Note that I do not work for Skype nor have any affiliation with them - I just thought I'd chuck together a few solutions to common sound problems in Linux using ALSA, seeing as I've dealt with a few of them. If you have something to add, please do! I'm doing this in the hopes that it will help new Skype/Linux users, and that we might avoid some repeat questions on the forums. Anyway, here goes.

Prerequisites:
You have already read and followed the Linux Sound FAQ written by futt

The main aim of this FAQ is to give hints on using Skype in combination with ArtsD and ALSA with dmix and dsnoop plugins!
This way you can configure all your Alsa-capable applications to use Alsa!

Question: Why does Skype "freeze" when I issue a call or try to answer one?
A very common reason for Skype to "freeze" is a busy sound device. The symptoms are: when calling someone Skype permanently displays "Connecting..." but nothing else happens. Hiding the main window make Skype unavailable; choosing file->close makes Skye freeze completely.
To get out of this, either kill all Skype processes, or the processes that are using your sound-card.
Ways to avoid this issue: [list]
Question: How do I know if I am running OSS drivers or ALSA drivers?
There are multiple ways to check this (if you don't have alsa, but have sound, then you have OSS drivers!):[list]Look if /proc/asound/ exists and has folders for all of your active audio devices (/proc/asound/card0, ...)
[*]With lsmod list your modules and look if audio-related modules have a name starting with "snd-"
[*]If you are using devfs or udev, check for the device folder /dev/snd/ and it's content to detect alsa.[list]

Question: How do I test my alsa configuration?
For testing playback, make sure you have a *.wav audio file available, best is 48kHz stereo 16-bit audio, but other formats can do it as well in most cases.

CODE

# Try play back

aplay -D plughw:0,0 Test.wav

# Try recording:

arecord -D plughw:0,0 RecTest.wav

Then of course listen to the file you did record to see if it worked correctly. These tests should done from Linux console with NO GRAPHICAL SESSION ACTIVE, just to avoid 99% of risks that some other application is using audio, potentially interfering with the tests. Normally when audio device is "blocked" aplay and arecord will wait for the device to get available, and output information once they start playing, so check running applications if either aplay or arecord stay silent! Mixer settings should be changed with alsamixer ("Capture" channel selected for capture (an volume = 0), "Mic" channel selected for capture and muted at maximum volume, "Mic Select" = Mic1 for nearly all configurations, "Mono Out" = mix)


Question: How to operate when I have multiple sound cards (onbard, PCI, usb, ...)
You can use common alsa tools which each of the sound cards as well as each of their sub-devices! This is done by the -D option of aplay and arecord. This option either takes a device name as in ~/.asoundrc, or "hw:x,y" or "plughw:x,y" where hw = hardware, plughw = hardware with auto-resampling when needed, x = index of soundard (0..n), y = index of sub-device (0..n).
The list of devices and card you have can be displayed with

CODE

# For playback:

aplay -l

**** List of PLAYBACK Hardware Devices ****

card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 0: Intel ICH [Intel 82801DB-ICH4]

 Subdevices: 0/1

 Subdevice #0: subdevice #0

card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 4: Intel ICH - IEC958 [Intel 82801DB-ICH4 - IEC958]

 Subdevices: 1/1

 Subdevice #0: subdevice #0

# For recording:

arecord -l

**** List of CAPTURE Hardware Devices ****

card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 0: Intel ICH [Intel 82801DB-ICH4]

 Subdevices: 0/1

 Subdevice #0: subdevice #0

card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 1: Intel ICH - MIC ADC [Intel 82801DB-ICH4 - MIC ADC]

 Subdevices: 1/1

 Subdevice #0: subdevice #0

card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 2: Intel ICH - MIC2 ADC [Intel 82801DB-ICH4 - MIC2 ADC]

 Subdevices: 1/1

 Subdevice #0: subdevice #0

card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 3: Intel ICH - ADC2 [Intel 82801DB-ICH4 - ADC2]

 Subdevices: 1/1

 Subdevice #0: subdevice #0

The example above shows my single onboard soundcard in my Laptop. For playback you have "hw:0,0" and "hw:0,4", and use your brain for recording :wink:.
Mixer settings can be adjusted with alsamixer, and you may use the -c option to select the soundcard (0..n).


Question: I have a Webcam with integrated mic, how do I use that with Skype?
There is not to much work to get this if previous step went right. You should just edit the SLAVE parameter for pcm.dsnoop if your Webcam-audio is not the first alsa sound device, or SLAVE for pcm.dmix in case your Webcam takes precedence to built-in soundcard!
Without dmix and dsnoop this would look like the following:

CODE

pcm.asymed {

       type asym

       playback.pcm "hw:0,0"

       capture.pcm "hw:1,0"

}



pcm.!default {

       type plug

       slave.pcm "asymed"

}

Question: How do I activate dmix for ALSA
To activate dmix in alsa you need to create a alsa configuration file "~/.asoundrc" (you can also create the file /etc/asound.conf to have a system-wide configuration):

CODE

pcm.asymed {

       type asym

       playback.pcm "dmix"

       capture.pcm "dsnoop"

}

pcm.!default {

       type plug

       slave.pcm "asymed"

}

The asymed PCM is used to combine input and output channels (dmix for playback, allowing multiple applications to play at the same time, dsnoop for recording, allowing many applications to record at the same time).
The reason why I redefined the default PCM to be of type plug is to allow automatic format conversion between the application and alsa-plugins.


Question: Depending on the sound I play I get "noise"
This may be because of sampling-rate conversions between 44.1kHz and 48kHz as the default pcm.dmix uses 48kHz rate. (CD-Audio and most common audio files are encoded in 44.1kHz, but suond-cards are moving to 48kHz which is also used for SPDIF output and half of new 96kHz sample rate) This is also true for pcm.dsnoop

The few lines you could add to the top of your .asoundrc will configure dmix to run with 44.1kHz, which is most commonly used for audio files as it's the one used on CDs.
These lines were extracted from the alsa.conf file installed with alsa-lib and adjusted for 44.1kHz:

CODE

pcm.!dmix {

       type dmix

       ipc_key 5678293

       ipc_key_add_uid yes

       slave {

               pcm "hw:0,0"

               format S16_LE

               rate 44100

       }

}

pcm.!dsnoop {

       type dsnoop

       ipc_key 5778293

       ipc_key_add_uid yes

       slave {

               pcm "hw:0,0"

               format S16_LE

               rate 44100

       }

}

Note, you need to call them "pcm.!dmix" and "pcm.!dsnoop" so that the "original" ones from the alsa configuration get overridden, like for the "pcm.!default".


Question: How do I tell Skype to use my new alsa default PCM?
You need to pipe Skype through a ALSA-aware sound-daemon. In my case I use artsd.
With the following configuration of artsd I get good results:[list] Run with highest priority
[*]Audio device: Advanced Linux Sound Architecture (Alsa)
[*][x] Full duplex
[*]Custom sampling rate: 48000 Hz (should be same as for dmix and dsnoop plugins!)[list]pcm.!dmix {
type dmix
ipc_key 5678293
ipc_key_add_uid yes
slave {
pcm "hw:0,0"
period_time 0
period_size 128
buffer_size 2048
format S16_LE
rate 48000
}
}
pcm.!dsnoop {
type dsnoop
ipc_key 5778293
ipc_key_add_uid yes
slave {
pcm "hw:0,0"
period_time 0
period_size 128
buffer_size 2048
format S16_LE
rate 48000
}
}
...[/CODE]Don't forget to configure both dmix AND dsnoop parts (buffersizes should be equal) as failing to do so will probably make artsd refuse starting with cryptic error messages like:

CODE

AudioSubSystem::handleIO: write failed

len = 4096, can_write = 6000, errno = 17 (File exists)

The buffer-sizes you should select shall be a multiple of the buffer used by Skype (in the example buffer_size = 20ms: 48000 * 2 * 2 = 192000 bytes/second => 3840 for 20ms [Rate, 48kHz * Channels, Stereo * Bytes/Sample, 16bit]) If this does not work as expected, or you have suggestions for improvement of this documentation, tell me!
After some further testing, and with suggestion from vgough, the most important is to have buffer_size and period_size EQUAL in Artsd and Alsa configuration, their exact size is far less important (it just affects the delaying behavior... currently delay is always growing over time on my Laptop, less than a second at start, over 10s after 10min of conversation. To small buffers can cause some sound-drops)


Question: How do I need to configure and run Skype
You run skype with

CODE

artsdsp -m skype

and you need to configure skype to use /dev/dsp as output device!! This is the only one monitored by artsdsp (and the other sound wrappers)


Question: Does this solve delay problems
The last updated were not yet tested for delay, just for sound quality, next update soon (after my next longer call :wink: )
With buffer size as updated on 15 January I got delay of 2-3s after 8 minutes of call (instead of 30 with previous values right from the start!)


Question: What should I do if I can't get reasonable sound quality with Skype?
Check above, and if this buffer-size adjusting still fails, the only solution left is to run Skype with direct access to your /dev/dsp kernel-emulated oss device, but don't forget to stop all audio software you have runnning, and in case of Artsd, let it release the sound-device. If you try to use Skype without sound-wrapper and your sound-device is blocked, you may have Skype (partially) freezing.


Changelog:
19 Nov 2004: Small fix for dsnoop and dmix sections in .asoundrc (added "!")
20 Dec 2004: Added combination of playback & record on different sound cards
12 Jan 2005: Update for buffer-settings which should fix much choppyness!
15 Jan 2005: Update for buffer-settings which should reduce delay very much

Can someone make this a "sticky"?
Even better can a Skype staff sorts
this forum and "Linux Sound FAQ"
started by futt to create/update Linux
FAQ. There's alot of garbages in the
latter topic, only futt's initial comments
and alittle bit of others are worthwhile
keeping. After creating/updating
Linux FAQ, encourage the users to see
this new FAQ before posting.

Sheng-Chieh

wrong topic, sry

You run skype with

CODE

artsdsp -m skype

and you need to configure skype to use /dev/dsp as output device!! This is the only one monitored by artsdsp (and the other sound wrappers)

Usefull script for those who not using kde but have artsd (thanks to Azarah):

CODE

#!/bin/bash



if [ -z "`pidof artsd`" ]; then

       artsd -a alsa -d -D default &



       sleep 1

fi



artsdsp -m /opt/skype/skype.bin

I'm using Mandrake 10.1 OE and skype-0.92.0.12-mdk.

Skype is configured for /dev/dsp.

No matter what I try, I still end up with the following skype error:

artsdsp -m skype
Engine::init invoked. user is
Sound device is not defined
Engine::init success
SkyWindowBase::onlineStatusChanged setting pixmap
DialPad: DialPad
language selected: 5

Alsa loads perfectly upon boot, all applications have sound working.

My modprobe.conf contains the following:

install snd-cs46xx /sbin/modprobe --first-time --ignore-install snd-cs46xx && { /sbin/modprobe snd-pcm-oss; /bin/true; }

# ALSA portion
alias sound-slot-0 snd-cs46xx
alias char-major-116 snd

# OSS/Free portion - card #1
alias char-major-14 soundcore
alias sound-slot-0 snd-card-0
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

alias /dev/mixer snd-mixer-oss
alias /dev/dsp snd-pcm-oss
alias /dev/midi snd-seq-oss

options enable int array=8
options mmap_valid int array=8
options snd cards_limit=1
options snd-pcm-oss nonblock_open=1

My .asoundrc file contains:

pcm.cs46xx {
type hw
card 0
}

ctl.cs46xx {
type hw
card 0
}

If anyone has any ideas to make this work, I'd sure like to see them because I'm at a roadblock....

George

[quote=gczerw]I'm using Mandrake 10.1 OE and skype-0.92.0.12-mdk.

Skype is configured for /dev/dsp.[/quote]
Did you check that /dev/dsp really exists? Usually for the first soundcard a /dev/dsp1 is created, and /dev/dsp is a symbolic link to the \"main\" soundcard.

[quote=gczerw]alias /dev/mixer snd-mixer-oss
alias /dev/dsp snd-pcm-oss
alias /dev/midi snd-seq-oss[/quote]
I would change \"alias /dev/dsp snd-pcm-oss\" to \"alias /dev/dsp1 snd-pcm-oss\" and make sure the symbolic link \"/dev/dsp -> /dev/dsp1\" exists.[/quote]

If you start Skype without sound-daemon wrapper, thenyour .asoundrc has no influence, and you need to take care that no other application uses the sound-device.
If you use a sound-daemon wrapper, then you need to try until you get the best setting...

Mandrake 10.1 uses UDEV, and not devfs.
/dev/dsp does exist, and is the only /dev/dsp device listed. /dev/midi and /dev/mixer also exist. All have RW permissions for the user as well as for group audio.
[/list]

I've installed the latest kernel version 2.6.9.1 with the latest ALSA driver 1.0.7 and the choppy sound I experienced seems to be gone! Any one else have a similar experience? Maybe the newest ALSA driver and Kernel fixed some backwards compatibility with OSS?

I've installed the latest kernel version 2.6.9.1 with the latest ALSA driver 1.0.7 and the choppy sound I experienced seems to be gone! Any one else have a similar experience? Maybe the newest ALSA driver and Kernel fixed some backwards compatibility with OSS?

Where did you get that kernel version? :shock:

I have no choppy sound when I run Skype on kernel-oss-emulation, but with sound-daemon it depends verymuch on the daemon/config/hardware combination :-(
My wish: get rid of the sound-daemon! Either by direct support of Alsa, or temporarly by support of aoss!

I spoke too soon, the sound quality improved significantly with the kernel and ALSA update but the sound becomes choppy after about 10 minutes of conversation. I'm still optimistic because skype is now usable in my linux box, while before the sound would be choppy afer a few seconds of conversation.

Hi!

I'm using mandrake 10.1 with an abit is7-e mainboard onboard sound card.
Intel Corp. 82801EB/ER (ICH5/ICH5R) AC'97 Audio Controller
with this driver snd-intel8x0

I can't here others, an other's cant here me ..
Alsa set in kde to full duplex mode, i'm already at the audio group, got rw permission to /dev/dsp, , and I get an error on the console screen when I start skype Soundcard not definied. Xmms works fine with alsa.

Any ideal ?

I'm using mandrake 10.1 with an abit is7-e mainboard onboard sound card.  
Intel Corp. 82801EB/ER (ICH5/ICH5R) AC'97 Audio Controller
with this driver snd-intel8x0

I can't here others, an other's cant here me ..  
Alsa set in kde to full duplex mode, i'm already at the audio group, got rw permission to /dev/dsp, , and I get an error on the console screen when I start skype Soundcard not definied. Xmms works fine with alsa.

Do you start skype with "artsdsp -m skype"?

I also try it, but when I try to call someone , it's do a segfault (core dumped)

I think quite a few Madrake 10.1 users had problems, which possibly are caused by the move to GCC 3.4.* compiler...

As I don't use Mandrake I can't tell very much :-(
Does skype work fine with direct OSS access?

How can I try the direct oss access? Just start simply skype from console?

How can I try the direct oss access? Just start simply skype from console?

That's it, just make sure arts has released the sound-card when you try calling!

same thing

any other ideal?

Hummmm.

I still have the problem of skype complaining about the in-famous "Sound device is not defined" oh, and by the way I'm a Gentoo User.

Could someone tell me what I need to do in order to get skype working.

[size=18]This is what happens when I try to run skype

CODE

Mustang ~ # artsdsp -m skype

Running artsd found

Starting artsd wrapped skype

Engine::init invoked. user is

Sound device is not defined

Engine::init success

UI: Engine::Engine credentials loaded

SkyWindowBase::onlineStatusChanged setting pixmap

DialPad: DialPad

language selected: 5

Engine::init invoked. user is **User-Name**

Sound device is not defined

Engine::init success

/etc/modules.d/alsa - Includes

CODE

options snd device_mode=0666

alias snd-card-0 snd-intel8x0

alias sound-slot-0 snd-intel8x0



alias sound-service-0-0 snd-mixer-oss

alias sound-service-0-3 snd-pcm-oss

alias sound-service-0-12 snd-pcm-oss



alias /dev/mixer snd-mixer-oss

alias /dev/dsp snd-pcm-oss

/etc/init.d/alsasound {stop/start} - Includes

CODE

Mustang ~ # /etc/init.d/alsasound stop

* WARNING:  you are stopping a boot service.

* Storing ALSA Mixer Levels ...                                          [ ok ]

* Unloading ALSA ...                                                     [ ok ]

* Unloading ALSA modules ...                                             [ ok ]

Mustang ~ # /etc/init.d/alsasound start

* Loading ALSA modules ...

*   Loading: snd-card-0 ...                                              [ ok ]

*   Loading: snd-seq-oss ...                                             [ ok ]

*   Loading: snd-pcm-oss ...                                             [ ok ]

* Restoring Mixer Levels ...                                             [ ok ]

lsmod - Gives

CODE

Mustang ~ # lsmod

Module                  Size  Used by

snd_pcm_oss            47528  0

snd_mixer_oss          17152  1 snd_pcm_oss

snd_seq_oss            31040  0

snd_seq_midi_event      7168  1 snd_seq_oss

snd_seq                47504  4 snd_seq_oss,snd_seq_midi_event

snd_seq_device          8204  2 snd_seq_oss,snd_seq

snd_intel8x0           29884  4

snd_ac97_codec         64352  1 snd_intel8x0

snd_pcm                85640  5 snd_pcm_oss,snd_intel8x0,snd_ac97_codec

snd_timer              22980  4 snd_seq,snd_pcm

snd                    50212  13 snd_pcm_oss,snd_mixer_oss,snd_seq_oss,snd_seq,snd_seq_device,snd_intel8x0,snd_ac97_codec,snd_pcm,snd_timer

snd_page_alloc          9416  2 snd_intel8x0,snd_pcm

fglrx                 222396  9

amixer - Gives

CODE

Mustang ~ # amixer

Simple mixer control 'Master',0

 Capabilities: pvolume pswitch pswitch-joined

 Playback channels: Front Left - Front Right

 Limits: Playback 0 - 31

 Front Left: Playback 19 [61%] [on]

 Front Right: Playback 19 [61%] [on]

Simple mixer control 'Master Mono',0

 Capabilities: pvolume pvolume-joined pswitch pswitch-joined

 Playback channels: Mono

 Limits: Playback 0 - 31

 Mono: Playback 19 [61%] [on]

Simple mixer control 'Headphone',0

 Capabilities: pvolume pswitch pswitch-joined

 Playback channels: Front Left - Front Right

 Limits: Playback 0 - 31

 Front Left: Playback 19 [61%] [on]

 Front Right: Playback 19 [61%] [on]

Simple mixer control '3D Control - Center',0

 Capabilities: volume volume-joined

 Playback channels: Mono

 Limits: 0 - 15

 Mono: 0 [0%]

Simple mixer control '3D Control - Depth',0

 Capabilities: volume volume-joined

 Playback channels: Mono

 Limits: 0 - 15

 Mono: 0 [0%]

Simple mixer control '3D Control - Switch',0

 Capabilities: pswitch pswitch-joined

 Playback channels: Mono

 Mono: Playback [on]

Simple mixer control 'PCM',0

 Capabilities: pvolume pswitch pswitch-joined

 Playback channels: Front Left - Front Right

 Limits: Playback 0 - 31

 Front Left: Playback 19 [61%] [on]

 Front Right: Playback 19 [61%] [on]

Simple mixer control 'PCM Out Path & Mute',0

 Capabilities:

 Mono:

Simple mixer control 'Line',0

 Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-joined cswitch-exclusive

 Capture exclusive group: 0

 Playback channels: Front Left - Front Right

 Capture channels: Front Left - Front Right

 Limits: Playback 0 - 31

 Front Left: Playback 19 [61%] [on] Capture [off]

 Front Right: Playback 19 [61%] [on] Capture [off]

Simple mixer control 'CD',0

 Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-joined cswitch-exclusive

 Capture exclusive group: 0

 Playback channels: Front Left - Front Right

 Capture channels: Front Left - Front Right

 Limits: Playback 0 - 31

 Front Left: Playback 19 [61%] [on] Capture [off]

 Front Right: Playback 19 [61%] [on] Capture [off]

Simple mixer control 'Mic',0

 Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch cswitch-joined cswitch-exclusive

 Capture exclusive group: 0

 Playback channels: Mono

 Capture channels: Front Left - Front Right

 Limits: Playback 0 - 31

 Mono: Playback 19 [61%] [on]

 Front Left: Capture [off]

 Front Right: Capture [off]

Simple mixer control 'Mic Boost (+20dB)',0

 Capabilities: pswitch pswitch-joined

 Playback channels: Mono

 Mono: Playback [on]

Simple mixer control 'Mic Select',0

 Capabilities:

 Mono:

Simple mixer control 'Video',0

 Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-joined cswitch-exclusive

 Capture exclusive group: 0

 Playback channels: Front Left - Front Right

 Capture channels: Front Left - Front Right

 Limits: Playback 0 - 31

 Front Left: Playback 19 [61%] [on] Capture [off]

 Front Right: Playback 19 [61%] [on] Capture [off]

Simple mixer control 'Phone',0

 Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch cswitch-joined cswitch-exclusive

 Capture exclusive group: 0

 Playback channels: Mono

 Capture channels: Front Left - Front Right

 Limits: Playback 0 - 31

 Mono: Playback 19 [61%] [on]

 Front Left: Capture [off]

 Front Right: Capture [off]

Simple mixer control 'IEC958',0

 Capabilities: pswitch pswitch-joined

 Playback channels: Mono

 Mono: Playback [on]

Simple mixer control 'IEC958 Playback AC97-SPSA',0

 Capabilities: volume volume-joined

 Playback channels: Mono

 Limits: 0 - 3

 Mono: 1 [33%]

Simple mixer control 'PC Speaker',0

 Capabilities: pvolume pvolume-joined pswitch pswitch-joined

 Playback channels: Mono

 Limits: Playback 0 - 15

 Mono: Playback 9 [60%] [on]

Simple mixer control 'Aux',0

 Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-joined cswitch-exclusive

 Capture exclusive group: 0

 Playback channels: Front Left - Front Right

 Capture channels: Front Left - Front Right

 Limits: Playback 0 - 31

 Front Left: Playback 19 [61%] [on] Capture [off]

 Front Right: Playback 19 [61%] [on] Capture [off]

Simple mixer control 'Mono Output Select',0

 Capabilities:

 Mono:

Simple mixer control 'Capture',0

 Capabilities: cvolume cswitch cswitch-joined

 Capture channels: Front Left - Front Right

 Limits: Capture 0 - 15

 Front Left: Capture 0 [0%] [on]

 Front Right: Capture 0 [0%] [on]

Simple mixer control 'Mix',0

 Capabilities: cswitch cswitch-joined cswitch-exclusive

 Capture exclusive group: 0

 Capture channels: Front Left - Front Right

 Front Left: Capture [on]

 Front Right: Capture [on]

Simple mixer control 'Mix Mono',0

 Capabilities: cswitch cswitch-joined cswitch-exclusive

 Capture exclusive group: 0

 Capture channels: Front Left - Front Right

 Front Left: Capture [off]

 Front Right: Capture [off]

Simple mixer control 'External Amplifier',0

 Capabilities: pswitch pswitch-joined

 Playback channels: Mono

 Mono: Playback [on]

[/code]

that's the same for me..
also not working :evil:

[quote=Mustang-Tech]I still have the problem of skype complaining about the in-famous "Sound device is not defined"
For me Skype often says sound device not defined, but still works (I'm on Gentoo as well, x86 stable)
Are you running x86 or ~x86, or amd64?