Участник с: 03 августа 2018
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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
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