PulseAudio Настройка микрофона

Доброго времени суток, ув.форумчане, достаточно давно бьюсь с своим арчиком, все не могу настроить микрофон, когда-то давно на старом ноутбуке сделал за пару минут, а на новой машине вот уже 3-й заход был и все безуспешно, очень надеюсь на вашу помощь.
Сейчас при записи
arecord -f cd -d 10 test-mic.wav
слышу небольшой шум, звука с микрофона нет вообще, гуляю по pavucontrol Input/Recording вкладке пробовал разные вариции не помогло ;(
Ноут ideapad 320 Lenovo
PulseAudio 12.2
Kernel: x86_64 Linux 4.17.11-arch1

Параметры запуска
GRUB_CMDLINE_LINUX_DEFAULT="intel_iommu=igfx_off"

╰─$ inxi -A
Audio:     Card-1: Intel Sunrise Point-LP HD Audio driver: snd_hda_intel
           Sound Server: ALSA v: k4.17.11-arch1
pacmd list-sources

3 source(s) available.
  * index: 0
	name: <alsa_output.pci-0000_00_1f.3.analog-stereo.monitor>
	driver: <module-alsa-card.c>
	flags: DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
	state: RUNNING
	suspend cause: (none)
	priority: 1030
	volume: front-left: 53199 /  81% / -5.43 dB,   front-right: 53199 /  81% / -5.43 dB
	        balance 0.00
	base volume: 65536 / 100% / 0.00 dB
	volume steps: 65537
	muted: no
	current latency: 0.00 ms
	max rewind: 4 KiB
	sample spec: s16le 2ch 48000Hz
	channel map: front-left,front-right
	             Stereo
	used by: 2
	linked by: 2
	configured latency: 40.00 ms; range is 0.50 .. 2000.00 ms
	monitor_of: 0
	card: 0 <alsa_card.pci-0000_00_1f.3>
	module: 6
	properties:
		device.description = "Monitor of Built-in Audio Analog Stereo"
		device.class = "monitor"
		alsa.card = "0"
		alsa.card_name = "HDA Intel PCH"
		alsa.long_card_name = "HDA Intel PCH at 0x94220000 irq 132"
		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 = "9d71"
		device.product.name = "Sunrise Point-LP HD Audio"
		device.form_factor = "internal"
		device.string = "0"
		module-udev-detect.discovered = "1"
		device.icon_name = "audio-card-pci"
    index: 1
	name: <alsa_input.pci-0000_00_1f.3.analog-stereo>
	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: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
	        balance 0.00
	base volume: 6554 /  10% / -60.00 dB
	volume steps: 65537
	muted: no
	current latency: 0.34 ms
	max rewind: 0 KiB
	sample spec: s16le 2ch 48000Hz
	channel map: front-left,front-right
	             Stereo
	used by: 2
	linked by: 2
	configured latency: 40.00 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 = "Generic Analog"
		alsa.id = "Generic 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 0x94220000 irq 132"
		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 = "9d71"
		device.product.name = "Sunrise Point-LP HD Audio"
		device.form_factor = "internal"
		device.string = "front:0"
		device.buffering.buffer_size = "384000"
		device.buffering.fragment_size = "192000"
		device.access_mode = "mmap+timer"
		device.profile.name = "analog-stereo"
		device.profile.description = "Analog Stereo"
		device.description = "Built-in Audio Analog Stereo"
		alsa.mixer_name = "Realtek Generic"
		alsa.components = "HDA:10ec0230,17aa3804,00100002 HDA:8086280b,80860101,00100000"
		module-udev-detect.discovered = "1"
		device.icon_name = "audio-card-pci"
	ports:
		analog-input-internal-mic: Internal Microphone (priority 8900, latency offset 0 usec, available: no)
			properties:
				device.icon_name = "audio-input-microphone"
		analog-input-mic: Microphone (priority 8700, latency offset 0 usec, available: yes)
			properties:
				device.icon_name = "audio-input-microphone"
	active port: <analog-input-mic>
    index: 2
	name: <record_mono>
	driver: <module-remap-source.c>
	flags: DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
	state: RUNNING
	suspend cause: (none)
	priority: 1000
	volume: mono: 65536 / 100% / 0.00 dB
	        balance 0.00
	base volume: 65536 / 100% / 0.00 dB
	volume steps: 65537
	muted: no
	current latency: 0.65 ms
	max rewind: 0 KiB
	sample spec: s16le 1ch 48000Hz
	channel map: mono
	             Mono
	used by: 1
	linked by: 1
	configured latency: 40.00 ms; range is 0.50 .. 2000.00 ms
	module: 20
	properties:
		device.master_device = "alsa_input.pci-0000_00_1f.3.analog-stereo"
		device.class = "filter"
		device.description = "Remapped Built-in Audio Analog Stereo"
		device.icon_name = "audio-input-microphone"

cat /etc/pulse/default.pa
#!/usr/bin/pulseaudio -nF
#
# This file is part of PulseAudio.
#
# PulseAudio is free software; you can redistribute it and/or modify it
# under the terms of the GNU Lesser General Public License as published by
# the Free Software Foundation; either version 2 of the License, or
# (at your option) any later version.
#
# PulseAudio is distributed in the hope that it will be useful, but
# WITHOUT ANY WARRANTY; without even the implied warranty of
# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
# General Public License for more details.
#
# You should have received a copy of the GNU Lesser General Public License
# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>.

# This startup script is used only if PulseAudio is started per-user
# (i.e. not in system mode)

.fail

### Automatically restore the volume of streams and devices
load-module module-device-restore
load-module module-stream-restore
load-module module-card-restore

### Automatically augment property information from .desktop files
### stored in /usr/share/application
load-module module-augment-properties

### Should be after module-*-restore but before module-*-detect
load-module module-switch-on-port-available

### Load audio drivers statically
### (it's probably better to not load these drivers manually, but instead
### use module-udev-detect -- see below -- for doing this automatically)
#load-module module-alsa-sink
#load-module module-alsa-source device=hw:0,0
#load-module module-oss device="/dev/dsp" sink_name=output source_name=input
#load-module module-oss-mmap device="/dev/dsp" sink_name=output source_name=input
#load-module module-null-sink
#load-module module-pipe-sink
#load-module module-alsa-source device=hw:0,1
### Automatically load driver modules depending on the hardware available
.ifexists module-udev-detect.so
load-module module-udev-detect
.else
### Use the static hardware detection module (for systems that lack udev support)
load-module module-detect
.endif

### Automatically connect sink and source if JACK server is present
.ifexists module-jackdbus-detect.so
.nofail
load-module module-jackdbus-detect channels=2
.fail
.endif

### Automatically load driver modules for Bluetooth hardware
.ifexists module-bluetooth-policy.so
load-module module-bluetooth-policy
.endif

.ifexists module-bluetooth-discover.so
load-module module-bluetooth-discover
.endif

### Load several protocols
.ifexists module-esound-protocol-unix.so
load-module module-esound-protocol-unix
.endif
load-module module-native-protocol-unix

### Network access (may be configured with paprefs, so leave this commented
### here if you plan to use paprefs)
#load-module module-esound-protocol-tcp
#load-module module-native-protocol-tcp
#load-module module-zeroconf-publish

### Load the RTP receiver module (also configured via paprefs, see above)
#load-module module-rtp-recv

### Load the RTP sender module (also configured via paprefs, see above)
#load-module module-null-sink sink_name=rtp format=s16be channels=2 rate=44100 sink_properties="device.description='RTP Multicast Sink'"
#load-module module-rtp-send source=rtp.monitor

### Load additional modules from GConf settings. This can be configured with the paprefs tool.
### Please keep in mind that the modules configured by paprefs might conflict with manually
### loaded modules.
.ifexists module-gconf.so
.nofail
load-module module-gconf
.fail
.endif

### Automatically restore the default sink/source when changed by the user
### during runtime
### NOTE: This should be loaded as early as possible so that subsequent modules
### that look up the default sink/source get the right value
load-module module-default-device-restore

### Automatically move streams to the default sink if the sink they are
### connected to dies, similar for sources
load-module module-rescue-streams

### Make sure we always have a sink around, even if it is a null sink.
load-module module-always-sink

### Honour intended role device property
load-module module-intended-roles

### Automatically suspend sinks/sources that become idle for too long
load-module module-suspend-on-idle

### If autoexit on idle is enabled we want to make sure we only quit
### when no local session needs us anymore.
.ifexists module-console-kit.so
load-module module-console-kit
.endif
.ifexists module-systemd-login.so
load-module module-systemd-login
.endif

### Enable positioned event sounds
load-module module-position-event-sounds

### Cork music/video streams when a phone stream is active
load-module module-role-cork

### Modules to allow autoloading of filters (such as echo cancellation)
### on demand. module-filter-heuristics tries to determine what filters
### make sense, and module-filter-apply does the heavy-lifting of
### loading modules and rerouting streams.
load-module module-filter-heuristics
load-module module-filter-apply

### Make some devices default
#set-default-sink output
#set-default-source input
#load-module module-echo-cancel use_master_format=1 aec_method=webrtc aec_args="analog_gain_control=0 digital_gain_control=1" source_name=echoCancel_source sink_name=echoCancel_sink
#set-default-source echoCancel_source
#set-default-sink echoCancel_sink

load-module module-remap-source source_name=record_mono master=alsa_input.pci-0000_00_1f.3.analog-stereo master_channel_map=front-left channel_map=mono
#load-module module-remap-source master=alsa_input.pci-0000_00_1f.3.analog-stereo master_channel_map=front-left,front-right channels=2 channel_map=mono,mono
set-default-source record_mono
load-module module-udev-detect tsched=0

arecord -f dat -r 60000 -D hw:0,0 -d 5 test.wav
любезно показывает 48000, что и было любезно установленно в конфиг
cat /etc/pulse/daemon.conf

# This file is part of PulseAudio.
#
# PulseAudio is free software; you can redistribute it and/or modify
# it under the terms of the GNU Lesser General Public License as published by
# the Free Software Foundation; either version 2 of the License, or
# (at your option) any later version.
#
# PulseAudio is distributed in the hope that it will be useful, but
# WITHOUT ANY WARRANTY; without even the implied warranty of
# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
# General Public License for more details.
#
# You should have received a copy of the GNU Lesser General Public License
# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>.

## Configuration file for the PulseAudio daemon. See pulse-daemon.conf(5) for
## more information. Default values are commented out.  Use either ; or # for
## commenting.

; daemonize = no
; fail = yes
; allow-module-loading = yes
; allow-exit = yes
; use-pid-file = yes
; system-instance = no
; local-server-type = user
; enable-shm = yes
; enable-memfd = yes
; shm-size-bytes = 0 # setting this 0 will use the system-default, usually 64 MiB
; lock-memory = no
; cpu-limit = no

; high-priority = yes
; nice-level = -11

; realtime-scheduling = yes
; realtime-priority = 5

; exit-idle-time = 20
; scache-idle-time = 20

; dl-search-path = (depends on architecture)

; load-default-script-file = yes
; default-script-file = /etc/pulse/default.pa

; log-target = auto
; log-level = notice
; log-meta = no
; log-time = no
; log-backtrace = 0

; resample-method = speex-float-1
avoid-resampling = yes
; enable-remixing = yes
; remixing-use-all-sink-channels = yes
; enable-lfe-remixing = no
; lfe-crossover-freq = 0

flat-volumes = no
; flat-volumes = yes

; rlimit-fsize = -1
; rlimit-data = -1
; rlimit-stack = -1
; rlimit-core = -1
; rlimit-as = -1
; rlimit-rss = -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

; default-sample-format = s16le
default-sample-rate = 48000
; alternate-sample-rate = 48000
; default-sample-channels = 2
; default-channel-map = front-left,front-right

; default-fragments = 4
; default-fragment-size-msec = 25

; enable-deferred-volume = yes
; deferred-volume-safety-margin-usec = 8000
; deferred-volume-extra-delay-usec = 0

https://i.imgur.com/PoiT8tt.jpg
https://i.imgur.com/CpbjNRZ.jpg
https://i.imgur.com/X1IH5pQ.jpg

з.ы надеюсь не промахнулся с разделом для темы
`--> pacman -Ql pulseaudio-alsa
pulseaudio-alsa /etc/
pulseaudio-alsa /etc/asound.conf
а вот такое имеется?
kefzce, попробуй вытащить наушники при прослушивании после записи.

EDIT 1 -
kefzce
любезно показывает 48000, что и было любезно установленно в конфиг
при настройке лучше все оставить по дефолту
Ошибки не исчезают с опытом - они просто умнеют
safocl
`--> pacman -Ql pulseaudio-alsa
pulseaudio-alsa /etc/
pulseaudio-alsa /etc/asound.conf
а вот такое имеется?
Да, прошу прощение за долгий ответ.
кстати есть вероятность аппаратного касяка либо подключения микрофона, либо обрыва проводков идущих к нему, если я правильно понял чо микр встроенный...
 
Зарегистрироваться или войдите чтобы оставить сообщение.