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#1 2026-01-25 18:55:34

migo
Member
Registered: 2025-12-01
Posts: 9

Something is limiting playback sample rate to max. 48kHz

Hi, I've tried to avoid resampling, searched all info about pipewire and allowing native playback but I've failed. sad System is up to date ArchLinux with integrated sound card.

migo@migoG17:~$ cat /proc/asound/card*/codec#*
Codec: Realtek ALC294
Address: 0
AFG Function Id: 0x1 (unsol 1)
Vendor Id: 0x10ec0294
Subsystem Id: 0x10431d1f
Revision Id: 0x100004
No Modem Function Group found
Default PCM:
    rates [0x560]: 44100 48000 96000 192000
    bits [0xe]: 16 20 24
    formats [0x1]: PCM
Default Amp-In caps: N/A
Default Amp-Out caps: N/A
State of AFG node 0x01:
  Power states:  D0 D1 D2 D3 D3cold CLKSTOP EPSS
  Power: setting=D0, actual=D0
GPIO: io=4, o=0, i=0, unsolicited=1, wake=0
  IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0
  IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0
  IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0
  IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0
Node 0x02 [Audio Output] wcaps 0x41d: Stereo Amp-Out
  Control: name="Headphone Playback Volume", index=0, device=0
    ControlAmp: chs=3, dir=Out, idx=0, ofs=0
  Device: name="ALC294 Analog", type="Audio", device=0
  Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0
  Amp-Out vals:  [0x00 0x00]
  Converter: stream=5, channel=0
  PCM:
    rates [0x60]: 44100 48000
    bits [0xe]: 16 20 24
    formats [0x1]: PCM
  Power states:  D0 D1 D2 D3 EPSS
  Power: setting=D0, actual=D0
Node 0x03 [Audio Output] wcaps 0x41d: Stereo Amp-Out
  Control: name="Speaker Playback Volume", index=0, device=0
    ControlAmp: chs=3, dir=Out, idx=0, ofs=0
  Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0
  Amp-Out vals:  [0x05 0x05]
  Converter: stream=5, channel=0
  PCM:
    rates [0x560]: 44100 48000 96000 192000
    bits [0xe]: 16 20 24
    formats [0x1]: PCM
  Power states:  D0 D1 D2 D3 EPSS
  Power: setting=D0, actual=D0
Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono
Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono
Node 0x06 [Audio Output] wcaps 0x411: Stereo
  Converter: stream=0, channel=0
  PCM:
    rates [0x40]: 48000
    bits [0xe]: 16 20 24
    formats [0x1]: PCM
  Power states:  D0 D1 D2 D3 EPSS
  Power: setting=D0, actual=D0
migo@migoG17:~$ cat .config/pipewire/pipewire.conf.d/clock-rate.conf
context.properties = {
    default.clock.allowed-rates = [ 48000 44100 96000 192000 ]
    }
migo@migoG17:~$

Here is pw-top output of Tonearm linux Tidal player playing 96k sampled song:

S   ID  QUANT   RATE    WAIT    BUSY   W/Q   B/Q  ERR FORMAT           NAME
S   30      0      0    ---     ---   ---   ---     0                  Dummy-Driver
S   31      0      0    ---     ---   ---   ---     0                  Freewheel-Driver
S   49      0      0    ---     ---   ---   ---     0                  Midi-Bridge
S   52      0      0    ---     ---   ---   ---     0                  bluez_midi.server
R   55   4096  96000  53,2us  71,5us  0,00  0,00    0    S32LE 2 48000 alsa_output.pci-0000_09_00.6.analog-stereo
R   67   8192  96000  24,6us  17,7us  0,00  0,00    0   S24_32 2 96000  + tonearm
S   56      0      0    ---     ---   ---   ---     0                  alsa_input.pci-0000_09_00.6.analog-stereo
S   60      0      0    ---     ---   ---   ---     0                  v4l2_input.pci-0000_09_00.3-usb-0_1_1.0

Thank you!

Last edited by migo (2026-01-26 07:41:14)

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#2 2026-01-25 22:44:21

seth
Member
From: Don't DM me only for attention
Registered: 2012-09-03
Posts: 72,690

Re: Something is limiting playback sample rate to max. 48kHz

Please use [code][/code] tags. Edit your post in this regard.

tail -n10000 /proc/asound/card*/codec#*

will implicitly print the file names (but there seems only one anyway)

R   55   4096  96000  53,2us  71,5us  0,00  0,00    0    S32LE 2 48000 alsa_output.pci-0000_09_00.6.analog-stereo
Node 0x06 [Audio Output] wcaps 0x411: Stereo
  PCM:
    rates [0x40]: 48000
pactl list sinks

Only

Node 0x03 [Audio Output] wcaps 0x41d: Stereo Amp-Out
  Control: name="Speaker Playback Volume", index=0, device=0
    ControlAmp: chs=3, dir=Out, idx=0, ofs=0
  Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0
  Amp-Out vals:  [0x05 0x05]
  Converter: stream=5, channel=0
  PCM:
    rates [0x560]: 44100 48000 96000 192000
    bits [0xe]: 16 20 24
    formats [0x1]: PCM
  Power states:  D0 D1 D2 D3 EPSS
  Power: setting=D0, actual=D0

actually lists > 48k

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#3 2026-01-26 07:53:31

migo
Member
Registered: 2025-12-01
Posts: 9

Re: Something is limiting playback sample rate to max. 48kHz

Hello, sorry for wrong formatting sad, corrected now.

migo@migoG17:/usr/lib/systemd/system$ pactl list sinks
Sink #56
        State: RUNNING
        Name: alsa_output.pci-0000_09_00.6.analog-stereo
        Description: Ryzen HD Audio Controller Analog Stereo
        Driver: PipeWire
        Sample Specification: s32le 2ch 48000Hz
        Channel Map: front-left,front-right
        Owner Module: 4294967295
        Mute: no
        Volume: front-left: 30927 /  47% / -19,57 dB,   front-right: 30927 /  47% / -19,57 dB
                balance 0,00
        Base Volume: 65536 / 100% / 0,00 dB
        Monitor Source: alsa_output.pci-0000_09_00.6.analog-stereo.monitor
        Latency: 0 usec, configured 0 usec
        Flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY
        Properties:
                object.path = "alsa:acp:Generic:3:playback"
                api.alsa.path = "front:0"
                api.alsa.pcm.card = "0"
                api.alsa.pcm.stream = "playback"
                port.group = "playback"
                device.icon_name = "audio-card-analog"
                device.bus = "pci"
                audio.channels = "2"
                audio.position = "FL,FR"
                device.routes = "2"
                alsa.resolution_bits = "16"
                alsa.card = "0"
                alsa.card_name = "HD-Audio Generic"
                alsa.long_card_name = "HD-Audio Generic at 0xfc5c0000 irq 147"
                alsa.driver_name = "snd_hda_intel"
                alsa.mixer_name = "Realtek ALC294"
                alsa.components = "HDA:10ec0294,10431d1f,00100004"
                alsa.id = "Generic"
                device.api = "alsa"
                device.class = "sound"
                alsa.class = "generic"
                alsa.subclass = "generic-mix"
                alsa.name = "ALC294 Analog"
                alsa.subdevice = "0"
                alsa.subdevice_name = "subdevice #0"
                alsa.device = "0"
                alsa.sync.id = "00000000:00000000:00000000:00000000"
                device.profile.name = "analog-stereo"
                device.profile.description = "Analog Stereo"
                card.profile.device = "3"
                device.id = "49"
                factory.name = "api.alsa.pcm.sink"
                node.pause-on-idle = "false"
                priority.driver = "1009"
                priority.session = "1009"
                media.class = "Audio/Sink"
                node.name = "alsa_output.pci-0000_09_00.6.analog-stereo"
                node.nick = "ALC294 Analog"
                device.description = "Ryzen HD Audio Controller"
                api.alsa.card.name = "HD-Audio Generic"
                api.alsa.card.longname = "HD-Audio Generic at 0xfc5c0000 irq 147"
                factory.id = "19"
                clock.quantum-limit = "8192"
                client.id = "48"
                node.driver = "true"
                node.loop.name = "data-loop.0"
                library.name = "audioconvert/libspa-audioconvert"
                object.id = "56"
                object.serial = "56"
                node.max-latency = "16384/48000"
                api.alsa.period-size = "1024"
                api.alsa.period-num = "32"
                api.alsa.headroom = "0"
                device.enum.api = "udev"
                api.alsa.card = "0"
                device.plugged.usec = "8755803"
                device.bus_path = "pci-0000:09:00.6"
                sysfs.path = "/devices/pci0000:00/0000:00:08.1/0000:09:00.6/sound/card0"
                device.subsystem = "sound"
                device.vendor.id = "0x1022"
                device.vendor.name = "Advanced Micro Devices, Inc. [AMD]"
                device.product.id = "0x15e3"
                device.product.name = "Ryzen HD Audio Controller"
                device.name = "alsa_card.pci-0000_09_00.6"
                device.nick = "HD-Audio Generic"
                api.alsa.use-acp = "true"
                api.acp.auto-profile = "false"
                api.acp.auto-port = "false"
                api.dbus.ReserveDevice1.Priority = "-20"
                api.alsa.split-enable = "true"
                api.dbus.ReserveDevice1 = "Audio0"
                spa.object.id = "2"
                device.string = "0"
        Ports:
                analog-output-speaker: Speakers (type: Speaker, priority: 10000, availability group: Legacy 2, availability unknown)
                analog-output-headphones: Headphones (type: Headphones, priority: 9900, availability group: Legacy 3, not available)
        Active Port: analog-output-speaker
        Formats:
                pcm
migo@migoG17:/usr/lib/systemd/system$

According to the pw-top rate is 96k on both ends, but output format in the alsa device is 48k only. Where can be the problem? Switching output sample format between 44.1k and 48k works fine, but not above 48k.  Thank you.

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#4 2026-01-26 13:23:51

seth
Member
From: Don't DM me only for attention
Registered: 2012-09-03
Posts: 72,690

Re: Something is limiting playback sample rate to max. 48kHz

As mentioned, the node only supports <= 48k
Afaict only "Node 0x03 [Audio Output] wcaps 0x41d: Stereo Amp-Out" will do > 48k (whether you will hear anything over that output or it's just the volume control I can't tell)

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#5 2026-01-26 16:50:22

migo
Member
Registered: 2025-12-01
Posts: 9

Re: Something is limiting playback sample rate to max. 48kHz

Understood, but this is true only for the Node 0x02, which is labeled as "Headphone Playback Volume":

Node 0x02 [Audio Output] wcaps 0x41d: Stereo Amp-Out
  Control: name="Headphone Playback Volume", index=0, device=0
    ControlAmp: chs=3, dir=Out, idx=0, ofs=0
  Device: name="ALC294 Analog", type="Audio", device=0
  Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0
  Amp-Out vals:  [0x00 0x00]
  Converter: stream=5, channel=0
  PCM:
    rates [0x60]: 44100 48000
    bits [0xe]: 16 20 24
    formats [0x1]: PCM
  Power states:  D0 D1 D2 D3 EPSS
  Power: setting=D0, actual=D0

Node 0x03 labeled "Speaker Playback Volume" has 192k max and I'm using internal speakers for playback.

Node 0x03 [Audio Output] wcaps 0x41d: Stereo Amp-Out
  Control: name="Speaker Playback Volume", index=0, device=0
    ControlAmp: chs=3, dir=Out, idx=0, ofs=0
  Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0
  Amp-Out vals:  [0x3d 0x3d]
  Converter: stream=5, channel=0
  PCM:
    rates [0x560]: 44100 48000 96000 192000
    bits [0xe]: 16 20 24
    formats [0x1]: PCM
  Power states:  D0 D1 D2 D3 EPSS
  Power: setting=D0, actual=D0

and main codec section list all sample rates:

Default PCM:
    rates [0x560]: 44100 48000 96000 192000
    bits [0xe]: 16 20 24
    formats [0x1]: PCM

It seem like there is a bug somewhere to me. Or I get all this wrong?
Thank you!

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#6 2026-01-26 21:08:50

seth
Member
From: Don't DM me only for attention
Registered: 2012-09-03
Posts: 72,690

Re: Something is limiting playback sample rate to max. 48kHz

this is true only for the Node 0x02

No.

I'm using internal speakers for playback.

Which is

seth wrote:
R   55   4096  96000  53,2us  71,5us  0,00  0,00    0    S32LE 2 48000 alsa_output.pci-0000_09_00.6.analog-stereo
Node 0x06 [Audio Output] wcaps 0x411: Stereo
  PCM:
    rates [0x40]: 48000

Or complete

Node 0x06 [Audio Output] wcaps 0x411: Stereo
  Converter: stream=0, channel=0
  PCM:
    rates [0x40]: 48000
    bits [0xe]: 16 20 24
    formats [0x1]: PCM
  Power states:  D0 D1 D2 D3 EPSS
  Power: setting=D0, actual=D0

The behavior of the output doesn't seem surprising - the question is whether you can utilize node 0x3 at all or that's a purely transactional device.

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#7 2026-01-27 12:05:45

migo
Member
Registered: 2025-12-01
Posts: 9

Re: Something is limiting playback sample rate to max. 48kHz

Thank you for explanation. Do you know the way I can try to play through node 0x3? Thank you!

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#8 2026-01-27 13:06:48

seth
Member
From: Don't DM me only for attention
Registered: 2012-09-03
Posts: 72,690

Re: Something is limiting playback sample rate to max. 48kHz

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#9 2026-01-27 19:14:03

migo
Member
Registered: 2025-12-01
Posts: 9

Re: Something is limiting playback sample rate to max. 48kHz

Thank you, will try do it and will report back. smile

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#10 Yesterday 08:18:43

migo
Member
Registered: 2025-12-01
Posts: 9

Re: Something is limiting playback sample rate to max. 48kHz

Hello, I still think you are wrong, according to the hdaanalyser is output set to 0x03 and changing volume slider in this property changes actual volume. You can see this here too 0x03* :

Node 0x17 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out
  Control: name="Speaker Playback Switch", index=0, device=0
    ControlAmp: chs=3, dir=Out, idx=0, ofs=0
  Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
  Amp-Out vals:  [0x00 0x00]
  Pincap 0x0001001c: OUT HP EAPD Detect
  EAPD 0x2: EAPD
  Pin Default 0x90170110: [Fixed] Speaker at Int N/A
    Conn = Analog, Color = Unknown
    DefAssociation = 0x1, Sequence = 0x0
    Misc = NO_PRESENCE
  Pin-ctls: 0x40: OUT
  Unsolicited: tag=00, enabled=0
  Power states:  D0 D1 D2 D3 EPSS
  Power: setting=D0, actual=D0
  Connection: 3
     0x02 0x03* 0x06

Speaker Playback Switch

https://pasteboard.co/4D9WpClQ1Rna.png

and

Speaker Playback Volume
https://pasteboard.co/v2shX0sMCA2I.png

in second picture you can see all supported rates too.

Last edited by migo (Yesterday 08:19:46)

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#11 Yesterday 09:24:06

seth
Member
From: Don't DM me only for attention
Registered: 2012-09-03
Posts: 72,690

Re: Something is limiting playback sample rate to max. 48kHz

And did this change?

R   55   4096  96000  53,2us  71,5us  0,00  0,00    0    S32LE 2 48000 alsa_output.pci-0000_09_00.6.analog-stereo

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#12 Yesterday 09:36:15

migo
Member
Registered: 2025-12-01
Posts: 9

Re: Something is limiting playback sample rate to max. 48kHz

No, unfortunately. I've restarted following services:

systemctl --user restart pipewire pipewire-pulse wireplumber

here is some more info from pw-cli:

migo@migoG17:~$ pw-cli info 60
        id: 60
        permissions: rwxm-
        type: PipeWire:Interface:Node/3
*       input ports: 2/65
*       output ports: 2/0
*       state: "idle"
*       properties:
*               object.path = "alsa:acp:Generic:3:playback"
*               api.alsa.path = "front:0"
*               api.alsa.pcm.card = "0"
*               api.alsa.pcm.stream = "playback"
*               port.group = "playback"
*               device.icon-name = "audio-card-analog"
*               device.bus = "pci"
*               audio.channels = "2"
*               audio.position = "FL,FR"
*               device.routes = "2"
*               alsa.resolution_bits = "16"
*               alsa.card = "0"
*               alsa.card_name = "HD-Audio Generic"
*               alsa.long_card_name = "HD-Audio Generic at 0xfc5c0000 irq 147"
*               alsa.driver_name = "snd_hda_intel"
*               alsa.mixer_name = "Realtek ALC294"
*               alsa.components = "HDA:10ec0294,10431d1f,00100004"
*               alsa.id = "Generic"
*               device.api = "alsa"
*               device.class = "sound"
*               alsa.class = "generic"
*               alsa.subclass = "generic-mix"
*               alsa.name = "ALC294 Analog"
*               alsa.subdevice = "0"
*               alsa.subdevice_name = "subdevice #0"
*               alsa.device = "0"
*               alsa.sync.id = "00000000:00000000:00000000:00000000"
*               device.profile.name = "analog-stereo"
*               device.profile.description = "Analog Stereo"
*               card.profile.device = "3"
*               device.id = "53"
*               factory.name = "api.alsa.pcm.sink"
*               node.pause-on-idle = "false"
*               priority.driver = "1009"
*               priority.session = "1009"
*               media.class = "Audio/Sink"
*               node.name = "alsa_output.pci-0000_09_00.6.analog-stereo"
*               node.nick = "ALC294 Analog"
*               node.description = "Laptop"
*               api.alsa.card.name = "HD-Audio Generic"
*               api.alsa.card.longname = "HD-Audio Generic at 0xfc5c0000 irq 147"
*               factory.id = "19"
*               clock.quantum-limit = "8192"
*               client.id = "52"
*               node.driver = "true"
*               node.loop.name = "data-loop.0"
*               library.name = "audioconvert/libspa-audioconvert"
*               object.id = "60"
*               object.serial = "60"
*               node.max-latency = "16384/44100"
*               api.alsa.period-size = "1024"
*               api.alsa.period-num = "32"
*               api.alsa.headroom = "0"
*       params: (9)
*         3 (Spa:Enum:ParamId:EnumFormat) r-
*         1 (Spa:Enum:ParamId:PropInfo) r-
*         2 (Spa:Enum:ParamId:Props) rw
*         4 (Spa:Enum:ParamId:Format) rw
*         10 (Spa:Enum:ParamId:EnumPortConfig) r-
*         11 (Spa:Enum:ParamId:PortConfig) rw
*         15 (Spa:Enum:ParamId:Latency) rw
*         16 (Spa:Enum:ParamId:ProcessLatency) rw
*         17 (Spa:Enum:ParamId:Tag) rw

I don't know if it is enough, maybe this enumeration is done during boot by alsa and is fix. What do you think? Thank you!

Last edited by migo (Yesterday 09:38:03)

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#13 Yesterday 09:40:37

seth
Member
From: Don't DM me only for attention
Registered: 2012-09-03
Posts: 72,690

Re: Something is limiting playback sample rate to max. 48kHz

I think you're speakers are "Node 0x06 [Audio Output] wcaps 0x411: Stereo" and "Node 0x03 [Audio Output] wcaps 0x41d: Stereo Amp-Out" is a transactional device.
Does the notebook advertise 24bit/192kHz output (this would likely be all over the manual and on some stickers a major marketing point)

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#14 Yesterday 10:02:14

migo
Member
Registered: 2025-12-01
Posts: 9

Re: Something is limiting playback sample rate to max. 48kHz

But when I select output to "Node 0x06 [Audio Output] wcaps 0x411: Stereo" in Node 0x17 there is no audio. But when I disable all connection to the  "Node 0x06 [Audio Output] wcaps 0x411: Stereo" in all widgets connected to it playback is running without interruption.

https://pasteboard.co/sAab0xQpmWjl.png

and node 0x06 is capable only playing 48k right? But output is playing 44.1k just fine.

S   ID  QUANT   RATE    WAIT    BUSY   W/Q   B/Q  ERR FORMAT           NAME
S   30      0      0    ---     ---   ---   ---     0                  Dummy-Driver
S   31      0      0    ---     ---   ---   ---     0                  Freewheel-Driver
S   54      0      0    ---     ---   ---   ---     0                  Midi-Bridge
S   57      0      0    ---     ---   ---   ---     0                  bluez_midi.server
R   60   1024  44100  26,6us  14,7us  0,00  0,00    0    S32LE 2 44100 alsa_output.pci-0000_09_00.6.analog-stereo
R   73   3969  44100  10,0us   7,0us  0,00  0,00    0    S16LE 2 44100  + tonearm
S   61      0      0    ---     ---   ---   ---     0                  alsa_input.pci-0000_09_00.6.analog-stereo
S   65      0      0    ---     ---   ---   ---     0                  v4l2_input.pci-0000_09_00.3-usb-0_1_1.0

https://pasteboard.co/ojtXM3OWjQeu.png

Last edited by migo (Yesterday 15:29:01)

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#15 Yesterday 16:18:27

seth
Member
From: Don't DM me only for attention
Registered: 2012-09-03
Posts: 72,690

Re: Something is limiting playback sample rate to max. 48kHz

But when I select output to "Node 0x06 [Audio Output] wcaps 0x411: Stereo" in Node 0x17 there is no audio.

https://pasteboard.co/sAab0xQpmWjl.png is 404 and I don't really understand what you're trying to say with this.

and node 0x06 is capable only playing 48k right?

https://wiki.archlinux.org/title/PipeWi … ample_rate
The stream will be upsampled.

seth wrote:

Does the notebook advertise 24bit/192kHz output (this would likely be all over the manual and on some stickers a major marketing point)

Otherwise what makes you believe some internal notebook speakers can run 24bit/192kHz ?
192kHz is typically required for processing to avoid interpolation artifacts but gets downsampled to 44kHz or 48kHz for actual playback - I'd frankly be surprised if it was marketed for notebook speakers (and marveled how to get that fidelity out of that size/tech)
The overall support probably exists to be able to link out the signal to some external HiFi DACs

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#16 Today 08:14:29

migo
Member
Registered: 2025-12-01
Posts: 9

Re: Something is limiting playback sample rate to max. 48kHz

seth wrote:

But when I select output to "Node 0x06 [Audio Output] wcaps 0x411: Stereo" in Node 0x17 there is no audio.

https://pasteboard.co/sAab0xQpmWjl.png is 404 and I don't really understand what you're trying to say with this.

Link is working fine here...

seth wrote:

and node 0x06 is capable only playing 48k right?

https://wiki.archlinux.org/title/PipeWi … ample_rate
The stream will be upsampled.

It is not upsampled it is playing 44100, please look at output from pw-top in my previous post.

seth wrote:
seth wrote:

Does the notebook advertise 24bit/192kHz output (this would likely be all over the manual and on some stickers a major marketing point)

Otherwise what makes you believe some internal notebook speakers can run 24bit/192kHz ?
192kHz is typically required for processing to avoid interpolation artifacts but gets downsampled to 44kHz or 48kHz for actual playback - I'd frankly be surprised if it was marketed for notebook speakers (and marveled how to get that fidelity out of that size/tech)
The overall support probably exists to be able to link out the signal to some external HiFi DACs

I'm aware that this doesn't have real sense for internal speakers, but for headphone use it have. 

Yes laptop is Hi-Res certification (for headphone), please look:

https://rog.asus.com/laptops/rog-strix/ … ries/spec/

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