Check the sound specs on my laptop

how do i check the sound specs on my laptop? bit depth and sample rates

Kernel: 5.15.6-zen2-1-zen x86_64 bits: 64 compiler: gcc v: 11.1.0
parameters: BOOT_IMAGE=/@/boot/vmlinuz-linux-zen root=UUID=453092bf-16eb-4fa2-9643-a0e413678109
rw [email protected] quiet
cryptdevice=UUID=42bc0ad4-90a9-4f07-8a4c-78f17bd64474:luks-42bc0ad4-90a9-4f07-8a4c-78f17bd64474
root=/dev/mapper/luks-42bc0ad4-90a9-4f07-8a4c-78f17bd64474 splash rd.udev.log_priority=3
vt.global_cursor_default=0 systemd.unified_cgroup_hierarchy=1
resume=/dev/mapper/luks-bd92b9d8-d6a0-4438-9c6e-a327dabdb6fc loglevel=3
Desktop: KDE Plasma 5.23.4 tk: Qt 5.15.2 info: latte-dock wm: kwin_x11 vt: 1 dm: SDDM
Distro: Garuda Linux base: Arch Linux
Machine:   Type: Laptop System: HP product: HP Pavilion Laptop 15-eh0xxx v: N/A
serial: <superuser required> Chassis: type: 10 serial: <superuser required>
Mobo: HP model: 87C5 v: 35.24 serial: <superuser required> UEFI: AMI v: F.12 date: 03/04/2021
Battery:   ID-1: BAT0 charge: 37.2 Wh (100.0%) condition: 37.2/37.2 Wh (100.0%) volts: 12.7 min: 11.3
model: Hewlett-Packard Primary type: Li-ion serial: N/A status: Full cycles: 334
CPU:       Info: 6-Core model: AMD Ryzen 5 4500U with Radeon Graphics bits: 64 type: MCP arch: Zen 2
family: 17 (23) model-id: 60 (96) stepping: 1 microcode: 8600106 cache: L1: 384 KiB L2: 3 MiB
L3: 8 MiB
flags: avx avx2 ht lm nx pae sse sse2 sse3 sse4_1 sse4_2 sse4a ssse3 svm bogomips: 28447
Speed: 2256 MHz min/max: 1400/2375 MHz boost: enabled Core speeds (MHz): 1: 3863 2: 2753
3: 1650 4: 1768 5: 2886 6: 1801
Vulnerabilities: Type: itlb_multihit status: Not affected
Type: l1tf status: Not affected
Type: mds status: Not affected
Type: meltdown status: Not affected
Type: spec_store_bypass mitigation: Speculative Store Bypass disabled via prctl
Type: spectre_v1 mitigation: usercopy/swapgs barriers and __user pointer sanitization
Type: spectre_v2
mitigation: Full AMD retpoline, IBPB: conditional, IBRS_FW, STIBP: disabled, RSB filling
Type: srbds status: Not affected
Type: tsx_async_abort status: Not affected
Graphics:  Device-1: AMD Renoir vendor: Hewlett-Packard driver: amdgpu v: kernel bus-ID: 04:00.0
chip-ID: 1002:1636 class-ID: 0300
Display: x11 server: X.Org 1.21.1.1 compositor: kwin_x11 driver: loaded: amdgpu,ati
unloaded: modesetting alternate: fbdev,vesa display-ID: :0 screens: 1
Screen-1: 0 s-res: 1921x2160 s-dpi: 96 s-size: 506x570mm (19.9x22.4") s-diag: 762mm (30")
Monitor-1: eDP res: 1920x1080 hz: 60 dpi: 142 size: 344x194mm (13.5x7.6") diag: 395mm (15.5")
Monitor-2: HDMI-A-0 res: 1920x1080 hz: 60 dpi: 305 size: 160x90mm (6.3x3.5") diag: 184mm (7.2")
OpenGL: renderer: AMD RENOIR (DRM 3.42.0 5.15.6-zen2-1-zen LLVM 13.0.0) v: 4.6 Mesa 21.2.5
direct render: Yes
Audio:     Device-1: AMD vendor: Hewlett-Packard driver: snd_hda_intel v: kernel bus-ID: 04:00.1
chip-ID: 1002:1637 class-ID: 0403
Device-2: AMD Raven/Raven2/FireFlight/Renoir Audio Processor vendor: Hewlett-Packard
driver: snd_rn_pci_acp3x v: kernel alternate: snd_pci_acp3x,snd_pci_acp5x bus-ID: 04:00.5
chip-ID: 1022:15e2 class-ID: 0480
Device-3: AMD Family 17h HD Audio vendor: Hewlett-Packard driver: snd_hda_intel v: kernel
bus-ID: 04:00.6 chip-ID: 1022:15e3 class-ID: 0403
Sound Server-1: ALSA v: k5.15.6-zen2-1-zen running: yes
Sound Server-2: JACK v: 1.9.19 running: no
Sound Server-3: PulseAudio v: 15.0 running: no
Sound Server-4: PipeWire v: 0.3.40 running: yes
Network:   Device-1: Intel Wi-Fi 6 AX200 driver: iwlwifi v: kernel bus-ID: 02:00.0 chip-ID: 8086:2723
class-ID: 0280
IF: wlo1 state: up mac: <filter>
Bluetooth: Device-1: Intel AX200 Bluetooth type: USB driver: btusb v: 0.8 bus-ID: 1-4:3 chip-ID: 8087:0029
class-ID: e001
Report: bt-adapter ID: hci0 rfk-id: 0 state: up address: <filter>
Drives:    Local Storage: total: 476.94 GiB used: 243.2 GiB (51.0%)
SMART Message: Unable to run smartctl. Root privileges required.
ID-1: /dev/nvme0n1 maj-min: 259:0 vendor: Samsung model: MZVLQ512HALU-000H1 size: 476.94 GiB
block-size: physical: 512 B logical: 512 B speed: 31.6 Gb/s lanes: 4 type: SSD serial: <filter>
rev: HPS4NFXV temp: 27.9 C scheme: GPT
Partition: ID-1: / raw-size: 467.84 GiB size: 467.84 GiB (100.00%) used: 243.2 GiB (52.0%) fs: btrfs
dev: /dev/dm-0 maj-min: 254:0 mapped: luks-42bc0ad4-90a9-4f07-8a4c-78f17bd64474
ID-2: /boot/efi raw-size: 300 MiB size: 299.4 MiB (99.80%) used: 720 KiB (0.2%) fs: vfat
dev: /dev/nvme0n1p1 maj-min: 259:1
ID-3: /home raw-size: 467.84 GiB size: 467.84 GiB (100.00%) used: 243.2 GiB (52.0%) fs: btrfs
dev: /dev/dm-0 maj-min: 254:0 mapped: luks-42bc0ad4-90a9-4f07-8a4c-78f17bd64474
ID-4: /var/log raw-size: 467.84 GiB size: 467.84 GiB (100.00%) used: 243.2 GiB (52.0%)
fs: btrfs dev: /dev/dm-0 maj-min: 254:0 mapped: luks-42bc0ad4-90a9-4f07-8a4c-78f17bd64474
ID-5: /var/tmp raw-size: 467.84 GiB size: 467.84 GiB (100.00%) used: 243.2 GiB (52.0%)
fs: btrfs dev: /dev/dm-0 maj-min: 254:0 mapped: luks-42bc0ad4-90a9-4f07-8a4c-78f17bd64474
Swap:      Kernel: swappiness: 133 (default 60) cache-pressure: 100 (default)
ID-1: swap-1 type: zram size: 7.19 GiB used: 1.33 GiB (18.5%) priority: 100 dev: /dev/zram0
ID-2: swap-2 type: partition size: 8.8 GiB used: 0 KiB (0.0%) priority: -2 dev: /dev/dm-1
maj-min: 254:1 mapped: luks-bd92b9d8-d6a0-4438-9c6e-a327dabdb6fc
Sensors:   System Temperatures: cpu: 46.0 C mobo: N/A gpu: amdgpu temp: 48.0 C
Fan Speeds (RPM): N/A
Info:      Processes: 266 Uptime: 1d 31m wakeups: 5406 Memory: 7.2 GiB used: 3.74 GiB (52.0%)
Init: systemd v: 249 tool: systemctl Compilers: gcc: 11.1.0 clang: 13.0.0 Packages:
pacman: 1738 lib: 514 Shell: fish v: 3.3.1 default: Bash v: 5.1.12 running-in: konsole
inxi: 3.3.09

I'm not an audio expert, but maybe the documentation of pipewire configuration could help you find what you need:

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Tip: In Garuda, pipewire.conf is in /usr/share/pipewire rather than /etc/pipewire :eyes:

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Previous answers are a tad off topic but depends on what is actually meant. Which is to say if you want to see what your settings are then yes you will have to look at the pulse or pipe conf files. It's very user unfriendly. However this will not tell you what your spec is. You will have to find your chipset and look up what its bit rate and frequency ranges are. Chipset will be in various dmesg/journalctl bits or there are some tools you can use to list that. The easiest would be just launch hardinfo and see what it gives you.

Now if you want to set different default bit rates and frequencies, resamplers etc then you will have to dig into pulse/alsa/pipe config details. If you are doing audio barring you are using an application that is now forcing pipewire it's best you get rid of it.

For example in my /etc/pulse/daemon.conf I have as a base. I used to set 96 as my sample rate but it just yields too many issues in games and I can just change when doing audio specific things.
default-sample-format = float32be
default-sample-rate = 48000
alternate-sample-rate = 44100
default-sample-channels = 2
default-channel-map = front-left,front-right

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im trying to see what bit depth and sample rates my laptop supports so i can do some live asio vst dsp , live effects on a microphone. if the dsp processor supports 96 kb or higher i plan to use 4 cores to process dsp and 2 cores to record in bitwig. im used to using windows to do this and multiple sound cards and mutiple computers however with jack i think i could do it all on one machine. it used to take an i7 and a high end sound card to do this im going to attempt this on my ryzen laptop hoping that the apu has accelerated dsp processing.

right now i'm researching before i dive in.

Yeah you will definitely want to nuke pipewire :wink:

I'm spitballing but I'd say you probably can do 96Khz given a lot of 7.1 systems are built for it and the Renoir HDMI or DP ports are probably assumed to plug into TV's with as well as normal monitors. The how well is the catch.

I'll try to toss in if I can with future questions but this stuff is tricky. Never mind with a card that isn't designed for low latency.

If you're going jack you can define your sample rate in the server line. You can do this to test and push the sample rate up, see if it works, kill, try next highest until you find your cards limits in supported freq/BR or just what is can process acceptably. I've sadly dealt with too many cards over the years that say ya ya I can TOTALLY do 32bit @ 192Khz...sure with a 5minute latency in the fine print.

/usr/bin/jackd -R -v -dalsa -dhw:UR-Interface-Here -r48000 <-- dat -r flag

Not sure if you have this covered yet but I also recommend linvst. GitHub - osxmidi/LinVst: Linux Windows vst wrapper/bridge It's what I use for my DAW. Simple to use, works well.

i'm aiming to do at least 96khz 24 bit at like 3 ms latency, never tried this on linux yet but I'm sure its possible, but i don't know if i can replace 2 high end sound cards an an i3 and i7 that it used to take to do this. thanks for the recommend, i say that wile browsing in pamac and was considering it.

infact that is what got me to thinking this was possible, cause many of my fav vst plugins are made in ms visual studios, i figured between wine and linvst i could figure out how to make it work.

i accomplished this once before but my windows vst didnt work on thatr setup and i was using some janky pitch correction that i wasnt willing to use. it was slow and it sounded terrible but that was years ago on my core 2 duo sager gaming laptop . it was top of the line but still wasnt close to fast enough. but hardware has came a long way.

also amd has this new audio architecture with true audio next and "sound open firmware",
point being i think amd is working on accelerating dsp and this could be great for audio engineers.

Well I'll be bold and say you can defiantly make it work. It's just the 3ms @ 96 that's looming with a question mark :wink:

I've always been able to get that 1/1.5ms for my drums during playback/tracking but my interface is designed for it.

Top of the line doesn't matter much once you say "laptop." They just aren't up to the task...yet. Now I reserve an exception because the new Ryzen machines with a big SSD may be just fine. Though I'd wager you will want to disable all the power saving because that stuff has no place in audio heh.

im not looking to do playback, i'm looking to do lots of live vst . i'm looking to run an audio effects chain of at least 10 plugins, chorus reverb delay pitch correction and surround processing..multiple instances of some of them as well. i hope i don't have to run windows to do this, since windows 8 i left and haven't looked back, i have a few windows laptops around that i use when i need windows but i never use them.

That sounds brutal...I'm really thinking you're going to need a legit audio interface for that. However that brings about a lot of other issues where in interfaces that might do some DSP become problematic when they don't know to do it because the vst's are behind 5 layers of obscuring audio subsystems. I will be interested to hear if the Renoir chip holds up.

It also sounds like you're setting up EZ Karaoke mode for singers heh. To which I say DON'T DO IT! Just film them butchering the songs, upload to YT and make a fortune off the cringe! :wink:

you're hillarious, i do however feel you are looking at this the wrong way. a sound card is processor, a simd processor, and so is my gpu amd supports several technologies to accelerate sound processing on their apu. think of how many voices a modern video game takes, the calculations for ambience of the gunshot sound echoing in an environment or the surround of a helicopter flying in circles above you.

point being, an apu could actually have more dsp power than a high end sound card or most interfaces, for instance... how much latency does usb have compared to a dsp processor that is literally part of the processor. it doesnt even have to use pci express bus, its part of the cpu itself.

I completely get what you're saying but much like the 5 layers removed thing it's more of a "how do you tell it to do the thing" situation. And while yeah the APU might (or 100% DOES) have more horse power than say a high end RME card and should have less latency it doesn't mean it will. There are some other things in the mix, some software, some low level that can kibosh it. Which isn't to discourage, just that until it's tested, it's all just conjecture. SO TEST THAT SUCKER! heh.

That said why go 96Khz with a chip where the DAC might be meh?

the way i see it if windows can do it then linux can do it too, all thats left is to do it.

I agree, and really I don't think there is a lot Linux specific in this. A lot of these "it should" issues are in Windows too. I ran Windows in the studio since Win98 and it was always underwhemling or laced with gotchas. I don't care if you used a Creative knock off you found in the trash or blew a few grand on Seq'd cards...audio was always "supposed" to do X and never delivered heh. Bit like how it looks on the menu vs when you get it. :0

i disagree...im a windows expert and i ran an x-fi fatality at 3 ms latency at 192 khz with all the asio i ever wanted and lots of headroom to boot. If you need help with a daw on winows im your man.

I developed in vs visual studios for like a decade and i also produced in windows for about 20 years with great results. i invented the sound that guys like travis scot roddy rich and others use today..

i know windows like the back of my hand i used to study everything windows from the msdn network and msdn channel 9. ive had windows kernel architects explain the windows kernel in detail on whiteboards, i even helped test and finalize windows xp embedded working with product manager mike hall on the forums,..

i might end up making me an xp embedded dsp box for my vocal rack. then just run that into my other laptop to record as long as linux low latency monitoring works ..

Lol I was just a lowly MSFT certified OEM :wink:

Been doing audio for about 35 years, live and studio :wink:

Can't say anything about X-Fi I've never used Creative or consumer stuff.

As a side note if you've ever seen boxes like the MUSE VST racks those were all just Linux boxes with there "we won't tell you s#it about how we do it" special sauce.

i was an oem for a while but just so i could work on certain projects. as far as x-fi....

at the time it was literally the fastest dsp processor on the planet bar none, specifically the titanium edition. i had the fatal1ty version. and when i say the fastest in the world, it really was...they made it to be such and marketed it as such. games and vst essentially use the same processing technologies,

youd be suprised, id put an x-fi against almost anything on the market, even today. These cards are highly under rated, literally about 4 times less latency than a usb interface and also about ten ten mes the processing power of something like an m audio m track. those cards are amazing.

these people invented hardware accelerated sound.

Don't get me started on M-Audio...flipping garbage heh. When I downsized I went with a USB interface. It's OK but certainly nothing great. Too many issues in the past with PCI/e and noise on the bus.

As for the dsp side of things I go back to my yeah but what knows to use it heh. I've had some vst that where the "only this card" versions to work with their DSP chip but really it never made any difference. Hence my point of unless something somewhere says "use this for processing" it just dumps it on the CPU. This is sadly there a lot of people buy into the Apple/Protools/Logic stuff because it's a forced eco system so this all happens naturally.

high quality motherboard and powersuply and a ups quiet those noises..also using a lower end gpu and even putting hot glue in the chokes of the mb all make the noise floor lower..you then adjust the wav balance and your inputs and outputs right and most noise is gone..also using a preamp and not running mic inputs high helps tremendously as well.

essentially if you keep your levels right then that noise floor is inaudible,