[solved] alsa: duplicate front signal

Hi,
I got a new sound device today, a Terratec Aureon Sky 5.1.
The driver of my old device offerd me a switch called "duplicate front", which made it possible to plug a second amplifier to the card's rear output and get the same signal as from the front output. As the new device doesn't offer this switch in alsamixer, is there another way to do this?
Another (worse) option would be to have both devices installed, but I can't get them to work simoultanously, I only get a signal on the default card (device 0). How can I make them work parallel?
Thanks, RhahkJin
Last edited by RhahkJin (2008-10-11 11:56:25)

Solved, here is the ~/.asoundrc that mada it work (found in gentoo wiki):
pcm.!dmix {
type dmix
ipc_key 1024
slave {
pcm "hw:0,0"
channels 6
pcm.!default {
type plug
slave.pcm "dmix"
slave.channels 6
route_policy duplicate

Similar Messages

[Solved]Alsa funkiness (sound breaking)

(Note: I originally thought this was a steam issue, but turns out it's not only steam that can break my sound (see the update on the bottom))
I get an issue with Steam where I get a short (less than 1s) sound on repeat/loop (usually it's some part of the message notification sound, or a trailer that I was watching) sounding a bit like a broken record, randomly occurring after keeping steam open for long periods of time in the background. Only fix I've found so far is to restart steam.
I have the Enlightenmen 18 Window Manager.
Notable errors:
Fontconfig error: "/etc/fonts/conf.d/10-scale-bitmap-fonts.conf", line 70: non-double matrix element
Fontconfig warning: "/etc/fonts/conf.d/10-scale-bitmap-fonts.conf", line 78: saw unknown, expected number
[0706/170950:ERROR:object_proxy.cc(239)] Failed to call method: org.freedesktop.DBus.Error.ServiceUnknown: The name org.freedesktop.NetworkManager was not provided by any .service files
[0706/170950:WARNING:proxy_service.cc(958)] PAC support disabled because there is no system implementation
PulseAudio connect failed (used only for Mic Volume Control) with error: Access denied
** (steam:7341): WARNING **: Could not initialize NMClient /org/freedesktop/NetworkManager: The name org.freedesktop.NetworkManager was not provided by any .service files
(steam:7341): LIBDBUSMENU-GLIB-WARNING **: Trying to remove a child that doesn't believe we're it's parent.
[0706/171536:ERROR:reference_audio_renderer.cc(46)] Not implemented reached in virtual void media::ReferenceAudioRenderer::OnCreated(media::AudioOutputController*)
[0706/171536:ERROR:reference_audio_renderer.cc(50)] Not implemented reached in virtual void media::ReferenceAudioRenderer::OnPlaying(media::AudioOutputController*)
[0706/171657:ERROR:reference_audio_renderer.cc(54)] Not implemented reached in virtual void media::ReferenceAudioRenderer::OnPaused(media::AudioOutputController*)
AL lib: pulseaudio.c:612: Context did not connect: Access denied
I don't have any reliable method to reproduce the issue, and it does not seem to output an error of it's own in the terminal output when it occurs. I have noticed that this is more likely to occur when steam has been open for prolonged periods. I am not using pulseaudio, just Alsa. I also noticed when I was running a Wine game (not through steam) that this issue occurred, and it seemed to contaminate the sound outputs for whine too (I got like a "double" sound for every sound effect the game played). If the sound was completely broken on steam and making a loop like I explained at the start of this post and then I close steam while the Wine game is running, the sound from steam will remain until I shut down the game too (i.e. seems that this isn't just breakage internally in steam, but in alsa itself)
The one thing I have tried to solve this was install lib32-libpulse, this did not seem to fix the errors nor the problem. I imagine installing pulseaudio could fix this problem, but pulseaudio downright hates my sound card, so I would rather not use it.
I also have two other issues too that are less severe.
1: Mouse cursor sometimes gets locked in steam instance (Right clicking will show the steam context menu, pressing "stop" in that context menu will solve the issue. This sometimes happens when I use "Ctrl + C"(copy) or "Ctrl + V"(paste) when I'm typing in the discussions)
2: Fullscreen videos will not go fullscreen (they will only fill the steam window)
Update:
I found a bulletproof way to make a similar issue occur.
If I open facebook in firefox, and get a message notification while I am playing a movie in Cmplayer (my media player of choice) the sound in CMplayer breaks (but firefox's sound keeps working normally)
Update2:
I was rather desperate to get this fixed so I installed pulseaudio as much as I despise it, the quality of my audio is damaged by it, but this problem was as I suspected fixed by installing it. I am guessing pulseaudio is set up partially to prevent issues like this one, but I would relaly prefer not to have it. This does not rid me of all the errors however (reference_audio_renderer errors persist) but it does create a new error for steam:
[0706/215704:ERROR:alsa_output.cc(684)] Failed querying delay: Input/output error
Last edited by rabcor (2014-07-11 16:28:34)

I always had to have an asound.conf in the name of upmixing 2.0 to 5.1 This is the content of my asound.conf before I made the changes. I also had a problem with a game where I had crackling noises on 48khz so I had to tell alsa to force 96khz (or 44.1khz but since my card supports 96khz that's what I went with) to avoid that.
pcm.!default {
slave.pcm "surround51"
slave.channels 6
type route
# Front and rear
ttable.0.0 0.7
ttable.1.1 0.7
ttable.2.2 0.6
ttable.3.3 0.6
# Center and LFE
ttable.4.4 1
ttable.5.5 1
# Front left/right to center
ttable.0.4 0.5
ttable.1.4 0.5
# Front left/right to rear
ttable.0.2 0.5
ttable.1.3 0.5
Now it looks like this and everything seems to be working fine (except that mono sound only plays from my left speaker and I haven't defined a way to downmix 7.1 to 5.1):
pcm.dmixed {
type asym
playback.pcm {
# See plugin:dmix at http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html
type dmix
# Don't block other users, e.g. the Timidity midi-player daemon
# http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html
ipc_key_add_uid true
ipc_key 5678293
ipc_perm 0660
ipc_gid audio
# Don't put the rate here! Otherwise it resets the rate & channels set below, as shown by: cat /proc/asound/card0/pcm0p/sub0/hw_params
slave {
# 2 for stereo, 6 for surround51, 8 for surround71
channels 6
pcm {
# mplayer chooses S32_LE, but others usually S16_LE
format S32_LE
#format S16_LE
# 44100 or 48000
# 44100 for music, 48000 is compatible with most h/w
#rate 44100
rate 96000
# http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html
# Maybe helps
nonblock true
type hw
card 0
device 0
subdevice 0
# mplayer2 chooses 1024
# period_size 512 with buffer_size 16384 stops crackling in xmame
# 320 breaks flash - https://bbs.archlinux.org/viewtopic.php?id=129458
#period_size 512
period_size 1024
#period_size 512
# 4096 might make sound crackle
# mplayer2 chooses 8192. Half-Life 2 chooses 16384.
# If too large, use CONFIG_SND_HDA_PREALLOC_SIZE=2048
buffer_size 16384
capture.pcm "hw:0"
pcm.!default{
type plug
slave.pcm "upmix20_51"
pcm.!surround20 {
type plug
slave.pcm "upmix20_51"
pcm.!surround40 {
type plug
slave.pcm "dmixed"
route_policy duplicate
pcm.!surround51 {
type plug
slave.pcm "dmixed"
pcm.upmix20_51 {
slave.pcm "dmixed"
slave.channels 6
type route
# Front and rear
ttable.0.0 0.7
ttable.1.1 0.7
ttable.2.2 0.6
ttable.3.3 0.6
# Center and LFE
ttable.4.4 1
ttable.5.5 1
# Front left/right to center
ttable.0.4 0.5
ttable.1.4 0.5
# Front left/right to rear
ttable.0.2 0.5
ttable.1.3 0.5
Is there a better way for me to achieve the results of the above config? Because I'm all ears.
Last edited by rabcor (2014-07-08 11:09:23)

[Solved] Alsa with 24 bit Output Exclusively

This might be an unusual request hoping to find any ALSA expert. Hope this is a right place to post.
I just made an asynchronous USB2 audio device supporting 44.1k, 48k, 88.2, 96k, 192k sampling frequencies. The firmware so far just supports 24bit word length. Since this is a truly audiophile device the mixer / re-sampler is no option although it works (e.g. Amarok with Xine backend). MPlayer plays bit-perfect as long as the content is 24 bit and one of the above sampling frequencies - works since ALSA update in February 2011. It is really fun this way! However, most peoples's music content is 16 bit.
For Windows the ASIO driver is solving the problem, but I doubt it transfers bit-perfect (Tests to follow soon).
I really would like to recommend Linux for the use, but in this case the problem must be resolved. i could imagine 2 ways of doing:
1. Simple: ALSA is configured to output any content with 24 bit word length. Means adding zeros after the 16th bit.
Disadvantage: Following Digital to Analog converters would accept the signal as true 24 bit and skip the quantization smoothing.
2. Perfect: If there is an available option for doing this re-quantization in Alsa or MPlayer it would be perfect. As long as the algorithm is good, it doesn't matter this process is done within the DAC or within the computer.
My goal is to make a clear instruction of how to use our device with Linux without compromise.
Thanks for any idea!
Last edited by jcci (2011-05-04 02:41:04)

First of all thanks for so much response!
LADSPA was not known to me and I will study what I can do with it. Do you know how to use it?
Edit: LADSPA is meant for production software. No popular Linux player can use those plugins. A solution that works with Mplayer, gstreamer or xine would be accepted by most Linux users, since there are a lot of players on top of those.
Every modern DAC is using re-quantization, synchronous up-sampling and depending on the architecture also noise shaping. The general requirements for s/n can not be achieved with a pure 16bit approach. This is not a problem of the DAC hardware. The USB device can not be initialized when using a 16bit audio file; so the music never reaches the DAC.
The re-sampler inside Linux (or Windows) is definitely downgrading the audio quality. You don't need to have "golden ears" for experiencing this, rather than just a good audio system.
A clean re-quantization could do good, depending on the algorithm.
In case of just enlarging the word length (means in fact no signal change) the re-quantization would be also skipped in the DAC - a very puristic approach which should be proofed to be better. So how to configure ALSA for this?
A dither below on the very low bits of the 24bit signal should be a good idea since it is proofed to statistically shuffle the linearity errors and make the less audible.
Today I did some tests with a very high grade audio system and Linux performs great with 24 bit content. It is not easy to beat top grade CD players on the same DAC. In fact this USB solution is part of a DAC.
The Windows driver works good on my desktop system, but gets choppy on my Macbook... Still some work to do before I can compare.
I admit the cleanest solution would be a firmware revision, but the original firmware making company has no interest in solving this problem and Linux user would have to wait too long or sacrifice sound quality by re-sampling.
Last edited by jcci (2011-04-28 08:42:31)

[SOLVED] Alsa: How to disable auto-mute when plugging in headphones?

I have a set of headphones plugged into my workstation permanently (the headphone jack is hard to reach so I don't want to plug those headphones in and out) as well as some speakers. Jack sensing works perfectly - when I unplug the headphones, the speakers are unmuted and when I plug the headphones back in the speakers are muted. Now as I said, my headphones are plugged in permanently and occasionally I want to have sound on the speakers at the same time. So far I have not found a way to have both headphones and speakers unmuted simultaneously.
I originally thought this was a problem with Pulseaudio (I am using Cinnamon as my desktop) but when I directly go to the Alsa hw device I get exactly the same behavior. The sound chip is a built-in VIA VT1818S and aplay displays the correct cards and devices:
$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: SB [HDA ATI SB], device 0: VT1818S Analog [VT1818S Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: SB [HDA ATI SB], device 1: VT1818S Digital [VT1818S Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: SB [HDA ATI SB], device 2: VT1818S HP [VT1818S HP]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: NVidia [HDA NVidia], device 3: HDMI 0 [HDMI 0]
Subdevices: 1/1
Subdevice #0: subdevice #0
I verified with "mpg123 -a hw:0,0" that I get no sound on the speakers when the headphones are plugged in and vice versa. I tried a lot of different settings with alsamixer and amixer to no avail. These are my current settings:
$ amixer -c 0 contents
numid=42,iface=CARD,name='Front Headphone Jack'
; type=BOOLEAN,access=r-------,values=1
: values=on
numid=44,iface=CARD,name='Front Mic Jack'
; type=BOOLEAN,access=r-------,values=1
: values=off
numid=46,iface=CARD,name='HDMI Phantom Jack'
; type=BOOLEAN,access=r-------,values=1
: values=on
numid=45,iface=CARD,name='Line Jack'
; type=BOOLEAN,access=r-------,values=1
: values=off
numid=40,iface=CARD,name='Line Out CLFE Jack'
; type=BOOLEAN,access=r-------,values=1
: values=off
numid=38,iface=CARD,name='Line Out Front Jack'
; type=BOOLEAN,access=r-------,values=1
: values=on
numid=41,iface=CARD,name='Line Out Side Jack'
; type=BOOLEAN,access=r-------,values=1
: values=off
numid=39,iface=CARD,name='Line Out Surround Jack'
; type=BOOLEAN,access=r-------,values=1
: values=off
numid=43,iface=CARD,name='Rear Mic Jack'
; type=BOOLEAN,access=r-------,values=1
: values=off
numid=47,iface=CARD,name='SPDIF Phantom Jack',index=1
; type=BOOLEAN,access=r-------,values=1
: values=on
numid=37,iface=MIXER,name='Master Playback Switch'
; type=BOOLEAN,access=rw------,values=1
: values=on
numid=36,iface=MIXER,name='Master Playback Volume'
; type=INTEGER,access=rw---R--,values=1,min=0,max=42,step=0
: values=23
| dBscale-min=-63.00dB,step=1.50dB,mute=0
numid=13,iface=MIXER,name='Headphone Playback Switch'
; type=BOOLEAN,access=rw------,values=2
: values=on,on
numid=12,iface=MIXER,name='PCM Playback Switch'
; type=BOOLEAN,access=rw------,values=2
: values=on,on
numid=11,iface=MIXER,name='PCM Playback Volume'
; type=INTEGER,access=rw---R--,values=2,min=0,max=31,step=0
: values=31,31
| dBscale-min=-34.50dB,step=1.50dB,mute=0
numid=28,iface=MIXER,name='Front Mic Boost Volume'
; type=INTEGER,access=rw---R--,values=2,min=0,max=3,step=0
: values=0,0
| dBscale-min=0.00dB,step=10.25dB,mute=0
numid=24,iface=MIXER,name='Front Mic Playback Switch'
; type=BOOLEAN,access=rw------,values=2
: values=on,on
numid=23,iface=MIXER,name='Front Mic Playback Volume'
; type=INTEGER,access=rw---R--,values=2,min=0,max=31,step=0
: values=31,31
| dBscale-min=-34.50dB,step=1.50dB,mute=0
numid=2,iface=MIXER,name='Front Playback Switch'
; type=BOOLEAN,access=rw------,values=2
: values=on,on
numid=1,iface=MIXER,name='Front Playback Volume'
; type=INTEGER,access=rw---R--,values=2,min=0,max=42,step=0
: values=42,42
| dBscale-min=-63.00dB,step=1.50dB,mute=0
numid=4,iface=MIXER,name='Surround Playback Switch'
; type=BOOLEAN,access=rw------,values=2
: values=on,on
numid=3,iface=MIXER,name='Surround Playback Volume'
; type=INTEGER,access=rw---R--,values=2,min=0,max=42,step=0
: values=42,42
| dBscale-min=-63.00dB,step=1.50dB,mute=0
numid=6,iface=MIXER,name='Center Playback Switch'
; type=BOOLEAN,access=rw------,values=1
: values=on
numid=5,iface=MIXER,name='Center Playback Volume'
; type=INTEGER,access=rw---R--,values=1,min=0,max=42,step=0
: values=42
| dBscale-min=-63.00dB,step=1.50dB,mute=0
numid=8,iface=MIXER,name='LFE Playback Switch'
; type=BOOLEAN,access=rw------,values=1
: values=on
numid=7,iface=MIXER,name='LFE Playback Volume'
; type=INTEGER,access=rw---R--,values=1,min=0,max=42,step=0
: values=42
| dBscale-min=-63.00dB,step=1.50dB,mute=0
numid=26,iface=MIXER,name='Line Playback Switch'
; type=BOOLEAN,access=rw------,values=2
: values=on,on
numid=25,iface=MIXER,name='Line Playback Volume'
; type=INTEGER,access=rw---R--,values=2,min=0,max=31,step=0
: values=31,31
| dBscale-min=-34.50dB,step=1.50dB,mute=0
numid=16,iface=MIXER,name='Capture Switch'
; type=BOOLEAN,access=rw------,values=2
: values=on,on
numid=18,iface=MIXER,name='Capture Switch',index=1
; type=BOOLEAN,access=rw------,values=2
: values=on,on
numid=15,iface=MIXER,name='Capture Volume'
; type=INTEGER,access=rw---R--,values=2,min=0,max=31,step=0
: values=0,0
| dBscale-min=-16.50dB,step=1.50dB,mute=0
numid=17,iface=MIXER,name='Capture Volume',index=1
; type=INTEGER,access=rw---R--,values=2,min=0,max=31,step=0
: values=0,0
| dBscale-min=-16.50dB,step=1.50dB,mute=0
numid=14,iface=MIXER,name='Loopback Mixing'
; type=ENUMERATED,access=rw------,values=1,items=2
; Item #0 'Disabled'
; Item #1 'Enabled'
: values=0
numid=35,iface=MIXER,name='IEC958 Default PCM Playback Switch'
; type=BOOLEAN,access=rw------,values=1
: values=off
numid=31,iface=MIXER,name='IEC958 Playback Con Mask'
; type=IEC958,access=r-------,values=1
: values=[AES0=0x0f AES1=0xff AES2=0x00 AES3=0x00]
numid=32,iface=MIXER,name='IEC958 Playback Pro Mask'
; type=IEC958,access=r-------,values=1
: values=[AES0=0x0f AES1=0x00 AES2=0x00 AES3=0x00]
numid=33,iface=MIXER,name='IEC958 Playback Default'
; type=IEC958,access=rw------,values=1
: values=[AES0=0x04 AES1=0x00 AES2=0x00 AES3=0x00]
numid=34,iface=MIXER,name='IEC958 Playback Switch'
; type=BOOLEAN,access=rw------,values=1
: values=off
numid=30,iface=MIXER,name='Dynamic Power-Control'
; type=ENUMERATED,access=rw------,values=1,items=2
; Item #0 'Disabled'
; Item #1 'Enabled'
: values=0
numid=29,iface=MIXER,name='Independent HP'
; type=ENUMERATED,access=rw------,values=1,items=2
; Item #0 'OFF'
; Item #1 'ON'
: values=0
numid=19,iface=MIXER,name='Input Source'
; type=ENUMERATED,access=rw------,values=1,items=4
; Item #0 'Rear Mic'
; Item #1 'Front Mic'
; Item #2 'Line'
; Item #3 'Stereo Mixer'
: values=1
numid=20,iface=MIXER,name='Input Source',index=1
; type=ENUMERATED,access=rw------,values=1,items=4
; Item #0 'Rear Mic'
; Item #1 'Front Mic'
; Item #2 'Line'
; Item #3 'Stereo Mixer'
: values=0
numid=27,iface=MIXER,name='Rear Mic Boost Volume'
; type=INTEGER,access=rw---R--,values=2,min=0,max=3,step=0
: values=0,0
| dBscale-min=0.00dB,step=10.25dB,mute=0
numid=22,iface=MIXER,name='Rear Mic Playback Switch'
; type=BOOLEAN,access=rw------,values=2
: values=off,off
numid=21,iface=MIXER,name='Rear Mic Playback Volume'
; type=INTEGER,access=rw---R--,values=2,min=0,max=31,step=0
: values=23,23
| dBscale-min=-34.50dB,step=1.50dB,mute=0
numid=10,iface=MIXER,name='Side Playback Switch'
; type=BOOLEAN,access=rw------,values=2
: values=on,on
numid=9,iface=MIXER,name='Side Playback Volume'
; type=INTEGER,access=rw---R--,values=2,min=0,max=42,step=0
: values=42,42
| dBscale-min=-63.00dB,step=1.50dB,mute=0
numid=49,iface=PCM,name='Capture Channel Map'
; type=INTEGER,access=r----R--,values=2,min=0,max=36,step=0
: values=0,0
| | TLV size error (257, 8, 0)!
numid=50,iface=PCM,name='Capture Channel Map',index=1
; type=INTEGER,access=r----R--,values=2,min=0,max=36,step=0
: values=0,0
| | TLV size error (257, 8, 0)!
numid=48,iface=PCM,name='Playback Channel Map'
; type=INTEGER,access=r----R--,values=8,min=0,max=36,step=0
: values=0,0,0,0,0,0,0,0
| | TLV size error (257, 8, 0)!
numid=51,iface=PCM,name='Playback Channel Map',device=1
; type=INTEGER,access=r----R--,values=2,min=0,max=36,step=0
: values=0,0
| | TLV size error (257, 8, 0)!
numid=52,iface=PCM,name='Playback Channel Map',device=2
; type=INTEGER,access=r----R--,values=2,min=0,max=36,step=0
: values=0,0
| | TLV size error (257, 8, 0)!
If I read the Alsa kernel docs right, the VIA driver simply does not provide any option to disable auto-muting: http://git.alsa-project.org/?p=alsa-ker … xt;hb=HEAD.
I would be glad if I had misread/misunderstood something or there were any other means of achieving what I am after? All I found so far is lots of posts where auto-muting did not work but very few people seem to want the opposite. FWIW, my old installation of Ubuntu 11.10 was working fine in that regard, i.e. it probably did not support auto-muting for my hardware.
Last edited by fax (2013-03-24 20:20:19)

Solved, well, good enough for me anyway. In alsamixer, I enabled "Independent HP". Independent HP means that two of the six surround channels get diverted to the headphones. Alsa creates a second device, one for line out and the other one for the headphones. It is a very neat feature that is meant to allow you to e.g. do a voice call with your headphones while you play music through the speakers at the same time. I can now choose in PulseAudio whether I want output to my headphones or the speakers. It required that in /etc/pulse/default.pa, I added the line "load-module module-alsa-sink device=hw:0,2" after this paragraph:
### Automatically load driver modules depending on the hardware available
.ifexists module-udev-detect.so
load-module module-udev-detect
.else
### Use the static hardware detection module (for systems that lack udev support)
load-module module-detect
.endif
# Take Independent HP into use.
load-module module-alsa-sink device=hw:0,2
All this really is a work-around for two bugs:
Call it a bug or a missing feature, but it does not look like the Alsa snd_hda_codec_via driver allows to disable auto-muting.
Before the change to default.pa, PulseAudio already allows me to switch between two output devices - "analog output" and "analog headphones". "Analog headphones" however does the same thing as "analog output" and plays sound on the external speakers and not the headphones. I suspect that the PulseAudio module-udev-detect detects both Alsa devices but wires them wrongly.
The above still doesn't give me what I originally asked, i.e. sound on both speakers and headphones at the same time, but I realized that I don't really need that. I just need a way to switch on speakers occasionally while my headphones are plugged in.
Last edited by fax (2013-03-24 23:06:27)

[SOLVED] ALSA input source not not saving on reboot

Just in case anyone else has a similar issue....
An annoying recent development was the ALSA input source not being restored on rebooting. In particular in my case the internal digital microphone on my Dell XPS m1330.
The audio chip is Intel-HDA, reported as "STAC9228" by alsamixer. The primary 'sound device' is alsamixer is listed as PulseAudio, with Intel HDA being an additional device (F6). Under the F4 capture tab, there are 3 input source options, the first kept on resetting to 'Mic' instead of 'Internal Mic 1'. Mic is the front analog mic jack, and as a result there was no Skype microphone functionality each reboot. Easy enough to use alsamixer to reset to the correct input when required of course, but annoying nonetheless.
I tried deleting asound.conf, ensuring alsa was (and wasn't) running as a daemon and using alsactl store, all to no avail. I already have !snd_pcm_oss and !snd-pcm-oss in my rc.conf, and Skype is not controlling any levels.
Finally solved by adding 'alsactl restore' to the startup programs (gnome-session-properties).
Last edited by sultanoswing (2011-05-17 00:56:27)

[UPDATE: Internal microphone no longer working under kernel 3.1.1-1 and alsa 0.24-3.]
Sigh.

[solved] Xorg -configuration crashes (signal 11)

I've been trying to get Xorg to work all day without success. What I've been doing:
* Install archlinux from scratch
* Upgrade with pacman -Syu
* Install Xorg
* Install fglrx
* Run Xorg -configure.
This is when it crashes with the following output:
X Window System Version 7.2.0
Release Date: 22 January 2007
X Protocol Version 11, Revision 0, Release 7.2
Build Operating System: UNKNOWN
Current Operating System: Linux twarkie 2.6.22-ARCH #1 SMP PREEMPT Tue Sep 25 19:18:19 CEST 2007 i686
Build Date: 08 April 2007
Before reporting problems, check http://wiki.x.org
to make sure that you have the latest version.
Module Loader present
Markers: (--) probed, (**) from config file, (==) default setting,
(++) from command line, (!!) notice, (II) informational,
(WW) warning, (EE) error, (NI) not implemented, (??) unknown.
(==) Log file: "/var/log/Xorg.0.log", Time: Wed Sep 26 22:11:11 2007
(WW) Open ACPI failed (/var/run/acpid.socket) (No such file or directory)
(II) No APM support in BIOS or kernel
(II) Loader magic: 0x81c44a0
(II) Module ABI versions:
X.Org ANSI C Emulation: 0.3
X.Org Video Driver: 1.1
X.Org XInput driver : 0.7
X.Org Server Extension : 0.3
X.Org Font Renderer : 0.5
(II) Loader running on linux
(II) LoadModule: "pcidata"
(II) Loading /usr/lib/xorg/modules//libpcidata.so
(II) Module pcidata: vendor="X.Org Foundation"
compiled for 7.2.0, module version = 1.0.0
ABI class: X.Org Video Driver, version 1.1
(--) using VT number 7
(II) PCI: PCI scan (all values are in hex)
(II) PCI: 00:00:0: chip 10de,00e1 card 0000,0000 rev a1 class 06,00,00 hdr 00
(II) PCI: 00:01:0: chip 10de,00e0 card 10de,0c11 rev a2 class 06,01,00 hdr 80
(II) PCI: 00:01:1: chip 10de,00e4 card 10de,0c11 rev a1 class 0c,05,00 hdr 80
(II) PCI: 00:02:0: chip 10de,00e7 card 10de,0c11 rev a1 class 0c,03,10 hdr 80
(II) PCI: 00:02:1: chip 10de,00e7 card 10de,0c11 rev a1 class 0c,03,10 hdr 80
(II) PCI: 00:02:2: chip 10de,00e8 card 10de,0c11 rev a2 class 0c,03,20 hdr 80
(II) PCI: 00:05:0: chip 10de,00df card 10de,0c11 rev a2 class 06,80,00 hdr 00
(II) PCI: 00:08:0: chip 10de,00e5 card f0de,0c11 rev a2 class 01,01,8a hdr 00
(II) PCI: 00:09:0: chip 10de,00ee card 10de,0c11 rev a2 class 01,01,85 hdr 00
(II) PCI: 00:0a:0: chip 10de,00e3 card 10de,0c11 rev a2 class 01,01,85 hdr 00
(II) PCI: 00:0b:0: chip 10de,00e2 card 0000,0000 rev a2 class 06,04,00 hdr 01
(II) PCI: 00:0e:0: chip 10de,00ed card 0000,0000 rev a2 class 06,04,00 hdr 01
(II) PCI: 00:18:0: chip 1022,1100 card 0000,0000 rev 00 class 06,00,00 hdr 80
(II) PCI: 00:18:1: chip 1022,1101 card 0000,0000 rev 00 class 06,00,00 hdr 80
(II) PCI: 00:18:2: chip 1022,1102 card 0000,0000 rev 00 class 06,00,00 hdr 80
(II) PCI: 00:18:3: chip 1022,1103 card 0000,0000 rev 00 class 06,00,00 hdr 80
(II) PCI: 01:00:0: chip 1002,4152 card 174b,7c29 rev 00 class 03,00,00 hdr 80
(II) PCI: 01:00:1: chip 1002,4172 card 174b,7c28 rev 00 class 03,80,00 hdr 00
(II) PCI: 02:0b:0: chip 1412,1724 card 3031,4553 rev 01 class 04,01,00 hdr 00
(II) PCI: End of PCI scan
(II) Host-to-PCI bridge:
(II) Bus 0: bridge is at (0:0:0), (0,0,2), BCTRL: 0x0008 (VGA_EN is set)
(II) Bus 0 I/O range:
[0] -1 0 0x00000000 - 0x0000ffff (0x10000) IX[b]
(II) Bus 0 non-prefetchable memory range:
[0] -1 0 0x00000000 - 0xffffffff (0x0) MX[b]
(II) Bus 0 prefetchable memory range:
[0] -1 0 0x00000000 - 0xffffffff (0x0) MX[b]
(II) PCI-to-ISA bridge:
(II) Bus -1: bridge is at (0:1:0), (0,-1,-1), BCTRL: 0x0008 (VGA_EN is set)
(II) PCI-to-PCI bridge:
(II) Bus 1: bridge is at (0:11:0), (0,1,1), BCTRL: 0x000a (VGA_EN is set)
(II) Bus 1 I/O range:
[0] -1 0 0x0000b000 - 0x0000bfff (0x1000) IX[b]
(II) Bus 1 non-prefetchable memory range:
[0] -1 0 0xe8000000 - 0xe8ffffff (0x1000000) MX[b]
(II) Bus 1 prefetchable memory range:
[0] -1 0 0xd8000000 - 0xe7ffffff (0x10000000) MX[b]
(II) PCI-to-PCI bridge:
(II) Bus 2: bridge is at (0:14:0), (0,2,2), BCTRL: 0x0202 (VGA_EN is cleared)
(II) Bus 2 I/O range:
[0] -1 0 0x0000c000 - 0x0000cfff (0x1000) IX[b]
(II) Bus 2 non-prefetchable memory range:
[0] -1 0 0xe9000000 - 0xe9ffffff (0x1000000) MX[b]
(--) PCI:*(1:0:0) ATI Technologies Inc RV350 AR [Radeon 9600] rev 0, Mem @ 0xd8000000/27, 0xe8030000/16, I/O @ 0xb000/8
(--) PCI: (1:0:1) ATI Technologies Inc RV350 AR [Radeon 9600] (Secondary) rev 0, Mem @ 0xe0000000/27, 0xe8020000/16
List of video drivers:
fglrx
(II) LoadModule: "fglrx"
(II) Loading /usr/lib/xorg/modules/drivers//fglrx_drv.so
(II) Module fglrx: vendor="FireGL - ATI Technologies Inc."
compiled for 7.1.0, module version = 8.40.4
Module class: X.Org Video Driver
ABI class: X.Org Video Driver, version 1.0
(II) Addressable bus resource ranges are
[0] -1 0 0x00000000 - 0xffffffff (0x0) MX[b]
[1] -1 0 0x00000000 - 0x0000ffff (0x10000) IX[b]
(II) OS-reported resource ranges:
[0] -1 0 0x00100000 - 0x3fffffff (0x3ff00000) MX[b]E(B)
[1] -1 0 0x000f0000 - 0x000fffff (0x10000) MX[b]
[2] -1 0 0x000c0000 - 0x000effff (0x30000) MX[b]
[3] -1 0 0x00000000 - 0x0009ffff (0xa0000) MX[b]
[4] -1 0 0x0000ffff - 0x0000ffff (0x1) IX[b]
[5] -1 0 0x00000000 - 0x000000ff (0x100) IX[b]
(II) PCI Memory resource overlap reduced 0xd0000000 from 0xd7ffffff to 0xcfffffff
(II) Active PCI resource ranges:
[0] -1 0 0xea000000 - 0xea000fff (0x1000) MX[b]
[1] -1 0 0xea004000 - 0xea0040ff (0x100) MX[b]
[2] -1 0 0xea003000 - 0xea003fff (0x1000) MX[b]
[3] -1 0 0xea002000 - 0xea002fff (0x1000) MX[b]
[4] -1 0 0xd0000000 - 0xcfffffff (0x0) MX[b]O
[5] -1 0 0xe8030000 - 0xe803ffff (0x10000) MX[b](B)
[6] -1 0 0xd8000000 - 0xdfffffff (0x8000000) MX[b](B)
[7] -1 0 0x0000c200 - 0x0000c27f (0x80) IX[b]
[8] -1 0 0x0000c100 - 0x0000c11f (0x20) IX[b]
[9] -1 0 0x0000df00 - 0x0000df7f (0x80) IX[b]
[10] -1 0 0x0000de00 - 0x0000de0f (0x10) IX[b]
[11] -1 0 0x00000b70 - 0x00000b73 (0x4) IX[b]
[12] -1 0 0x00000970 - 0x00000977 (0x8) IX[b]
[13] -1 0 0x00000bf0 - 0x00000bf3 (0x4) IX[b]
[14] -1 0 0x000009f0 - 0x000009f7 (0x8) IX[b]
[15] -1 0 0x0000d900 - 0x0000d97f (0x80) IX[b]
[16] -1 0 0x0000d800 - 0x0000d80f (0x10) IX[b]
[17] -1 0 0x00000b60 - 0x00000b63 (0x4) IX[b]
[18] -1 0 0x00000960 - 0x00000967 (0x8) IX[b]
[19] -1 0 0x00000be0 - 0x00000be3 (0x4) IX[b]
[20] -1 0 0x000009e0 - 0x000009e7 (0x8) IX[b]
[21] -1 0 0x0000f000 - 0x0000f00f (0x10) IX[b]
[22] -1 0 0x0000d100 - 0x0000d107 (0x8) IX[b]
[23] -1 0 0x00004c40 - 0x00004c7f (0x40) IX[b]
[24] -1 0 0x00004c00 - 0x00004c3f (0x40) IX[b]
[25] -1 0 0x0000e000 - 0x0000e01f (0x20) IX[b]
[26] -1 0 0x0000b000 - 0x0000b0ff (0x100) IX[b](B)
(II) Inactive PCI resource ranges:
[0] -1 0 0xe8020000 - 0xe802ffff (0x10000) MX[b](B)
[1] -1 0 0xe0000000 - 0xe7ffffff (0x8000000) MX[b](B)
(II) Active PCI resource ranges after removing overlaps:
[0] -1 0 0xea000000 - 0xea000fff (0x1000) MX[b]
[1] -1 0 0xea004000 - 0xea0040ff (0x100) MX[b]
[2] -1 0 0xea003000 - 0xea003fff (0x1000) MX[b]
[3] -1 0 0xea002000 - 0xea002fff (0x1000) MX[b]
[4] -1 0 0xd0000000 - 0xcfffffff (0x0) MX[b]O
[5] -1 0 0xe8030000 - 0xe803ffff (0x10000) MX[b](B)
[6] -1 0 0xd8000000 - 0xdfffffff (0x8000000) MX[b](B)
[7] -1 0 0x0000c200 - 0x0000c27f (0x80) IX[b]
[8] -1 0 0x0000c100 - 0x0000c11f (0x20) IX[b]
[9] -1 0 0x0000df00 - 0x0000df7f (0x80) IX[b]
[10] -1 0 0x0000de00 - 0x0000de0f (0x10) IX[b]
[11] -1 0 0x00000b70 - 0x00000b73 (0x4) IX[b]
[12] -1 0 0x00000970 - 0x00000977 (0x8) IX[b]
[13] -1 0 0x00000bf0 - 0x00000bf3 (0x4) IX[b]
[14] -1 0 0x000009f0 - 0x000009f7 (0x8) IX[b]
[15] -1 0 0x0000d900 - 0x0000d97f (0x80) IX[b]
[16] -1 0 0x0000d800 - 0x0000d80f (0x10) IX[b]
[17] -1 0 0x00000b60 - 0x00000b63 (0x4) IX[b]
[18] -1 0 0x00000960 - 0x00000967 (0x8) IX[b]
[19] -1 0 0x00000be0 - 0x00000be3 (0x4) IX[b]
[20] -1 0 0x000009e0 - 0x000009e7 (0x8) IX[b]
[21] -1 0 0x0000f000 - 0x0000f00f (0x10) IX[b]
[22] -1 0 0x0000d100 - 0x0000d107 (0x8) IX[b]
[23] -1 0 0x00004c40 - 0x00004c7f (0x40) IX[b]
[24] -1 0 0x00004c00 - 0x00004c3f (0x40) IX[b]
[25] -1 0 0x0000e000 - 0x0000e01f (0x20) IX[b]
[26] -1 0 0x0000b000 - 0x0000b0ff (0x100) IX[b](B)
(II) Inactive PCI resource ranges after removing overlaps:
[0] -1 0 0xe8020000 - 0xe802ffff (0x10000) MX[b](B)
[1] -1 0 0xe0000000 - 0xe7ffffff (0x8000000) MX[b](B)
(II) OS-reported resource ranges after removing overlaps with PCI:
[0] -1 0 0x00100000 - 0x3fffffff (0x3ff00000) MX[b]E(B)
[1] -1 0 0x000f0000 - 0x000fffff (0x10000) MX[b]
[2] -1 0 0x000c0000 - 0x000effff (0x30000) MX[b]
[3] -1 0 0x00000000 - 0x0009ffff (0xa0000) MX[b]
[4] -1 0 0x0000ffff - 0x0000ffff (0x1) IX[b]
[5] -1 0 0x00000000 - 0x000000ff (0x100) IX[b]
(II) All system resource ranges:
[0] -1 0 0x00100000 - 0x3fffffff (0x3ff00000) MX[b]E(B)
[1] -1 0 0x000f0000 - 0x000fffff (0x10000) MX[b]
[2] -1 0 0x000c0000 - 0x000effff (0x30000) MX[b]
[3] -1 0 0x00000000 - 0x0009ffff (0xa0000) MX[b]
[4] -1 0 0xea000000 - 0xea000fff (0x1000) MX[b]
[5] -1 0 0xea004000 - 0xea0040ff (0x100) MX[b]
[6] -1 0 0xea003000 - 0xea003fff (0x1000) MX[b]
[7] -1 0 0xea002000 - 0xea002fff (0x1000) MX[b]
[8] -1 0 0xd0000000 - 0xcfffffff (0x0) MX[b]O
[9] -1 0 0xe8030000 - 0xe803ffff (0x10000) MX[b](B)
[10] -1 0 0xd8000000 - 0xdfffffff (0x8000000) MX[b](B)
[11] -1 0 0xe8020000 - 0xe802ffff (0x10000) MX[b](B)
[12] -1 0 0xe0000000 - 0xe7ffffff (0x8000000) MX[b](B)
[13] -1 0 0x0000ffff - 0x0000ffff (0x1) IX[b]
[14] -1 0 0x00000000 - 0x000000ff (0x100) IX[b]
[15] -1 0 0x0000c200 - 0x0000c27f (0x80) IX[b]
[16] -1 0 0x0000c100 - 0x0000c11f (0x20) IX[b]
[17] -1 0 0x0000df00 - 0x0000df7f (0x80) IX[b]
[18] -1 0 0x0000de00 - 0x0000de0f (0x10) IX[b]
[19] -1 0 0x00000b70 - 0x00000b73 (0x4) IX[b]
[20] -1 0 0x00000970 - 0x00000977 (0x8) IX[b]
[21] -1 0 0x00000bf0 - 0x00000bf3 (0x4) IX[b]
[22] -1 0 0x000009f0 - 0x000009f7 (0x8) IX[b]
[23] -1 0 0x0000d900 - 0x0000d97f (0x80) IX[b]
[24] -1 0 0x0000d800 - 0x0000d80f (0x10) IX[b]
[25] -1 0 0x00000b60 - 0x00000b63 (0x4) IX[b]
[26] -1 0 0x00000960 - 0x00000967 (0x8) IX[b]
[27] -1 0 0x00000be0 - 0x00000be3 (0x4) IX[b]
[28] -1 0 0x000009e0 - 0x000009e7 (0x8) IX[b]
[29] -1 0 0x0000f000 - 0x0000f00f (0x10) IX[b]
[30] -1 0 0x0000d100 - 0x0000d107 (0x8) IX[b]
[31] -1 0 0x00004c40 - 0x00004c7f (0x40) IX[b]
[32] -1 0 0x00004c00 - 0x00004c3f (0x40) IX[b]
[33] -1 0 0x0000e000 - 0x0000e01f (0x20) IX[b]
[34] -1 0 0x0000b000 - 0x0000b0ff (0x100) IX[b](B)
(II) Primary Device is: PCI 01:00:0
(II) ATI Proprietary Linux Driver Version Identifier:8.40.4
(II) ATI Proprietary Linux Driver Release Identifier: UNSUPPORTED-8.402
(II) ATI Proprietary Linux Driver Build Date: Jul 31 2007 22:20:14
Backtrace:
0: Xorg(xf86SigHandler+0x84) [0x80d0574]
1: [0xb7fdc420]
2: /usr/lib/xorg/modules/drivers//fglrx_drv.so(atiddxProbeMain+0x11b) [0xb7a95bfb]
3: Xorg(DoConfigure+0x20c) [0x80ce37c]
4: Xorg(InitOutput+0x66a) [0x809fc0a]
5: Xorg(main+0x27b) [0x806e36b]
6: /lib/libc.so.6(__libc_start_main+0xe0) [0xb7df5f90]
7: Xorg(FontFileCompleteXLFD+0xa9) [0x806d8b1]
Fatal server error:
Caught signal 11. Server aborting
If I enable the testing repository and perform a full system upgrade, I can get X to work with the open source drivers for ATI. But I need to get fglrx to work in order to use tv-out. So there seem to be a problem with Xorg itself, not the driver in this case.
Any ideas?
Last edited by twarkie (2007-09-26 23:45:54)

Well, in your previous log, the crash backtrace:
Backtrace:
0: Xorg(xf86SigHandler+0x84) [0x80d0574]
1: [0xb7fdc420]
2: /usr/lib/xorg/modules/drivers//fglrx_drv.so(atiddxProbeMain+0x11b) [0xb7a95bfb]
3: Xorg(DoConfigure+0x20c) [0x80ce37c]
4: Xorg(InitOutput+0x66a) [0x809fc0a]
5: Xorg(main+0x27b) [0x806e36b]
6: /lib/libc.so.6(__libc_start_main+0xe0) [0xb7df5f90]
7: Xorg(FontFileCompleteXLFD+0xa9) [0x806d8b1]
died inside /usr/lib/xorg/modules/drivers/fglrx_drv.so - so I figured if you removed fglrx, Xorg -configure wouldn't try to use that module, and the crash would go away. I'm glad I guessed right.
Could you please add [solved] to your topic title?

Duplicate Inbox after trying to solve Gmail duplicate sent messages...

.. was trying to solve the problem of duplicate sent messages with Gmail email on the BB Curve 8350. In following instructions from this message board, I instead ended up with duplicate INBOXES. Now, when I unregister and re-register my gmail account on my phone, it creates a new icon. This has resulted in my original problem being multiplied by four. What I mean is, now I have duplicate sent messages in TWO DIFFERENT INBOXES. Originally, my gmail account went to the main Inbox on the phone. I cannot delete the main message system.. HELP!

It seems I found a bug!
I now understand what happens, although don't know the reason. The symptoms are caused by the "Auto-Expunge" feature on Gmail. I disabled it and fooled around the options and this is what I discovered:
When I set "the message will be archived when disappear from the last visible folder", the message is archived whenever I try to move the sent messages to any other folder than the "Sent" one;
When I set "move the message to trash when disappear from the last visible folder", the message is deleted and moved to trash whenever I try to move it from the "Sent" folder to any other.
In other words, for some awkward reason, Lion's Apple Mail is not handling folder-to-folder operations and Gmail thinks that the messages has been removed from the last visible folder and does with it whatever it was instructed to, which is either to archive or move to trash.
Since by disabling Auto-Expunge Gmail waits for the client to take the action, it seems that Apple Mail is sending the wrong command or sending it on the wrong timing.
I wonder is Apple is aware of this.
Any suggestions?

[Solved]logroate duplicate entry error

Every night the cron daemon ones and one of its tasks is to run logroate (by default). I keep getting these errors in my inbox. error: samba.pacsave:1 duplicate log entry for /var/log/samba/log.0.0.0.0 I've tried deleting all the samba log files and still no aval. So any help would be great!
Thanks!
Last edited by mezoko (2007-08-23 10:48:52)

I think the problem is that you have in /etc/logrotate.d two files (samba and samba.pacsave) which have the same entry. Make a diff to see which one you want to keep and delete the other one. This should solves it.

[SOLVED] alsa 1.0.23

Hello,
First time arch user, and so far its going pretty well. Currently I have no sound, but am looking to upgrade ALSA to version 1.0.23 which came out on 4/16/2010. I am trying to use pacman, but it only finds 1.0.22-2. Do I need to wait a bit before the repositories are updated? Or am I looking in the wrong spot? I'm aware I can compile it from source, but I'm not very comfortable with doing this yet.
Thanks
Last edited by bubbleman (2010-04-19 04:47:59)

bubbleman wrote:
Ok, l have a Xonar DS in the mail, which alsa 1.0.23 adds support for. I have looked at ABS, but wasn't sure what to do about changing the PKGBUILD to reflect that I wanted version 1.0.23. Do I simply just change the version number, or is there more to it than that? The wiki doesnt really go into explaining that part.
the wiki im looking at is:
http://wiki.archlinux.org/index.php/Arch_Build_System
Try upping the pkgver and see if it compiles. If it doesn't you need to do some problem-solving based on the errors generated.

[SOLVED] ALSA no sound interrnal speaker with HDA Intel PCH

Hi, i have no sound from the internal speaker of my laptop but the headphones work.
configuration : http://www.alsa-project.org/db/?f=68a08 … 6d1a1242ae
i have tried to install the driver from realtek and to add the option "model=auto", "model=generic", "model=3stack" to the snd-hda-intel" module without success.
the command "aplay -vv -D hw:0,0 /usr/share/sounds/alsa/test.wav" plays a sound only through the headphones.
have you any idea to help me ?
thanks
Last edited by walkyrie (2012-06-02 13:22:49)

This seems pretty similar to my issue... No sound from my from headphone output at all... I do get sound from the rear output BUT it stops when I plug in the front one!
Ran the monitor option on the python script and see the following...
PLUG HEADPHONES IN:
======================================
Diff for codec 0/2 (0x11060441):
+++
@@ -220,17 +220,16 @@
Control: name="Line-Out Jack", index=0, device=0
Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-Out vals: [0x00 0x00]
Pincap 0x0001001c: OUT HP EAPD Detect
EAPD 0x2: EAPD
Pin Default 0x01014010: [Jack] Line Out at Ext Rear
Conn = 1/8, Color = Green
DefAssociation = 0x1, Sequence = 0x0
- Pin-ctls: 0x40: OUT
Unsolicited: tag=0x02, enabled=1
Power: setting=D0, actual=D0
Connection: 1
0x18
Node 0x25 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out
Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-Out vals: [0x80 0x80]
Pincap 0x0000001c: OUT HP Detect
======================================
PULL HEADPHONES OUT:
======================================
Diff for codec 0/2 (0x11060441):
+++
@@ -220,16 +220,17 @@
Control: name="Line-Out Jack", index=0, device=0
Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-Out vals: [0x00 0x00]
Pincap 0x0001001c: OUT HP EAPD Detect
EAPD 0x2: EAPD
Pin Default 0x01014010: [Jack] Line Out at Ext Rear
Conn = 1/8, Color = Green
DefAssociation = 0x1, Sequence = 0x0
+ Pin-ctls: 0x40: OUT
Unsolicited: tag=0x02, enabled=1
Power: setting=D0, actual=D0
Connection: 1
0x18
Node 0x25 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out
Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-Out vals: [0x80 0x80]
Pincap 0x0000001c: OUT HP Detect
======================================
Not a lot of difference, but enough to see that the system sees the event...
Now I need to know what to do to make sure the sound starts on the front headphone output when the rear one is disconnected, or if they are parallel, not have the sound stop ;-)

[SOLVED]ALSA sequencer(/dev/snd/seq) missing = broken midi

I get this error when attempting to start qsynth (frontend for fluidsynth)
fluidsynth: warning: Failed to pin the sample data to RAM; swapping is possible.
fluidsynth: warning: Requested a period size of 1024, got 940 instead
ALSA lib seq_hw.c:457:(snd_seq_hw_open) open /dev/snd/seq failed: No such file or directory
fluidsynth: error: Error opening ALSA sequencer
It's right, /dev/snd/seq really does not exist on my system. I have the midi driver set to alsa_seq in the qsynth settings, and it's been working just fine up until now. I haven't tinkered with anything related to alsa, so I'm not sure how it disappeared. The only possible cause I can think of is that today I installed kernel26-lts and have been using it for a few hours. However, rebooting and switching to the standard kernel doesn't solve the problem, as /dev/snd/seq is still missing and I get an identical error message. Any ideas/suggestions/wild guesses?
Last edited by splittercode (2010-08-25 15:14:54)

Maybe you have blacklisted some module in rc.conf?
[karol@black ~]$ lsmod | grep -i seq
snd_seq_dummy 1071 0
snd_seq_oss 25016 0
snd_seq_midi_event 4488 1 snd_seq_oss
snd_seq 41656 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi_event
snd_seq_device 4353 3 snd_seq_dummy,snd_seq_oss,snd_seq
snd_timer 15601 2 snd_seq,snd_pcm
snd 42602 11 snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_mixer_oss,snd_intel8x0,snd_ac97_codec,snd_pcm,snd_timer
[karol@black ~]$ l /dev/snd/seq
crw-rw---- 1 116, 3 2010-08-25 13:56 /dev/snd/seq

[Solved] ALSA "This sound device does not have any controls" XDJ-R1

Hey guys,
I have a XDJ-R1 which includes a 24bit 44.1khz USB soundcard that can be used to record audio from the mixer.
When I plug it in it's detected fine. However using alsamixer (or amixer) I don't see any audio controls for volume.
If I start pulseaudio it will detect the device just fine. However, at 100% volume the audio seems rather low. I'm seeing about -24db while my mixer shows the output at around +2db. I suspect the ALSA volume is rather low. But have no way of changing it as I can't seem to see any controls!
Here are some command outputs:
alpaly -l
**** List of PLAYBACK Hardware Devices ****
card 0: Intel [HDA Intel], device 0: ALC269 Analog [ALC269 Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: XDJR1 [PIONEER XDJ-R1], device 0: USB Audio [USB Audio]
Subdevices: 1/1
Subdevice #0: subdevice #0
aplay -L
null
Discard all samples (playback) or generate zero samples (capture)
default:CARD=Intel
HDA Intel, ALC269 Analog
Default Audio Device
sysdefault:CARD=Intel
HDA Intel, ALC269 Analog
Default Audio Device
front:CARD=Intel,DEV=0
HDA Intel, ALC269 Analog
Front speakers
surround40:CARD=Intel,DEV=0
HDA Intel, ALC269 Analog
4.0 Surround output to Front and Rear speakers
surround41:CARD=Intel,DEV=0
HDA Intel, ALC269 Analog
4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=Intel,DEV=0
HDA Intel, ALC269 Analog
5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=Intel,DEV=0
HDA Intel, ALC269 Analog
5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=Intel,DEV=0
HDA Intel, ALC269 Analog
7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
default:CARD=XDJR1
PIONEER XDJ-R1, USB Audio
Default Audio Device
sysdefault:CARD=XDJR1
PIONEER XDJ-R1, USB Audio
Default Audio Device
front:CARD=XDJR1,DEV=0
PIONEER XDJ-R1, USB Audio
Front speakers
surround40:CARD=XDJR1,DEV=0
PIONEER XDJ-R1, USB Audio
4.0 Surround output to Front and Rear speakers
surround41:CARD=XDJR1,DEV=0
PIONEER XDJ-R1, USB Audio
4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=XDJR1,DEV=0
PIONEER XDJ-R1, USB Audio
5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=XDJR1,DEV=0
PIONEER XDJ-R1, USB Audio
5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=XDJR1,DEV=0
PIONEER XDJ-R1, USB Audio
7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
iec958:CARD=XDJR1,DEV=0
PIONEER XDJ-R1, USB Audio
IEC958 (S/PDIF) Digital Audio Output
amixer -c 1
(output is empty)
cat /proc/asound/devices
2: [ 0- 0]: digital audio playback
3: [ 0- 0]: digital audio capture
4: [ 0- 0]: hardware dependent
5: [ 0] : control
6: [ 1- 0]: raw midi
7: [ 1- 0]: digital audio playback
8: [ 1- 0]: digital audio capture
9: [ 1] : control
33: : timer
(with pulseaudio on) pacmd list-sources
3 source(s) available.
index: 2
name: <alsa_input.usb-Pioneer_PIONEER_XDJ-R1_PIONEER_XDJ-R1-00-XDJR1.analog-stereo>
driver: <module-alsa-card.c>
flags: HARDWARE DECIBEL_VOLUME LATENCY
state: SUSPENDED
suspend cause: IDLE
priority: 9049
volume: front-left: 65536 / 100% / 0.00 dB, front-right: 65536 / 100% / 0.00 dB
balance 0.00
base volume: 65536 / 100% / 0.00 dB
volume steps: 65537
muted: no
current latency: 0.00 ms
max rewind: 0 KiB
sample spec: s24le 2ch 44100Hz
channel map: front-left,front-right
Stereo
used by: 0
linked by: 1
fixed latency: 99.95 ms
card: 1 <alsa_card.usb-Pioneer_PIONEER_XDJ-R1_PIONEER_XDJ-R1-00-XDJR1>
module: 5
properties:
alsa.resolution_bits = "24"
device.api = "alsa"
device.class = "sound"
alsa.class = "generic"
alsa.subclass = "generic-mix"
alsa.name = "USB Audio"
alsa.id = "USB Audio"
alsa.subdevice = "0"
alsa.subdevice_name = "subdevice #0"
alsa.device = "0"
alsa.card = "1"
alsa.card_name = "PIONEER XDJ-R1"
alsa.long_card_name = "Pioneer PIONEER XDJ-R1 at usb-0000:00:1d.1-2, full speed"
alsa.driver_name = "snd_usb_audio"
device.bus_path = "pci-0000:00:1d.1-usb-0:2:1.0"
sysfs.path = "/devices/pci0000:00/0000:00:1d.1/usb2/2-2/2-2:1.0/sound/card1"
udev.id = "usb-Pioneer_PIONEER_XDJ-R1_PIONEER_XDJ-R1-00-XDJR1"
device.bus = "usb"
device.vendor.id = "08e4"
device.vendor.name = "Pioneer Corp."
device.product.id = "0170"
device.product.name = "PIONEER XDJ-R1"
device.serial = "Pioneer_PIONEER_XDJ-R1_PIONEER_XDJ-R1"
device.string = "front:1"
device.buffering.buffer_size = "26448"
device.buffering.fragment_size = "6612"
device.access_mode = "mmap"
device.profile.name = "analog-stereo"
device.profile.description = "Analog Stereo"
device.description = "PIONEER XDJ-R1 Analog Stereo"
alsa.mixer_name = "USB Mixer"
alsa.components = "USB08e4:0170"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-usb"
ports:
analog-input: Analog Input (priority 10000, latency offset 0 usec, available: unknown)
properties:
active port: <analog-input>
What might I be missing? Why can PA figure the device out but ALSA can't?
Last edited by EvanPurkhiser (2014-04-16 09:27:13)

I'm interested in recording from the mixer over USB.
The device also supports playback into the mixer, but I assume this is designed for use in conjunction with using the device as a MIDI-Controller, which I don't have any intentions of doing.
What I've been doing to check the levels of the device is to play a track through the device (making sure that the VU meter for the channel is peaking at around 0db), record using `ffmpeg -f pulse -i 1 recording.wav` and then running `ffmpeg -i recording.wav -af "volumedetect" -f null /dev/null`
Which looks something like this:
[Parsed_volumedetect_0 @ 0x1f0eb80] n_samples: 10265206
[Parsed_volumedetect_0 @ 0x1f0eb80] mean_volume: -26.1 dB
[Parsed_volumedetect_0 @ 0x1f0eb80] max_volume: -12.0 dB
[Parsed_volumedetect_0 @ 0x1f0eb80] histogram_12db: 21
[Parsed_volumedetect_0 @ 0x1f0eb80] histogram_13db: 238
[Parsed_volumedetect_0 @ 0x1f0eb80] histogram_14db: 1619
[Parsed_volumedetect_0 @ 0x1f0eb80] histogram_15db: 7703
[Parsed_volumedetect_0 @ 0x1f0eb80] histogram_16db: 29370
I actually _can_ record directly from alsa, but I have to tell it the format, rate, and channels of the device.
[netbook/scratch/]> arecord -D hw:1 test.wav
Recording WAVE 'test.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono
arecord: set_params:1233: Sample format non available
Available formats:
- S24_3LE
[netbook/scratch/]> arecord -D hw:1 -f S24_3LE test.wav
Recording WAVE 'test.wav' : Signed 24 bit Little Endian in 3bytes, Rate 8000 Hz, Mono
arecord: set_params:1239: Channels count non available
[netbook/scratch/]> arecord -D hw:1 -f S24_3LE -c 2 test.wav
Recording WAVE 'test.wav' : Signed 24 bit Little Endian in 3bytes, Rate 8000 Hz, Stereo
Warning: rate is not accurate (requested = 8000Hz, got = 44100Hz)
please, try the plug plugin
[netbook/scratch/]> arecord -D hw:1 -f S24_3LE -c 2 -r 44100 test.wav
Recording WAVE 'test.wav' : Signed 24 bit Little Endian in 3bytes, Rate 44100 Hz, Stereo
The volume is still just as low as recording with PA though

Anyone got this problem solved? Duplicate titles on iPad's iPod?

Dear All,
Hope all reading are well! My problem seems to clear itself up - by itself -sometimes....
Problem is after syncing up then disconnecting iPad, duplicates of songs are seen. You know on the display of the iPod's usual display of your tunes.
Sometimes after resyncing the dupes go away or more are duplicated of different tunes!
Apple Support does not offer any advise. Please comment. These dupes bother me.
Thank you for your time and advice.
Sincerely,
MacOwner

Dear OrangeMarlin,
Much thanks in your reply. I must be more correct in describing my problem. I get these dupes but have no duplicates of anykind in my iTunes library. This is what is confusing me. The duplicated titles are right after one another and will play consecutively if I permit them.
I was talked through (by Apple Support) on finding duplicated songs but this did not solve anything because there were no duplicated songs/tunes in the library.
Thanks again for your input!
Much appreciated!
Sincerely,
MacOwner

[SOLVED] ALSA and intel soundcard - not detected, /dev/snd is empty

Hi, there was some topics mounths ago about this but I still can't figure out how to get my sound to work.
The main problem is that my soundcard is not detected and /dev/snd seems a little empty. I can't find the controlc0,1 files either....
Here is some useful info.
root@red red]# ls -l /sys/module/snd/holders/
total 0
lrwxrwxrwx 1 root root 0 Jun 15 17:09 snd_hda_codec -> ../../snd_hda_codec
lrwxrwxrwx 1 root root 0 Jun 15 17:09 snd_hda_intel -> ../../snd_hda_intel
lrwxrwxrwx 1 root root 0 Jun 15 17:09 snd_hwdep -> ../../snd_hwdep
lrwxrwxrwx 1 root root 0 Jun 15 17:09 snd_mixer_oss -> ../../snd_mixer_oss
lrwxrwxrwx 1 root root 0 Jun 15 17:09 snd_pcm -> ../../snd_pcm
lrwxrwxrwx 1 root root 0 Jun 15 17:09 snd_pcm_oss -> ../../snd_pcm_oss
lrwxrwxrwx 1 root root 0 Jun 15 17:09 snd_seq -> ../../snd_seq
lrwxrwxrwx 1 root root 0 Jun 15 17:09 snd_seq_device -> ../../snd_seq_device
lrwxrwxrwx 1 root root 0 Jun 15 17:09 snd_seq_oss -> ../../snd_seq_oss
lrwxrwxrwx 1 root root 0 Jun 15 17:09 snd_timer -> ../../snd_timer
[root@red red]# lsmod | grep '^snd' | column -t
snd_seq_dummy 1439 0
snd_seq_oss 28928 0
snd_seq_midi_event 5412 1 snd_seq_oss
snd_seq 50530 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi_event
snd_seq_device 5233 3 snd_seq_dummy,snd_seq_oss,snd_seq
snd_hda_intel 22514 0
snd_pcm_oss 39096 0
snd_hda_codec 68863 1 snd_hda_intel
snd_hwdep 6150 1 snd_hda_codec
snd_mixer_oss 16956 1 snd_pcm_oss
snd_pcm 70924 3 snd_hda_intel,snd_pcm_oss,snd_hda_codec
snd_timer 19684 2 snd_seq,snd_pcm
snd 57209 10 snd_seq_oss,snd_seq,snd_seq_device,snd_hda_intel,snd_pcm_oss,snd_hda_codec,snd_hwdep,snd_mixer_oss,snd_pcm,snd_timer
snd_page_alloc 7161 2 snd_hda_intel,snd_pcm
[root@red red]# ls -l /dev/snd/
total 0
crw-rw---- 1 root audio 116, 3 Jun 15 17:00 seq
crw-rw---- 1 root audio 116, 2 Jun 15 17:00 timer
Souncard : Don't know the exact name, the "intel hd" one with an ICH7 bridge
Mobo : intel d945gclf (little atom board, not the newer ones)
Notice that line : snd_hda_intel 22514 0
Seems to me that the module is loaded, am I right?
EDIT: Sorry, maybe I am in the wrong section...
Last edited by Lrouge (2010-06-18 18:33:54)

Looks pretty complete to me. (OK, /dev/snd does look sparse.) What do you see when you run alsamixer and press the F5 key?
$ sudo su -
# pacman -Sy alsa-utils
# alsamixer
Press F5 and check for:
* Master, Speaker, Front, Line, PCM volume down (if any)
* Any MM (bottom of vertical bar, signifies: line is muted) -- mute / un-mute with the 'm' key
Press Esc key to exit alsamixer + save changes.
Also, right after booting, run: dmesg | grep -i codec
Look for something like "Realtek" or "ALC662" or similar.
Manually add codec driver: modprobe snd_hda_codec_realtek
ls -al /dev/snd
drwxr-xr-x 2 root root 60 May 28 19:35 by-path
crw-rw---- 1 root audio 116, 7 May 28 19:35 controlC0
crw-rw---- 1 root audio 116, 6 May 28 19:35 hwC0D2
crw-rw---- 1 root audio 116, 5 May 28 19:35 pcmC0D0c
crw-rw---- 1 root audio 116, 4 May 28 19:35 pcmC0D0p
crw-rw---- 1 root audio 116, 3 May 28 19:35 seq
crw-rw---- 1 root audio 116, 2 May 28 19:35 timer
lsmod | grep snd
snd_hda_codec_realtek 179443 1
snd_seq_dummy 1099 0
snd_seq_oss 25304 0
snd_seq_midi_event 4452 1 snd_seq_oss
snd_hda_intel 18921 0
snd_seq 42628 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi_event
snd_seq_device 4313 3 snd_seq_dummy,snd_seq_oss,snd_seq
snd_hda_codec 56728 2 snd_hda_codec_realtek,snd_hda_intel
snd_pcm_oss 33693 0
snd_hwdep 5102 1 snd_hda_codec
snd_pcm 57479 3 snd_hda_intel,snd_hda_codec,snd_pcm_oss
snd_mixer_oss 14810 1 snd_pcm_oss
snd_timer 16117 2 snd_seq,snd_pcm
snd 43847 11 snd_hda_codec_realtek,snd_seq_oss,snd_hda_intel,snd_seq,snd_seq_device,snd_hda_codec,snd_pcm_oss,snd_hwdep,snd_pcm,snd_mixer_oss,snd_timer
soundcore 5007 1 snd
snd_page_alloc 5809 2 snd_hda_intel,snd_pcm

[SOLVED] ALSA firmware not loading properly with kernel 3.8.6-1

Hi All,
I have a E-MU 1820 card + breakout box that have been working beautiful under Archlinux for years but now something happen.
It still work OK under windows (dual boot same computer).
Normally during boot all indicators on the breakout box blink and then only thin show is clock setting indicator. But now all the indicators still blink but instead it is a -12 dB indicator that light up and not the clock settings and no sound.
dmesg show following.
[christer@Arch ~]$ dmesg | grep emu
[ 5.269612] emu1010: Special config.
[ 5.269731] emu1010: EMU_HANA_ID = 0x7f
[ 5.274685] emu1010: firmware file = emu/hana.fw, size = 0x133a4
[ 13.465727] emu1010: Hana Firmware loaded
[ 13.465776] emu1010: Hana version: 3.4
[ 13.465864] emu1010: Card options = 0x0
[ 13.465888] emu1010: Card options = 0x0
[ 13.466379] emu1010: Card options3 = 0x0
[ 14.486723] emu1010: Loading Audio Dock Firmware
[ 16.477785] emu1010: EMU_HANA+DOCK_IRQ_STATUS = 0x36
[ 16.477809] emu1010: EMU_HANA+DOCK_ID = 0x55
[ 16.477809] emu1010: Audio Dock Firmware loaded
[christer@Arch ~]$
aplay -l looks like it did before.
[christer@Arch ~]$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: EMU1010 [E-mu 1010 [MAEM8810]], device 0: emu10k1 [ADC Capture/Standard PCM Playback]
Subdevices: 32/32
Subdevice #0: subdevice #0
Subdevice #1: subdevice #1
Subdevice #2: subdevice #2
Subdevice #3: subdevice #3
Subdevice #4: subdevice #4
Subdevice #5: subdevice #5
Subdevice #6: subdevice #6
Subdevice #7: subdevice #7
Subdevice #8: subdevice #8
Subdevice #9: subdevice #9
Subdevice #10: subdevice #10
Subdevice #11: subdevice #11
Subdevice #12: subdevice #12
Subdevice #13: subdevice #13
Subdevice #14: subdevice #14
Subdevice #15: subdevice #15
Subdevice #16: subdevice #16
Subdevice #17: subdevice #17
Subdevice #18: subdevice #18
Subdevice #19: subdevice #19
Subdevice #20: subdevice #20
Subdevice #21: subdevice #21
Subdevice #22: subdevice #22
Subdevice #23: subdevice #23
Subdevice #24: subdevice #24
Subdevice #25: subdevice #25
Subdevice #26: subdevice #26
Subdevice #27: subdevice #27
Subdevice #28: subdevice #28
Subdevice #29: subdevice #29
Subdevice #30: subdevice #30
Subdevice #31: subdevice #31
card 0: EMU1010 [E-mu 1010 [MAEM8810]], device 2: emu10k1 efx [Multichannel Capture/PT Playback]
Subdevices: 8/8
Subdevice #0: subdevice #0
Subdevice #1: subdevice #1
Subdevice #2: subdevice #2
Subdevice #3: subdevice #3
Subdevice #4: subdevice #4
Subdevice #5: subdevice #5
Subdevice #6: subdevice #6
Subdevice #7: subdevice #7
card 0: EMU1010 [E-mu 1010 [MAEM8810]], device 3: emu10k1 [Multichannel Playback]
Subdevices: 1/1
Subdevice #0: subdevice #0
[christer@Arch ~]$
lsmod
[christer@Arch ~]$ lsmod | grep emu
snd_emu10k1 140245 1
snd_util_mem 2403 1 snd_emu10k1
snd_hwdep 6428 1 snd_emu10k1
snd_ac97_codec 113456 1 snd_emu10k1
snd_rawmidi 18831 1 snd_emu10k1
snd_seq_device 5268 5 snd_seq,snd_rawmidi,snd_seq_oss,snd_emu10k1,snd_seq_dummy
snd_pcm 78146 3 snd_pcm_oss,snd_ac97_codec,snd_emu10k1
snd_page_alloc 7426 2 snd_pcm,snd_emu10k1
snd_timer 18934 3 snd_pcm,snd_seq,snd_emu10k1
snd 60156 13 snd_pcm_oss,snd_ac97_codec,snd_hwdep,snd_timer,snd_pcm,snd_seq,snd_rawmidi,snd_seq_oss,snd_emu10k1,snd_seq_device,snd_mixer_oss
[christer@Arch ~]$
From what I can see all looks OK, like it did when it worked, but something is not at it should.
Any thoughts or ideas about what might have happen or how to progress in troubleshooting.
All input most welcome.
I found info describing problem with firmware loading, but it is over my head.
http://git.kernel.org/cgit/linux/kernel … 54cf7008ab
Output from Aadebug ( http://alsa.opensrc.org/Aadebug )
[christer@Arch ~]$ ./alsadebug
ALSA Audio Debug v0.2.0 - Fri Apr 12 20:46:05 CEST 2013
http://alsa.opensrc.org/aadebug
http://www.gnu.org/licenses/agpl-3.0.txt
Kernel ----------------------------------------------------
Linux Arch 3.8.6-1-ARCH #1 SMP PREEMPT Sat Apr 6 07:27:01 CEST 2013 x86_64 GNU/Linux
Advanced Linux Sound Architecture Driver Version k3.8.6-1-ARCH.
Loaded Modules --------------------------------------------
snd_emu10k1 139271 2
snd_util_mem 2339 1 snd_emu10k1
snd_hwdep 6364 1 snd_emu10k1
snd_pcm_oss 38511 0
snd_mixer_oss 14995 1 snd_pcm_oss
snd_ac97_codec 112216 1 snd_emu10k1
snd_seq_dummy 1463 0
snd_seq_oss 29098 0
snd_seq_midi_event 5660 1 snd_seq_oss
snd_rawmidi 18742 1 snd_emu10k1
snd_seq 49946 5 snd_seq_midi_event,snd_seq_oss,snd_seq_dummy
snd_seq_device 5180 5 snd_seq,snd_rawmidi,snd_seq_oss,snd_emu10k1,snd_seq_dummy
snd_pcm 76956 4 snd_pcm_oss,snd_ac97_codec,snd_emu10k1
snd_page_alloc 7298 2 snd_pcm,snd_emu10k1
snd_timer 18687 3 snd_pcm,snd_seq,snd_emu10k1
snd 58893 14 snd_pcm_oss,snd_ac97_codec,snd_hwdep,snd_timer,snd_pcm,snd_seq,snd_rawmidi,snd_seq_oss,snd_emu10k1,snd_seq_device,snd_mixer_oss
Proc Asound -----------------------------------------------
0 [EMU1010 ]: Audigy2 - E-mu 1010 [MAEM8810]
E-mu 1010 [MAEM8810] (rev.3, serial:0x40011102) at 0xd000, irq 19
1: : sequencer
2: [ 0- 0]: hardware dependent
3: [ 0- 1]: raw midi
4: [ 0- 0]: raw midi
5: [ 0- 3]: digital audio playback
6: [ 0- 2]: digital audio playback
7: [ 0- 2]: digital audio capture
8: [ 0- 1]: digital audio capture
9: [ 0- 0]: digital audio playback
10: [ 0- 0]: digital audio capture
11: [ 0] : control
33: : timer
00-00: EMU10K1 (FX8010)
00-00: emu10k1 : ADC Capture/Standard PCM Playback : playback 32 : capture 1
00-01: emu10k1 mic : Mic Capture : capture 1
00-02: emu10k1 efx : Multichannel Capture/PT Playback : playback 8 : capture 1
00-03: emu10k1 : Multichannel Playback : playback 1
Client info
cur clients : 3
peak clients : 3
max clients : 192
Client 0 : "System" [Kernel]
Port 0 : "Timer" (Rwe-)
Port 1 : "Announce" (R-e-)
Connecting To: 15:0
Client 14 : "Midi Through" [Kernel]
Port 0 : "Midi Through Port-0" (RWe-)
Client 15 : "OSS sequencer" [Kernel]
Port 0 : "Receiver" (-we-)
Connected From: 0:1
Client 16 : "E-mu 1010 [MAEM8810]" [Kernel]
Port 0 : "Audigy MPU-401 (UART)" (RWeX)
Port 32 : "Audigy MPU-401 #2" (RWeX)
Client 17 : "Emu10k1 WaveTable" [Kernel]
Port 0 : "Emu10k1 Port 0" (-We-)
Port 1 : "Emu10k1 Port 1" (-We-)
Port 2 : "Emu10k1 Port 2" (-We-)
Port 3 : "Emu10k1 Port 3" (-We-)
Dev Snd ---------------------------------------------------
total 0
drwxr-xr-x 2 root root 60 Apr 12 20:42 by-path
crw-rw----+ 1 root audio 116, 11 Apr 12 20:42 controlC0
crw-rw----+ 1 root audio 116, 2 Apr 12 20:42 hwC0D0
crw-rw----+ 1 root audio 116, 12 Apr 12 20:46 hwC0D2
crw-rw----+ 1 root audio 116, 4 Apr 12 20:42 midiC0D0
crw-rw----+ 1 root audio 116, 3 Apr 12 20:42 midiC0D1
crw-rw----+ 1 root audio 116, 13 Apr 12 20:46 midiC0D2
crw-rw----+ 1 root audio 116, 14 Apr 12 20:46 midiC0D3
crw-rw----+ 1 root audio 116, 10 Apr 12 20:42 pcmC0D0c
crw-rw----+ 1 root audio 116, 9 Apr 12 20:43 pcmC0D0p
crw-rw----+ 1 root audio 116, 8 Apr 12 20:42 pcmC0D1c
crw-rw----+ 1 root audio 116, 7 Apr 12 20:42 pcmC0D2c
crw-rw----+ 1 root audio 116, 6 Apr 12 20:42 pcmC0D2p
crw-rw----+ 1 root audio 116, 5 Apr 12 20:42 pcmC0D3p
crw-rw----+ 1 root audio 116, 1 Apr 12 20:42 seq
crw-rw----+ 1 root audio 116, 33 Apr 12 20:42 timer
CPU -------------------------------------------------------
model name : Intel(R) Core(TM)2 Duo CPU E8400 @ 3.00GHz
cpu MHz : 2000.000
model name : Intel(R) Core(TM)2 Duo CPU E8400 @ 3.00GHz
cpu MHz : 2000.000
RAM -------------------------------------------------------
MemTotal: 4051696 kB
SwapTotal: 51196 kB
Hardware --------------------------------------------------
01:00.0 VGA compatible controller: NVIDIA Corporation G94 [GeForce 9600 GT] (rev a1)
05:01.0 Multimedia audio controller: Creative Labs SB Audigy (rev 03)
Interupts -------------------------------------------------
CPU0 CPU1
0: 48 0 IO-APIC-edge timer
1: 1 1 IO-APIC-edge i8042
6: 1 2 IO-APIC-edge floppy
8: 0 1 IO-APIC-edge rtc0
9: 0 0 IO-APIC-fasteoi acpi
12: 4 1 IO-APIC-edge i8042
16: 529 493 IO-APIC-fasteoi uhci_hcd:usb1, pata_jmicron, nvidia
18: 10654 10645 IO-APIC-fasteoi uhci_hcd:usb3, ehci_hcd:usb4, uhci_hcd:usb8, firewire_ohci, i801_smbus
19: 3621 3626 IO-APIC-fasteoi uhci_hcd:usb7, snd_emu10k1
21: 0 0 IO-APIC-fasteoi uhci_hcd:usb2
23: 0 0 IO-APIC-fasteoi ehci_hcd:usb5, uhci_hcd:usb6
44: 7637 7652 PCI-MSI-edge ahci
45: 11719 11736 PCI-MSI-edge eth0
NMI: 47 47 Non-maskable interrupts
LOC: 49551 45081 Local timer interrupts
SPU: 0 0 Spurious interrupts
PMI: 47 47 Performance monitoring interrupts
IWI: 0 0 IRQ work interrupts
RTR: 0 0 APIC ICR read retries
RES: 28730 28313 Rescheduling interrupts
CAL: 94 54 Function call interrupts
TLB: 3503 3509 TLB shootdowns
TRM: 0 0 Thermal event interrupts
THR: 0 0 Threshold APIC interrupts
MCE: 0 0 Machine check exceptions
MCP: 1 1 Machine check polls
ERR: 0
MIS: 0
[christer@Arch ~]$
output from systemctl show somw inacive sound related sevices.
alsa-restore.service loaded inactive dead Restore Sound Card State
alsa-store.service loaded inactive dead Store Sound Card State
auditd.service error inactive dead auditd.service
display-manager.service error inactive dead display-manager.service
dmeventd.service loaded inactive dead Device-mapper event daemon
emergency.service loaded inactive dead Emergency Shell
lvmetad.service loaded inactive dead LVM2 metadata daemon
plymouth-quit-wait.service error inactive dead plymouth-quit-wait.service
plymouth-start.service error inactive dead plymouth-start.service
rc-local-shutdown.service loaded inactive dead /etc/rc.local.shutdown Compatibility
rescue.service loaded inactive dead Rescue Shell
systemd-...console.service loaded inactive dead Dispatch Password Requests to Console
systemd-...rd-wall.service loaded inactive dead Forward Password Requests to Wall
systemd-fsck-root.service loaded inactive dead File System Check on Root Device
systemd-...e8ee844.service loaded inactive dead File System Check on /dev/disk/by-uuid/9751ee37-e769-4bbe-9436-2d8e5e8ee844
systemd-...ev-sda2.service loaded inactive dead File System Check on /dev/sda2
systemd-...ev-sda4.service loaded inactive dead File System Check on /dev/sda4
systemd-initctl.service loaded inactive dead /dev/initctl Compatibility Daemon
systemd-...l-flush.service loaded inactive dead Trigger Flushing of Journal to Persistent Storage
systemd-...ed-load.service loaded inactive dead Load Random Seed
systemd-...ed-save.service loaded inactive dead Save Random Seed
systemd-...ad-done.service loaded inactive dead Stop Read-Ahead Data Collection
systemd-shutdownd.service loaded inactive dead Delayed Shutdown Service
systemd-...s-clean.service loaded inactive dead Cleanup of Temporary Directories
systemd-...unlevel.service loaded inactive dead Update UTMP about System Runlevel Changes
systemd-...hutdown.service loaded inactive dead Update UTMP about System Shutdown
ypbind.service error inactive dead ypbind.service
emergency.target loaded inactive dead Emergency Mode
final.target loaded inactive dead Final Step
nss-lookup.target loaded inactive dead Host and Network Name Lookups
nss-user-lookup.target loaded inactive dead User and Group Name Lookups
remote-fs-pre.target loaded inactive dead Remote File Systems (Pre)
remote-fs-setup.target error inactive dead remote-fs-setup.target
rescue.target loaded inactive dead Rescue Mode
runlevel1.target error inactive dead runlevel1.target
runlevel2.target error inactive dead runlevel2.target
runlevel3.target error inactive dead runlevel3.target
runlevel4.target error inactive dead runlevel4.target
runlevel5.target error inactive dead runlevel5.target
shutdown.target loaded inactive dead Shutdown
syslog.target error inactive dead syslog.target
umount.target loaded inactive dead Unmount All Filesystems
All the best!
Christer
Last edited by agkbill (2013-04-15 18:41:03)

Thank you mich41!
It looks like a kernel bug.
I downgraded to 3.7.10-1 then firmware loaded OK and I could play sounds.
Upgraded to 3.8.6-1 then the firmware fail to load.
Something looks not ok with 3.8.6-1.
Best regards,
Christer

[solved] alsa: duplicate front signal

Similar Messages

Maybe you are looking for