New to Linux, How can I make Bazzite audio sound as good as Windows 10?

Ask Fedora
21

That makes sense since you are using an outboard DAC.

22

The only player that can do bitperfect IMO is MPD, Deadbeef is also good.

I use cantata as the frontend for mpd.

$ cat /etc/mpd.conf 
############################################################################
### begin of mpd configuration file
### created by `mpd-configure' (version 0.9.8) on 2019-01-14T20:06:51+00:00.
### see: https://gitlab.com/ronalde/mpd-configure/
############################################################################

audio_output {
	type             "alsa"
	name             "D10s - USB Audio"
	device           "hw:CARD=D10s,DEV=0"
	auto_resample    "no"
	auto_format      "no"
	auto_channels    "no"
	replay_gain_handler "none"
	mixer_type                 "none"
}
replaygain                 "off"

## start processing `02_paths.conf'
music_directory             "/home/leigh/Music"
log_file                    "/var/lib/mpd/mpd.log"
playlist_directory          "/var/lib/mpd/playlists"
db_file                     "/var/lib/mpd/mpd.db"
state_file_interval         "1"
state_file                  "/var/lib/mpd/state"
sticker_file                "/var/lib/mpd/sticker"
save_absolute_paths_in_playlists "no"
## done processing

## start processing `03_general.conf'
filesystem_charset          "UTF-8"
log_level                   "default"
auto_update                 "yes"
auto_update_depth           "4096"
## done processing

## start processing `04_client-limits.conf'
connection_timeout          "60"
max_connections             "10"
max_playlist_length         "25876"
max_command_list_size       "3234"
max_output_buffer_size      "12938"
## done processing

## start processing `05_network.conf'
bind_to_address             "0.0.0.0"
## done processing

## start processing `06_zeroconf.conf'
zeroconf_enabled            "yes"
## done processing

decoder {
        plugin "wildmidi"
        enabled "no"
}

decoder {
    plugin          "hybrid_dsd"
    enabled         "no"
    }

input {
        plugin "qobuz"
        enabled "no"
}

input {
        plugin "tidal"
        enabled "no"
}

############################################################################
### end of mpd configuration file
############################################################################

Also run the mpd systemd service as a root process as it provides realtime scheduling,

23

latency is rarely constant.

System Load (CPU Strain) As more tracks, effects plugins, and other processes are added, the computer’s CPU has to work harder. If the CPU cannot keep up, the system may introduce variable delays, clicks, or pops to manage the workload, or the latency may simply increase overall.

Driver Inconsistencies The quality and efficiency of audio drivers (such as ASIO for PCs or Core Audio for Macs) significantly impact latency stability. Poorly optimized or generic drivers can lead to inconsistent delay times.

player > pulseaudio > pipewire > alsa = too many layers with high variable latency

mpd > alsa = single layer with much lower and more constant latency

24

Respectfully, this is the first time in ~25 years of pro audio that I’ve ever heard of high latency causing inaccurate sample streams. Maybe you’re describing jitter - which is related to sample clocks, not system latency.

Either way, neither of those would cause ‘muffled’ audio. Low latency causes ‘underruns’ due to a cpu not processing data in time for the next sample - which is why we set a safety “buffer” where samples are queued up in memory before processing. The most likely scenarios in this case might be:

  • system/DAC samplerate mismatch (Fedora defaults to 48Khz check DAC is set the same),
  • gain-staging (ALSA overdriving the DAC),
  • Source material bitrate (internet connection/ vpn)

I’d goto alsamixer , F6 to choose the affected interface, set all audio channels to 50%. Test playback.
If this clears up the issue, go back to alsamixer and set the faders to 70, exit and save the settings with sudo alsactl store .

Also in alsamixer, there could be a samplerate control (unusual but I’ve seen it exposed here), 48khz is pretty standard, now. Make sure both the system AND DAC are set the same. (Linux Pipewire *conf files are usually set at this rate by default so you shouldn’t need to mess with those)

And honestly, just don’t use digital effects for music playback. It’s hard to say how they’re coded and sampling or phase problems from poor code will cause all sorts of terrible smearing issues. If you must use digital EQ, look for “linear phase” Or use EQs built into high end equipment.

I’d start with those.

(FWIW - ALSA is the low level system, Pipewire or Jack are the ASIO of Linux)