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#1 2022-01-30 06:18:48

superuserdada
Member
Registered: 2022-01-05
Posts: 10

ALSA static noise microphone

While recording audio there is a static background noise while using external and internal microhone both
    Audio sample : https://voca.ro/1cfFs2eKwagt


 
$pacmd list-sources
2 source(s) available.
    index: 0
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo.monitor>
	driver: <module-alsa-card.c>
	flags: DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
	state: SUSPENDED
	suspend cause: IDLE
	priority: 1030
	volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
	        balance 0.00
	base volume: 65536 / 100% / 0.00 dB
	volume steps: 65537
	muted: no
	current latency: 0.00 ms
	max rewind: 0 KiB
	sample spec: s16le 2ch 44100Hz
	channel map: front-left,front-right
	             Stereo
	used by: 0
	linked by: 0
	configured latency: 0.00 ms; range is 0.50 .. 2000.00 ms
	monitor_of: 0
	card: 1 <alsa_card.pci-0000_00_1b.0>
	module: 7
	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 0xf1a10000 irq 30"
		alsa.driver_name = "snd_hda_intel"
		device.bus_path = "pci-0000:00:1b.0"
		sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0"
		device.bus = "pci"
		device.vendor.id = "8086"
		device.vendor.name = "Intel Corporation"
		device.product.id = "1e20"
		device.product.name = "7 Series/C216 Chipset Family High Definition Audio Controller (LIFEBOOK E752)"
		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_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
	state: SUSPENDED
	suspend cause: IDLE
	priority: 9039
	volume: front-left: 20726 /  32% / -30.00 dB,   front-right: 20726 /  32% / -30.00 dB
	        balance 0.00
	base volume: 6554 /  10% / -60.00 dB
	volume steps: 65537
	muted: no
	current latency: 0.00 ms
	max rewind: 0 KiB
	sample spec: s16le 2ch 44100Hz
	channel map: front-left,front-right
	             Stereo
	used by: 0
	linked by: 0
	configured latency: 0.00 ms; range is 0.50 .. 2000.00 ms
	card: 1 <alsa_card.pci-0000_00_1b.0>
	module: 7
	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 0xf1a10000 irq 30"
		alsa.driver_name = "snd_hda_intel"
		device.bus_path = "pci-0000:00:1b.0"
		sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0"
		device.bus = "pci"
		device.vendor.id = "8086"
		device.vendor.name = "Intel Corporation"
		device.product.id = "1e20"
		device.product.name = "7 Series/C216 Chipset Family High Definition Audio Controller (LIFEBOOK E752)"
		device.form_factor = "internal"
		device.string = "front:0"
		device.buffering.buffer_size = "352800"
		device.buffering.fragment_size = "176400"
		device.access_mode = "mmap+timer"
		device.profile.name = "analog-stereo"
		device.profile.description = "Analog Stereo"
		device.description = "Built-in Audio Analog Stereo"
		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"
		analog-input-aux: Analog Input (priority 8000, latency offset 0 usec, available: unknown)
			properties:
				
	active port: <analog-input-mic>
$ arecord -lL
null
    Discard all samples (playback) or generate zero samples (capture)
lavrate
    Rate Converter Plugin Using Libav/FFmpeg Library
samplerate
    Rate Converter Plugin Using Samplerate Library
speexrate
    Rate Converter Plugin Using Speex Resampler
jack
    JACK Audio Connection Kit
oss
    Open Sound System
pipewire
    PipeWire Sound Server
pulse
    PulseAudio Sound Server
speex
    Plugin using Speex DSP (resample, agc, denoise, echo, dereverb)
upmix
    Plugin for channel upmix (4,6,8)
vdownmix
    Plugin for channel downmix (stereo) with a simple spacialization
default
    Default ALSA Output (currently PulseAudio Sound Server)
sysdefault:CARD=PCH
    HDA Intel PCH, Generic Analog
    Default Audio Device
front:CARD=PCH,DEV=0
    HDA Intel PCH, Generic Analog
    Front output / input
usbstream:CARD=PCH
    HDA Intel PCH
    USB Stream Output
usbstream:CARD=NVidia
    HDA NVidia
    USB Stream Output
**** List of CAPTURE Hardware Devices ****
card 0: PCH [HDA Intel PCH], device 0: Generic Analog [Generic Analog]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
fuser -v /dev/snd/*
 USER        PID ACCESS COMMAND
/dev/snd/controlC0:  will        468 F.... pulseaudio
/dev/snd/controlC1:  will        468 F.... pulseaudio

Both external and internal microphones worked fine on windows previously.

What I've tried:
https://wiki.archlinux.org/title/PulseA … _recording

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#2 2022-01-30 11:29:07

V1del
Forum Moderator
Registered: 2012-10-16
Posts: 25,292

Re: ALSA static noise microphone

The sample rate is almost never the problem here. One thing to try is a potential misrepresentation of the channel counts since normally a mic input is simply not stereo, try completely muting the left-/or the right channel of the mic source and check whether this improves things.

Another good candidate is a too high mic boost, check "alsamixer -c0" and turn down a potential mic boost (test all the way to 0 and instead bump the volume of the actual mic source).

Somewhat more obscure, broken power_save handling, try 

echo 0 | sudo tee -a /sys/module/snd_hda_intel/parameters/power_save

to disable the power save default.

If that doesn't help post the outputs of 

amixer -c0
pacmd list-sources
pacmd list-source-outputs

during an active recording attempt.

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#3 2022-01-30 14:51:39

superuserdada
Member
Registered: 2022-01-05
Posts: 10

Re: ALSA static noise microphone

Muting one of the two channels dint help and the microphone boost is set to zero.

echo 0 | sudo tee -a /sys/module/snd_hda_intel/parameters/power_save

and rebooting dint help either .

Here are the outputs of following commands while actively recording in audacity :


$amixer -c0
 
Simple mixer control 'Master',0
  Capabilities: pvolume pvolume-joined pswitch pswitch-joined
  Playback channels: Mono
  Limits: Playback 0 - 87
  Mono: Playback 57 [66%] [-22.50dB] [on]
Simple mixer control 'Headphone',0
  Capabilities: pvolume pswitch
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 87
  Mono:
  Front Left: Playback 87 [100%] [0.00dB] [on]
  Front Right: Playback 87 [100%] [0.00dB] [on]
Simple mixer control 'Speaker',0
  Capabilities: pvolume pswitch
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 87
  Mono:
  Front Left: Playback 0 [0%] [-65.25dB] [off]
  Front Right: Playback 0 [0%] [-65.25dB] [off]
Simple mixer control 'PCM',0
  Capabilities: pvolume
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 255
  Mono:
  Front Left: Playback 255 [100%] [0.00dB]
  Front Right: Playback 255 [100%] [0.00dB]
Simple mixer control 'Mic',0
  Capabilities: pvolume pswitch
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 0 [0%] [-34.50dB] [off]
  Front Right: Playback 0 [0%] [-34.50dB] [off]
Simple mixer control 'Mic Boost',0
  Capabilities: volume
  Playback channels: Front Left - Front Right
  Capture channels: Front Left - Front Right
  Limits: 0 - 3
  Front Left: 0 [0%] [0.00dB]
  Front Right: 0 [0%] [0.00dB]
Simple mixer control 'IEC958',0
  Capabilities: pswitch pswitch-joined
  Playback channels: Mono
  Mono: Playback [off]
Simple mixer control 'Beep',0
  Capabilities: pvolume pswitch
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 31
  Mono:
  Front Left: Playback 0 [0%] [-34.50dB] [off]
  Front Right: Playback 0 [0%] [-34.50dB] [off]
Simple mixer control 'Capture',0
  Capabilities: cvolume cswitch
  Capture channels: Front Left - Front Right
  Limits: Capture 0 - 63
  Front Left: Capture 63 [100%] [30.00dB] [on]
  Front Right: Capture 63 [100%] [30.00dB] [on]
Simple mixer control 'Auto-Mute Mode',0
  Capabilities: enum
  Items: 'Disabled' 'Enabled'
  Item0: 'Disabled'
Simple mixer control 'Digital',0
  Capabilities: cvolume
  Capture channels: Front Left - Front Right
  Limits: Capture 0 - 120
  Front Left: Capture 0 [0%] [-30.00dB]
  Front Right: Capture 0 [0%] [-30.00dB]
Simple mixer control 'Internal Mic Boost',0
  Capabilities: volume
  Playback channels: Front Left - Front Right
  Capture channels: Front Left - Front Right
  Limits: 0 - 3
  Front Left: 0 [0%] [0.00dB]
  Front Right: 0 [0%] [0.00dB]
Simple mixer control 'Loopback Mixing',0
  Capabilities: enum
  Items: 'Disabled' 'Enabled'
  Item0: 'Disabled
 
$pacmd list-sources
 
2 source(s) available.
    index: 0
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo.monitor>
	driver: <module-alsa-card.c>
	flags: DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
	state: SUSPENDED
	suspend cause: IDLE
	priority: 1030
	volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
	        balance 0.00
	base volume: 65536 / 100% / 0.00 dB
	volume steps: 65537
	muted: yes
	current latency: 0.00 ms
	max rewind: 0 KiB
	sample spec: s16le 2ch 44100Hz
	channel map: front-left,front-right
	             Stereo
	used by: 0
	linked by: 0
	configured latency: 0.00 ms; range is 0.50 .. 2000.00 ms
	monitor_of: 0
	card: 1 <alsa_card.pci-0000_00_1b.0>
	module: 7
	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 0xf1a10000 irq 30"
		alsa.driver_name = "snd_hda_intel"
		device.bus_path = "pci-0000:00:1b.0"
		sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0"
		device.bus = "pci"
		device.vendor.id = "8086"
		device.vendor.name = "Intel Corporation"
		device.product.id = "1e20"
		device.product.name = "7 Series/C216 Chipset Family High Definition Audio Controller (LIFEBOOK E752)"
		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_1b.0.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: 22957 /  35% / -27.33 dB,   front-right: 22957 /  35% / -27.33 dB
	        balance 0.00
	base volume: 6554 /  10% / -60.00 dB
	volume steps: 65537
	muted: no
	current latency: 2.40 ms
	max rewind: 0 KiB
	sample spec: s16le 2ch 44100Hz
	channel map: front-left,front-right
	             Stereo
	used by: 1
	linked by: 1
	configured latency: 24.99 ms; range is 0.50 .. 2000.00 ms
	card: 1 <alsa_card.pci-0000_00_1b.0>
	module: 7
	properties:
		alsa.resolution_bits = "16"
		device.api = "alsa"
		device.class = "sound"
		alsa.class = "generic"
		alsa.subclass = "generic-mix"
		alsa.name = "ALC269VC Analog"
		alsa.id = "ALC269VC 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 0xf1a10000 irq 30"
		alsa.driver_name = "snd_hda_intel"
		device.bus_path = "pci-0000:00:1b.0"
		sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0"
		device.bus = "pci"
		device.vendor.id = "8086"
		device.vendor.name = "Intel Corporation"
		device.product.id = "1e20"
		device.product.name = "7 Series/C216 Chipset Family High Definition Audio Controller (LIFEBOOK E752)"
		device.form_factor = "internal"
		device.string = "front:0"
		device.buffering.buffer_size = "352800"
		device.buffering.fragment_size = "176400"
		device.access_mode = "mmap+timer"
		device.profile.name = "analog-stereo"
		device.profile.description = "Analog Stereo"
		device.description = "Built-in Audio Analog Stereo"
		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>
$pacmd list-source-outputs
 
1 source output(s) available.
    index: 46
	driver: <protocol-native.c>
	flags: 
	state: RUNNING
	source: 1 <alsa_input.pci-0000_00_1b.0.analog-stereo>
	volume: mono: 65536 / 100% / 0.00 dB
	        balance 0.00
	muted: no
	current latency: 0.00 ms
	requested latency: 24.99 ms
	sample spec: float32le 1ch 44100Hz
	channel map: mono
	             Mono
	resample method: copy
	owner module: 9
	client: 84 <ALSA plug-in [audacity]>
	properties:
		media.name = "ALSA Capture"
		application.name = "ALSA plug-in [audacity]"
		native-protocol.peer = "UNIX socket client"
		native-protocol.version = "35"
		application.process.id = "643"
		application.process.user = "will"
		application.process.host = "chroma"
		application.process.binary = "audacity"
		application.language = "C"
		window.x11.display = ":0"
		application.process.machine_id = "10498068ce20442b951afb6d77ee59c8"
		application.process.session_id = "1"
		application.icon_name = "audacity"
		module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [audacity]"

Last edited by superuserdada (2022-01-30 14:52:46)

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#4 2022-01-30 14:59:30

V1del
Forum Moderator
Registered: 2012-10-16
Posts: 25,292

Re: ALSA static noise microphone

The power_save command is for runtime adjustment, if you reboot it's reset again.

alsa-plugin based recordings can often be dicey, try something that directly uses pulse e.g. parecord or so or in a browser. 2.40ms latency is plain unrealistic, do things improve if you start audacity/your recording application wiht the PULSE_LATENCY_MSEC=60 argument? i.e.  PULSE_LATENCY_MSEC=60 audacity

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#5 2022-01-30 15:11:34

superuserdada
Member
Registered: 2022-01-05
Posts: 10

Re: ALSA static noise microphone

Power-save command doesn't help. Recorded audio using parecord but its the same. Things do not improve by launching applications by changing latency either.

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