X1C7 Pulseaudio and subwoofer

rosseba · 2019-09-13 21:26:08

Hi all,
I'm working on audio config of my X1 Carbon 7th Gen (based on Dolby Atmos system).
What I have done so far:

fixed the 0 to 100 volume issue with this tip at link: https://wiki.archlinux.org/index.php/Le … e_controls
enabled subwoofer with enable-lfe-remixing and lfe-crossover-freq settings in /etc/pulse/daemon.conf
configured default-sample-channels = 3 so I can switch back and forth to headphones without restoring a sound profile config without subwoofer

Despite I'm quite happy of this config, the overall volume is low.
Volume Up and Down is now working on alsamixer Master channel on high freqs only, bass freqs seems always enabled.
If I disable subwoofer sound is louder but quite bad (high freqs only).
Some details about the scenario:

$ uname -a
Linux x1c7 5.2.14-arch2-1-ARCH #1 SMP PREEMPT Thu Sep 12 10:42:38 UTC 2019 x86_64 GNU/Linux

$ pacman -Q | grep pulse
libcanberra-pulse 0.30+2+gc0620e4-2
libpulse 12.99.3-1
pulseaudio 12.99.3-1
pulseeffects 4.6.7-1
xfce4-pulseaudio-plugin 0.4.2-2

$ pulseaudio --dump-conf
### Read from configuration file: /etc/pulse/daemon.conf ###
daemonize = no
fail = yes
high-priority = yes
nice-level = -11
realtime-scheduling = yes
realtime-priority = 5
allow-module-loading = yes
allow-exit = yes
use-pid-file = yes
system-instance = no
local-server-type = user
cpu-limit = no
enable-shm = yes
flat-volumes = no
lock-memory = no
exit-idle-time = 20
scache-idle-time = 20
dl-search-path = /usr/lib/pulse-12.99/modules
default-script-file = /etc/pulse/default.pa
load-default-script-file = yes
log-target = 
log-level = notice
resample-method = auto
avoid-resampling = no
enable-remixing = yes
remixing-use-all-sink-channels = yes
enable-lfe-remixing = yes
lfe-crossover-freq = 250
default-sample-format = s16le
default-sample-rate = 44100
alternate-sample-rate = 48000
default-sample-channels = 3
default-channel-map = front-left,front-right,front-center
default-fragments = 4
default-fragment-size-msec = 25
enable-deferred-volume = yes
deferred-volume-safety-margin-usec = 8000
deferred-volume-extra-delay-usec = 0
shm-size-bytes = 0
log-meta = no
log-time = no
log-backtrace = 0
rlimit-fsize = -1
rlimit-data = -1
rlimit-stack = -1
rlimit-core = -1
rlimit-rss = -1
rlimit-as = -1
rlimit-nproc = -1
rlimit-nofile = 256
rlimit-memlock = -1
rlimit-locks = -1
rlimit-sigpending = -1
rlimit-msgqueue = -1
rlimit-nice = 31
rlimit-rtprio = 9
rlimit-rttime = 200000

$ pacmd list-sinks
1 sink(s) available.
  * index: 8
	name: <alsa_output.pci-0000_00_1f.3.analog-surround-21>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
	state: RUNNING
	suspend cause: (none)
	priority: 9039
	volume: front-left: 26090 /  40% / -24.00 dB,   front-right: 26090 /  40% / -24.00 dB,   lfe: 26090 /  40% / -24.00 dB
	        balance 0.00
	base volume: 65536 / 100% / 0.00 dB
	volume steps: 65537
	muted: no
	current latency: 683.84 ms
	max request: 275 KiB
	max rewind: 275 KiB
	monitor source: 10
	sample spec: s16le 3ch 48000Hz
	channel map: front-left,front-right,lfe
	used by: 1
	linked by: 2
	configured latency: 980.00 ms; range is 0.50 .. 2000.00 ms
	card: 0 <alsa_card.pci-0000_00_1f.3>
	module: 6
	properties:
		alsa.resolution_bits = "16"
		device.api = "alsa"
		device.class = "sound"
		alsa.class = "generic"
		alsa.subclass = "generic-mix"
		alsa.name = "ALC285 Analog"
		alsa.id = "ALC285 Analog"
		alsa.subdevice = "0"
		alsa.subdevice_name = "subdevice #0"
		alsa.device = "0"
		alsa.card = "0"
		alsa.card_name = "HDA Intel PCH"
		alsa.long_card_name = "HDA Intel PCH at 0xea23c000 irq 168"
		alsa.driver_name = "snd_hda_intel"
		device.bus_path = "pci-0000:00:1f.3"
		sysfs.path = "/devices/pci0000:00/0000:00:1f.3/sound/card0"
		device.bus = "pci"
		device.vendor.id = "8086"
		device.vendor.name = "Intel Corporation"
		device.product.id = "9dc8"
		device.product.name = "Cannon Point-LP High Definition Audio Controller"
		device.form_factor = "internal"
		device.string = "surround21:0"
		device.buffering.buffer_size = "576000"
		device.buffering.fragment_size = "288000"
		device.access_mode = "mmap+timer"
		device.profile.name = "analog-surround-21"
		device.profile.description = "Analog Surround 2.1"
		device.description = "Built-in Audio Analog Surround 2.1"
		alsa.mixer_name = "Realtek ALC285"
		alsa.components = "HDA:10ec0285,17aa2293,00100002 HDA:8086280b,80860101,00100000"
		module-udev-detect.discovered = "1"
		device.icon_name = "audio-card-pci"
	ports:
		analog-output-speaker: Speakers (priority 10000, latency offset 0 usec, available: unknown)
			properties:
				device.icon_name = "audio-speakers"
	active port: <analog-output-speaker>

Thanks for any tips,
-f

V1del · 2019-09-14 10:35:59

What are you using to react to "Volume up/down"? if you do that with alsamixer/amixer you are using a mixer that's not guaranteed to reflect pulse state properly. Literally from this example you are on 40% sink volume across the board.

pactl set-sink-volume alsa_output.pci-0000_00_1f.3.analog-surround-21 80%

rosseba · 2019-09-14 16:25:43

V1del wrote:

What are you using to react to "Volume up/down"?

Right now I'm using the default xfce pulseaudio volume slider in the panel.
Using it I can reach 150% of the volume, above it the sound become clipped or distorted.
Volume up/down just controls the alsa Master channel, alsa Speaker and PCM channels are 100% stable.
Again bass freqs are always present, Volume up/down control high freqs only.
Lenovo forums talk about sound improvements in kernel 5.3 and SOF: https://forums.lenovo.com/t5/Ubuntu/Gui … -p/4489823 do you agree?
Regards,
-f

Arch Linux

#1 2019-09-13 21:26:08

X1C7 Pulseaudio and subwoofer

#2 2019-09-14 10:35:59

Re: X1C7 Pulseaudio and subwoofer

#3 2019-09-14 16:25:43

Re: X1C7 Pulseaudio and subwoofer

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