You can use amixer. It's in the alsa-utils package on Ubuntu and Debian.
Run amixer without parameters to get an overview about your controls for the default device.
You can also use alsamixer without parameters (from the same package) to get a more visual overview. Use F6 to see and switch between devices. Commonly you might have PulseAudio and a hardware ...
As @Teresa-e-Junior pointed out pactl is the tool to use:
First of all we might want to get the IDs of our PA sinks. On my system this is what I get:
$ pactl list short sinks
0 alsa_output.pci-0000_01_00.1.hdmi-surround module-alsa-card.c s16le 6ch 44100Hz SUSPENDED
1 alsa_output.pci-0000_00_1b.0.analog-stereo module-alsa-card.c ...
I think the reason this isn't working for you is because that interface has been deprecated. You normally can't write audio using /dev/dsp anymore, at least without being tricky.
There is a program that will accomplish this for you on your system: padsp.
This will map the /dev/audio or /dev/dsp file to the new Audio Server system.
Fire up the terminal and ...
When setting up a new Arch install I usually do the following:
pacman -S alsa-utils
Then fire up alsamixer with
Go to 'Master' then press M and then keep hitting the up key to bring the sound levels up. I do that for any other inputs I want like pcm etc.
with the number being how many channels you want to test. You ...
I think you can do this with PulseAudio. I found this tutorial that shows how, titled: Redirect Audio Out to Mic In (Linux).
Run the application pavucontrol.
Go to the "Input Devices" tab, and select "Show: Monitors" from the bottom of the window. If your computer is currently playing audio, you should see a bar showing the volume of the ...
It's all about multiplexing.
I don't need to multiplex sounds from or to several sound cards
Ah, but you do! If you want to be able to play audio from two sources at once, ever, for any reason, you need multiplexing.
OSX and Windows handle Multiplexing in the Kernel (but still in software), which is why this never/rarely comes up on those platforms.
You can get the number of sink Input with pactl command.
$ pactl list sink-inputs
Sink Input #7119
Owner Module: 12
application.icon_name = "google-chrome"
media.name = "Playback"
application.name = "Chromium"
Using this number(#7119), you specify the sink ...
Turning all hardware mixers up worked.
Edit: Actually just found another, slightly hacky, work around to this issue after it came back a few boots later by using an ladpsa amplifier and compressor in Arch. Something similar to this could probably work in other distros too.
Install ladspa plugins:
pacman -S ladspa-plugins
Create an .asoundrc file in the ...
I recommend asoundconf, which will take care of generating a proper .asoundrc .
List the available sound cards:
Set the default one:
asoundconf set-default-card PARAMETER
Unmute it if necessary through either alsamixer(ncurses) or amixer (cli)
One way could perhaps be to use amixer.
Jack plugged in:
$ amixer -c 0 contents
numid=29,iface=CARD,name='Front Headphone Jack'
Jack not plugged in:
$ amixer -c 0 contents
numid=29,iface=CARD,name='Front Headphone Jack'
So for that ...
Set Headphone Mic Boost gain to 10dB. Any other value seems to cause the irritating background noise in headphones. This can be done with amixer:
amixer -c0 sset 'Headphone Mic Boost' 10dB
To make this happen automatically every time you headphones are connected install acpid.
Start it by running:
sudo systemctl start acpid.service
Enable it by running:
I finally found an answer to this problem based on this post from another forum (see "Attempt 1b" for reference). According to it, one can specify the module id, even if the driver used is the same:
alias char-major-116 snd
alias snd-card-0 snd-hda-intel
alias snd-card-1 snd-hda-intel
options snd cards_limit=2
options snd-hda-intel id=SB index=0
The following commands are use to manipulate the PulseAudio sound server:
pacmd - Used to reconfigure a PulseAudio sound server during runtime.
pactl - Used to control a running PulseAudio sound server.
Here are some examples of how they function.
pacmd list-sinks :: list name or index number of possible sinks
pacmd set-default-sink [sinkname] :: set the ...
When looking around, the issue arised when I installed GDM to handle logins, which of course has Gnome attached to it, which in turn requires Pulseaudio. That set my default device to Pulseaudio which messed stuff up.
What I did to solve my problem was to edit /etc/pulse/daemon.conf. Uncomment the following line default-sample-rate = 42100 and ...
Apparently there is no option for firefox, but you can manipulate the ALSA output through environment variables.
Try for example:
Alternatively, if this does not work, try scripting a little your .asoundrc
vars [ ALSAPCM ]
cat /dev/urandom | aplay
is the command that needs to be typed. If you aren't in "audio" group, you could prefix aplay with sudo. This also doesn't interfere with any daemons (I was running pulseaudio while this command was active and correctly heard the "noise".
EDIT (Aug 6, 2019): In an older version of the command I also had a padsp tee thing between the ...
The solution turned out to be simpler than it appeared. The output of fuser -v /dev/snd/* revealed jackd was silently hogging the audio card even after QjackCtl supposedly killed it. Running killall jackd fixed the problem. The problem wasn't with PulseAudio, but rather jackd running invisibly in the background.
ALSA stands for Advanced Linux Sound Architecture, I'd encourage you to poke around their project website if you're truly curious. Specifically I'd take a look at the "I'm new to ALSA pages & tutorials.
The ArchLinux wiki probably describes it the best.
The Advanced Linux Sound Architecture (ALSA) is a Linux kernel component which replaced the ...
I just wandered upon this post and see you are struggling with the answer to this as I was. This is what fixed it for me:
Go into alsamixer and set everything the way you want it, then exit and type this:
sudo alsactl store
That will store the current config of alsamixer and it should keep the config.
I solved this issue as follows:
First, load the snd-aloop module:
sudo modprobe snd-aloop
This will create a new device called Loopback:
± % cat /proc/asound/cards !10017
0 [PCH ]: HDA-Intel - HDA Intel PCH
I found solution here:
there I've found :
$ amixer -c0 cset iface=MIXER,name='Input Source',index=1 'Front Mic' # (Record from Front Mic)
slightly modified according to my sound-card and setup (default sound-card, different items ordering) :
$ amixer cset name='Input Source',index=0 'Rear Mic'
After asking this questions multiple times on gentoo IRC and finally the forums, I was pushed in the right direction and able to solve the problem.
~ $ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: PCH [HDA Intel PCH], device 0: ALC892 Analog [ALC892 Analog]
Subdevice #0: subdevice #0
card 0: PCH [HDA Intel PCH], device 1: ...
I've done this in the past and have had pretty good success with boosting the volume up. The only problem with it is if you turn it down, then you'll have to come back into the system preferences under the speaker icon to turn it back up to 150%.
However with the steps below should remain "sticky" from reboot to reboot.
Alsa by default uses the same sampling rate and format as the source. It is however possible to force the sample rate up (or down).
Here is one way you could do it. (in /etc/asound.conf or ~/.asoundrc)
Then just make that pcm a slave to another one, like ...
First identify your microphone device file; should be something similar to /dev/snd/pcmC0D0c. To help you find the device file, you can start a test recording with arecord or such, then do lsof | grep '/dev/snd'; it will list all programs and their associated device file.
Then you can peek usage of the microphone using fuser /dev/snd/pcmC0D0c. It will ...
Finally fixed it by going through absolutely every bar in alsamixer, even the ones that I wasn't sure would matter, and unmuting them all. As it turns out, the mute toggle mutes everything, but only unmutes the master volume.