[SOLVED] Pulse, Teamspeak, USB Mic doesn't work, USB low bandwith

deafeningsylence · 2016-10-18 16:16:34

Dear Arch Community,

I am desperate. We are talking pulseaudio.

My USB Mic (Samson Meteor Mic) is not recording any input, althout it is being recognised by the system and can be chosen as input source.

I tried manually setting it as default source in the pulse audio config (/etc/pulse/default.pa), changing output from stereo to mono via pavucontrol and via the pulseaudio daemon.conf (/etc/pulse/daemon.conf).
I tried various combinations of device configuration (analog input, anaolog duplex, digital duplex) in pavucontrol.
I unmuted the shit out of the system.
I tried biting the microphone.
I resignated.
I cried.
Help.

Can't I just use my microphone, be it in teamspeak or google hangout? :'(

I sincerely hope, somebody can point me towards something I have not tried yet.

Maybe install the git version instead? Would that change a lot?

All the best,
deafeningsylence

PS: I attached the pulseaudio sources list and the same for alsa for further analysis, I could not find anything there.

[silence@arch ~]$ pacmd list-sources
5 source(s) available.
    index: 0
        name: <alsa_input.usb-046d_HD_Webcam_C525_D4488030-00.analog-mono>
        driver: <module-alsa-card.c>
        flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
        state: RUNNING
        suspend cause: 
        priority: 9049
        volume: mono: 49144 /  75% / -7.50 dB
                balance 0.00
        base volume: 20724 /  32% / -30.00 dB
        volume steps: 65537
        muted: no
        current latency: 0.00 ms
        max rewind: 0 KiB
        sample spec: s16le 1ch 48000Hz
        channel map: mono
                     Mono
        used by: 1
        linked by: 1
        configured latency: 30.00 ms; range is 0.50 .. 2000.00 ms
        card: 0 <alsa_card.usb-046d_HD_Webcam_C525_D4488030-00>
        module: 6
        properties:
                alsa.resolution_bits = "16"
                device.api = "alsa"
                device.class = "sound"
                alsa.class = "generic"
                alsa.subclass = "generic-mix"
                alsa.name = "USB Audio"
                alsa.id = "USB Audio"
                alsa.subdevice = "0"
                alsa.subdevice_name = "subdevice #0"
                alsa.device = "0"
                alsa.card = "0"
                alsa.card_name = "HD Webcam C525"
                alsa.long_card_name = "HD Webcam C525 at usb-0000:00:1a.0-1.4, high speed"
                alsa.driver_name = "snd_usb_audio"
                device.bus_path = "pci-0000:00:1a.0-usb-0:1.4:1.0"
                sysfs.path = "/devices/pci0000:00/0000:00:1a.0/usb1/1-1/1-1.4/1-1.4:1.0/sound/card0"
                udev.id = "usb-046d_HD_Webcam_C525_D4488030-00"
                device.bus = "usb"
                device.vendor.id = "046d"
                device.vendor.name = "Logitech, Inc."
                device.product.id = "0826"
                device.product.name = "HD Webcam C525"
                device.serial = "046d_HD_Webcam_C525_D4488030"
                device.form_factor = "webcam"
                device.string = "hw:0"
                device.buffering.buffer_size = "192000"
                device.buffering.fragment_size = "96000"
                device.access_mode = "mmap+timer"
                device.profile.name = "analog-mono"
                device.profile.description = "Analog Mono"
                device.description = "HD Webcam C525 Analog Mono"
                alsa.mixer_name = "USB Mixer"
                alsa.components = "USB046d:0826"
                module-udev-detect.discovered = "1"
                device.icon_name = "camera-web-usb"
        ports:
                analog-input-mic: Microphone (priority 8700, latency offset 0 usec, available: unknown)
                        properties:
                                device.icon_name = "audio-input-microphone"
        active port: <analog-input-mic>
    index: 3
        name: <alsa_output.pci-0000_06_01.0.analog-stereo.monitor>
        driver: <module-alsa-card.c>
        flags: DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
        state: IDLE
        suspend cause: 
        priority: 1050
        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: 344 KiB
        sample spec: s16le 2ch 44100Hz
        channel map: front-left,front-right
                     Stereo
        used by: 0
        linked by: 0
        configured latency: 2000.00 ms; range is 0.50 .. 2000.00 ms
        monitor_of: 1
        card: 2 <alsa_card.pci-0000_06_01.0>
        module: 8
        properties:
                device.description = "Monitor of CMI8788 [Oxygen HD Audio] (CMI8786 (Xonar DG)) Analog Stereo"
                device.class = "monitor"
                alsa.card = "1"
                alsa.card_name = "Xonar DG"
                alsa.long_card_name = "C-Media Oxygen HD Audio at 0xb000, irq 18"
                alsa.driver_name = "snd_oxygen"
                device.bus_path = "pci-0000:06:01.0"
                sysfs.path = "/devices/pci0000:00/0000:00:1c.5/0000:05:00.0/0000:06:01.0/sound/card1"
                device.bus = "pci"
                device.vendor.id = "13f6"
                device.vendor.name = "C-Media Electronics Inc"
                device.product.id = "8788"
                device.product.name = "CMI8788 [Oxygen HD Audio] (CMI8786 (Xonar DG))"
                device.string = "1"
                module-udev-detect.discovered = "1"
                device.icon_name = "audio-card-pci"
  * index: 4
        name: <alsa_input.pci-0000_06_01.0.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: 9059
        volume: front-left: 26090 /  40% / -24.00 dB,   front-right: 26090 /  40% / -24.00 dB
                balance 0.00
        base volume: 41350 /  63% / -12.00 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 .. 743.04 ms
        card: 2 <alsa_card.pci-0000_06_01.0>
        module: 8
        properties:
                alsa.resolution_bits = "16"
                device.api = "alsa"
                device.class = "sound"
                alsa.class = "generic"
                alsa.subclass = "generic-mix"
                alsa.name = "Multichannel"
                alsa.id = "Multichannel"
                alsa.subdevice = "0"
                alsa.subdevice_name = "subdevice #0"
                alsa.device = "0"
                alsa.card = "1"
                alsa.card_name = "Xonar DG"
                alsa.long_card_name = "C-Media Oxygen HD Audio at 0xb000, irq 18"
                alsa.driver_name = "snd_oxygen"
                device.bus_path = "pci-0000:06:01.0"
                sysfs.path = "/devices/pci0000:00/0000:00:1c.5/0000:05:00.0/0000:06:01.0/sound/card1"
                device.bus = "pci"
                device.vendor.id = "13f6"
                device.vendor.name = "C-Media Electronics Inc"
                device.product.id = "8788"
                device.product.name = "CMI8788 [Oxygen HD Audio] (CMI8786 (Xonar DG))"
                device.string = "front:1"
                device.buffering.buffer_size = "131072"
                device.buffering.fragment_size = "131072"
                device.access_mode = "mmap+timer"
                device.profile.name = "analog-stereo"
                device.profile.description = "Analog Stereo"
                device.description = "CMI8788 [Oxygen HD Audio] (CMI8786 (Xonar DG)) Analog Stereo"
                alsa.mixer_name = "CMI8786"
                alsa.components = "CS4245 CMI8786"
                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: unknown)
                        properties:
                                device.icon_name = "audio-input-microphone"
                analog-input-mic: Microphone (priority 8700, latency offset 0 usec, available: unknown)
                        properties:
                                device.icon_name = "audio-input-microphone"
                analog-input-linein: Line In (priority 8100, latency offset 0 usec, available: unknown)
                        properties:

                analog-input-aux: Analog Input (priority 8000, latency offset 0 usec, available: unknown)
                        properties:

        active port: <analog-input-aux>
    index: 8
        name: <alsa_output.usb-Samson_Technologies_Samson_Meteor_Mic-00.analog-stereo.monitor>
        driver: <module-alsa-card.c>
        flags: DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
        state: SUSPENDED
        suspend cause: IDLE 
        priority: 1040
        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: 4
        card: 1 <alsa_card.usb-Samson_Technologies_Samson_Meteor_Mic-00>
        module: 7
        properties:
                device.description = "Monitor of Meteor condenser microphone Analog Stereo"
                device.class = "monitor"
                alsa.card = "2"
                alsa.card_name = "Samson Meteor Mic"
                alsa.long_card_name = "Samson Technologies Samson Meteor Mic at usb-0000:00:1a.0-1.5, full speed"
                alsa.driver_name = "snd_usb_audio"
                device.bus_path = "pci-0000:00:1a.0-usb-0:1.5:1.0"
                sysfs.path = "/devices/pci0000:00/0000:00:1a.0/usb1/1-1/1-1.5/1-1.5:1.0/sound/card2"
                udev.id = "usb-Samson_Technologies_Samson_Meteor_Mic-00"
                device.bus = "usb"
                device.vendor.id = "17a0"
                device.vendor.name = "Samson Technologies Corp."
                device.product.id = "0310"
                device.product.name = "Meteor condenser microphone"
                device.serial = "Samson_Technologies_Samson_Meteor_Mic"
                device.form_factor = "microphone"
                device.string = "2"
                module-udev-detect.discovered = "1"
                device.icon_name = "audio-input-microphone-usb"
    index: 9
        name: <alsa_input.usb-Samson_Technologies_Samson_Meteor_Mic-00.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: 9049
        volume: front-left: 55596 /  85% / -4.29 dB,   front-right: 55706 /  85% / -4.23 dB
                balance 0.00
        base volume: 28172 /  43% / -22.00 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.usb-Samson_Technologies_Samson_Meteor_Mic-00>
        module: 7
        properties:
                alsa.resolution_bits = "16"
                device.api = "alsa"
                device.class = "sound"
                alsa.class = "generic"
                alsa.subclass = "generic-mix"
                alsa.name = "USB Audio"
                alsa.id = "USB Audio"
                alsa.subdevice = "0"
                alsa.subdevice_name = "subdevice #0"
                alsa.device = "0"
                alsa.card = "2"
                alsa.card_name = "Samson Meteor Mic"
                alsa.long_card_name = "Samson Technologies Samson Meteor Mic at usb-0000:00:1a.0-1.5, full speed"
                alsa.driver_name = "snd_usb_audio"
                device.bus_path = "pci-0000:00:1a.0-usb-0:1.5:1.0"
                sysfs.path = "/devices/pci0000:00/0000:00:1a.0/usb1/1-1/1-1.5/1-1.5:1.0/sound/card2"
                udev.id = "usb-Samson_Technologies_Samson_Meteor_Mic-00"
                device.bus = "usb"
                device.vendor.id = "17a0"
                device.vendor.name = "Samson Technologies Corp."
                device.product.id = "0310"
                device.product.name = "Meteor condenser microphone"
                device.serial = "Samson_Technologies_Samson_Meteor_Mic"
                device.form_factor = "microphone"
                device.string = "front:2"
                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 = "Meteor condenser microphone Analog Stereo"
                alsa.mixer_name = "USB Mixer"
                alsa.components = "USB17a0:0310"
                module-udev-detect.discovered = "1"
                device.icon_name = "audio-input-microphone-usb"
        ports:
                analog-input-mic: Microphone (priority 8700, latency offset 0 usec, available: unknown)
                        properties:
                                device.icon_name = "audio-input-microphone"
        active port: <analog-input-mic>

And the output of arecord -l

[silence@arch ~]$ arecord -l
**** List of CAPTURE Hardware Devices ****
card 0: C525 [HD Webcam C525], device 0: USB Audio [USB Audio]
  Subdevices: 0/1
  Subdevice #0: subdevice #0
card 1: DG [Xonar DG], device 0: Multichannel [Multichannel]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 1: DG [Xonar DG], device 1: Digital [Digital]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 2: Mic [Samson Meteor Mic], device 0: USB Audio [USB Audio]
  Subdevices: 1/1
  Subdevice #0: subdevice #0

Last edited by deafeningsylence (2016-10-19 12:58:07)

deafeningsylence · 2016-10-18 18:47:45

Update:

I think it might have to do something with ALSA rather than pulseaudio. In Teamspeak there is no error at all when I choose the USB Mic, just no sound is recorded.
Now when I select ALSA and not Pulseaudio and then the USB Microphone, I get an error message: "Could not open capture device.". When I use fuser for the alsa usb device I get an error as well, the device being busy.

So my theory now is, ALSA somehow cannot open the USB Mic and therefore pulseaudio cannot grab it from ALSA. However, so far I had no success in playing around wir ALSA to get the Mic to work.

Ideas appreciated.

All the best,

deafeningsylence

deafeningsylence · 2016-10-19 11:00:47

Update 2:

So, I am back to assuming it is a pulseaudio issue.

I ran a few commands to figure out why Teamspeak "cannot open capture device" and at least got that message to disappear by removing alsa and pulseaudio configs.

arecord hw:2,0 -d10 /tmp/test-mic.wav

[silence@arch ~]$ arecord hw:2,0 -d10 /tmp/test-mic.wav
ALSA lib pcm_dsnoop.c:618:(snd_pcm_dsnoop_open) unable to open slave
arecord: main:786: audio open error: Device or resource busy

ls /proc/asound/card*/pcm*/sub*/status

[silence@arch ~]$ ls /proc/asound/card*/pcm*/sub*/status
/proc/asound/card0/pcm0c/sub0/status /proc/asound/card0/pcm1p/sub0/status /proc/asound/card2/pcm0p/sub0/status
/proc/asound/card0/pcm0p/sub0/status /proc/asound/card1/pcm0c/sub0/status
/proc/asound/card0/pcm1c/sub0/status /proc/asound/card2/pcm0c/sub0/status

grep owner_pid /proc/asound/card*/pcm*/sub*/status

[silence@arch ~]$ grep owner_pid /proc/asound/card*/pcm*/sub*/status
/proc/asound/card0/pcm0c/sub0/status:owner_pid : 1060
/proc/asound/card0/pcm0p/sub0/status:owner_pid : 1059
/proc/asound/card1/pcm0c/sub0/status:owner_pid : 1052
/proc/asound/card2/pcm0c/sub0/status:owner_pid : 1057
/proc/asound/card2/pcm0p/sub0/status:owner_pid : 1055

Then I checked these PIDs in top and checked what is using them and it is pulseaudio. So pulseaudio is using the mic after all, but still does not get any input out of it.
In pavucontrol it shows the microphone as well as its monitor in input devices. Though also there no bar is shown when I speak, as opposed to my webcam mic right above it.
Microphone configuration is Analog Duplex, which is the correct one.

pavucontrol

http://imgur.com/a/UBhda
And why the hell can't I post this image.

In the process I tried different combinations of setting default sources in ~/.asoundrc and ~/.config/pulse/default.pa. Even after this nothing changed, so I reverted it to detecting all the devices automatically (aka delete the two files), which at least made the Teamspeak error "could not open capture device" disappear.

Maybe now someone has an idea?

I really do not understand what pulseaudio is doing to the mic, that it has no input.
I also do not know where I can search for any error messages regarding this. Although, maybe there are none, since no errors are shown.

Regards,

deafeningsylence

Last edited by deafeningsylence (2016-10-19 11:02:57)

V1del · 2016-10-19 11:14:36

Slow down, take a breather... Teamspeak can use pulseaudio natively, tell Teamspeak to use pulseaudio and try to record and then post

pacmd list-source-outputs
sudo fuser -v /dev/snd/*
amixer -c2

also note that the default source in pulseaudio (ie. the source that gets used if a new program that pulse hasn't seen yet (if it has seen it already because you tried to use it, it will use whatever source it was last on) opens up a source stream gets pointed to) is set to your internal card as opposed to your USB headset use

pacmd set-default-source alsa_input.usb-Samson_Technologies_Samson_Meteor_Mic-00.analog-stereo

to change that, if you at the moment, have any modifications to the pulse files or ALSA configs (~/.asoundrc, /etc/asound.conf for example) please post them as well

deafeningsylence · 2016-10-19 11:45:53

Hey V1del,

first of all, thank you for your help

Here are the outputs you asked. I already tried setting the mic to default in alsa (asoundrc) as well as in pulse (default.pa), however the problem is not that the device does not show up in teamspeak or anywhere else, the problem is that it does not record any sound.

First the list of pulseaudio sources.

[silence@arch ~]$ pacmd list-source-outputs
7 source output(s) available.
    index: 0
        driver: <protocol-native.c>
        flags: DONT_MOVE 
        state: RUNNING
        source: 0 <alsa_input.usb-046d_HD_Webcam_C525_D4488030-00.analog-mono>
        volume: mono: 65536 / 100% / 0.00 dB
                balance 0.00
        muted: no
        current latency: 0.00 ms
        requested latency: 40.00 ms
        sample spec: float32le 1ch 25Hz
        channel map: mono
                     Mono
        resample method: peaks
        owner module: 10
        client: 7 <PulseAudio Volume Control>
        properties:
                media.name = "Peak detect"
                application.name = "PulseAudio Volume Control"
                native-protocol.peer = "UNIX socket client"
                native-protocol.version = "31"
                application.id = "org.PulseAudio.pavucontrol"
                application.icon_name = "audio-card"
                application.version = "3.0"
                application.process.id = "1318"
                application.process.user = "silence"
                application.process.host = "arch"
                application.process.binary = "pavucontrol"
                application.language = "C"
                window.x11.display = ":0"
                application.process.machine_id = "6a3acac23aef43f79c17bb05b0955d9e"
                application.process.session_id = "c1"
                module-stream-restore.id = "source-output-by-application-id:org.PulseAudio.pavucontrol"
    index: 1
        driver: <protocol-native.c>
        flags: DONT_MOVE 
        state: RUNNING
        source: 1 <alsa_output.usb-Samson_Technologies_Samson_Meteor_Mic-00.analog-stereo.monitor>
        volume: mono: 65536 / 100% / 0.00 dB
                balance 0.00
        muted: no
        current latency: 22.47 ms
        requested latency: 40.00 ms
        sample spec: float32le 1ch 25Hz
        channel map: mono
                     Mono
        resample method: peaks
        owner module: 10
        client: 7 <PulseAudio Volume Control>
        properties:
                media.name = "Peak detect"
                application.name = "PulseAudio Volume Control"
                native-protocol.peer = "UNIX socket client"
                native-protocol.version = "31"
                application.id = "org.PulseAudio.pavucontrol"
                application.icon_name = "audio-card"
                application.version = "3.0"
                application.process.id = "1318"
                application.process.user = "silence"
                application.process.host = "arch"
                application.process.binary = "pavucontrol"
                application.language = "C"
                window.x11.display = ":0"
                application.process.machine_id = "6a3acac23aef43f79c17bb05b0955d9e"
                application.process.session_id = "c1"
                module-stream-restore.id = "source-output-by-application-id:org.PulseAudio.pavucontrol"
    index: 2
        driver: <protocol-native.c>
        flags: DONT_MOVE 
        state: RUNNING
        source: 2 <alsa_input.usb-Samson_Technologies_Samson_Meteor_Mic-00.analog-stereo>
        volume: mono: 65536 / 100% / 0.00 dB
                balance 0.00
        muted: no
        current latency: 0.00 ms
        requested latency: 40.00 ms
        sample spec: float32le 1ch 25Hz
        channel map: mono
                     Mono
        resample method: peaks
        owner module: 10
        client: 7 <PulseAudio Volume Control>
        properties:
                media.name = "Peak detect"
                application.name = "PulseAudio Volume Control"
                native-protocol.peer = "UNIX socket client"
                native-protocol.version = "31"
                application.id = "org.PulseAudio.pavucontrol"
                application.icon_name = "audio-card"
                application.version = "3.0"
                application.process.id = "1318"
                application.process.user = "silence"
                application.process.host = "arch"
                application.process.binary = "pavucontrol"
                application.language = "C"
                window.x11.display = ":0"
                application.process.machine_id = "6a3acac23aef43f79c17bb05b0955d9e"
                application.process.session_id = "c1"
                module-stream-restore.id = "source-output-by-application-id:org.PulseAudio.pavucontrol"
    index: 3
        driver: <protocol-native.c>
        flags: DONT_MOVE 
        state: RUNNING
        source: 3 <alsa_output.pci-0000_06_01.0.analog-stereo.monitor>
        volume: mono: 65536 / 100% / 0.00 dB
                balance 0.00
        muted: no
        current latency: 3.63 ms
        requested latency: 40.00 ms
        sample spec: float32le 1ch 25Hz
        channel map: mono
                     Mono
        resample method: peaks
        owner module: 10
        client: 7 <PulseAudio Volume Control>
        properties:
                media.name = "Peak detect"
                application.name = "PulseAudio Volume Control"
                native-protocol.peer = "UNIX socket client"
                native-protocol.version = "31"
                application.id = "org.PulseAudio.pavucontrol"
                application.icon_name = "audio-card"
                application.version = "3.0"
                application.process.id = "1318"
                application.process.user = "silence"
                application.process.host = "arch"
                application.process.binary = "pavucontrol"
                application.language = "C"
                window.x11.display = ":0"
                application.process.machine_id = "6a3acac23aef43f79c17bb05b0955d9e"
                application.process.session_id = "c1"
                module-stream-restore.id = "source-output-by-application-id:org.PulseAudio.pavucontrol"
    index: 4
        driver: <protocol-native.c>
        flags: DONT_MOVE 
        state: RUNNING
        source: 4 <alsa_input.pci-0000_06_01.0.analog-stereo>
        volume: mono: 65536 / 100% / 0.00 dB
                balance 0.00
        muted: no
        current latency: 0.00 ms
        requested latency: 40.00 ms
        sample spec: float32le 1ch 25Hz
        channel map: mono
                     Mono
        resample method: peaks
        owner module: 10
        client: 7 <PulseAudio Volume Control>
        properties:
                media.name = "Peak detect"
                application.name = "PulseAudio Volume Control"
                native-protocol.peer = "UNIX socket client"
                native-protocol.version = "31"
                application.id = "org.PulseAudio.pavucontrol"
                application.icon_name = "audio-card"
                application.version = "3.0"
                application.process.id = "1318"
                application.process.user = "silence"
                application.process.host = "arch"
                application.process.binary = "pavucontrol"
                application.language = "C"
                window.x11.display = ":0"
                application.process.machine_id = "6a3acac23aef43f79c17bb05b0955d9e"
                application.process.session_id = "c1"
                module-stream-restore.id = "source-output-by-application-id:org.PulseAudio.pavucontrol"
    index: 17
        driver: <protocol-native.c>
        flags: DONT_MOVE 
        state: RUNNING
        source: 3 <alsa_output.pci-0000_06_01.0.analog-stereo.monitor>
        volume: mono: 65536 / 100% / 0.00 dB
                balance 0.00
        muted: no
        current latency: 3.65 ms
        requested latency: 40.00 ms
        sample spec: float32le 1ch 25Hz
        channel map: mono
                     Mono
        resample method: peaks
        owner module: 10
        client: 7 <PulseAudio Volume Control>
        direct on input: 10
        properties:
                media.name = "Peak detect"
                application.name = "PulseAudio Volume Control"
                native-protocol.peer = "UNIX socket client"
                native-protocol.version = "31"
                application.id = "org.PulseAudio.pavucontrol"
                application.icon_name = "audio-card"
                application.version = "3.0"
                application.process.id = "1318"
                application.process.user = "silence"
                application.process.host = "arch"
                application.process.binary = "pavucontrol"
                application.language = "C"
                window.x11.display = ":0"
                application.process.machine_id = "6a3acac23aef43f79c17bb05b0955d9e"
                application.process.session_id = "c1"
                module-stream-restore.id = "source-output-by-application-id:org.PulseAudio.pavucontrol"
    index: 37
        driver: <protocol-native.c>
        flags: START_CORKED 
        state: RUNNING
        source: 2 <alsa_input.usb-Samson_Technologies_Samson_Meteor_Mic-00.analog-stereo>
        volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
                balance 0.00
        muted: no
        current latency: 0.00 ms
        requested latency: 30.00 ms
        sample spec: s16le 2ch 48000Hz
        channel map: front-left,front-right
                     Stereo
        resample method: speex-float-1
        owner module: 10
        client: 9 <TeamSpeak3>
        properties:
                media.name = "alsa_input.usb-Samson_Technologies_Samson_Meteor_Mic-00.analog-stereocapture23"
                application.name = "TeamSpeak3"
                native-protocol.peer = "UNIX socket client"
                native-protocol.version = "31"
                media.role = "phone"
                media.icon_name = "teamspeak3"
                application.process.id = "1346"
                application.process.user = "silence"
                application.process.host = "arch"
                application.process.binary = "ts3client_linux_amd64"
                application.language = "C"
                window.x11.display = ":0"
                application.process.machine_id = "6a3acac23aef43f79c17bb05b0955d9e"
                application.process.session_id = "c1"
                module-stream-restore.id = "source-output-by-media-role:phone"

The output for fuser.

                     USER        PID ACCESS COMMAND
/dev/snd/controlC0:  silence     951 F.... pulseaudio
/dev/snd/controlC1:  silence     951 F.... pulseaudio
/dev/snd/controlC2:  silence     951 F.... pulseaudio
/dev/snd/pcmC0D0c:   silence     951 F...m pulseaudio
/dev/snd/pcmC0D0p:   silence     951 F...m pulseaudio
/dev/snd/pcmC1D0c:   silence     951 F...m pulseaudio
/dev/snd/pcmC2D0c:   silence     951 F...m pulseaudio
/dev/snd/pcmC2D0p:   silence     951 F...m pulseaudio

Then the ouput for amixer -c2, aka the Mic.

[silence@arch ~]$ amixer -c2
Simple mixer control 'Speaker',0
  Capabilities: pvolume pswitch pswitch-joined
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 40
  Mono:
  Front Left: Playback 40 [100%] [0.00dB] [on]
  Front Right: Playback 40 [100%] [0.00dB] [on]
Simple mixer control 'Mic',0
  Capabilities: pvolume cvolume pswitch pswitch-joined cswitch cswitch-joined
  Playback channels: Front Left - Front Right
  Capture channels: Front Left - Front Right
  Limits: Playback 0 - 60 Capture 0 - 26
  Front Left: Playback 60 [100%] [30.00dB] [on] Capture 18 [69%] [14.00dB] [on]
  Front Right: Playback 60 [100%] [30.00dB] [on] Capture 18 [69%] [14.00dB] [on]

And the content of /etc/pulse/default.pa, I do not have ~/.asoundrc anymore since I deleted it, after creating and editing it did not help.

#!/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:2,0
#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

### 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

###Changes I tried
#set-default-source input alsa_output.usb-Samson_Technologies_Samson_Meteor_Mic-00.analog-stereo.monitor
#set-default-source input alsa_input.usb-Samson_Technologies_Samson_Meteor_Mic-00.analog-stereo
load-module module-device-manager
#set-default-source input alsa_input.hw_2_0
#load-module module-alsa-sink
#load-module module-alsa-source device=hw:2,0

I hope these logs help you.

Also I checked journalctl and found this, when testing the Mic under pulseaudio specifically in teamspeak settings. Maybe it is some kernel issue? I am using the linux-vfio kernel, which is outdated atm, however only very little and the Mic worked on an earlier install on the same kernel.

Oct 19 13:11:35 arch konsole[1093]: org.kde.kurifilter-ikws: "Oct 19 11:05:01 arch kernel: usb 1-1.5: cannot submit

Thanks again!

Last edited by deafeningsylence (2016-10-19 11:53:53)

V1del · 2016-10-19 12:37:20

A quick googler for that message (would be good to have some more context to this, look through dmesg if there are more lines regarding the USB) show that one issue might be the USB hub into which the headset is plugged, as it seems that it requires full power. use lsusb -v to verify if any other device is on the same bus and make sure that the USB mic has one 2.0 bus to itself. Another thing, the base volume for the microphone seems to be configured quite low, use

alsamixer -c2 #Verify with aplay -l that it is still card 2 and the number hasn't switched around through udev

to increase that base volume. Another thing, though unlikely as teamspeak is the corking stream, is to comment out the load-module module-role-cork to make sure that your mic isnt' accidentally muted due to some weird interpretation error.

Finally if none of that helped, try with the default kernel, it's quite possible that there have been relevant fixes, or maybe even a simple recompile of your current kernel might help, due to some compiler differences.

deafeningsylence · 2016-10-19 12:53:42

It wooorks! Wow, I cannot believe what the issue was.

It seems the usb I plugged it into had not enough bandwith, although it is the back of my computer, there should be enough power for about every device I put in there. Next time I will buy a Gigabyte motherboard or EVGA and not an apparently low quality MSI board.

So yes, just switching it to an USB 3.0 port together with my mouse (I figured a mouse does not need much) fixed this.

Thank you so much V1del. I thought about that this could be it before, but I refused to believe it, since it is my freaking motherboard I plugged it in. Unbelievable really.

Thanks again!

Last edited by deafeningsylence (2016-10-19 12:59:04)

V1del · 2016-10-19 13:01:46

Well just because a usb port is at the back of the motherboard doesn't really have to mean anything. If it is an old 1.0 hub those are usually too slow for most modern requirements. As mentioned lsusb will show you which device is hooked to which bus, and if the usb you plugged it into happens to be a 1.0 bus (that might have even more other devices hanging on it) then the result isn't that surprising. However glad to hear that this was solvable with this.

deafeningsylence · 2016-10-19 13:20:04

Very true,
however, there should be only USB 2.0 and USB 3.0 bus's on that motherboard. The mic probably hang on a bus with my keyboard that also functions as a hub for one usb, I figure that this took all the bandwith.

[silence@arch ~]$ lsusb
Bus 004 Device 002: ID 8087:0024 Intel Corp. Integrated Rate Matching Hub
Bus 004 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub
Bus 006 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub
Bus 005 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub
Bus 003 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub
Bus 002 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub
Bus 001 Device 011: ID 046d:c24d Logitech, Inc. G710 Gaming Keyboard
Bus 001 Device 017: ID 046d:c080 Logitech, Inc. 
Bus 001 Device 016: ID 17a0:0310 Samson Technologies Corp. Meteor condenser microphone
Bus 001 Device 014: ID 046d:c227 Logitech, Inc. G15 Refresh Keyboard
Bus 001 Device 013: ID 046d:c226 Logitech, Inc. G15 Refresh Keyboard
Bus 001 Device 012: ID 046d:c223 Logitech, Inc. G11/G15 Keyboard / USB Hub
Bus 001 Device 015: ID 046d:0826 Logitech, Inc. HD Webcam C525
Bus 001 Device 002: ID 8087:0024 Intel Corp. Integrated Rate Matching Hub
Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub

Here it says that I have 6 USB bus'es. Although all the devices seem to be on bus 1. Apparently the switching did not change anything about the mic being on bus 1, just another device of the same bus.
Is that normal? Since I do have 3.0 USBs and 2.0 USBs I figured these would be on different bus'es.
This is what it looked like before I switched USBs.

Bus 004 Device 002: ID 8087:0024 Intel Corp. Integrated Rate Matching Hub
Bus 004 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub
Bus 006 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub
Bus 005 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub
Bus 003 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub
Bus 002 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub
Bus 001 Device 018: ID 17a0:0310 Samson Technologies Corp. Meteor condenser microphone
Bus 001 Device 017: ID 046d:c080 Logitech, Inc. 
Bus 001 Device 014: ID 046d:c227 Logitech, Inc. G15 Refresh Keyboard
Bus 001 Device 013: ID 046d:c226 Logitech, Inc. G15 Refresh Keyboard
Bus 001 Device 012: ID 046d:c223 Logitech, Inc. G11/G15 Keyboard / USB Hub
Bus 001 Device 015: ID 046d:0826 Logitech, Inc. HD Webcam C525
Bus 001 Device 002: ID 8087:0024 Intel Corp. Integrated Rate Matching Hub
Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub

So yeah probably I should have assumed something more hardware related earlier. It is just that one tends to think very software-sided when solving Linux problems, at least when one is a newbie .

Arch Linux

#1 2016-10-18 16:16:34

[SOLVED] Pulse, Teamspeak, USB Mic doesn't work, USB low bandwith

#2 2016-10-18 18:47:45

Re: [SOLVED] Pulse, Teamspeak, USB Mic doesn't work, USB low bandwith

#3 2016-10-19 11:00:47

Re: [SOLVED] Pulse, Teamspeak, USB Mic doesn't work, USB low bandwith

#4 2016-10-19 11:14:36

Re: [SOLVED] Pulse, Teamspeak, USB Mic doesn't work, USB low bandwith

#5 2016-10-19 11:45:53

Re: [SOLVED] Pulse, Teamspeak, USB Mic doesn't work, USB low bandwith

#6 2016-10-19 12:37:20

Re: [SOLVED] Pulse, Teamspeak, USB Mic doesn't work, USB low bandwith

#7 2016-10-19 12:53:42

Re: [SOLVED] Pulse, Teamspeak, USB Mic doesn't work, USB low bandwith

#8 2016-10-19 13:01:46

Re: [SOLVED] Pulse, Teamspeak, USB Mic doesn't work, USB low bandwith

#9 2016-10-19 13:20:04

Re: [SOLVED] Pulse, Teamspeak, USB Mic doesn't work, USB low bandwith

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