Advanced Linux Sound Architecture/Troubleshooting
Volume
No output
If speaker-test produces sound but some other program does not, determine whether PulseAudio is being used:
# fuser -v /dev/snd/*
If it is, try using apulseAUR, as described in Advanced Linux Sound Architecture#PulseAudio compatibility. Alternatively, killing the PulseAudio process may cause sound to start working in the desired process.
Output is muted after reboot
Run the following command:
# alsactl restore
If the problem persists, verify that the Auto-Mute option in alsamixer is set to Disabled.
Volume is too low
Run alsamixer and try to increase the value of the sliders, unmuting channels if necessary. Note that if you have many sliders, you may have to scroll to the right to see any missing sliders.
If all the sliders are maxed out, and the volume is still too low, you can try running the following script to reset your codec settings:
$ wget -O hda-analyzer.py https://git.alsa-project.org/?p=alsa.git;a=blob_plain;f=hda-analyzer/run.py
Close the analyzer, and when prompted as to whether you want to reset the codecs, say "yes".
If the volume is still too low, run alsamixer again: resetting the codecs may have caused new sliders to become enabled and some of them may be set to a low value.
Volume is still too low
If you are facing low volume even after maxing out your speakers/headphones, you can give the softvol plugin a try. Add the following to /etc/asound.conf.
/etc/asound.conf
pcm.!default {
type plug
slave.pcm "softvol"
}
pcm.softvol {
type softvol
slave {
pcm "dmix"
}
control {
name "Pre-Amp"
card 0
}
min_dB -5.0
max_dB 20.0
resolution 6
}
plug with hw in the above configuration.After the changes are loaded successfully, you will see a Pre-Amp section in alsamixer. You can adjust the levels there.
- Setting a high value for
Pre-Ampcan cause sound distortion, so adjust it according to the level that suits you. - Some audio codecs may need to have settings adjusted in the HDA Analyzer (see #Volume is too low) in order to achieve proper volume without distortion. Checking the HP option under widget control in the Playback Switch (Node[0x14] PIN in the ALC892 codec, for instance) can sometimes improve audio quality and volume significantly.
Random lack of sound on startup
You can quickly test sound by running speaker-test. If there is no sound, you may see something similar to:
ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave Playback open error: -16 Device or resource busy
If you have no sound on startup, this may be because your system has multiple sound cards, and their order may sometimes change on startup. If this is the case, try setting the default sound card.
If you use MPD and the above configuration tips do not work, try following https://mpd.wikia.com/wiki/Configuration#ALSA_MPD_software_volume_control.
Microphone
No microphone input
In alsamixer, make sure that all the volume levels are up under recording, and that CAPTURE is toggled active on the microphone (e.g. Mic, Internal Mic) and/or on Capture (in alsamixer, select these items and press space). Try making positive Mic Boost and raising Capture and Digital levels higher; this may make static or distortion, but then you can adjust them back down once you are hearing something when you record
As the pulseaudio wrapper is shown as "default" in alsamixer, you may have to press F6 to select your actual soundcard first. You may also need to enable and increase the volume of Line-in in the Playback section.
To test the microphone, run these commands (see arecord(1) for further information):
$ arecord --duration=5 --format=dat test-mic.wav $ aplay test-mic.wav
Alternatively, you can run this command:
$ arecord -vv --format=dat /dev/null
alongside alsamixer to easily identify channel which you should select and unmute.
To test a particular device, use the --device parameter followed by the hardware PCM name in the form hw:C,D for card C device D, or plughw:C,D for plugged hardware. For instance:
$ arecord -vvv --format=dat --device=plughw:0,0 /dev/null
If all fails, you may want to eliminate hardware failure by testing the microphone with a different device.
For at least some computers, muting a microphone (MM) simply means its input does not go immediately to the speakers. It still receives input.
Many Dell laptops need "-dmic" to be appended to the model kernel module parameter:
options snd-hda-intel model=dell-m6-dmic
Some programs use try to use OSS as the main input software. If you have enabled the snd_pcm_oss, snd_mixer_oss or snd_seq_oss kernel modules previously (they are not loaded by default), try unloading them.
See also:
- https://www.alsa-project.org/main/index.php/SoundcardTesting
- https://alsa.opensrc.org/Record_from_mic
Setting the default microphone/capture device
Some applications (Pidgin, Adobe Flash) do not provide an option to change the capture device. It becomes a problem if your microphone is on a separate device (e.g. USB webcam or microphone) than your internal sound card. To change only the default capture device, leaving the default playback device as is, you can modify your ~/.asoundrc file to include the following:
~/.asoundrc
pcm.usb
{
type hw
card U0x46d0x81d
}
pcm.!default
{
type asym
playback.pcm
{
type plug
slave.pcm "dmix"
}
capture.pcm
{
type plug
slave.pcm "usb"
}
}
Replace U0x46d0x81d with your capture device's card name in ALSA. You can use arecord -L to list all the capture devices detected by ALSA.
Internal microphone not working
First make sure the volume is enabled under the Capture view in alsamixer. In some cases, the "Internal Microphone" is not displayed in the capture list available when pressing F4. If so, specifying the card number given by aplay -l to start alsamixer (for example alsamixer -c 0 ) can make it appear.
If still unsucessful, add the following kernel module parameter:
/etc/modprobe.d/snd-hda-intel.conf
options snd-hda-intel enable_msi=1
Then reload the module:
# rmmod snd-hda-intel && modprobe snd-hda-intel
Now there should be an additional input under the previously mentioned Capture view.
Crackling microphone
If you are getting a crackling or popping sound from your microphone that cannot be resolved with ALSA settings or cleaning your microphone jack, try adding the following kernel module parameter:
options snd-hda-intel model=MODEL position_fix=3
This option will fix crackling on pure ALSA, but will cause issues with PulseAudio. To let PulseAudio use these settings effectively, edit /etc/pulse/default.pa and add the tsched=0 parameter to module-udev-detect:
/etc/pulse/default.pa
load-module module-udev-detect tsched=0
See https://docs.kernel.org/sound/hd-audio/notes.html#dma-position-problem
Audio Quality
Crackling sound through mini-jack (headphones connector)
Following Advanced Linux Sound Architecture#Simultaneous output might lead to crackling sound through headphones or external speakers. This can be fixed by muting or setting the volume to 0% on Mic. Use alsamixer or amixer:
$ amixer sset "Mic" 0% $ amixer sset "Mic" mute
Popping sound after resuming from suspension
You might hear a popping sound after resuming the computer from suspension. This can be fixed by editing /etc/pm/sleep.d/90alsa and removing the line that says aplay -d 1 /dev/zero
Sound skipping during playback
Run alsamixer, and if channels exist for nonexistent output devices then disable them (e.g. alsamixer showing a center speaker but you not having one).
Poor sound quality or clipping
If you experience poor sound quality, try setting the PCM volume (in alsamixer) to a level such that gain is 0.
If snd-usb-audio driver has been loaded, you could try to enable softvol:
/etc/asound.conf
pcm.!default {
type plug
slave.pcm "softvol"
}
pcm.dmixer {
type dmix
ipc_key 1024
slave {
pcm "hw:0"
period_size 4096
buffer_size 131072
rate 50000
}
bindings {
0 0
1 1
}
}
pcm.dsnooper {
type dsnoop
ipc_key 1024
slave {
pcm "hw:0"
channels 2
period_size 4096
buffer_size 131072
rate 50000
}
bindings {
0 0
1 1
}
}
pcm.softvol {
type softvol
slave { pcm "dmixer" }
control {
name "Master"
card 0
}
}
ctl.!default {
type hw
card 0
}
ctl.softvol {
type hw
card 0
}
ctl.dmixer {
type hw
card 0
}
Pops when starting and stopping playback
Some modules (e.g. snd_ac97_codec and snd_hda_intel) can power off your sound card when it is not used. This can make an audible noise (like a crack/pop/scratch) when turning on/off your sound card. Sometimes even when moving the volume slider or opening and closing windows on some desktop environments. If you find this annoying, try modinfo your_module and look for a module option that adjusts or disables this feature.
For example, to disable the power saving mode for the snd_hda_intel module, add the following kernel module parameter:
options snd_hda_intel power_save=0
You may also need to disable power saving for the audio card controller:
options snd_hda_intel power_save=0 power_save_controller=N
You may also have to unmute the 'Line' ALSA channel for this to work. Any value will do (other than '0' or something too high). For example, on an onboard VIA VT1708S (using the snd_hda_intel module) these cracks occurred even when power_save was set to 0. Unmuting the 'Line' channel and setting a value of '1' solved the problem.
See https://docs.kernel.org/sound/designs/powersave.html
Sound skipping while using dynamic CPU frequency scaling
Some combinations of ALSA drivers and chipsets may cause audio from all sources to skip when used in combination with a dynamic frequency scaling governor such as ondemand or conservative. Currently, the solution is to switch back to the performance governor.
Refer to CPU frequency scaling for more information.
Hardware and Cards
Verifying output parameters
Check the contents of /proc/asound/cardX/pcmYp/subZ/hw_params, where X, Y, and Z are system dependent.
In order to find this file, execute the following command while outputting anything via ALSA:
$ find /proc/asound/ -name hw_params | xargs -I FILE grep -v -l "closed" FILE | grep '/proc/asound/card./pcm.p/sub./hw_params'
If nothing is playing there should be no results.
Here is an example output for audio with a bit depth of 24 bits and a sampling frequency of 44.1 kilohertz:
$ cat /proc/asound/card1/pcm0p/sub0/hw_params
access: RW_INTERLEAVED format: S24_3LE subformat: STD channels: 2 rate: 44100 (44100/1) period_size: 5513 buffer_size: 22050
More info is available in the ALSA documentation.
Error 'Unknown hardware' appears after kernel update
The following messages may be displayed during ALSA's initialization:
Unknown hardware "foo" "bar" ... Hardware is initialized using a guess method /usr/bin/alsactl: set_control:nnnn:failed to obtain info for control #mm (No such file or directory)
or:
Found hardware: "HDA-Intel" "VIA VT1705" "HDA:11064397,18490397,00100000" "0x1849" "0x0397" Hardware is initialized using a generic method /usr/bin/alsactl: set_control:1328: failed to obtain info for control #1 (No such file or directory) /usr/bin/alsactl: set_control:1328: failed to obtain info for control #2 (No such file or directory) /usr/bin/alsactl: set_control:1328: failed to obtain info for control #25 (No such file or directory) /usr/bin/alsactl: set_control:1328: failed to obtain info for control #26 (No such file or directory)
Simply store ALSA mixer settings again:
# alsactl -f /var/lib/alsa/asound.state store
It may be necessary configure ALSA again with alsamixer
Fix wrong audio pin mapping
If the mappings to your audio pins(plugs) do not correspond but ALSA works fine, you could try HDA Analyzer -- a pyGTK2 GUI for HD-audio control can be found at the ALSA wiki. Try tweaking the Widget Control section of the PIN nodes, to make microphones IN and headphone jacks OUT. Referring to the Config Defaults heading is a good idea.
python in the run.py file with python2 to point the script to the Python 2 version. Then make the script executable and run it.S/PDIF output does not work
If the optical/coaxial digital output of your motherboard/sound card is not working or stopped working, and have already enabled and unmuted it in alsamixer, try running the following:
$ iecset audio on
You can also put this command in an enabled systemd service as it sometimes it may stop working after a reboot.
Conflicting PC speaker
If you are sure nothing is muted, that your drivers are installed correctly, and that your volume is right, but you still do not hear anything, then try the following kernel module parameters:
options snd-NAME-OF-MODULE ac97_quirk=0
The above fix has been observed to work with via82xx
options snd-NAME-OF-MODULE ac97_quirk=1
The above fix has been reported to work with snd-intel8x0
HP TX2500
Use these kernel module parameters:
options snd-cmipci mpu_port=0x330 fm_port=0x388 options snd-hda-intel index=0 model=toshiba position_fix=1 options snd-hda-intel model=hp (works for tx2000cto)
No sound when S/PDIF video card is installed
Discover available modules and their order:
$ cat /proc/asound/modules
0 snd_hda_intel 1 snd_ca0106
Disable the undesired video card audio codec in /etc/modprobe.d/modprobe.conf:
/etc/modprobe.d/modprobe.conf
install snd_hda_intel /bin/false
If both devices use the same module then you can use the enable kernel module parameter from snd_hda_intel module; it is an array of booleans that can enable/disable the desired sound card.
options snd_hda_intel enable=1,0
Wrong sound card model type
Although ALSA detects your soundcard through the BIOS, at times ALSA may not be able to recognize your model type. The soundcard chip can be found in alsamixer (e.g. ALC662) and the model can be set as kernel module parameters:
options snd-hda-intel model=MODEL
There are other model settings too. For most cases ALSA defaults will do. If you want to look at more specific settings for your soundcard take a look at the ALSA Soundcard List find your model, then Details, then look at the "Setting up modprobe..." section. Enter these values in /etc/modprobe.d/modprobe.conf. For example, for an Intel AC97 audio:
# ALSA portion alias char-major-116 snd alias snd-card-0 snd-intel8x0 # module options should go here # OSS/Free portion alias char-major-14 soundcore alias sound-slot-0 snd-card-0 # card #1 alias sound-service-0-0 snd-mixer-oss alias sound-service-0-1 snd-seq-oss alias sound-service-0-3 snd-pcm-oss alias sound-service-0-8 snd-seq-oss alias sound-service-0-12 snd-pcm-oss
Intel onboard sound
No sound with onboard Intel sound card
There may be a problem with two conflicting modules loaded, namely snd-intel8x0 and snd-intel8x0m. In this case, blacklist snd-intel8x0m.
Muting the "External Amplifier" in alsamixer or amixer may also help. See the ALSA wiki.
Unmuting the "Mix" setting in the mixer might help, also.
No headphone sound with onboard intel sound card
With some laptops, you may need to add the following kernel module parameter:
options snd-hda-intel model=model
Where model is any of the ones listed in the kernel documentation, but check that it is the correct version of that document for your kernel version.
To know your chip name type the following command (with * being corrected to match your files). Note that some chips could have been renamed and do not directly match the available ones in the file.
$ grep Codec /proc/asound/card*/codec*
Note that there is a high chance none of the input devices (all internal and external mics) will work if you choose to do this, so it is either your headphones or your mic. Please report to ALSA if you are affected by this bug.
And also, if you have problems getting beeps to work (pcspkr):
options snd-hda-intel model=model enable=1 index=0
HDMI
HDMI Output does not work
The procedure described below can be used to test HDMI audio. Before proceeding, make sure you have enabled and unmuted the output with alsamixer.
Connect your PC to the Display via HDMI cable and enable the display with xrandr.
Use aplay -l to discover the card and device number. For example:
$ aplay -l
**** 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: Generic [HD-Audio Generic], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0
Send sound to the device. Following the example in the previous step, you would send sound to card 1, device 3:
$ aplay -D plughw:1,3 /usr/share/sounds/alsa/Front_Center.wav
If aplay does not output any errors, but still no sound is heard, "reboot" the receiver, monitor or tv set. Since the HDMI interface executes a handshake on connection, it might have noticed before that there was no audio stream embedded, and disabled audio decoding. If you are using a standalone window manager, you may need to have sound playing while plugging in the HDMI cable.
mplay and other application could be configured to use special HDMI device as audio output. But flashplugin could only use default device. The following method is used to override default device. But you need to change it back when your TV is disconnected from HDMI port.
If the test is successful, create or edit your ~/.asoundrc file to set HDMI as the default audio device.
~/.asoundrc
pcm.!default {
type hw
card 1
device 3
}
Or if the above configuration does not work try:
~/.asoundrc
defaults.pcm.card 1 defaults.pcm.device 3 defaults.ctl.card 1
Or if you alternatively succeed with
$ speaker-test -Dplug:hdmi
for your HDMI or DisplayPort port the following configuration will work (successfully tested on Lenovo ThinkPad T430s):
~/.asoundrc
pcm.!default {
type plug
slave.pcm "hdmi"
}
PCM through HDMI does not work (Intel Gfx)
As of Linux 3.1 multi-channel PCM output through HDMI with a Intel card (Intel Eaglelake, IbexPeak/Ironlake,SandyBridge/CougarPoint and IvyBridge/PantherPoint) is not yet supported. Support for it has been recently added and expected to be available in Linux 3.2. To make it work in Linux 3.1 you need to apply the following patches:
HDMI 5.1 sound goes to wrong speakers
Sound can be redirected to the intended speakers using ALSA's remap function.
/etc/asound.conf
pcm.!hdmi-remap {
type asym
playback.pcm {
type plug
slave.pcm "remap-surround51"
}
}
pcm.!remap-surround51 {
type route
slave.pcm "hw:0,3"
ttable {
0.0= 1
1.1= 1
2.4= 1
3.5= 1
4.2= 1
5.3= 1
}
}
Intel Cannon Lake PCH cAVS
On Intel Cannon Lake (eg. HP ZBook 15 G6), the integrated sound chipset requires ALSA firmware, and the following kernel module parameters are required:
options snd-hda-intel dmic_detect=0 options snd-hda-intel model=laptop-amic enable=yes
That should enable both sound and microphone.
Applications
SDL: No sound with SDL applications
If you get no sound using SDL based applications, try setting the environment variable SDL_AUDIODRIVER to alsa.
OpenAL: No sound in applications that use OpenAL
OpenAL defaults to PulseAudio. To instruct it to try ALSA first:
/etc/openal/alsoft.conf
drivers=alsa,pulse
VirtualBox: Virtual machine has no sound
If you experience problems with VirtualBox, the following command might be helpful:
$ alsactl init
Found hardware: "ICH" "SigmaTel STAC9700,83,84" "AC97a:83847600" "0x8086" "0x0000" Hardware is initialized using a generic method
You might need to activate the ALSA output in your audio software as well. You might also try selecting different sound devices in your virtual machine settings to find one that works.
Others: Generic application problems
For other applications who insist on their own audio setup, e.g., XMMS or MPlayer, you would need to set their specific options.
For MPlayer or mpv, add the following line to the respective configuration file:
ao=alsa
Eg. for XMMS2, go into their options and make sure the sound driver is set to ALSA, not oss.
To do this in XMMS:
- Open XMMS
- Options > Preferences.
- Choose the ALSA output plugin.
For applications which do not provide a ALSA output, you can use aoss from the alsa-oss package. To use aoss, when you run the program, prefix it with aoss, e.g.:
aoss realplay
pcm.!default{ ... } doesnt work for me anymore. but this does:
pcm.default pcm.dmixer
Other Issues
Simultaneous playback problems
If you are having problems with simultaneous playback, and if PulseAudio is installed, its default configuration is set to "hijack" the soundcard. Some users of ALSA may not want to use PulseAudio and are quite content with their current ALSA settings. One fix is to edit /etc/asound.conf and comment out the following lines:
# Use PulseAudio by default
pcm.!default {
type pulse
fallback "sysdefault"
hint {
show on
description "Default ALSA Output (currently PulseAudio Sound Server)"
}
}
Commenting the following out also may help:
ctl.!default {
type pulse
fallback "sysdefault"
}
This may be a much simpler solution than completely uninstalling PulseAudio.
Effectively, here is an example of a working /etc/asound.conf:
pcm.dmixer {
type dmix
ipc_key 1024
ipc_key_add_uid 0
ipc_perm 0660
}
pcm.dsp {
type plug
slave.pcm "dmix"
}
/etc/asound.conf file was intended for and used successfully with a global MPD configuration. See #Problems with availability to only one user at a time.dmix to work with vanilla ALSA, see the upstream documentation. In particular, you probably want to replace dsp in the above config with !default. Also, if you notice this causes certain applications to skip while playing (i.e. sound "glitchy"), and complain about underrun occurring, you may want to tweak the slave.buffer_size inside pcm.dmixer.Removing old ALSA state file (asound.state)
The alsa-utils package provides alsa-store.service which automatically stores the current ALSA state to /var/lib/alsa/asound.state upon system shutdown. This can be problematic for users who are trying to reset their current ALSA state as the asound.state file will be recreated with the current state upon every shutdown (e.g., attempting to remove user-defined channels from the mixer). The alsa-store.service service may be temporarily disabled by creating the following empty file:
# mkdir -p /etc/alsa # touch /etc/alsa/state-daemon.conf
The presence of state-daemon.conf prevents alsa-store.service from saving asound.state during shutdown. After disabling this service, the asound.state file may be removed as such:
# rm /var/lib/alsa/asound.state
After rebooting, the previous ALSA state should be lost and the current state should be reset to defaults. Re-enable alsa-store.service by deleting the condition file we created:
# rm /etc/alsa/state-daemon.conf
On the next shutdown, the asound.state file should be recreated with ALSA defaults. The file may also be generated immediately using:
# alsactl store
If you want to clean ALSA state without rebooting, you can use rmmod to remove the sound driver module, then manually delete the unwanted entries in asound.state, and then use modprobe to reinstall the sound driver module.
Problems with availability to only one user at a time
You might find that only one user can use the dmixer at a time. This is probably ok for most, but for those who run mpd as a separate user this poses a problem. When mpd is playing a normal user cannot play sounds though the dmixer. While it is quite possible to just run mpd under a user's login account, another solution has been found. Adding the line ipc_key_add_uid 0 to the pcm.dmixer block disables this locking. The following is a snippet from asound.conf, the rest is the same as above.
...
pcm.dmixer {
type dmix
ipc_key 1024
ipc_key_add_uid 0
ipc_perm 0660
slave {
...
Crackling/popping on Dell laptops
Check if you have i8kutilsAUR installed and if anything (e.g. i8kmon.service) is reading or writing to the interface exposed by the module, as i8kutils BIOS system calls block the kernel for a moment on some systems. See warning in Fan speed control#Dell laptops for more details.