Amarok 2.5.0-1 - no sound

Updated to Amarok 2.5.0-1 and now there is no sound, although the track appears to 'progress' normally.
Going to Settings->Configure Amarok->Playback->configure Phonon I was able to select the preferred audio device and got sound when pressing the 'Test' button. Still no sound when playing a (ogg) file though.
Downgraded back to Amarok 2.4.3-3 and all is well again.
Anyone else seeing this?

@Chembro
Thanks for the update regarding the 'death' of the Xine backend
The Amarok 2.5.0-1 has fixed a really annoying problem I was having. Every time the previous amarok started, it sat the system at >60% load until you opened the Amarok window, then all was well. I did try to diagnose things, I got a 'flood' of X errors when running 'amarok' from a console (no debug flag needed). These ceased as soon as you opened the Amarok window. I worked round it by remembering to open Amarok every time I started the system.
Still one 'cosmetic' glitch left: if you use the 'Now Playing' KDE widget, Amarok 'forgets' to update the artwork on the first track, when playing an album. When track 2 starts, the artwork changes. Until then you either get a 'blank' or whatever the previous album was.
The above I can live with, I still prefer Amarok to the other players available for Linux by a long way.

Similar Messages

No sound troubleshooting

I have a problem with my Arch install having no sound since my last reboot.
I am mainly running KDE and KDE apps, so it *might* be phonon-related, but I'm not conviced that's the issue. In the past, I used to have 7 or 8 soundcards in the KDE settings module for Phonon, and specifically I had hw(0,0), hw(0,1) and hw(1,3). But these no longer shows up. Though, in the past I also had pulseaudio installed. (Which after my reboot caused all sort of errors, so I removed it.) In the really distant past, I didn't have pulseaudio installed and to the best of my memory, hw(0,0), hw(0,1) and hw(1,3) didn't show up in the settings module either.
I have already tried google for all error messages I have got from mpg123, mplayer, xine and aplay, and tried to follow diagnostics and troubleshootings I have found in arch bbs and arch wiki. Of course, I could have overlooked something, so please be kind with me and assume I have forgot to do all the basics... :-(
I have speakers connected to both SPEAKER (green 3.5 mm jack) and LINE OUT (black 3.5 mm jack) in the back panel. I have no speakers attached to my green 3.5 mm speaker jack on the front panel. My user is added to the audio group, and I have also tried to run all music applications as root.
My main theory is that something is blocking the sound card, so ALSA can't use it. But I can't figure out what.
Here is most of what I believe could be of use:
Packages installed:
phonon, gstreamer, xine-lib, esound, jack, alsa, mplayer, vlc, amarok
/proc/asound/version:
Advanced Linux Sound Architecture Driver Version 1.0.22.1
/proc/asound/cards:
0 [SB ]: HDA-Intel - HDA ATI SB
HDA ATI SB at 0xfe024000 irq 16
1 [HDMI ]: HDA-Intel - HDA ATI HDMI
HDA ATI HDMI at 0xfdffc000 irq 19
/proc/asound/devices:
2: : timer
3: [ 0- 2]: digital audio capture
4: [ 0- 1]: digital audio playback
5: [ 0- 1]: digital audio capture
6: [ 0- 0]: digital audio playback
7: [ 0- 0]: digital audio capture
8: [ 0- 0]: hardware dependent
9: [ 0] : control
10: [ 1- 3]: digital audio playback
11: [ 1- 0]: hardware dependent
12: [ 1] : control
13: : sequencer
/proc/asound/timers:
G0: system timer : 3333.333us (10000000 ticks)
P0-0-0: PCM playback 0-0-0 : SLAVE
P0-0-1: PCM capture 0-0-1 : SLAVE
P0-1-0: PCM playback 0-1-0 : SLAVE
P0-1-1: PCM capture 0-1-1 : SLAVE
P0-2-1: PCM capture 0-2-1 : SLAVE
P0-2-3: PCM capture 0-2-3 : SLAVE
P1-3-0: PCM playback 1-3-0 : SLAVE
/proc/asound/pcm:
00-00: ALC889A Analog : ALC889A Analog : playback 1 : capture 1
00-01: ALC889A Digital : ALC889A Digital : playback 1 : capture 1
00-02: ALC889A Analog : ALC889A Analog : capture 2
01-03: ATI HDMI : ATI HDMI : playback 1
Chipset:
Realtek ALC889A
Volumes on all channels have been confirmed with alsamixer.
aplay -l:
**** List of PLAYBACK Hardware Devices ****
card 0: SB [HDA ATI SB], device 0: ALC889A Analog [ALC889A Analog]
Subdevices: 0/1
Subdevice #0: subdevice #0
card 0: SB [HDA ATI SB], device 1: ALC889A Digital [ALC889A Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: HDMI [HDA ATI HDMI], device 3: ATI HDMI [ATI HDMI]
Subdevices: 1/1
Subdevice #0: subdevice #0
Thus, I tried aplay on all cards/devices:
aplay -D plughw:0,0 /usr/share/sounds/alsa/Front_Center.wav
aplay: main:654: audio open error: Device or resource busy
aplay -D plughw:0,1 /usr/share/sounds/alsa/Front_Center.wav:
Playing WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono
aplay -D plughw:1,3 /usr/share/sounds/alsa/Front_Center.wav:
Playing WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono
Note: Nothing ever played in any speakers, despite the fact that aplay claimed so.
So I thought, maybe sound levels was set to 0 on plughw0,1. So I ran
alsamixer -D plughw:0,1
but it only gave me
ALSA lib control.c:902:(snd_ctl_open_noupdate) Invalid CTL plughw:0,1
cannot open mixer: File or directory not found
aplay -L:
null
Discard all samples (playback) or generate zero samples (capture)
default:CARD=SB
HDA ATI SB, ALC889A Analog
Default Audio Device
front:CARD=SB,DEV=0
HDA ATI SB, ALC889A Analog
Front speakers
surround40:CARD=SB,DEV=0
HDA ATI SB, ALC889A Analog
4.0 Surround output to Front and Rear speakers
surround41:CARD=SB,DEV=0
HDA ATI SB, ALC889A Analog
4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=SB,DEV=0
HDA ATI SB, ALC889A Analog
5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=SB,DEV=0
HDA ATI SB, ALC889A Analog
5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=SB,DEV=0
HDA ATI SB, ALC889A Analog
7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
iec958:CARD=SB,DEV=0
HDA ATI SB, ALC889A Digital
IEC958 (S/PDIF) Digital Audio Output
hdmi:CARD=HDMI
HDA ATI HDMI, ATI HDMI
HDMI Audio Output
aplay <mp3 file>:
ALSA lib pcm_dmix.c:1018:(snd_pcm_dmix_open) unable to open slave
aplay: main:654: audio open error: Device or resource busy
xine <mp3 file>:
This is xine (X11 gui) - a free video player v0.99.6-[DEBUG].
jackd 0.118.0
JACK compiled with System V SHM support.
loading driver ..
creating alsa driver ... hw:0|hw:0|1024|2|48000|0|0|nomon|swmeter|-|32bit
control device hw:0
the playback device "hw:0" is already in use. Please stop the application using it and run JACK again
cannot load driver module alsa
params.c:OpenConfFile() - Unable to open configuration file "/home/ac/.smb/smb.conf":
Filen eller katalogen finns inte
params.c:OpenConfFile() - Unable to open configuration file "/etc/samba/smb.conf":
Filen eller katalogen finns inte
xine: could not connect to socket
xine: Filen eller katalogen finns inte
Playlist '/home/ac/.xine/xine-ui_old_playlist.tox' saved.
mplayer <mp3 file>:
MPlayer SVN-r31774-4.5.0 (C) 2000-2010 MPlayer Team
158 audio & 340 video codecs
mplayer: could not connect to socket
mplayer: No such file or directory
Opening audio decoder: [mp3lib] MPEG layer-2, layer-3
AUDIO: 44100 Hz, 2 ch, s16le, 320.0 kbit/22.68% (ratio: 40000->176400)
Selected audio codec: [mp3] afm: mp3lib (mp3lib MPEG layer-2, layer-3)
==========================================================================
[AO OSS] audio_setup: Can't open audio device /dev/dsp: Device or resource busy
[AO_ALSA] alsa-lib: pcm_hw.c:1293:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-16): Device or resource busy
[AO_ALSA] alsa-lib: pcm_dmix.c:1018:(snd_pcm_dmix_open) unable to open slave
[AO_ALSA] Playback open error: Device or resource busy
[JACK] cannot open server
[AO SDL] Samplerate: 44100Hz Channels: Stereo Format s16le
[AO_ALSA] alsa-lib: pcm_hw.c:1293:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-16): Device or resource busy
[AO_ALSA] alsa-lib: pcm_dmix.c:1018:(snd_pcm_dmix_open) unable to open slave
[AO SDL] Unable to open audio: No available audio device
DVB card number must be between 1 and 4
AO: [null] 44100Hz 2ch s16le (2 bytes per sample)
Video: no video
Starting playback...
vlc <mp3 file>:
pops up a dialog saying:
Potential ALSA version problem:
VLC failed to initialize your sound output device (if any).
Please update alsa-lib to version 1.0.23-2-g8d80d5f or higher to try to fix this issue.
Also it outputs to tty
VLC media player 1.1.1 The Luggage (revision exported)
Blocked: call to unsetenv("DBUS_ACTIVATION_ADDRESS")
Blocked: call to unsetenv("DBUS_ACTIVATION_BUS_TYPE")
Warning: call to signal(13, 0x1)
[0xa55120] main libvlc: Kör vlc med standardgränssnittet. Använd "cvlc" för att använda vlc utan gränssnitt.
Blocked: call to setlocale(6, "")
Blocked: call to sigaction(17, 0x7f0133433ac0, 0x7f0133433b60)
Blocked: call to setlocale(6, "")
Warning: call to signal(13, 0x1)
Warning: call to rand()
Warning: call to rand()
Warning: call to rand()
Blocked: call to setlocale(1, "C")
Blocked: call to setlocale(1, "sv_SE.UTF-8")
Blocked: call to setlocale(1, "C")
ALSA lib pcm_dmix.c:1018:(snd_pcm_dmix_open) unable to open slave
[0x7f012c013de0] oss audio output error: cannot open audio device (/dev/dsp)
[0x7f012c013de0] jack audio output error: failed to connect to JACK server
ALSA lib pcm_dmix.c:1018:(snd_pcm_dmix_open) unable to open slave
Blocked: call to signal(13, 0x7f012706cad0)
Warning: call to signal(13, (nil))
Blocked: call to sigaction(11, 0x7f0127b74850, (nil))
Blocked: call to sigaction(7, 0x7f0127b74850, (nil))
Blocked: call to sigaction(8, 0x7f0127b74850, (nil))
Blocked: call to sigaction(3, 0x7f0127b74850, (nil))
Warning: call to sigaction(14, 0x7f0127b74850, (nil))
[0x7f012c013de0] main audio output error: couldn't find a filter for the conversion f32l -> mpga
[0x7f012c013de0] main audio output error: couldn't create audio output pipeline
Warning: call to rand()
Warning: call to rand()
I'm unable to update alsa-lib to 1.0.23-2, 1.0.23-1 is the latest I can find in any repo. I tried
sudo pacman -Syu
sudo pacman -S alsa-lib
but it didn't help
Interesting modules loaded (lsmod | grep snd):
snd_seq_dummy 1447 0
snd_seq_oss 28928 0
snd_seq_midi_event 5420 1 snd_seq_oss
snd_seq 50530 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi_event
snd_seq_device 5241 3 snd_seq_dummy,snd_seq_oss,snd_seq
snd_pcm_oss 39096 0
snd_mixer_oss 16932 1 snd_pcm_oss
snd_hda_codec_atihdmi 2723 1
snd_hda_codec_realtek 267667 1
snd_hda_intel 21906 3
snd_hda_codec 76595 3 snd_hda_codec_atihdmi,snd_hda_codec_realtek,snd_hda_intel
snd_hwdep 6126 1 snd_hda_codec
snd_pcm 71653 4 snd_pcm_oss,snd_hda_intel,snd_hda_codec
snd_timer 19660 2 snd_seq,snd_pcm
snd 57225 16 snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_mixer_oss,snd_hda_codec_realtek,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm,snd_timer
soundcore 6089 1 snd
snd_page_alloc 7201 2 snd_hda_intel,snd_pcm
dmesg | grep -i hda
HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16
hda_codec: ALC889A: BIOS auto-probing.
input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:14.2/input/input3
HDA Intel 0000:01:05.1: PCI INT B -> GSI 19 (level, low) -> IRQ 19
HDA Intel 0000:01:05.1: setting latency timer to 64
hda-intel: IRQ timing workaround is activated for card #1. Suggest a bigger bdl_pos_adj.
Any ideas?

Could it possibly be something with ALSA?
ALSA in the Wiki
I always have problems with Sound at first and have to go in and unmute everything and set ALSA to start at system start. I was able to fix it from the Wiki guide.
Apologies if you're already past this step.

No sound in multimedia playback after update (xine,vlc,mplayer)

Hi!
For some days now I don't have sound anymore in amarok and vlc. Since the sound works on e.g. youtube I don't know how to find the root of the problem and of course not the solution. In amarok my sound card doesn't show up any more. I'm using ALSA. "aplay /usr/share/sounds/alsa/Front_Center.wav" doesn't give any output. (I don't know if it matters.)
I already spent time in searching (googling) by myself without success. Can anyone give me a hint?
Thanks in advance,
Maximalminimalist
Last edited by Maximalminimalist (2010-12-14 20:05:41)

Ultraman wrote:What vitals did you check? What DE/WM are you running? KDE by any chance?
I'm using i3. (tiling WM)
Ultraman wrote:Did you check the mixer? Anything muted there? Probably not, but worth a check.
Alsamixer should be fine. (Sound works on youtube anyway.)
Ultraman wrote:Have you blacklisted the snd-pcm-oss module to prevent some app using OSS instead of ALSA and claiming the soundcard that way? (rc.conf -> MODULES=(!snd-pcm-oss) )
No, I even had it in the modules but the "!" makes no difference.
Ultraman wrote:If you are running KDE: Have you checked Phonon's settings? Does it recognize any backend (e.g. Xine, Gstreamer)? What about sound hardware?
I'm not using KDE but amarok recognises xine.
Ultraman wrote:Tried starting VLC and Amarok from a terminal and looked at the output they send to it? Perhaps any error messages among them?
When I start VLC from terminal I get this:
vlc Weird\ Al\ Yankovic\ -\ White\ And\ Nerdy\ \[HQ\].avi
VLC media player 1.1.5 The Luggage (revision exported)
Blocked: call to unsetenv("DBUS_ACTIVATION_ADDRESS")
Blocked: call to unsetenv("DBUS_ACTIVATION_BUS_TYPE")
Blocked: call to setlocale(6, "")
Blocked: call to sigaction(17, 0x7fec095fbac0, 0x7fec095fbb60)
Blocked: call to setlocale(6, "")
frame skip 8
frame skip 8
When I tell vlc to use alsa. (Which wasn't necessary before.) I get this as a "pop-up error":
Potential ALSA version problem:
VLC failed to initialize your sound output device (if any).
Please update alsa-lib to version 1.0.23-2-g8d80d5f or higher to try to fix this issue.
But pacman tells me:
pacman -Qs alsa-lib
local/alsa-lib 1.0.23-2
An alternative implementation of Linux sound support
local/lib32-alsa-lib 1.0.23-4
An alternative implementation of Linux sound support (32 bit)
EDIT: I just tried KDE. (I have it installed but don't use it.) Everything is weird. I can't press anything right with the mouse because everything is mirror inverted and some zones are covered with a weird mix of pixels...fact is KDE doesn't work fine on my computer right now.
Last edited by Maximalminimalist (2010-12-12 04:01:53)

Problem with sound in Amarok

Hi guys, I'm having problem with Amarok 2. When I start it the notification says "The audio playback device HDA Intel (CONEXANT Analog) does not work. Falling back to HDA NVidia, HDMI 0 (HDMI Audio Output) ". Amarok starts normally but can't open any song. The same thing happens when I try to open a video with Dragon player(but I don't use so this doesn't matter).
VLC, MPlayer and Youtube have no problem playing videos or sounds, so I could play music with VLC but you'd agree it's not really a music player.
(I tried installing Clementine, but it has it's own problems, saying "Your GStreamer installation is missing a plug-in." (but I don't like Clementine anyway))
Another thing that bothers me is that the hardware sound buttons don't affect these programs.
Could someone help me with this?

there are these errors (several times):
amarok: [CollectionDB] [ERROR!] [virtual QStringList SqliteConnection::query(const QString&)] sqlite3_compile error:
amarok: [CollectionDB] [ERROR!] no such table: directories
amarok: [CollectionDB] [ERROR!] on query: SELECT dir, changedate FROM directories;
amarok: [void CollectionDB::createTables(DbConnection*)]
amarok: [CollectionDB] [ERROR!] [virtual QStringList SqliteConnection::query(const QString&)] sqlite3_compile error:
amarok: [CollectionDB] [ERROR!] table amazon already exists
amarok: [CollectionDB] [ERROR!] on query: CREATE TABLE amazon ( asin VARCHAR(20), locale VARCHAR(2), filename VARCHAR(33), refetchdate INTEGER );
amarok: [CollectionDB] [ERROR!] [virtual int SqliteConnection::insert(const QString&, const QString&)] sqlite3_compile error:
amarok: [CollectionDB] [ERROR!] no such table: directories
amarok: [CollectionDB] [ERROR!] on insert: INSERT INTO directories SELECT * FROM directories_temp;
QDom: saving invalid character , the document will not be well-formed
and when I try to create a playlist it looks normal:
amarok: [PlaylistBrowser] Creating playlist
amarok: [PlaylistBrowser] Saving Playlist to: /home/matej/.kde/share/apps/amarok/playlists/Untitled.m3u
amarok: Loading playlist
amarok: [ThreadWeaver] Job completed: PlaylistReader. Jobs pending: 0
and also load one:
amarok: Loading playlist
amarok: [ThreadWeaver] Job completed: PlaylistReader. Jobs pending: 0
There are also some errors of GUI (QT, KDE), but these are I think a little bit off-topic.

Mplyer sound doesn't work, instead amarok's sond work !!!

hi everyone, the problem is that mplayer's sound doesn't work.
the sound with Amarok and Firefox(e.g youtube) works, but with mpalyer no !!!
i've posted some commands:
thanks
/etc/mplayer/
codecs.conf example.conf input.conf
/etc/mplayer/example.conf
# MPlayer configuration file
# Configuration files are read system-wide from /usr/local/etc/mplayer.conf
# and per-user from ~/.mplayer/config, where per-user settings override
# system-wide settings, all of which are overrriden by the command line.
# The configuration file settings are the same as the command line
# options without the preceding '-'.
# See the CONFIGURATION FILES section in the man page
# for a detailed description of the syntax.
# Profiles should be placed at the bottom of the configuration file to ensure
# that settings wanted as defaults are not restricted to specific profiles.
# video settings #
# Specify default video driver (see -vo help for a list).
#vo=xv
# Use SDL video with the aalib subdriver by default.
#vo = sdl:aalib
# FBdev driver:
# mode to use (read from fb.modes)
#fbmode = 640x480-120
# location of the fb.modes file
#fbmodeconfig = /etc/fb.modes
# Specify your monitor timings for the vesa and fbdev video output drivers.
# See /etc/X11/XF86Config for timings. Be careful; if you specify settings
# that exceed the capabilities of your monitor, you may damage it.
# horizontal frequency range (k stands for 1000)
#monitor-hfreq = 31.5k-50k,70k
# vertical frequency range
#monitor-vfreq = 50-90
# dotclock (or pixelclock) range (m stands for 1000000)
#monitor-dotclock = 30M-300M
# Start in fullscreen mode by default.
#fs=yes
# Change to a different videomode when going fullscreen.
#vm=yes
# Override the autodetected color depth, may need 'vm=yes' as well.
#bpp=0
# Enable software scaling (powerful CPU needed) for video output
# drivers that do not support hardware scaling.
#zoom=yes
# standard monitor size, with square pixels
#monitoraspect=4:3
# Use this for a widescreen monitor, non-square pixels.
#monitoraspect=16:9
# Keep the player window on top of all other windows.
#ontop=yes
# audio settings #
# Specify default audio driver (see -ao help for a list).
#ao=oss
# Use SDL audio driver with the esd subdriver by default.
#ao = sdl:esd
# Specify the mixer device.
#mixer = /dev/mixer
# Resample the sound to 44100Hz with the lavcresample audio filter.
#af=lavcresample=44100
# Output audio to S/PDIF
#ao=alsa:device=spdif
#ac=hwac3,hwdts,hwmpa,
# other settings #
# Pretend to be Window Media Player.
# Fixes playback when playlist and media file use the same URL.
#user-agent=NSPlayer/4.1.0.3856
# Pretend to be Quicktime
# Fixes playback for apple.com/trailers redirects
#user-agent="QuickTime"
# Pretend to be Realplayer SP
# Fixes playback for some video streaming sites
#user-agent=RMA/1.0
# Pretend to have lots of bandwidth
# Speeds up realmedia rtsp:// streams
#bandwidth=2000000
# Pretend to be Winamp
# Fixes playback of some NSV streams
#user-agent="Winamp NSV Player/5.12 (ultravox/2.0)"
# Drop frames to preserve audio/video sync.
#framedrop = yes
# Specify your preferred skin here (skins are searched for in
# /usr/local/share/mplayer/skins/<name> and ~/.mplayer/skins/<name>).
#skin = Abyss
# Resample the font alphamap.
# 0 plain white fonts
# 0.75 very narrow black outline (default)
# 1 narrow black outline
# 10 bold black outline
#ffactor = 0.75
# cache settings
# Use 8MB input cache by default.
#cache = 8192
# Prefill 20% of the cache before starting playback.
#cache-min = 20.0
# Prefill 50% of the cache before restarting playback after the cache emptied.
#cache-seek-min = 50
# DVD: Display English subtitles if available.
#slang = en
# DVD: Play English audio tracks if available.
#alang = en
# Profiles #
# The options declared as part of profiles override global default settings,
# but only take effect when the profile is active.
[protocol.dvdnav]
#vc=ffmpeg12,
#mouse-movements=yes
#nocache=yes
#[vo.vdpau]
#vc=ffmpeg12vdpau,ffwmv3vdpau,ffvc1vdpau,ffh264vdpau,ffodivxvdpau,
# Most video filters do not work with vdpau.
#vf-clr=yes
# OSD progress bar vertical alignment
#progbar-align=50
# You can also include other configuration files.
#include = /path/to/the/file/you/want/to/include
~/.mplayer/config
# Write your default config options here!
# [default] viene applicata a ogni file
[default]
# usa il server grafico X come video output
vo=xv
# usa alsa per l'audio output
ao=alsa
# ao=oss # Use OSS4
# decodifica multithreaded H264/MPEG-1/2 (valido: 1-8)
lavdopts=threads=2
# impostare il canale audio preferito (in questo esempio il numero sei)
channels = 6
# scala i sottotitoli al 3% della dimensione dello schermo
subfont-text-scale = 3
# non usa fontconfig
nofontconfig = 1
# aggiunge bordi neri ai filmati che non hanno lo stesso aspetto di forma dello schermo
# per wide screen monitors
vf-add=expand=::::1:16/9:16
# per non wide screen traditional monitors
#vf-add=expand=::::1:4/3:16
#profilo di up-mixing da due canali audio a sei canali
# use -profile 2chto6ch to activate
[2chto6ch]
af-add=pan=6:1:0:.4:0:.6:2:0:1:0:.4:.6:2
#profilo di down-mixing da sei canali audio a due canali
# use -profile 6chto2ch to activate
[6chto2ch]
af-add=pan=2:0.7:0:0:0.7:0.5:0:0:0.5:0.6:0.6:0:0
# Disabilita screensaver
heartbeat-cmd="xscreensaver-command -deactivate &" # stop xscreensaver
stop-xscreensaver="yes" # stop gnome-screensaver
ls /etc/modprobe.d/
alsa-base.conf snd-hda-intel.conf
etc/modprobe.d/alsa-base.conf
options snd slots=snd_mia,snd_hda_intel
options snd_mia index=1
options snd_hda_intel index=0
/etc/modprobe.d/snd-hda-intel.conf
options snd-hda-intel model=auto
cat /proc/asound/modules
0 snd_hda_intel
lsmod |grep snd
snd_hda_codec_hdmi 30162 1
snd_hda_codec_conexant 36759 1
snd_hda_intel 37000 2
snd_hda_codec 150305 3 snd_hda_codec_hdmi,snd_hda_codec_conexant,snd_hda_intel
snd_hwdep 6340 1 snd_hda_codec
snd_pcm 77645 3 snd_hda_codec_hdmi,snd_hda_codec,snd_hda_intel
snd_page_alloc 7210 2 snd_pcm,snd_hda_intel
snd_timer 18726 1 snd_pcm
snd 59109 11 snd_hwdep,snd_timer,snd_hda_codec_hdmi,snd_hda_codec_conexant,snd_pcm,snd_hda_codec,snd_hda_intel
soundcore 5450 1 snd

Neburski wrote:
ryanjacobs wrote:So, apparently, your system cannot find the ALSA output driver. I suggest that you use a different driver than ALSA.
OP should first go through the wiki page of ALSA to ensure that his ALSA setup is working correctly. There aren't many sound setups that do not work under ALSA. If OP is relying on another sound setup then he should follow the appropriate wiki and tell us so.
@otto88: It would also help if you would try to explain what your intentions were. I'm assuming you merely want to have sound working and not configure some fancy setup. In that case you should just stick to a basic setup and follow the wiki step by step (don't skip any line and if you don't understand something, you figure that out or you ask people, don't just ignore it).
For instance I don't even have those /etc/modprobe.d/alsa-base.conf and /etc/modprobe.d/snd-hda-intel.conf even though I have 2 sound devices, while you seem to only have 1 sound device
otto88 wrote:cat /proc/asound/modules
0 snd_hda_intel
So your /etc/modprobe.d/alsa-base.conf in that case is totally unnecessary, let alone that you don't even have the snd_mia module loaded so why do you even configure alsa to load it at index 1. You aren't supposed to simply copy and paste the configurations from the wiki. Those configurations are merely there to give you an idea on how to make your own configuration.
When you have ensured that your sound is been set up correctly then you can start with the most basic mplayer configuration. You keep the /etc/mplayer/example.cong configuration file totally commented out (like you have atm) and your ~/.mplayer/config should only contain
ao=alsa
ok i just followed the wiki and your advices and doesn't work, yet
mplayer -ao alsa /dati/Musica/StudyDue.mp3
MPlayer SVN-r36498-snapshot-4.8.2 (C) 2000-2013 MPlayer Team
206 audio & 433 video codecs
mplayer: could not connect to socket
mplayer: No such file or directory
Failed to open LIRC support. You will not be able to use your remote control.
Playing /dati/Musica/StudyDue.mp3.
libavformat version 55.19.104 (internal)
Audio only file format detected.
Load subtitles in /dati/Musica/
==========================================================================
Opening audio decoder: [mpg123] MPEG 1.0/2.0/2.5 layers I, II, III
AUDIO: 22050 Hz, 2 ch, s16le, 32.0 kbit/4.54% (ratio: 4000->88200)
Selected audio codec: [mpg123] afm: mpg123 (MPEG 1.0/2.0/2.5 layers I, II, III)
==========================================================================
[AO_ALSA] Playback open error: No such file or directory
Failed to initialize audio driver 'alsa'
Could not open/initialize audio device -> no sound.
Audio: no sound
Video: no video
Exiting... (End of file)

No sound from Amarok in Gnome

Just installed Arch, from having used Ubuntu
Sound seems to work with everything else - totem and the mplayer command get music playing just fine.
But in Amarok, while every other aspect of it seems to behave fine, it doesn't make any sound. Alsa is selected as the output plugin. I get this on opening Amarok from terminal:
$ amarok
Xlib: extension "Generic Event Extension" missing on display ":0.0".
Xlib: extension "Generic Event Extension" missing on display ":0.0".
Xlib: extension "Generic Event Extension" missing on display ":0.0".
Amarok: [Loader] Starting amarokapp..
Amarok: [Loader] Don't run gdb, valgrind, etc. against this binary! Use amarokapp.
Xlib: extension "Generic Event Extension" missing on display ":0.0".
Xlib: extension "Generic Event Extension" missing on display ":0.0".
Xlib: extension "Generic Event Extension" missing on display ":0.0".
Xlib: extension "Generic Event Extension" missing on display ":0.0".
Xlib: extension "Generic Event Extension" missing on display ":0.0".
Xlib: extension "Generic Event Extension" missing on display ":0.0".
kbuildsycoca running...
kdecore (KAction): WARNING: KAction::insertKAccel( kaccel = 0x91f5f40 ): KAccel object already contains an action name "play_pause"
QLayout "unnamed" added to QVBox "unnamed", which already has a layout
kdecore (KAction): WARNING: KAction::insertKAccel( kaccel = 0x91f5f40 ): KAccel object already contains an action name "play_pause"
QLayout: Adding KToolBar/mainToolBar (child of QVBox/unnamed) to layout for PlaylistWindow/PlaylistWindow
QObject::connect: Incompatible sender/receiver arguments
StarManager::ratingsColorsChanged() --> ContextBrowser::ratingOrScoreOrLabelsChanged(const QString&)
$ Xlib: extension "Generic Event Extension" missing on display ":0.0".
Xlib: extension "Generic Event Extension" missing on display ":0.0".
QColor::setRgb: RGB parameter(s) out of range
QWidget::setMinimumSize: The smallest allowed size is (0,0)
QComboBox::setCurrentItem: (speakerComboBox) Index 1 out of range
I figure that something's going wrong there, and I'm guessing the issue is to do with Amarok being run through Gnome and not KDE. However, that's where my ideas end. As I enjoyed using Amarok in Ubuntu, any help would be much appreciated.

Kde 4.7.4 and libltdl:
core/libltdl 2.4.2-2 [installed]
A system independent dlopen wrapper for GNU libtool
multilib/lib32-libltdl 2.4.2-2 [installed]
A system independent dlopen wrapper for GNU libtool (32-bit)

[SOLVED] No sound in Clementine nor Amarok

I'm using KDE with GStreamer as phonon backend.
This morning I tried to play some mp3 files on Amarok and they didn't work. I installed Clementine and I had the same results.
I can play videos and audio files on mplayer, so there is no problem with my soundcard.
On clementine, it is said that "the installation of GStreamer is missing one complement".
I installed all it is said in the wiki, but apparently I'm missing something.
Any ideas?
Last edited by doblerone (2014-05-28 19:16:32)

You need to install the gstreamer0.10-* packages. But yeah for amarok my advice would be to switch to phonon vlc, as phonon gstreamer tends to crash on track change.

Can't play sound from multiple programs

So I have this new shiny box running Arch64 with a ASRock N68-S motherboard. I installed alsa and I am able to play music just fine using amarok or mpd. However, when I try playing two sounds at once, there is a short pause and the first program becomes silent, and then the second kicks in.
According to the wiki, dmix should work automatically with this version of alsa. I tried seting up dmix "the old way" using /etc/asound.conf (hey, you never know) but no luck either.
Lspci lists my card as
00:05.0 Audio device: nVidia Corporation MCP61 High Definition Audio (rev a2)
I figured this was a bit vague and went to the manufacturer's site and found out the actual name is
5.1 CH Windows® Vista™ Premium Level HD Audio (Realtek ALC662 / VIA® VT1708S / VT1705 Audio Codec)
The Alsa wiki suggests to test dmix using these instructions but this simple test fails miserably, alsaplayer windows show up and complain "no stream found". So it seems and one of the unlucky few for which dmix doesn't just work.
However, OSS4 works with hardly no setup. But I just don't want to use it since most applications aren't OSS-aware, there is no KDE integration and so on.
What sould I do? I'd happy to report a bug upstream if necessary but I realised audio in Linux is way more complicated than previously thought and I'd rather make sure I haven't over looked something. I'm so confused I can't tell what my actual soundcard vendor is: nVidia, Realtek or Via?
edit: oh and of course, nothing of interest in /var/log/everything.log
Thanks!
Last edited by eldalion (2010-02-21 17:28:53)

eldalion wrote:I tried seting up dmix "the old way" using /etc/asound.conf (hey, you never know) but no luck either.
Not necessary. ALSA supports dmix by default. Don't edit that file.
Let me explain what's happening. mpd has a stupid default config, of using OSS rather than ALSA. And OSS screws up dmix.
For example, my ~/.mpdconf is:
music_directory "~/gnutella/downloads/Cool"
playlist_directory "~/.mpd/playlists"
db_file "~/.mpd/mpd.db"
log_file "~/.mpd/mpd.log"
error_file "~/.mpd/mpd.error"
audio_output {
type "alsa"
name "Sound Card"
options "dev=dmixer"
device "plug:dmix"
Try that, to prove to yourself that I'm right Also use this good debugging command, from the same ALSA wiki entry (because I put it there):
fuser -v /dev/snd/* /dev/dsp*
eldalion wrote:What should I do?
Step 1: File a bug with your distro, for the app (mpd) to default to ALSA instead of OSS. Defaulting to OSS these days is ridiculous - ALSA is the standard sound system in Linux, and that's not gonna change.
Please do this. I'm not doing this, because I don't use Arch or mpd any more.
Step 2: File a bug upstream, to default to ALSA instead of OSS.
Edit: I notice that Ubuntu has the right idea:
* Update mpd.conf from upstream
- Configure an ALSA output by default
Last edited by brebs (2010-02-23 07:56:16)

The Sorry State of Sound In Linux

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READ: http://insanecoding.blogspot.com/2007/0 … linux.html
I've only discovered this because of an ALSA bug was making my life miserable. That article was written about a month or two before OSS4 (Open Sound System v4) was released under GPL and CDDL. This past January, it was released under BSD as well. Unfortunately for the developer, he does not understand open source very well. He is now reporting that his revenue went significantly down..
OSS4 does not equal to the crap OSS in Linux 2.4 kernels. Even OSS3 != OSS3 in former linux kernels. He was neglecting the open source version in favour of the commercial one. Instead of improving it, someone forked it. And, the fork became popular. Unfortunately, the original author had no interest in working on the fork. So, he only focused on the commercial version. Now, many years later, the commercial version is fully open source. There is no more commercial version.
I've tried it, and I really like it. Music sounds much better than on ALSA. For example, screaming in alternative rock is more legible. It's like enabling "enhance voice" in a phone. It supports higher PCM values without noise.
The mixer (vmix) is capable of 18 channels. It can also do per application volume control (like PulseAudio).
It totally makes sense that OSS gets back into the kernel because it works on almost every UNIX and UNIX-like system (except OS X). ALSA only works on Linux, and according to that article, developers still prefer the OSS API, even on ALSA. However, the ALSA OSS API is lacking according to an ALSA developer. I can confirm that the comment on the ALSA OSS Emulation API working better than the ALSA API. XMMS with ALSA enabled freezes my system. XMMS with OSS enabled on ALSA does not.
It worked without any configuration besides muting some channels to kill the noise.
THE BAD
ossmix and ossxmix are totally unusable because they do not name the channels properly. ossxmix uses 100% CPU if Compiz is enabled. Version 4.1 will fix this.
I had to figure out WTF ossxmix.codec1.connector.jack14.jack 54:54 means. Sane names need to be added to the jacks such as "mater, front, input, auxiliary, microphone, etc.."
It's best to use ossxmix (GTK+ mixer) and playing with all the jacks to figure out what each one does.
One interesting control is how OSS4 should behave (Fast, Medium High, Professional, etc.). It's probably a latency control. ossxmix is a demo app on how you can control the mixer from GTK. It needs to be made usable.
If you use Media Player Daemon, VLC, and MPlayer, OSS4 works beautifully. They use OSS directly. Others have problems.
Totem does not play sound. Totem-Xine uses 100% CPU (Xine-UI works with low CPU usage).
In terminal I noticed some output from oss mixer control saying it received bad arguments. OSS4 is backwards compatible with OSS3, but xine-lib may be using the API badly.
The progress bar (seeker) in GStreamer based applications (Rhythmbox, Banshee) does not work if you use vmix, or it may work, but it will be unsure of the length of a song. You will see it constantly adjusting a few seconds up or down. By the middle of the song, the length reporting generally stabilises. You have to enable softoss (the old mixer). vmix and Gstreamer behave very badly. The Gstreamer developer responsible for OSS claims that it does on his system. Though, he is most likely using the trunk version.
You have to apply this patched gstreamer to make volume control in GNOME work.
While the progress bar problem is fixed with softoss, volume control does not work, even with the patched GStreamer. It only works with vmix. System > Preferences > Sound lists nothing under "Default Mixer Tracks".
Sound does not work in KDE4 at all. While Phonon is based on Xine, which has OSS support, it queries HAL, which does not support OSS4 yet. Therefore, it thinks that there is no sound card.
As for KDE 3.5.9. Amarok works, Noatun works, and anything based on MPlayer should work. KMIX does not work. Same in KDE4.
DOWNLOAD IT (Popular Distributions Linux and Unix Distributions)
Gentoo ebuild
Arch Linux
The Arch Linux wiki has a good article. Read it.
A Ubuntu user has published an install guide as well.
Check Configuring Applications for OSSv4 after you install it.
Overall, I think it's a million times better than ALSA. It's cross platform, and it's stable. It was released a year ago on 15th of March 2007. Since he does not have a marketing department, no one has heard of it.
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Last edited by SpookyET (2008-03-29 15:01:19)

wyvern wrote:
Hmm, so it's a choice - have to enable effects and use an older mixer, or put up with a 'stuck' progress bar... Well, as I said, I use MOC, and I'm happy with my mixer setup so far, so I'll stick to what I've got.
I can't believe I finally have my headphones working, or external speakers, should I so wish - and what with the ATI divers *finally* providing tear-free video, it's like it's my birthday
Since you mentioned ATI and went offtopic, can you pastebin your ATI xorg? It works fine for me, but If I enable compiz, i get flickering video and slow scrolling. So, I'm stuck with the open source driver, which uses a lot more CPU when I play video.
I can also confirm what the article said about the ALSA OSS emulation. It does work better than ALSA with the ALSA API. xmms with ALSA enabled freezes my sytem. xmms with OSS enabled running ALSA does not. I'm tired of this bollocks. We have to mount a campaign to get OSS back into the kernel.
Anyway, this is my OSS output.
OSSINFO
Version info: OSS 4.0 (b1014/200803130443) (0x00040003) GPL
Platform: Linux/i686 2.6.24-ARCH #1 SMP PREEMPT Wed Mar 5 12:07:52 UTC 2008 (mercurius)
Number of audio devices: 10
Number of audio engines: 13
Number of mixer devices: 2
Device objects
0: osscore0 OSS core services
1: hdaudio0 Intel HD Audio interrupts=1389433 (1526696)
HD Audio controller Intel HD Audio
Vendor ID 0x80862668
Subvendor ID 0x10250070
Codec 0: ALC880 (0x10ec0880/0x08800000)
Codec 1: Unknown (0x11c13026)
2: softoss0 OSS Virtual Mixer v3.0
Mixer devices
0: High Definition Audio ALC880 (Mixer 0 of device object 1)
Device file /dev/oss/hdaudio0/mix0, Legacy device /dev/mixer0
Priority: 10
Caps:
Device handle: PCI00701025-0000:00:1b.0-mx01
Device priority: 10
1: Virtual Mixer (Mixer 0 of device object 2)
Device file /dev/oss/softoss0/mix0, Legacy device /dev/mixer1
Priority: 1
Caps: VIRTUAL
Device handle: softoss0-mx01
Device priority: 1
Audio devices
HD Audio front /dev/oss/hdaudio0/pcm0 (device index 0)
Legacy device /dev/dsp0
Caps: TRIGGER MMAP
Modes: OUTPUT
Out engine 1: 0/HD Audio front
Available for use
Input formats (0x00001010):
AFMT_S16_LE - 16 bit signed little endian
AFMT_S32_LE - 32 bit signed little endian
Output formats (0x00001010):
AFMT_S16_LE - 16 bit signed little endian
AFMT_S32_LE - 32 bit signed little endian
Device handle: PCI00701025-0000:00:1b.0-au01
Related mixer dev: 0
Sample rate source: 0
Preferred channel configuration: Not indicated
Supported number of channels (min - max): 2 - 8
Native sample rates (min - max): 44100 - 192000 (44100,48000,96000,192000)
HW Type: Not indicated.
Minimum latency: Not indicated
HD Audio rear /dev/oss/hdaudio0/pcm1 (device index 1)
Legacy device /dev/dsp1
Caps: TRIGGER MMAP
Modes: OUTPUT
Out engine 1: 1/HD Audio rear
Available for use
Input formats (0x00001010):
AFMT_S16_LE - 16 bit signed little endian
AFMT_S32_LE - 32 bit signed little endian
Output formats (0x00001010):
AFMT_S16_LE - 16 bit signed little endian
AFMT_S32_LE - 32 bit signed little endian
Device handle: PCI00701025-0000:00:1b.0-au02
Related mixer dev: 0
Sample rate source: 0
Preferred channel configuration: Not indicated
Supported number of channels (min - max): 2 - 2
Native sample rates (min - max): 44100 - 192000 (44100,48000,96000,192000)
HW Type: Not indicated.
Minimum latency: Not indicated
HD Audio center/LFE /dev/oss/hdaudio0/pcm2 (device index 2)
Legacy device /dev/dsp2
Caps: TRIGGER MMAP
Modes: OUTPUT
Out engine 1: 2/HD Audio center/LFE
Available for use
Input formats (0x00001010):
AFMT_S16_LE - 16 bit signed little endian
AFMT_S32_LE - 32 bit signed little endian
Output formats (0x00001010):
AFMT_S16_LE - 16 bit signed little endian
AFMT_S32_LE - 32 bit signed little endian
Device handle: PCI00701025-0000:00:1b.0-au03
Related mixer dev: 0
Sample rate source: 0
Preferred channel configuration: Not indicated
Supported number of channels (min - max): 2 - 2
Native sample rates (min - max): 44100 - 96000 (44100,48000,96000)
HW Type: Not indicated.
Minimum latency: Not indicated
HD Audio side /dev/oss/hdaudio0/pcm3 (device index 3)
Legacy device /dev/dsp3
Caps: TRIGGER MMAP
Modes: OUTPUT
Out engine 1: 3/HD Audio side
Available for use
Input formats (0x00001010):
AFMT_S16_LE - 16 bit signed little endian
AFMT_S32_LE - 32 bit signed little endian
Output formats (0x00001010):
AFMT_S16_LE - 16 bit signed little endian
AFMT_S32_LE - 32 bit signed little endian
Device handle: PCI00701025-0000:00:1b.0-au04
Related mixer dev: 0
Sample rate source: 0
Preferred channel configuration: Not indicated
Supported number of channels (min - max): 2 - 2
Native sample rates (min - max): 44100 - 96000 (44100,48000,96000)
HW Type: Not indicated.
Minimum latency: Not indicated
HD Audio spdif-out /dev/oss/hdaudio0/spdout0 (device index 4)
Legacy device /dev/dsp4
Caps: TRIGGER MMAP
Modes: OUTPUT
Out engine 1: 4/HD Audio spdif-out
Available for use
Input formats (0x00001410):
AFMT_S16_LE - 16 bit signed little endian
AFMT_AC3 - AC3 (Dolby Digital) encoded audio
AFMT_S32_LE - 32 bit signed little endian
Output formats (0x00001410):
AFMT_S16_LE - 16 bit signed little endian
AFMT_AC3 - AC3 (Dolby Digital) encoded audio
AFMT_S32_LE - 32 bit signed little endian
Device handle: PCI00701025-0000:00:1b.0-au05
Related mixer dev: 0
Sample rate source: 0
Preferred channel configuration: Not indicated
Supported number of channels (min - max): 2 - 2
Native sample rates (min - max): 44100 - 96000 (44100,48000,96000)
HW Type: Not indicated.
Minimum latency: Not indicated
High Definition Audio rec1 /dev/oss/hdaudio0/pcmin0 (device index 5)
Legacy device /dev/dsp5
Caps: TRIGGER MMAP
Modes: INPUT
In engine 1: 5/High Definition Audio rec1
Available for use
Input formats (0x00001010):
AFMT_S16_LE - 16 bit signed little endian
AFMT_S32_LE - 32 bit signed little endian
Output formats (0x00001010):
AFMT_S16_LE - 16 bit signed little endian
AFMT_S32_LE - 32 bit signed little endian
Device handle: PCI00701025-0000:00:1b.0-au06
Related mixer dev: 0
Sample rate source: 0
Preferred channel configuration: Not indicated
Supported number of channels (min - max): 2 - 2
Native sample rates (min - max): 44100 - 96000 (44100,48000,96000)
HW Type: Not indicated.
Minimum latency: Not indicated
High Definition Audio rec2 /dev/oss/hdaudio0/pcmin1 (device index 6)
Legacy device /dev/dsp6
Caps: TRIGGER MMAP
Modes: INPUT
In engine 1: 6/High Definition Audio rec2
Available for use
Input formats (0x00001010):
AFMT_S16_LE - 16 bit signed little endian
AFMT_S32_LE - 32 bit signed little endian
Output formats (0x00001010):
AFMT_S16_LE - 16 bit signed little endian
AFMT_S32_LE - 32 bit signed little endian
Device handle: PCI00701025-0000:00:1b.0-au07
Related mixer dev: 0
Sample rate source: 0
Preferred channel configuration: Not indicated
Supported number of channels (min - max): 2 - 2
Native sample rates (min - max): 44100 - 96000 (44100,48000,96000)
HW Type: Not indicated.
Minimum latency: Not indicated
High Definition Audio rec3 /dev/oss/hdaudio0/pcmin2 (device index 7)
Legacy device /dev/dsp7
Caps: TRIGGER MMAP
Modes: INPUT
In engine 1: 7/High Definition Audio rec3
Available for use
Input formats (0x00001010):
AFMT_S16_LE - 16 bit signed little endian
AFMT_S32_LE - 32 bit signed little endian
Output formats (0x00001010):
AFMT_S16_LE - 16 bit signed little endian
AFMT_S32_LE - 32 bit signed little endian
Device handle: PCI00701025-0000:00:1b.0-au08
Related mixer dev: 0
Sample rate source: 0
Preferred channel configuration: Not indicated
Supported number of channels (min - max): 2 - 2
Native sample rates (min - max): 44100 - 96000 (44100,48000,96000)
HW Type: Not indicated.
Minimum latency: Not indicated
High Definition Audio spdif-in /dev/oss/hdaudio0/spdin0 (device index 8)
Legacy device /dev/dsp8
Caps: TRIGGER MMAP
Modes: INPUT
In engine 1: 8/High Definition Audio spdif-in
Available for use
Input formats (0x00001410):
AFMT_S16_LE - 16 bit signed little endian
AFMT_AC3 - AC3 (Dolby Digital) encoded audio
AFMT_S32_LE - 32 bit signed little endian
Output formats (0x00001410):
AFMT_S16_LE - 16 bit signed little endian
AFMT_AC3 - AC3 (Dolby Digital) encoded audio
AFMT_S32_LE - 32 bit signed little endian
Device handle: PCI00701025-0000:00:1b.0-au09
Related mixer dev: 0
Sample rate source: 0
Preferred channel configuration: Not indicated
Supported number of channels (min - max): 2 - 2
Native sample rates (min - max): 44100 - 96000 (44100,48000,96000)
HW Type: Not indicated.
Minimum latency: Not indicated
OSS Virtual Mixer v3.0 Playback /dev/oss/softoss0/pcm0 (device index 9)
Legacy device /dev/dsp9
Caps: TRIGGER MMAP VIRTUAL
Modes: OUTPUT
Out engine 1: 9/OSS Virtual Mixer v3.0 Playback
Available for use
Out engine 2: 10/OSS Virtual Mixer v3.0 Playback
Available for use
Out engine 3: 11/OSS Virtual Mixer v3.0 Playback
Available for use
Out engine 4: 12/OSS Virtual Mixer v3.0 Playback
Available for use
Input formats (0x00000010):
AFMT_S16_LE - 16 bit signed little endian
Output formats (0x00000010):
AFMT_S16_LE - 16 bit signed little endian
Device handle: softoss0-au01
Related mixer dev: 1
Sample rate source: 0
Preferred channel configuration: Not indicated
Supported number of channels (min - max): 1 - 2
Native sample rates (min - max): 48000 - 48000
HW Type: Not indicated.
Minimum latency: Not indicated
OSSMIX -d0
Selected mixer 0/High Definition Audio ALC880
Known controls are:
codec1.connector.jack14.mode <jack|input> (currently jack)
codec1.connector.jack14.mute ON|OFF (currently OFF)
codec1.connector.jack14.front-m ON|OFF (currently OFF)
codec1.connector.jack14.inputmi ON|OFF (currently ON)
codec1.connector.jack14.jack <both/leftvol>[:<rightvol>] (currently 57.9:57.9 dB)
codec1.connector.jack15.mode <jack|input> (currently jack)
codec1.connector.jack15.mute ON|OFF (currently OFF)
codec1.connector.jack15.rear-mu ON|OFF (currently OFF)
codec1.connector.jack15.inputmi ON|OFF (currently OFF)
codec1.connector.jack15.jack <both/leftvol>[:<rightvol>] (currently 57.9:57.9 dB)
codec1.connector.jack16.mode <jack|input> (currently jack)
codec1.connector.jack16.mute ON|OFF (currently OFF)
codec1.connector.jack16.center/ ON|OFF (currently OFF)
codec1.connector.jack16.inputmi ON|OFF (currently OFF)
codec1.connector.jack16.jack <both/leftvol>[:<rightvol>] (currently 57.9:57.9 dB)
codec1.connector.jack17.mode <jack|input> (currently jack)
codec1.connector.jack17.mute ON|OFF (currently OFF)
codec1.connector.jack17.side-mu ON|OFF (currently OFF)
codec1.connector.jack17.inputmi ON|OFF (currently OFF)
codec1.connector.jack17.jack <both/leftvol>[:<rightvol>] (currently 57.9:57.9 dB)
codec1.connector.jack18.mode <jack|input> (currently jack)
codec1.connector.jack18.mute ON|OFF (currently OFF)
codec1.connector.jack18.jack <jack|jack|jack|jack> (currently jack)
codec1.connector.jack19.mode <jack|input> (currently jack)
codec1.connector.jack19.mute ON|OFF (currently OFF)
codec1.connector.jack19.jack <jack|jack|jack|jack> (currently jack)
codec1.connector.jack1a.mode <jack|input> (currently jack)
codec1.connector.jack1a.mute ON|OFF (currently OFF)
codec1.connector.jack1a.jack <jack|jack|jack|jack> (currently jack)
codec1.connector.jack1b.mode <jack|input> (currently jack)
codec1.connector.jack1b.mute ON|OFF (currently OFF)
codec1.connector.jack1b.jack <jack|jack|jack|jack> (currently jack)
codec1.record.rec1 <both/leftvol>[:<rightvol>] (currently 31.9:31.9 dB)
codec1.record.rec1.rec1 <jack|jack|jack|jack|jack|jack|jack> (currently jack)
codec1.record.rec2 <both/leftvol>[:<rightvol>] (currently 31.9:31.9 dB)
codec1.record.rec2.rec2 <jack|jack|jack|jack|jack|jack|jack> (currently jack)
codec1.record.rec3 <both/leftvol>[:<rightvol>] (currently 31.9:31.9 dB)
codec1.record.rec3.rec3 <jack|jack|jack|jack|jack|inputmix|jack|jack|jack|jack> (currently jack)
codec1.misc.jack1 <both/leftvol>[:<rightvol>] (currently 58.9:58.9 dB)
codec1.misc.jack2 <both/leftvol>[:<rightvol>] (currently 58.9:58.9 dB)
codec1.misc.jack3 <both/leftvol>[:<rightvol>] (currently 58.9:58.9 dB)
codec1.misc.jack4 <both/leftvol>[:<rightvol>] (currently 58.9:58.9 dB)
codec1.misc.jack5 <both/leftvol>[:<rightvol>] (currently 58.9:58.9 dB)
codec1.misc.jack6 <both/leftvol>[:<rightvol>] (currently 58.9:58.9 dB)
codec1.misc.jack7 <both/leftvol>[:<rightvol>] (currently 58.9:58.9 dB)
codec1.misc.jack8 <both/leftvol>[:<rightvol>] (currently 58.9:58.9 dB)
codec1.misc.inputmix <jack|jack|jack|jack|jack|jack|jack|jack> (currently jack)
OSSMIX -d1
Selected mixer 1/Virtual Mixer
Known controls are:
synth <both/leftvol>[:<rightvol>] (currently 100:100)
pcm <both/leftvol>[:<rightvol>] (currently 100:100)
2 <leftVU>:<rightVU>] (currently 137:136)
autoreset ON|OFF (currently OFF)
effects.eq.prescale <monovol> (currently 255)
effects.eq.lo <monovol> (currently 128)
effects.eq.mid <monovol> (currently 128)
effects.eq.hi <monovol> (currently 128)
effects.eq.xhi <monovol> (currently 128)
effects.eq.bypass ON|OFF (currently OFF)
voices.pcm9 <both/leftvol>[:<rightvol>] (currently 100:100)
voices1 <leftVU>:<rightVU>] (currently 135:135)
voices.pcm10 <both/leftvol>[:<rightvol>] (currently 100:100)
voices2 <leftVU>:<rightVU>] (currently 118:119)
voices.pcm11 <both/leftvol>[:<rightvol>] (currently 100:100)
voices3 <leftVU>:<rightVU>] (currently 0:0)
voices.pcm12 <both/leftvol>[:<rightvol>] (currently 100:100)
voices4 <leftVU>:<rightVU>] (currently 0:0)
Last edited by SpookyET (2008-03-15 01:17:07)

No Sound or Kicker in KDE

About two days ago I noticed that when I went to play a video using smplayer that I had no sound when I did about 3 hours before. I tried a few different players and nothing worked, i remember seeing a notice in amarok though that ALSA was in use by another program but it wasn't. I opened mplayer from the command line and it said that it couldn't open ALSA and OSS sound devices because they were in use. I tried restarting the ALSA daemon but that didnt do anything so I rebooted and the sound worked again. The same problem kept happening every day at about 2-10 hour intervals.
After one of the reboots to fix ALSA yesterday I noticed that I couldn't see my kicker panel at the bottom, I switched from compiz to kwin and it still wasn't there, back to compiz and still not there, I restarted X and it still wasnt there. I just booted back in to see if it was there and its still invisable, yet it is running since I can see it in the process list.

Ok, i figured out what the problem was. For some reason I had to install Flip4Mac to make the previews work again. Hope this helps everyone else.

(Solved) one sound application at a time

hello,archers! I moved from gnome to Kde(at last) and faced the issue. E.G if amarok plays musik there's no system notifications and smplayer plays no sound t.i only one programm can use alsa at a time. I have latest kde, alsa and gstreamer and xine backends for phonon. alsamixer shows everything's ok. What should I do? thnx.
Last edited by off220 (2011-01-15 10:30:01)

Ok. I've been able to solve the problem by creating /home/user/.asoundrc file with this content
pcm.softvolPhonon {
type softvol
slave.pcm "default:CARD=0"
control {
name "Phonon"
card 0
min_dB -51.0
max_dB 0.0
resolution 100
hint {
show on
description "My Soundcard with extra Volume Control"
Then I just choose Phonon as the mail output in sound preferences. Solved.

Most, but not all of my sound is broken.

It all started when I upgraded to KDEmod 4.3. First, my sound stopped working, but after reading the notes I realized I needed to install a Phonon backend. So I did. I installed kdemod-phonon-backend-xine and for a little while all was well. Then sound just stopped working again. I've tried Amarok, VLC, Kaffeine, Flash videos in Firefox and a couple other various programs with various files. For some strange reason Amarok DOES have sound, but nothing else does, even when playing the same files I can play in Amarok. So it doesn't seem to be a problem with the sound itself, but with certain programs, but I have no idea why. I literally did not change a single package between when it was working and when it broke. I'm not sure whether the upgrade to KDE 4.3 was related, but since it also gave me sound problems until fixing them I thought I'd throw it out there.
I'm at a bit of a loss here. Any ideas?

Sorry to revive an old topic, but I'm having the same issues again, and this time much worse. Before it was kind of sporadic and I could live with it. Now no sound outside of Amarok works, ever. I even took out my sound card and tried to use the onboard sound in case it was something crazy like that, but no results. I notice when I start KDE there's a message telling me the device doesn't work, though it doesn't explain why. When I try to play sound in anything but Amarok, I find this message in everything.log
Nov 22 23:43:23 workstation pulseaudio[3139]: alsa-sink.c: Error opening PCM device front:0: Device or resource busy
Despite all this Amarok STILL works, and if I go into the system settings for KDE and play the test sound, that works. Nothing else does.
I'm at a loss as to why the device would be busy.

Setting ALSA sound card system wide (rather than just in KDE)

Hi,
I've got an HTPC with an ATI Radeon HD 4350 graphics card running Arch x86. It has its own integrated sound card for outputting audio via HDMI, which is detected by ALSA as a separate device. I've managed to set it as the default audio device in KDE 4 (via System Settings). This allows applications such as Amarok and SMPlayer to properly output audio to the HD 4350's integrated sound card, but software such as Flash and MPlayer (from the command line) are still outputting audio to the motherboard's built-in sound card. How can I change this setting on a system-wide level? I've tried running alsaconf as root and adding
pcm.!default {
type hw
card 2
device 3
to ~/.asoundrc; I got the card and device numbers from aplay:
[htpc@exia ~]$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: Intel [HDA Intel], device 0: HDA Generic [HDA Generic]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: HDMI [HDA ATI HDMI], device 3: ATI HDMI [ATI HDMI]
Subdevices: 1/1
Subdevice #0: subdevice #0
Here is the output from MPlayer when I try to play a video, complete with errors regarding audio:
MPlayer UNKNOWN-4.4.0 (C) 2000-2009 MPlayer Team
137 audio & 296 video codecs
mplayer: could not connect to socket
mplayer: No such file or directory
Failed to open LIRC support. You will not be able to use your remote control.
Playing /tmp/FlashmvWXPy.
libavformat file format detected.
[flv @ 0xac7d250]skipping flv packet: type 18, size 294, flags 0
[lavf] Video stream found, -vid 0
[lavf] Audio stream found, -aid 1
VIDEO: [H264] 854x480 0bpp 29.970 fps 0.0 kbps ( 0.0 kbyte/s)
==========================================================================
Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family
Selected video codec: [ffh264] vfm: ffmpeg (FFmpeg H.264)
==========================================================================
==========================================================================
Opening audio decoder: [faad] AAC (MPEG2/4 Advanced Audio Coding)
FAAD: compressed input bitrate missing, assuming 128kbit/s!
AUDIO: 44100 Hz, 2 ch, s16le, 128.0 kbit/9.07% (ratio: 16000->176400)
Selected audio codec: [faad] afm: faad (FAAD AAC (MPEG-2/MPEG-4 Audio))
==========================================================================
[AO OSS] audio_setup: Can't open audio device /dev/dsp: Device or resource busy
[AO_ALSA] alsa-lib: confmisc.c:768:(parse_card) cannot find card '2'
[AO_ALSA] alsa-lib: conf.c:3513:(_snd_config_evaluate) function snd_func_card_driver returned error: No such file or directory
[AO_ALSA] alsa-lib: confmisc.c:392:(snd_func_concat) error evaluating strings
[AO_ALSA] alsa-lib: conf.c:3513:(_snd_config_evaluate) function snd_func_concat returned error: No such file or directory
[AO_ALSA] alsa-lib: confmisc.c:1251:(snd_func_refer) error evaluating name
[AO_ALSA] alsa-lib: conf.c:3513:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory
[AO_ALSA] alsa-lib: conf.c:3985:(snd_config_expand) Evaluate error: No such file or directory
[AO_ALSA] alsa-lib: pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM default
[AO_ALSA] Playback open error: No such file or directory
[JACK] cannot open server
[AO SDL] Samplerate: 44100Hz Channels: Stereo Format s16le
[AO_ALSA] alsa-lib: confmisc.c:768:(parse_card) cannot find card '2'
[AO_ALSA] alsa-lib: conf.c:3513:(_snd_config_evaluate) function snd_func_card_driver returned error: No such file or directory
[AO_ALSA] alsa-lib: confmisc.c:392:(snd_func_concat) error evaluating strings
[AO_ALSA] alsa-lib: conf.c:3513:(_snd_config_evaluate) function snd_func_concat returned error: No such file or directory
[AO_ALSA] alsa-lib: confmisc.c:1251:(snd_func_refer) error evaluating name
[AO_ALSA] alsa-lib: conf.c:3513:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory
[AO_ALSA] alsa-lib: conf.c:3985:(snd_config_expand) Evaluate error: No such file or directory
[AO_ALSA] alsa-lib: pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM default
[AO SDL] Unable to open audio: No available audio device
DVB card number must be between 1 and 4
AO: [null] 44100Hz 2ch s16le (2 bytes per sample)
Starting playback...
VDec: vo config request - 854 x 480 (preferred colorspace: Planar YV12)
VDec: using Planar YV12 as output csp (no 0)
Movie-Aspect is 1.78:1 - prescaling to correct movie aspect.
VO: [xv] 854x480 => 854x480 Planar YV12
Thanks!
Last edited by w1ntermute (2009-06-28 18:40:03)

whoops wrote:Do you use the motherboard sound-card?
No.
whoops wrote:If not - did you try just turning it off in bios (or blacklisting the driver)?
No, I haven't tried that. I'll give it a shot. Thanks for the tip!
Last edited by w1ntermute (2009-07-02 18:11:53)

Sound card issue

I have onboard sound, it's been working perfectly fine in Arch. The other day I took apart my computer to move it into a new case, dropped a screwdriver on the motherboard a couple of times, straightened a pin that I noticed was bent on the front panel audio connector (which I wasn't using), and when I hooked everything back up and turned the system on, I no longer heard sound. I didn't get any error messages from the sound system, and when I play something in, say, Amarok, everything looks normal, except for the fact that no sound is coming out of my speakers. So I suspect I physically damaged the part of the motherboard responsible for playing sound.
lspci still returns the name of the onboard device, though, and dmesg doesn't report anything different from what it normally reports about the device. Is there anything else I might try to see if there's a software solution to this?
I'd like to fix the onboard sound, but I thought that might not be possible, so I put in a cheap PCI sound card. lspci detects this too, but when I plug my speakers into that card's line out I also get no sound, although again everything on the software end seems to be fine. I tried disabling the onboard sound in the bios but the system still doesn't seem to detect that I want to use that card.
So, if anyone has any thoughts on what I'm doing wrong with the PCI sound card or (preferably) a way to try to fix the onboard sound, please post.

Have you run "alsaconf"? It could be your PCI sound-card isn't configured in ALSA yet.
Have you checked whether the line-out is muted? You could for example by "alsamixer" adjust sound levels.
If the on-board sound-card is disabled, there shouldn't be any other card for ALSA to pick up. Otherwise you need to check in /etc/rc.conf which card is picked up as default, in other words which module is put first in line.
Dropping screwdrivers on the motherboard isn't a very good idea. Even static electricity could severely damage components.

Can't play MIDI the same time as any other sound.

Hello there.
I am a bass player and therefore I use GTick from [extra] as a metronome and TuxGuitar from [community] as GuitarPro tablature reader. I have a problem with (at least) this two programs: when I try to start the metronome or to play a tablature, I get error messages if any sound is playing, no matter if it is Amarok playing music, or TuxGuitar playing a tablature.
For non-musicians, a GuitarPro tablature is a transcription of a song which can be played through MIDI at the same time as you can see the partiture, or the notes played on an image of a guitar neck or a keyboard.
This happens every time I want to play some MIDI from this two apps: if I have a tablature open with TuxGuitar I can't play another tablature on another instance of TuxGuitar, or even start GTick to play alone with the metronome. I have to close TuxGuitar and start GTick, or open the other tablature in the same instance of TuxGuitar, which is annoying when, for example, trying to compare two tablatures of the same song.
This makes me think that my system can't play any MIDI when some other sound is playing, but I don't know any other MIDI apps to try. Any ideas on how to setup multiple MIDI playing? I've read on the Wiki [1] that it was necessary to enable dmix some time ago to allow multiple programs play sound at once, but this is not necessary since ALSA 1.0.9rc2, and I have alsa-lib 1.0.17a-1.
Thank you in advance.
Bye!
[1] http://wiki.archlinux.org/index.php/All … ing_system.

Surgat_ wrote:What does "rewrite ALSA so it software-mixes OSS" mean?
It means, "do some horrendously geeky stuff which only a very small proportion of the population have the knowledge, competence, ability and free time to do for no pay."
You're looking for an easy solution. There is no easy solution.
It's the app that says "ALSA-compatible", and it's the app that I'm criticizing for that statement. Not you.
Welcome to the ridiculous world of Linux audio - see jungle and my recommendation. This is what free software produces - an unholy mess of conflicting, contradicting and overlapping standards.
Last edited by brebs (2008-10-01 21:04:34)

Amarok 2.5.0-1 - no sound

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