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Hello all,
I just noticed that playback stop working on my machine on the three above mentioned browsers. On Firefox sound works perfectly. I have no idea when it stopped working and no idea on how to debug the issue.
The only thing I can say is that while playing sound, e.g. Youtube, no playback application appears in pavucontrol and this error is printed into the terminal used to launch for example chrome:
[4383:4383:0528/213237.753746:ERROR:audio_manager_base.cc(363)] Invalid audio output parameters received; using fake audio path: format: 1, channel_layout: 3, channels: 2, sample_rate: 1600, frames_per_buffer: 1024, effects: 0, mic_positions:
which looks very suspicious but googling it did not bring anything interesting.
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During attempted playback, what's your output for
sudo fuser -v /dev/snd/*
aplay -lL
pacmd list-sinks
pacmd list-sink-inputs
any custom .asoundrc/asound.conf and/or pulse adjustments?
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Hi!
Outputs below. Only adjustments to pulseaudio are for the webcam mic and aredefault-sample-rate = 1600 in daemon.conf and set-default-source alsa_input.usb-046d_0825_09769740-02.analog-mono in default.pa.
It really looks like Chrome/Qutebrowser/Chromium do not pick the correct pulseaudio server channel somehow.
~ % sudo fuser -v /dev/snd/*
[sudo] password for user:
USER PID ACCESS COMMAND
/dev/snd/controlC0: user 16212 F.... pulseaudio
/dev/snd/controlC1: user 16212 F.... pulseaudio
/dev/snd/controlC2: user 16212 F.... pulseaudio
~ % aplay -lL
null
Discard all samples (playback) or generate zero samples (capture)
jack
JACK Audio Connection Kit
pulse
PulseAudio Sound Server
default
Default ALSA Output (currently PulseAudio Sound Server)
sysdefault:CARD=SB
HDA ATI SB, ALC892 Analog
Default Audio Device
front:CARD=SB,DEV=0
HDA ATI SB, ALC892 Analog
Front speakers
surround21:CARD=SB,DEV=0
HDA ATI SB, ALC892 Analog
2.1 Surround output to Front and Subwoofer speakers
surround40:CARD=SB,DEV=0
HDA ATI SB, ALC892 Analog
4.0 Surround output to Front and Rear speakers
surround41:CARD=SB,DEV=0
HDA ATI SB, ALC892 Analog
4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=SB,DEV=0
HDA ATI SB, ALC892 Analog
5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=SB,DEV=0
HDA ATI SB, ALC892 Analog
5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=SB,DEV=0
HDA ATI SB, ALC892 Analog
7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
iec958:CARD=SB,DEV=0
HDA ATI SB, ALC892 Digital
IEC958 (S/PDIF) Digital Audio Output
usbstream:CARD=SB
HDA ATI SB
USB Stream Output
hdmi:CARD=HDMI,DEV=0
HDA ATI HDMI, HDMI 0
HDMI Audio Output
hdmi:CARD=HDMI,DEV=1
HDA ATI HDMI, HDMI 1
HDMI Audio Output
hdmi:CARD=HDMI,DEV=2
HDA ATI HDMI, HDMI 2
HDMI Audio Output
hdmi:CARD=HDMI,DEV=3
HDA ATI HDMI, HDMI 3
HDMI Audio Output
hdmi:CARD=HDMI,DEV=4
HDA ATI HDMI, HDMI 4
HDMI Audio Output
usbstream:CARD=HDMI
HDA ATI HDMI
USB Stream Output
usbstream:CARD=U0x46d0x825
USB Device 0x46d:0x825
USB Stream Output
**** List of PLAYBACK Hardware Devices ****
card 0: SB [HDA ATI SB], device 0: ALC892 Analog [ALC892 Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: SB [HDA ATI SB], device 1: ALC892 Digital [ALC892 Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: HDMI [HDA ATI HDMI], device 3: HDMI 0 [HDMI 0]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: HDMI [HDA ATI HDMI], device 7: HDMI 1 [HDMI 1]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: HDMI [HDA ATI HDMI], device 8: HDMI 2 [HDMI 2]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: HDMI [HDA ATI HDMI], device 9: HDMI 3 [HDMI 3]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: HDMI [HDA ATI HDMI], device 10: HDMI 4 [HDMI 4]
Subdevices: 1/1
Subdevice #0: subdevice #0
~ % pacmd list-sinks
1 sink(s) available.
* index: 0
name: <alsa_output.pci-0000_00_14.2.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: 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 request: 0 KiB
max rewind: 0 KiB
monitor source: 1
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 .. 1999.82 ms
card: 2 <alsa_card.pci-0000_00_14.2>
module: 8
properties:
alsa.resolution_bits = "16"
device.api = "alsa"
device.class = "sound"
alsa.class = "generic"
alsa.subclass = "generic-mix"
alsa.name = "ALC892 Analog"
alsa.id = "ALC892 Analog"
alsa.subdevice = "0"
alsa.subdevice_name = "subdevice #0"
alsa.device = "0"
alsa.card = "0"
alsa.card_name = "HDA ATI SB"
alsa.long_card_name = "HDA ATI SB at 0xfeb00000 irq 16"
alsa.driver_name = "snd_hda_intel"
device.bus_path = "pci-0000:00:14.2"
sysfs.path = "/devices/pci0000:00/0000:00:14.2/sound/card0"
device.bus = "pci"
device.vendor.id = "1002"
device.vendor.name = "Advanced Micro Devices, Inc. [AMD/ATI]"
device.product.id = "4383"
device.product.name = "SBx00 Azalia (Intel HDA)"
device.form_factor = "internal"
device.string = "front:0"
device.buffering.buffer_size = "352768"
device.buffering.fragment_size = "176384"
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 ALC892"
alsa.components = "HDA:10ec0892,10438436,00100302"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-pci"
ports:
analog-output-lineout: Line Out (priority 9900, latency offset 0 usec, available: yes)
properties:
analog-output-headphones: Headphones (priority 9000, latency offset 0 usec, available: no)
properties:
device.icon_name = "audio-headphones"
active port: <analog-output-lineout>
~ % pacmd list-sink-inputs
0 sink input(s) available.
Last edited by positronik (2019-05-29 16:35:44)
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Well yes, don't adjust your default sample rate to that low of a value to fix a mic device, that doesn't make much sense. switch that back to 44100 and if anything, switch the alternate-sampling-rate. And you likely rather want to use 16000 and not 1600
Probably even better than setting this globally would be to use e.g. the avoid-resampling=true option so that pulse should be more proactive in simply picking supported sample rates or if you really absolutely have to, define a custom alsa-card with the preferred sample rate explicitly for the mic.
Or in general, why are you mucking around with this in the first place? Are you sure you need to set this?
Last edited by V1del (2019-05-29 17:53:41)
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So, I removed all custom pulseaudio configs and now sound seems to be working!
I would have never expected that some options like that one where related to the issue, thanks a lot!
BTW I had to play around with the sampling rate because otherwise I was getting the 'chipmunk' sound when recording. Now it seems that with plain config the issue is not there, but unfortunately I remember that the issue appears randomly.
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