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# Advanced Linux Sound Architecture/Troubleshooting

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From ArchWiki
< Advanced Linux Sound Architecture

## 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
    }

Note: You will probably have to restart the computer, as restarting the alsa daemon did not load the new configuration. Also, if the configuration does not work even after restarting, try changing `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.

Note:

- Setting a high value for `Pre-Amp` can 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

This article or section is out of date.

Reason: The `ondemand` governor was the default for a while, but has been replaced for years by `schedutil`, is this still applicable? (Discuss in Talk:Advanced Linux Sound Architecture/Troubleshooting)

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.

Note: The script is incompatible with Python 3, which is the default Python implementation on Arch Linux. In order to use the script, replace all occurrences of `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.

Note: It may be necessary to put this "options" line below (after) any "alias" lines about your card.

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:

- drm: support routines for HDMI/DP ELD
- drm/i915: pass ELD to HDMI/DP audio driver

#### 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

This article or section is out of date.

Reason: User reports of the kernel parameter no longer being needed. No bug report linked to follow up… (Discuss in Talk:Advanced Linux Sound Architecture/Troubleshooting)

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"
    }

Note: This `/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.
Note: Alternatively, if you do not have PulseAudio installed, and just want to get `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.

Category:

- Sound

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