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Using a Hyperx cloud core on asrock b350 mobo. Others sound find to me through my headset but they report static from my mic is audible to them. I've added
default-sample-rate = 48000
resample-method = src-sinc-best-qualityto the bottom of
/etc/pulse/daemon.confand added
use_ucm=0 tsched=0to the end of
load-module module-udev-detectin
/etc/pulse/default.paEven with
load-module module-remap-source source_name=record_mono master=alsa_input.pci-0000_28_00.3.analog-stereo master_channel_map=front-left channel_map=monoset-default-source record_mono
from $ pacmd list-sources | grep 'name:.*input'
name: <alsa_input.pci-0000_28_00.3.analog-stereo>at the end of default.conf I can hear the static in the playback.
This is using discord by the way, is it just a side effect of discord on linux. I see numerous other topics with the same issue but they just say the above changes fixed it where its still there for anyone I talk to.
Last edited by cj360 (2018-10-13 00:47:59)
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Have you tried without doing these changes? Those are emergency options for fixing broken drivers and are not generally necessary nor recommended. src-sinc-best-quality doesn't make sense either way (even if you indeed were to fix something with e.g. tsched=0), as that resampler has been removed from the pulse package. use speex-float-5 or something. FWIW post
pacmd list-sources
pacmd list-source-outputsduring crackling playback.
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$ pacmd list-sources
3 source(s) available.
index: 0
name: <alsa_output.pci-0000_26_00.1.hdmi-stereo-extra1.monitor>
driver: <module-alsa-card.c>
flags: DECIBEL_VOLUME 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 48000Hz
channel map: front-left,front-right
Stereo
used by: 0
linked by: 0
fixed latency: 100.00 ms
monitor_of: 0
card: 0 <alsa_card.pci-0000_26_00.1>
module: 6
properties:
device.description = "Monitor of GP104 High Definition Audio Controller Digital Stereo (HDMI 2)"
device.class = "monitor"
alsa.card = "1"
alsa.card_name = "HDA NVidia"
alsa.long_card_name = "HDA NVidia at 0xf7080000 irq 52"
alsa.driver_name = "snd_hda_intel"
device.bus_path = "pci-0000:26:00.1"
sysfs.path = "/devices/pci0000:00/0000:00:03.1/0000:26:00.1/sound/card1"
device.bus = "pci"
device.vendor.id = "10de"
device.vendor.name = "NVIDIA Corporation"
device.product.id = "10f0"
device.product.name = "GP104 High Definition Audio Controller"
device.string = "1"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-pci"
index: 1
name: <alsa_output.pci-0000_28_00.3.analog-stereo.monitor>
driver: <module-alsa-card.c>
flags: DECIBEL_VOLUME LATENCY
state: IDLE
suspend cause: (none)
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: 18 KiB
sample spec: s16le 2ch 48000Hz
channel map: front-left,front-right
Stereo
used by: 0
linked by: 0
fixed latency: 100.00 ms
monitor_of: 1
card: 1 <alsa_card.pci-0000_28_00.3>
module: 7
properties:
device.description = "Monitor of Family 17h (Models 00h-0fh) HD Audio Controller Analog Stereo"
device.class = "monitor"
alsa.card = "0"
alsa.card_name = "HD-Audio Generic"
alsa.long_card_name = "HD-Audio Generic at 0xf7800000 irq 54"
alsa.driver_name = "snd_hda_intel"
device.bus_path = "pci-0000:28:00.3"
sysfs.path = "/devices/pci0000:00/0000:00:08.1/0000:28:00.3/sound/card0"
device.bus = "pci"
device.vendor.id = "1022"
device.vendor.name = "Advanced Micro Devices, Inc. [AMD]"
device.product.id = "1457"
device.product.name = "Family 17h (Models 00h-0fh) HD Audio Controller"
device.string = "0"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-pci"
* index: 2
name: <alsa_input.pci-0000_28_00.3.analog-stereo>
driver: <module-alsa-card.c>
flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY
state: SUSPENDED
suspend cause: IDLE
priority: 9039
volume: front-left: 10387 / 16% / -48.00 dB, front-right: 10387 / 16% / -48.00 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 48000Hz
channel map: front-left,front-right
Stereo
used by: 0
linked by: 0
fixed latency: 100.00 ms
card: 1 <alsa_card.pci-0000_28_00.3>
module: 7
properties:
alsa.resolution_bits = "16"
device.api = "alsa"
device.class = "sound"
alsa.class = "generic"
alsa.subclass = "generic-mix"
alsa.name = "ALC1220 Analog"
alsa.id = "ALC1220 Analog"
alsa.subdevice = "0"
alsa.subdevice_name = "subdevice #0"
alsa.device = "0"
alsa.card = "0"
alsa.card_name = "HD-Audio Generic"
alsa.long_card_name = "HD-Audio Generic at 0xf7800000 irq 54"
alsa.driver_name = "snd_hda_intel"
device.bus_path = "pci-0000:28:00.3"
sysfs.path = "/devices/pci0000:00/0000:00:08.1/0000:28:00.3/sound/card0"
device.bus = "pci"
device.vendor.id = "1022"
device.vendor.name = "Advanced Micro Devices, Inc. [AMD]"
device.product.id = "1457"
device.product.name = "Family 17h (Models 00h-0fh) HD Audio Controller"
device.string = "front:0"
device.buffering.buffer_size = "19200"
device.buffering.fragment_size = "3840"
device.access_mode = "mmap"
device.profile.name = "analog-stereo"
device.profile.description = "Analog Stereo"
device.description = "Family 17h (Models 00h-0fh) HD Audio Controller Analog Stereo"
alsa.mixer_name = "Realtek ALC1220"
alsa.components = "HDA:10ec1220,18491221,00100003"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-pci"
ports:
analog-input-front-mic: Front Microphone (priority 8500, latency offset 0 usec, available: no)
properties:
device.icon_name = "audio-input-microphone"
analog-input-rear-mic: Rear Microphone (priority 8200, latency offset 0 usec, available: yes)
properties:
device.icon_name = "audio-input-microphone"
analog-input-linein: Line In (priority 8100, latency offset 0 usec, available: no)
properties:
active port: <analog-input-rear-mic>$ pacmd list-source-outputs
0 source output(s) available.I can hear static along with my voice with the default pulseaudio setup. The only change that I see I should keep is setting the rate to the correct value:
$ arecord --list-devices
**** List of CAPTURE Hardware Devices ****
card 0: Generic [HD-Audio Generic], device 0: ALC1220 Analog [ALC1220 Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: Generic [HD-Audio Generic], device 2: ALC1220 Alt Analog [ALC1220 Alt Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
$ arecord -f dat -r 60000 -D hw:0,2 -d 5 test.wav
Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 60000 Hz, Stereo
Warning: rate is not accurate (requested = 60000Hz, got = 48000Hz)
please, try the plug pluginOffline
That test completely bypasses pulse, so if you already get crackling with that, your error is either on the driver or the hardware level. can you post
amixer -c0
systool -vm snd-hda-intelLast edited by V1del (2018-10-11 15:32:03)
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$ amixer -c0
Simple mixer control 'Master',0
Capabilities: pvolume pvolume-joined pswitch pswitch-joined
Playback channels: Mono
Limits: Playback 0 - 87
Mono: Playback 39 [45%] [-36.00dB] [on]
Simple mixer control 'Headphone',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 'Front',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 'Front 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 'Front 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 'Surround',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 'Center',0
Capabilities: pvolume pvolume-joined pswitch pswitch-joined
Playback channels: Mono
Limits: Playback 0 - 87
Mono: Playback 87 [100%] [0.00dB] [on]
Simple mixer control 'LFE',0
Capabilities: pvolume pvolume-joined pswitch pswitch-joined
Playback channels: Mono
Limits: Playback 0 - 87
Mono: Playback 87 [100%] [0.00dB] [on]
Simple mixer control 'Line',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 'Line 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 'IEC958 Default PCM',0
Capabilities: pswitch pswitch-joined
Playback channels: Mono
Mono: Playback [on]
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 'Capture',1
Capabilities: cvolume cswitch
Capture channels: Front Left - Front Right
Limits: Capture 0 - 63
Front Left: Capture 0 [0%] [-17.25dB] [off]
Front Right: Capture 0 [0%] [-17.25dB] [off]
Simple mixer control 'Auto-Mute Mode',0
Capabilities: enum
Items: 'Disabled' 'Enabled'
Item0: 'Enabled'
Simple mixer control 'Input Source',0
Capabilities: cenum
Items: 'Front Mic' 'Rear Mic' 'Line'
Item0: 'Rear Mic'
Simple mixer control 'Input Source',1
Capabilities: cenum
Items: 'Front Mic' 'Rear Mic' 'Line'
Item0: 'Front Mic'
Simple mixer control 'Loopback Mixing',0
Capabilities: enum
Items: 'Disabled' 'Enabled'
Item0: 'Disabled'
Simple mixer control 'Rear 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 'Rear Mic Boost',0
Capabilities: volume
Playback channels: Front Left - Front Right
Capture channels: Front Left - Front Right
Limits: 0 - 3
Front Left: 2 [67%] [20.00dB]
Front Right: 2 [67%] [20.00dB]$ systool -vm snd-hda-intel
Error opening module snd-hda-intelLast edited by cj360 (2018-10-13 00:41:07)
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Ah whoops would have to be
systool -vm snd_hda_intel. Any way, decrease the boost and up the actual volume instead:
amixer -c0 set 'Rear Mic Boost',0 0
amixer -c0 set 'Rear Mic',0 50%,50% unmute #Not sure what the exact relation between these 2 is, but still bring them down to a more sensible value
amixer -c0 set 'Capture',0 50%,50%Offline
Tested after changing amixer values, playback was still my voice with static around it. Enabling echo-canceling seems to be a solution however. Adding
/etc/pulse/default.pa:
load-module module-echo-cancel use_master_format=1 aec_method=webrtc aec_args="analog_gain_control=0 digital_gain_control=1" source_name=HyperX_echocancel sink_name=HyperX_echocancel source_properties=device.description=HyperX_echocancel
set-default-source HyperX_echocancel_source
set-default-sink HyperX_echocancel_sinkSeems to provide to clear-er recording I was looking for. Again thank you for the help and feedback.
fyi the "source_properties=" isn't needed it just makes the sink easier to read in pavucontrol.
Last edited by cj360 (2018-10-13 00:49:08)
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