BASICS - PulseAudio - No Sound

Hello!
I have a fresh Arch installation and everything works fine, but I lost my sound.
That's right, it worked at first but now it doesn't. Didn't change anything regarding sound packages, only had a reboot inbetween.
pavucontrol shows me all my devices (I have an USB DAC in use) and the mixer is unmuted.
The volume meters/bars are also acting when sound is played back, but it just won't reach my headphones, it's total silence.
I don't know where to look anymore but will provide any information needed to solve this.
Really want to keep using Arch again.
Cheers!
Last edited by detto (2014-10-30 20:01:09)

I'm back. My sound doesn't work anymore.
It's not Arch specific though as a fresh installation of Lubuntu has the exact same problem.
I can play an audio file with
aplay -D plughw:1,0 /usr/share/sounds/alsa/Front_Center.wav
and I hear distorted crap.
Replugging the USB port doesn't help anymore, switching USB port doesn't help either.
It is something related to alsa (I GUESS!) because Lubuntu has the same problem withouth an installed pulseaudio package.
Can anybody help me out fix this?

Similar Messages

Pavucontrol/Pulseaudio hangs sound playback

Hello everyone
I've posted on this before, and the bug went away, but now it's back and I just wish it'd go away permanently. For reasons I can't explain, trying to control the volume with pavucontrol (or, it seems, any volume control that interacts with pulseaudio) suspends sound playback. And then pavucontrol crashes with a callback timeout failure error.
I've followed all the various guides on setting up Pulseaudio. I've experimented with setting sinks, with setting sound outputs, everything. But time and again the pavucontrol applet brings the sound to a halt. As far as I know I've installed everything that might be needed. I used to like Pulseaudio, the whole control-volume-of-individual-streams is great. But it just doesn't work. And yet it's one of those features, which once you get used to it, is really hard to live without.
How can I fix this once and for all?
EDIT: Reading /var/log/errors.log I've noticed that Pulseaudio is failing to set hardware parameters with a connection timeout.

Have you tried setting the pulse volume from alsa (amixer)? You should really bring this to the pulse mailing list, since this sort of bug is probably hardware/driver level and not very simple to debug.

[Pulseaudio] Pop sound when the first stream starts, auto-suspend?

Hi,
I've been having a popping sound issues whenever a Pulseaudio sink was opened. At first I thought it had a link with the power saving settings of snd_hda_intel, so I changed those two settings:
power_save setting = 0
power_save_controller = N
Sadly, it doesn't change anything.
I also tried to disable auto-mute in alsamixer. It didn't work.
Then, I tried to investigate on pulseaudio's side, and figured the sinks are disabled when they are idle, so I commented out this statement in /etc/pulse/default.pa:
### Automatically suspend sinks/sources that become idle for too long
#load-module module-suspend-on-idle
However, if I do that, the device will still be suspended when the last stream ends, but won't come back to life except if I select the device again in pavucontrol while a sound stream is running (super weird).
Any leads? I'm a bit desperate...
EDIT: I'm experiencing what's been described here: Red Hat - Bug #520403 - but the solution they proposed does not work for me. I also followed the guide for ALSA Setup on the wiki, and read the section talking about Popping Noises, but nothing works for me, I un-muted the Line-In too. FYI, my audio codec is a Realtek ALC892.
EDIT 2: Noticed it does the same thing when I mute the volume.
Thanks!
Gabriel
Last edited by gferon (2012-05-16 17:02:06)

emeres wrote:
Right, an array of unsigned integers to be exact. You would have to look at the code, either in your sources or here and here. They may have changed somewhat, but I doubt any drastic changes. First link is to linux 3.14 version. At least I assume it gets passed on directly, if that really is the case, I do not know. Just try it, hardware damage risk is minimal.
Looking at the mask, I think it takes up to 31 samples. So try with 1, 2, 4, 8, 16, 24. I may be very well wrong about this.
Since I already have 0 and 1 in the 'trial matrix' I'll add 24 only. Adding the combinations with 2, 4, 8, 16, 24 is too much for me
emeres wrote:Why? Modprobe should not care about hyphen or underscore. They should be always converted when parsing. On most keyboard layouts (also on the Polish programmer one) hyphen is one key one keystroke, where underscore is two key one keystroke, therefore more error prone and less efficient.
Wow, I didn't know that! What do you know! Linux being bitchy even about capital letters made me think it would be even more bitchy about '_' vs '-'.

Pulseaudio no sound

Followed the wiki and started daemon, used the fuser commands to check that nothing was using sound first and restarted daemon after closing all apps but no sound.
Here is my /etc/asound.conf:
# Use Pulseaudio by default
pcm.pulse {
type pulse
ctl.pulse {
type pulse
# 6 channel dmix:
pcm.dmix6 {
type dmix
ipc_key 1024
ipc_key_add_uid false
ipc_perm 0666
slave {
pcm "hw:0,0"
rate 48000
format "S32_LE"
channels 6
period_time 0
period_size 1024
buffer_time 0
buffer_size 8192
# upmixing:
pcm.ch51dup {
type route
slave.pcm dmix6
slave.channels 6
ttable.0.0 1
ttable.1.1 1
ttable.0.2 0.6
ttable.1.3 0.6
ttable.0.4 0.5
ttable.1.4 0.5
ttable.0.5 0.5
ttable.1.5 0.5
pcm.duplex {
type asym
playback.pcm "ch51dup" # upmix first
capture.pcm "hw:0"
# change default device:
#pcm.!default {
# type plug
# slave {
# pcm "duplex"
# for aoss
pcm.dsp "duplex"
pcm.dsp1 "duplex"
pcm.10to20 {
type route
slave.pcm default
slave.channels 2
ttable.0.0 1
ttable.0.1 1
pcm.headset {
type bluetooth
profile "voice"
ctl.headset {
type bluetooth
profile "voice"
ctl.equal {
type equal;
#pcm.plugequal {
# type equal;
# Modify the line below if you don't
# want to use sound card 0.
#slave.pcm "plughw:0,0";
#by default we want to play from more sources at time:
# slave.pcm "plug:dmix6";
#pcm.equal {
# Or if you want the equalizer to be your
# default soundcard uncomment the following
# line and comment the above line.
#pcm.!default {
# type plug;
# slave.pcm plugequal;
#------------------------end of working file
So my setup works like this: I use plug:dmix6 for anything that is 5.1, while default upmixes stereo to all speakers for anything else (music, 2channel video, youtube, etc.). It all works great and has for a long time, but now I started using XBMC and I need to be able to switch between using dmix6 for movies/shows, and default for Music.
I have tried various video/audio files with mplayer using -ao alsa:device=pulse but get no sound.
Here is end of /var/log/messages.log if it helps:
Jan 18 14:51:30 localhost pulseaudio[7793]: sink-input.c: application.language = "C"
Jan 18 14:51:30 localhost pulseaudio[7793]: sink-input.c: window.x11.display = ":0"
Jan 18 14:51:30 localhost pulseaudio[7793]: sink-input.c: application.process.machine_id = "f4cc2664f136cd7500a22d7d00000367"
Jan 18 14:51:30 localhost pulseaudio[7793]: sink-input.c: application.process.session_id = "f4cc2664f136cd7500a22d7d00000367-1295238838.187447-631589587"
Jan 18 14:51:30 localhost pulseaudio[7793]: sink-input.c: module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [mplayer]"
Jan 18 14:51:30 localhost pulseaudio[7793]: protocol-native.c: Requested tlength=500.00 ms, minreq=31.25 ms
Jan 18 14:51:30 localhost pulseaudio[7793]: protocol-native.c: Final latency 531.25 ms = 437.50 ms + 2*31.25 ms + 31.25 ms
Jan 18 14:51:36 localhost pulseaudio[7793]: sap.c: sendmsg() failed: Connection refused
Jan 18 14:51:39 localhost pulseaudio[7793]: module-device-restore.c: Storing volume/mute/port for device sink:alsa_output.pci-0000_05_01.0.analog-surround-51.
Jan 18 14:51:39 localhost pulseaudio[7793]: module-device-restore.c: Storing volume/mute/port for device sink:alsa_output.pci-0000_05_01.0.analog-surround-51.
Jan 18 14:51:39 localhost pulseaudio[7793]: module-device-restore.c: Storing volume/mute/port for device sink:alsa_output.pci-0000_05_01.0.analog-surround-51.
Jan 18 14:51:39 localhost pulseaudio[7793]: module-device-restore.c: Storing volume/mute/port for device sink:alsa_output.pci-0000_05_01.0.analog-surround-51.
Jan 18 14:51:40 localhost pulseaudio[7793]: module-device-restore.c: Storing volume/mute/port for device sink:alsa_output.pci-0000_05_01.0.analog-surround-51.
Jan 18 14:51:40 localhost pulseaudio[7793]: module-device-restore.c: Storing volume/mute/port for device sink:alsa_output.pci-0000_05_01.0.analog-surround-51.
Jan 18 14:51:40 localhost pulseaudio[7793]: module-device-restore.c: Storing volume/mute/port for device sink:alsa_output.pci-0000_05_01.0.analog-surround-51.
Jan 18 14:51:40 localhost pulseaudio[7793]: module-device-restore.c: Storing volume/mute/port for device sink:alsa_output.pci-0000_05_01.0.analog-surround-51.
Jan 18 14:51:41 localhost pulseaudio[7793]: sink-input.c: Freeing input 23 "ALSA Playback"
Jan 18 14:51:41 localhost pulseaudio[7793]: client.c: Freed 12 "ALSA plug-in [mplayer]"
Jan 18 14:51:41 localhost pulseaudio[7793]: protocol-native.c: Connection died.
Jan 18 14:51:46 localhost pulseaudio[7793]: sap.c: sendmsg() failed: Connection refused
Jan 18 14:51:46 localhost pulseaudio[7793]: module-suspend-on-idle.c: Sink rtp idle for too long, suspending ...
Jan 18 14:51:49 localhost pulseaudio[7793]: module-device-restore.c: Synced.
Jan 18 14:51:56 localhost pulseaudio[7793]: sap.c: sendmsg() failed: Connection refused
Jan 18 14:51:56 localhost pulseaudio[7793]: client.c: Created 13 "Native client (UNIX socket client)"
Jan 18 14:51:56 localhost pulseaudio[7793]: protocol-native.c: Got credentials: uid=1000 gid=100 success=1
Jan 18 14:51:56 localhost pulseaudio[7793]: client.c: Freed 13 "ALSA plug-in [amixer]"
Jan 18 14:51:56 localhost pulseaudio[7793]: protocol-native.c: Connection died.
Jan 18 14:52:06 localhost pulseaudio[7793]: sap.c: sendmsg() failed: Connection refused
Jan 18 14:52:16 localhost pulseaudio[7793]: sap.c: sendmsg() failed: Connection refused
Jan 18 14:52:26 localhost pulseaudio[7793]: sap.c: sendmsg() failed: Connection refused
Thanks for any help!
Last edited by colbert (2011-01-18 19:58:42)

Well; complete the circle and mark your thread as solved so people don't need to click to find out you fixed it

With Pulseaudio, no sound with other apps when jack is running

I need to be able to run other sound apps with Jack running (and, ideally, be able to route sound from those apps into Ardour via Jack). So I installed PulseAudio, along with pulseaudio-alsa, and blacklisted the snd_pcm_oss module as per the Arch wiki. At that point I logged out and back in and I still had sound in all my apps. I then set up pulseaudio over jack as per the Arch wiki instructions here:
https://wiki.archlinux.org/index.php/Pu … rough_JACK
including the scripts for qjackctl. I now have sound only in my apps that use Jack; Ardour, for instance. Can anyone help? I'm using KDE 4.6 with the Phonon-VLC backend for sound.

Do some debugging first, please. The wiki instructions are really quite hackish (but there's no real good way to automate this as of now).
Kill pulseaudio, start jack (till here you're working fine I see), restart pulseaudio, try and load the jack modules and check qjackctl or pavucontrol to see if they're connected. If not, find the error logs and google them. Then come back here and ask more.

Pulseaudio delayed sound issues with simultaneous output

I've got Pulseaudio set up to send simulataneous output to analog speakers and HDMI monitor-internal speakers. However, frequently when no audio has been playing for a while, when I do something involving audio, the simulataneous output is out of sync (one of the two output is receiving delayed signal) and so it sounds odd, echo-y, and awful. I usually have to kill pulseaudio and restart to get it back working.
It's not always been like this; the issue seemed to have started a couple of months ago.
Any suggestions on a fix?
my default.pa settings are below:
#!/usr/bin/pulseaudio -nF
# This file is part of PulseAudio.
# PulseAudio is free software; you can redistribute it and/or modify it
# under the terms of the GNU Lesser General Public License as published by
# the Free Software Foundation; either version 2 of the License, or
# (at your option) any later version.
# PulseAudio is distributed in the hope that it will be useful, but
# WITHOUT ANY WARRANTY; without even the implied warranty of
# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
# General Public License for more details.
# You should have received a copy of the GNU Lesser General Public License
# along with PulseAudio; if not, write to the Free Software Foundation,
# Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
# This startup script is used only if PulseAudio is started per-user
# (i.e. not in system mode)
.nofail
### Load something into the sample cache
#load-sample-lazy x11-bell /usr/share/sounds/gtk-events/activate.wav
#load-sample-lazy pulse-hotplug /usr/share/sounds/startup3.wav
#load-sample-lazy pulse-coldplug /usr/share/sounds/startup3.wav
#load-sample-lazy pulse-access /usr/share/sounds/generic.wav
.fail
### Automatically restore the volume of streams and devices
load-module module-device-restore
load-module module-stream-restore
load-module module-card-restore
### Automatically augment property information from .desktop files
### stored in /usr/share/application
load-module module-augment-properties
### Should be after module-*-restore but before module-*-detect
load-module module-switch-on-port-available
### Load audio drivers statically
### (it's probably better to not load these drivers manually, but instead
### use module-udev-detect -- see below -- for doing this automatically)
#load-module module-alsa-sink
#load-module module-alsa-source device=hw:1,0
#load-module module-oss device="/dev/dsp" sink_name=output source_name=input
#load-module module-oss-mmap device="/dev/dsp" sink_name=output source_name=input
#load-module module-null-sink
#load-module module-pipe-sink
load-module module-alsa-sink device=hw:1,9
### 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
### Automatically connect sink and source if JACK server is present
.ifexists module-jackdbus-detect.so
.nofail
load-module module-jackdbus-detect channels=2
.fail
.endif
### Automatically load driver modules for Bluetooth hardware
.ifexists module-bluetooth-policy.so
load-module module-bluetooth-policy
.endif
.ifexists module-bluetooth-discover.so
load-module module-bluetooth-discover
.endif
### Load several protocols
.ifexists module-esound-protocol-unix.so
load-module module-esound-protocol-unix
.endif
load-module module-native-protocol-unix
### Network access (may be configured with paprefs, so leave this commented
### here if you plan to use paprefs)
#load-module module-esound-protocol-tcp
#load-module module-native-protocol-tcp
#load-module module-zeroconf-publish
### Load the RTP receiver module (also configured via paprefs, see above)
#load-module module-rtp-recv
### Load the RTP sender module (also configured via paprefs, see above)
#load-module module-null-sink sink_name=rtp format=s16be channels=2 rate=44100 sink_properties="device.description='RTP Multicast Sink'"
#load-module module-rtp-send source=rtp.monitor
### Load additional modules from GConf settings. This can be configured with the paprefs tool.
### Please keep in mind that the modules configured by paprefs might conflict with manually
### loaded modules.
.ifexists module-gconf.so
.nofail
load-module module-gconf
.fail
.endif
### Automatically restore the default sink/source when changed by the user
### during runtime
### NOTE: This should be loaded as early as possible so that subsequent modules
### that look up the default sink/source get the right value
load-module module-default-device-restore
### Automatically move streams to the default sink if the sink they are
### connected to dies, similar for sources
load-module module-rescue-streams
### Make sure we always have a sink around, even if it is a null sink.
load-module module-always-sink
### Honour intended role device property
load-module module-intended-roles
### Automatically suspend sinks/sources that become idle for too long
load-module module-suspend-on-idle
### If autoexit on idle is enabled we want to make sure we only quit
### when no local session needs us anymore.
.ifexists module-console-kit.so
load-module module-console-kit
.endif
.ifexists module-systemd-login.so
load-module module-systemd-login
.endif
### Enable positioned event sounds
load-module module-position-event-sounds
### Cork music/video streams when a phone stream is active
load-module module-role-cork
### Modules to allow autoloading of filters (such as echo cancellation)
### on demand. module-filter-heuristics tries to determine what filters
### make sense, and module-filter-apply does the heavy-lifting of
### loading modules and rerouting streams.
load-module module-filter-heuristics
load-module module-filter-apply
# X11 modules should not be started from default.pa so that one daemon
# can be shared by multiple sessions.
### Load X11 bell module
#load-module module-x11-bell sample=bell-windowing-system
### Register ourselves in the X11 session manager
#load-module module-x11-xsmp
### Publish connection data in the X11 root window
#.ifexists module-x11-publish.so
#.nofail
#load-module module-x11-publish
#.fail
#.endif
### shell commands ?
load-module module-native-protocol-tcp auth-ip-acl=127.0.0.1;192.168.0.0/24
### Make some devices default
#set-default-sink output
#set-default-source input
set-default-sink 2
Last edited by emacsomancer (2015-05-23 23:18:34)

if you open up 'pavucontrol' play around with the latency offset value for either output path

PulseAudio stops sounding when I switch to other display (Ctrl+Alt+F1)

Hello.
When I am playing any sound and I switch to a display (Ctrl+Alt+F1, Ctrl+Alt+F2...) where I have not loged in or a am not loged with my default user account, the sound stops, and when I switch back, I continues sounding.
Anyone have an idea of what could be causing this?
Thanks.

Thanks for the reply.
Is there a way to change that behavior?
Edit: After killing console-kit-daemon, sound works on all displays, but I think that it is not the best solution.
Last edited by eduardosm (2011-12-02 21:29:57)

External HDD Stor E Basics Strange clicking sound once in a while...

Hi...
Well the title says it all.. but... when I wrote a folder with files on it and tried to open it says That the file location on HDD is corrupted and uinreadable, but when i try to delete this folder it gives me error like 0x80070091 The directory is not empty... I just bought the hard drive today and didnt drop it... And when i listen to music from the HDD and i hear the clicking sound, but the music plays normally.. what'swrong??

A clicking HD is never a good sign!
http://www.seagate.com/support/downloads/item/seatools-win-master/
When I Google your error code, it takes me here. Sence I have never had this error code,it's the best I can do for you.http://answers.microsoft.com/en-us/windows/forum/windows_vista-files/deleting-a-corrupt-folder-error...
Dokie!!
I Love my Satellite L775D-S7222 Laptop. Some days you're the windshield, Some days you're the bug. The Computer world is crazy. If you have answers to computer problems, pass them forward.

Iphone screen is basically black, but sound still works aswell as the light beside the camera

Hi. I have an iphone 4s and the screen is barely readable. it is nearly completely black. The light beside the camera seems to come on every time that I turn on the phone but yet the screen doesn't seem to come on. i have restored the iphone and done everything it says in other forums but yet nothing is working. it is on the set up screen ready to set up as a new iphone but it is so hard to see everything that i cannot set it up. im planning on bringing it back to vodafone tomorrow to see if they can do anything. Any suggestions on what to do?

Sounds like it may be a hardware issue.
Did the issue happen over time or did something cause it (dropped it, water, etc.)

[Solved] Sound with ALSA and PulseAudio at the same time?

Hi, I am using arch linux to play Counter Strike on.
While it works fine and all, I just don't get any sound in CS because I am using Mangler, which only seems to work with my microphone if I use PulseAudio.
I hope this is a rather simple issue, basicly, I just want to enable sound in PulseAudio and ALSA at the same time, is this possible?
Last edited by Paaskehare (2010-08-12 15:39:09)

I have read that PulseAudio decreases sound quality, and therefore I would not like to use this as a systemwide driver, but rather only where it is needed.
Please correct me if I am wrong
Edit:
Ok, sound quality isn't noticeably worse, so this will do. HOWEVER, while I am in Counter Strike my sound is delayed by ~1 second, and it is therefore useless, any thoughts on that matter?
Edit again .. Solved this time:
I set my ALSA plugin in winecfg to "Hardware Acceleration: Emulation" - that seemed to do the trick.
Last edited by Paaskehare (2010-08-12 15:38:56)

No TV sound - Problem with Pulseaudio and Gnome

Hi Community!
Since January 2013 I got big problems with the sound of my tvcard. The card is a Sundtek Media TV Pro. But after all the problems have nothing to to with the card itself!
So far I found out, that the problem is with Pulseaudio.
- Sound works under Windows on same PC.
- Sound don't work under Arch - not with tvtime, not with vlc, etc.
- Problem occurs even on a new installed system with the Arch-2013-02-01 install medium.
cat /etc/mediasrv.log
[1033] Pulseaudio connect reported an error (6 - Connection refused)
[1033] select error() in audio processor (can be caused because of pulseaudio) (Interrupted system call)
After setting up a new Arch installation with just xorg-server, gnome, tvtime and the sundtek driver from AUR I run journalctl -b:
Feb 10 16:25:15 hightower pulseaudio[636]: [pulseaudio] pid.c:
Daemon already running.
With sudo journalctl | grep 'Feb 10 16:2' I get the following output:
Feb 10 16:25:01 hightower rtkit-daemon[489]: Successfully made thread 488 of process 488 (/usr/bin/pulseaudio) owned by '120' high priority at nice level -11.
Feb 10 16:25:14 hightower rtkit-daemon[489]: Successfully made thread 606 of process 606 (/usr/bin/pulseaudio) owned by '1000' high priority at nice level -11.
Feb 10 16:25:15 hightower rtkit-daemon[489]: Successfully made thread 636 of process 636 (/usr/bin/pulseaudio) owned by '1000' high priority at nice level -11.
Feb 10 16:25:15 hightower pulseaudio[636]: [pulseaudio] pid.c: Daemon already running.
Feb 10 16:28:55 hightower pulseaudio[1000]: [pulseaudio] module-jackdbus-detect.c: Unable to contact D-Bus session bus: org.freedesktop.DBus.Error.NotSupported: Unable to autolaunch a dbus-daemon without a $DISPLAY for X11
Feb 10 16:28:55 hightower pulseaudio[1000]: [pulseaudio] module.c: Failed to load module "module-jackdbus-detect" (argument:""): initialization failed.
Feb 10 16:28:55 hightower pulseaudio[1000]: [pulseaudio] socket-server.c: bind(): Die Adresse wird bereits verwendet
Feb 10 16:28:55 hightower pulseaudio[1000]: [pulseaudio] module.c: Failed to load module "module-esound-protocol-unix" (argument: ""): initialization failed.
Feb 10 16:28:55 hightower pulseaudio[1000]: [pulseaudio] main.c: Module load failed.
Feb 10 16:28:55 hightower pulseaudio[1000]: [pulseaudio] main.c: Module load failed.
Feb 10 16:28:55 hightower pulseaudio[1000]: [pulseaudio] main.c: Konnte Daemon nicht initialisieren.
Feb 10 16:28:55 hightower pulseaudio[997]: [pulseaudio] main.c: Start des Daemons fehlgeschlagen.
So, if I understand right, pulseaudio is alreade started by the rtkit-daemon and gnome couldn't start it again.
At https://bugs.archlinux.org/task/30926 I found a hint concerning /etc/xdg/autostart/pulseaudio.desktop and /etc/xdg/autostart/pulseaudio-kde.desktop.
After removing them from /etc/xdg/autostart and running journalctl -b there are no more errors.
With sudo journalctl | grep 'pulseaudio' I get:
sudo journalctl | grep 'Feb 10 17:4' | grep 'pulse*'
Feb 10 17:41:42 hightower rtkit-daemon[485]: Successfully made thread 484 of process 484 (/usr/bin/pulseaudio) owned by '120' high priority at nice level -11.
Feb 10 17:41:55 hightower rtkit-daemon[485]: Successfully made thread 625 of process 625 (/usr/bin/pulseaudio) owned by '1000' high priority at nice level -11.
In my opinion all looks ok! I also checked the channels with the alsamixer and it looks good, too.
So I ran vlc v4l2://:dev=/dev/video0:width=720:height=576:tuner-frequency=510750:adev=/dev/null:standard=255 - but still no sound.
With cat /var/log/mediasrv.log I checked the drivers logfile:
2013-02-10 17:48:55 [343] SETTING DVB-C
2013-02-10 17:48:55 [343] switching to analog TV
2013-02-10 17:48:56 [343] SETTING PAL/SECAM
2013-02-10 17:48:58 [343] invalid audio input selected: 2
2013-02-10 17:48:58 [343] Setting frequency: 510750000
2013-02-10 17:48:58 [343] Using settings for Europe
2013-02-10 17:48:58 [343] Set Pixelformat: 0 YUV 4:2:2 (YUYV)
2013-02-10 17:48:58 [1033] Starting audio processor (PID 1033)
2013-02-10 17:48:58 [1033] initializing audio in audio processor
2013-02-10 17:48:58 [1033] loaded local pulseaudio driver
2013-02-10 17:48:58 [1033] select error() in audio processor (can be caused because of pulseaudio) (Interrupted system call)
2013-02-10 17:48:59 [1033] Pulseaudio connect reported an error (6 - Connection refused)
2013-02-10 17:48:59 [1033] select error() in audio processor (can be caused because of pulseaudio) (Interrupted system call)
2013-02-10 17:48:59 [1033] Pulseaudio connect reported an error (6 - Connection refused)
I started tvtime again. This time I ran rm -rf ~/.config/pulse/ and killall pulseaudio to kill the Pulseaudio server. I instantly got a loud noise out of my speakers.
I closed tvtime and re-opened it. And there it was: TV SOUND OUT OF MY SPEAKERS!
But now, when I run pavucontrol I get:
Connection to PulseAudio failed. Automatic retry in 5s.
In this cas this is likely because PULSE_SERVER in the Environment/X11 root Windows Properties or default-server in client.conf is misconfigured.
This situation can also arrise when PulseAudio crashed and left stale details in the X11 Root Windows.
If this is the case, then PuseAudio should autospawn again, or if this is not configured you sould run start-pulseaudio-x11 manually.
I got the information from Sundtek, that Pulseaudio should not be started as system service. Instead, every user should have an own Pulseaudio daemon and that PA was concepted this way. But they don't know how it is implemented in ArchLinux.
I also don't know it exactly. So the question to all you ArchLinu pro's:
How is PA implemented in Arch? And how can I sustainable solve my problem?
Thank you for your time and help!
Durag

There is just one more thing I found out.
After logging into Gnome and running ps aux | grep pulse I get:
rebel 727 0.7 0.2 444316 10288 ? S<l 03:02 0:00 /usr/bin/pulseaudio --start --log-target=syslog
rebel 731 0.0 0.0 71236 2652 ? S 03:02 0:00 /usr/lib/pulse/gconf-helper
rebel 843 0.0 0.0 17396 1184 pts/0 D+ 03:03 0:00 grep pulse
To get the sound to work I run:
rm -rf .config/pulse/
killall pulseaudio
killall -9 pulseaudio
I again run ps aux | grep pulse
rebel 4016 4.2 0.2 510124 11528 ? Sl 03:06 0:02 /usr/bin/pulseaudio --start --log-target=syslog
rebel 4020 0.0 0.0 71368 2700 ? S 03:06 0:00 /usr/lib/pulse/gconf-helper
rebel 4046 0.0 0.0 19456 1216 pts/0 R+ 03:07 0:00 grep pulse
I can recognize that the STATs have changed. Maybe this can help to solve the problem?

No Sound in ALSA & Pulseaudio despite configuring

Right.
So, I installed ALSA, and fully followed the wiki's instructions to no avail. On a previous install on another computer, I was able to fix the problem by installing Pulseaudio, and so I did it again on my current comp. I got sound on this computer on a singular occasion after the Pulseaudio install, but now sound is gone again. Any idea why?
I am using compiz-standalone, and in my ~/.start-fusion.sh, I already have
pulseaudio --start
in it.
However, when I run pacmd list, it still states that there is no Pulseaudio daemon running. I have to start it manually in the end, and still, sound doesn't work.
I also noticed that when executing
$ fuser -v /dev/snd/*
$ fuser -v /dev/dsp
/dev/dsp does not exist.
asound.conf is already to let pulseaudio be the default device, and everything in alsamixer is unmuted. I have also installed alsa-plugins, so there, really, should be no reason why pulseaudio, and sound in general doesn't work.

I'm not sure what volume control/mixer you are using but I installed pulse audio yesterday along with gnome-media-pulseaudio and through gnome's applet I was able to change my hardware's profile and manage to get sound and volume controls back hope it helps (i'm running compiz stand alone as well but have Gnome DE too)

PULSEAUDIO Crackling / Distortion

This distortion is like a crackly echoey distortion. It actually has an echo component to it, When I change the output and it goes normal for a few seonds and tehre is a song playing, then it kicks in, the sound quality goes way down, it has this crackling sound, and very noticeable echo comes on the voices and the drums (which you can barely make out over the crackling). Output is perfect if forced to use ALSA directly, or if I boot into windows Seven. This is all pulse audio (probably), the piece of crap.
When I go "pulseaudio --kill" and then "mplayer -ao alsa:device=hw=0.0 <file>" it plays perfectly without distortion. When I go alsamixer -c0 none of the sliders have any gain, they are all at 0db or -1db or so.
To stop the distortion, chanigng from 44100 to 48000 and so on in the /etc/pulse/ whatever file doesnt work. What does work is opening pavucontrol and changing the internal audio on the configuration page. I hav a Barts thing that is turned off and an Internal Audio that can be set to a huge list of things like Analog Surround 5.0 Out, Analog Stereo Duplex, Analog Surround 7.1 Output + Analog Stereo Input, so on and so forth.
Now changing from one to another will randomly make the distortion go away for 30 seconds, until I open a new program, close pavucontrol, or just wait a while, then it will come back on, and I'll have to change it to another, which won't work at first usually but randomly will. There is NO CORRELATION between what is actually chosen and what does and doesnt have distortion straight away.
Basically PulseAudio is $#!+ all the sudden, and I don't know why. I have a fully updated system, hardware is as follows:
$ lspci | grep udio
00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia (Intel HDA) (rev 40)
01:00.1 Audio device: ATI Technologies Inc Barts HDMI Audio [Radeon HD 6800 Series]
Also plays it fine as testing with mplayer, so this is a pulseaudio thing (i think). WTH????
[EDIT] And here is the lsmod:
$ lsmod
Module Size Used by
uas 8170 0
usb_storage 43628 1
ipv6 280362 48
md5 4127 2
hmac 2937 1
nls_utf8 1320 5
cifs 265094 6
fscache 39883 1 cifs
it87 28123 0
hwmon_vid 2796 1 it87
usbhid 36375 0
hid 78087 1 usbhid
snd_hda_codec_hdmi 22857 1
snd_hda_codec_realtek 295717 1
snd_seq_dummy 1479 0
snd_seq_oss 29240 0
fglrx 2679751 87
snd_seq_midi_event 5516 1 snd_seq_oss
vboxdrv 1760035 0
snd_seq 50562 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi_event
snd_hda_intel 21837 6
snd_hda_codec 74609 3 snd_hda_codec_hdmi,snd_hda_codec_realtek,snd_hda_intel
snd_seq_device 5281 3 snd_seq_dummy,snd_seq_oss,snd_seq
r8169 36414 0
ohci_hcd 21338 0
ppdev 5854 0
firewire_ohci 26921 0
snd_hwdep 6222 1 snd_hda_codec
snd_pcm_oss 39509 0
parport_pc 31800 1
snd_mixer_oss 17730 1 snd_pcm_oss
lp 8992 0
firewire_core 50038 1 firewire_ohci
snd_pcm 72321 6 snd_hda_codec_hdmi,snd_hda_intel,snd_hda_codec,snd_pcm_oss
mii 3842 1 r8169
evdev 9361 12
fuse 65179 5
xhci_hcd 65218 0
parport 30087 3 ppdev,parport_pc,lp
edac_core 35034 0
i2c_piix4 8176 0
ehci_hcd 38878 0
crc_itu_t 1313 1 firewire_core
snd_timer 19537 2 snd_seq,snd_pcm
button 4882 1 fglrx
i2c_core 19217 1 i2c_piix4
processor 25265 0
wmi 8061 0
pcspkr 1835 0
sg 25972 0
k10temp 2771 0
serio_raw 4566 0
usbcore 139496 7 uas,usb_storage,usbhid,ohci_hcd,xhci_hcd,ehci_hcd
edac_mce_amd 9223 0
shpchp 26725 0
snd 58906 22 snd_hda_codec_hdmi,snd_hda_codec_realtek,snd_seq_oss,snd_seq,snd_hda_intel,snd_hda_codec,snd_seq_device,snd_hwdep,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer
soundcore 6161 1 snd
snd_page_alloc 7361 2 snd_hda_intel,snd_pcm
pci_hotplug 24687 1 shpchp
ext4 334690 1
mbcache 5802 1 ext4
jbd2 69898 1 ext4
crc16 1313 1 ext4
sd_mod 27120 4
ahci 20465 1
pata_jmicron 2464 0
libahci 17952 1 ahci
pata_acpi 3296 0
libata 169396 4 ahci,pata_jmicron,libahci,pata_acpi
scsi_mod 125814 5 uas,usb_storage,sg,sd_mod,libata
Also my last message on dmesg is:
hda-intel: IRQ timing workaround is activated for card #0. Suggest a bigger bdl_pos_adj.
But there is only one of these lines in the output and the distortion kicks in several times. This was right after the line:
chromium-sandbo (2511): /proc/2509/oom_adj is deprecated, please use /proc/2509/oom_score_adj instead.
So it happened after I opened my web browser (I only just booted this pc up I had it turned off over the night). This is a recurring problem over the past couple of days, BTW. It was probably the last update because when I done it I didn't reboot for awhile.
Last edited by me4tw (2011-04-03 05:03:26)

hi - i had the same problem, but only with vlc - but I love vlc *sad*.
00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia (Intel HDA) (rev 40)
Subsystem: Micro-Star International Co., Ltd. Device 7599
Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx-
Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=slow >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx-
Latency: 64, Cache Line Size: 64 bytes
Interrupt: pin ? routed to IRQ 16
Region 0: Memory at f9ff4000 (64-bit, non-prefetchable) [size=16K]
Capabilities: <access denied>
Kernel driver in use: HDA Intel
Kernel modules: snd-hda-intel
lsmod | grep -i snd:
snd_seq_dummy 1455 0
snd_seq_oss 28052 0
snd_seq_midi_event 5332 1 snd_seq_oss
snd_seq 48705 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi_event
snd_seq_device 5100 3 snd_seq_dummy,snd_seq_oss,snd_seq
snd_pcm_oss 37890 0
snd_mixer_oss 14851 1 snd_pcm_oss
snd_hda_codec_realtek 294053 1
snd_hda_intel 21738 2
snd_hda_codec 73739 2 snd_hda_codec_realtek,snd_hda_intel
snd_hwdep 6134 1 snd_hda_codec
snd_pcm 71032 3 snd_pcm_oss,snd_hda_intel,snd_hda_codec
snd_timer 18992 2 snd_seq,snd_pcm
snd 55132 15 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 5986 1 snd
snd_page_alloc 7017 2 snd_hda_intel,snd_pcm
uname -r : 2.6.38-ARCH
the hint with pulse an tsched=0 does help too :-)
@heftig: danke ;-)
Last edited by debijan (2011-04-18 18:03:43)

Gnome3 without pulseaudio (again)

Hi all,
I always have had a particular hate for pulseaudio, a sound daemon that stands as a layer between apps and the hardware. Great idea, if it only worked! It doesn't!
Configuring and getting it to work is a real pain, as it doesn't work out-of-the box.
Gnome 3 now seems to depend on pulseaudio, which I think it's a bad decision. At least KDE let's you choose!
I was about to ditch Gnome3 - even though I was enjoying the experience - then I came accross a thread in this forum where other people have the same problems with pulseaudio. The topic is this: https://bbs.archlinux.org/viewtopic.php?id=117986 (post #9)
So I did all that. Renamed and later deleted all pulseaudio-* under /etc/xdg, launched gnome-session-properties to disable pulseaudio, deleted .desktop files under ~/.config/autostart and even then it seemed to start.
Installed alsa-utils, ran alsaconf, rebooted.. And pulseaudio was still there. Don't know what process is is invoking it, but I'm guessing it's the neat volume manager in the tray.
Finally I ran gstreamer-properties, chose ALSA instead of pulseaudio, and then....renamed pulseaudio itself mv /usr/bin/pulseaudio /usr/bin/pulseaudio.o.
Ah.............................everything was just working normally, except for the volume keys (as the neat volume manager disappeared)
So, how Can I replace the default app in the tray?
The sound definitions under System seem to be pulseaudio related and now no soundcard is shown.
Any ideas?
Last edited by aurocha (2011-05-09 23:22:17)

You can blow Pulse off your system entirely by doing pacman -Rnsdd pulseaudio. (I also got rid of pulseaudio-alsa. I don't know if you need that if you're using something other than OSS). After removing it, I unchecked/disabled all references to Pulse everywhere I could find in config utilities, and then deleted every file that had that nasty p-word in its name anywhere in ~/.
Then I had to go through this harrowing process to get sound working as one would expect it to: pacman -S oss.
I had the same issues as you, though, thanks to Pulse integration. Basically, no volume control: Volume icon does nothing, multimedia keys don't work, and there's no OSD indicator when the volume is/should be changing.
For the first problem, oh well, I removed the volume icon from the panel. I always did that, anyway, because I just use the keyboard.
For the second, I created three one-line scripts and bound them to my multimedia keys. For oss,
Volume up: ossxmix vmix0-outvol +1
Volume down: ossxmix vmix0-outvol -- -1
Mute: ossxmix vmix0-outvol 0
These are the laziest things ever. Mute just turns the volume all the way down, it can't return you to where you were. The OSS FAQ has more sophisticated script examples, if you're into that. I think the third problem (no OSD) can be resolved with scripting, too.
ALSA can be controlled via scripting with amixer: http://manpages.ubuntu.com/manpages/gut … xer.1.html
Rant time: I have no idea why the GNOME guys decided to integrate with a particular sound system. Linux sound is a total mess. Users' needs and mileage vary so greatly -- nothing works for everyone. In my personal experiences, OSS has always worked 100% out of the box with zero effort or configuration, whereas with Pulse the closest I ever got was, "I can make sound from one application, but please don't try to use two at the same time or everything will explode." I gave Pulse about 45 minutes after GNOME3 came along. Then it was like, "Why am I wrestling with this when OSS was already working with no effort?"

Can not get output from Objective DAC without Pulseaudio [SOLVED]

I have owned the ObjectiveDAC/O2 Amp combo for a couple of months now. Unfortunately I was unable to get the device to function properly without pulseaudio installed. When trying to configure it on a pure alsa system, I can set it as the default control card in alsamixer. The issue is that when I set it as the default pcm device, it fails to work. I can select the device in pulseaudio fine, and my system recognizes the device in alsa, however I cannot, for some reason, actually use the device. It seems that pulseaudio butchers sound quality, and I would greatly appreciate any help being able to get my device working without it. I can't believe that any sane individual would actually make part of the audio stack downmix audio by default.
It was just a lack of understanding of Alsa's config syntax. I do apologize for taking up space on the forums. I now have Alsa running with my DAC just fine.
Last edited by agahnim (2015-05-01 13:50:28)

Yeah that's what I meant. Try to burn a new cd at a slower speed. When the new one is done create a hashsum of the new burn and compare it to the iso you used to burn it with. I only say this because a few times in the past when I had sr0 errors while booting live cds it was because the disk was messed up and I had to burn it again.

BASICS - PulseAudio - No Sound

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