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#1 2019-01-25 07:58:02

nan0s7
Member
Registered: 2018-10-28
Posts: 6

Recent package update breaks microphone usage

I noticed the other day that we had an upgrade to the webrtc package among other things, I'm not sure how to check exactly what changed but surely someone else has encountered this issue other than myself.

My basic setup is using Pulseaudio. I have a USB microphone and headset/headphones. I noticed that the public test version of Discord didn't like my microphone. What happens exactly is that it can detect that my microphone is getting a signal, but as soon as I go to hear the audio either by testing the output of the microphone being sent straight back into my headphones via Discord, or I join a chat, all output from the microphone stops. It stays active for less than a second.

Using paprefs I was able to get a loopback working, so something is working. I can't passthrough my microphone output to my virtual machine, which uses pulseaudio to do so. I've tried the usual reinstalling of packages such as pulseaudio; nothing of note has changed. To be clear, I can hear audio perfectly fine - which is probably why I didn't notice anything was wrong until today.

So my basic understanding of the issue is that whenever an application attempts to use the microphone, it just stops working. I did also check pacmd for any useful information and I couldn't find anything unusual. All my settings are the same as when it worked.

If it's already a known issue, just let me know. Thanks for your help. smile

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#2 2019-01-25 10:47:32

V1del
Forum Moderator
Registered: 2012-10-16
Posts: 6,017

Re: Recent package update breaks microphone usage

FWIW you really shouldn't expect or hope that anyone else has encountered your exact issue. Especially in audio there's a lot of different hardware that can have different reactions to things.

Discord does sometimes have weird ways to initialize the devices, does something simpler like parec work? What's your output for

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

during the 'getting a signal' and not getting a signal situation?

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#3 2019-01-25 11:38:53

nan0s7
Member
Registered: 2018-10-28
Posts: 6

Re: Recent package update breaks microphone usage

Yeah it's never usually the exact issue, but something similar. I guess it just depends.

So here's the output for when the device is clearly registering within Discord; where it shows the live microphone activity to determine the input sensitivity.

Edit: Just thought I'd let you know that the Razor Megalodon's are a set of headphones with a built-in microphone that I do not use, but I do use the headphones themselves. The microphone I use is the Blue Snowball.

 ~ $ sudo fuser -v /dev/snd/*
                     USER        PID ACCESS COMMAND
/dev/snd/controlC0:  scott       639 F.... pulseaudio
/dev/snd/controlC1:  scott       639 F.... pulseaudio
/dev/snd/controlC2:  scott       639 F.... pulseaudio
/dev/snd/pcmC0D0c:   scott       639 F...m pulseaudio
/dev/snd/pcmC1D0p:   scott       639 F...m pulseaudio
/dev/snd/pcmC2D0c:   scott       639 F...m pulseaudio 
 ~ $ pacmd list-sources
5 source(s) available.
    index: 0
	name: <alsa_output.usb-Razer_Razer_Megalodon-00.analog-surround-21.monitor>
	driver: <module-alsa-card.c>
	flags: DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
	state: RUNNING
	suspend cause: (none)
	priority: 1040
	volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB,   lfe: 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: 7 KiB
	sample spec: s16le 3ch 48000Hz
	channel map: front-left,front-right,lfe
	used by: 1
	linked by: 1
	configured latency: 26.00 ms; range is 26.00 .. 1820.44 ms
	monitor_of: 0
	card: 0 <alsa_card.usb-Razer_Razer_Megalodon-00>
	module: 6
	properties:
		device.description = "Monitor of Razer Megalodon Analog Surround 2.1"
		device.class = "monitor"
		alsa.card = "1"
		alsa.card_name = "Razer Megalodon"
		alsa.long_card_name = "Razer Razer Megalodon at usb-0000:00:14.0-3, full speed"
		alsa.driver_name = "snd_usb_audio"
		device.bus_path = "pci-0000:00:14.0-usb-0:3:1.0"
		sysfs.path = "/devices/pci0000:00/0000:00:14.0/usb1/1-3/1-3:1.0/sound/card1"
		udev.id = "usb-Razer_Razer_Megalodon-00"
		device.bus = "usb"
		device.vendor.id = "1532"
		device.vendor.name = "Razer USA, Ltd"
		device.product.id = "000e"
		device.product.name = "Razer Megalodon"
		device.serial = "Razer_Razer_Megalodon"
		device.string = "1"
		module-udev-detect.discovered = "1"
		device.icon_name = "audio-card-usb"
    index: 2
	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: 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: 6554 /  10% / -60.00 dB
	volume steps: 65537
	muted: no
	current latency: 0.00 ms
	max rewind: 0 KiB
	sample spec: s32le 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: 2 <alsa_card.pci-0000_00_1f.3>
	module: 8
	properties:
		alsa.resolution_bits = "32"
		device.api = "alsa"
		device.class = "sound"
		alsa.class = "generic"
		alsa.subclass = "generic-mix"
		alsa.name = "ALC1150 Analog"
		alsa.id = "ALC1150 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 0xdf120000 irq 133"
		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 = "a170"
		device.product.name = "100 Series/C230 Series Chipset Family HD Audio Controller"
		device.form_factor = "internal"
		device.string = "front:0"
		device.buffering.buffer_size = "705600"
		device.buffering.fragment_size = "352800"
		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 ALC1150"
		alsa.components = "HDA:10ec0900,18491151,00100001 HDA:80862809,80860101,00100000"
		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: yes)
			properties:
				device.icon_name = "audio-input-microphone"
		analog-input-rear-mic: Rear Microphone (priority 8200, latency offset 0 usec, available: no)
			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-front-mic>
    index: 3
	name: <rtp.monitor>
	driver: <module-null-sink.c>
	flags: DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
	state: RUNNING
	suspend cause: (none)
	priority: 1000
	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: 1 KiB
	sample spec: s16be 2ch 44100Hz
	channel map: front-left,front-right
	             Stereo
	used by: 1
	linked by: 1
	configured latency: 7.26 ms; range is 0.50 .. 2000.00 ms
	monitor_of: 1
	module: 27
	properties:
		device.description = "Monitor of RTP Multicast"
		device.class = "monitor"
		device.icon_name = "audio-input-microphone"
    index: 4
	name: <combined.monitor>
	driver: <module-combine-sink.c>
	flags: DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
	state: SUSPENDED
	suspend cause: IDLE
	priority: 1000
	volume: front-left: 65533 / 100% / -0.00 dB,   front-right: 65533 / 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: float32le 2ch 44100Hz
	channel map: front-left,front-right
	             Stereo
	used by: 0
	linked by: 0
	configured latency: 0.00 ms; range is 0.50 .. 200.00 ms
	monitor_of: 2
	module: 30
	properties:
		device.description = "Monitor Source of Simultaneous output to Razer Megalodon Analog Surround 2.1"
		device.class = "monitor"
		device.icon_name = "audio-input-microphone"
  * index: 6
	name: <alsa_input.usb-BLUE_MICROPHONE_Blue_Snowball_201305-00.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: 9049
	volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
	        balance 0.00
	base volume: 38295 /  58% / -14.00 dB
	volume steps: 65537
	muted: no
	current latency: 0.38 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: 10.00 ms; range is 0.50 .. 2000.00 ms
	card: 1 <alsa_card.usb-BLUE_MICROPHONE_Blue_Snowball_201305-00>
	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 = "Blue Snowball"
		alsa.long_card_name = "BLUE MICROPHONE Blue Snowball at usb-0000:00:14.0-8, full speed"
		alsa.driver_name = "snd_usb_audio"
		device.bus_path = "pci-0000:00:14.0-usb-0:8:1.0"
		sysfs.path = "/devices/pci0000:00/0000:00:14.0/usb1/1-8/1-8:1.0/sound/card2"
		udev.id = "usb-BLUE_MICROPHONE_Blue_Snowball_201305-00"
		device.bus = "usb"
		device.vendor.id = "0d8c"
		device.vendor.name = "C-Media Electronics, Inc."
		device.product.id = "0005"
		device.product.name = "Blue Snowball"
		device.serial = "BLUE_MICROPHONE_Blue_Snowball_201305"
		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 = "Blue Snowball Analog Stereo"
		alsa.mixer_name = "USB Mixer"
		alsa.components = "USB0d8c:0005"
		module-udev-detect.discovered = "1"
		device.icon_name = "audio-card-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> 
 ~ $ pacmd list-source-outputs
3 source output(s) available.
    index: 5
	driver: <protocol-native.c>
	flags: 
	state: RUNNING
	source: 0 <alsa_output.usb-Razer_Razer_Megalodon-00.analog-surround-21.monitor>
	volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
	        balance 0.00
	muted: no
	current latency: 20.85 ms
	requested latency: 10.00 ms
	sample spec: s16le 2ch 44100Hz
	channel map: front-left,front-right
	             Stereo
	resample method: speex-float-1
	owner module: 11
	client: 21 <qemu-system-x86_64>
	properties:
		media.name = "qemu"
		application.name = "qemu-system-x86_64"
		native-protocol.peer = "UNIX socket client"
		native-protocol.version = "32"
		application.process.id = "6294"
		application.process.user = "root"
		application.process.host = "scott-linux"
		application.process.binary = "qemu-system-x86_64"
		application.language = "C"
		window.x11.display = ":0"
		application.process.machine_id = "2eea978869e74d178a4ccb3d1d772c6a"
		module-stream-restore.id = "source-output-by-application-name:qemu-system-x86_64"
    index: 49
	driver: <protocol-native.c>
	flags: 
	state: RUNNING
	source: 6 <alsa_input.usb-BLUE_MICROPHONE_Blue_Snowball_201305-00.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: 10.00 ms
	sample spec: s16le 2ch 44100Hz
	channel map: front-left,front-right
	             Stereo
	resample method: (null)
	owner module: 11
	client: 20 <WEBRTC VoiceEngine>
	properties:
		media.name = "recStream"
		application.name = "WEBRTC VoiceEngine"
		native-protocol.peer = "UNIX socket client"
		native-protocol.version = "32"
		application.process.id = "3926"
		application.process.user = "scott"
		application.process.host = "scott-linux"
		application.process.binary = "DiscordPTB"
		application.language = "en_AU.UTF-8"
		window.x11.display = ":0"
		application.process.machine_id = "2eea978869e74d178a4ccb3d1d772c6a"
		application.process.session_id = "2"
		module-stream-restore.id = "source-output-by-application-name:WEBRTC VoiceEngine"
    index: 52
	driver: <module-rtp-send.c>
	flags: 
	state: RUNNING
	source: 3 <rtp.monitor>
	volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
	        balance 0.00
	muted: no
	current latency: 4.31 ms
	requested latency: 7.26 ms
	sample spec: s16be 2ch 44100Hz
	channel map: front-left,front-right
	             Stereo
	resample method: (null)
	owner module: 32
	properties:
		media.name = "RTP Monitor Stream"
		rtp.source = "0.0.0.0"
		rtp.destination = "224.0.0.56"
		rtp.mtu = "1280"
		rtp.port = "46220"
		rtp.ttl = "1"
		module-stream-restore.id = "source-output-by-media-name:RTP Monitor Stream" 

Now for when I'm "testing" the microphone within Discord (this occurs with each mic I use, although all have been USB);

 ~ $ sudo fuser -v /dev/snd/*
                     USER        PID ACCESS COMMAND
/dev/snd/controlC0:  scott       639 F.... pulseaudio
/dev/snd/controlC1:  scott       639 F.... pulseaudio
/dev/snd/controlC2:  scott       639 F.... pulseaudio
/dev/snd/pcmC1D0p:   scott       639 F...m pulseaudio
/dev/snd/pcmC2D0c:   scott       639 F...m pulseaudio 
 ~ $ pacmd list-sources
5 source(s) available.
    index: 0
	name: <alsa_output.usb-Razer_Razer_Megalodon-00.analog-surround-21.monitor>
	driver: <module-alsa-card.c>
	flags: DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
	state: RUNNING
	suspend cause: (none)
	priority: 1040
	volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB,   lfe: 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: 7 KiB
	sample spec: s16le 3ch 48000Hz
	channel map: front-left,front-right,lfe
	used by: 1
	linked by: 1
	configured latency: 26.00 ms; range is 26.00 .. 1820.44 ms
	monitor_of: 0
	card: 0 <alsa_card.usb-Razer_Razer_Megalodon-00>
	module: 6
	properties:
		device.description = "Monitor of Razer Megalodon Analog Surround 2.1"
		device.class = "monitor"
		alsa.card = "1"
		alsa.card_name = "Razer Megalodon"
		alsa.long_card_name = "Razer Razer Megalodon at usb-0000:00:14.0-3, full speed"
		alsa.driver_name = "snd_usb_audio"
		device.bus_path = "pci-0000:00:14.0-usb-0:3:1.0"
		sysfs.path = "/devices/pci0000:00/0000:00:14.0/usb1/1-3/1-3:1.0/sound/card1"
		udev.id = "usb-Razer_Razer_Megalodon-00"
		device.bus = "usb"
		device.vendor.id = "1532"
		device.vendor.name = "Razer USA, Ltd"
		device.product.id = "000e"
		device.product.name = "Razer Megalodon"
		device.serial = "Razer_Razer_Megalodon"
		device.string = "1"
		module-udev-detect.discovered = "1"
		device.icon_name = "audio-card-usb"
    index: 2
	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: 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: 6554 /  10% / -60.00 dB
	volume steps: 65537
	muted: no
	current latency: 0.00 ms
	max rewind: 0 KiB
	sample spec: s32le 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: 2 <alsa_card.pci-0000_00_1f.3>
	module: 8
	properties:
		alsa.resolution_bits = "32"
		device.api = "alsa"
		device.class = "sound"
		alsa.class = "generic"
		alsa.subclass = "generic-mix"
		alsa.name = "ALC1150 Analog"
		alsa.id = "ALC1150 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 0xdf120000 irq 133"
		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 = "a170"
		device.product.name = "100 Series/C230 Series Chipset Family HD Audio Controller"
		device.form_factor = "internal"
		device.string = "front:0"
		device.buffering.buffer_size = "705600"
		device.buffering.fragment_size = "352800"
		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 ALC1150"
		alsa.components = "HDA:10ec0900,18491151,00100001 HDA:80862809,80860101,00100000"
		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: yes)
			properties:
				device.icon_name = "audio-input-microphone"
		analog-input-rear-mic: Rear Microphone (priority 8200, latency offset 0 usec, available: no)
			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-front-mic>
    index: 3
	name: <rtp.monitor>
	driver: <module-null-sink.c>
	flags: DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
	state: RUNNING
	suspend cause: (none)
	priority: 1000
	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: 1 KiB
	sample spec: s16be 2ch 44100Hz
	channel map: front-left,front-right
	             Stereo
	used by: 1
	linked by: 1
	configured latency: 7.26 ms; range is 0.50 .. 2000.00 ms
	monitor_of: 1
	module: 27
	properties:
		device.description = "Monitor of RTP Multicast"
		device.class = "monitor"
		device.icon_name = "audio-input-microphone"
    index: 4
	name: <combined.monitor>
	driver: <module-combine-sink.c>
	flags: DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
	state: SUSPENDED
	suspend cause: IDLE
	priority: 1000
	volume: front-left: 65533 / 100% / -0.00 dB,   front-right: 65533 / 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: float32le 2ch 44100Hz
	channel map: front-left,front-right
	             Stereo
	used by: 0
	linked by: 0
	configured latency: 0.00 ms; range is 0.50 .. 200.00 ms
	monitor_of: 2
	module: 30
	properties:
		device.description = "Monitor Source of Simultaneous output to Razer Megalodon Analog Surround 2.1"
		device.class = "monitor"
		device.icon_name = "audio-input-microphone"
  * index: 6
	name: <alsa_input.usb-BLUE_MICROPHONE_Blue_Snowball_201305-00.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: 9049
	volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
	        balance 0.00
	base volume: 38295 /  58% / -14.00 dB
	volume steps: 65537
	muted: no
	current latency: 0.20 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: 10.00 ms; range is 0.50 .. 2000.00 ms
	card: 1 <alsa_card.usb-BLUE_MICROPHONE_Blue_Snowball_201305-00>
	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 = "Blue Snowball"
		alsa.long_card_name = "BLUE MICROPHONE Blue Snowball at usb-0000:00:14.0-8, full speed"
		alsa.driver_name = "snd_usb_audio"
		device.bus_path = "pci-0000:00:14.0-usb-0:8:1.0"
		sysfs.path = "/devices/pci0000:00/0000:00:14.0/usb1/1-8/1-8:1.0/sound/card2"
		udev.id = "usb-BLUE_MICROPHONE_Blue_Snowball_201305-00"
		device.bus = "usb"
		device.vendor.id = "0d8c"
		device.vendor.name = "C-Media Electronics, Inc."
		device.product.id = "0005"
		device.product.name = "Blue Snowball"
		device.serial = "BLUE_MICROPHONE_Blue_Snowball_201305"
		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 = "Blue Snowball Analog Stereo"
		alsa.mixer_name = "USB Mixer"
		alsa.components = "USB0d8c:0005"
		module-udev-detect.discovered = "1"
		device.icon_name = "audio-card-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> 
 ~ $ pacmd list-source-outputs
3 source output(s) available.
    index: 5
	driver: <protocol-native.c>
	flags: 
	state: RUNNING
	source: 0 <alsa_output.usb-Razer_Razer_Megalodon-00.analog-surround-21.monitor>
	volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
	        balance 0.00
	muted: no
	current latency: 24.60 ms
	requested latency: 10.00 ms
	sample spec: s16le 2ch 44100Hz
	channel map: front-left,front-right
	             Stereo
	resample method: speex-float-1
	owner module: 11
	client: 21 <qemu-system-x86_64>
	properties:
		media.name = "qemu"
		application.name = "qemu-system-x86_64"
		native-protocol.peer = "UNIX socket client"
		native-protocol.version = "32"
		application.process.id = "6294"
		application.process.user = "root"
		application.process.host = "scott-linux"
		application.process.binary = "qemu-system-x86_64"
		application.language = "C"
		window.x11.display = ":0"
		application.process.machine_id = "2eea978869e74d178a4ccb3d1d772c6a"
		module-stream-restore.id = "source-output-by-application-name:qemu-system-x86_64"
    index: 49
	driver: <protocol-native.c>
	flags: 
	state: RUNNING
	source: 6 <alsa_input.usb-BLUE_MICROPHONE_Blue_Snowball_201305-00.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: 10.00 ms
	sample spec: s16le 2ch 44100Hz
	channel map: front-left,front-right
	             Stereo
	resample method: (null)
	owner module: 11
	client: 20 <WEBRTC VoiceEngine>
	properties:
		media.name = "recStream"
		application.name = "WEBRTC VoiceEngine"
		native-protocol.peer = "UNIX socket client"
		native-protocol.version = "32"
		application.process.id = "3926"
		application.process.user = "scott"
		application.process.host = "scott-linux"
		application.process.binary = "DiscordPTB"
		application.language = "en_AU.UTF-8"
		window.x11.display = ":0"
		application.process.machine_id = "2eea978869e74d178a4ccb3d1d772c6a"
		application.process.session_id = "2"
		module-stream-restore.id = "source-output-by-application-name:WEBRTC VoiceEngine"
    index: 52
	driver: <module-rtp-send.c>
	flags: 
	state: RUNNING
	source: 3 <rtp.monitor>
	volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
	        balance 0.00
	muted: no
	current latency: 6.62 ms
	requested latency: 7.26 ms
	sample spec: s16be 2ch 44100Hz
	channel map: front-left,front-right
	             Stereo
	resample method: (null)
	owner module: 32
	properties:
		media.name = "RTP Monitor Stream"
		rtp.source = "0.0.0.0"
		rtp.destination = "224.0.0.56"
		rtp.mtu = "1280"
		rtp.port = "46220"
		rtp.ttl = "1"
		module-stream-restore.id = "source-output-by-media-name:RTP Monitor Stream" 

Thanks for the reply smile

Last edited by nan0s7 (2019-01-25 11:40:17)

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#4 2019-01-25 14:11:32

V1del
Forum Moderator
Registered: 2012-10-16
Posts: 6,017

Re: Recent package update breaks microphone usage

That looks alright, did you configure that webrtc sink yourself, or is that something enforced by discord? FWIW do you also get no input if you just try

parec -d alsa_input.usb-BLUE_MICROPHONE_Blue_Snowball_201305-00.analog-stereo test.wav

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#5 2019-01-25 22:06:28

McDuck
Member
Registered: 2007-03-07
Posts: 41

Re: Recent package update breaks microphone usage

No intention to hijack the thread but the author is not the only one with this issue, I have the exact same issue.
It's a recent issue, a couple of days at most.
I too have the Blue Snowball USB mic, but I did try with an old logitech USB mic and also a "standard mic" plugged directly into the sound card, same result.
If it helps: Error message when trying to use the mic in Discord

[000:000] [12590] (audio_send_stream.cc:519): Unable to create encoder for {name: opus, clockrate_hz: 48000, num_channels: 1, parameters: {stereo: 0}}
[000:000] [12590] (audio_send_stream.cc:261): Failed to set up send codec state.

Last edited by McDuck (2019-01-25 22:07:21)

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#6 2019-01-25 22:52:13

ZSmith
Member
Registered: 2017-02-25
Posts: 13

Re: Recent package update breaks microphone usage

I can confirm that this issue is affecting me as well, but only on my desktop. On my laptop (also an Arch Linux system) everything works correctly. I can confirm a similar time frame as the OP (last several days). I'm seeing similar log output as McDuck, but only on my desktop system. No other applications seem to have any trouble recording audio input. I am using a standard analog headset connected to my computer's front panel jacks. I would like to add that the web client works and is a reasonable workaround until this is resolved.

Last edited by ZSmith (2019-01-25 22:59:41)

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#7 2019-01-26 03:51:57

nan0s7
Member
Registered: 2018-10-28
Posts: 6

Re: Recent package update breaks microphone usage

V1del wrote:

That looks alright, did you configure that webrtc sink yourself, or is that something enforced by discord? FWIW do you also get no input if you just try

parec -d alsa_input.usb-BLUE_MICROPHONE_Blue_Snowball_201305-00.analog-stereo test.wav

Yeah I don't notice any input with that command. I didn't set up webrtc myself, I'm assuming it's just done by Discord.

Can confirm I see that message the other's have seen when monitoring Discord in the terminal.

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#8 2019-01-26 08:45:22

nan0s7
Member
Registered: 2018-10-28
Posts: 6

Re: Recent package update breaks microphone usage

I noticed that Discord (the PTB release at least) just had an update. Decided to install it and it seems to have fixed the microphone issues we've been experiencing.

For the record, I checked it in a terminal to see the live logs too, and the error mentioned above still occurs when the mic is being used. But, it has no effect at all anyway.

 [000:000] [23641] (audio_device_pulse_linux.cc:1857): Can't query latency
[000:001] [23638] (audio_send_stream.cc:519): Unable to create encoder for {name: opus, clockrate_hz: 48000, num_channels: 1, parameters: {stereo: 0}}
[000:001] [23638] (audio_send_stream.cc:261): Failed to set up send codec state. 

So I guess thanks for all the help! Too bad we still don't have much of an idea for what actually went wrong, but considering they managed to fix it within 48 hours I hope they know for next time. smile

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#9 2019-01-26 10:15:19

V1del
Forum Moderator
Registered: 2012-10-16
Posts: 6,017

Re: Recent package update breaks microphone usage

nan0s7 wrote:

Yeah I don't notice any input with that command. I didn't set up webrtc myself, I'm assuming it's just done by Discord.

You will have a test.wav in your working directory, which should contain audio, however from the other messages it does look like a discord issue.

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#10 2019-01-27 17:12:55

ZSmith
Member
Registered: 2017-02-25
Posts: 13

Re: Recent package update breaks microphone usage

I can confirm working mic in discord-ptb

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