pci_bus
snd_dev_ids=( Babyface2359686 UA25 )

for snd_dev_id in ${snd_dev_ids[@]}; do
  snd_dev_card_number=$(awk '/'$snd_dev_id'/ {print $1}' $proc_path/cards)

  if [ -n "$snd_dev_card_number" ]; then
    snd_dev_card=card$snd_dev_card_number
    snd_dev_pci_bus_ref=$(grep -Eo "usb-[^[:space:],-]+" $proc_path/$snd_dev_card/stream0 | sed "s/usb-\(.*\)/\1/")
    snd_dev_pci_bus_ref_short=$(awk -F ':' '{print $1":"$2}' <<<$snd_dev_pci_bus_ref)
    snd_dev_irq=$(cat $sys_pci_bus_path/$snd_dev_pci_bus_ref_short/device/$snd_dev_pci_bus_ref/irq)
    snd_dev_irq_pid=$(pgrep $snd_dev_irq-xhci)

    chrt -f -p $prio $snd_dev_irq_pid
    
    prio=$((prio-prio_step))
  fi

done
New notebook – The day after