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Hello,
I spent the afternoon trying to fix this issue on a Beelink SER6-P featuring a Realtek ALC897 sound card. The output level is extremely low. I also tried other live distros and the problem seems the same: only the HDMI audio is usable.
Following the wiki (link) I tried to change /etc/asound.conf to have a Pre-amp setting in alsamixer but without any luck, is it possible that I am doing anything wrong? Is there anything else I can try?
Thanks
#/etc/asoud.conf
defaults.pcm.card 1
defaults.ctl.card 1
pcm.!default {
type hw
slave.pcm "softvol"
}
pcm.softvol {
type softvol
slave {
pcm "dmix"
}
control {
name "Pre-Amp"
card 0
}
min_dB -5.0
max_dB 20.0
resolution 6
}aplay -l
...
card 1: Generic_1 [HD-Audio Generic], device 0: ALC897 Analog [ALC897 Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0Offline
I soon realized after posting I could not add the Pre-Amp because the configuration used card0 instead of the correct card1. Nonetheless the pre-amp is of no use. Sound clips much before getting to any reasonable volume.
Open for alternative suggestions.
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You'd want to use actual processing tools/loudness ladspa modules or so. Those can be setup with ALSA alone but is somewhat cumbersome, a much easier and intuitive approach would be using pipewire together with e.g. easyeffects or so.
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You'd want to use actual processing tools/loudness ladspa modules or so. Those can be setup with ALSA alone but is somewhat cumbersome, a much easier and intuitive approach would be using pipewire together with e.g. easyeffects or so.
I am trying right now. In easyeffect like with pre-amp i can increase the signal but get to saturation well before I get a usable signal output I do not think an user-space solution is possible. It must be those damned realtek drivers that do not enable some hardware amplifier for the headphones as they do on windows.
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Did you check your HW options with
alsamixer -c1some toggles for preamps and the like are somewhat hidden as booleans/enumerations rather than a "pure" volume control.
Other than that issues here are often tied to UEFI updates, maybe check that, and/or ensuring fast-boot is disabled on Windows
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Did you check your HW options with
alsamixer -c1some toggles for preamps and the like are somewhat hidden as booleans/enumerations rather than a "pure" volume control.
Other than that issues here are often tied to UEFI updates, maybe check that, and/or ensuring fast-boot is disabled on Windows
No options in alsamixer were left unturned. The computer is new and I wiped out windows without testing the headphones output. But I see online that this is also a possible issue on windows unless you install the Realtek super mega pro drivers specific for this interface.
As beelink is clueless on the issue, tomorrow I will write to Realtek directly.
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I have the same ALC897 chip on an Asrock B650M PG Riptide. Volume is still a little low but increasing the clock from 48000 to 192000Hz improved the quality by a bit, noticeably.
The chip doesn't seem to support 32-bit yet it runs at that by default according to monitoring tools.
~/.config/pipewire/pipewire.conf
## Properties for the DSP configuration.
default.clock.rate = 192000
default.clock.allowed-rates = [ 192000 256000 384000 ]
#default.clock.quantum = 16384
default.clock.min-quantum = 4096
default.clock.max-quantum = 16384
default.clock.quantum-limit = 16384
default.clock.quantum-floor = 32
~/.config/pipewire/client.conf & ~/.config/pipewire/pipewire-pulse.conf
resample.quality = 10
These are my settings.
format.dsp 32 bit float mono audio
Or is 32-bit float something different?
Last edited by erayrafet (2025-02-08 06:48:30)
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I have the same ALC897 chip on an Asrock B650M PG Riptide. Volume is still a little low but increasing the clock from 48000 to 192000Hz improved the quality by a bit, noticeably.
The chip doesn't seem to support 32-bit yet it runs at that by default according to monitoring tools.~/.config/pipewire/pipewire.conf
## Properties for the DSP configuration.
default.clock.rate = 192000
default.clock.allowed-rates = [ 192000 256000 384000 ]
#default.clock.quantum = 16384
default.clock.min-quantum = 4096
default.clock.max-quantum = 16384
default.clock.quantum-limit = 16384
default.clock.quantum-floor = 32~/.config/pipewire/client.conf & ~/.config/pipewire/pipewire-pulse.conf
resample.quality = 10These are my settings.
format.dsp 32 bit float mono audio
Or is 32-bit float something different?
I am on pulseaudio wihtout pipewire atm, I tried changing bitrate and encoding from float32le to s16le but there is no impact on volume, on sound quality I cannot tell as I can barely hear anything.
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