Yrurtj

Context

I have been running archlinux on my laptop for years and never needed the microphone. Now I need it but cannot find a way to configure it properly.

I use alsa and pulseadio.

Running alsamixer as a user I have two gauges:

  ||      ||

Master  Capture

I have both of them unmuted and enabled to sensible (~50%) values.

Running alsamixer as root I have several gauges:

  ||        ||        ||      ||      ||       ||

Master  Headphones  Speaker  PCM  Microphone MIC Boot 



 ||      ||       ||        ||       ||

Beep  Loopback Internal  Internal  Capture

If I enable the Loopback as root I can hear the microphone back through the speakers.

If I explicitly select my sound card in alsamixer as a user I get the same commands as running as root (and if I enable Loopback here I do hear the microphone from the speakers as well). I get the following:

(To the right are two more Internal gauges, the Loopback (enable/disable) and a mutable Off-hook)

Yet, no matter what I do and how much I meddle with the alsa gauges I never manage to pass the microphone to the user application. In essence, every time I run arecord I get the following output:

$ arecord -vv -f dat /dev/null 

Recording WAVE '/dev/null' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo

ALSA <-> PulseAudio PCM I/O Plugin

Its setup is:

  stream       : CAPTURE

  access       : RW_INTERLEAVED

  format       : S16_LE

  subformat    : STD

  channels     : 2

  rate         : 48000

  exact rate   : 48000 (48000/1)

  msbits       : 16

  buffer_size  : 24000

  period_size  : 6000

  period_time  : 125000

  tstamp_mode  : NONE

  tstamp_type  : GETTIMEOFDAY

  period_step  : 1

  avail_min    : 6000

  period_event : 0

  start_threshold  : 1

  stop_threshold   : 24000

  silence_threshold: 0

  silence_size : 0

  boundary     : 6755399441055744000

#+                                                 | 00%

aplay, on the other hand, works perfectly.

My sound card is (lspci):

# lspci| grep -i audio

00:14.2 Audio device: Advanced Micro Devices, Inc. [AMD/ATI] SBx00 Azalia (Intel HDA)

I looked at the arch wiki and tried setting the following in /etc/modprobe.d/modprobe.conf (and rebooted the machine):

options snd-hda-intel model=dell-m6-dmic

(I cannot get the microphone to send data to the user application with or without that configuration.)

My current hda modules are:

# lsmod |grep hda

snd_hda_codec_si3054    16384  1

snd_hda_codec_realtek    69632  1

snd_hda_codec_generic    69632  1 snd_hda_codec_realtek

snd_hda_intel          32768  5

snd_hda_codec         106496  4 snd_hda_codec_realtek,snd_hda_codec_si3054,snd_hda_codec_generic,snd_hda_intel

snd_hda_core           61440  5 snd_hda_codec_realtek,snd_hda_codec_si3054,snd_hda_codec_generic,snd_hda_codec,snd_hda_intel

snd_hwdep              16384  1 snd_hda_codec

snd_pcm                86016  4 snd_hda_codec_si3054,snd_hda_codec,snd_hda_intel,snd_hda_core

snd                    65536  18 snd_hda_codec_realtek,snd_hda_codec_si3054,snd_hwdep,snd_timer,snd_pcm,snd_hda_codec_generic,snd_hda_codec,snd_hda_intel

For completeness, this is my pulseaudio config (I have not tried meddling with it):

$ pulseaudio --dump-conf

### Read from configuration file: /etc/pulse/daemon.conf ###

daemonize = no

fail = yes

high-priority = yes

nice-level = -11

realtime-scheduling = yes

realtime-priority = 5

allow-module-loading = yes

allow-exit = yes

use-pid-file = yes

system-instance = no

local-server-type = user

cpu-limit = no

enable-shm = yes

flat-volumes = no

lock-memory = no

exit-idle-time = 20

scache-idle-time = 20

dl-search-path = /usr/lib/pulse-9.0/modules

default-script-file = /etc/pulse/default.pa

load-default-script-file = yes

log-target = 

log-level = notice

resample-method = auto

enable-remixing = yes

enable-lfe-remixing = no

lfe-crossover-freq = 0

default-sample-format = s16le

default-sample-rate = 44100

alternate-sample-rate = 48000

default-sample-channels = 2

default-channel-map = front-left,front-right

default-fragments = 4

default-fragment-size-msec = 25

enable-deferred-volume = yes

deferred-volume-safety-margin-usec = 8000

deferred-volume-extra-delay-usec = 0

shm-size-bytes = 0

log-meta = no

log-time = no

log-backtrace = 0

rlimit-fsize = -1

rlimit-data = -1

rlimit-stack = -1

rlimit-core = -1

rlimit-rss = -1

rlimit-as = -1

rlimit-nproc = -1

rlimit-nofile = 256

rlimit-memlock = -1

rlimit-locks = -1

rlimit-sigpending = -1

rlimit-msgqueue = -1

rlimit-nice = 31

rlimit-rtprio = 9

rlimit-rttime = 200000

As per Faheem Mitha's comment, I'm confident the pulseaudio tmpfs has the right permissions:

# ls -la /run/user/861213/pulse/

total 4

drwx------ 2 grochmal users  80 Jul  4 21:44 .

drwx------ 5 grochmal users 120 Jul  4 21:46 ..

srw-rw-rw- 1 grochmal users   0 Jul  4 21:44 native

-rw------- 1 grochmal users   4 Jul  4 21:44 pid

And pavucontrol was muted. I was not aware of pavucontrol which resulted in the issue in the first place:

Question

I'm out of ideas. I believe that the Loopback audio interface exists in kernel space, therefore my problem is that I cannot bring the microphone data into user space. Yet, I'm not 100% sure of it.

What other options can I give to snd-hda-intel (or the other modules) that might be relevant for a microphone that works on the loopback interface but does not propagate to the application layer?

Could I be doing something wrong with pulseaudio? But then again, pulseaudio worked correctly (together with alsa) for all that time, and I use audacity quite often.

Based on correspondence with the poster (see comments), it turned out that the sound card was muted inside pavucontrol.

I also met the problem of my microphone not working on Arch Linux. In my case there's no device shown under "Input Devices" in pavucontrol (It says "no input devices available"). Following the following advice from Arch Wiki fixed it for me:

1. Run arecord -l to find out the id of the microphone in hw:CARD,DEVICE notation, e.g. if the output is


**** List of CAPTURE Hardware Devices ****
card 0: PCH [HDA Intel PCH], device 0: ALC269VC Analog[ALC269VC Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0


Then the id is hw:0,0

2. Add the line load-module module-alsa-source device=hw:0,0 in /etc/pulse/default.pa




3. pulseaudio -k; pulseaudio -D to restart pulseaudio and activate the changes.

I'll leave it here in case somebody runs into the same problem as mine.

1.rm -rf ~/.config/pulse

2.rm -rf /etc/pulse/

3.Plug in the usb headphone into laptop

4.pacman -S pulseaudio pavucontrol

5. Add "load-module module-loopback latency_msec=1" in /etc/pulse/default.pa

6.reboot

6.pavucontrol

OK.