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uvdevnull
September 1st, 2009, 04:40 AM
If anyone is still having issues after following this guide, as I was, may I suggest following this one, it worked for me:

http://ubuntuforums.org/showthread.php?t=1046137

Stromatolyte
September 5th, 2009, 08:40 AM
I just wanted to say thanks followed part A & my sound now works on both the card and usb.
(Juanty 9.04):D

prp123
September 15th, 2009, 05:31 AM
also have to remove tons of stuff, like ubuntu-desktop. I'll tery and get the hardy deb packages and install them, to see if that solves my problem

Research Paper Writing (http://www.pureresearchpapers.com)

Artemis3
September 19th, 2009, 02:20 AM
I'm on Jaunty x86_64

I have pulseaudio working perfectly, so i gave a try to the global equalizer part of the guide. After finishing, i logged out then logged in, and the standard welcoming sound sure enough sounds with the eq values i had chosen, however, thats the only sound i ever hear. After that, any other attempts to play back sound doesn't produce anything.

If i attempt to play back any sound, Pulseaudio Manager shows the client, but there is no sound.

bunorama
September 21st, 2009, 02:21 AM
The instruction (first page) states this does not work in Jaunty, maybe need to line that out since anyone running Jaunty should probably stop there...

It might be nice to include a big bold notice at the top stating this is an Edit-A-Text-File solution, which if it works is fine, but very disappointing to discover this later on.

wizard1974uk
September 21st, 2009, 07:37 AM
The instruction (first page) states this does not work in Jaunty, maybe need to line that out since anyone running Jaunty should probably stop there...

It might be nice to include a big bold notice at the top stating this is an Edit-A-Text-File solution, which if it works is fine, but very disappointing to discover this later on.

It does work with Jaunty, read the post carefully, you just follow the instructions in Part A.

Doncr
September 27th, 2009, 04:42 AM
Note 4: Kubuntu users: Don't follow this guide - PulseAudio isn't used in your distribution

Really? KDE seems to depend on PulseAudio. I cannot remove it without a list of dependencies that include a vast amount of kde packages - including kate.

I am so sick of trying to fix bloody pulse audio crap - is there a linux distribution that does not use pulse?

Crimson Kaze
September 28th, 2009, 08:05 AM
Stuck on step 5 of part A Keep getting Connection failed: Connection refused Ubuntu 9.04 Jaunty

redcharlie
September 28th, 2009, 07:06 PM
Just in case anyone else has this issue:
No gnome app seemed to play sound, specifically rhythmbox, sound-juicer, totem, and the nautilus mouse-over sound preview, they all were mute.
They all acted like they were playing (even had the little music note icon appear in nautilus) but no sound (and DID NOT appear in the pulseaudio applet->Volume Control->playback window)

Kaffeine, mplayer, xine, they worked fine.
(and DID appear in the pulseaudio applet->Volume Control->playback window)

Running sound-juicer or totem in a terminal window with the option "--gst-debug=3" revealed that the alsa libs were complaining that the sound device was busy. (if you do this redirect the debug output to a file cause it's a firehose, e.g. "totem --gst-debug=3 >totem.log 2>&1")

Solution: goto System->Preferences->Sound->Devices and change everything from autodetect to pulseaudio (despite what Part A, point 4 says) and now it all works!:guitar:

Using Jaunty on a Gigabyte MA78GPM

RgnKjnVA
October 3rd, 2009, 09:25 PM
Running 8.10 on a Dell Latitude D610 laptop, Intel drivers. At some point the alsa module stopped initializing. I'm having some trouble making sense of the logs. Is it failing because of the 'Operation not permitted' problems early in the logs? This used to work earlier in the summer.

my script contains...


pactl load-module module-alsa-sink device=Headset; sleep 2
pactl load-module module-alsa-source device=Headset; sleep 2



Oct 3 16:17:01 laptop /USR/SBIN/CRON[15369]: (root) CMD ( cd / && run-parts --report /etc/cron.ho
urly)
Oct 3 16:18:06 laptop kernel: [ 8404.031297] Intel ICH 0000:00:1e.2: PCI INT A disabled
Oct 3 16:18:06 laptop kernel: [ 8404.720305] Intel ICH 0000:00:1e.2: PCI INT A -> GSI 16 (level, lo
w) -> IRQ 16
Oct 3 16:18:06 laptop kernel: [ 8404.720337] Intel ICH 0000:00:1e.2: setting latency timer to 64
Oct 3 16:18:07 laptop kernel: [ 8405.548045] intel8x0_measure_ac97_clock: measured 55206 usecs
Oct 3 16:18:07 laptop kernel: [ 8405.548059] intel8x0: clocking to 48000
Oct 3 16:18:18 laptop pulseaudio[15797]: main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation
not permitted
Oct 3 16:18:18 laptop pulseaudio[15797]: main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation
not permitted
Oct 3 16:18:19 laptop pulseaudio[15797]: alsa-util.c: Error opening PCM device hw:0: Device or reso
urce busy
Oct 3 16:18:19 laptop pulseaudio[15797]: module.c: Failed to load module "module-alsa-sink" (argum
ent: "device_id=0 sink_name=alsa_output.pci_8086_266e_sound_card_0_a lsa_playback_0"): initialization
failed.
Oct 3 16:18:19 laptop pulseaudio[15797]: sap.c: sendmsg() failed: Operation not permitted
Oct 3 16:18:19 laptop pulseaudio[15797]: rtp.c: sendmsg() failed: Operation not permitted
Oct 3 16:18:20 laptop last message repeated 90 times
Oct 3 16:18:24 laptop pulseaudio[15797]: sap.c: sendmsg() failed: Operation not permitted
Oct 3 16:18:34 laptop last message repeated 2 times
Oct 3 16:18:34 laptop pulseaudio[15797]: alsa-util.c: Error opening PCM device Headset: Unknown err
or 240
Oct 3 16:18:34 laptop pulseaudio[15797]: module.c: Failed to load module "module-alsa-sink" (argum
ent: "device=Headset"): initialization failed.
Oct 3 16:18:36 laptop pulseaudio[15797]: alsa-util.c: Error opening PCM device Headset: Unknown err
or 240
Oct 3 16:18:36 laptop pulseaudio[15797]: module.c: Failed to load module "module-alsa-source" (arg
ument: "device=Headset"): initialization failed.
Oct 3 16:18:39 laptop pulseaudio[15797]: sap.c: sendmsg() failed: Operation not permitted

Artemis3
October 3rd, 2009, 11:01 PM
Warning 3: The equalizer currently does not work for Jaunty users, as there seems to be missing LADSPA plugins in the libasound2-plugins package. I'm investigating the issue currently.

As i said, it does work briefly, namely, the sound that plays right after logon sounds equalized and all, but it just doesn't work after that.

I don't think there are any LADSPA plugins missing, i installed ladspa-sdk and did the equalization using ReZound (which in turn uses said ladspa plugin).

The problem must lie elsewhere...

A made this ~/.asoundrc (plughw fails horribly, so used plug:dmix)

# ALSA library configuration file

# Include settings that are under the control of asoundconf(1).
# (To disable these settings, comment out this line.)
</home/artemis3/.asoundrc.asoundconf>

pcm.equalized {
type plug
slave.pcm "equalizer";
}

pcm.equalizer {
type ladspa

# The output from the EQ can either go direct to a hardware device
# (if you have a hardware mixer, e.g. SBLive/Audigy) or it can go
# to the software mixer shown here.
#slave.pcm "plughw"
slave.pcm "plug:dmix"

# Sometimes you may need to specify the path to the plugins,
# especially if you've just installed them. Once you've logged
# out/restarted this shouldn't be necessary, but if you get errors
# about being unable to find plugins, try uncommenting this.
path "/usr/lib/ladspa"

plugins [
{
label mbeq
id 1197
input {
#this setting is here by example, edit to your own taste
#bands: 50hz, 100hz, 156hz, 220hz, 311hz, 440hz, 622hz, 880hz,
# 1250hz, 1750hz, 2500hz, 3500hz, 5000hz, 10000hz, 20000hz
#range: -70 to 30
controls [ 3 5 -8 -10 -10 -9 -12 -12 -12 -12 -14 -20 -12 -9 0 ]
}
}
]
}


And here is my /etc/pulse/default.pa


#!/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
load-sample-dir-lazy /usr/share/sounds/ubuntu/stereo

.fail

### 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 suspend sinks/sources that become idle for too long
load-module module-suspend-on-idle

### Automatically restore the volume of streams and devices
load-module module-device-restore
load-module module-stream-restore

### Load audio drivers statically (it's probably better to not load
### these drivers manually, but instead use module-hal-detect --
### see below -- for doing this automatically)
#load-module module-alsa-sink
load-module module-alsa-sink device=equalized
#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

### Automatically load driver modules depending on the hardware available
.ifexists module-hal-detect.so
load-module module-hal-detect tsched=0
.else
### Alternatively use the static hardware detection module (for systems that
### lack HAL support)
load-module module-detect
.endif

### Automatically load driver modules for Bluetooth hardware
#.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 reciever 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 description="RTP Multicast Sink"
#load-module module-rtp-send source=rtp.monitor

### Automatically restore the default sink/source when changed by the user during runtime
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

### If autoexit on idle is enabled we want to make sure we only quit
### when no local session needs us anymore.
load-module module-console-kit

### Enable positioned event sounds
load-module module-position-event-sounds

# 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

### Make some devices default
#set-default-sink output
#set-default-source input

Since it doesn't work, i just comment out the line with:

##load-module module-alsa-sink device=equalized and all is back to normal.

Perhaps something is wrong in loading modules and HAL getting in the way?

Anyone knows if there is a launchpad petition for a system wide equalizer?

vixmusic01
October 11th, 2009, 08:28 PM
Hey anyone out there having difficulty trying to get Pulseaudio working with a hardware playback device that uses ICE1712 (as displayed when you run the command "aplay -l", there is a bug I have found which seems to affect all cards using this chipset.

For example my card is card 0: H71 [Hoontech STA DSP24 Media 7.1], device 0: ICE1712 multi [ICE1712 multi] but according to this bug other cards such as M-Audio type cards are affected too.

There is a manual workaround listed in this bug.
https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/178442
There is a posting about an M-Audio Delta 44 card...
https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/178442/comments/8

I have found that this workaround works for my card too.

If it helps people I have included the steps I made to get my ICE1712 type card working in pulseaudio.

In my example you will need to replace the "H71" with whatever the device says when you run
asoundconf list

By editing this file,

gksudo gedit /etc/pulse/default.pa
and adding the following lines at the end, I now have a working pulseaudio in 8.10 64 bit.


# Workaround for MAudio Audiophile
# https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/178442/comments/7
# Added comment rod40cool - (STA Audio Media 7.1 card uses same ICE1712 driver)
# Added comment rod40cool - Run command "asoundconf list" to determine what to add after "device=hw:????"
# Added comment rod40cool - Make sure after pasting that there are only 2 lines below starting each with "load-module module..."
load-module module-alsa-sink sink_name=H71_out device=hw:H71 format=s32le channels=10 channel_map=left,right,aux0,aux1,aux2,aux3,aux4,au x5,aux6,aux7
load-module module-alsa-source source_name=H71_in device=hw:H71 format=s32le channels=12 channel_map=left,right,aux0,aux1,aux2,aux3,aux4,au x5,aux6,aux7,aux8,aux9

I hope this helps someone who is trying to get pulseaudio running for a ST Audio Media 7.1 or M-Audio Delta card or in fact any card that uses the ICE1712 driver.

Cheers

THANK YOU!!!

My M-Audio card NOW WORKS!

See my steps on https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/425435

and on my blog: http://www.livemorelightly.com/Tech/2009/10/m-audio-delta4…in-ubuntu-9-04

Thanks also to:

Laurent Moussault wrote on 2008-04-06: #8

Same problem with a M-Audio Delta 44 (same alsa driver ice1712).

The workaround above works, with "device=hw:M44".

From what I read in the link posted above, the problem seems to be that there is no default channel map (i.e. knowing which channel is left and which is right) for 10 channels cards, and that the driver (or may be the card) refuse to open for just 2 channels.

IMHO the bug importance is more than "low", since the ice1712 based cards are quite common among "music-oriented" linux users (these are affordable high-quality cards, supported by alsa for a long time).

I have to agree with this statement. It was difficult for me, even though I have some grasp of technology, to get this to work. Gutsy worked out of the box!

Thanks for your kind attention to this matter in future distros.

Thanks again to those that helped me.

Victoria:KS

ownsuall
October 12th, 2009, 01:36 AM
I'm in the Ubuntu 64-bit Beta atm(at the moment) and the sound works with out need to use the guide so I believe they have fixed this in 9.10. :)
Now I can play FoFix :guitar:

UeB
October 19th, 2009, 08:04 PM
Thanks!!!

Now amarok and flash do not bock each other's sound anymore!

fungiside
October 20th, 2009, 12:06 AM
I've been trying to figure out how to get pulseaudio to run at system startup in Jaunty. I have another thread (http://ubuntuforums.org/showthread.php?t=1295602) but I'll just paste the same stuff in here.




fungi@ren:~$ aplay -l
aplay: device_list:217: no soundcards found...


But after I log in to gnome:



fungi@ren:~$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: HDMI [HDA ATI HDMI], device 3: ATI HDMI [ATI HDMI]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: CMI8788 [C-Media CMI8788], device 0: Multichannel [Multichannel]
Subdevices: 0/1
Subdevice #0: subdevice #0
card 1: CMI8788 [C-Media CMI8788], device 1: Digital [Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0


Also strange, starting pulseaudio manually seems to do nothing:



fungi@ren:~$ pulseaudio -vv -D
D: main.c: Started as real root: no, suid root: yes
I: main.c: We're in the group 'pulse-rt', allowing high-priority scheduling.
I: main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted
I: main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation not permitted
I: core-util.c: Successfully gained nice level -11.
D: main.c: Can realtime: yes, can high-priority: yes
I: main.c: Giving up CAP_NICE
D: main.c: Can realtime: no, can high-priority: no
I: main.c: Daemon startup successful.

fungi@ren:~$ aplay -l
aplay: device_list:217: no soundcards found...



If I start with --v



fungi@ren:~$ pulseaudio -vv
D: main.c: Started as real root: no, suid root: yes
I: main.c: We're in the group 'pulse-rt', allowing high-priority scheduling.
I: main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted
I: main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation not permitted
I: core-util.c: Successfully gained nice level -11.
D: main.c: Can realtime: yes, can high-priority: yes
I: main.c: Giving up CAP_NICE
D: main.c: Can realtime: no, can high-priority: no
I: main.c: This is PulseAudio 0.9.15
D: main.c: Compilation host: i486-pc-linux-gnu
D: main.c: Compilation CFLAGS: -g -O2 -g -Wall -O3 -Wall -W -Wextra -pipe -Wno-long-long -Winline -Wvla -Wno-overlength-strings -Wunsafe-loop-optimizations -Wundef -Wformat=2 -Wlogical-op -Wsign-compare -Wformat-security -Wmissing-include-dirs -Wformat-nonliteral -Wold-style-definition -Wpointer-arith -Winit-self -Wdeclaration-after-statement -Wfloat-equal -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wmissing-declarations -Wmissing-noreturn -Wshadow -Wendif-labels -Wcast-align -Wstrict-aliasing=2 -Wwrite-strings -Wno-unused-parameter -ffast-math -Wp,-D_FORTIFY_SOURCE=2 -fno-common -fdiagnostics-show-option
D: main.c: Running on host: Linux i686 2.6.28-15-generic #52-Ubuntu SMP Wed Sep 9 10:49:34 UTC 2009
D: main.c: Found 1 CPUs.
I: main.c: Page size is 4096 bytes
D: main.c: Compiled with Valgrind support: no
D: main.c: Running in valgrind mode: no
D: main.c: Optimized build: yes
D: main.c: All asserts enabled.
I: main.c: Machine ID is c175c33c6ee5c964d4d8e84b4a8f1cab.
I: main.c: Session ID is c175c33c6ee5c964d4d8e84b4a8f1cab-1255993290.807568-1833262847.
I: main.c: Using runtime directory /home/fungi/.pulse/c175c33c6ee5c964d4d8e84b4a8f1cab:runtime.
I: main.c: Using state directory /home/fungi/.pulse.
I: main.c: Running in system mode: no
I: main.c: Fresh high-resolution timers available! Bon appetit!
D: rtclock.c: Timer slack is set to 50 us.
D: memblock.c: Using shared memory pool with 1024 slots of size 64.0 KiB each, total size is 64.0 MiB, maximum usable slot size is 65496
I: module.c: Loaded "module-suspend-on-idle" (index: #0; argument: "").
I: module-device-restore.c: Sucessfully opened database file '/home/fungi/.pulse/c175c33c6ee5c964d4d8e84b4a8f1cab:device-volumes.i486-pc-linux-gnu.gdbm'.
I: module.c: Loaded "module-device-restore" (index: #1; argument: "").
I: module-stream-restore.c: Sucessfully opened database file '/home/fungi/.pulse/c175c33c6ee5c964d4d8e84b4a8f1cab:stream-volumes.i486-pc-linux-gnu.gdbm'.
I: module.c: Loaded "module-stream-restore" (index: #2; argument: "").
I: module-card-restore.c: Sucessfully opened database file '/home/fungi/.pulse/c175c33c6ee5c964d4d8e84b4a8f1cab:card-database.i486-pc-linux-gnu.gdbm'.
I: module.c: Loaded "module-card-restore" (index: #3; argument: "").
I: module.c: Loaded "module-augment-properties" (index: #4; argument: "").
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.15/modules/module-hal-detect.so': success
D: dbus-util.c: Successfully connected to D-Bus system bus 4cee9ce6de24e5be0579d3144adce563 as :1.40
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/computer_alsa_timer
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/computer_alsa_sequencer
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_13f6_8788_sound_card_0_alsa_playback_1
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_13f6_8788_sound_card_0_alsa_capture_1
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_13f6_8788_sound_card_0_alsa_playback_0
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_13f6_8788_sound_card_0_alsa_capture_0
D: module-hal-detect.c: Loading module-alsa-card with arguments 'device_id=1 name=pci_13f6_8788_sound_card_0 card_name=alsa_card.pci_13f6_8788_sound_card_0 tsched=0'
E: module-alsa-card.c: Card '1' doesn't exist: No such file or directory
E: module.c: Failed to load module "module-alsa-card" (argument: "device_id=1 name=pci_13f6_8788_sound_card_0 card_name=alsa_card.pci_13f6_8788_sound_card_0 tsched=0"): initialization failed.
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_13f6_8788_sound_card_0_alsa_control__1
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1002_aa38_sound_card_0_alsa_playback_3
D: module-hal-detect.c: Loading module-alsa-card with arguments 'device_id=0 name=pci_1002_aa38_sound_card_0 card_name=alsa_card.pci_1002_aa38_sound_card_0 tsched=0'
E: module-alsa-card.c: Card '0' doesn't exist: No such file or directory
E: module.c: Failed to load module "module-alsa-card" (argument: "device_id=0 name=pci_1002_aa38_sound_card_0 card_name=alsa_card.pci_1002_aa38_sound_card_0 tsched=0"): initialization failed.
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1002_aa38_sound_card_0_alsa_control__1
I: module-hal-detect.c: Loaded 0 modules.
I: module.c: Loaded "module-hal-detect" (index: #5; argument: "tsched=0").
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.15/modules/module-bluetooth-discover.so': failure
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.15/modules/module-esound-protocol-unix.so': success
I: module.c: Loaded "module-esound-protocol-unix" (index: #6; argument: "").
I: module.c: Loaded "module-native-protocol-unix" (index: #7; argument: "").
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.15/modules/module-gconf.so': success
I: module.c: Loaded "module-gconf" (index: #8; argument: "").
I: module-default-device-restore.c: Saved default sink 'alsa_output.pci_13f6_8788_sound_card_0_alsa_playb ack_0' not existant, not restoring default sink setting.
I: module-default-device-restore.c: Saved default source 'alsa_input.pci_13f6_8788_sound_card_0_alsa_captur e_0' not existant, not restoring default source setting.
I: module.c: Loaded "module-default-device-restore" (index: #9; argument: "").
I: module.c: Loaded "module-rescue-streams" (index: #10; argument: "").
D: module-always-sink.c: Autoloading null-sink as no other sinks detected.
I: module-device-restore.c: Restoring volume for sink auto_null.
I: module-device-restore.c: Restoring mute state for sink auto_null.
I: sink.c: Created sink 0 "auto_null" with sample spec s16le 6ch 44100Hz and channel map front-left,rear-left,front-center,front-right,rear-right,lfe
I: sink.c: device.description = "Null Output"
I: sink.c: device.class = "abstract"
I: sink.c: device.icon_name = "audio-card"
I: module-device-restore.c: Restoring volume for source auto_null.monitor.
I: module-device-restore.c: Restoring mute state for source auto_null.monitor.
I: source.c: Created source 0 "auto_null.monitor" with sample spec s16le 6ch 44100Hz and channel map front-left,rear-left,front-center,front-right,rear-right,lfe
I: source.c: device.description = "Monitor of Null Output"
I: source.c: device.class = "monitor"
I: source.c: device.icon_name = "audio-input-microphone"
D: module-null-sink.c: Thread starting up
D: rtpoll.c: Acquired POSIX realtime signal SIGRTMIN+29
D: module-suspend-on-idle.c: Sink auto_null becomes idle.
I: module.c: Loaded "module-null-sink" (index: #11; argument: "sink_name=auto_null").
I: module.c: Loaded "module-always-sink" (index: #12; argument: "").
I: client.c: Created 0 "ConsoleKit Session /org/freedesktop/ConsoleKit/Session10"
D: module-console-kit.c: Added new session /org/freedesktop/ConsoleKit/Session10
I: module.c: Loaded "module-console-kit" (index: #13; argument: "").
I: module.c: Loaded "module-position-event-sounds" (index: #14; argument: "").
I: module.c: Loaded "module-cork-music-on-phone" (index: #15; argument: "").
W: main.c: Unable to contact D-Bus: org.freedesktop.DBus.Error.Spawn.ExecFailed: dbus-launch failed to autolaunch D-Bus session: Autolaunch error: X11 initialization failed.
I: main.c: Daemon startup complete.
D: module-hal-detect.c: dbus: interface=org.freedesktop.DBus, path=/org/freedesktop/DBus, member=NameAcquired
D: module-console-kit.c: dbus: interface=org.freedesktop.DBus, path=/org/freedesktop/DBus, member=NameAcquired

tiggsy
October 20th, 2009, 02:48 PM
The sound worked fine on Hardy (after about a week of mucking about)

I just upgraded to Jaunty Jackalope i386 and it doesn't any more - problem A : does not play audio, appears in the playback tab

I followed your instructions though some of the files listed in step 1 weren't present on this machine.

In step 5, the Pulse Audio Device Chooser doesn't work. I get the starting message, then it disappears. However, it is listed in the System Monitor Processes tab as sleeping. I just went straight to the volume control instead.


$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: Audigy2 [Audigy 2 ZS [SB0350]], device 0: emu10k1 [ADC Capture/Standard PCM Playback]
Subdevices: 31/32
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: Audigy2 [Audigy 2 ZS [SB0350]], 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: Audigy2 [Audigy 2 ZS [SB0350]], device 3: emu10k1 [Multichannel Playback]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: Audigy2 [Audigy 2 ZS [SB0350]], device 4: p16v [p16v]
Subdevices: 1/1
Subdevice #0: subdevice #0
tiggsy@tiggsys-desktop:~$



pkill pulseaudio; sleep 2; pulseaudio -vv
I: caps.c: Limited capabilities successfully to CAP_SYS_NICE.
I: caps.c: Dropping root privileges.
I: caps.c: Limited capabilities successfully to CAP_SYS_NICE.
D: main.c: Started as real root: no, suid root: yes
I: main.c: PolicyKit refuses acquire-high-priority privilege.
N: main.c: Called SUID root and real-time and/or high-priority scheduling was requested in the configuration. However, we lack the necessary privileges:
N: main.c: We are not in group 'pulse-rt', PolicyKit refuse to grant us the requested privileges and we have no increase RLIMIT_NICE/RLIMIT_RTPRIO resource limits.
N: main.c: For enabling real-time/high-priority scheduling please acquire the appropriate PolicyKit privileges, or become a member of 'pulse-rt', or increase the RLIMIT_NICE/RLIMIT_RTPRIO resource limits for this user.
I: main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted
I: main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation not permitted
D: main.c: Can realtime: no, can high-priority: no
D: main.c: Can realtime: no, can high-priority: no
I: main.c: This is PulseAudio 0.9.14
D: main.c: Compilation host: i486-pc-linux-gnu
D: main.c: Compilation CFLAGS: -g -O2 -g -Wall -O3 -Wall -W -Wextra -pedantic -pipe -Wno-long-long -Wvla -Wno-overlength-strings -Wconversion -Wundef -Wformat -Wlogical-op -Wpacked -Wformat-security -Wmissing-include-dirs -Wformat-nonliteral -Wold-style-definition -Wdeclaration-after-statement -Wfloat-equal -Wmissing-declarations -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wmissing-noreturn -Wshadow -Wendif-labels -Wpointer-arith -Wcast-align -Wwrite-strings -Wno-unused-parameter -ffast-math
D: main.c: Running on host: Linux i686 2.6.28-15-generic #52-Ubuntu SMP Wed Sep 9 10:49:34 UTC 2009
I: main.c: Page size is 4096 bytes
D: main.c: Compiled with Valgrind support: no
D: main.c: Running in valgrind mode: no
D: main.c: Optimized build: yes
I: main.c: Machine ID is 4ae84242b0b9cc433d09f91448fdff23.
I: main.c: Using runtime directory /home/tiggsy/.pulse/4ae84242b0b9cc433d09f91448fdff23:runtime.
I: main.c: Using state directory /home/tiggsy/.pulse.
I: main.c: Running in system mode: no
I: main.c: Fresh high-resolution timers available! Bon appetit!
D: memblock.c: Using shared memory pool with 1024 slots of size 64.0 KiB each, total size is 64.0 MiB, maximum usable slot size is 65496
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9/modules/module-gconf.so': success
I: module.c: Loaded "module-gconf" (index: #0; argument: "").
I: module.c: Loaded "module-suspend-on-idle" (index: #1; argument: "").
I: module-device-restore.c: Sucessfully opened database file '/home/tiggsy/.pulse/4ae84242b0b9cc433d09f91448fdff23:device-volumes.i486-pc-linux-gnu.gdbm'.
I: module.c: Loaded "module-device-restore" (index: #2; argument: "").
I: module-stream-restore.c: Sucessfully opened database file '/home/tiggsy/.pulse/4ae84242b0b9cc433d09f91448fdff23:stream-volumes.i486-pc-linux-gnu.gdbm'.
I: module.c: Loaded "module-stream-restore" (index: #3; argument: "").
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9/modules/module-hal-detect.so': success
I: module-hal-detect.c: Trying capability alsa
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/computer_alsa_timer
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/computer_alsa_sequencer
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_playback_4
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_capture_4
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_playback_3
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_playback_2
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_capture_2
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_capture_1
D: module-hal-detect.c: Loading module-alsa-sink with arguments 'device_id=0 sink_name=alsa_output.pci_1102_4_sound_card_0_alsa _playback_0 tsched=0'
D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
I: module-alsa-sink.c: Successfully opened device front:0.
I: module-alsa-sink.c: Successfully enabled mmap() mode.
I: (alsa-lib)control.c: Invalid CTL front:0
I: alsa-util.c: Unable to attach to mixer front:0: No such file or directory
I: alsa-util.c: Successfully attached to mixer 'hw:0'
I: alsa-util.c: Cannot find mixer control "Master" or mixer control is no combination of switch/volume.
W: alsa-util.c: Cannot find fallback mixer control "PCM" or mixer control is no combination of switch/volume.
I: alsa-util.c: Using mixer control "PCM".
I: module-device-restore.c: Restoring volume for sink alsa_output.pci_1102_4_sound_card_0_alsa_playback_ 0.
I: module-device-restore.c: Restoring mute state for sink alsa_output.pci_1102_4_sound_card_0_alsa_playback_ 0.
I: sink.c: Created sink 0 "alsa_output.pci_1102_4_sound_card_0_alsa_playback_ 0" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: module-device-restore.c: Restoring volume for source alsa_output.pci_1102_4_sound_card_0_alsa_playback_ 0.monitor.
I: module-device-restore.c: Restoring mute state for source alsa_output.pci_1102_4_sound_card_0_alsa_playback_ 0.monitor.
I: source.c: Created source 0 "alsa_output.pci_1102_4_sound_card_0_alsa_playback_ 0.monitor" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: module-alsa-sink.c: Using 8 fragments of size 1764 bytes, buffer time is 80.00ms
D: module-alsa-sink.c: hwbuf_unused=0
D: module-alsa-sink.c: setting avail_min=1
I: module-alsa-sink.c: Volume ranges from 0 to 100.
I: module-alsa-sink.c: Volume ranges from -40.00 dB to 0.00 dB.
I: alsa-util.c: ALSA device lacks independant volume controls for each channel.
I: module-alsa-sink.c: Using hardware volume control. Hardware dB scale supported.
I: module-alsa-sink.c: Using software mute control.
D: alsa-util.c: snd_pcm_dump():
D: alsa-util.c: Hooks PCM
D: alsa-util.c: Its setup is:
D: alsa-util.c: stream : PLAYBACK
D: alsa-util.c: access : MMAP_INTERLEAVED
D: alsa-util.c: format : S16_LE
D: alsa-util.c: subformat : STD
D: alsa-util.c: channels : 2
D: alsa-util.c: rate : 44100
D: alsa-util.c: exact rate : 44100 (44100/1)
D: alsa-util.c: msbits : 16
D: alsa-util.c: buffer_size : 3528
D: alsa-util.c: period_size : 441
D: alsa-util.c: period_time : 10000
D: alsa-util.c: tstamp_mode : ENABLE
D: alsa-util.c: period_step : 1
D: alsa-util.c: avail_min : 441
D: alsa-util.c: period_event : 0
D: alsa-util.c: start_threshold : -1
D: alsa-util.c: stop_threshold : 1849688064
D: alsa-util.c: silence_threshold: 0
D: alsa-util.c: silence_size : 0
D: alsa-util.c: boundary : 1849688064
D: alsa-util.c: Slave: Hardware PCM card 0 'Audigy 2 ZS [SB0350]' device 0 subdevice 0
D: alsa-util.c: Its setup is:
D: alsa-util.c: stream : PLAYBACK
D: alsa-util.c: access : MMAP_INTERLEAVED
D: alsa-util.c: format : S16_LE
D: alsa-util.c: subformat : STD
D: alsa-util.c: channels : 2
D: alsa-util.c: rate : 44100
D: alsa-util.c: exact rate : 44100 (44100/1)
D: alsa-util.c: msbits : 16
D: alsa-util.c: buffer_size : 3528
D: alsa-util.c: period_size : 441
D: alsa-util.c: period_time : 10000
D: alsa-util.c: tstamp_mode : ENABLE
D: alsa-util.c: period_step : 1
D: alsa-util.c: avail_min : 441
D: alsa-util.c: period_event : 0
D: alsa-util.c: start_threshold : -1
D: alsa-util.c: stop_threshold : 1849688064
D: alsa-util.c: silence_threshol
D: module-alsa-sink.c: Thread starting up
D: rtpoll.c: Acquired POSIX realtime signal SIGRTMIN+29
D: module-alsa-sink.c: Requested volume: 0: 100% 1: 100%
D: module-alsa-sink.c: Got hardware volume: 0: 100% 1: 100%
D: module-alsa-sink.c: Calculated software volume: 0: 100% 1: 100%
I: module-alsa-sink.c: Starting playback.
D: module-suspend-on-idle.c: Source alsa_output.pci_1102_4_sound_card_0_alsa_playback_ 0.monitor becomes idle.
D: module-suspend-on-idle.c: Sink alsa_output.pci_1102_4_sound_card_0_alsa_playback_ 0 becomes idle.
I: module.c: Loaded "module-alsa-sink" (index: #4; argument: "device_id=0 sink_name=alsa_output.pci_1102_4_sound_card_0_alsa _playback_0 tsched=0").
D: module-hal-detect.c: Loading module-alsa-source with arguments 'device_id=0 source_name=alsa_input.pci_1102_4_sound_card_0_als a_capture_0 tsched=0'
D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)setup.c: Cannot lock ctl elem
D: alsa-util.c: Trying front:0 without SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)setup.c: Cannot lock ctl elem
D: alsa-util.c: Trying plug:front:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Trying plug:front:0 without SND_PCM_NO_AUTO_FORMAT ...
I: alsa-util.c: PCM device plug:front:0 refused our hw parameters: Invalid argument
D: alsa-util.c: Trying surround40:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm_hw.c: open /dev/snd/pcmC0D0c failed
I: alsa-util.c: Couldn't open PCM device surround40:0: Device or resource busy
D: alsa-util.c: Trying surround41:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm_hw.c: open /dev/snd/pcmC0D0c failed
I: alsa-util.c: Couldn't open PCM device surround41:0: Device or resource busy
D: alsa-util.c: Trying surround50:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm_hw.c: open /dev/snd/pcmC0D0c failed
I: alsa-util.c: Couldn't open PCM device surround50:0: Device or resource busy
D: alsa-util.c: Trying surround51:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm_hw.c: open /dev/snd/pcmC0D0c failed
I: alsa-util.c: Couldn't open PCM device surround51:0: Device or resource busy
D: alsa-util.c: Trying surround71:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm_hw.c: open /dev/snd/pcmC0D0c failed
I: alsa-util.c: Couldn't open PCM device surround71:0: Device or resource busy
D: alsa-util.c: Trying hw:0 as last resort...
D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ...
I: module-alsa-source.c: Successfully opened device hw:0.
I: module-alsa-source.c: Successfully enabled mmap() mode.
I: alsa-util.c: Successfully attached to mixer 'hw:0'
I: alsa-util.c: Cannot find mixer control "Capture" or mixer control is no combination of switch/volume.
W: alsa-util.c: Cannot find fallback mixer control "Mic" or mixer control is no combination of switch/volume.
I: alsa-util.c: Using mixer control "Mic".
I: module-device-restore.c: Restoring volume for source alsa_input.pci_1102_4_sound_card_0_alsa_capture_0.
I: module-device-restore.c: Restoring mute state for source alsa_input.pci_1102_4_sound_card_0_alsa_capture_0.
I: source.c: Created source 1 "alsa_input.pci_1102_4_sound_card_0_alsa_capture_0" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: module-alsa-source.c: Using 2 fragments of size 7168 bytes, buffer time is 81.27ms
D: module-alsa-source.c: hwbuf_unused=0
D: module-alsa-source.c: setting avail_min=1
I: module-alsa-source.c: Volume ranges from 0 to 100.
I: module-alsa-source.c: Volume ranges from -40.00 dB to 0.00 dB.
I: alsa-util.c: All 2 channels can be mapped to mixer channels.
I: module-alsa-source.c: Using hardware volume control. Hardware dB scale supported.
I: module-alsa-source.c: Using software mute control.
D: alsa-util.c: snd_pcm_dump():
D: alsa-util.c: Hardware PCM card 0 'Audigy 2 ZS [SB0350]' device 0 subdevice 0
D: alsa-util.c: Its setup is:
D: alsa-util.c: stream : CAPTURE
D: alsa-util.c: access : MMAP_INTERLEAVED
D: alsa-util.c: format : S16_LE
D: alsa-util.c: subformat : STD
D: alsa-util.c: channels : 2
D: alsa-util.c: rate : 44100
D: alsa-util.c: exact rate : 44100 (44100/1)
D: alsa-util.c: msbits : 16
D: alsa-util.c: buffer_size : 3584
D: alsa-util.c: period_size : 1792
D: alsa-util.c: period_time : 40634
D: alsa-util.c: tstamp_mode : ENABLE
D: alsa-util.c: period_step : 1
D: alsa-util.c: avail_min : 1792
D: alsa-util.c: period_event : 0
D: alsa-util.c: start_threshold : -1
D: alsa-util.c: stop_threshold : 1879048192
D: alsa-util.c: silence_threshold: 0
D: alsa-util.c: silence_size : 0
D: alsa-util.c: boundary : 1879048192
D: module-alsa-source.c: Thread starting up
D: rtpoll.c: Acquired POSIX realtime signal SIGRTMIN+28
D: module-alsa-source.c: Requested volume: 0: 74% 1: 74%
D: module-alsa-source.c: Got hardware volume: 0: 74% 1: 74%
D: module-alsa-source.c: Calculated software volume: 0: 99% 1: 99%
D: module-suspend-on-idle.c: Source alsa_input.pci_1102_4_sound_card_0_alsa_capture_0 becomes idle.
I: module.c: Loaded "module-alsa-source" (index: #5; argument: "device_id=0 source_name=alsa_input.pci_1102_4_sound_card_0_als a_capture_0 tsched=0").
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_midi_3
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_midi_2
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_midi_1
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_midi_0
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_hw_specific_2
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_hw_specific_0
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_control__1
I: module-hal-detect.c: Loaded 2 modules.
I: module.c: Loaded "module-hal-detect" (index: #6; argument: "tsched=0").
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9/modules/module-esound-protocol-unix.so': success
I: module.c: Loaded "module-esound-protocol-unix" (index: #7; argument: "").
I: module.c: Loaded "module-native-protocol-unix" (index: #8; argument: "").
I: module-default-device-restore.c: Restored default sink 'alsa_output.pci_1102_4_sound_card_0_alsa_playback _0'.
D: core-subscribe.c: Dropped redundant event due to change event.
I: module-default-device-restore.c: Restored default source 'alsa_input.pci_1102_4_sound_card_0_alsa_capture_0 '.
I: module.c: Loaded "module-default-device-restore" (index: #9; argument: "").
I: module.c: Loaded "module-rescue-streams" (index: #10; argument: "").
I: module.c: Loaded "module-always-sink" (index: #11; argument: "").
I: client.c: Created 0 "ConsoleKit Session /org/freedesktop/ConsoleKit/Session1"
D: module-console-kit.c: Added new session /org/freedesktop/ConsoleKit/Session1
I: module.c: Loaded "module-console-kit" (index: #12; argument: "").
I: module.c: Loaded "module-position-event-sounds" (index: #13; argument: "").
I: main.c: Daemon startup complete.
D: module-hal-detect.c: dbus: interface=org.freedesktop.DBus, path=/org/freedesktop/DBus, member=NameAcquired
D: module-console-kit.c: dbus: interface=org.freedesktop.DBus, path=/org/freedesktop/DBus, member=NameAcquired
I: module-suspend-on-idle.c: Sink alsa_output.pci_1102_4_sound_card_0_alsa_playback_ 0 idle for too long, suspending ...
I: module-alsa-sink.c: Device suspended...
I: module-suspend-on-idle.c: Source alsa_output.pci_1102_4_sound_card_0_alsa_playback_ 0.monitor idle for too long, suspending ...
I: module-suspend-on-idle.c: Source alsa_input.pci_1102_4_sound_card_0_alsa_capture_0 idle for too long, suspending ...
I: module-alsa-source.c: Device suspended...
I: client.c: Created 1 "Native client (UNIX socket client)"
D: protocol-native.c: Protocol version: remote 14, local 14
I: protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
D: protocol-native.c: SHM possible: yes
D: protocol-native.c: Negotiated SHM: yes
I: module-stream-restore.c: Not restoring device for stream source-output-by-application-id:org.PulseAudio.pavucontrol, because already set
D: module-suspend-on-idle.c: Source alsa_output.pci_1102_4_sound_card_0_alsa_playback_ 0.monitor becomes idle.
D: module-suspend-on-idle.c: Source alsa_output.pci_1102_4_sound_card_0_alsa_playback_ 0.monitor becomes busy.
D: resampler.c: Channel matrix:
D: resampler.c: I00 I01
D: resampler.c: +------------
D: resampler.c: O00 | 1.000 1.000
I: resampler.c: Using resampler 'peaks'
I: resampler.c: Using float32le as working format.
D: memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
D: memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
I: source-output.c: Created output 0 "Peak detect" on alsa_output.pci_1102_4_sound_card_0_alsa_playback_ 0.monitor with sample spec float32le 1ch 25Hz and channel map mono
D: memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
D: memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
I: protocol-native.c: Final latency 60.00 ms = 40.00 ms + 20.00 ms
I: module-stream-restore.c: Not restoring device for stream source-output-by-application-id:org.PulseAudio.pavucontrol, because already set
I: module-alsa-source.c: Trying resume...
D: module-alsa-source.c: hwbuf_unused=0
D: module-alsa-source.c: setting avail_min=1
I: module-alsa-source.c: Resumed successfully...
D: module-suspend-on-idle.c: Source alsa_input.pci_1102_4_sound_card_0_alsa_capture_0 becomes idle.
D: module-suspend-on-idle.c: Source alsa_input.pci_1102_4_sound_card_0_alsa_capture_0 becomes busy.
D: resampler.c: Channel matrix:
D: resampler.c: I00 I01
D: resampler.c: +------------
D: resampler.c: O00 | 1.000 1.000
I: resampler.c: Using resampler 'peaks'
I: resampler.c: Using float32le as working format.
D: memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
D: memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
I: source-output.c: Created output 1 "Peak detect" on alsa_input.pci_1102_4_sound_card_0_alsa_capture_0 with sample spec float32le 1ch 25Hz and channel map mono
D: memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
D: memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
I: protocol-native.c: Final latency 60.00 ms = 40.00 ms + 20.00 ms
D: module-alsa-source.c: hwbuf_unused=0
D: module-alsa-source.c: setting avail_min=1
D: module-alsa-source.c: hwbuf_unused=0
D: module-alsa-source.c: setting avail_min=1
I: module-stream-restore.c: Storing volume/mute/device for stream source-output-by-application-id:org.PulseAudio.pavucontrol.
I: module-stream-restore.c: Synced.
D: module-suspend-on-idle.c: Source alsa_output.pci_1102_4_sound_card_0_alsa_playback_ 0.monitor becomes idle.
D: module-suspend-on-idle.c: Source alsa_output.pci_1102_4_sound_card_0_alsa_playback_ 0.monitor becomes idle.
I: source-output.c: Freeing output 0 "Peak detect"
D: module-alsa-source.c: hwbuf_unused=0
D: module-alsa-source.c: setting avail_min=1
D: module-suspend-on-idle.c: Source alsa_input.pci_1102_4_sound_card_0_alsa_capture_0 becomes idle.
D: module-suspend-on-idle.c: Source alsa_input.pci_1102_4_sound_card_0_alsa_capture_0 becomes idle.
I: source-output.c: Freeing output 1 "Peak detect"
I: client.c: Freed 1 "PulseAudio Volume Control"
I: protocol-native.c: Connection died.
I: module-suspend-on-idle.c: Source alsa_input.pci_1102_4_sound_card_0_alsa_capture_0 idle for too long, suspending ...
I: module-alsa-source.c: Device suspended...
I: module-suspend-on-idle.c: Source alsa_output.pci_1102_4_sound_card_0_alsa_playback_ 0.monitor idle for too long, suspending ...
I: client.c: Created 2 "Native client (UNIX socket client)"
D: protocol-native.c: Protocol version: remote 14, local 14
I: protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
D: protocol-native.c: SHM possible: yes
D: protocol-native.c: Negotiated SHM: yes
I: module-stream-restore.c: Not restoring device for stream source-output-by-application-id:org.PulseAudio.pavucontrol, because already set
D: module-suspend-on-idle.c: Source alsa_output.pci_1102_4_sound_card_0_alsa_playback_ 0.monitor becomes idle.
D: module-suspend-on-idle.c: Source alsa_output.pci_1102_4_sound_card_0_alsa_playback_ 0.monitor becomes busy.
D: resampler.c: Channel matrix:
D: resampler.c: I00 I01
D: resampler.c: +------------
D: resampler.c: O00 | 1.000 1.000
I: resampler.c: Using resampler 'peaks'
I: resampler.c: Using float32le as working format.
D: memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
D: memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
I: source-output.c: Created output 2 "Peak detect" on alsa_output.pci_1102_4_sound_card_0_alsa_playback_ 0.monitor with sample spec float32le 1ch 25Hz and channel map mono
D: memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
D: memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
I: protocol-native.c: Final latency 60.00 ms = 40.00 ms + 20.00 ms
I: module-stream-restore.c: Storing volume/mute/device for stream source-output-by-application-id:org.PulseAudio.pavucontrol.
I: module-stream-restore.c: Not restoring device for stream source-output-by-application-id:org.PulseAudio.pavucontrol, because already set
I: module-alsa-source.c: Trying resume...
D: module-alsa-source.c: hwbuf_unused=0
D: module-alsa-source.c: setting avail_min=1
I: module-alsa-source.c: Resumed successfully...
D: module-suspend-on-idle.c: Source alsa_input.pci_1102_4_sound_card_0_alsa_capture_0 becomes idle.
D: module-suspend-on-idle.c: Source alsa_input.pci_1102_4_sound_card_0_alsa_capture_0 becomes busy.
D: resampler.c: Channel matrix:
D: resampler.c: I00 I01
D: resampler.c: +------------
D: resampler.c: O00 | 1.000 1.000
I: resampler.c: Using resampler 'peaks'
I: resampler.c: Using float32le as working format.
D: memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
D: memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
I: source-output.c: Created output 3 "Peak detect" on alsa_input.pci_1102_4_sound_card_0_alsa_capture_0 with sample spec float32le 1ch 25Hz and channel map mono
D: memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
D: memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
I: protocol-native.c: Final latency 60.00 ms = 40.00 ms + 20.00 ms
D: module-alsa-source.c: hwbuf_unused=0
D: module-alsa-source.c: setting avail_min=1
D: module-alsa-source.c: hwbuf_unused=0
D: module-alsa-source.c: setting avail_min=1
I: module-stream-restore.c: Storing volume/mute/device for stream source-output-by-application-id:org.PulseAudio.pavucontrol.
I: module-stream-restore.c: Synced.
I: client.c: Created 3 "Native client (UNIX socket client)"
D: protocol-native.c: Protocol version: remote 14, local 14
I: protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
D: protocol-native.c: SHM possible: yes
D: protocol-native.c: Negotiated SHM: yes
I: module-alsa-sink.c: Trying resume...
D: module-alsa-sink.c: hwbuf_unused=0
D: module-alsa-sink.c: setting avail_min=1
I: module-alsa-sink.c: Resumed successfully...
D: module-suspend-on-idle.c: Sink alsa_output.pci_1102_4_sound_card_0_alsa_playback_ 0 becomes idle.
D: module-suspend-on-idle.c: Sink alsa_output.pci_1102_4_sound_card_0_alsa_playback_ 0 becomes busy.
D: memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
D: memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
I: sink-input.c: Created input 0 "Playback Stream" on alsa_output.pci_1102_4_sound_card_0_alsa_playback_ 0 with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: protocol-native.c: Requested tlength=200.00 ms, minreq=10.00 ms
D: protocol-native.c: Adjust latency mode enabled, configuring sink latency to half of overall latency.
D: memblockq.c: memblockq requested: maxlength=70560, tlength=19404, base=4, prebuf=17644, minreq=1764 maxrewind=0
D: memblockq.c: memblockq sanitized: maxlength=70560, tlength=19404, base=4, prebuf=17644, minreq=1764 maxrewind=0
I: protocol-native.c: Final latency 200.00 ms = 90.00 ms + 2*10.00 ms + 90.00 ms
I: module-alsa-sink.c: Starting playback.
D: module-alsa-sink.c: hwbuf_unused=0
D: module-alsa-sink.c: setting avail_min=1
I: module-stream-restore.c: Storing volume/mute/device for stream sink-input-by-media-role:music.
D: module-suspend-on-idle.c: Source alsa_output.pci_1102_4_sound_card_0_alsa_playback_ 0.monitor becomes busy.
D: resampler.c: Channel matrix:
D: resampler.c: I00 I01
D: resampler.c: +------------
D: resampler.c: O00 | 1.000 1.000
I: resampler.c: Using resampler 'peaks'
I: resampler.c: Using float32le as working format.
D: memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
D: memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
I: source-output.c: Created output 4 "Peak detect" on alsa_output.pci_1102_4_sound_card_0_alsa_playback_ 0.monitor with sample spec float32le 1ch 25Hz and channel map mono
D: memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=4, prebuf=1, minreq=0 maxrewind=0
D: memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=4, prebuf=4, minreq=4 maxrewind=0
I: protocol-native.c: Final latency 60.00 ms = 40.00 ms + 20.00 ms
D: module-alsa-sink.c: hwbuf_unused=0
D: module-alsa-sink.c: setting avail_min=1
D: module-alsa-sink.c: hwbuf_unused=0
D: module-alsa-sink.c: setting avail_min=1
I: module-stream-restore.c: Storing volume/mute/device for stream source-output-by-application-id:org.PulseAudio.pavucontrol.
D: sink-input.c: Requesting rewind due to uncorking
D: module-suspend-on-idle.c: Sink alsa_output.pci_1102_4_sound_card_0_alsa_playback_ 0 becomes busy.
D: protocol-native.c: Requesting rewind due to end of underrun.
I: module-stream-restore.c: Synced.



Right. There's a problem. The output is still going on, but I can't find the top, it isn't available. And I can't work out where the bit above is in the output, as it all seems to be very similar. Should I run it with output to file?


UPDATE:
I left the thing running. It carried on for several hours, and was still going last time I looked. then i forgot about it and rebooted after trying to get autokey working properly again. If it's really needed I will start it running again first thing tomorrow morning, hopefully the output will be complete by the time i go to bed.

jrolland
October 20th, 2009, 06:49 PM
I'm sorry if this has already been posted, but I couldn't find it in my search.

I have a I have a Dell Dimension 2350 with a 1.7 GHz (single core) processor and 1 GB of RAM running Ubuntu Hardy Heron.

Using the on-board sound card, an Intel 82801DB-ICH4, has not yielded any sound whatsoever, no matter how many walkthroughs I follow, so I bought a new sound card, a Rocketfish sound card from Best Buy, and it appears to be a re-branded Creative Labs Sound Blaster Audigy LS SB0310 sound card.

When I type


alsamixer -Dhw

I get the output


alsamixer: function snd_ctl_open failed for hw: No such file or directory

When I type


user@machine:~$ speaker-test -Dplug:front -c2

I get the following output


speaker-test 1.0.15

Playback device is plug:front
Stream parameters are 48000Hz, S16_LE, 2 channels
Using 16 octaves of pink noise
ALSA lib confmisc.c:768:(parse_card) cannot find card 'I82801DBICH4'
ALSA lib conf.c:3513:(_snd_config_evaluate) function snd_func_card_driver returned error: No such device
ALSA lib confmisc.c:392:(snd_func_concat) error evaluating strings
ALSA lib conf.c:3513:(_snd_config_evaluate) function snd_func_concat returned error: No such device
ALSA lib confmisc.c:1251:(snd_func_refer) error evaluating name
ALSA lib conf.c:3513:(_snd_config_evaluate) function snd_func_refer returned error: No such device
ALSA lib conf.c:3985:(snd_config_expand) Evaluate error: No such device
ALSA lib pcm.c:2145:(snd_pcm_open_noupdate) Unknown PCM front
Playback open error: -19,No such device

I have disable the onboard sound card (which is what it appears to be trying to find) in BIOS.

Thanks in advance for any assistance you can provide.

theneoindian
October 20th, 2009, 06:54 PM
Thanx a bunch .. I finally got pulseaudio working on my system .... :-)

lemmy15
October 20th, 2009, 11:27 PM
Hello,

One thing I am unsure about (and hence afraid to make any changes) is the difference between 32-bit Ubuntu (which I have) and 64-bit (mentioned in post #1, Part B: Hardy Heron (8.04)). Just so I can be sure, to make the changes, should I just skip item #0, or is the whole process nullified because I have 32-bit?

I've experienced the things mentioned here. And when we use our Skype, we can talk for, maybe, five minutes before the other end gets a message stating that there's a problem with the audio card, whereas on our end the message we get is something about the audio back (I'm getting this info second-hand, so pardon me if it's not verbatim). It's also possible that this Skype problem is unrelated, but I thought I would mention it.

THANKS in advance!

lemmy

psyke83
October 24th, 2009, 09:32 PM
Edit: the Karmic PulseAudio equalizer now has a new home (http://ubuntuforums.org/showthread.php?t=1308838)!

tiggsy
October 25th, 2009, 08:51 PM
Do you have anything to turn the sound on?

topdownjimmy
October 27th, 2009, 03:41 PM
Why not put this in the user documentation wiki? Isn't this precisely the kind of thing the wiki is for? What if psyke83 suddenly disappears or stops using Ubuntu? Somebody has to duplicate these instructions in another thread and update it themselves?

Why are the forums the primary source of Ubuntu documentation? Forums are for discussion.

psyke83
October 27th, 2009, 07:34 PM
Why not put this in the user documentation wiki? Isn't this precisely the kind of thing the wiki is for? What if psyke83 suddenly disappears or stops using Ubuntu? Somebody has to duplicate these instructions in another thread and update it themselves?

Why are the forums the primary source of Ubuntu documentation? Forums are for discussion.

Parts of the guide are not appropriate for a wiki, as they use packages from my PPA. I don't want that posted elsewhere.

Most of the wiki page quite out of date. I'm not too interested in helping to maintain that wiki page, because less-informed users may re-add useless information (such as the libflashsupport junk that's still listed).

tiggsy
October 30th, 2009, 08:19 PM
I'm now proud user of Karmic Koala - but still with NO SOUND.

How do i get the sound working on KK?

topdownjimmy
October 30th, 2009, 11:23 PM
Most of the wiki page quite out of date. I'm not too interested in helping to maintain that wiki page, because less-informed users may re-add useless information (such as the libflashsupport junk that's still listed).

Isn't that the point though? Wikis are meant to be improved. They're out of date because of people's indifference toward them, but they are the more appropriate medium for this type of documentation.

Logan 1229
October 31st, 2009, 04:51 PM
I'm now proud user of Karmic Koala - but still with NO SOUND.

How do i get the sound working on KK?

Had same problem. Tried this guide & it created more problems (applet disappeared for logging off,etc, amoung other issues).

After re-installing fresh, discovered I was missing a driver for my Audigy 4 sound card (drivers for Audigy 2 card are same). Then discovered that in 'Sound Preferences', the default setting (under 'Output' tab) was set to my integrated motherboard audio. Switched it to my sound card & had sound back. Note though, that both may integrated audio & the sound card descriptions were the same but were listed separately as two devices.

Lastly, & am I am gloriously happy to say this, my sound quality level increased by 50%!! I was considering going back to Windows to get the sound quality level back but now I don't need to! To all who contributed to this release -- great job & thank you very much!

sbersier
October 31st, 2009, 08:13 PM
Here is a 15-bands system-wide graphical user interface for the pulseaudio-equalizer.sh script running under Ubuntu 9.10 Karmic Koala (not tested on other distributions).

This post has been moved to a new thread: http://ubuntuforums.org/showpost.php?p=8248217&postcount=2 (http://ubuntuforums.org/showthread.php?t=1308838)
Regards,
Steph

blacksm1th
November 1st, 2009, 02:04 AM
@psyke83 It work but only for one audio device. I have two devices - speakers and headset. Is it possible for other device to use equalizer? This is the situation now:
http://img33.imageshack.us/img33/930/screenshotjv.th.png (http://img33.imageshack.us/i/screenshotjv.png/)

@sbersier It work but interface did not load previous selected values.

Thank you both for the efforts.

psyke83
November 1st, 2009, 02:19 AM
@psyke83 It work but only for one audio device. I have two devices - speakers and headset. Is it possible for other device to use equalizer? This is the situation now:
http://img33.imageshack.us/img33/930/screenshotjv.th.png (http://img33.imageshack.us/i/screenshotjv.png/)

@sbersier It work but interface did not load previous selected values.

Thank you both for the efforts.

Well, the LADSPA sink will use a master sink, which is the default sink chosen by PulseAudio.

I can change the script so that it will use the user-selected sink as the master for the LADSPA sink, but I'm not sure if that's going to help in your case.

Edit: actually, the script should do this already. Whatever sink is currently used as default by PulseAudio is used for the equalizer sink, when you execute the script.

blacksm1th
November 1st, 2009, 03:37 AM
Yes it work now. :) The key option is "Set as fallback" in "Output devices".

sbersier
November 1st, 2009, 07:22 AM
@psyke83 It work but only for one audio device. I have two devices - speakers and headset. Is it possible for other device to use equalizer? This is the situation now:
http://img33.imageshack.us/img33/930/screenshotjv.th.png (http://img33.imageshack.us/i/screenshotjv.png/)

@sbersier It work but interface did not load previous selected values.

Thank you both for the efforts.
Yes, indeed I have to change this. It was a first move.
The list of things to do:
1) Make the GUI interface read the .equalizerrc file (which contains the settings) at launch
2) Make a setup.sh script in order to install everything in a more simple way
3) Make the GUI (maybe) a bit more sexy... For example by adding a possibility of loading (user) predefined settings - headphones, speaker, movie, rock,...

Regards.

sbersier
November 1st, 2009, 09:43 AM
@psyke83 It work but only for one audio device. I have two devices - speakers and headset. Is it possible for other device to use equalizer? This is the situation now:
http://img33.imageshack.us/img33/930/screenshotjv.th.png (http://img33.imageshack.us/i/screenshotjv.png/)

@sbersier It work but interface did not load previous selected values.

Thank you both for the efforts.
OK. I've modified the java code in order to load existing settings at launch.
You just have to delete the $HOME/.equalizerrc and replace the previous equalizer.java by the new one (on the same post: http://ubuntuforums.org/showpost.php?p=8207442&postcount=1527 ).
Then compile it the same way as before.
Regards.

tipiglen
November 1st, 2009, 04:47 PM
Thanks for all that effort. Seems to work, but I can only get it up via terminal. Any idea how it might be called from a menu item?

Thanks in advance
ed
(listening to Horace Silver)

sbersier
November 1st, 2009, 05:47 PM
Thanks for all that effort. Seems to work, but I can only get it up via terminal. Any idea how it might be called from a menu item?

Thanks in advance
ed
(listening to Horace Silver)
Sorry, for the moment I don't see how... Indeed, it would be much more practical! If anyone has an idea he/she's welcome.
Regards.

tipiglen
November 1st, 2009, 06:30 PM
OK, I've done it:

1, make a script and save it as 'loadequalizer' wherever you want.
I saved it in my /home/ed/bin directory.

#!/bin/sh

# NEED_SYMLINK

# this shell script has to be named "loadequalizer" to
# launch Equalizer by way of /usr/local/bin symlink

java equalizer & "$@"

2. (as root) make a symbolic link (to wherever you've saved the wee script)

cd /usr/local/bin
ln -s /home/ed/bin/loadequalizer equalizer


3. Quit being root (for safety's sake)

4. in the main GUI, go to programs>system>main menu
and choose sound & Video
Make a new item with command: 'equalizer %u'

That should do it. Works for me

sbersier
November 1st, 2009, 09:30 PM
OK, I've done it:

1, make a script and save it as 'loadequalizer' wherever you want.
I saved it in my /home/ed/bin directory.

#!/bin/sh

# NEED_SYMLINK

# this shell script has to be named "loadequalizer" to
# launch Equalizer by way of /usr/local/bin symlink

java equalizer & "$@"2. (as root) make a symbolic link (to wherever you've saved the wee script)

cd /usr/local/bin
ln -s /home/ed/bin/loadequalizer equalizer
3. Quit being root (for safety's sake)

4. in the main GUI, go to programs>system>main menu
and choose sound & Video
Make a new item with command: 'equalizer %u'

That should do it. Works for me
Hello,
I followed carefully your procedure (for 3-4 times). Unfortunately, for me it doesn't work.
I don't know why... In fact, the point is that I can't execute the result of 'javac xxx.java' (let's call it 'xxx') with the 'java xxx' command when I issue it from a different directory than the one containing the 'xxx*.class' classes. Weird... I can show you what I get:


steph@steph-desktop:/usr/local/bin$ sudo ln -s /home/steph/bin/EQUALIZER/loadequalizer equalizer
steph@steph-desktop:/usr/local/bin$ ls -l equalizer
lrwxrwxrwx 1 root root 39 2009-11-01 21:00 equalizer -> /home/steph/bin/EQUALIZER/loadequalizer
(so the link exists and is executable)
steph@steph-desktop:/usr/local/bin$ cd /home/steph/bin/EQUALIZER/
steph@steph-desktop:~/bin/EQUALIZER$ ls -l
total 64
-rw-r--r-- 1 steph steph 363 2009-11-01 20:58 equalizer.class
-rw-r--r-- 1 steph steph 509 2009-11-01 20:58 equalizerFrame$1.class
-rw-r--r-- 1 steph steph 2027 2009-11-01 20:58 equalizerFrame$2.class
-rw-r--r-- 1 steph steph 536 2009-11-01 20:58 equalizerFrame$3.class
-rw-r--r-- 1 steph steph 2310 2009-11-01 20:58 equalizerFrame$4.class
-rw-r--r-- 1 steph steph 6064 2009-11-01 20:58 equalizerFrame.class
-rw-r--r-- 1 steph steph 13770 2009-11-01 20:58 equalizer.java
-rw-r--r-- 1 steph steph 13772 2009-11-01 20:58 equalizer.java~
-rwxr-xr-x 1 steph steph 158 2009-11-01 20:58 loadequalizer
(everything is there...)
steph@steph-desktop:~/bin/EQUALIZER$ equalizer
(It works. Ok. Let's close equalizer and move up from one directory...)
steph@steph-desktop:~/bin/EQUALIZER$ cd ..
steph@steph-desktop:~/bin$ equalizer (The result is shown below)
steph@steph-desktop:~/bin$ Exception in thread "main" java.lang.NoClassDefFoundError: equalizer
Caused by: java.lang.ClassNotFoundException: equalizer
at java.net.URLClassLoader$1.run(URLClassLoader.java: 217)
at java.security.AccessController.doPrivileged(Native Method)
at java.net.URLClassLoader.findClass(URLClassLoader.j ava:205)
at java.lang.ClassLoader.loadClass(ClassLoader.java:3 23)
at sun.misc.Launcher$AppClassLoader.loadClass(Launche r.java:294)
at java.lang.ClassLoader.loadClass(ClassLoader.java:2 6
at java.lang.ClassLoader.loadClassInternal(ClassLoade r.java:336)
Could not find the main class: equalizer. Program will exit.

Putting the link in the Main menu doesn't help. Nothing happens. In fact, the same happens than above but since there is no terminal window associated with it, I don't see anything.
My Ubuntu 9.10 Karmic was recently installed from scratch (I didn't upgrade from 9.04). So, it might be that I miss something in my present installation.
Best regards,
Steph

blacksm1th
November 1st, 2009, 09:41 PM
I use this script to run EQ:

#!/bin/sh
cd /home/azot/equalizer/ # <- this path must be edited
java equalizer &
And with shortcut on panel with this:

/home/azot/equalizer/run-eq.sh
in command textbox.

sbersier
November 1st, 2009, 09:56 PM
Yep! It works perfectly. Thanks a lot. I will modify my original post according to your suggestion.
Best regards,
Steph

undertakingyou
November 2nd, 2009, 01:13 AM
It may be of note that with the Audigy 2 soundcard that I have (and I assume all the Audigy cards that use the same driver snd-emu10k1) also requires one additional thing to make the audio work correctly.

In a terminal:

alsamixer

Go over to the Audigy Analog/Digital output jack and press 'm' to switch it on.
Press escape to leave alsamixer.
Logout and then Login.

tipiglen
November 2nd, 2009, 01:36 AM
Blacksmith, A similar solution, and simpler than mine, And tidier.

Steph, Glad it works for you

Thanks to both of you

http://home2.btconnect.com/tipiglen/loveandpeace3.gif
<b>Salaam/Shalom/Shanthi/Peace</b>

ed

GNUbee40
November 2nd, 2009, 01:15 PM
This Q might come as slightly unrelated:

I use a config file to load Alsa sound settings at every startup. This is a workaround for PulseAudio Muting all sound at every boot.

However I have completely forgotten how I proceeded back then. Does anybody know which file it could have been? Have searched and searched to no avail...

Edit: Finally found out - no need to answer this

tjccjt
November 2nd, 2009, 08:47 PM
I've recently upgraded my HP2133 from 9.04 to 9.10. Audio playback is not working.

The hardware device appears for a few seconds in the Output tab of the volume control applet but then vanishes.

Does anyone have a fix for this? I've tried the guide here but it has had no effect.

Thanks

Tony



tjc@shiny:~$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: VT82xx [HDA VIA VT82xx], device 0: AD198x Analog [AD198x Analog]
Subdevices: 1/1
I: client.c: Created 5 "Native client (UNIX socket client)"
I: client.c: Freed 5 "Native client (UNIX socket client)"
I: protocol-native.c: Connection died.
Subdevice #0: subdevice #0




E: alsa-util.c: snd_pcm_delay() returned a value that is exceptionally large: -641504 bytes (-3636 ms).
E: alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers.
E: alsa-util.c: snd_pcm_dump():

brjoon1021
November 3rd, 2009, 07:35 PM
I want to install Karmic from the minimal installation CD with the Gnome-core package, synaptic and a few choice apps.

What packages should I install to have the BEST working sound with pulseaudio that I can have for Mplayer, Flash video via Firefox, CDs and MP3's.

Should I enable any additional repositories or PPA's before installing sound. As soon as I a working desktop I will add to synaptic whatever you advise.

Thanks,

B

trayzz
November 4th, 2009, 02:20 AM
I found a link to this thread somewhere when I was looking for a fix how to get rid of this error message in my terminal

bt_audio_service_open: connect() failed: Connection refused (111)
i followed part A for karmic koala. My problem is, that when I open pavucontrol, i get an error message "connection failed: permission denied" and this in my terminal

E: socket-server.c: bind(): Address already in use
E: module.c: Failed to load module "module-esound-protocol-unix" (argument: ""): initialization failed.
E: main.c: Module load failed.
E: main.c: Failed to initialize daemon.
The application still opens though. strange is then that there are no devices listed: neither for input, output, nor for devices using the driver, even though rhythmbox is running.

..any suggestions?

tiggsy
November 4th, 2009, 03:09 AM
It looks as if i'm going to have to go back to Hardy, the last version where I had functional sound. I am not a happy tigger.

psyke83
November 4th, 2009, 09:27 AM
Here is a 15-bands system-wide graphical user interface for the pulseaudio-equalizer.sh script running under Ubuntu 9.10 Karmic Koala (not tested on other distributions).

This is a java applet which makes use of the MODIFIED (see below) pulseaudio-equalizer.sh shell script (see http://ubuntuforums.org/showpost.php?p=8159199&postcount=1520 )

Although I haven't personally tested your work (I don't have any Java development packages installed, I'm not a huge fan of Java due to limited memory on my laptop and I don't like to see non-native widgets on interfaces), I really appreciate your effort in coding this GUI for the script. :)

I've updated the script to v1.6. What's relevant to you from this update is the following:


The script will now check for the presence of an ~/.equalizerrc file and read the control settings automatically - so there's no longer any need to ask users to edit my script.
Your GUI could possibly take advantage of the new "always-on" / "always-off" options. You could modify the interface to include a "Save" and/or Reset button associated to these options. Users can then make permanent changes without having to launch your program (or my script script) upon each session.
Also, I think you mentioned something about pre-set equalizer values (rock, pop, classical, etc.). All you need is to figure out the control values suitable for the profiles (that's the hard bit), and then copy the values to the ~/.equalizerrc file and apply changes depending on the profile a user chooses from your interface.


Finally, a suggestion. The next time you edit your post, I recommend that you encapsulate your code using the CODE tags (see the # icon on the formatting toolbar), as it will make things much neater and prevent any loss of formatting.

sbersier
November 4th, 2009, 01:24 PM
Although I haven't personally tested your work (I don't have any Java development packages installed, I'm not a huge fan of Java due to limited memory on my laptop and I don't like to see non-native widgets on interfaces), I really appreciate your effort in coding this GUI for the script. :)

I've updated the script to v1.6. What's relevant to you from this update is the following:


The script will now check for the presence of an ~/.equalizerrc file and read the control settings automatically - so there's no longer any need to ask users to edit my script.
Your GUI could possibly take advantage of the new "always-on" / "always-off" options. You could modify the interface to include a "Save" and/or Reset button associated to these options. Users can then make permanent changes without having to launch your program (or my script script) upon each session.
Also, I think you mentioned something about pre-set equalizer values (rock, pop, classical, etc.). All you need is to figure out the control values suitable for the profiles (that's the hard bit), and then copy the values to the ~/.equalizerrc file and apply changes depending on the profile a user chooses from your interface.


Finally, a suggestion. The next time you edit your post, I recommend that you encapsulate your code using the CODE tags (see the # icon on the formatting toolbar), as it will make things much neater and prevent any loss of formatting.
I've made the changes according to your suggestions and changed my original post.
I still have to think a bit about user-defined profiles (headphone, speakers, ....) But anyway, I didn't think about predefining myself these profiles but intended to give the user the possibility of defining them him/herself and naming it as he/she wants.

Best regards,
Steph

blacksm1th
November 4th, 2009, 03:34 PM
I want to install Karmic from the minimal installation CD with the Gnome-core package, synaptic and a few choice apps.

What packages should I install to have the BEST working sound with pulseaudio that I can have for Mplayer, Flash video via Firefox, CDs and MP3's.

Should I enable any additional repositories or PPA's before installing sound. As soon as I a working desktop I will add to synaptic whatever you advise.

Thanks,

B
Just follow the part A from first post.

psyke83
November 4th, 2009, 09:17 PM
I've made the changes according to your suggestions and changed my original post.
I still have to think a bit about user-defined profiles (headphone, speakers, ....) But anyway, I didn't think about predefining myself these profiles but intended to give the user the possibility of defining them him/herself and naming it as he/she wants.

Best regards,
Steph

Thanks, I've finally tested your interface.

Some comments:

Since v1.7 I renamed the script options (on -> enable, off -> disable, always-on -> enable-config, always-off -> disable-config), so you'll need to update the source. I renamed the functions and options in your code to conform with v1.7 and later of my script.
I also modified your script to read the user's home directory automatically:

public static String Home=System.getProperty("user.home");
public static String filename=Home+"/.equalizerrc";
public static File default_pa=new File (Home+"/.pulse/default.pa");

Suggestions:

Your interface detects a "persistent" configuration merely if the file ~/.pulse/default.pa exists. This is not necessarily accurate, as some users may have a default.pa with customizations unrelated to the equalizer. I've added better detection to the actual script since v1.7, which perhaps you could somehow take advantage of?
If the ~/.equalizerrc file does not exist, your interface will not use the preset values from the script. Is this what you intended? I've added a new undocumented option to v1.8, "write-equalizerrc". Basically, if the ~/.equalizerrc file does not exist, that command will create the file using the preset values from my script. You may want to update your interface with that in mind. I've also added "delete-equalizerrc" for convenience (and if needed).
In a similar vein to the previous point, the new "write-equalizerrc" option could serve as a way of implementing a "Reset to defaults" button on your interface.
When a "Persistent" configuration is already enabled, and you click the "Persistent" button to disable the configuration (i.e., disable-config), the equalizer will also be disabled on the running PulseAudio server. I suggest that you change your script to run "pulseaudio-equalizer.sh enable" immediately after "pulseaudio-equalizer.sh disable-config".
Why not create a JAR file and attach to your post along with the source? This should do it:

conn@inspiron:~/work$ javac equalizer.java
conn@inspiron:~/work$ jar cfe equalizer.jar equalizer equalizer*.class

Then you can run the program like so, without the need for wrapper scripts:

conn@inspiron:~/work$ java -jar equalizer.jar


For your convenience, I'm posting the full modified code below (not including the suggested changes, as I'm not familiar enough with Java to change myself).

Oh and by the way, I moved the script to its own thread (http://ubuntuforums.org/showthread.php?t=1308838), you may want to post any further updates to your interface there... ;)


/*** @version 1.3 2009-11-4
* @author Stéphane Bersier
This java applet shows an equalizer and simply run pulseaudio-equalizer.sh
shell script from Conn O'Griofa.
For more information on pulseaudio-equalizer.sh
http://ubuntuforums.org/showpost.php?p=8159199&postcount=1520
Also part of the code was taken and modified from an example by Cay Horstmann
http://www.java2s.com/Code/Java/Swing-JFC/SliderTest.htm
And from:
http://www.roseindia.net/java/beginners/java-read-file-line-by-line.shtml

The resulting code is quiet ugly and it should be considered as my own fault...

*/

import java.awt.*;
import java.awt.event.*;
import java.util.*;
import javax.swing.*;
import javax.swing.event.*;
import java.io.*;


public class equalizer {

public static void main(String[] args)
{ JFrame frame = new equalizerFrame();
frame.setVisible(true);
}
}



class equalizerFrame extends JFrame
{
public static String Home=System.getProperty("user.home");
public static String filename=Home+"/.equalizerrc";
public static File default_pa=new File (Home+"/.pulse/default.pa");

public static int val_slider;
public static int source_slider;
public static int slider1_source;
public static int slider2_source;
public static int slider3_source;
public static int slider4_source;
public static int slider5_source;
public static int slider6_source;
public static int slider7_source;
public static int slider8_source;
public static int slider9_source;
public static int slider10_source;
public static int slider11_source;
public static int slider12_source;
public static int slider13_source;
public static int slider14_source;
public static int slider15_source;

public static int slider1_value;
public static int slider2_value;
public static int slider3_value;
public static int slider4_value;
public static int slider5_value;
public static int slider6_value;
public static int slider7_value;
public static int slider8_value;
public static int slider9_value;
public static int slider10_value;
public static int slider11_value;
public static int slider12_value;
public static int slider13_value;
public static int slider14_value;
public static int slider15_value;
public static String token;
public static String[] tokens;
public static int enable_config;
public static int[] sliders_values;

public static void initializeSettings(){
sliders_values=new int[16];
for (int i = 1; i < 16; i = i+1) {
sliders_values[i]=0;
}
}

public static void readSettings()
{
try{
FileInputStream fstream = new FileInputStream(filename);
// Get the object of DataInputStream
DataInputStream in = new DataInputStream(fstream);
BufferedReader br = new BufferedReader(new InputStreamReader(in));
String strLine;
//

while ((strLine = br.readLine()) != null) {
// Print the content on the console
int i=0;
tokens=strLine.split(",");
String result="";
for (String token : tokens){
result+=(token+"-");
i=i+1;
sliders_values[i]=Integer.parseInt(token);
;}

}
//Close the input stream
in.close();
}catch (Exception e){


}
}
public equalizerFrame()
{ setTitle("Equalizer");
setSize(1000, 300);
addWindowListener(new WindowAdapter()
{ public void windowClosing(WindowEvent e)
{ System.exit(0);
}
} );

// set up grid bag layout and constraints
Container cp=getContentPane();
cp.setLayout(new GridBagLayout());
constraints = new GridBagConstraints();
constraints.weighty = 100;
constraints.gridwidth = 1;
constraints.gridheight = 1;
constraints.gridx = 0;
constraints.gridy = 0;



// add sliders with various decorations

// First initialize settings to 0 and then read existing settings file (if it exist)
initializeSettings();
readSettings();
slider1_value=sliders_values[1];
slider2_value=sliders_values[2];
slider3_value=sliders_values[3];
slider4_value=sliders_values[4];
slider5_value=sliders_values[5];
slider6_value=sliders_values[6];
slider7_value=sliders_values[7];
slider8_value=sliders_values[8];
slider9_value=sliders_values[9];
slider10_value=sliders_values[10];
slider11_value=sliders_values[11];
slider12_value=sliders_values[12];
slider13_value=sliders_values[13];
slider14_value=sliders_values[14];
slider15_value=sliders_values[15];



JSlider slider1 = new JSlider(JSlider.VERTICAL,-70,30, sliders_values[1]);
addSlider(slider1, "50Hz");
slider1_source=slider1.hashCode();
slider1.setMajorTickSpacing(10);
slider1.setPaintTicks(true);

JSlider slider2 = new JSlider(JSlider.VERTICAL,-70,30,sliders_values[2]);
addSlider(slider2, "100Hz");
slider2_source=slider2.hashCode();
slider2.setMajorTickSpacing(10);
slider2.setPaintTicks(true);

JSlider slider3 = new JSlider(JSlider.VERTICAL,-70,30,sliders_values[3]);
addSlider(slider3, "156Hz");
slider3_source=slider3.hashCode();
slider3.setMajorTickSpacing(10);
slider3.setPaintTicks(true);

JSlider slider4 = new JSlider(JSlider.VERTICAL,-70,30,sliders_values[4]);
addSlider(slider4, "220Hz");
slider4_source=slider4.hashCode();
slider4.setMajorTickSpacing(10);
slider4.setPaintTicks(true);

JSlider slider5 = new JSlider(JSlider.VERTICAL,-70,30,sliders_values[5]);
addSlider(slider5, "311Hz");
slider5_source=slider5.hashCode();
slider5.setMajorTickSpacing(10);
slider5.setPaintTicks(true);

JSlider slider6 = new JSlider(JSlider.VERTICAL,-70,30,sliders_values[6]);
addSlider(slider6, "440Hz");
slider6_source=slider6.hashCode();
slider6.setMajorTickSpacing(10);
slider6.setPaintTicks(true);

JSlider slider7 = new JSlider(JSlider.VERTICAL,-70,30,sliders_values[7]);
addSlider(slider7, "622Hz");
slider7_source=slider7.hashCode();
slider7.setMajorTickSpacing(10);
slider7.setPaintTicks(true);

JSlider slider8 = new JSlider(JSlider.VERTICAL,-70,30,sliders_values[8]);
addSlider(slider8, "880Hz");
slider8_source=slider8.hashCode();
slider8.setMajorTickSpacing(10);
slider8.setPaintTicks(true);

JSlider slider9 = new JSlider(JSlider.VERTICAL,-70,30,sliders_values[9]);
addSlider(slider9, "1250Hz");
slider9_source=slider9.hashCode();
slider9.setMajorTickSpacing(10);
slider9.setPaintTicks(true);

JSlider slider10 = new JSlider(JSlider.VERTICAL,-70,30,sliders_values[10]);
addSlider(slider10, "1750Hz");
slider10_source=slider10.hashCode();
slider10.setMajorTickSpacing(10);
slider10.setPaintTicks(true);

JSlider slider11 = new JSlider(JSlider.VERTICAL,-70,30,sliders_values[11]);
addSlider(slider11, "2.5kHz");
slider11_source=slider11.hashCode();
slider11.setMajorTickSpacing(10);
slider11.setPaintTicks(true);

JSlider slider12 = new JSlider(JSlider.VERTICAL,-70,30,sliders_values[12]);
addSlider(slider12, "3.5kHz");
slider12_source=slider12.hashCode();
slider12.setMajorTickSpacing(10);
slider12.setPaintTicks(true);

JSlider slider13 = new JSlider(JSlider.VERTICAL,-70,30,sliders_values[13]);
addSlider(slider13, "5kHz");
slider13_source=slider13.hashCode();
slider13.setMajorTickSpacing(10);
slider13.setPaintTicks(true);

JSlider slider14 = new JSlider(JSlider.VERTICAL,-70,30,sliders_values[14]);
addSlider(slider14, "10kHz");
slider14_source=slider14.hashCode();
slider14.setMajorTickSpacing(10);
slider14.setPaintTicks(true);

JSlider slider15 = new JSlider(JSlider.VERTICAL,-70,30,sliders_values[15]);
addSlider(slider15, "20kHz");
slider15_source=slider15.hashCode();
slider15.setMajorTickSpacing(10);
slider15.setPaintTicks(true);
slider15.setPaintLabels(true);

JButton Apply = new JButton("Apply");
ActionListener actionListenerapply =
new ActionListener() {
public void actionPerformed(ActionEvent actionEvent)
{
try {
BufferedWriter out = new BufferedWriter(new FileWriter(filename));
out.write(""+slider1_value+","
+slider2_value+","
+slider3_value+","
+slider4_value+","
+slider5_value+","
+slider6_value+","
+slider7_value+","
+slider8_value+","
+slider9_value+","
+slider10_value+","
+slider11_value+","
+slider12_value+","
+slider13_value+","
+slider14_value+","
+slider15_value);

out.close();
}
catch (IOException e) {
System.out.println("exception happened - here's what I know: ");
e.printStackTrace();
System.exit(-1);
}
try{ Process p = Runtime.getRuntime().exec("pulseaudio-equalizer.sh enable");
}
catch (IOException e) {
System.out.println("exception happened - here's what I know: ");
e.printStackTrace();
System.exit(-1);}
};
};


Apply.addActionListener(actionListenerapply);
cp.add(Apply);

JToggleButton EnableConfig = new JToggleButton("Persistent",default_pa.exists());

ActionListener actionListenerEnableConfig = new ActionListener() {
public void actionPerformed(ActionEvent actionEvent){

if (default_pa.exists()){
try{ Process p = Runtime.getRuntime().exec("pulseaudio-equalizer.sh disable-config");
}
catch (IOException e) {
System.out.println("exception happened - here's what I know: ");
e.printStackTrace();
System.exit(-1);}
}
else {
try{ Process p = Runtime.getRuntime().exec("pulseaudio-equalizer.sh enable-config");
}
catch (IOException e) {
System.out.println("exception happened - here's what I know: ");
e.printStackTrace();
System.exit(-1);}}

}
};

EnableConfig.addActionListener(actionListenerEnabl eConfig);

cp.add(EnableConfig);

JButton Quit = new JButton("Quit");
ActionListener actionListenerquit = new ActionListener() {
public void actionPerformed(ActionEvent actionEvent) {
System.exit(0);
}
};
Quit.addActionListener(actionListenerquit);
cp.add(Quit);
}

public void addSlider(JSlider s, String description)
{ // create text field that is shown next to slider
final TextField textField = new TextField("",4);

// update text field when the slider value changes
s.addChangeListener(new ChangeListener()
{ public void stateChanged(ChangeEvent event)
{ JSlider source = (JSlider)event.getSource();
val_slider=source.getValue();
val_slider=val_slider;
int slider_source = source.hashCode();
if (slider_source==slider1_source){
slider1_value=val_slider;
}
if (slider_source==slider2_source){
slider2_value=val_slider;
}
if (slider_source==slider3_source){
slider3_value=val_slider;
}
if (slider_source==slider4_source){
slider4_value=val_slider;
}
if (slider_source==slider5_source){
slider5_value=val_slider;
}
if (slider_source==slider6_source){
slider6_value=val_slider;
}
if (slider_source==slider7_source){
slider7_value=val_slider;
}
if (slider_source==slider8_source){
slider8_value=val_slider;
}
if (slider_source==slider9_source){
slider9_value=val_slider;
}
if (slider_source==slider10_source){
slider10_value=val_slider;
}
if (slider_source==slider11_source){
slider11_value=val_slider;
}
if (slider_source==slider12_source){
slider12_value=val_slider;
}
if (slider_source==slider13_source){
slider13_value=val_slider;
}
if (slider_source==slider14_source){
slider14_value=val_slider;
}
if (slider_source==slider15_source){
slider15_value=val_slider;
}
textField.setText("" + val_slider);
}
});

// add three components into the next row

constraints.gridy = 0;
constraints.anchor = GridBagConstraints.WEST;
constraints.fill = GridBagConstraints.NONE;
constraints.weightx = 0;
getContentPane().add(new JLabel(description), constraints);

constraints.gridy++;
constraints.anchor = GridBagConstraints.CENTER;
constraints.fill = GridBagConstraints.HORIZONTAL;
constraints.weightx = 100;
getContentPane().add(s, constraints);

constraints.gridy++;
constraints.anchor = GridBagConstraints.WEST;
constraints.fill = GridBagConstraints.NONE;
constraints.weightx = 0;
getContentPane().add(textField, constraints);

// advance row
constraints.gridx++;
}

private GridBagConstraints constraints;
}

blacksm1th
November 5th, 2009, 12:52 PM
@psyke83 This "toggle" option is a nice feature. It give me an idea for script that can change the default output device. This script can be used with launcher on panel. If user have 3 output devices in a worst case scenario with 2 clicks the desired device will be selected. Can you make that kind of script?
Regards

sbersier
November 5th, 2009, 01:18 PM
OK. So...
- There is no more reference to an existing or non-existing default.pa file. It performs a "pulseaudio-equalizer status" operation.
- I didn't make use of the write-equalizerrc feature since I prefer to start in a neutral position (zeros everywhere)
- I separated in the GUI the two functions: (1)(enable, disable) and (2)(enable-config, disable-config)
It means that you can select the "Keep config" option without necessarily starting the equalizer
and as well you can deselect the "Keep config" option without stopping the equalizer
I did this because it was easier for me (I'm not a programmer. In fact, this is my first java program...)
- I've build a equalizer.jar and modified the install instructions.

The result is posted on the new thread and I will no more interfere in this one. My first post will be obsolete and will point to the new thread.
Thanks and best regards.
Steph

llenchikk
November 5th, 2009, 07:36 PM
Hello!
I boot Ubuntu 9.10 on LiveCD. There was no problems with play/capture sound on my notebook.
I installed ubuntu on hard disc. And there is no sound capture... But playing sounds is ok in all applications. Updates from repo change nothing. Sound capture don't work. Capture monitor show nothing. All work when boot from LiveCD.
There is bug https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/460351. (https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/460351)
I make all steps for Karmic Koala in first message this thread.
Where is different in pulseaudio configuration at LiveCD and right after installation?
What can I do for check the problem? I'll be glad any ideas!

gareim
November 6th, 2009, 07:59 AM
This is a great guide, and after following it, most of my sound problems were fixed. There is one problem though, and it's been following me since 8.10. I can't have multiple applications playing sound at the same time. I have to close Amarok to have sound on Firefox, and after following this guide, it worked (had Firefox, Amarok, and VLC), but after some logging in and out, this no longer works. Any ideas for what might be wrong? Thanks a lot. =]

Edit: Ok, it turns out that Firefox gets no sound at all, while VLC and Amarok can both have sound at the same time. Does this mean that Firefox isn't working with Pulseaudio??

RayArdia
November 6th, 2009, 11:47 AM
I 'theeeenk' I have followed most of the excellent guide, but before I go to the Equalizer bit I'd be grateful for comments on my 'Notes so far':-


Pulse Audio/Skype Problems
Using Karmic on Acer Aspire 5735, 3GB RAM 250GB HDD.
Skype seems to function OK, test call OK but when calling a contact can see and hear them (tho they can't hear me), but as soon as I switch my video on, we break contact.
Sound Devices:-aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: Intel [HDA Intel], device 0: ALC268 Analog [ALC268 Analog]
Subdevices: 0/1
Subdevice #0: subdevice #0
Verbose output:-
~$ pkill pulseaudio; sleep 2; pulseaudio -vv
I: main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted
I: main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation not permitted
D: core-rtclock.c: Timer slack is set to 50 us.
I: core-util.c: Failed to acquire high-priority scheduling: No such file or directory
I: main.c: This is PulseAudio 0.9.19
D: main.c: Compilation host: i486-pc-linux-gnu
D: main.c: Compilation CFLAGS: -g -O2 -g -Wall -O3 -Wall -W -Wextra -pipe -Wno-long-long -Winline -Wvla -Wno-overlength-strings -Wunsafe-loop-optimizations -Wundef -Wformat=2 -Wlogical-op -Wsign-compare -Wformat-security -Wmissing-include-dirs -Wformat-nonliteral -Wold-style-definition -Wpointer-arith -Winit-self -Wdeclaration-after-statement -Wfloat-equal -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wmissing-declarations -Wmissing-noreturn -Wshadow -Wendif-labels -Wcast-align -Wstrict-aliasing=2 -Wwrite-strings -Wno-unused-parameter -ffast-math -Wp,-D_FORTIFY_SOURCE=2 -fno-common -fdiagnostics-show-option
D: main.c: Running on host: Linux i686 2.6.31-14-generic #48-Ubuntu SMP Fri Oct 16 14:04:26 UTC 2009
D: main.c: Found 2 CPUs.
I: main.c: Page size is 4096 bytes
D: main.c: Compiled with Valgrind support: no
D: main.c: Running in valgrind mode: no
D: main.c: Optimized build: yes
D: main.c: All asserts enabled.
I: main.c: Machine ID is 4854c8e7f9f878813720c37a4a5e64a9.
I: main.c: Session ID is 4854c8e7f9f878813720c37a4a5e64a9-1257450588.198234-2042037969.
I: main.c: Using runtime directory /home/ray/.pulse/4854c8e7f9f878813720c37a4a5e64a9-runtime.
I: main.c: Using state directory /home/ray/.pulse.
I: main.c: Using modules directory /usr/lib/pulse-0.9.19/modules.
I: main.c: Running in system mode: no
E: pid.c: Daemon already running.
E: main.c: pa_pid_file_create() failed.
Installed packages for equalizer:_
Reading package lists... Done
Building dependency tree
Reading state information... Done
The following packages were automatically installed and are no longer required:
ttf-wqy-zenhei binutils-static
Use 'apt-get autoremove' to remove them.
The following NEW packages will be installed
ladspa-sdk swh-plugins
0 upgraded, 2 newly installed, 0 to remove and 25 not upgraded.
Need to get 541kB of archives.
After this operation, 2,351kB of additional disk space will be used.
Get: 1 http://es.archive.ubuntu.com karmic/universe ladspa-sdk 1.1-6 [39.6kB]
Get: 2 http://es.archive.ubuntu.com karmic/universe swh-plugins 0.4.15-2 [502kB]
Fetched 541kB in 7s (73.9kB/s)
Selecting previously deselected package ladspa-sdk.
(Reading database ... 149319 files and directories currently installed.)
Unpacking ladspa-sdk (from .../ladspa-sdk_1.1-6_i386.deb) ...
Selecting previously deselected package swh-plugins.
Unpacking swh-plugins (from .../swh-plugins_0.4.15-2_i386.deb) ...
Processing triggers for doc-base ...
Processing 2 added doc-base file(s)...
Registering documents with scrollkeeper...
Processing triggers for man-db ...
Setting up ladspa-sdk (1.1-6) ...

Setting up swh-plugins (0.4.15-2) ...
sudo mv -$/home/ray/Downloads/pulseaudio-equalizer.sh /usr/local/bin

Skype test still functions OK.

RayArdia
November 6th, 2009, 08:01 PM
After hibernating and waking again my system has now gone back to its old habits. Skype test voice is intermittent and a little 'crackly'. No response from either the internal mic or a plugged-in (jacks) headset.
Since I have now failed to get Skype to work on both Jaunty and Karmic, either I'm doing something wrong (most likely) or there's a glitch of some kind in Skype or Ubuntu.
Can anyone help please?

srp010
November 6th, 2009, 08:15 PM
Another user in need of pulseaudio assistance. I need streaming radio to write docs. Without music I wander the halls avoiding my work. I am no Linux Jedi and could use some help figuring this out. Thanks In advance.

Recently upgraded to 9.10 and find everything seems to be functioning except pulseadio which plays choppy/stuttering sounds.

My box is a Del Optiplex 755 running Windows Pro 5.1 (they make me) with VM server 1.0.5 (80187) hosting Ubuntu 9.10. I found info on ubuntu hosting windows vms but not the other way around. The problem looks to be accessing the virtual sound card but I am clueless on what to dig into next.

aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: AudioPCI [Ensoniq AudioPCI], device 0: ES1371/1 [ES1371 DAC2/ADC]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: AudioPCI [Ensoniq AudioPCI], device 1: ES1371/2 [ES1371 DAC1]
Subdevices: 1/1
Subdevice #0: subdevice #0

pkill pulseaudio; sleep 2; pulseaudio -vv
I: main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted
I: main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation not permitted
D: core-rtclock.c: Timer slack is set to 50 us.
I: core-util.c: Failed to acquire high-priority scheduling: No such file or directory
I: main.c: This is PulseAudio 0.9.19
D: main.c: Compilation host: i486-pc-linux-gnu
D: main.c: Compilation CFLAGS: -g -O2 -g -Wall -O3 -Wall -W -Wextra -pipe -Wno-long-long -Winline -Wvla -Wno-overlength-strings -Wunsafe-loop-optimizations -Wundef -Wformat=2 -Wlogical-op -Wsign-compare -Wformat-security -Wmissing-include-dirs -Wformat-nonliteral -Wold-style-definition -Wpointer-arith -Winit-self -Wdeclaration-after-statement -Wfloat-equal -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wmissing-declarations -Wmissing-noreturn -Wshadow -Wendif-labels -Wcast-align -Wstrict-aliasing=2 -Wwrite-strings -Wno-unused-parameter -ffast-math -Wp,-D_FORTIFY_SOURCE=2 -fno-common -fdiagnostics-show-option
D: main.c: Running on host: Linux i686 2.6.31-14-generic #48-Ubuntu SMP Fri Oct 16 14:04:26 UTC 2009
D: main.c: Found 1 CPUs.
I: main.c: Page size is 4096 bytes
D: main.c: Compiled with Valgrind support: no
D: main.c: Running in valgrind mode: no
D: main.c: Optimized build: yes
D: main.c: All asserts enabled.
I: main.c: Machine ID is 2e8d1d8c5fbe5c88bf83dd24484509e6.
I: main.c: Session ID is 2e8d1d8c5fbe5c88bf83dd24484509e6-1257530575.722059-1229792856.
I: main.c: Using runtime directory /home/srp/.pulse/2e8d1d8c5fbe5c88bf83dd24484509e6-runtime.
I: main.c: Using state directory /home/srp/.pulse.
I: main.c: Using modules directory /usr/lib/pulse-0.9.19/modules.
I: main.c: Running in system mode: no
I: main.c: Fresh high-resolution timers available! Bon appetit!
I: cpu-x86.c: CPU flags: MMX SSE SSE2 SSE3 SSSE3
I: svolume_mmx.c: Initialising MMX optimized functions.
I: remap_mmx.c: Initialising MMX optimized remappers.
I: svolume_sse.c: Initialising SSE2 optimized functions.
I: remap_sse.c: Initialising SSE2 optimized remappers.
I: sconv_sse.c: Initialising SSE2 optimized conversions.
D: memblock.c: Using shared memory pool with 1024 slots of size 64.0 KiB each, total size is 64.0 MiB, maximum usable slot size is 65496
D: database-tdb.c: Opened TDB database '/home/srp/.pulse/2e8d1d8c5fbe5c88bf83dd24484509e6-device-volumes.tdb'
I: module-device-restore.c: Sucessfully opened database file '/home/srp/.pulse/2e8d1d8c5fbe5c88bf83dd24484509e6-device-volumes'.
I: module.c: Loaded "module-device-restore" (index: #0; argument: "").
D: database-tdb.c: Opened TDB database '/home/srp/.pulse/2e8d1d8c5fbe5c88bf83dd24484509e6-stream-volumes.tdb'
I: module-stream-restore.c: Sucessfully opened database file '/home/srp/.pulse/2e8d1d8c5fbe5c88bf83dd24484509e6-stream-volumes'.
I: module.c: Loaded "module-stream-restore" (index: #1; argument: "").
D: database-tdb.c: Opened TDB database '/home/srp/.pulse/2e8d1d8c5fbe5c88bf83dd24484509e6-card-database.tdb'
I: module-card-restore.c: Sucessfully opened database file '/home/srp/.pulse/2e8d1d8c5fbe5c88bf83dd24484509e6-card-database'.
I: module.c: Loaded "module-card-restore" (index: #2; argument: "").
I: module.c: Loaded "module-augment-properties" (index: #3; argument: "").
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.19/modules/module-udev-detect.so': success
D: module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
D: module-udev-detect.c: /devices/pci0000:00/0000:00:12.0/sound/card0 is busy: no
D: module-udev-detect.c: Loading module-alsa-card with arguments 'device_id="0" name="pci-0000_00_12.0" card_name="alsa_card.pci-0000_00_12.0" tsched=yes ignore_dB=no card_properties="module-udev-detect.discovered=1"'
D: dbus-util.c: Successfully connected to D-Bus session bus 2d0708ae7e5b1c8f8829a6944af464d0 as :1.93
D: reserve-wrap.c: Successfully acquired reservation lock on device 'Audio0'
D: alsa-mixer.c: Looking at profile output:analog-mono
D: alsa-mixer.c: Checking for playback on Analog Mono (analog-mono)
D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open hw:0
D: alsa-util.c: Maximum hw buffer size is 743 ms
D: alsa-util.c: Set buffer size first, period size second.
D: alsa-mixer.c: Profile output:analog-mono supported.
D: alsa-mixer.c: Looking at profile output:analog-mono+input:analog-mono
D: alsa-mixer.c: Checking for recording on Analog Mono (analog-mono)
D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open hw:0
D: alsa-util.c: Maximum hw buffer size is 743 ms
D: alsa-util.c: Set buffer size first, period size second.
I: alsa-util.c: Device hw:0 doesn't support 44100 Hz, changed to 44099 Hz.
D: alsa-mixer.c: Profile output:analog-mono+input:analog-mono supported.
D: alsa-mixer.c: Looking at profile output:analog-mono+input:analog-stereo
D: alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo)
D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open front:0
D: alsa-util.c: Maximum hw buffer size is 371 ms
D: alsa-util.c: Set buffer size first, period size second.
I: alsa-util.c: Device front:0 doesn't support 44100 Hz, changed to 44099 Hz.
D: alsa-mixer.c: Profile output:analog-mono+input:analog-stereo supported.
D: alsa-mixer.c: Looking at profile output:analog-mono+input:iec958-stereo
D: alsa-mixer.c: Checking for recording on Digital Stereo (IEC95 (iec958-stereo)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)setup.c: Cannot obtain info for CTL elem (PCM,'IEC958 Playback PCM Stream',0,0,0): No such file or directory
I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:analog-mono+input:iec958-surround-40
D: alsa-mixer.c: Checking for recording on Digital Surround 4.0 (IEC95 (iec958-surround-40)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)setup.c: Cannot obtain info for CTL elem (PCM,'IEC958 Playback PCM Stream',0,0,0): No such file or directory
I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:analog-stereo
D: alsa-mixer.c: Checking for playback on Analog Stereo (analog-stereo)
D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open front:0
D: alsa-util.c: Maximum hw buffer size is 371 ms
D: alsa-util.c: Set buffer size first, period size second.
D: alsa-mixer.c: Profile output:analog-stereo supported.
D: alsa-mixer.c: Looking at profile output:analog-stereo+input:analog-mono
D: alsa-mixer.c: Checking for recording on Analog Mono (analog-mono)
D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open hw:0
D: alsa-util.c: Maximum hw buffer size is 743 ms
D: alsa-util.c: Set buffer size first, period size second.
I: alsa-util.c: Device hw:0 doesn't support 44100 Hz, changed to 44099 Hz.
D: alsa-mixer.c: Profile output:analog-stereo+input:analog-mono supported.
D: alsa-mixer.c: Looking at profile output:analog-stereo+input:analog-stereo
D: alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo)
D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open front:0
D: alsa-util.c: Maximum hw buffer size is 371 ms
D: alsa-util.c: Set buffer size first, period size second.
I: alsa-util.c: Device front:0 doesn't support 44100 Hz, changed to 44099 Hz.
D: alsa-mixer.c: Profile output:analog-stereo+input:analog-stereo supported.
D: alsa-mixer.c: Looking at profile output:analog-stereo+input:iec958-stereo
D: alsa-mixer.c: Checking for recording on Digital Stereo (IEC95 (iec958-stereo)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)setup.c: Cannot obtain info for CTL elem (PCM,'IEC958 Playback PCM Stream',0,0,0): No such file or directory
I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:analog-stereo+input:iec958-surround-40
D: alsa-mixer.c: Checking for recording on Digital Surround 4.0 (IEC95 (iec958-surround-40)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)setup.c: Cannot obtain info for CTL elem (PCM,'IEC958 Playback PCM Stream',0,0,0): No such file or directory
I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:analog-surround-40
D: alsa-mixer.c: Checking for playback on Analog Surround 4.0 (analog-surround-40)
D: alsa-util.c: Trying surround40:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)setup.c: Cannot obtain info for CTL elem (MIXER,'AC97 2ch->4ch Copy Switch',0,0,0): No such file or directory
I: alsa-util.c: Error opening PCM device surround40:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:analog-surround-40+input:analog-mono
D: alsa-mixer.c: Checking for playback on Analog Surround 4.0 (analog-surround-40)
D: alsa-util.c: Trying surround40:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)setup.c: Cannot obtain info for CTL elem (MIXER,'AC97 2ch->4ch Copy Switch',0,0,0): No such file or directory
I: alsa-util.c: Error opening PCM device surround40:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:analog-surround-40+input:analog-stereo
D: alsa-mixer.c: Checking for playback on Analog Surround 4.0 (analog-surround-40)
D: alsa-util.c: Trying surround40:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)setup.c: Cannot obtain info for CTL elem (MIXER,'AC97 2ch->4ch Copy Switch',0,0,0): No such file or directory
I: alsa-util.c: Error opening PCM device surround40:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:analog-surround-40+input:iec958-stereo
D: alsa-mixer.c: Checking for playback on Analog Surround 4.0 (analog-surround-40)
D: alsa-util.c: Trying surround40:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)setup.c: Cannot obtain info for CTL elem (MIXER,'AC97 2ch->4ch Copy Switch',0,0,0): No such file or directory
I: alsa-util.c: Error opening PCM device surround40:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:analog-surround-40+input:iec958-surround-40
D: alsa-mixer.c: Checking for playback on Analog Surround 4.0 (analog-surround-40)
D: alsa-util.c: Trying surround40:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)setup.c: Cannot obtain info for CTL elem (MIXER,'AC97 2ch->4ch Copy Switch',0,0,0): No such file or directory
I: alsa-util.c: Error opening PCM device surround40:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:analog-surround-41
D: alsa-mixer.c: Checking for playback on Analog Surround 4.1 (analog-surround-41)
D: alsa-util.c: Trying surround41:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM surround41:0
I: alsa-util.c: Error opening PCM device surround41:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:analog-surround-41+input:analog-mono
D: alsa-mixer.c: Checking for playback on Analog Surround 4.1 (analog-surround-41)
D: alsa-util.c: Trying surround41:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM surround41:0
I: alsa-util.c: Error opening PCM device surround41:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:analog-surround-41+input:analog-stereo
D: alsa-mixer.c: Checking for playback on Analog Surround 4.1 (analog-surround-41)
D: alsa-util.c: Trying surround41:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM surround41:0
I: alsa-util.c: Error opening PCM device surround41:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:analog-surround-41+input:iec958-stereo
D: alsa-mixer.c: Checking for playback on Analog Surround 4.1 (analog-surround-41)
D: alsa-util.c: Trying surround41:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM surround41:0
I: alsa-util.c: Error opening PCM device surround41:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:analog-surround-41+input:iec958-surround-40
D: alsa-mixer.c: Checking for playback on Analog Surround 4.1 (analog-surround-41)
D: alsa-util.c: Trying surround41:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM surround41:0
I: alsa-util.c: Error opening PCM device surround41:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:analog-surround-50
D: alsa-mixer.c: Checking for playback on Analog Surround 5.0 (analog-surround-50)
D: alsa-util.c: Trying surround50:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM surround50:0
I: alsa-util.c: Error opening PCM device surround50:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:analog-surround-50+input:analog-mono
D: alsa-mixer.c: Checking for playback on Analog Surround 5.0 (analog-surround-50)
D: alsa-util.c: Trying surround50:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM surround50:0
I: alsa-util.c: Error opening PCM device surround50:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:analog-surround-50+input:analog-stereo
D: alsa-mixer.c: Checking for playback on Analog Surround 5.0 (analog-surround-50)
D: alsa-util.c: Trying surround50:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM surround50:0
I: alsa-util.c: Error opening PCM device surround50:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:analog-surround-50+input:iec958-stereo
D: alsa-mixer.c: Checking for playback on Analog Surround 5.0 (analog-surround-50)
D: alsa-util.c: Trying surround50:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM surround50:0
I: alsa-util.c: Error opening PCM device surround50:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:analog-surround-50+input:iec958-surround-40
D: alsa-mixer.c: Checking for playback on Analog Surround 5.0 (analog-surround-50)
D: alsa-util.c: Trying surround50:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM surround50:0
I: alsa-util.c: Error opening PCM device surround50:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:analog-surround-51
D: alsa-mixer.c: Checking for playback on Analog Surround 5.1 (analog-surround-51)
D: alsa-util.c: Trying surround51:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM surround51:0
I: alsa-util.c: Error opening PCM device surround51:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:analog-surround-51+input:analog-mono
D: alsa-mixer.c: Checking for playback on Analog Surround 5.1 (analog-surround-51)
D: alsa-util.c: Trying surround51:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM surround51:0
I: alsa-util.c: Error opening PCM device surround51:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:analog-surround-51+input:analog-stereo
D: alsa-mixer.c: Checking for playback on Analog Surround 5.1 (analog-surround-51)
D: alsa-util.c: Trying surround51:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM surround51:0
I: alsa-util.c: Error opening PCM device surround51:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:analog-surround-51+input:iec958-stereo
D: alsa-mixer.c: Checking for playback on Analog Surround 5.1 (analog-surround-51)
D: alsa-util.c: Trying surround51:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM surround51:0
I: alsa-util.c: Error opening PCM device surround51:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:analog-surround-51+input:iec958-surround-40
D: alsa-mixer.c: Checking for playback on Analog Surround 5.1 (analog-surround-51)
D: alsa-util.c: Trying surround51:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM surround51:0
I: alsa-util.c: Error opening PCM device surround51:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:analog-surround-71
D: alsa-mixer.c: Checking for playback on Analog Surround 7.1 (analog-surround-71)
D: alsa-util.c: Trying surround71:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM surround71:0
I: alsa-util.c: Error opening PCM device surround71:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:analog-surround-71+input:analog-mono
D: alsa-mixer.c: Checking for playback on Analog Surround 7.1 (analog-surround-71)
D: alsa-util.c: Trying surround71:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM surround71:0
I: alsa-util.c: Error opening PCM device surround71:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:analog-surround-71+input:analog-stereo
D: alsa-mixer.c: Checking for playback on Analog Surround 7.1 (analog-surround-71)
D: alsa-util.c: Trying surround71:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM surround71:0
I: alsa-util.c: Error opening PCM device surround71:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:analog-surround-71+input:iec958-stereo
D: alsa-mixer.c: Checking for playback on Analog Surround 7.1 (analog-surround-71)
D: alsa-util.c: Trying surround71:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM surround71:0
I: alsa-util.c: Error opening PCM device surround71:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:analog-surround-71+input:iec958-surround-40
D: alsa-mixer.c: Checking for playback on Analog Surround 7.1 (analog-surround-71)
D: alsa-util.c: Trying surround71:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM surround71:0
I: alsa-util.c: Error opening PCM device surround71:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:iec958-stereo
D: alsa-mixer.c: Checking for playback on Digital Stereo (IEC958) (iec958-stereo)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)setup.c: Cannot obtain info for CTL elem (PCM,'IEC958 Playback PCM Stream',0,0,0): No such file or directory
I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-stereo+input:analog-mono
D: alsa-mixer.c: Checking for playback on Digital Stereo (IEC95 (iec958-stereo)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)setup.c: Cannot obtain info for CTL elem (PCM,'IEC958 Playback PCM Stream',0,0,0): No such file or directory
I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-stereo+input:analog-stereo
D: alsa-mixer.c: Checking for playback on Digital Stereo (IEC95 (iec958-stereo)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)setup.c: Cannot obtain info for CTL elem (PCM,'IEC958 Playback PCM Stream',0,0,0): No such file or directory
I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-stereo+input:iec958-stereo
D: alsa-mixer.c: Checking for playback on Digital Stereo (IEC95 (iec958-stereo)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)setup.c: Cannot obtain info for CTL elem (PCM,'IEC958 Playback PCM Stream',0,0,0): No such file or directory
I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-stereo+input:iec958-surround-40
D: alsa-mixer.c: Checking for playback on Digital Stereo (IEC95 (iec958-stereo)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)setup.c: Cannot obtain info for CTL elem (PCM,'IEC958 Playback PCM Stream',0,0,0): No such file or directory
I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-surround-40
D: alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC95 (iec958-surround-40)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)setup.c: Cannot obtain info for CTL elem (PCM,'IEC958 Playback PCM Stream',0,0,0): No such file or directory
I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-surround-40+input:analog-mono
D: alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC95 (iec958-surround-40)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)setup.c: Cannot obtain info for CTL elem (PCM,'IEC958 Playback PCM Stream',0,0,0): No such file or directory
I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-surround-40+input:analog-stereo
D: alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC95 (iec958-surround-40)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)setup.c: Cannot obtain info for CTL elem (PCM,'IEC958 Playback PCM Stream',0,0,0): No such file or directory
I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-surround-40+input:iec958-stereo
D: alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC95 (iec958-surround-40)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)setup.c: Cannot obtain info for CTL elem (PCM,'IEC958 Playback PCM Stream',0,0,0): No such file or directory
I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-surround-40+input:iec958-surround-40
D: alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC95 (iec958-surround-40)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)setup.c: Cannot obtain info for CTL elem (PCM,'IEC958 Playback PCM Stream',0,0,0): No such file or directory
I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-40
D: alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC958/AC3) (iec958-ac3-surround-40)
D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-40+input:analog-mono
D: alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC958/AC3) (iec958-ac3-surround-40)
D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-40+input:analog-stereo
D: alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC958/AC3) (iec958-ac3-surround-40)
D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-40+input:iec958-stereo
D: alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC958/AC3) (iec958-ac3-surround-40)
D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-40+input:iec958-surround-40
D: alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC958/AC3) (iec958-ac3-surround-40)
D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-51
D: alsa-mixer.c: Checking for playback on Digital Surround 5.1 (IEC958/AC3) (iec958-ac3-surround-51)
D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-51+input:analog-mono
D: alsa-mixer.c: Checking for playback on Digital Surround 5.1 (IEC958/AC3) (iec958-ac3-surround-51)
D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-51+input:analog-stereo
D: alsa-mixer.c: Checking for playback on Digital Surround 5.1 (IEC958/AC3) (iec958-ac3-surround-51)
D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-51+input:iec958-stereo
D: alsa-mixer.c: Checking for playback on Digital Surround 5.1 (IEC958/AC3) (iec958-ac3-surround-51)
D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-51+input:iec958-surround-40
D: alsa-mixer.c: Checking for playback on Digital Surround 5.1 (IEC958/AC3) (iec958-ac3-surround-51)
D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:hdmi-stereo
D: alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo)
D: alsa-util.c: Trying hdmi:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM hdmi:0
I: alsa-util.c: Error opening PCM device hdmi:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:hdmi-stereo+input:analog-mono
D: alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo)
D: alsa-util.c: Trying hdmi:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM hdmi:0
I: alsa-util.c: Error opening PCM device hdmi:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:hdmi-stereo+input:analog-stereo
D: alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo)
D: alsa-util.c: Trying hdmi:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM hdmi:0
I: alsa-util.c: Error opening PCM device hdmi:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:hdmi-stereo+input:iec958-stereo
D: alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo)
D: alsa-util.c: Trying hdmi:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM hdmi:0
I: alsa-util.c: Error opening PCM device hdmi:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:hdmi-stereo+input:iec958-surround-40
D: alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo)
D: alsa-util.c: Trying hdmi:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM hdmi:0
I: alsa-util.c: Error opening PCM device hdmi:0: Invalid argument
D: alsa-mixer.c: Looking at profile input:analog-mono
D: alsa-mixer.c: Checking for recording on Analog Mono (analog-mono)
D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open hw:0
D: alsa-util.c: Maximum hw buffer size is 743 ms
D: alsa-util.c: Set buffer size first, period size second.
I: alsa-util.c: Device hw:0 doesn't support 44100 Hz, changed to 44099 Hz.
D: alsa-mixer.c: Profile input:analog-mono supported.
D: alsa-mixer.c: Looking at profile input:analog-stereo
D: alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo)
D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open front:0
D: alsa-util.c: Maximum hw buffer size is 371 ms
D: alsa-util.c: Set buffer size first, period size second.
I: alsa-util.c: Device front:0 doesn't support 44100 Hz, changed to 44099 Hz.
D: alsa-mixer.c: Profile input:analog-stereo supported.
D: alsa-mixer.c: Looking at profile input:iec958-stereo
D: alsa-mixer.c: Checking for recording on Digital Stereo (IEC95 (iec958-stereo)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)setup.c: Cannot obtain info for CTL elem (PCM,'IEC958 Playback PCM Stream',0,0,0): No such file or directory
I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
D: alsa-mixer.c: Looking at profile input:iec958-surround-40
D: alsa-mixer.c: Checking for recording on Digital Surround 4.0 (IEC95 (iec958-surround-40)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)setup.c: Cannot obtain info for CTL elem (PCM,'IEC958 Playback PCM Stream',0,0,0): No such file or directory
I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
I: card.c: Created 0 "alsa_card.pci-0000_00_12.0"
D: reserve-wrap.c: Successfully create reservation lock monitor for device 'Audio0'
D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open front:0
D: alsa-util.c: Maximum hw buffer size is 371 ms
D: alsa-util.c: Set buffer size first, period size second.
I: alsa-sink.c: Successfully opened device front:0.
I: alsa-sink.c: Selected mapping 'Analog Stereo' (analog-stereo).
I: alsa-sink.c: Successfully enabled mmap() mode.
I: alsa-sink.c: Successfully enabled timer-based scheduling mode.
I: (alsa-lib)control.c: Invalid CTL front:0
I: alsa-mixer.c: Unable to attach to mixer front:0: No such file or directory
I: alsa-mixer.c: Successfully attached to mixer 'hw:0'
D: alsa-mixer.c: Probing path 'analog-output'
D: alsa-mixer.c: Probing path 'analog-output-headphones'
D: alsa-mixer.c: Probe of element 'Headphone' failed.
D: alsa-mixer.c: Probing path 'analog-output-mono'
D: alsa-mixer.c: Probe of element 'Master Mono' failed.
D: alsa-mixer.c: Probing path 'analog-output-lfe-on-mono'
D: alsa-mixer.c: Probe of element 'Master Mono' failed.
D: alsa-sink.c: Probed mixer paths:
D: alsa-mixer.c: Path Set 0x9538c38, direction=1, probed=yes
D: alsa-mixer.c: Path analog-output (Analog Output), direction=1, priority=100, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=63, min_dB=-129, max_dB=60
D: alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes
D: alsa-mixer.c: Element PCM, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x404f600000000f66, n_channels=2, override_map=yes
I: module-device-restore.c: Restoring volume for sink alsa_output.pci-0000_00_12.0.analog-stereo.
I: module-device-restore.c: Restoring mute state for sink alsa_output.pci-0000_00_12.0.analog-stereo.
I: sink.c: Created sink 0 "alsa_output.pci-0000_00_12.0.analog-stereo" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: sink.c: alsa.resolution_bits = "16"
I: sink.c: device.api = "alsa"
I: sink.c: device.class = "sound"
I: sink.c: alsa.class = "generic"
I: sink.c: alsa.subclass = "generic-mix"
I: sink.c: alsa.name = "ES1371 DAC2/ADC"
I: sink.c: alsa.id = "ES1371/1"
I: sink.c: alsa.subdevice = "0"
I: sink.c: alsa.subdevice_name = "subdevice #0"
I: sink.c: alsa.device = "0"
I: sink.c: alsa.card = "0"
I: sink.c: alsa.card_name = "Ensoniq AudioPCI"
I: sink.c: alsa.long_card_name = "Ensoniq AudioPCI ENS1371 at 0x1480, irq 19"
I: sink.c: alsa.driver_name = "snd_ens1371"
I: sink.c: device.bus_path = "pci-0000:00:12.0"
I: sink.c: sysfs.path = "/devices/pci0000:00/0000:00:12.0/sound/card0"
I: sink.c: device.bus = "pci"
I: sink.c: device.vendor.id = "1274"
I: sink.c: device.vendor.name = "Ensoniq"
I: sink.c: device.product.id = "1371"
I: sink.c: device.product.name = "ES1371 [AudioPCI-97]"
I: sink.c: device.form_factor = "internal"
I: sink.c: device.string = "front:0"
I: sink.c: device.buffering.buffer_size = "65536"
I: sink.c: device.buffering.fragment_size = "65536"
I: sink.c: device.access_mode = "mmap+timer"
I: sink.c: device.profile.name = "analog-stereo"
I: sink.c: device.profile.description = "Analog Stereo"
I: sink.c: device.description = "Internal Audio Analog Stereo"
I: sink.c: alsa.mixer_name = "Cirrus Logic CS4297A rev 3"
I: sink.c: alsa.components = "AC97a:43525913"
I: sink.c: module-udev-detect.discovered = "1"
I: sink.c: device.icon_name = "audio-card-pci"
D: core-subscribe.c: Dropped redundant event due to change event.
I: source.c: Created source 0 "alsa_output.pci-0000_00_12.0.analog-stereo.monitor" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: source.c: device.description = "Monitor of Internal Audio Analog Stereo"
I: source.c: device.class = "monitor"
I: source.c: alsa.card = "0"
I: source.c: alsa.card_name = "Ensoniq AudioPCI"
I: source.c: alsa.long_card_name = "Ensoniq AudioPCI ENS1371 at 0x1480, irq 19"
I: source.c: alsa.driver_name = "snd_ens1371"
I: source.c: device.bus_path = "pci-0000:00:12.0"
I: source.c: sysfs.path = "/devices/pci0000:00/0000:00:12.0/sound/card0"
I: source.c: device.bus = "pci"
I: source.c: device.vendor.id = "1274"
I: source.c: device.vendor.name = "Ensoniq"
I: source.c: device.product.id = "1371"
I: source.c: device.product.name = "ES1371 [AudioPCI-97]"
I: source.c: device.form_factor = "internal"
I: source.c: device.string = "0"
I: source.c: module-udev-detect.discovered = "1"
I: source.c: device.icon_name = "audio-card-pci"
I: alsa-sink.c: Using 1.0 fragments of size 65536 bytes (371.52ms), buffer size is 65536 bytes (371.52ms)
I: alsa-sink.c: Time scheduling watermark is 20.00ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=15502
D: alsa-mixer.c: Activating path analog-output
D: alsa-mixer.c: Path analog-output (Analog Output), direction=1, priority=100, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=63, min_dB=-129, max_dB=60
D: alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes
D: alsa-mixer.c: Element PCM, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x404f600000000f66, n_channels=2, override_map=yes
I: alsa-sink.c: Hardware volume ranges from -129.00 dB to 60.00 dB.
I: alsa-sink.c: Fixing base volume to -60.00 dB
I: alsa-sink.c: Using hardware volume control. Hardware dB scale supported.
I: alsa-sink.c: Using hardware mute control.
D: alsa-util.c: snd_pcm_dump():
D: alsa-util.c: Hardware PCM card 0 'Ensoniq AudioPCI' device 0 subdevice 0
D: alsa-util.c: Its setup is:
D: alsa-util.c: stream : PLAYBACK
D: alsa-util.c: access : MMAP_INTERLEAVED
D: alsa-util.c: format : S16_LE
D: alsa-util.c: subformat : STD
D: alsa-util.c: channels : 2
D: alsa-util.c: rate : 44100
D: alsa-util.c: exact rate : 44101 (1445100000/3276
D: alsa-util.c: msbits : 16
D: alsa-util.c: buffer_size : 16384
D: alsa-util.c: period_size : 16384
D: alsa-util.c: period_time : 371510
D: alsa-util.c: tstamp_mode : ENABLE
D: alsa-util.c: period_step : 1
D: alsa-util.c: avail_min : 16384
D: alsa-util.c: period_event : 0
D: alsa-util.c: start_threshold : -1
D: alsa-util.c: stop_threshold : 1073741824
D: alsa-util.c: silence_threshold: 0
D: alsa-util.c: silence_size : 0
D: alsa-util.c: boundary : 1073741824
D: alsa-util.c: appl_ptr : 0
D: alsa-util.c: hw_ptr : 0
D: alsa-sink.c: Thread starting up
I: core-util.c: Successfully enabled SCHED_RR scheduling for thread, with priority 4, which is lower than the requested 5.
D: alsa-sink.c: Requested volume: 0: 100% 1: 100%
D: alsa-sink.c: Got hardware volume: 0: 100% 1: 100%
D: alsa-sink.c: Calculated software volume: 0: 100% 1: 100% (accurate-enough=yes)
I: alsa-sink.c: Starting playback.
D: alsa-sink.c: Cutting sleep time for the initial iterations by half.
D: alsa-sink.c: Cutting sleep time for the initial iterations by half.
D: alsa-sink.c: Cutting sleep time for the initial iterations by half.
D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open front:0
D: alsa-util.c: Maximum hw buffer size is 371 ms
D: alsa-util.c: Set buffer size first, period size second.
I: alsa-util.c: Device front:0 doesn't support 44100 Hz, changed to 44099 Hz.
I: alsa-source.c: Successfully opened device front:0.
I: alsa-source.c: Selected mapping 'Analog Stereo' (analog-stereo).
I: alsa-source.c: Successfully enabled mmap() mode.
I: alsa-source.c: Successfully enabled timer-based scheduling mode.
I: (alsa-lib)control.c: Invalid CTL front:0
I: alsa-mixer.c: Unable to attach to mixer front:0: No such file or directory
I: alsa-mixer.c: Successfully attached to mixer 'hw:0'
D: alsa-mixer.c: Probing path 'analog-input'
D: alsa-mixer.c: Probe of element 'Mic' failed.
D: alsa-mixer.c: Probing path 'analog-input-microphone'
D: alsa-mixer.c: Probing path 'analog-input-linein'
D: alsa-mixer.c: Probing path 'analog-input'
D: alsa-mixer.c: Probing path 'analog-input-video'
D: alsa-mixer.c: Probing path 'analog-input-video'
D: alsa-mixer.c: Probe of element 'TV Tuner' failed.
D: alsa-mixer.c: Probing path 'analog-input-radio'
D: alsa-mixer.c: Probe of element 'FM' failed.
D: alsa-mixer.c: Probing path 'analog-input'
D: alsa-mixer.c: Probe of element 'Mic/Line' failed.
D: alsa-source.c: Probed mixer paths:
D: alsa-mixer.c: Path Set 0x954b008, direction=2, probed=yes
D: alsa-mixer.c: Path analog-input-microphone (Analog Microphone), direction=2, priority=100, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=15, min_dB=0, max_dB=22.5
D: alsa-mixer.c: Element Capture, direction=2, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x4037e00000000f66, n_channels=2, override_map=yes
D: alsa-mixer.c: Element Mic, direction=2, switch=1, volume=0, enumeration=0, required=4, required_absent=0, mask=0x0, n_channels=0, override_map=yes
D: alsa-mixer.c: Element Line, direction=2, switch=2, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Element Aux, direction=2, switch=2, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Element Video, direction=2, switch=2, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Path analog-input-linein (Analog Line-In), direction=2, priority=90, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=15, min_dB=0, max_dB=22.5
D: alsa-mixer.c: Element Capture, direction=2, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x4037e00000000f66, n_channels=2, override_map=yes
D: alsa-mixer.c: Element Mic, direction=2, switch=2, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Element Line, direction=2, switch=1, volume=0, enumeration=0, required=4, required_absent=0, mask=0x0, n_channels=0, override_map=yes
D: alsa-mixer.c: Element Aux, direction=2, switch=2, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Element Video, direction=2, switch=2, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Path analog-input (Analog Input), direction=2, priority=90, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=15, min_dB=0, max_dB=22.5
D: alsa-mixer.c: Element Capture, direction=2, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x4037e00000000f66, n_channels=2, override_map=yes
D: alsa-mixer.c: Element Mic, direction=2, switch=2, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Element Line, direction=2, switch=2, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Element Aux, direction=2, switch=1, volume=0, enumeration=0, required=4, required_absent=0, mask=0x0, n_channels=0, override_map=yes
D: alsa-mixer.c: Element Video, direction=2, switch=2, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Path analog-input-video (Analog Video), direction=2, priority=70, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=15, min_dB=0, max_dB=22.5
D: alsa-mixer.c: Element Capture, direction=2, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x4037e00000000f66, n_channels=2, override_map=yes
D: alsa-mixer.c: Element Mic, direction=2, switch=2, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Element Line, direction=2, switch=2, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Element Aux, direction=2, switch=2, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Element Video, direction=2, switch=1, volume=0, enumeration=0, required=4, required_absent=0, mask=0x0, n_channels=0, override_map=yes
D: alsa-mixer.c: Added 4 ports
D: core-subscribe.c: Dropped redundant event due to change event.
I: source.c: Created source 1 "alsa_input.pci-0000_00_12.0.analog-stereo" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: source.c: alsa.resolution_bits = "16"
I: source.c: device.api = "alsa"
I: source.c: device.class = "sound"
I: source.c: alsa.class = "generic"
I: source.c: alsa.subclass = "generic-mix"
I: source.c: alsa.name = "ES1371 DAC2/ADC"
I: source.c: alsa.id = "ES1371/1"
I: source.c: alsa.subdevice = "0"
I: source.c: alsa.subdevice_name = "subdevice #0"
I: source.c: alsa.device = "0"
I: source.c: alsa.card = "0"
I: source.c: alsa.card_name = "Ensoniq AudioPCI"
I: source.c: alsa.long_card_name = "Ensoniq AudioPCI ENS1371 at 0x1480, irq 19"
I: source.c: alsa.driver_name = "snd_ens1371"
I: source.c: device.bus_path = "pci-0000:00:12.0"
I: source.c: sysfs.path = "/devices/pci0000:00/0000:00:12.0/sound/card0"
I: source.c: device.bus = "pci"
I: source.c: device.vendor.id = "1274"
I: source.c: device.vendor.name = "Ensoniq"
I: source.c: device.product.id = "1371"
I: source.c: device.product.name = "ES1371 [AudioPCI-97]"
I: source.c: device.form_factor = "internal"
I: source.c: device.string = "front:0"
I: source.c: device.buffering.buffer_size = "65536"
I: source.c: device.buffering.fragment_size = "65536"
I: source.c: device.access_mode = "mmap+timer"
I: source.c: device.profile.name = "analog-stereo"
I: source.c: device.profile.description = "Analog Stereo"
I: source.c: device.description = "Internal Audio Analog Stereo"
I: source.c: alsa.mixer_name = "Cirrus Logic CS4297A rev 3"
I: source.c: alsa.components = "AC97a:43525913"
I: source.c: module-udev-detect.discovered = "1"
I: source.c: device.icon_name = "audio-card-pci"
I: alsa-source.c: Using 1.0 fragments of size 65536 bytes (371.52ms), buffer size is 65536 bytes (371.52ms)
I: alsa-source.c: Time scheduling watermark is 20.00ms
D: alsa-source.c: hwbuf_unused=0
D: alsa-source.c: setting avail_min=15502
D: alsa-mixer.c: Activating path analog-input-microphone
D: alsa-mixer.c: Path analog-input-microphone (Analog Microphone), direction=2, priority=100, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=15, min_dB=0, max_dB=22.5
D: alsa-mixer.c: Element Capture, direction=2, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x4037e00000000f66, n_channels=2, override_map=yes
D: alsa-mixer.c: Element Mic, direction=2, switch=1, volume=0, enumeration=0, required=4, required_absent=0, mask=0x0, n_channels=0, override_map=yes
D: alsa-mixer.c: Element Line, direction=2, switch=2, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Element Aux, direction=2, switch=2, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Element Video, direction=2, switch=2, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
I: alsa-source.c: Hardware volume ranges from 0.00 dB to 22.50 dB.
I: alsa-source.c: Fixing base volume to -22.50 dB
I: alsa-source.c: Using hardware volume control. Hardware dB scale supported.
I: alsa-source.c: Using hardware mute control.
D: alsa-util.c: snd_pcm_dump():
D: alsa-util.c: Hardware PCM card 0 'Ensoniq AudioPCI' device 0 subdevice 0
D: alsa-util.c: Its setup is:
D: alsa-util.c: stream : CAPTURE
D: alsa-util.c: access : MMAP_INTERLEAVED
D: alsa-util.c: format : S16_LE
D: alsa-util.c: subformat : STD
D: alsa-util.c: channels : 2
D: alsa-util.c: rate : 44099
D: alsa-util.c: exact rate : 44099.8 (1572864000/35666)
D: alsa-util.c: msbits : 16
D: alsa-util.c: buffer_size : 16384
D: alsa-util.c: period_size : 16384
D: alsa-util.c: period_time : 371519
D: alsa-util.c: tstamp_mode : ENABLE
D: alsa-util.c: period_step : 1
D: alsa-util.c: avail_min : 16384
D: alsa-util.c: period_event : 0
D: alsa-util.c: start_threshold : -1
D: alsa-util.c: stop_threshold : 1073741824
D: alsa-util.c: silence_threshold: 0
D: alsa-util.c: silence_size : 0
D: alsa-util.c: boundary : 1073741824
D: alsa-util.c: appl_ptr : 0
D: alsa-util.c: hw_ptr : 0
D: alsa-source.c: Thread starting up
I: core-util.c: Successfully enabled SCHED_RR scheduling for thread, with priority 4, which is lower than the requested 5.
D: alsa-source.c: Read hardware volume: 0: 42% 1: 42%
I: module.c: Loaded "module-alsa-card" (index: #4; argument: "device_id="0" name="pci-0000_00_12.0" card_name="alsa_card.pci-0000_00_12.0" tsched=yes ignore_dB=no card_properties="module-udev-detect.discovered=1"").
I: module-udev-detect.c: Card /devices/pci0000:00/0000:00:12.0/sound/card0 (alsa_card.pci-0000_00_12.0) module loaded.
I: module-udev-detect.c: Found 1 cards.
I: module.c: Loaded "module-udev-detect" (index: #5; argument: "").
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.19/modules/module-bluetooth-discover.so': failure
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.19/modules/module-esound-protocol-unix.so': success
D: alsa-sink.c: Wakeup from ALSA!
I: alsa-sink.c: Underrun!
I: alsa-sink.c: Increasing wakeup watermark to 30.00 ms
I: module.c: Loaded "module-esound-protocol-unix" (index: #6; argument: "").
I: module.c: Loaded "module-native-protocol-unix" (index: #7; argument: "").
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.19/modules/module-gconf.so': success
D: alsa-sink.c: Wakeup from ALSA!
I: alsa-sink.c: Underrun!
I: alsa-sink.c: Increasing wakeup watermark to 40.00 ms
D: alsa-sink.c: Wakeup from ALSA!
I: alsa-sink.c: Underrun!
I: alsa-sink.c: Increasing wakeup watermark to 50.00 ms
D: module-gconf.c: Loading module 'module-zeroconf-discover' with args '' due to GConf configuration.
D: alsa-sink.c: Wakeup from ALSA!
I: alsa-sink.c: Underrun!
I: alsa-sink.c: Increasing wakeup watermark to 60.00 ms
I: alsa-sink.c: Increasing wakeup watermark to 70.00 ms
I: module.c: Loaded "module-zeroconf-discover" (index: #8; argument: "").
I: module.c: Loaded "module-gconf" (index: #9; argument: "").
D: core-subscribe.c: Dropped redundant event due to change event.
I: module-default-device-restore.c: Restored default sink 'alsa_output.pci-0000_00_12.0.analog-stereo'.
D: core-subscribe.c: Dropped redundant event due to change event.
I: module-default-device-restore.c: Restored default source 'alsa_input.pci-0000_00_12.0.analog-stereo'.
I: module.c: Loaded "module-default-device-restore" (index: #10; argument: "").
I: module.c: Loaded "module-rescue-streams" (index: #11; argument: "").
I: module.c: Loaded "module-always-sink" (index: #12; argument: "").
I: module.c: Loaded "module-intended-roles" (index: #13; argument: "").
D: module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_12.0.analog-stereo becomes idle, timeout in 5 seconds.
D: module-suspend-on-idle.c: Source alsa_input.pci-0000_00_12.0.analog-stereo becomes idle, timeout in 5 seconds.
I: module.c: Loaded "module-suspend-on-idle" (index: #14; argument: "").
I: alsa-sink.c: Increasing wakeup watermark to 80.00 ms
D: dbus-util.c: Successfully connected to D-Bus system bus 66f6628bed0b768fa7352fdc4af46478 as :1.95
I: client.c: Created 0 "ConsoleKit Session /org/freedesktop/ConsoleKit/Session2"
D: module-console-kit.c: Added new session /org/freedesktop/ConsoleKit/Session2
I: module.c: Loaded "module-console-kit" (index: #15; argument: "").
I: module.c: Loaded "module-position-event-sounds" (index: #16; argument: "").
I: alsa-sink.c: Increasing wakeup watermark to 90.00 ms
D: main.c: Got org.pulseaudio.Server!
I: main.c: Daemon startup complete.
D: module-console-kit.c: dbus: interface=org.freedesktop.DBus, path=/org/freedesktop/DBus, member=NameAcquired
D: module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
I: alsa-sink.c: Increasing wakeup watermark to 100.00 ms
I: alsa-sink.c: Increasing wakeup watermark to 110.00 ms
I: alsa-sink.c: Increasing wakeup watermark to 120.00 ms
I: alsa-sink.c: Increasing wakeup watermark to 130.00 ms
I: alsa-sink.c: Increasing wakeup watermark to 140.00 ms
I: alsa-sink.c: Increasing wakeup watermark to 150.00 ms
I: alsa-sink.c: Increasing wakeup watermark to 160.00 ms
I: alsa-sink.c: Increasing wakeup watermark to 170.00 ms
I: alsa-sink.c: Increasing wakeup watermark to 180.00 ms
I: alsa-sink.c: Increasing wakeup watermark to 190.00 ms
I: alsa-sink.c: Increasing wakeup watermark to 200.00 ms
I: alsa-sink.c: Increasing wakeup watermark to 210.00 ms
I: alsa-sink.c: Increasing wakeup watermark to 220.00 ms
I: alsa-sink.c: Increasing wakeup watermark to 230.00 ms
I: alsa-sink.c: Increasing wakeup watermark to 240.00 ms
I: alsa-sink.c: Increasing wakeup watermark to 250.00 ms
I: alsa-sink.c: Increasing wakeup watermark to 260.00 ms
I: alsa-sink.c: Increasing wakeup watermark to 270.00 ms
I: alsa-sink.c: Increasing wakeup watermark to 280.00 ms
I: alsa-sink.c: Increasing wakeup watermark to 290.00 ms
I: alsa-sink.c: Increasing wakeup watermark to 300.00 ms
I: alsa-sink.c: Increasing wakeup watermark to 310.00 ms
I: alsa-sink.c: Increasing wakeup watermark to 320.00 ms
I: alsa-sink.c: Increasing wakeup watermark to 330.00 ms
I: alsa-sink.c: Increasing wakeup watermark to 340.00 ms
I: alsa-sink.c: Increasing wakeup watermark to 350.00 ms
I: alsa-sink.c: Increasing wakeup watermark to 360.00 ms
I: alsa-sink.c: Increasing wakeup watermark to 361.52 ms
I: alsa-sink.c: Increasing minimal latency to 1.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 2.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 4.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 8.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 16.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 26.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 36.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 46.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 56.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 66.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 76.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 86.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 96.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 106.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 116.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 126.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 136.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 146.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 156.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 166.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 176.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 186.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 196.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 206.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 216.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 226.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 236.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 246.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 256.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 266.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 276.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 286.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 296.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 306.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 316.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 326.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 336.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 346.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 356.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 366.00 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: alsa-sink.c: Increasing minimal latency to 371.52 ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=442
I: client.c: Created 1 "Native client (UNIX socket client)"
D: protocol-native.c: Protocol version: remote 16, local 16
I: protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
D: protocol-native.c: SHM possible: yes
D: protocol-native.c: Negotiated SHM: yes
D: module-augment-properties.c: Looking for .desktop file for gnome-settings-daemon
E: alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write!
E: alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_ens1371'. Please report this issue to the ALSA developers.
E: alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail.
I: client.c: Created 2 "Native client (UNIX socket client)"
D: protocol-native.c: Protocol version: remote 16, local 16
I: protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
D: protocol-native.c: SHM possible: yes
D: protocol-native.c: Negotiated SHM: yes
D: module-augment-properties.c: Looking for .desktop file for gnome-volume-control-applet
W: ratelimit.c: 838 events suppressed
D: alsa-sink.c: Wakeup from ALSA!
I: alsa-sink.c: Underrun!
D: alsa-sink.c: Wakeup from ALSA!
I: alsa-sink.c: Underrun!
D: alsa-sink.c: Wakeup from ALSA!
I: alsa-sink.c: Underrun!
D: alsa-sink.c: Wakeup from ALSA!
I: alsa-sink.c: Underrun!
D: alsa-sink.c: Wakeup from ALSA!
I: alsa-sink.c: Underrun!
D: alsa-sink.c: Wakeup from ALSA!
I: module-suspend-on-idle.c: Source alsa_input.pci-0000_00_12.0.analog-stereo idle for too long, suspending ...
D: source.c: Suspend cause of source alsa_input.pci-0000_00_12.0.analog-stereo is 0x0004, suspending
I: alsa-source.c: Device suspended...
I: module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_12.0.analog-stereo idle for too long, suspending ...
D: sink.c: Suspend cause of sink alsa_output.pci-0000_00_12.0.analog-stereo is 0x0004, suspending
I: alsa-sink.c: Device suspended...
D: reserve-wrap.c: Device lock status of reserve-monitor-wrapper@Audio0 changed: not busy
D: module-udev-detect.c: /dev/snd/controlC0 is accessible: yes

RayArdia
November 6th, 2009, 08:25 PM
Further to the above I have read, and posted a comment on the Bug #460351 in pulseaudio..... thread.

cesium62
November 8th, 2009, 07:27 AM
I don't understand why I can't install a new version of Linux and have sound and the screen resolution Just Work (tm).

Don't listen to those people telling you how wonderful this guide is. The guide *doesn't work*. pavucontrol says cannot connect, and running pulseaudio first gets a "address already in use error". Do you actually know what you're talking about?







jasmine@monkies:~$ pavucontrol

(pavucontrol:3109): Gtk-CRITICAL **: gtk_main_quit: assertion `main_loops != NULL' failed
jasmine@monkies:~$ pulseaudio & pavucontrol
[1] 3115
E: socket-server.c: bind(): Address already in use
E: module.c: Failed to load module "module-esound-protocol-unix" (argument: ""): initialization failed.
E: main.c: Module load failed.
E: main.c: Failed to initialize daemon.

(pavucontrol:3116): Gtk-CRITICAL **: gtk_main_quit: assertion `main_loops != NULL' failed
[1]+ Exit 1 pulseaudio

Djzn.BR
November 8th, 2009, 03:54 PM
I encourage everyone to upgrade to Karmic, so that we can make this guide definitely obsolete. PulseAudio is working fine in my box. I found out that Aqualung can definitely use the equalizer plugins to boost your sound. Funny enough, it is the same multi-band equalizer used by this guide. Aqualung has a graphical UI with sliders for the equalizer. Anyways if you are not a Aqualung fan, you can use the java UI provided in the mentioned post.

Saronn
November 8th, 2009, 04:25 PM
I use Karmic, and my sound doesn't work. I've turned up every single audio bar as high as it can go, but theres no use. There isn't any sound at all.

lplay -l

**** List of PLAYBACK Hardware Devices ****
card 0: Intel [HDA Intel], device 0: ALC268 Analog [ALC268 Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: Intel [HDA Intel], device 6: Si3054 Modem [Si3054 Modem]
Subdevices: 0/1
Subdevice #0: subdevice #0


pkill pulseaudio; sleep 2; pulseaudio -vv

I: main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted
I: main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation not permitted
D: core-rtclock.c: Timer slack is set to 50 us.
I: core-util.c: Failed to acquire high-priority scheduling: No such file or directory
I: main.c: This is PulseAudio 0.9.19
D: main.c: Compilation host: i486-pc-linux-gnu
D: main.c: Compilation CFLAGS: -g -O2 -g -Wall -O3 -Wall -W -Wextra -pipe -Wno-long-long -Winline -Wvla -Wno-overlength-strings -Wunsafe-loop-optimizations -Wundef -Wformat=2 -Wlogical-op -Wsign-compare -Wformat-security -Wmissing-include-dirs -Wformat-nonliteral -Wold-style-definition -Wpointer-arith -Winit-self -Wdeclaration-after-statement -Wfloat-equal -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wmissing-declarations -Wmissing-noreturn -Wshadow -Wendif-labels -Wcast-align -Wstrict-aliasing=2 -Wwrite-strings -Wno-unused-parameter -ffast-math -Wp,-D_FORTIFY_SOURCE=2 -fno-common -fdiagnostics-show-option
D: main.c: Running on host: Linux i686 2.6.31-14-generic #48-Ubuntu SMP Fri Oct 16 14:04:26 UTC 2009
D: main.c: Found 2 CPUs.
I: main.c: Page size is 4096 bytes
D: main.c: Compiled with Valgrind support: no
D: main.c: Running in valgrind mode: no
D: main.c: Optimized build: yes
D: main.c: All asserts enabled.
I: main.c: Machine ID is 061fc23bd95b07eac51c13ce4ab52da0.
I: main.c: Session ID is 061fc23bd95b07eac51c13ce4ab52da0-1257693427.205161-1713632306.
I: main.c: Using runtime directory /home/saronn/.pulse/061fc23bd95b07eac51c13ce4ab52da0-runtime.
I: main.c: Using state directory /home/saronn/.pulse.
I: main.c: Using modules directory /usr/lib/pulse-0.9.19/modules.
I: main.c: Running in system mode: no
E: pid.c: Daemon already running.
E: main.c: pa_pid_file_create() failed.

rbolio
November 9th, 2009, 12:59 AM
Hi hi hi!...

Heres the deal!:

Last night followed this guide throughly! it worked excelently actually!... I logged out and back in, and it kept working excelently!

However, when i Rebooted the laptop, pulse audio server was not connecting, when attempting to run it from terminal I got this output!


E: module.c: Failed to open module "module-rygel-media-server": file not found
E: module-gconf.c: pa_module_load() failed
E: module.c: Failed to open module "module-raop-discover": file not found
E: module-gconf.c: pa_module_load() failed
E: module-ladspa-sink.c: Master sink not found
E: module.c: Failed to load module "module-ladspa-sink" (argument: "sink_name=ladspa_output.mbeq_1197.mbeq master= plugin=mbeq_1197 label=mbeq control=-1,-1,-1,-1,-5,-10,-18,-15,-10,-5,-5,-5,-5,0,0"): initialization failed.
E: main.c: Module load failed.
E: main.c: Failed to initialize daemon.


I mean, sound works through ALSA (have to control volume through alsamixer....which kinda sucks) but it works.... .

pulse audio manager says (at the very bottom)
"Failure, connection Terminated"..


.....

I own a Dell Inspiron 6400 (E150?) with a HDA intel(STAC92xxx) soundcard... Karmik Koala (ftl?) , and as i mentioned, i already followed this guide...


Thanks for your time! :D

rbolio
November 9th, 2009, 02:03 AM
I just realized how stupid my question is!

Djzn.BR
November 9th, 2009, 02:10 AM
TO WHOM it may concern: I have a Sound Blaster SB128 PCI (Ensoniq chip) pretty ancient hardware, but this little one has an 8W RMS amplifier and I never give up on it. Sorry to all folks running Karmic and not getting PulseAudio to work. Consider what type of hardware you have got. I dislike Realtek chips beyond belief. SB128 Works out of the box with Ubuntu.

rbolio
November 9th, 2009, 03:55 AM
Oh.....sorry to be such a troublesome lad....

I tried to fix pulse...and I still have this output



rbolio@rbolio-laptop:~$ pulseaudio
E: module-ladspa-sink.c: Master sink not found
E: module.c: Failed to load module "module-ladspa-sink" (argument: "sink_name=ladspa_output.mbeq_1197.mbeq master= plugin=mbeq_1197 label=mbeq control=-1,-1,-1,-1,-5,-10,-18,-15,-10,-5,-5,-5,-5,0,0"): initialization failed.
E: main.c: Module load failed.
E: main.c: Failed to initialize daemon.

em3raldxiii
November 9th, 2009, 07:43 AM
Oh.....sorry to be such a troublesome lad....

I tried to fix pulse...and I still have this output



rbolio@rbolio-laptop:~$ pulseaudio
E: module-ladspa-sink.c: Master sink not found
E: module.c: Failed to load module "module-ladspa-sink" (argument: "sink_name=ladspa_output.mbeq_1197.mbeq master= plugin=mbeq_1197 label=mbeq control=-1,-1,-1,-1,-5,-10,-18,-15,-10,-5,-5,-5,-5,0,0"): initialization failed.
E: main.c: Module load failed.
E: main.c: Failed to initialize daemon.



This is precisely what I am getting now at step A5. I have a fresh install of Ubuntu 9.10 karmic; sound was working, but was broken in Chromium (fun game) ... it was all scratchy and horrid sounding. Now it's b0rkd! Will continue muddling for a solution.

EDIT: Upon logout & login, I was able to continue with the instructions without the above error. I will update again once I find out if it fixes the sound in Chromium.

em3raldxiii
November 9th, 2009, 08:21 AM
Followed the guide, everything worked fine, sound is good, except in the game: Chromium-BSUHere is the required information:


desktop:~$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: nForce2 [NVidia nForce2], device 0: Intel ICH [NVidia nForce2]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: nForce2 [NVidia nForce2], device 2: Intel ICH - IEC958 [NVidia nForce2 - IEC958]
Subdevices: 1/1
Subdevice #0: subdevice #0


desktop:~$ pkill pulseaudio; sleep 2; pulseaudio -vv
I: main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted
I: main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation not permitted
D: core-rtclock.c: Timer slack is set to 50 us.
I: core-util.c: Failed to acquire high-priority scheduling: No such file or directory
I: main.c: This is PulseAudio 0.9.19
D: main.c: Compilation host: i486-pc-linux-gnu
D: main.c: Compilation CFLAGS: -g -O2 -g -Wall -O3 -Wall -W -Wextra -pipe -Wno-long-long -Winline -Wvla -Wno-overlength-strings -Wunsafe-loop-optimizations -Wundef -Wformat=2 -Wlogical-op -Wsign-compare -Wformat-security -Wmissing-include-dirs -Wformat-nonliteral -Wold-style-definition -Wpointer-arith -Winit-self -Wdeclaration-after-statement -Wfloat-equal -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wmissing-declarations -Wmissing-noreturn -Wshadow -Wendif-labels -Wcast-align -Wstrict-aliasing=2 -Wwrite-strings -Wno-unused-parameter -ffast-math -Wp,-D_FORTIFY_SOURCE=2 -fno-common -fdiagnostics-show-option
D: main.c: Running on host: Linux i686 2.6.31-14-generic #48-Ubuntu SMP Fri Oct 16 14:04:26 UTC 2009
D: main.c: Found 1 CPUs.
I: main.c: Page size is 4096 bytes
D: main.c: Compiled with Valgrind support: no
D: main.c: Running in valgrind mode: no
D: main.c: Optimized build: yes
D: main.c: All asserts enabled.
I: main.c: Machine ID is 8102534df702661de774ae154ae9ad5b.
I: main.c: Session ID is 8102534df702661de774ae154ae9ad5b-1257749094.618189-33603730.
I: main.c: Using runtime directory /home/christian/.pulse/8102534df702661de774ae154ae9ad5b-runtime.
I: main.c: Using state directory /home/christian/.pulse.
I: main.c: Using modules directory /usr/lib/pulse-0.9.19/modules.
I: main.c: Running in system mode: no
E: pid.c: Daemon already running.
E: main.c: pa_pid_file_create() failed.

I suspect that it is possibly due to Chromium directly accessing the hardware. I will continue to investigate.

psyke83
November 9th, 2009, 11:48 AM
This is precisely what I am getting now at step A5. I have a fresh install of Ubuntu 9.10 karmic; sound was working, but was broken in Chromium (fun game) ... it was all scratchy and horrid sounding. Now it's b0rkd! Will continue muddling for a solution.

EDIT: Upon logout & login, I was able to continue with the instructions without the above error. I will update again once I find out if it fixes the sound in Chromium.

Your problem is the equalizer, which you need to disable. What instructions did you use to enable it?

Djzn.BR
November 9th, 2009, 03:36 PM
I have one issue to report: Suddenly the volume goes very low, this after using the equalizer and rebooting. I get the normal volume back if I run the Python GUI EQ again and click apply.

There is also one issue with the deb file in GNOME Menu:
It misleads the executable to /usr/share/pulseaudio-equalizer.py when it should be /usr/share/pulseaudio-equalizer/pulseaudio-equalizer.py

ntetreau
November 9th, 2009, 06:32 PM
Just a quick question. I'm not sure if this happens only on my setup, if it's a bug or a misunderstood "feature". In the volume properties, I can set the output above 100%, up to 150% in fact. However, the volume applet is limited to 100% which means that when I want to adjust the sound from 150% to 130% for example, using the keyboard or the applet drops the sound directly to 100% or lower. I cannot bring it back up above 100% unless I go back to the volume properties. I would definitely prefer to be able to reach 150% from the applet. Any thoughts? Thanks.

EDIT: I found a bug report on this here (https://bugzilla.gnome.org/show_bug.cgi?id=591285)

It is marked as fixed, but I still have this issue...

jon8798
November 12th, 2009, 09:27 PM
Hi, sorry if this has already been done, but I haven't been able to read through all 157 pages of this thread yet...
I lost all sound after upgrading to Karmic yesterday and tried this guide to see if it would help. On trying to launch the PulseAudio volume control I get the "Connection Failed: Connection refused" message. On trying to manually start through the terminal I get the following:

jon@jon-laptop:~$ pulseaudio & pavucontrol
[1] 4124
E: socket-server.c: bind(): Address already in use
E: module.c: Failed to load module "module-esound-protocol-unix" (argument: ""): initialization failed.
E: main.c: Module load failed.
E: main.c: Failed to initialize daemon.

(pavucontrol:4125): Gtk-CRITICAL **: gtk_main_quit: assertion `main_loops != NULL' failed
[1]+ Exit 1 pulseaudio
jon@jon-laptop:~$



Really grateful for any suggestions.
Thanks,
Jon

jon8798
November 12th, 2009, 10:04 PM
Oh yeah, I just rebooted and now get the little Ubuntu logo then it clears to a terminal screen asking me to logon - only the screen flickers continually and my keyboard inputs flicker too meaning that I can't acurately type - so can't login.
How on earth has playing around with PulseAudio done that?

I'm stuck on XP only for now, any thoughts before I format and reinstall an older Ubuntu version?

ownsuall
November 13th, 2009, 05:58 AM
Hi, sorry if this has already been done, but I haven't been able to read through all 157 pages of this thread yet...
I lost all sound after upgrading to Karmic yesterday and tried this guide to see if it would help. On trying to launch the PulseAudio volume control I get the "Connection Failed: Connection refused" message. On trying to manually start through the terminal I get the following:

jon@jon-laptop:~$ pulseaudio & pavucontrol
[1] 4124
E: socket-server.c: bind(): Address already in use
E: module.c: Failed to load module "module-esound-protocol-unix" (argument: ""): initialization failed.
E: main.c: Module load failed.
E: main.c: Failed to initialize daemon.

(pavucontrol:4125): Gtk-CRITICAL **: gtk_main_quit: assertion `main_loops != NULL' failed
[1]+ Exit 1 pulseaudio
jon@jon-laptop:~$



Really grateful for any suggestions.
Thanks,
Jon
Try running the following in console:
killall pulseaudio
sudo alsa force-reload
pulseaudio -D

raps
November 15th, 2009, 11:55 PM
Hello,

I have a suggestion to complete the following point:
Q. PulseAudio is working correctly, but I am noticing some stuttering on my system. Is there anything I can do to help?

It is said : Note 1: you must restart pulseaudio for any configuration changes to take effect.
Please change it in Note 1: you must restart pulseaudio for any configuration changes to take effect (use "pulseaudio -k" in a terminal).

So, no need to search for the command.

Otherwise I have a new point for your FAQ.
See here:
https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/352732/
The solution that worked for me:
https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/352732/comments/38


Details of the point:
After every boot, the master channel of the sound card is muted (the volume level is set to zero). It seems that the volume level is set to zero when shutting down.
I had that problem on Xubuntu Jaunty 9.04.
The solution is:

comment out line 372 in /etc/init.d/alsa-utils:
# mute_and_zero_levels "$TARGET_CARD" || EXITSTATUS=1

I did not noticed a problem caused by this change, and the master channel is not muted any more *glad*

raps
November 16th, 2009, 12:04 AM
Other problem I solved:
I had no sound. Level were at maximum. I only could get an error: connection failed (or something equivalent).

Running "pulseaudio" in a terminal returned the error (sorry it's french):
$ pulseaudio
I: caps.c: Limited capabilities successfully to CAP_SYS_NICE.
I: caps.c: Dropping root privileges.
I: caps.c: Limited capabilities successfully to CAP_SYS_NICE.
W: alsa-util.c: Device front:0 doesn't support 44100 Hz, changed to 48000 Hz.
W: authkey.c: Failed to open cookie file '/home/thomas/.pulse-cookie': Permission non accordée
W: authkey.c: Failed to load authorization key '/home/thomas/.pulse-cookie': Permission non accordée (means no rights)
E: module.c: Failed to load module "module-native-protocol-unix" (argument: ""): initialization failed.
E: main.c: Module load failed.
E: main.c: Échec lors de l'initialisation du démon

A diagnostic gave me:
$ ls -l .pulse-cookie
-rw------- 1 root root 256 2009-08-16 19:03 .pulse-cookie

I solved it with (with username replaced by my user name):
sudo chown username:username ~/.pulse-cookie

My system:
Xubuntu Jaunty 9.04

bytecode
November 16th, 2009, 07:13 PM
I have one issue to report: Suddenly the volume goes very low, this after using the equalizer and rebooting. I get the normal volume back if I run the Python GUI EQ again and click apply.

There is also one issue with the deb file in GNOME Menu:
It misleads the executable to /usr/share/pulseaudio-equalizer.py when it should be /usr/share/pulseaudio-equalizer/pulseaudio-equalizer.py


Ah ha! I was getting the


"Connection refused"


"module-ladspa-sink.c: Master sink not found"

errors, but linking to correct the above path issues sorted it:

sudo ln -s /usr/share/pulseaudio-equalizer/pulseaudio-equalizer.py /usr/share/pulseaudio-equalizer.py

wbb
November 24th, 2009, 06:10 PM
Thanks for the great help, I now have sound on my ZOTAC ION ITX A.

aimdeey
November 24th, 2009, 07:16 PM
Hello everybody!

Yesterday I decided to upgrade from Ubuntu 9.04 (i386) to 9.10. Sound in 9.04 used to go OK, but after the upgrade there's no sound at all.

I've followed your steps for Karmic Koala in part A, and I still get no sound. Moreover, if I type "aplay -l" in the terminal it says

aplay: device_list:223: no se encontraron tarjetas de sonido...

That is, "no sound card was detected". Do you know what to do?

Thanks! =)

EDIT: I found a link that helped me fix it: https://bugs.launchpad.net/ubuntu/+source/linux-rt/+bug/394282

One of the last comments says:


sudo apt-get --purge remove linux-sound-base alsa-base alsa-utils alsa-source pulseaudio

sudo apt-get install build-essential linux-headers-$(uname -r) module-assistant alsa-source

sudo dpkg-reconfigure alsa-source

sudo module-assistant a-i alsa-source

reboot

It worked for me =D

ncpokrt
November 24th, 2009, 08:43 PM
THANK YOU SO-O-O-O-O MUCH!!!

I have been going through threads and trying everything. I learned a lot in the process but it was a frustrating experience. Nothing worked until I came upon your HOWTO. Hearing the Ubuntu login was joyful.

KiaZiShiru
November 25th, 2009, 11:26 AM
Try running the following in console:
killall pulseaudio
sudo alsa force-reload
pulseaudio -D


I had the same problem and tried this but now I get this error:

kiazishiru@blackybrother:~$ sudo alsa force-reload
lsof: WARNING: can't stat() fuse.gvfs-fuse-daemon file system /home/kiazishiru/.gvfs
Output information may be incomplete.
Terminating processes: 2615lsof: WARNING: can't stat() fuse.gvfs-fuse-daemon file system /home/kiazishiru/.gvfs
Output information may be incomplete.
lsof: WARNING: can't stat() fuse.gvfs-fuse-daemon file system /home/kiazishiru/.gvfs
Output information may be incomplete.
.
lsof: WARNING: can't stat() fuse.gvfs-fuse-daemon file system /home/kiazishiru/.gvfs
Output information may be incomplete.
Unloading ALSA sound driver modules: snd-hda-codec-realtek snd-hda-intel snd-hda-codec snd-hwdep snd-pcm-oss snd-mixer-oss snd-pcm snd-seq-dummy snd-seq-oss snd-seq-midi snd-rawmidi snd-seq-midi-event snd-seq snd-timer snd-seq-device snd-page-alloc (failed: modules still loaded: snd-hda-codec-realtek snd-hda-codec snd-hwdep snd-pcm snd-timer snd-page-alloc).
Loading ALSA sound driver modules: snd-hda-codec-realtek snd-hda-intel snd-hda-codec snd-hwdep snd-pcm-oss snd-mixer-oss snd-pcm snd-seq-dummy snd-seq-oss snd-seq-midi snd-rawmidi snd-seq-midi-event snd-seq snd-timer snd-seq-device snd-page-alloc.
kiazishiru@blackybrother:~$ pulseaudio -D
E: main.c: Daemon startup failed.


Iǘe been trying all different kinds of things in the past but still don have my sound working, itś a clean install.


*edit*
after reboot and rerun of pavucontrol sound is now working.

ncpokrt
November 25th, 2009, 06:25 PM
I got it working yesterday following your instructions but this morning the sound was gone and I can't get it back.

My program shows up in the pulseaudio Volume Control but I get no sound. I tried to attach the output you requested in a file because there is so much of it. However, that file exceeds the forum file size limit.

I will go through the output to see if I can get any clues as to the problem.

BTW - I could not redirect the output to a file and the terminal scroll cut off the output. I increased the scroll length in the terminal; then I selected all of the text in the terminal window and pasted it into a text file.

HeadHunter00
November 26th, 2009, 01:32 AM
Finally a solution for the sound problems

HeadHunter00
November 26th, 2009, 01:32 AM
I love you man

christopherreay
November 26th, 2009, 04:10 PM
Hi there,

Im tryin to get my first open source project on the board, and with so MANY things I would like contribute to, I thought I would start with a rock. Computer Games. I love Spring, but the sound simply doesnt work with PA. Now ive read (really quite a lot) of posts you have made about glitch free, etc, and Spring RTS has an ubunutu repo and all that.

So here's the issue. Spring connects through ALSA plugin to pulse audio. Works, sort of... and then (very soon) there comes a LOUD stutter grainy greating noise, intermittently filled by perfect audio.
So I open the Pavu volume control... Wierd. Spring is there (ALSA Plugin (spring)) however during the periods that it is noisy, its entry in the playing tab flickers. Spring is being dismissed/reattached to pulse in a kind of dance of the sugar plum graters.

Im sure this is something to do with how spring is ... pfft.. nope im not sure of anything. Do you have any ideas. Id love to contribute to the Spring codebase and this would be inline with ubuntu/gnome/repos/LINUX yey

p.s. also skype [edit: (through PA)] still crashes my sound card intermittently [edit: (like the the mike hard crashes and I have to turn my machine OFF to get it to work again)], and in koala i do not have a single player that syncs a/v except totem. pfft, pfft and PFTT again!

Christopher

ierotheo
November 27th, 2009, 07:12 PM
I run Ubuntu 9.10. I followed your instructions and still I have no microphone.
Also, I use a USB phone for Skype and every time I have to go to Sound Preferences and choose the USB phone as an output and input to use it with Skype. At the end of a call if I want to listen to music I have to go back to Sound Preferences and Choose the internal Audio output.
Is there a way to select one thing for Skype totally independent of the rest of the system so I do not have to go back and forth?
Thank you,
ierotheo

detyabozhye
November 28th, 2009, 03:08 AM
Awesome, my sound mixing works great now!

ktronicon5
November 28th, 2009, 09:27 PM
Hello,
Does anyone know any workarounds for the following bug/issue:


The application does not play audio and does list an entry in the Playback tab; the application is using PulseAudio but there is a problem, such as: a bug in PulseAudio, a problem with your ALSA kernel module or libraries, or your PCM/Master volume is muted.

I finally got pulse up and running, but now i'm stumped. I'm pretty new to all this, don't know how to check my ALSA kernel module for problems, for example.

Volume control is not muted, and lists a single "dummy output."
My soundcard is a Hoontech DSP24, 8-in, 8-out.
Alsamixer displays all inputs and outputs. HW 0-7 for example defaults PCM Out and have no volume control or mute options.
I managed to get the sound to work once, but only for Rhythmbox, and only that once. No idea. Can anyone help??
Thanks,
Keith

garvinrick4
November 29th, 2009, 05:25 AM
I used this from the original thread on a Laptop I bought
11/24/09 a HP G71-34US and it is the only one that made my
Intel sound card work after hours of looking and trying.
I decided to put another partition of Ubuntu for 10.04 so I
can keep one stable and one testing this time.
I lost this web page and I had to hunt forever and when I put
in the Karmic fix I will be darned if it did not work again.
It acts like it doe's not have a chance to work and then bingo
noise from my speakers. I will not lose this Thread again. I was up
all night looking for something to work again. Sound problems are
so baffling. Look at this Grub went nuts last night and wiped it
and started over in a fit of energy. Partitioning software is so
good now. The only thing that stops me from doing more is the size of the boot menu gets so long when you get a couple of different Linux kernels for each partition.
Anyway I sure am greatfull for this fix for my sound.


Device Boot Start End Blocks Id System
/dev/sda1 * 1 26 203776 7 HPFS/NTFS
Partition 1 does not end on cylinder boundary.
/dev/sda2 26 26132 209696248 7 HPFS/NTFS
/dev/sda3 26133 37330 89947935 5 Extended
/dev/sda4 37331 38914 12709888 7 HPFS/NTFS
/dev/sda5 32107 37111 40202631 83 Linux
/dev/sda6 37112 37330 1759086 82 Linux swap / Solaris
/dev/sda7 26133 31856 45977967 83 Linux
/dev/sda8 31857 32106 2008093+ 82 Linux swap / Solaris

Partition table entries are not in disk order

Greencoat1982
November 29th, 2009, 05:17 PM
All this guide did was show me how to crash my audio.

HeadHunter00
November 30th, 2009, 05:13 PM
Thanks for the tutorial. I am having problems with mplayer. On a stock install, mplayer just wouldn't work, due to some pulseaudio connection refused problem. I solved it by using -ao alsa. Now after using your tutorial, none of the methods seems to work :( I get proper sound in firefox/flash, but now mplayer is borked :( I can't use amarok either, it says - "xine: unable to initialize any audio drivers".

Here is the (relevant) mplayer output:

rohan@ubuntu:/media/sda6/Fav_songs/Hindi$ mplayer Saher/Palkein\ Jhukaao\ Na.mp3 -ao pulse
AO: [pulse] Failed to connect to server: Connection refused
Could not open/initialize audio device -> no sound.
Audio: no sound
Video: no video

rohan@ubuntu:/media/sda6/Fav_songs/Hindi$ mplayer Saher/Palkein\ Jhukaao\ Na.mp3 -ao alsa
*** PULSEAUDIO: Unable to connect: Connection refused
[AO_ALSA] Playback open error: Connection refused
Could not open/initialize audio device -> no sound.
Audio: no sound
Video: no video

rohan@ubuntu:/media/sda6/Fav_songs/Hindi$ mplayer Saher/Palkein\ Jhukaao\ Na.mp3 -ao oss
[AO OSS] audio_setup: Can't open audio device /dev/dsp: Device or resource busy
Could not open/initialize audio device -> no sound.
Audio: no sound
Video: no video

I am using Kubuntu, and not Ubuntu, does that make any difference? I know I can roll back all the changes, but somehow what you've done seems correct, so I don't want to roll back. Is this a problem specific to my system?
Kubuntu can't use this equalizer. The volume should be fine in kubuntu by disabling it. This script only works with ubuntu, as far as I know.

HeadHunter00
November 30th, 2009, 05:15 PM
All this guide did was show me how to crash my audio.
Again, it doesn't work in xubuntu either; just the normal gnome based ubuntu

psyke83
November 30th, 2009, 06:31 PM
Oh.....sorry to be such a troublesome lad....

I tried to fix pulse...and I still have this output



rbolio@rbolio-laptop:~$ pulseaudio
E: module-ladspa-sink.c: Master sink not found
E: module.c: Failed to load module "module-ladspa-sink" (argument: "sink_name=ladspa_output.mbeq_1197.mbeq master= plugin=mbeq_1197 label=mbeq control=-1,-1,-1,-1,-5,-10,-18,-15,-10,-5,-5,-5,-5,0,0"): initialization failed.
E: main.c: Module load failed.
E: main.c: Failed to initialize daemon.


Your problem is the equalizer, not PulseAudio itself.

Make sure you're using the latest script (from the dedicated thread), and then use the "Reset to defaults" menu option to get sound working again.

If you every see errors mentioning LADSPA, it's *always* related to the equalizer. Keep that in mind, and please reply in the other thread if you need further assistance.

jamesisin
December 1st, 2009, 08:29 AM
Any audio experts want to take a crack at my 9.10/Rhythmbox problem?

http://ubuntuforums.org/showthread.php?t=1326861

Basically, Rb jumps from song to song very rapidly. No other audio application is being affected and PA doesn't seem to be the culprit (nor does gstreamer for that matter). I'm stumped and really could use a solution.

lkraemer
December 2nd, 2009, 03:18 AM
psyke83

I am trying to get sound functional on a New Compaq CQ61-313US that has the following:

AMD Athlon II M300 with 3 Gig RAM and a 250 Gig Hitachi SATA Drive.
I have installed Ubuntu karmic (9.10) and the Kernel that is being
used is 2.6.31-15-generic.

I have followed your instructions in this posting and still don't
have any sound.


I have followed the instructions in Appendix A - General Troubleshooting

1. Close all applications that may be accessing the sound card.
2. Open the "PulseAudio Device Chooser" from Applications/Sound & Video. Click on the applet icon, and click "Volume Control...". Click on the "Playback" tab.
3. Launch the application you wish to test and allow it to play sound.
4. Check PulseAudio Volume Control's Playback tab and see if the application displays an entry while the application is (or should be) playing audio.

My result is:
The application does not play audio and does list an entry in the Playback tab;
- the application is using PulseAudio but there is a problem, such as: a bug in PulseAudio, a problem with your ALSA kernel module or libraries, or your PCM/Master volume is muted.


rick@rick-laptop:~$ uname -r
2.6.31-15-generic

rick@rick-laptop:~$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: SB [HDA ATI SB], device 0: STAC92xx Analog [STAC92xx Analog]
Subdevices: 0/1
Subdevice #0: subdevice #0

rick@rick-laptop:~$ pkill pulseaudio; sleep 2; pulseaudio -vv
There is so much output that the beginning is lost. And I haven't
found a way to capture that information. I tried to pipe the output
to a log file, but that did not work.

D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-40+input:iec958-surround-40
D: alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC958/AC3) (iec958-ac3-surround-40)
D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-51
D: alsa-mixer.c: Checking for playback on Digital Surround 5.1 (IEC958/AC3) (iec958-ac3-surround-51)
D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-51+input:analog-mono
D: alsa-mixer.c: Checking for playback on Digital Surround 5.1 (IEC958/AC3) (iec958-ac3-surround-51)
D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-51+input:analog-stereo
D: alsa-mixer.c: Checking for playback on Digital Surround 5.1 (IEC958/AC3) (iec958-ac3-surround-51)
D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-51+input:iec958-stereo
D: alsa-mixer.c: Checking for playback on Digital Surround 5.1 (IEC958/AC3) (iec958-ac3-surround-51)
D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-51+input:iec958-surround-40
D: alsa-mixer.c: Checking for playback on Digital Surround 5.1 (IEC958/AC3) (iec958-ac3-surround-51)
D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:hdmi-stereo
D: alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo)
D: alsa-util.c: Trying hdmi:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm_hw.c: open /dev/snd/pcmC0D3p failed
I: alsa-util.c: Error opening PCM device hdmi:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:hdmi-stereo+input:analog-mono
D: alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo)
D: alsa-util.c: Trying hdmi:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm_hw.c: open /dev/snd/pcmC0D3p failed
I: alsa-util.c: Error opening PCM device hdmi:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:hdmi-stereo+input:analog-stereo
D: alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo)
D: alsa-util.c: Trying hdmi:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm_hw.c: open /dev/snd/pcmC0D3p failed
I: alsa-util.c: Error opening PCM device hdmi:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:hdmi-stereo+input:iec958-stereo
D: alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo)
D: alsa-util.c: Trying hdmi:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm_hw.c: open /dev/snd/pcmC0D3p failed
I: alsa-util.c: Error opening PCM device hdmi:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:hdmi-stereo+input:iec958-surround-40
D: alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo)
D: alsa-util.c: Trying hdmi:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm_hw.c: open /dev/snd/pcmC0D3p failed
I: alsa-util.c: Error opening PCM device hdmi:0: No such file or directory
D: alsa-mixer.c: Looking at profile input:analog-mono
D: alsa-mixer.c: Checking for recording on Analog Mono (analog-mono)
D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid argument
D: alsa-mixer.c: Looking at profile input:analog-stereo
D: alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo)
D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open front:0
D: alsa-util.c: Maximum hw buffer size is 371 ms
D: alsa-util.c: Set buffer size first, period size second.
D: alsa-mixer.c: Profile input:analog-stereo supported.
D: alsa-mixer.c: Looking at profile input:iec958-stereo
D: alsa-mixer.c: Checking for recording on Digital Stereo (IEC958) (iec958-stereo)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm_hw.c: open /dev/snd/pcmC0D1c failed
I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
D: alsa-mixer.c: Looking at profile input:iec958-surround-40
D: alsa-mixer.c: Checking for recording on Digital Surround 4.0 (IEC958) (iec958-surround-40)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm_hw.c: open /dev/snd/pcmC0D1c failed
I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
I: card.c: Created 0 "alsa_card.pci-0000_00_14.2"
D: reserve-wrap.c: Successfully create reservation lock monitor for device 'Audio0'
D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open front:0
D: alsa-util.c: Maximum hw buffer size is 371 ms
D: alsa-util.c: Set buffer size first, period size second.
I: alsa-sink.c: Successfully opened device front:0.
I: alsa-sink.c: Selected mapping 'Analog Stereo' (analog-stereo).
I: alsa-sink.c: Successfully enabled mmap() mode.
I: alsa-sink.c: Successfully enabled timer-based scheduling mode.
I: (alsa-lib)control.c: Invalid CTL front:0
I: alsa-mixer.c: Unable to attach to mixer front:0: No such file or directory
I: alsa-mixer.c: Successfully attached to mixer 'hw:0'
D: alsa-mixer.c: Probing path 'analog-output'
D: alsa-mixer.c: Probing path 'analog-output-headphones'
D: alsa-mixer.c: Probing path 'analog-output-mono'
D: alsa-mixer.c: Probe of element 'Master Mono' failed.
D: alsa-mixer.c: Probing path 'analog-output-lfe-on-mono'
D: alsa-mixer.c: Probe of element 'Master Mono' failed.
D: alsa-sink.c: Probed mixer paths:
D: alsa-mixer.c: Path Set 0x827fb08, direction=1, probed=yes
D: alsa-mixer.c: Path analog-output (Analog Output), direction=1, priority=100, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=64, min_dB=-147, max_dB=0
D: alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes
D: alsa-mixer.c: Element Headphone, direction=1, switch=1, volume=3, enumeration=0, required=0, required_absent=0, mask=0x6, n_channels=2, override_map=no
D: alsa-mixer.c: Element Speaker, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes
D: alsa-mixer.c: Element PCM, direction=1, switch=0, volume=1, enumeration=0, required=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes
D: alsa-mixer.c: Path analog-output-headphones (Analog Headphones), direction=1, priority=90, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=64, min_dB=-147, max_dB=0
D: alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes
D: alsa-mixer.c: Element Headphone, direction=1, switch=1, volume=1, enumeration=0, required=4, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes
D: alsa-mixer.c: Element Speaker, direction=1, switch=2, volume=2, enumeration=0, required=0, required_absent=0, mask=0x6, n_channels=2, override_map=no
D: alsa-mixer.c: Element PCM, direction=1, switch=0, volume=1, enumeration=0, required=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes
D: alsa-mixer.c: Added 2 ports
I: module-device-restore.c: Restoring port for sink sink:alsa_output.pci-0000_00_14.2.analog-stereo.
I: module-device-restore.c: Restoring volume for sink alsa_output.pci-0000_00_14.2.analog-stereo.
I: module-device-restore.c: Restoring mute state for sink alsa_output.pci-0000_00_14.2.analog-stereo.
I: sink.c: Created sink 0 "alsa_output.pci-0000_00_14.2.analog-stereo" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: sink.c: alsa.resolution_bits = "16"
I: sink.c: device.api = "alsa"
I: sink.c: device.class = "sound"
I: sink.c: alsa.class = "generic"
I: sink.c: alsa.subclass = "generic-mix"
I: sink.c: alsa.name = "STAC92xx Analog"
I: sink.c: alsa.id = "STAC92xx Analog"
I: sink.c: alsa.subdevice = "0"
I: sink.c: alsa.subdevice_name = "subdevice #0"
I: sink.c: alsa.device = "0"
I: sink.c: alsa.card = "0"
I: sink.c: alsa.card_name = "HDA ATI SB"
I: sink.c: alsa.long_card_name = "HDA ATI SB at 0xd0400000 irq 16"
I: sink.c: alsa.driver_name = "snd_hda_intel"
I: sink.c: device.bus_path = "pci-0000:00:14.2"
I: sink.c: sysfs.path = "/devices/pci0000:00/0000:00:14.2/sound/card0"
I: sink.c: device.bus = "pci"
I: sink.c: device.vendor.id = "1002"
I: sink.c: device.vendor.name = "ATI Technologies Inc"
I: sink.c: device.product.id = "4383"
I: sink.c: device.product.name = "SBx00 Azalia (Intel HDA)"
I: sink.c: device.form_factor = "internal"
I: sink.c: device.string = "front:0"
I: sink.c: device.buffering.buffer_size = "65536"
I: sink.c: device.buffering.fragment_size = "32768"
I: sink.c: device.access_mode = "mmap+timer"
I: sink.c: device.profile.name = "analog-stereo"
I: sink.c: device.profile.description = "Analog Stereo"
I: sink.c: device.description = "Internal Audio Analog Stereo"
I: sink.c: alsa.mixer_name = "IDT 92HD75B2X5"
I: sink.c: alsa.components = "HDA:111d7608,103c363f,00100202 HDA:11c11040,103c137e,00100200"
I: sink.c: module-udev-detect.discovered = "1"
I: sink.c: device.icon_name = "audio-card-pci"
D: core-subscribe.c: Dropped redundant event due to change event.
I: source.c: Created source 0 "alsa_output.pci-0000_00_14.2.analog-stereo.monitor" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: source.c: device.description = "Monitor of Internal Audio Analog Stereo"
I: source.c: device.class = "monitor"
I: source.c: alsa.card = "0"
I: source.c: alsa.card_name = "HDA ATI SB"
I: source.c: alsa.long_card_name = "HDA ATI SB at 0xd0400000 irq 16"
I: source.c: alsa.driver_name = "snd_hda_intel"
I: source.c: device.bus_path = "pci-0000:00:14.2"
I: source.c: sysfs.path = "/devices/pci0000:00/0000:00:14.2/sound/card0"
I: source.c: device.bus = "pci"
I: source.c: device.vendor.id = "1002"
I: source.c: device.vendor.name = "ATI Technologies Inc"
I: source.c: device.product.id = "4383"
I: source.c: device.product.name = "SBx00 Azalia (Intel HDA)"
I: source.c: device.form_factor = "internal"
I: source.c: device.string = "0"
I: source.c: module-udev-detect.discovered = "1"
I: source.c: device.icon_name = "audio-card-pci"
I: alsa-sink.c: Using 2.0 fragments of size 32768 bytes (185.76ms), buffer size is 65536 bytes (371.52ms)
I: alsa-sink.c: Time scheduling watermark is 20.00ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=15502
D: alsa-mixer.c: Activating path analog-output
D: alsa-mixer.c: Path analog-output (Analog Output), direction=1, priority=100, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=64, min_dB=-147, max_dB=0
D: alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes
D: alsa-mixer.c: Element Headphone, direction=1, switch=1, volume=3, enumeration=0, required=0, required_absent=0, mask=0x6, n_channels=2, override_map=no
D: alsa-mixer.c: Element Speaker, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes
D: alsa-mixer.c: Element PCM, direction=1, switch=0, volume=1, enumeration=0, required=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes
I: alsa-sink.c: Hardware volume ranges from -147.00 dB to 0.00 dB.
I: alsa-sink.c: No particular base volume set, fixing to 0 dB
I: alsa-sink.c: Using hardware volume control. Hardware dB scale supported.
I: alsa-sink.c: Using hardware mute control.
D: alsa-util.c: snd_pcm_dump():
D: alsa-util.c: Soft volume PCM
D: alsa-util.c: Control: PCM Playback Volume
D: alsa-util.c: min_dB: -51
D: alsa-util.c: max_dB: 0
D: alsa-util.c: resolution: 256
D: alsa-util.c: Its setup is:
D: alsa-util.c: stream : PLAYBACK
D: alsa-util.c: access : MMAP_INTERLEAVED
D: alsa-util.c: format : S16_LE
D: alsa-util.c: subformat : STD
D: alsa-util.c: channels : 2
D: alsa-util.c: rate : 44100
D: alsa-util.c: exact rate : 44100 (44100/1)
D: alsa-util.c: msbits : 16
D: alsa-util.c: buffer_size : 16384
D: alsa-util.c: period_size : 8192
D: alsa-util.c: period_time : 185759
D: alsa-util.c: tstamp_mode : ENABLE
D: alsa-util.c: period_step : 1
D: alsa-util.c: avail_min : 15502
D: alsa-util.c: period_event : 0
D: alsa-util.c: start_threshold : -1
D: alsa-util.c: stop_threshold : 1073741824
D: alsa-util.c: silence_threshold: 0
D: alsa-util.c: silence_size : 0
D: alsa-util.c: boundary : 1073741824
D: alsa-util.c: Slave: Hardware PCM card 0 'HDA ATI SB' device 0 subdevice 0
D: alsa-util.c: Its setup is:
D: alsa-util.c: stream : PLAYBACK
D: alsa-util.c: access : MMAP_INTERLEAVED
D: alsa-util.c: format : S16_LE
D: alsa-util.c: subformat : STD
D: alsa-util.c: channels : 2
D: alsa-util.c: rate : 44100
D: alsa-util.c: exact rate : 44100 (44100/1)
D: alsa-util.c: msbits : 16
D: alsa-util.c: buffer_size : 16384
D: alsa-util.c: period_size : 8192
D: alsa-util.c: period_time : 185759
D: alsa-util.c: tstamp_mode : ENABLE
D: alsa-util.c: period_step : 1
D: alsa-util.c: avail_min : 15502
D: alsa-util.c: period_event : 0
D: alsa-util.c: start_threshold : -1
D: alsa-util.c: stop_threshold : 1073741824
D: alsa-util.c: silence_threshold: 0
D: alsa-util.c: silence_size : 0
D: alsa-util.c: boundary : 1073741824
D: alsa-util.c: appl_ptr : 0
D: alsa-util.c: hw_ptr : 0
D: alsa-sink.c: Thread starting up
I: core-util.c: Successfully enabled SCHED_RR scheduling for thread, with priority 4, which is lower than the requested 5.
D: alsa-sink.c: Requested volume: 0: 96% 1: 96%
D: alsa-sink.c: Got hardware volume: 0: 96% 1: 96%
D: alsa-sink.c: Calculated software volume: 0: 100% 1: 100% (accurate-enough=yes)
I: alsa-sink.c: Starting playback.
D: alsa-sink.c: Cutting sleep time for the initial iterations by half.
D: alsa-sink.c: Cutting sleep time for the initial iterations by half.
D: alsa-sink.c: Cutting sleep time for the initial iterations by half.
D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open front:0
D: alsa-util.c: Maximum hw buffer size is 371 ms
D: alsa-util.c: Set buffer size first, period size second.
I: alsa-source.c: Successfully opened device front:0.
I: alsa-source.c: Selected mapping 'Analog Stereo' (analog-stereo).
I: alsa-source.c: Successfully enabled mmap() mode.
I: alsa-source.c: Successfully enabled timer-based scheduling mode.
I: (alsa-lib)control.c: Invalid CTL front:0
I: alsa-mixer.c: Unable to attach to mixer front:0: No such file or directory
I: alsa-mixer.c: Successfully attached to mixer 'hw:0'
D: alsa-mixer.c: Probing path 'analog-input'
D: alsa-mixer.c: Probing path 'analog-input-microphone'
D: alsa-mixer.c: Probe of element 'Mic' failed.
D: alsa-mixer.c: Probing path 'analog-input-linein'
D: alsa-mixer.c: Probe of element 'Line' failed.
D: alsa-mixer.c: Probing path 'analog-input'
D: alsa-mixer.c: Probe of element 'Aux' failed.
D: alsa-mixer.c: Probing path 'analog-input-video'
D: alsa-mixer.c: Probe of element 'Video' failed.
D: alsa-mixer.c: Probing path 'analog-input-video'
D: alsa-mixer.c: Probe of element 'TV Tuner' failed.
D: alsa-mixer.c: Probing path 'analog-input-radio'
D: alsa-mixer.c: Probe of element 'FM' failed.
D: alsa-mixer.c: Probing path 'analog-input'
D: alsa-mixer.c: Probe of element 'Mic/Line' failed.
D: alsa-source.c: Probed mixer paths:
D: alsa-mixer.c: Path Set 0x828d138, direction=2, probed=yes
D: alsa-mixer.c: Path analog-input (Analog Input), direction=2, priority=100, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=15, min_dB=0, max_dB=22.5
D: alsa-mixer.c: Element Capture, direction=2, switch=1, volume=1, enumeration=0, required=2, required_absent=0, mask=0x4037e00000000f66, n_channels=2, override_map=yes
D: alsa-mixer.c: Element Input Source, direction=2, switch=0, volume=0, enumeration=1, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Option Mic (input-microphone-1/Microphone 1) index=0, priority=20
D: alsa-mixer.c: Option Front Mic (input-microphone-2/Microphone 2) index=1, priority=19
D: alsa-mixer.c: Setting input-microphone-1 (Microphone 1) priority=20
D: alsa-mixer.c: Setting input-microphone-2 (Microphone 2) priority=19
D: alsa-mixer.c: Added 2 ports
D: core-subscribe.c: Dropped redundant event due to change event.
I: module-device-restore.c: Restoring port for source source:alsa_input.pci-0000_00_14.2.analog-stereo.
I: source.c: Created source 1 "alsa_input.pci-0000_00_14.2.analog-stereo" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: source.c: alsa.resolution_bits = "16"
I: source.c: device.api = "alsa"
I: source.c: device.class = "sound"
I: source.c: alsa.class = "generic"
I: source.c: alsa.subclass = "generic-mix"
I: source.c: alsa.name = "STAC92xx Analog"
I: source.c: alsa.id = "STAC92xx Analog"
I: source.c: alsa.subdevice = "0"
I: source.c: alsa.subdevice_name = "subdevice #0"
I: source.c: alsa.device = "0"
I: source.c: alsa.card = "0"
I: source.c: alsa.card_name = "HDA ATI SB"
I: source.c: alsa.long_card_name = "HDA ATI SB at 0xd0400000 irq 16"
I: source.c: alsa.driver_name = "snd_hda_intel"
I: source.c: device.bus_path = "pci-0000:00:14.2"
I: source.c: sysfs.path = "/devices/pci0000:00/0000:00:14.2/sound/card0"
I: source.c: device.bus = "pci"
I: source.c: device.vendor.id = "1002"
I: source.c: device.vendor.name = "ATI Technologies Inc"
I: source.c: device.product.id = "4383"
I: source.c: device.product.name = "SBx00 Azalia (Intel HDA)"
I: source.c: device.form_factor = "internal"
I: source.c: device.string = "front:0"
I: source.c: device.buffering.buffer_size = "65536"
I: source.c: device.buffering.fragment_size = "32768"
I: source.c: device.access_mode = "mmap+timer"
I: source.c: device.profile.name = "analog-stereo"
I: source.c: device.profile.description = "Analog Stereo"
I: source.c: device.description = "Internal Audio Analog Stereo"
I: source.c: alsa.mixer_name = "IDT 92HD75B2X5"
I: source.c: alsa.components = "HDA:111d7608,103c363f,00100202 HDA:11c11040,103c137e,00100200"
I: source.c: module-udev-detect.discovered = "1"
I: source.c: device.icon_name = "audio-card-pci"
I: alsa-source.c: Using 2.0 fragments of size 32768 bytes (185.76ms), buffer size is 65536 bytes (371.52ms)
I: alsa-source.c: Time scheduling watermark is 20.00ms
D: alsa-source.c: hwbuf_unused=0
D: alsa-source.c: setting avail_min=15502
D: alsa-mixer.c: Activating path analog-input
D: alsa-mixer.c: Path analog-input (Analog Input), direction=2, priority=100, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=15, min_dB=0, max_dB=22.5
D: alsa-mixer.c: Element Capture, direction=2, switch=1, volume=1, enumeration=0, required=2, required_absent=0, mask=0x4037e00000000f66, n_channels=2, override_map=yes
D: alsa-mixer.c: Element Input Source, direction=2, switch=0, volume=0, enumeration=1, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Option Mic (input-microphone-1/Microphone 1) index=0, priority=20
D: alsa-mixer.c: Option Front Mic (input-microphone-2/Microphone 2) index=1, priority=19
D: alsa-mixer.c: Setting input-microphone-1 (Microphone 1) priority=20
D: alsa-mixer.c: Setting input-microphone-2 (Microphone 2) priority=19
I: alsa-source.c: Hardware volume ranges from 0.00 dB to 22.50 dB.
I: alsa-source.c: Fixing base volume to -22.50 dB
I: alsa-source.c: Using hardware volume control. Hardware dB scale supported.
I: alsa-source.c: Using hardware mute control.
D: alsa-util.c: snd_pcm_dump():
D: alsa-util.c: Soft volume PCM
D: alsa-util.c: Control: PCM Playback Volume
D: alsa-util.c: min_dB: -51
D: alsa-util.c: max_dB: 0
D: alsa-util.c: resolution: 256
D: alsa-util.c: Its setup is:
D: alsa-util.c: stream : CAPTURE
D: alsa-util.c: access : MMAP_INTERLEAVED
D: alsa-util.c: format : S16_LE
D: alsa-util.c: subformat : STD
D: alsa-util.c: channels : 2
D: alsa-util.c: rate : 44100
D: alsa-util.c: exact rate : 44100 (44100/1)
D: alsa-util.c: msbits : 16
D: alsa-util.c: buffer_size : 16384
D: alsa-util.c: period_size : 8192
D: alsa-util.c: period_time : 185759
D: alsa-util.c: tstamp_mode : ENABLE
D: alsa-util.c: period_step : 1
D: alsa-util.c: avail_min : 15502
D: alsa-util.c: period_event : 0
D: alsa-util.c: start_threshold : -1
D: alsa-util.c: stop_threshold : 1073741824
D: alsa-util.c: silence_threshold: 0
D: alsa-util.c: silence_size : 0
D: alsa-util.c: boundary : 1073741824
D: alsa-util.c: Slave: Hardware PCM card 0 'HDA ATI SB' device 0 subdevice 0
D: alsa-util.c: Its setup is:
D: alsa-util.c: stream : CAPTURE
D: alsa-util.c: access : MMAP_INTERLEAVED
D: alsa-util.c: format : S16_LE
D: alsa-util.c: subformat : STD
D: alsa-util.c: channels : 2
D: alsa-util.c: rate : 44100
D: alsa-util.c: exact rate : 44100 (44100/1)
D: alsa-util.c: msbits : 16
D: alsa-util.c: buffer_size : 16384
D: alsa-util.c: period_size : 8192
D: alsa-util.c: period_time : 185759
D: alsa-util.c: tstamp_mode : ENABLE
D: alsa-util.c: period_step : 1
D: alsa-util.c: avail_min : 15502
D: alsa-util.c: period_event : 0
D: alsa-util.c: start_threshold : -1
D: alsa-util.c: stop_threshold : 1073741824
D: alsa-util.c: silence_threshold: 0
D: alsa-util.c: silence_size : 0
D: alsa-util.c: boundary : 1073741824
D: alsa-util.c: appl_ptr : 0
D: alsa-util.c: hw_ptr : 0
D: alsa-source.c: Thread starting up
I: core-util.c: Successfully enabled SCHED_RR scheduling for thread, with priority 4, which is lower than the requested 5.
D: alsa-source.c: Read hardware volume: 0: 42% 1: 42%
I: module.c: Loaded "module-alsa-card" (index: #4; argument: "device_id="0" name="pci-0000_00_14.2" card_name="alsa_card.pci-0000_00_14.2" tsched=yes ignore_dB=no card_properties="module-udev-detect.discovered=1"").
I: module-udev-detect.c: Card /devices/pci0000:00/0000:00:14.2/sound/card0 (alsa_card.pci-0000_00_14.2) module loaded.
I: module-udev-detect.c: Found 1 cards.
I: module.c: Loaded "module-udev-detect" (index: #5; argument: "").
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.19/modules/module-bluetooth-discover.so': success
D: dbus-util.c: Successfully connected to D-Bus system bus 238df70a29d4b22a98fdfa234b16658a as :1.98
D: bluetooth-util.c: dbus: interface=org.freedesktop.DBus, path=/org/freedesktop/DBus, member=NameAcquired
D: bluetooth-util.c: Bluetooth daemon is apparently not available.
I: module.c: Loaded "module-bluetooth-discover" (index: #6; argument: "").
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.19/modules/module-esound-protocol-unix.so': success
I: module.c: Loaded "module-esound-protocol-unix" (index: #7; argument: "").
I: module.c: Loaded "module-native-protocol-unix" (index: #8; argument: "").
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.19/modules/module-gconf.so': success
I: module.c: Loaded "module-gconf" (index: #9; argument: "").
D: core-subscribe.c: Dropped redundant event due to change event.
I: module-default-device-restore.c: Restored default sink 'alsa_output.pci-0000_00_14.2.analog-stereo'.
D: core-subscribe.c: Dropped redundant event due to change event.
I: module-default-device-restore.c: Restored default source 'alsa_input.pci-0000_00_14.2.analog-stereo'.
I: module.c: Loaded "module-default-device-restore" (index: #10; argument: "").
I: module.c: Loaded "module-rescue-streams" (index: #11; argument: "").
I: module.c: Loaded "module-always-sink" (index: #12; argument: "").
I: module.c: Loaded "module-intended-roles" (index: #13; argument: "").
D: module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
D: module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds.
I: module.c: Loaded "module-suspend-on-idle" (index: #14; argument: "").
I: client.c: Created 0 "ConsoleKit Session /org/freedesktop/ConsoleKit/Session2"
D: module-console-kit.c: Added new session /org/freedesktop/ConsoleKit/Session2
I: module.c: Loaded "module-console-kit" (index: #15; argument: "").
I: module.c: Loaded "module-position-event-sounds" (index: #16; argument: "").
D: main.c: Got org.pulseaudio.Server!
I: main.c: Daemon startup complete.
D: bluetooth-util.c: dbus: interface=org.freedesktop.DBus.Introspectable, path=/, member=Introspect
D: module-console-kit.c: dbus: interface=org.freedesktop.DBus.Introspectable, path=/, member=Introspect
D: module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
D: alsa-source.c: Wakeup from ALSA!
D: alsa-source.c: Wakeup from ALSA!
D: alsa-source.c: Wakeup from ALSA!
D: alsa-source.c: Wakeup from ALSA!
I: client.c: Created 1 "Native client (UNIX socket client)"
I: client.c: Created 2 "Native client (UNIX socket client)"
D: protocol-native.c: Protocol version: remote 16, local 16
I: protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
D: protocol-native.c: SHM possible: yes
D: protocol-native.c: Negotiated SHM: yes
D: protocol-native.c: Protocol version: remote 16, local 16
I: protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
D: protocol-native.c: SHM possible: yes
D: protocol-native.c: Negotiated SHM: yes
D: module-augment-properties.c: Looking for .desktop file for gnome-settings-daemon
D: module-augment-properties.c: Looking for .desktop file for gnome-volume-control-applet
D: alsa-source.c: Wakeup from ALSA!
D: alsa-source.c: Wakeup from ALSA!
D: alsa-source.c: Wakeup from ALSA!
D: alsa-source.c: Wakeup from ALSA!
I: module-suspend-on-idle.c: Source alsa_input.pci-0000_00_14.2.analog-stereo idle for too long, suspending ...
D: source.c: Suspend cause of source alsa_input.pci-0000_00_14.2.analog-stereo is 0x0004, suspending
I: alsa-source.c: Device suspended...
I: module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo idle for too long, suspending ...
D: sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0004, suspending
I: alsa-sink.c: Device suspended...
D: reserve-wrap.c: Device lock status of reserve-monitor-wrapper@Audio0 changed: not busy
D: module-udev-detect.c: /dev/snd/controlC0 is accessible: yes


AND it HANGS HERE.......WAITING for EVER unless I use CNTL C

^C

I: main.c: Got signal SIGINT.
I: main.c: Exiting.
I: main.c: Daemon shutdown initiated.
I: module.c: Unloading "module-device-restore" (index: #0).
I: module.c: Unloaded "module-device-restore" (index: #0).
I: module.c: Unloading "module-stream-restore" (index: #1).
I: module.c: Unloaded "module-stream-restore" (index: #1).
I: module.c: Unloading "module-card-restore" (index: #2).
I: module.c: Unloaded "module-card-restore" (index: #2).
I: module.c: Unloading "module-augment-properties" (index: #3).
I: module.c: Unloaded "module-augment-properties" (index: #3).
I: module.c: Unloading "module-alsa-card" (index: #4).
D: alsa-sink.c: Thread shutting down
I: sink.c: Freeing sink 0 "alsa_output.pci-0000_00_14.2.analog-stereo"
I: source.c: Freeing source 0 "alsa_output.pci-0000_00_14.2.analog-stereo.monitor"
D: core-subscribe.c: Dropped redundant event due to change event.
D: alsa-source.c: Thread shutting down
I: source.c: Freeing source 1 "alsa_input.pci-0000_00_14.2.analog-stereo"
I: card.c: Freed 0 "alsa_card.pci-0000_00_14.2"
I: module.c: Unloaded "module-alsa-card" (index: #4).
I: module.c: Unloading "module-udev-detect" (index: #5).
I: module.c: Unloaded "module-udev-detect" (index: #5).
I: module.c: Unloading "module-bluetooth-discover" (index: #6).
I: module.c: Unloaded "module-bluetooth-discover" (index: #6).
I: module.c: Unloading "module-esound-protocol-unix" (index: #7).
I: module.c: Unloaded "module-esound-protocol-unix" (index: #7).
I: module.c: Unloading "module-native-protocol-unix" (index: #8).
I: client.c: Freed 1 "GNOME Volume Control Applet"
I: client.c: Freed 2 "GNOME Volume Control Media Keys"
I: module.c: Unloaded "module-native-protocol-unix" (index: #8).
I: module.c: Unloading "module-gconf" (index: #9).
I: module.c: Unloaded "module-gconf" (index: #9).
I: module.c: Unloading "module-default-device-restore" (index: #10).
I: module.c: Unloaded "module-default-device-restore" (index: #10).
I: module.c: Unloading "module-rescue-streams" (index: #11).
I: module.c: Unloaded "module-rescue-streams" (index: #11).
I: module.c: Unloading "module-always-sink" (index: #12).
I: module.c: Unloaded "module-always-sink" (index: #12).
I: module.c: Unloading "module-intended-roles" (index: #13).
I: module.c: Unloaded "module-intended-roles" (index: #13).
I: module.c: Unloading "module-suspend-on-idle" (index: #14).
I: module.c: Unloaded "module-suspend-on-idle" (index: #14).
I: module.c: Unloading "module-console-kit" (index: #15).
D: module-console-kit.c: Removing session /org/freedesktop/ConsoleKit/Session2
I: client.c: Freed 0 "ConsoleKit Session /org/freedesktop/ConsoleKit/Session2"
I: module.c: Unloaded "module-console-kit" (index: #15).
I: module.c: Unloading "module-position-event-sounds" (index: #16).
I: module.c: Unloaded "module-position-event-sounds" (index: #16).
I: main.c: Daemon terminated.
rick@rick-laptop:~$


Anything that you could offer to assist me getting the sound functional would be appreciated.

Thanks.

lkraemer

Skerit
December 3rd, 2009, 11:46 PM
How do I define multiple inputs?

I need to add multiple recording devices (in order to use ultrastar-dx at wild, wild parties)

However, using Ubuntu's tools I can only select one or the other.

In ultrastar I *can* select different inputs, but since only one is defined I can't use it.

For example, whi is this not correct?



pcm.singstarmic {
type pulse
device hw:1,0
}

ctl.singstarmic {
type pulse
device hw:1,0
}

fizikz
December 5th, 2009, 05:01 AM
Any audio experts want to take a crack at my 9.10/Rhythmbox problem?

http://ubuntuforums.org/showthread.php?t=1326861

Basically, Rb jumps from song to song very rapidly. No other audio application is being affected and PA doesn't seem to be the culprit (nor does gstreamer for that matter). I'm stumped and really could use a solution.

My guess is that perhaps you are missing some audio codecs. That happened to me with Amarok (after a clean install of 9.10) before the right codecs were installed.

jamesisin
December 5th, 2009, 05:16 AM
My guess is that perhaps you are missing some audio codecs. That happened to me with Amarok (after a clean install of 9.10) before the right codecs were installed.

I don't imagine that is the problem as this would cause other applications to have issues and only Rb while in its main library is seeing this problem. DAAP in Rb is running fine and Movie Player (Totem) is able to play any file with which Rb encounters troubles (and the DAAP share is serving the exact same files). Even Rb can sometimes be coaxed into play the entire file by going back to it--or at least more than it played on its first attempt. Besides, all of the files in question are FLAC files and a missing codec ought to effect all of them equally.

Thanks for your efforts and if I've misunderstood your suggestion please help me understand better.

dasbooter
December 6th, 2009, 03:51 AM
Wow nice ... totally seems to have killed my pulse audio sound server :)

wondering if this screen shot of the term that I usually take before I do something I feel is about to break the system ;) . The next screen shot is the dialog when I try to open system->preferences->sound. Also amarok is asking me nicely if it can remove the pulse audio sound server from the list of outputs since it no longer exists and all system sounds have died.I followed directions from the A part only. Clean install and update of Karmic... I only wanted to try this b/c audio was stuttering in flash and actually this seems to have fixed it but I think only because everthing has fallen back to ALSA. Is it possible to undo this or fix it..?

guarana5
December 6th, 2009, 02:17 PM
I never had troubles with sound until installing Jaunty... Now I'm stuck at Section A part 5. There I get:

maekja@Rattennest:~$ pulseaudio & pavucontrol
I: caps.c: Limited capabilities successfully to CAP_SYS_NICE.
I: caps.c: Dropping root privileges.
I: caps.c: Limited capabilities successfully to CAP_SYS_NICE.
[1] 4618
W: alsa-util.c: Device front:0 doesn't support 44100 Hz, changed to 48000 Hz.
E: main.c: Default sink name (EMU10k1) does not exist in name register.
I: caps.c: Limited capabilities successfully to CAP_SYS_NICE.
I: caps.c: Dropping root privileges.
I: caps.c: Limited capabilities successfully to CAP_SYS_NICE.
[1]+ Exit 1 pulseaudio


I'm running Jaunty Jackalope x86, kernel version 2.6.28-17-generic


maekja@Rattennest:~$ aplay -l
**** Liste von PLAYBACK Geräten ****
I: caps.c: Limited capabilities successfully to CAP_SYS_NICE.
I: caps.c: Dropping root privileges.
I: caps.c: Limited capabilities successfully to CAP_SYS_NICE.
Karte 0: V8237 [VIA 8237], Gerät 0: VIA 8237 [VIA 8237]
Untergeordnete Geräte: 4/4
Untergeordnetes Gerät '0: subdevice #0
Untergeordnetes Gerät '1: subdevice #1
Untergeordnetes Gerät '2: subdevice #2
Untergeordnetes Gerät '3: subdevice #3
Karte 0: V8237 [VIA 8237], Gerät 1: VIA 8237 [VIA 8237]
Untergeordnete Geräte: 1/1
Untergeordnetes Gerät '0: subdevice #0
Karte 1: Live [SB Live 5.1 [SB0220]], Gerät 0: emu10k1 [ADC Capture/Standard PCM Playback]
Untergeordnete Geräte: 32/32
Untergeordnetes Gerät '0: subdevice #0
Untergeordnetes Gerät '1: subdevice #1
Untergeordnetes Gerät '2: subdevice #2
Untergeordnetes Gerät '3: subdevice #3
Untergeordnetes Gerät '4: subdevice #4
Untergeordnetes Gerät '5: subdevice #5
Untergeordnetes Gerät '6: subdevice #6
Untergeordnetes Gerät '7: subdevice #7
Untergeordnetes Gerät '8: subdevice #8
Untergeordnetes Gerät '9: subdevice #9
Untergeordnetes Gerät '10: subdevice #10
Untergeordnetes Gerät '11: subdevice #11
Untergeordnetes Gerät '12: subdevice #12
Untergeordnetes Gerät '13: subdevice #13
Untergeordnetes Gerät '14: subdevice #14
Untergeordnetes Gerät '15: subdevice #15
Untergeordnetes Gerät '16: subdevice #16
Untergeordnetes Gerät '17: subdevice #17
Untergeordnetes Gerät '18: subdevice #18
Untergeordnetes Gerät '19: subdevice #19
Untergeordnetes Gerät '20: subdevice #20
Untergeordnetes Gerät '21: subdevice #21
Untergeordnetes Gerät '22: subdevice #22
Untergeordnetes Gerät '23: subdevice #23
Untergeordnetes Gerät '24: subdevice #24
Untergeordnetes Gerät '25: subdevice #25
Untergeordnetes Gerät '26: subdevice #26
Untergeordnetes Gerät '27: subdevice #27
Untergeordnetes Gerät '28: subdevice #28
Untergeordnetes Gerät '29: subdevice #29
Untergeordnetes Gerät '30: subdevice #30
Untergeordnetes Gerät '31: subdevice #31
Karte 1: Live [SB Live 5.1 [SB0220]], Gerät 2: emu10k1 efx [Multichannel Capture/PT Playback]
Untergeordnete Geräte: 8/8
Untergeordnetes Gerät '0: subdevice #0
Untergeordnetes Gerät '1: subdevice #1
Untergeordnetes Gerät '2: subdevice #2
Untergeordnetes Gerät '3: subdevice #3
Untergeordnetes Gerät '4: subdevice #4
Untergeordnetes Gerät '5: subdevice #5
Untergeordnetes Gerät '6: subdevice #6
Untergeordnetes Gerät '7: subdevice #7
Karte 1: Live [SB Live 5.1 [SB0220]], Gerät 3: emu10k1 [Multichannel Playback]
Untergeordnete Geräte: 1/1
Untergeordnetes Gerät '0: subdevice #0
maekja@Rattennest:~$ pkill pulseaudio; sleep 2; pulseaudio -vv
I: caps.c: Limited capabilities successfully to CAP_SYS_NICE.
I: caps.c: Dropping root privileges.
I: caps.c: Limited capabilities successfully to CAP_SYS_NICE.
D: main.c: Started as real root: no, suid root: yes
I: main.c: We're in the group 'pulse-rt', allowing high-priority scheduling.
I: main.c: setrlimit(RLIMIT_NICE, (31, 31)) fehlgeschlagen: Operation not permitted
I: main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) fehlgeschlagen: Operation not permitted
I: core-util.c: Successfully gained nice level -11.
D: main.c: Can realtime: yes, can high-priority: yes
I: main.c: Giving up CAP_NICE
D: main.c: Can realtime: no, can high-priority: no
I: main.c: Dies ist PulseAudio 0.9.14
D: main.c: Compilation host: i486-pc-linux-gnu
D: main.c: Compilation CFLAGS: -g -O2 -g -Wall -O3 -Wall -W -Wextra -pedantic -pipe -Wno-long-long -Wvla -Wno-overlength-strings -Wconversion -Wundef -Wformat -Wlogical-op -Wpacked -Wformat-security -Wmissing-include-dirs -Wformat-nonliteral -Wold-style-definition -Wdeclaration-after-statement -Wfloat-equal -Wmissing-declarations -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wmissing-noreturn -Wshadow -Wendif-labels -Wpointer-arith -Wcast-align -Wwrite-strings -Wno-unused-parameter -ffast-math
D: main.c: Running on host: Linux i686 2.6.28-17-generic #58-Ubuntu SMP Tue Dec 1 18:57:07 UTC 2009
I: main.c: Seitengröße ist 4096 Bytes.
D: main.c: Compiled with Valgrind support: no
D: main.c: Running in valgrind mode: no
D: main.c: Optimized build: yes
I: main.c: System- ID ist c79938709bfe37a9ec9caa4c4ad910ca.
I: main.c: Using runtime directory /home/maekja/.pulse/c79938709bfe37a9ec9caa4c4ad910ca:runtime.
I: main.c: Using state directory /home/maekja/.pulse.
I: main.c: Running in system mode: no
I: main.c: Fresh high-resolution timers available! Bon appetit!
D: memblock.c: Using shared memory pool with 1024 slots of size 64,0 KiB each, total size is 64,0 MiB, maximum usable slot size is 65496
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9/modules/module-gconf.so': success
I: module.c: Loaded "module-gconf" (index: #0; argument: "").
I: module.c: Loaded "module-suspend-on-idle" (index: #1; argument: "").
I: module-device-restore.c: Sucessfully opened database file '/home/maekja/.pulse/c79938709bfe37a9ec9caa4c4ad910ca:device-volumes.i486-pc-linux-gnu.gdbm'.
I: module.c: Loaded "module-device-restore" (index: #2; argument: "").
I: module-stream-restore.c: Sucessfully opened database file '/home/maekja/.pulse/c79938709bfe37a9ec9caa4c4ad910ca:stream-volumes.i486-pc-linux-gnu.gdbm'.
I: module.c: Loaded "module-stream-restore" (index: #3; argument: "").
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9/modules/module-hal-detect.so': success
I: module-hal-detect.c: Trying capability alsa
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/computer_alsa_timer
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/computer_alsa_sequencer
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1106_3059_sound_card_0_alsa_playback_1
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1106_3059_sound_card_0_alsa_capture_1
D: module-hal-detect.c: Loading module-alsa-sink with arguments 'device_id=0 sink_name=alsa_output.pci_1106_3059_sound_card_0_a lsa_playback_0 tsched=0'
D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
I: module-alsa-sink.c: Successfully opened device front:0.
I: module-alsa-sink.c: Successfully enabled mmap() mode.
I: (alsa-lib)control.c: Invalid CTL front:0
I: alsa-util.c: Unable to attach to mixer front:0: No such file or directory
I: alsa-util.c: Successfully attached to mixer 'hw:0'
I: alsa-util.c: Using mixer control "Master".
I: sink.c: Created sink 0 "alsa_output.pci_1106_3059_sound_card_0_alsa_playba ck_0" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: source.c: Created source 0 "alsa_output.pci_1106_3059_sound_card_0_alsa_playba ck_0.monitor" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: module-alsa-sink.c: Using 8 fragments of size 1764 bytes, buffer time is 80,00ms
D: module-alsa-sink.c: hwbuf_unused=0
D: module-alsa-sink.c: setting avail_min=1
I: module-alsa-sink.c: Volume ranges from 0 to 31.
I: module-alsa-sink.c: Volume ranges from -46,50 dB to 0,00 dB.
I: alsa-util.c: All 2 channels can be mapped to mixer channels.
I: module-alsa-sink.c: Using hardware volume control. Hardware dB scale supported.
D: alsa-util.c: snd_pcm_dump():
D: alsa-util.c: Hardware PCM card 0 'VIA 8237' device 0 subdevice 0
D: alsa-util.c: Its setup is:
D: alsa-util.c: stream : PLAYBACK
D: alsa-util.c: access : MMAP_INTERLEAVED
D: alsa-util.c: format : S16_LE
D: alsa-util.c: subformat : STD
D: alsa-util.c: channels : 2
D: alsa-util.c: rate : 44100
D: alsa-util.c: exact rate : 44100 (44100/1)
D: alsa-util.c: msbits : 16
D: alsa-util.c: buffer_size : 3528
D: alsa-util.c: period_size : 441
D: alsa-util.c: period_time : 10000
D: alsa-util.c: tstamp_mode : ENABLE
D: alsa-util.c: period_step : 1
D: alsa-util.c: avail_min : 441
D: alsa-util.c: period_event : 0
D: alsa-util.c: start_threshold : -1
D: alsa-util.c: stop_threshold : 1849688064
D: alsa-util.c: silence_threshold: 0
D: alsa-util.c: silence_size : 0
D: alsa-util.c: boundary : 1849688064
D: module-alsa-sink.c: Thread starting up
D: rtpoll.c: Acquired POSIX realtime signal SIGRTMIN+29
D: module-alsa-sink.c: Read hardware volume: 0: 87% 1: 87%
I: module-alsa-sink.c: Starting playback.
D: module-suspend-on-idle.c: Source alsa_output.pci_1106_3059_sound_card_0_alsa_playba ck_0.monitor becomes idle.
D: module-suspend-on-idle.c: Sink alsa_output.pci_1106_3059_sound_card_0_alsa_playba ck_0 becomes idle.
I: module.c: Loaded "module-alsa-sink" (index: #4; argument: "device_id=0 sink_name=alsa_output.pci_1106_3059_sound_card_0_a lsa_playback_0 tsched=0").
D: module-hal-detect.c: Loading module-alsa-source with arguments 'device_id=0 source_name=alsa_input.pci_1106_3059_sound_card_0_ alsa_capture_0 tsched=0'
D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
W: alsa-util.c: Device front:0 doesn't support 44100 Hz, changed to 48000 Hz.
I: module-alsa-source.c: Successfully opened device front:0.
I: module-alsa-source.c: Successfully enabled mmap() mode.
I: (alsa-lib)control.c: Invalid CTL front:0
I: alsa-util.c: Unable to attach to mixer front:0: No such file or directory
I: alsa-util.c: Successfully attached to mixer 'hw:0'
I: alsa-util.c: Using mixer control "Capture".
I: source.c: Created source 1 "alsa_input.pci_1106_3059_sound_card_0_alsa_capture _0" with sample spec s16le 2ch 48000Hz and channel map front-left,front-right
I: module-alsa-source.c: Using 8 fragments of size 1920 bytes, buffer time is 80,00ms
D: module-alsa-source.c: hwbuf_unused=0
D: module-alsa-source.c: setting avail_min=1
I: module-alsa-source.c: Volume ranges from 0 to 15.
I: module-alsa-source.c: Volume ranges from 0,00 dB to 22,50 dB.
I: alsa-util.c: All 2 channels can be mapped to mixer channels.
I: module-alsa-source.c: Using hardware volume control. Hardware dB scale supported.
D: alsa-util.c: snd_pcm_dump():
D: alsa-util.c: Hardware PCM card 0 'VIA 8237' device 0 subdevice 0
D: alsa-util.c: Its setup is:
D: alsa-util.c: stream : CAPTURE
D: alsa-util.c: access : MMAP_INTERLEAVED
D: alsa-util.c: format : S16_LE
D: alsa-util.c: subformat : STD
D: alsa-util.c: channels : 2
D: alsa-util.c: rate : 48000
D: alsa-util.c: exact rate : 48000 (48000/1)
D: alsa-util.c: msbits : 16
D: alsa-util.c: buffer_size : 3840
D: alsa-util.c: period_size : 480
D: alsa-util.c: period_time : 10000
D: alsa-util.c: tstamp_mode : ENABLE
D: alsa-util.c: period_step : 1
D: alsa-util.c: avail_min : 480
D: alsa-util.c: period_event : 0
D: alsa-util.c: start_threshold : -1
D: alsa-util.c: stop_threshold : 2013265920
D: alsa-util.c: silence_threshold: 0
D: alsa-util.c: silence_size : 0
D: alsa-util.c: boundary : 2013265920
D: module-alsa-source.c: Thread starting up
D: rtpoll.c: Acquired POSIX realtime signal SIGRTMIN+28
D: module-alsa-source.c: Read hardware volume: 0: 62% 1: 62%
D: module-suspend-on-idle.c: Source alsa_input.pci_1106_3059_sound_card_0_alsa_capture _0 becomes idle.
I: module.c: Loaded "module-alsa-source" (index: #5; argument: "device_id=0 source_name=alsa_input.pci_1106_3059_sound_card_0_ alsa_capture_0 tsched=0").
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1106_3059_sound_card_0_alsa_control__1
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_2_sound_card_0_alsa_playback_3
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_2_sound_card_0_alsa_playback_2
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_2_sound_card_0_alsa_capture_2
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_2_sound_card_0_alsa_capture_1
D: module-hal-detect.c: Loading module-alsa-sink with arguments 'device_id=1 sink_name=alsa_output.pci_1102_2_sound_card_0_alsa _playback_0 tsched=0'
D: alsa-util.c: Trying front:1 with SND_PCM_NO_AUTO_FORMAT ...
I: module-alsa-sink.c: Successfully opened device front:1.
I: module-alsa-sink.c: Successfully enabled mmap() mode.
I: (alsa-lib)control.c: Invalid CTL front:1
I: alsa-util.c: Unable to attach to mixer front:1: No such file or directory
I: alsa-util.c: Successfully attached to mixer 'hw:1'
I: alsa-util.c: Using mixer control "Master".
I: sink.c: Created sink 1 "alsa_output.pci_1102_2_sound_card_0_alsa_playback_ 0" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: source.c: Created source 2 "alsa_output.pci_1102_2_sound_card_0_alsa_playback_ 0.monitor" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: module-alsa-sink.c: Using 8 fragments of size 1764 bytes, buffer time is 80,00ms
D: module-alsa-sink.c: hwbuf_unused=0
D: module-alsa-sink.c: setting avail_min=1
I: module-alsa-sink.c: Volume ranges from 0 to 31.
I: module-alsa-sink.c: Volume ranges from -46,50 dB to 0,00 dB.
I: alsa-util.c: All 2 channels can be mapped to mixer channels.
I: module-alsa-sink.c: Using hardware volume control. Hardware dB scale supported.
D: alsa-util.c: snd_pcm_dump():
D: alsa-util.c: Hooks PCM
D: alsa-util.c: Its setup is:
D: alsa-util.c: stream : PLAYBACK
D: alsa-util.c: access : MMAP_INTERLEAVED
D: alsa-util.c: format : S16_LE
D: alsa-util.c: subformat : STD
D: alsa-util.c: channels : 2
D: alsa-util.c: rate : 44100
D: alsa-util.c: exact rate : 44100 (44100/1)
D: alsa-util.c: msbits : 16
D: alsa-util.c: buffer_size : 3528
D: alsa-util.c: period_size : 441
D: alsa-util.c: period_time : 10000
D: alsa-util.c: tstamp_mode : ENABLE
D: alsa-util.c: period_step : 1
D: alsa-util.c: avail_min : 441
D: alsa-util.c: period_event : 0
D: alsa-util.c: start_threshold : -1
D: alsa-util.c: stop_threshold : 1849688064
D: alsa-util.c: silence_threshold: 0
D: alsa-util.c: silence_size : 0
D: alsa-util.c: boundary : 1849688064
D: alsa-util.c: Slave: Hardware PCM card 1 'SB Live 5.1 [SB0220]' device 0 subdevice 0
D: alsa-util.c: Its setup is:
D: alsa-util.c: stream : PLAYBACK
D: alsa-util.c: access : MMAP_INTERLEAVED
D: alsa-util.c: format : S16_LE
D: alsa-util.c: subformat : STD
D: alsa-util.c: channels : 2
D: alsa-util.c: rate : 44100
D: alsa-util.c: exact rate : 44100 (44100/1)
D: alsa-util.c: msbits : 16
D: alsa-util.c: buffer_size : 3528
D: alsa-util.c: period_size : 441
D: alsa-util.c: period_time : 10000
D: alsa-util.c: tstamp_mode : ENABLE
D: alsa-util.c: period_step : 1
D: alsa-util.c: avail_min : 441
D: alsa-util.c: period_event : 0
D: alsa-util.c: start_threshold : -1
D: alsa-util.c: stop_threshold : 1849688064
D: alsa-util.c: silence_threshol
D: module-alsa-sink.c: Thread starting up
D: rtpoll.c: Acquired POSIX realtime signal SIGRTMIN+27
D: module-alsa-sink.c: Read hardware volume: 0: 87% 1: 87%
I: module-alsa-sink.c: Starting playback.
D: module-suspend-on-idle.c: Source alsa_output.pci_1102_2_sound_card_0_alsa_playback_ 0.monitor becomes idle.
D: module-suspend-on-idle.c: Sink alsa_output.pci_1102_2_sound_card_0_alsa_playback_ 0 becomes idle.
I: module.c: Loaded "module-alsa-sink" (index: #6; argument: "device_id=1 sink_name=alsa_output.pci_1102_2_sound_card_0_alsa _playback_0 tsched=0").
D: module-hal-detect.c: Loading module-alsa-source with arguments 'device_id=1 source_name=alsa_input.pci_1102_2_sound_card_0_als a_capture_0 tsched=0'
D: alsa-util.c: Trying front:1 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)setup.c: Cannot lock ctl elem
D: alsa-util.c: Trying front:1 without SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)setup.c: Cannot lock ctl elem
D: alsa-util.c: Trying plug:front:1 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Trying plug:front:1 without SND_PCM_NO_AUTO_FORMAT ...
I: alsa-util.c: PCM device plug:front:1 refused our hw parameters: Invalid argument
D: alsa-util.c: Trying surround40:1 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm_hw.c: open /dev/snd/pcmC1D0c failed
I: alsa-util.c: Couldn't open PCM device surround40:1: Device or resource busy
D: alsa-util.c: Trying surround41:1 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm_hw.c: open /dev/snd/pcmC1D0c failed
I: alsa-util.c: Couldn't open PCM device surround41:1: Device or resource busy
D: alsa-util.c: Trying surround50:1 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm_hw.c: open /dev/snd/pcmC1D0c failed
I: alsa-util.c: Couldn't open PCM device surround50:1: Device or resource busy
D: alsa-util.c: Trying surround51:1 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm_hw.c: open /dev/snd/pcmC1D0c failed
I: alsa-util.c: Couldn't open PCM device surround51:1: Device or resource busy
D: alsa-util.c: Trying surround71:1 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 1
I: (alsa-lib)pcm.c: Unknown PCM surround71:1
I: alsa-util.c: Couldn't open PCM device surround71:1: Invalid argument
D: alsa-util.c: Trying hw:1 as last resort...
D: alsa-util.c: Trying hw:1 with SND_PCM_NO_AUTO_FORMAT ...
I: module-alsa-source.c: Successfully opened device hw:1.
I: module-alsa-source.c: Successfully enabled mmap() mode.
I: alsa-util.c: Successfully attached to mixer 'hw:1'
I: alsa-util.c: Using mixer control "Capture".
I: source.c: Created source 3 "alsa_input.pci_1102_2_sound_card_0_alsa_capture_0" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: module-alsa-source.c: Using 2 fragments of size 7168 bytes, buffer time is 81,27ms
D: module-alsa-source.c: hwbuf_unused=0
D: module-alsa-source.c: setting avail_min=1
I: module-alsa-source.c: Volume ranges from 0 to 15.
I: module-alsa-source.c: Volume ranges from 0,00 dB to 22,50 dB.
I: alsa-util.c: All 2 channels can be mapped to mixer channels.
I: module-alsa-source.c: Using hardware volume control. Hardware dB scale supported.
D: alsa-util.c: snd_pcm_dump():
D: alsa-util.c: Hardware PCM card 1 'SB Live 5.1 [SB0220]' device 0 subdevice 0
D: alsa-util.c: Its setup is:
D: alsa-util.c: stream : CAPTURE
D: alsa-util.c: access : MMAP_INTERLEAVED
D: alsa-util.c: format : S16_LE
D: alsa-util.c: subformat : STD
D: alsa-util.c: channels : 2
D: alsa-util.c: rate : 44100
D: alsa-util.c: exact rate : 44100 (44100/1)
D: alsa-util.c: msbits : 16
D: alsa-util.c: buffer_size : 3584
D: alsa-util.c: period_size : 1792
D: alsa-util.c: period_time : 40634
D: alsa-util.c: tstamp_mode : ENABLE
D: alsa-util.c: period_step : 1
D: alsa-util.c: avail_min : 1792
D: alsa-util.c: period_event : 0
D: alsa-util.c: start_threshold : -1
D: alsa-util.c: stop_threshold : 1879048192
D: alsa-util.c: silence_threshold: 0
D: alsa-util.c: silence_size : 0
D: alsa-util.c: boundary : 1879048192
D: module-alsa-source.c: Thread starting up
D: rtpoll.c: Acquired POSIX realtime signal SIGRTMIN+26
D: module-alsa-source.c: Read hardware volume: 0: 75% 1: 75%
D: module-suspend-on-idle.c: Source alsa_input.pci_1102_2_sound_card_0_alsa_capture_0 becomes idle.
I: module.c: Loaded "module-alsa-source" (index: #7; argument: "device_id=1 source_name=alsa_input.pci_1102_2_sound_card_0_als a_capture_0 tsched=0").
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_2_sound_card_0_alsa_midi_2
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_2_sound_card_0_alsa_midi_1
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_2_sound_card_0_alsa_midi_0
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_2_sound_card_0_alsa_hw_specific_2
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_2_sound_card_0_alsa_hw_specific_0
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_2_sound_card_0_alsa_control__1
I: module-hal-detect.c: Loaded 4 modules.
I: module.c: Loaded "module-hal-detect" (index: #8; argument: "tsched=0").
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9/modules/module-esound-protocol-unix.so': success
I: module.c: Loaded "module-esound-protocol-unix" (index: #9; argument: "").
I: module.c: Loaded "module-native-protocol-unix" (index: #10; argument: "").
I: module.c: Loaded "module-default-device-restore" (index: #11; argument: "").
I: module.c: Loaded "module-rescue-streams" (index: #12; argument: "").
I: module.c: Loaded "module-always-sink" (index: #13; argument: "").
I: client.c: Created 0 "ConsoleKit Session /org/freedesktop/ConsoleKit/Session3"
D: module-console-kit.c: Added new session /org/freedesktop/ConsoleKit/Session3
I: module.c: Loaded "module-console-kit" (index: #14; argument: "").
I: module.c: Loaded "module-position-event-sounds" (index: #15; argument: "").
E: main.c: Default sink name (EMU10k1) does not exist in name register.
I: module.c: Unloading "module-gconf" (index: #0).
I: module.c: Unloaded "module-gconf" (index: #0).
I: module.c: Unloading "module-suspend-on-idle" (index: #1).
I: module.c: Unloaded "module-suspend-on-idle" (index: #1).
I: module.c: Unloading "module-device-restore" (index: #2).
I: module.c: Unloaded "module-device-restore" (index: #2).
D: core-subscribe.c: Dropped redundant event due to remove event.
I: module.c: Unloading "module-stream-restore" (index: #3).
I: module.c: Unloaded "module-stream-restore" (index: #3).
D: core-subscribe.c: Dropped redundant event due to remove event.
I: module.c: Unloading "module-alsa-sink" (index: #4).
D: module-rescue-streams.c: No sink inputs to move away.
D: module-rescue-streams.c: No source outputs to move away.
D: core-subscribe.c: Dropped redundant event due to remove event.
D: core-subscribe.c: Dropped redundant event due to remove event.
D: module-alsa-sink.c: Thread shutting down
I: sink.c: Freeing sink 0 "alsa_output.pci_1106_3059_sound_card_0_alsa_playba ck_0"
I: source.c: Freeing source 0 "alsa_output.pci_1106_3059_sound_card_0_alsa_playba ck_0.monitor"
I: module.c: Unloaded "module-alsa-sink" (index: #4).
D: core-subscribe.c: Dropped redundant event due to remove event.
I: module.c: Unloading "module-alsa-source" (index: #5).
D: module-rescue-streams.c: No source outputs to move away.
D: core-subscribe.c: Dropped redundant event due to remove event.
D: module-alsa-source.c: Thread shutting down
I: source.c: Freeing source 1 "alsa_input.pci_1106_3059_sound_card_0_alsa_capture _0"
I: module.c: Unloaded "module-alsa-source" (index: #5).
D: core-subscribe.c: Dropped redundant event due to remove event.
I: module.c: Unloading "module-alsa-sink" (index: #6).
D: module-always-sink.c: Autoloading null-sink as no other sinks detected.
I: sink.c: Created sink 2 "auto_null" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: source.c: Created source 4 "auto_null.monitor" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
D: module-null-sink.c: Thread starting up
D: rtpoll.c: Acquired POSIX realtime signal SIGRTMIN+29
I: module.c: Loaded "module-null-sink" (index: #16; argument: "sink_name=auto_null").
D: module-rescue-streams.c: No sink inputs to move away.
D: module-rescue-streams.c: No source outputs to move away.
D: core-subscribe.c: Dropped redundant event due to remove event.
D: core-subscribe.c: Dropped redundant event due to remove event.
D: module-alsa-sink.c: Thread shutting down
I: sink.c: Freeing sink 1 "alsa_output.pci_1102_2_sound_card_0_alsa_playback_ 0"
I: source.c: Freeing source 2 "alsa_output.pci_1102_2_sound_card_0_alsa_playback_ 0.monitor"
I: module.c: Unloaded "module-alsa-sink" (index: #6).
D: core-subscribe.c: Dropped redundant event due to remove event.
I: module.c: Unloading "module-alsa-source" (index: #7).
D: module-rescue-streams.c: No source outputs to move away.
D: core-subscribe.c: Dropped redundant event due to remove event.
D: module-alsa-source.c: Thread shutting down
I: source.c: Freeing source 3 "alsa_input.pci_1102_2_sound_card_0_alsa_capture_0"
I: module.c: Unloaded "module-alsa-source" (index: #7).
D: core-subscribe.c: Dropped redundant event due to remove event.
I: module.c: Unloading "module-hal-detect" (index: #8).
I: module.c: Unloaded "module-hal-detect" (index: #8).
D: core-subscribe.c: Dropped redundant event due to remove event.
I: module.c: Unloading "module-esound-protocol-unix" (index: #9).
I: module.c: Unloaded "module-esound-protocol-unix" (index: #9).
D: core-subscribe.c: Dropped redundant event due to remove event.
I: module.c: Unloading "module-native-protocol-unix" (index: #10).
I: module.c: Unloaded "module-native-protocol-unix" (index: #10).
D: core-subscribe.c: Dropped redundant event due to remove event.
I: module.c: Unloading "module-default-device-restore" (index: #11).
I: module.c: Unloaded "module-default-device-restore" (index: #11).
D: core-subscribe.c: Dropped redundant event due to remove event.
I: module.c: Unloading "module-rescue-streams" (index: #12).
I: module.c: Unloaded "module-rescue-streams" (index: #12).
D: core-subscribe.c: Dropped redundant event due to remove event.
I: module.c: Unloading "module-always-sink" (index: #13).
I: module.c: Unloaded "module-always-sink" (index: #13).
D: core-subscribe.c: Dropped redundant event due to remove event.
I: module.c: Unloading "module-console-kit" (index: #14).
D: module-console-kit.c: Removing session /org/freedesktop/ConsoleKit/Session3
I: client.c: Freed 0 "ConsoleKit Session /org/freedesktop/ConsoleKit/Session3"
D: core-subscribe.c: Dropped redundant event due to remove event.
I: module.c: Unloaded "module-console-kit" (index: #14).
D: core-subscribe.c: Dropped redundant event due to remove event.
I: module.c: Unloading "module-position-event-sounds" (index: #15).
I: module.c: Unloaded "module-position-event-sounds" (index: #15).
D: core-subscribe.c: Dropped redundant event due to remove event.
I: module.c: Unloading "module-null-sink" (index: #16).
D: core-subscribe.c: Dropped redundant event due to remove event.
D: core-subscribe.c: Dropped redundant event due to remove event.
D: module-null-sink.c: Thread shutting down
I: sink.c: Freeing sink 2 "auto_null"
I: source.c: Freeing source 4 "auto_null.monitor"
I: module.c: Unloaded "module-null-sink" (index: #16).
D: core-subscribe.c: Dropped redundant event due to remove event.
I: main.c: Dämon beendet.

I would really appreciate your help!

knavarathna92
December 6th, 2009, 08:57 PM
Hey so I upgraded from Jaunty to Karmic today because I thought the bugs would have been patched by now, but after I upgraded, I had no sound.

In a nutshell, here's what happens: I log in, I hear the log in sound perfectly. I play any video, the sound plays nicely in the beginning but maybe 10 seconds after goes down to very low levels until it's essentially muted. I followed the tutorial and looking at volume control, nothing's muted and the bars for each program on the playback tab are moving. My only input and output devices are internal analog stereo (don't know what this means, I don't have speakers but I use headphones). Alsamixer shows PCM is at 100% but there is no option to mute/unmute shown (maybe it's muted? Can't tell)

I'm pretty confused and have extremely limited computer skills so any help would be appreciated.

Edit: Funny thing is, a couple of moments later after I stop playing sound, it comes back only to die down a couple of seconds later (this process repeats itself).

IMPORTANT : Pulse Audio device chooser won't run (in the teminal,I run the command and nothing shows up)

MSdefecter
December 6th, 2009, 10:55 PM
Hi,
I'm sorry to hijack this one, but my problems came when I upgraded to 9.10. I now have no sound at all. The trouble is I have a sound card on my graphics card and pulse seems to use that instead of my actual Sound card which is a ICE1712 based card. Pulse only see's it as a input and not an output source but I need it to use this card for both. How can I add my sound card to pulse? Please see the output for $ aplay -l $

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

card 0: HDMI [HDA ATI HDMI], device 3: ATI HDMI [ATI HDMI]

Subdevices: 1/1

Subdevice #0: subdevice #0

card 1: M44 [M Audio Delta 44], device 0: ICE1712 multi [ICE1712 multi]

Subdevices: 1/1

Subdevice #0: subdevice #0

ktronicon5
December 9th, 2009, 05:11 AM
I am also using an ICE1712 card and having the same problems since I installed karmic. Volume control seems to work but is not routing audio properly. It lists a "dummy output" but i have no sound in any application unless i'm running the Jack server. Is there a fix for this, or else a way to rout everything through the jack server? (firefox and rhythmbox etc)?

The card is a multi-in/out Hoontech DSP24



$ pulseaudio
E: pid.c: Daemon already running.
E: main.c: pa_pid_file_create() failed.


no idea what that means; pretty new to all this.

Tedward
December 9th, 2009, 11:24 PM
After doing part A and setting up the equalizer, I got the equalizer to come up and act like it was working, but it didn't effect my sound at all. I had to go to sleep, and decided to finish getting it to work today.

I got home today and when I turned my computer on there was no sound. If I go into the pulseaudio volume control it comes up with the error E that you labeled, the daemon isn't running.

When I try to run pulseaudio this is what comes up:
tedward@Tedwardbuntu:~$ pulseaudio
E: socket-server.c: bind(): Address already in use
E: module.c: Failed to load module "module-esound-protocol-unix" (argument: ""): initialization failed.
E: main.c: Module load failed.
E: main.c: Failed to initialize daemon.

I don't quite know how to address this problem. I am running Karmic 64 bit

AgenT
December 17th, 2009, 01:11 PM
I had a problem where Flash would cause 95-100% CPU usage in Firefox and other browsers. The movie would first stop having sound then would freeze altogether. Closing the tab where the flash movie was playing would not help. Only restarting the browser helped.

The solution?

I had timidity installed, from before my upgrade to 9.10. Timidity steals the soundcard which produces problems. After removing timidity, the problem went away. Now I get 30-40% CPU and after the movie finishes playing, it drops to CPU usage before playing the movie!

For those having any PulseAudio problems, READ OVER THE KARMIC CAVEATS (https://wiki.ubuntu.com/DebuggingSoundProblems/KarmicCaveats) WIKI!

waveman77
December 18th, 2009, 03:23 AM
"I get a lot of flak from Ubuntu users, and I am pretty sure the vast
amount of it is undeserving and not my fault."

Even with pages and pages of manual fixes, there are still problems.
pulseaudio was advertised as plug-compatible with ALSA but it is
anything but.

Why the Ubuntu team persisted with this garbage setup over two releases
is incomprehensible. Why did they go with a radical change, in an LTS
release but not test it until too late. Then they persisted in the next
release. Even with the pages of manual config changes, there are still
numerous unresolved problems.

This to fix - what? Sound worked for me before 8.04..

Why is the registration process so cumbersome? You have to register
twice for some reason, and the Ubuntu confirmation email took about 10
minutes to arrive, contrary to the false claim that it "has been sent".

Djzn.BR
December 18th, 2009, 01:23 PM
The equalizer is working on my system. Pulseaudio is working. However, I have to kill it before using Skype. Hell on earth emerges if I try to use Skype with Pulseaudio... it won't work properly, voice calls get echos, stuttering, lagging and loops. For everything serious, I'd use alsa.

But one question I have... no one could implement this mbeq LADSPA plugin directly with alsa? One could write another tutorial to remove pulseaudio and use EQ with alsa.

mulluysavage
December 20th, 2009, 05:08 PM
Suddenly my Songbird is returning the error:

mediacore error: autoaudiosink and alsasink elements are missing

How can I repair this? I am running Hardy - I started down the road of part A, but got the error

cannot stat /.pulse no such file or directory
cannot stat /.asound* no such file or directory
cannot stat /etc/asound.conf no such file or directory

any specific repairs for the mediacore error given?

in fact i cannot find an asound.conf file using a catfish search of the file system. I find an asoundconf bin that runs though.

Baasha
December 21st, 2009, 02:51 AM
I am running Karmic 9.10 64 bit and my sound is inoperable. When I boot up I have no sound. If I go to sound preferences I see I have a dummy output. I can fix that with a sudo modprobe snd-hda-intel (my builtin card on the mb) and then I get a stereo output. However this does not survive the next boot-up even if I do a sudo alsactl store.

I am completely stuck with this and suspect this is what is preventing me from setting up pulseaudio. Eventually, I want to set up 5.1 sound.

Any help would be appreciated.

tjk
December 28th, 2009, 07:25 AM
I am running Karmic 9.10 64 bit and my sound is inoperable. When I boot up I have no sound. If I go to sound preferences I see I have a dummy output.

I am running exactly the same as above, and as well only have a dummy output... I've no sound now since I upgraded to 9.10 (abt a month). I've tried almost all troubleshooting ideas that I could find in this forum. This problem is VERY annoying! Please give some good advice... is there a conflict? Why will AudioPulse-Alsa not recognize my sound devices.

biji
January 1st, 2010, 07:00 AM
thanks.. nice tutorial

Ian Clark
January 2nd, 2010, 11:15 AM
I can't get sound to come out of Kdenlive 0.7.6 on Kubuntu 9.10 Karmic.

1. My system-wide sound is OK, everything is still set to pulsaudio, but after doing part A of this tut I find that playing an AVI video has cracks in the audio, though plain mp3 files play fine.
2. Before doing part A, Kdenlive gave me an SDL error message. It didn't crash the program, and video has always played fine through Kdenlive, no matter what audio settings I choose within the program (I'm now set to pulse)
3. After following this tut, Kdenlive doesn't give me the error, but it also doesn't produce sound.
4. So far this means that after upgrading to Karmic, my video editing has been borked, and though Kdenlive functions wonderfully otherwise, without sound, I cannot do videos.

My system hardware (as read by the applet):
CPU: Intel(R) Pentium(R) 4 CPU 2.80GHz
GPU: ATI Technologies Inc RV350 AS [Radeon 9550]
音效: saa7130[0] ()
音效: Intel ICH5 with ALC202 (Intel ICH5)
音效: saa7130[0] (SAA7134 PCM)
音效: Intel ICH5 with ALC202 (Intel ICH5 - IEC958)
音效: Intel ICH5 with ALC202 (Intel ICH5 - MIC ADC)
音效: Intel ICH5 with ALC202 (Intel ICH5 - MIC2 ADC)
音效: Intel ICH5 with ALC202 (Intel ICH5 - ADC2)
音效: Intel ICH5 with ALC202 ()
網路: Loopback device Interface
網路: Networking Interface

Aplay:
aplay: device_list:223: no soundcards found...

Alsa output file is attached due to length.

Thanks to anyone who might check this out and provide some help! A bug report (https://bugs.launchpad.net/ubuntu/+source/libsdl1.2/+bug/491335) has been filed, but I'm not 100% this applies to my situation.

Scott O'Nanski
January 4th, 2010, 11:02 PM
This broken my sound. Ubuntu 9.10.

Thanks. Great post.

pstickar
January 5th, 2010, 10:24 PM
Removed.

suzenon
January 7th, 2010, 12:46 AM
Awesome topic. I think because of some tips in it my pulseaudio is finally working as it should work (or at least close to).

I see so much replies in this topic, it's probably flooded with questions, but i have one too :) Maybe someone will answer.
How can i turn off pulseaudio without removing it ? I'd like get back to using ALSA instead of pulseaudio but i don't know how to achieve it without removing pulseaudio. (doh - newb) Is there any magic switch?

And to everyone having sound issues with pa, have you tried removing it and using ALSA instead? It was working like a charm when i had issues and get rid of PA. :)
yeah i know its not much of a solution.

treehouse
January 7th, 2010, 07:25 PM
I'm on Karmic Koala and here's my issue, in hopes someone can at the very least point me towards a solution: when I have a program running that uses constant sound (for instance, an mp3), it'll run fine and then seemingly at random, it starts skipping and popping, and my cpu usage shoots up to 100%, where Pulseaudio is using every resource that wasn't already taken. This usueally happens within less than two minutes of opening any music or video file (and with videos the huge resource usage screws up the video).

I've gone through part a of the tutorial, and it made no change. When I play an mp3, it show up in Pavucontrol normally and plays fine (when it skips it just skips, but everything else seems fine.)

If you require assistance with a particular application - or simply cannot get PulseAudio to work - provide the following information:

1. Karmic Koala i386

2.


$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: CMI8738 [C-Media CMI8738], device 0: CMI8738 [C-Media PCI DAC/ADC]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: CMI8738 [C-Media CMI8738], device 1: CMI8738 [C-Media PCI 2nd DAC]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: CMI8738 [C-Media CMI8738], device 2: CMI8738 [C-Media PCI IEC958]
Subdevices: 1/1
Subdevice #0: subdevice #0


3. The verbose output from pulseaudio on your system: Right here on pastebin cuz it's long and I am worried about editting some of the lines out (http://pastebin.com/me008da6)


One of the things I'm noticing is the block that shows up when it skips.


D: protocol-native.c: Requesting rewind due to end of underrun.
D: alsa-sink.c: Requested to rewind 65536 bytes.
D: alsa-sink.c: Mhmm, actually there is nothing to rewind.
D: protocol-native.c: Underrun on 'My MPD PulseAudio Output', 0 bytes in queue.

I'm at a loss as to how to fix this rewind/underrun thing.

woodmaster
January 8th, 2010, 04:42 PM
Originally posted by: nullrend-
It worked great for me. No more suppressed audio events X 100 in my log files and sound works great.

Here is what I did to REMOVE pulse audio in Karmic:

Code:
$ sudo apt-get purge pulseaudio gstreamer0.10-pulseaudio
Code:
$ sudo apt-get autoremove
Code:
$ sudo apt-get install alsa-base alsa-tools alsa-tools-gui alsa-utils alsa-oss linux-sound-base alsamixergui
Code:
$ sudo apt-get install esound esound-clients esound-common libesd-alsa0 gnome-alsamixer
restart your computer!

Notes:
-run gstreamer-properties in terminal to set defaults to alsa (the old system/preferences/sound in jaunty)
This means:
Code:
$ gstreamer-properties
-remove gstreamer0.10-pulseaudio to get sound in totem.
The above commands should have taken care of that, but it never hurts to be sure when talking about the hydra that is PulseAudio.
-gnome-alsamixer is for changing the volume, not an applet but better that nothing
This means you are left without a panel applet to control volume across the board, thanks to the thoughtful efforts of Canonical to shove PulseAudio down our throats. So you need to have something to control audio with:
Code:
$ sudo apt-get install gnome-alsamixer
And something to keep gnome-alsamixer within easy reach, provided by AllTray:
Code:
$ sudo apt-get install alltray
Now, one last word of advice. Usually the best solution is to use aptitude or synaptic to do these moves, but you'll run headfirst into their dependency resolution efforts, which command them to put PulseAudio back in your system when reinstalling gnome-alsamixer. So either you use apt-get or you reconfigure aptitude/synaptic dependency resolution to work the way you want.

baserunner_ams
January 10th, 2010, 01:48 PM
hello i tried this tutorial, because i had issues with TeamSpeak.
I could hear ppl in the room, but my mic was not understandable (crackling as if i had wrong codecs).
Befor ei found this thread i tried to install all codecs i could find in synaptic without success.

If you require assistance with a particular application - or simply cannot get PulseAudio to work - provide the following information:

1. Your distribution version and architecture (e.g. Hardy Heron i386, Intrepid Ibex amd64, etc.).
Ubuntu AMD-64 Release 9.04 (jaunty), Kernel Linux 2.6.28-17-generic, GNOME 2.26.1

2. A listing of your sound devices:
Code:

$ aplay -l

matt@matt64ubuntu:~$ aplay -l
**** List of PLAYBACK Hardware Devices ****
I: caps.c: Limited capabilities successfully to CAP_SYS_NICE.
I: caps.c: Dropping root privileges.
I: caps.c: Limited capabilities successfully to CAP_SYS_NICE.
card 0: Intel [HDA Intel], device 0: ALC1200 Analog [ALC1200 Analog]
Subdevices: 0/1
Subdevice #0: subdevice #0
card 0: Intel [HDA Intel], device 1: ALC1200 Digital [ALC1200 Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0
matt@matt64ubuntu:~$


3. The verbose output from pulseaudio on your system:
Code:

$ pkill pulseaudio; sleep 2; pulseaudio -vv


matt@matt64ubuntu:~$ pkill pulseaudio; sleep 2; pulseaudio -vv
I: caps.c: Limited capabilities successfully to CAP_SYS_NICE.
I: caps.c: Dropping root privileges.
I: caps.c: Limited capabilities successfully to CAP_SYS_NICE.
D: main.c: Started as real root: no, suid root: yes
I: main.c: PolicyKit refuses acquire-high-priority privilege.
N: main.c: Called SUID root and real-time and/or high-priority scheduling was requested in the configuration. However, we lack the necessary privileges:
N: main.c: We are not in group 'pulse-rt', PolicyKit refuse to grant us the requested privileges and we have no increase RLIMIT_NICE/RLIMIT_RTPRIO resource limits.
N: main.c: For enabling real-time/high-priority scheduling please acquire the appropriate PolicyKit privileges, or become a member of 'pulse-rt', or increase the RLIMIT_NICE/RLIMIT_RTPRIO resource limits for this user.
I: main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted
I: main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation not permitted
D: main.c: Can realtime: no, can high-priority: no
D: main.c: Can realtime: no, can high-priority: no
I: main.c: This is PulseAudio 0.9.14
D: main.c: Compilation host: x86_64-pc-linux-gnu
D: main.c: Compilation CFLAGS: -g -O2 -g -Wall -O3 -Wall -W -Wextra -pedantic -pipe -Wno-long-long -Wvla -Wno-overlength-strings -Wconversion -Wundef -Wformat -Wlogical-op -Wpacked -Wformat-security -Wmissing-include-dirs -Wformat-nonliteral -Wold-style-definition -Wdeclaration-after-statement -Wfloat-equal -Wmissing-declarations -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wmissing-noreturn -Wshadow -Wendif-labels -Wpointer-arith -Wcast-align -Wwrite-strings -Wno-unused-parameter -ffast-math
D: main.c: Running on host: Linux x86_64 2.6.28-17-generic #58-Ubuntu SMP Tue Dec 1 21:27:25 UTC 2009
I: main.c: Page size is 4096 bytes
D: main.c: Compiled with Valgrind support: no
D: main.c: Running in valgrind mode: no
D: main.c: Optimized build: yes
I: main.c: Machine ID is dd08204480ccaa8b779b02be4af34359.
I: main.c: Using runtime directory /home/matt/.pulse/dd08204480ccaa8b779b02be4af34359:runtime.
I: main.c: Using state directory /home/matt/.pulse.
I: main.c: Running in system mode: no
I: main.c: Fresh high-resolution timers available! Bon appetit!
D: memblock.c: Using shared memory pool with 1024 slots of size 64.0 KiB each, total size is 64.0 MiB, maximum usable slot size is 65472
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9/modules/module-gconf.so': success
D: module-gconf.c: Loading module 'module-combine' with args '' due to GConf configuration.
I: sink.c: Created sink 0 "combined" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: source.c: Created source 0 "combined.monitor" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
D: module-combine.c: Thread starting up
D: rtpoll.c: Acquired POSIX realtime signal SIGRTMIN+29
I: module.c: Loaded "module-combine" (index: #0; argument: "").
D: module-gconf.c: Loading module 'module-native-protocol-tcp' with args '' due to GConf configuration.
I: module.c: Loaded "module-native-protocol-tcp" (index: #1; argument: "").
D: module-gconf.c: Loading module 'module-esound-protocol-tcp' with args '' due to GConf configuration.
I: module.c: Loaded "module-esound-protocol-tcp" (index: #2; argument: "").
I: module.c: Loaded "module-gconf" (index: #3; argument: "").
D: module-suspend-on-idle.c: Sink combined becomes idle.
D: module-suspend-on-idle.c: Source combined.monitor becomes idle.
I: module.c: Loaded "module-suspend-on-idle" (index: #4; argument: "").
I: module-device-restore.c: Sucessfully opened database file '/home/matt/.pulse/dd08204480ccaa8b779b02be4af34359:device-volumes.x86_64-pc-linux-gnu.gdbm'.
I: module.c: Loaded "module-device-restore" (index: #5; argument: "").
I: module-stream-restore.c: Sucessfully opened database file '/home/matt/.pulse/dd08204480ccaa8b779b02be4af34359:stream-volumes.x86_64-pc-linux-gnu.gdbm'.
I: module.c: Loaded "module-stream-restore" (index: #6; argument: "").
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9/modules/module-hal-detect.so': success
I: module-hal-detect.c: Trying capability alsa
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_8086_3a3e_sound_card_0_alsa_capture_1
D: module-hal-detect.c: Loading module-alsa-source with arguments 'device_id=0 source_name=alsa_input.pci_8086_3a3e_sound_card_0_ alsa_capture_0 tsched=0'
D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
I: module-alsa-source.c: Successfully opened device front:0.
I: module-alsa-source.c: Successfully enabled mmap() mode.
I: (alsa-lib)control.c: Invalid CTL front:0
I: alsa-util.c: Unable to attach to mixer front:0: No such file or directory
I: alsa-util.c: Successfully attached to mixer 'hw:0'
I: alsa-util.c: Using mixer control "Capture".
I: module-device-restore.c: Restoring volume for source alsa_input.pci_8086_3a3e_sound_card_0_alsa_capture _0.
I: module-device-restore.c: Restoring mute state for source alsa_input.pci_8086_3a3e_sound_card_0_alsa_capture _0.
I: source.c: Created source 1 "alsa_input.pci_8086_3a3e_sound_card_0_alsa_capture _0" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: module-alsa-source.c: Using 8 fragments of size 1792 bytes, buffer time is 81.27ms
D: module-alsa-source.c: hwbuf_unused=0
D: module-alsa-source.c: setting avail_min=1
I: module-alsa-source.c: Volume ranges from 0 to 31.
I: module-alsa-source.c: Volume ranges from -16.50 dB to 30.00 dB.
I: alsa-util.c: All 2 channels can be mapped to mixer channels.
I: module-alsa-source.c: Using hardware volume control. Hardware dB scale supported.
D: alsa-util.c: snd_pcm_dump():
D: alsa-util.c: Soft volume PCM
D: alsa-util.c: Control: PCM Playback Volume
D: alsa-util.c: min_dB: -51
D: alsa-util.c: max_dB: 0
D: alsa-util.c: resolution: 256
D: alsa-util.c: Its setup is:
D: alsa-util.c: stream : CAPTURE
D: alsa-util.c: access : MMAP_INTERLEAVED
D: alsa-util.c: format : S16_LE
D: alsa-util.c: subformat : STD
D: alsa-util.c: channels : 2
D: alsa-util.c: rate : 44100
D: alsa-util.c: exact rate : 44100 (44100/1)
D: alsa-util.c: msbits : 16
D: alsa-util.c: buffer_size : 3584
D: alsa-util.c: period_size : 448
D: alsa-util.c: period_time : 10158
D: alsa-util.c: tstamp_mode : ENABLE
D: alsa-util.c: period_step : 1
D: alsa-util.c: avail_min : 448
D: alsa-util.c: period_event : 0
D: alsa-util.c: start_threshold : -1
D: alsa-util.c: stop_threshold : 8070450532247928832
D: alsa-util.c: silence_threshold: 0
D: alsa-util.c: silence_size : 0
D: alsa-util.c: boundary : 8070450532247928832
D: alsa-util.c: Slave: Hardware PCM card 0 'HDA Intel' device 0 subdevice 0
D: alsa-util.c: Its setup is:
D: alsa-util.c: stream : CAPTURE
D: alsa-util.c: access : MMAP_INTERLEAVED
D: alsa-util.c: format : S16_LE
D: alsa-util.c: subformat : STD
D: alsa-util.c: channels : 2
D: alsa-util.c: rate : 44100
D: alsa-util.c: exact rate : 44100 (44100/1)
D: alsa-util.c: msbits : 16
D: alsa-util.c: buffer_size : 3584
D: alsa-util.c: period_size : 448
D: alsa-util.c: period_time : 10158
D: alsa-util.c: tstamp_mode : ENABLE
D: alsa-util.c: period_step : 1
D: alsa-util.c: avail_min : 448
D: alsa-util.c: period_event
D: module-alsa-source.c: Thread starting up
D: module-alsa-source.c: Requested volume: 0: 100% 1: 100%
D: module-alsa-source.c: Got hardware volume: 0: 100% 1: 100%
D: rtpoll.c: Acquired POSIX realtime signal SIGRTMIN+28
D: module-alsa-source.c: Calculated software volume: 0: 100% 1: 100%
D: module-suspend-on-idle.c: Source alsa_input.pci_8086_3a3e_sound_card_0_alsa_capture _0 becomes idle.
I: module.c: Loaded "module-alsa-source" (index: #7; argument: "device_id=0 source_name=alsa_input.pci_8086_3a3e_sound_card_0_ alsa_capture_0 tsched=0").
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_8086_3a3e_sound_card_0_alsa_control__1
D: module-hal-detect.c: Loading module-alsa-sink with arguments 'device_id=0 sink_name=alsa_output.pci_8086_3a3e_sound_card_0_a lsa_playback_0 tsched=0'
D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
I: module-alsa-sink.c: Successfully opened device front:0.
I: module-alsa-sink.c: Successfully enabled mmap() mode.
I: (alsa-lib)control.c: Invalid CTL front:0
I: alsa-util.c: Unable to attach to mixer front:0: No such file or directory
I: alsa-util.c: Successfully attached to mixer 'hw:0'
I: alsa-util.c: Using mixer control "Master".
I: module-device-restore.c: Restoring volume for sink alsa_output.pci_8086_3a3e_sound_card_0_alsa_playba ck_0.
I: module-device-restore.c: Restoring mute state for sink alsa_output.pci_8086_3a3e_sound_card_0_alsa_playba ck_0.
I: sink.c: Created sink 1 "alsa_output.pci_8086_3a3e_sound_card_0_alsa_playba ck_0" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: module-device-restore.c: Restoring volume for source alsa_output.pci_8086_3a3e_sound_card_0_alsa_playba ck_0.monitor.
I: module-device-restore.c: Restoring mute state for source alsa_output.pci_8086_3a3e_sound_card_0_alsa_playba ck_0.monitor.
I: source.c: Created source 2 "alsa_output.pci_8086_3a3e_sound_card_0_alsa_playba ck_0.monitor" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: module-alsa-sink.c: Using 8 fragments of size 1792 bytes, buffer time is 81.27ms
D: module-alsa-sink.c: hwbuf_unused=0
D: module-alsa-sink.c: setting avail_min=1
I: module-alsa-sink.c: Volume ranges from 0 to 31.
I: module-alsa-sink.c: Volume ranges from -46.50 dB to 0.00 dB.
I: alsa-util.c: ALSA device lacks independant volume controls for each channel.
I: module-alsa-sink.c: Using hardware volume control. Hardware dB scale supported.
D: alsa-util.c: snd_pcm_dump():
D: alsa-util.c: Soft volume PCM
D: alsa-util.c: Control: PCM Playback Volume
D: alsa-util.c: min_dB: -51
D: alsa-util.c: max_dB: 0
D: alsa-util.c: resolution: 256
D: alsa-util.c: Its setup is:
D: alsa-util.c: stream : PLAYBACK
D: alsa-util.c: access : MMAP_INTERLEAVED
D: alsa-util.c: format : S16_LE
D: alsa-util.c: subformat : STD
D: alsa-util.c: channels : 2
D: alsa-util.c: rate : 44100
D: alsa-util.c: exact rate : 44100 (44100/1)
D: alsa-util.c: msbits : 16
D: alsa-util.c: buffer_size : 3584
D: alsa-util.c: period_size : 448
D: alsa-util.c: period_time : 10158
D: alsa-util.c: tstamp_mode : ENABLE
D: alsa-util.c: period_step : 1
D: alsa-util.c: avail_min : 448
D: alsa-util.c: period_event : 0
D: alsa-util.c: start_threshold : -1
D: alsa-util.c: stop_threshold : 8070450532247928832
D: alsa-util.c: silence_threshold: 0
D: alsa-util.c: silence_size : 0
D: alsa-util.c: boundary : 8070450532247928832
D: alsa-util.c: Slave: Hardware PCM card 0 'HDA Intel' device 0 subdevice 0
D: alsa-util.c: Its setup is:
D: alsa-util.c: stream : PLAYBACK
D: alsa-util.c: access : MMAP_INTERLEAVED
D: alsa-util.c: format : S16_LE
D: alsa-util.c: subformat : STD
D: alsa-util.c: channels : 2
D: alsa-util.c: rate : 44100
D: alsa-util.c: exact rate : 44100 (44100/1)
D: alsa-util.c: msbits : 16
D: alsa-util.c: buffer_size : 3584
D: alsa-util.c: period_size : 448
D: alsa-util.c: period_time : 10158
D: alsa-util.c: tstamp_mode : ENABLE
D: alsa-util.c: period_step : 1
D: alsa-util.c: avail_min : 448
D: alsa-util.c: period_even
D: module-alsa-sink.c: Thread starting up
D: module-alsa-sink.c: Requested volume: 0: 94% 1: 94%
D: rtpoll.c: Acquired POSIX realtime signal SIGRTMIN+27
D: module-alsa-sink.c: Got hardware volume: 0: 94% 1: 94%
D: module-alsa-sink.c: Calculated software volume: 0: 100% 1: 100%
I: module-alsa-sink.c: Starting playback.
D: module-suspend-on-idle.c: Source alsa_output.pci_8086_3a3e_sound_card_0_alsa_playba ck_0.monitor becomes idle.
D: module-suspend-on-idle.c: Sink alsa_output.pci_8086_3a3e_sound_card_0_alsa_playba ck_0 becomes idle.
I: module-combine.c: Configuring new sink: alsa_output.pci_8086_3a3e_sound_card_0_alsa_playba ck_0
D: memblockq.c: memblockq requested: maxlength=16777216, tlength=16777216, base=4, prebuf=1, minreq=0 maxrewind=0
D: memblockq.c: memblockq sanitized: maxlength=16777216, tlength=16777216, base=4, prebuf=4, minreq=4 maxrewind=0
I: module-stream-restore.c: Not restore device for stream sink-input-by-media-role:filter, because already set.
I: module-stream-restore.c: Restoring volume for sink input sink-input-by-media-role:filter.
I: module-stream-restore.c: Restoring mute state for sink input sink-input-by-media-role:filter.
D: module-suspend-on-idle.c: Sink alsa_output.pci_8086_3a3e_sound_card_0_alsa_playba ck_0 becomes busy.
I: resampler.c: Using resampler 'trivial'
I: resampler.c: Using s16le as working format.
D: memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
D: memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
I: sink-input.c: Created input 0 "Simultaneous output on HDA Intel - ALC1200 Analog" on alsa_output.pci_8086_3a3e_sound_card_0_alsa_playba ck_0 with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
D: module-alsa-sink.c: hwbuf_unused=0
D: module-alsa-sink.c: setting avail_min=1
I: module.c: Loaded "module-alsa-sink" (index: #8; argument: "device_id=0 sink_name=alsa_output.pci_8086_3a3e_sound_card_0_a lsa_playback_0 tsched=0").
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_8086_3a3e_sound_card_0_alsa_playback_1
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_8086_3a3e_sound_card_0_alsa_capture_2
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/computer_alsa_sequencer
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/computer_alsa_timer
I: module-hal-detect.c: Loaded 2 modules.
I: module.c: Loaded "module-hal-detect" (index: #9; argument: "tsched=0").
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9/modules/module-esound-protocol-unix.so': success
I: module.c: Loaded "module-esound-protocol-unix" (index: #10; argument: "").
I: module.c: Loaded "module-native-protocol-unix" (index: #11; argument: "").
I: module-default-device-restore.c: Restored default sink 'combined'.
D: core-subscribe.c: Dropped redundant event due to change event.
I: module-default-device-restore.c: Restored default source 'alsa_input.pci_8086_3a3e_sound_card_0_alsa_captur e_0'.
I: module.c: Loaded "module-default-device-restore" (index: #12; argument: "").
I: module.c: Loaded "module-rescue-streams" (index: #13; argument: "").
I: module.c: Loaded "module-always-sink" (index: #14; argument: "").
I: client.c: Created 0 "ConsoleKit Session /org/freedesktop/ConsoleKit/Session19"
D: module-console-kit.c: Added new session /org/freedesktop/ConsoleKit/Session19
I: module.c: Loaded "module-console-kit" (index: #15; argument: "").
I: module.c: Loaded "module-position-event-sounds" (index: #16; argument: "").
I: main.c: Daemon startup complete.
D: module-hal-detect.c: dbus: interface=org.freedesktop.DBus, path=/org/freedesktop/DBus, member=NameAcquired
D: module-console-kit.c: dbus: interface=org.freedesktop.DBus, path=/org/freedesktop/DBus, member=NameAcquired
I: module-suspend-on-idle.c: Source combined.monitor idle for too long, suspending ...
I: module-suspend-on-idle.c: Sink combined idle for too long, suspending ...
D: module-alsa-sink.c: hwbuf_unused=0
D: module-alsa-sink.c: setting avail_min=1
D: module-suspend-on-idle.c: Sink alsa_output.pci_8086_3a3e_sound_card_0_alsa_playba ck_0 becomes idle.
D: module-suspend-on-idle.c: Sink alsa_output.pci_8086_3a3e_sound_card_0_alsa_playba ck_0 becomes idle.
I: sink-input.c: Freeing input 0 "Simultaneous output on HDA Intel - ALC1200 Analog"
I: module-combine.c: Device suspended...
I: module-suspend-on-idle.c: Source alsa_input.pci_8086_3a3e_sound_card_0_alsa_capture _0 idle for too long, suspending ...
I: module-alsa-source.c: Device suspended...
I: module-suspend-on-idle.c: Source alsa_output.pci_8086_3a3e_sound_card_0_alsa_playba ck_0.monitor idle for too long, suspending ...
I: module-suspend-on-idle.c: Sink alsa_output.pci_8086_3a3e_sound_card_0_alsa_playba ck_0 idle for too long, suspending ...
I: module-alsa-sink.c: Device suspended...

D: module-hal-detect.c: dbus: interface=org.freedesktop.ConsoleKit.Seat, path=/org/freedesktop/ConsoleKit/Seat1, member=SessionAdded
D: module-console-kit.c: dbus: interface=org.freedesktop.ConsoleKit.Seat, path=/org/freedesktop/ConsoleKit/Seat1, member=SessionAdded
D: module-hal-detect.c: dbus: interface=org.freedesktop.ConsoleKit.Seat, path=/org/freedesktop/ConsoleKit/Seat1, member=SessionRemoved
D: module-console-kit.c: dbus: interface=org.freedesktop.ConsoleKit.Seat, path=/org/freedesktop/ConsoleKit/Seat1, member=SessionRemoved

D: module-hal-detect.c: dbus: interface=org.freedesktop.ConsoleKit.Seat, path=/org/freedesktop/ConsoleKit/Seat1, member=SessionAdded
D: module-console-kit.c: dbus: interface=org.freedesktop.ConsoleKit.Seat, path=/org/freedesktop/ConsoleKit/Seat1, member=SessionAdded
D: module-hal-detect.c: dbus: interface=org.freedesktop.ConsoleKit.Seat, path=/org/freedesktop/ConsoleKit/Seat1, member=SessionRemoved
D: module-console-kit.c: dbus: interface=org.freedesktop.ConsoleKit.Seat, path=/org/freedesktop/ConsoleKit/Seat1, member=SessionRemoved
D: module-hal-detect.c: dbus: interface=org.freedesktop.ConsoleKit.Seat, path=/org/freedesktop/ConsoleKit/Seat1, member=SessionAdded
D: module-console-kit.c: dbus: interface=org.freedesktop.ConsoleKit.Seat, path=/org/freedesktop/ConsoleKit/Seat1, member=SessionAdded
D: module-hal-detect.c: dbus: interface=org.freedesktop.ConsoleKit.Seat, path=/org/freedesktop/ConsoleKit/Seat1, member=SessionRemoved
D: module-console-kit.c: dbus: interface=org.freedesktop.ConsoleKit.Seat, path=/org/freedesktop/ConsoleKit/Seat1, member=SessionRemoved
^CI: main.c: Got signal SIGINT.
I: main.c: Exiting.
I: main.c: Daemon shutdown initiated.
I: module.c: Unloading "module-combine" (index: #0).
D: module-rescue-streams.c: No sink inputs to move away.
D: module-rescue-streams.c: No source outputs to move away.
D: core-subscribe.c: Dropped redundant event due to remove event.
D: core-subscribe.c: Dropped redundant event due to remove event.
D: module-combine.c: Thread shutting down
I: sink.c: Freeing sink 0 "combined"
I: source.c: Freeing source 0 "combined.monitor"
I: module.c: Unloaded "module-combine" (index: #0).
I: module.c: Unloading "module-native-protocol-tcp" (index: #1).
I: module.c: Unloaded "module-native-protocol-tcp" (index: #1).
I: module.c: Unloading "module-esound-protocol-tcp" (index: #2).
I: module.c: Unloaded "module-esound-protocol-tcp" (index: #2).
I: module.c: Unloading "module-gconf" (index: #3).
D: module-gconf.c: Unloading module #0
D: module-gconf.c: Unloading module #1
D: module-gconf.c: Unloading module #2
I: module.c: Unloaded "module-gconf" (index: #3).
I: module.c: Unloading "module-suspend-on-idle" (index: #4).
I: module.c: Unloaded "module-suspend-on-idle" (index: #4).
I: module.c: Unloading "module-device-restore" (index: #5).
I: module.c: Unloaded "module-device-restore" (index: #5).
I: module.c: Unloading "module-stream-restore" (index: #6).
I: module.c: Unloaded "module-stream-restore" (index: #6).
I: module.c: Unloading "module-alsa-source" (index: #7).
D: module-rescue-streams.c: No source outputs to move away.
D: module-alsa-source.c: Thread shutting down
I: source.c: Freeing source 1 "alsa_input.pci_8086_3a3e_sound_card_0_alsa_capture _0"
I: module.c: Unloaded "module-alsa-source" (index: #7).
I: module.c: Unloading "module-alsa-sink" (index: #8).
D: module-always-sink.c: Autoloading null-sink as no other sinks detected.
I: sink.c: Created sink 2 "auto_null" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: source.c: Created source 3 "auto_null.monitor" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
D: module-null-sink.c: Thread starting up
D: rtpoll.c: Acquired POSIX realtime signal SIGRTMIN+29
I: module.c: Loaded "module-null-sink" (index: #17; argument: "sink_name=auto_null").
D: module-rescue-streams.c: No sink inputs to move away.
D: module-rescue-streams.c: No source outputs to move away.
D: module-alsa-sink.c: Thread shutting down
I: sink.c: Freeing sink 1 "alsa_output.pci_8086_3a3e_sound_card_0_alsa_playba ck_0"
I: source.c: Freeing source 2 "alsa_output.pci_8086_3a3e_sound_card_0_alsa_playba ck_0.monitor"
I: module.c: Unloaded "module-alsa-sink" (index: #8).
I: module.c: Unloading "module-hal-detect" (index: #9).
I: module.c: Unloaded "module-hal-detect" (index: #9).
I: module.c: Unloading "module-esound-protocol-unix" (index: #10).
I: module.c: Unloaded "module-esound-protocol-unix" (index: #10).
I: module.c: Unloading "module-native-protocol-unix" (index: #11).
I: module.c: Unloaded "module-native-protocol-unix" (index: #11).
I: module.c: Unloading "module-default-device-restore" (index: #12).
I: module.c: Unloaded "module-default-device-restore" (index: #12).
I: module.c: Unloading "module-rescue-streams" (index: #13).
I: module.c: Unloaded "module-rescue-streams" (index: #13).
I: module.c: Unloading "module-always-sink" (index: #14).
I: module.c: Unloaded "module-always-sink" (index: #14).
I: module.c: Unloading "module-console-kit" (index: #15).
D: module-console-kit.c: Removing session /org/freedesktop/ConsoleKit/Session19
I: client.c: Freed 0 "ConsoleKit Session /org/freedesktop/ConsoleKit/Session19"
I: module.c: Unloaded "module-console-kit" (index: #15).
I: module.c: Unloading "module-position-event-sounds" (index: #16).
I: module.c: Unloaded "module-position-event-sounds" (index: #16).
I: module.c: Unloading "module-null-sink" (index: #17).
D: core-subscribe.c: Dropped redundant event due to remove event.
D: core-subscribe.c: Dropped redundant event due to remove event.
D: module-null-sink.c: Thread shutting down
I: sink.c: Freeing sink 2 "auto_null"
I: source.c: Freeing source 3 "auto_null.monitor"
I: module.c: Unloaded "module-null-sink" (index: #17).
D: core-subscribe.c: Dropped redundant event due to remove event.
I: main.c: Daemon terminated.
matt@matt64ubuntu:~$

since the provided line of coded did not return to the command prompt, i hit enter twice, which produced the empty line in above code snipped. After a while i ended it with ctrl-c

4. If you are having a problem only with a specific application, specify the application's name and result you received from the instructions above (A-F).
when i start eve-online it turns up in the volume control and produces sound. so it should be A.
The problem starts when i start a second instance to play with a second character. I have seperated the charactetrs by installing them in differnet directories, so they should behave like 2 different applications. It used to work fine i usually played with 3 chars continuosly without issues.
When i start the second char, it also tunrs up in the pa volume control, but after a few seconds i get the connection lost popup and the volume control closes when i click ok. at the same moment all eve instances crashes with an error. When i try to restart eve it cloeses again befor ei get to login screen.

I have tested TS and still have the same issues as before this tutorial and even worse so that TS gets quite and i cant hear anything. Only when i change the TS settings i get sound back for a coule of minutes.

any help is appreciated if nothing else helps i might follow one of the earlier replioes to remove pa completly and se eif that helps.

PS: oh when i log-off and logon i can start eve again and play with one char without a issue.

baserunner_ams
January 10th, 2010, 05:18 PM
i got a workaround:
disable sound on all eve clients. :(

this seems to work, but of course it is not satisfying. I should be able to hav emulitple applications using sound....

cheers

shadfc
January 13th, 2010, 05:32 PM
Ill just add that I followed part A for Karmic and the optional part of adjusting the default-fragments and default-fragment-size-msec in order to get Skype to stop stuttering. All other sound on my system worked fine but Skype was unusable. I fiddled with those settings a bunch and got a little better and then worse but never good. Finally, I skipped out on the medibuntu version of Skype and installed the beta deb directly from skype's website. The audio devices (including microphone) only show Pulse Audio Server (local) but it works like a charm now. No stuttering at all. Hope this helps someone else.

ArponerO
January 16th, 2010, 07:33 PM
I had a problem with my Ubuntu 9.10, and now I know that Amarok2 somehow broke my sound after launching it. The solution: use pulseaudio also with Amarok2.

Hope it helps other people with the same problem.

LiteDrive
January 21st, 2010, 09:09 AM
Yo.

I'm a little confused. Even though I'm running 9.10, the fact that mine is 64 bit leaves me a little hesitant.

Should that make a difference? I'm a total Ubu noob; been running this for about a week ever since Windows went kaput on me.

danam1
January 24th, 2010, 11:50 PM
Psyche83,
Re: Your May 10, 2008 post.

This is my first post on Ubuntuforums.

I would like your help, please.

First, thank you for this helpful guide.

I followed the steps through to Appendix A - General Troubleshooting. My result was "C". The programs show but their sound is either almost nil, nil, or good until "bugs" interrupt the sound.

1. In Sound Recorder there is no sound (I can tap on my microphone and hear the tapping on the speakers).

2. With Skype there is a brief bit of sound before silence.


3. With Rhythmbox Music Player MP3 files play clearly until either 1) the sound becomes "fuzzed" and needs the first of many volume control setting changes to find a clear position or 2) a random noise blip of half a second will occur followed by more clear music.


I've got Karmic Koala on a Thinkpad T20 with a P3 650Mhz, and 375MB RAM.


dana@dana-laptop:~$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: CS46xx [Sound Fusion CS46xx], device 0: CS46xx [CS46xx]
Subdevices: 30/31
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
card 0: CS46xx [Sound Fusion CS46xx], device 1: CS46xx - Rear [CS46xx - Rear]
Subdevices: 31/31
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
card 0: CS46xx [Sound Fusion CS46xx], device 2: CS46xx - IEC958 [CS46xx - IEC958]
Subdevices: 1/1
Subdevice #0: subdevice #0


dana@dana-laptop:~$ pkill pulseaudio; sleep 2; pulseaudio -vv
I: main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted
I: main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation not permitted
D: core-rtclock.c: Timer slack is set to 50 us.
I: core-util.c: Failed to acquire high-priority scheduling: No such file or directory
I: main.c: This is PulseAudio 0.9.19
D: main.c: Compilation host: i486-pc-linux-gnu
D: main.c: Compilation CFLAGS: -g -O2 -g -Wall -O3 -Wall -W -Wextra -pipe -Wno-long-long -Winline -Wvla -Wno-overlength-strings -Wunsafe-loop-optimizations -Wundef -Wformat=2 -Wlogical-op -Wsign-compare -Wformat-security -Wmissing-include-dirs -Wformat-nonliteral -Wold-style-definition -Wpointer-arith -Winit-self -Wdeclaration-after-statement -Wfloat-equal -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wmissing-declarations -Wmissing-noreturn -Wshadow -Wendif-labels -Wcast-align -Wstrict-aliasing=2 -Wwrite-strings -Wno-unused-parameter -ffast-math -Wp,-D_FORTIFY_SOURCE=2 -fno-common -fdiagnostics-show-option
D: main.c: Running on host: Linux i686 2.6.31-17-generic #54-Ubuntu SMP Thu Dec 10 16:20:31 UTC 2009
D: main.c: Found 1 CPUs.
I: main.c: Page size is 4096 bytes
D: main.c: Compiled with Valgrind support: no
D: main.c: Running in Valgrind mode: no
D: main.c: Optimized build: yes
D: main.c: All asserts enabled.
I: main.c: Machine ID is a1845448c2b840bbc91940804b47d8f0.
I: main.c: Session ID is a1845448c2b840bbc91940804b47d8f0-1264361201.272609-1438599701.
I: main.c: Using runtime directory /home/dana/.pulse/a1845448c2b840bbc91940804b47d8f0-runtime.
I: main.c: Using state directory /home/dana/.pulse.
I: main.c: Using modules directory /usr/lib/pulse-0.9.19/modules.
I: main.c: Running in system mode: no
E: pid.c: Daemon already running.
E: main.c: pa_pid_file_create() failed.

Shown in red in Terminal:

"Daemon already running.
pa_pid_file_create() failed."


Ideas please!

Dana

jackzh
January 26th, 2010, 09:40 PM
Thank you so much for the solution!!!!

FINAAAAAAAAAAAAAAALY, got the usb headset to work in all of the following:

Music and Movie
Sound recording
Flash (Chrome, youbute etc).

Thanks a lot!!

Jack

Lechatelier
January 28th, 2010, 03:50 AM
This is unbelievable! Do I need to read 163 pages to make my microphone work in Ubuntu 9.10 or 9.04 (well maybe) and 2 different computers. What is worse I regularly used Skype with another version but do not know which because it was a long time ago. same equipment.

I have been using ubuntu since version 4.10 and never seen a problem like this since I tried to make it ẁork with a usb modem (never did).

I have been trying many things in the last 4 day and the damn thing hang the machine when trying recording, and skype return only guttural sounds.

Do you think I could operate skype by installing it on a specially installed virtual machine? In such case which Ubuntu/Skype versions would you recommend.

Ian Clark
January 28th, 2010, 05:03 AM
This is unbelievable! Do I need to read 163 pages to make my microphone work in Ubuntu 9.10 or 9.04 (well maybe) and 2 different computers. What is worse I regularly used Skype with another version but do not know which because it was a long time ago. same equipment.

I have been using ubuntu since version 4.10 and never seen a problem like this since I tried to make it ẁork with a usb modem (never did).

I have been trying many things in the last 4 day and the damn thing hang the machine when trying recording, and skype return only guttural sounds.

Do you think I could operate skype by installing it on a specially installed virtual machine? In such case which Ubuntu/Skype versions would you recommend.
Install skype by the .deb package from the official skype download page and it should detect your mic OK. If your cam doesn't work, then make a text file with this (and only this) as the body:

LD_PRELOAD=/usr/lib/libv4l/v4l2convert.so skype
Save it in any directory you want. Click on it, and chose "run". Bob's your uncle!

elvino
January 30th, 2010, 03:01 AM
I have searched these forums for over an hour and cannot get the answer I need!
I use Ubuntu 9.10, Skype 2.1x, a usb phone and my problem is simply that Skype is working perfectly. No lag, good quality etc. BUT: I need to speakers on my PC to ring rather than the USB phone. I cannot work out how to configure pulseaudio to selectively ring the speakers for an incoming call. I am sure this is a known problem. It's just that I don't have the solution. Mt usb phone does come with a MS windows driver that makes the handset ring when a call is received. Could I perhaps use that in some way?

almightybunghole
January 30th, 2010, 03:21 AM
Hi there,

I'm having terrible trouble trying to get Pulseaudio to work correctly on my Jaunty box, wondered if anyone might have an idea what I can try.

I've followed the instructions in the post (great post by the way, very clear) and got PA installed. If I send a stream to the analogue output of my card I can hear it no problem, but nothing comes out of the digital output (I know the cables etc are ok as I have tested using speaker-test)

I can see the output is getting to PA so it must be between there and the HW, but I'm not sure how to troubleshoot this?

The other (I'm sure unrelated) problem I have is that I would like to play VLC output over PA but it seems to not be supported?

aplay -l output:

**** List of PLAYBACK Hardware Devices ****
card 0: Intel [HDA Intel], device 0: ALC662 Analog [ALC662 Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: Intel [HDA Intel], device 1: ALC662 Digital [ALC662 Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0




Thanks in advance!

George

libhart
January 30th, 2010, 06:36 AM
Would someone mind cat'ing his/her volume-restore.table. Don't know if it's a bug or not, but pulse refuses to create this file for me so xine ends up back at no sound even though I have the fix in for the sound not muting on reboot. I just don't know what that file's supposed to look like, figure I'll just manually create it. Thanks.

listerdl
February 3rd, 2010, 06:56 PM
At the very beginning of this tutorial there was advice to all users to:

1. Backup (and then delete) your previous configuration files:
Code:

Sorry to be so basic, but can someone please tell me how this is done?

The code is here:


$ mkdir ~/pulse-backup && cp -r ~/.pulse ~/.asound* /etc/asound.conf /etc/pulse -t ~/pulse-backup/
$ rm -r ~/.pulse ~/.asound*
$ sudo rm /etc/asound

Do I navigate to each line of code above?

Thanks.....

tiggsy
February 3rd, 2010, 07:15 PM
Open a terminal (Applications -> Terminal).

Copy (or type) each line one at a time (without the $, eg. the first line starts mkdir...) and press enter. When the commands have all finished and it goes back to the prompt you paste the next one in, and so on.

listerdl
February 3rd, 2010, 07:26 PM
Open a terminal (Applications -> Terminal).

Copy (or type) each line one at a time (without the $, eg. the first line starts mkdir...) and press enter. When the commands have all finished and it goes back to the prompt you paste the next one in, and so on.

Thanks...i did as suggested but it didnt work, instead I got this message, is that because I need to put sudo at the beginning?


cp: cannot stat `/home/henry/.asound*': No such file or directory
cp: cannot stat `/etc/asound.conf': No such file or directory


I cant see why my file directories structure should be wrong since it is a clean install (if that is what the problem is)....

Thanks for all help!

Zendir
February 8th, 2010, 06:16 PM
Hello, great job whit this tutorial, and for its support!!!
I followed the steps of this guide, but many strange errors appeared in the process. First i explain my initial situation: Ubuntu hardy, all ok, but only one aplication can play sound at time (i am a older pentium III and the sound card is a ISA Soundblaster 16, and for enable it i do this: Assign IRQ 5 at BIOS, and add snd-sb16 in last line of etc/modules file), after near 2 years i decided to fix this problem and i found this tutorial


The final situation: no sound except whit OSS selected in System/Preferences/Sound, and very poor quality if play movies , and no flash at all (firefox seems to believe that not is installed and suggest to install it)

The stranges things:
a) in the part A, step 5 : the "Applications/Sound & Video/PulseAudio Device Chooser" is installed and seem to run good, but when i select Volume Control from this applet's menu the window appear 1 or 2 seconds & close , the same result if type pavucontrol in the terminal, no error messages or other things, the windows open and close inmediately.
b) after finalize the tutorial (i ommited the part a step 5) an reboot many times , in the top panel appear a lamp who give me this message:


Refresh Advanced Linux Sound Architecture (ALSA) configuration presets

New Advanced Linux Sound Architecture (ALSA) configuration presets have been added. Please execute the asoundconf(1) set-default-card macro in a Terminal now to refresh your user's configuration presets. You may accomplish this task by executing the following command in a Terminal: asoundconf set-default-card I followed this message :


filipaushkas@filipaushkas-desktop:~$ asoundconf set-default-card
You have omitted a necessary parameter. Please see the output from `asoundconf list`, and use one of those sound card(s) as the parameter.;)

filipaushkas@filipaushkas-desktop:~$ asoundconf list
Names of available sound cards:
S16
filipaushkas@filipaushkas-desktop:~$ asoundconf S16
Usage:
asoundconf is-active
asoundconf get|delete PARAMETER
asoundconf set PARAMETER VALUE
asoundconf list

Convenience macro functions:
asoundconf set-default-card PARAMETER
asoundconf reset-default-card
asoundconf set-pulseaudio
asoundconf unset-pulseaudio
asoundconf set-oss PARAMETER
asoundconf unset-oss c) In the part a,step 2 this is the terminal photo-finish (im sorry my computer is in spanish language, but i marked in blue the suspicious part):

filipaushkas@filipaushkas-desktop:~$ sudo aptitude install libasound2-plugins padevchooser libsdl1.2debian-pulseaudio
Leyendo lista de paquetes... Hecho
Creando árbol de dependencias
Leyendo la información de estado... Hecho
Leyendo la información de estado extendido
Inicializando el estado de los paquetes... Hecho
Construir la base de datos de etiquetas... Hecho
Se instalarán automáticamente los siguientes paquetes NUEVOS:
libgconfmm-2.6-1c2 libpulse-mainloop-glib0 paman paprefs pavucontrol
pavumeter pulseaudio-module-zeroconf
Se ELIMINARÁN automáticamente los siguientes paquetes:
libsdl1.2debian-alsa
Se instalarán los siguiente paquetes NUEVOS:
libasound2-plugins libgconfmm-2.6-1c2 libpulse-mainloop-glib0
libsdl1.2debian-pulseaudio padevchooser paman paprefs pavucontrol
pavumeter pulseaudio-module-zeroconf
Se ELIMINARÁN los siguientes paquetes:
libsdl1.2debian-alsa
0 paquetes actualizados, 10 nuevos instalados, 1 para eliminar y 0 sin actualizar.
Necesito descargar 607kB de ficheros. Después de desempaquetar se usarán 2105kB.
¿Quiere continuar? [Y/n/?] y
Escribiendo información de estado extendido... Hecho
Des:1 http://es.archive.ubuntu.com hardy/main libasound2-plugins 1.0.15-1ubuntu3 [106kB]
Des:2 http://es.archive.ubuntu.com hardy/main libgconfmm-2.6-1c2 2.22.0-1 [31,4kB]
Des:3 http://es.archive.ubuntu.com hardy-updates/main libpulse-mainloop-glib0 0.9.10-1ubuntu1.1 [23,9kB]
Des:4 http://es.archive.ubuntu.com hardy/universe paman 0.9.4-1ubuntu1 [95,3kB]
Des:5 http://es.archive.ubuntu.com hardy-updates/main pulseaudio-module-zeroconf 0.9.10-1ubuntu1.1 [17,5kB]
Des:6 http://es.archive.ubuntu.com hardy/universe paprefs 0.9.6-1ubuntu1 [31,8kB]
Des:7 http://es.archive.ubuntu.com hardy/universe pavucontrol 0.9.5-1ubuntu1 [46,2kB]
Des:8 http://es.archive.ubuntu.com hardy/universe pavumeter 0.9.3-1ubuntu1 [29,2kB]
Des:9 http://es.archive.ubuntu.com hardy/universe padevchooser 0.9.3-2ubuntu3 [20,2kB]
Des:10 http://es.archive.ubuntu.com hardy/universe libsdl1.2debian-pulseaudio 1.2.13-1ubuntu1 [206kB]
Descargados 607kB en 2s (235kB/s).
dpkg: libsdl1.2debian-alsa: problemas de dependencias, pero se desinstalará de todas formas
tal y como se solicitó:
libsdl1.2debian depende de libsdl1.2debian-alsa (= 1.2.13-1ubuntu1) | libsdl1.2debian-all (= 1.2.13-1ubuntu1) | libsdl1.2debian-esd (= 1.2.13-1ubuntu1) | libsdl1.2debian-arts (= 1.2.13-1ubuntu1) | libsdl1.2debian-oss (= 1.2.13-1ubuntu1) | libsdl1.2debian-nas (= 1.2.13-1ubuntu1) | libsdl1.2debian-pulseaudio (= 1.2.13-1ubuntu1); sin embargo:
El paquete `libsdl1.2debian-alsa' va a ser desinstalado.
El paquete `libsdl1.2debian-all' no está instalado.
El paquete `libsdl1.2debian-esd' no está instalado.
El paquete `libsdl1.2debian-arts' no está instalado.
El paquete `libsdl1.2debian-oss' no está instalado.
El paquete `libsdl1.2debian-nas' no está instalado.
El paquete `libsdl1.2debian-pulseaudio' no está instalado.
(Leyendo la base de datos ...
206252 ficheros y directorios instalados actualmente.)
Desinstalando libsdl1.2debian-alsa ...
Processing triggers for libc6 ...
ldconfig deferred processing now taking place
Seleccionando el paquete libasound2-plugins previamente no seleccionado.
(Leyendo la base de datos ...
206244 ficheros y directorios instalados actualmente.)
Desempaquetando libasound2-plugins (de .../libasound2-plugins_1.0.15-1ubuntu3_i386.deb) ...
Seleccionando el paquete libgconfmm-2.6-1c2 previamente no seleccionado.
Desempaquetando libgconfmm-2.6-1c2 (de .../libgconfmm-2.6-1c2_2.22.0-1_i386.deb) ...
Seleccionando el paquete libpulse-mainloop-glib0 previamente no seleccionado.
Desempaquetando libpulse-mainloop-glib0 (de .../libpulse-mainloop-glib0_0.9.10-1ubuntu1.1_i386.deb) ...
Seleccionando el paquete paman previamente no seleccionado.
Desempaquetando paman (de .../paman_0.9.4-1ubuntu1_i386.deb) ...
Seleccionando el paquete pulseaudio-module-zeroconf previamente no seleccionado.
Desempaquetando pulseaudio-module-zeroconf (de .../pulseaudio-module-zeroconf_0.9.10-1ubuntu1.1_i386.deb) ...
Seleccionando el paquete paprefs previamente no seleccionado.
Desempaquetando paprefs (de .../paprefs_0.9.6-1ubuntu1_i386.deb) ...
Seleccionando el paquete pavucontrol previamente no seleccionado.
Desempaquetando pavucontrol (de .../pavucontrol_0.9.5-1ubuntu1_i386.deb) ...
Seleccionando el paquete pavumeter previamente no seleccionado.
Desempaquetando pavumeter (de .../pavumeter_0.9.3-1ubuntu1_i386.deb) ...
Seleccionando el paquete padevchooser previamente no seleccionado.
Desempaquetando padevchooser (de .../padevchooser_0.9.3-2ubuntu3_i386.deb) ...
Seleccionando el paquete libsdl1.2debian-pulseaudio previamente no seleccionado.
Desempaquetando libsdl1.2debian-pulseaudio (de .../libsdl1.2debian-pulseaudio_1.2.13-1ubuntu1_i386.deb) ...
Configurando libasound2-plugins (1.0.15-1ubuntu3) ...
Configurando libgconfmm-2.6-1c2 (2.22.0-1) ...

Configurando libpulse-mainloop-glib0 (0.9.10-1ubuntu1.1) ...

Configurando paman (0.9.4-1ubuntu1) ...

Configurando pulseaudio-module-zeroconf (0.9.10-1ubuntu1.1) ...
Configurando paprefs (0.9.6-1ubuntu1) ...

Configurando pavucontrol (0.9.5-1ubuntu1) ...

Configurando pavumeter (0.9.3-1ubuntu1) ...

Configurando padevchooser (0.9.3-2ubuntu3) ...

Configurando libsdl1.2debian-pulseaudio (1.2.13-1ubuntu1) ...

Processing triggers for libc6 ...
ldconfig deferred processing now taking place
Leyendo lista de paquetes... Hecho
Creando árbol de dependencias
Leyendo la información de estado... Hecho
Leyendo la información de estado extendido
Inicializando el estado de los paquetes... Hecho
Escribiendo información de estado extendido... Hecho
Construir la base de datos de etiquetas... Hecho
filipaushkas@filipaushkas-desktop:~$ sudo aptitude remove --purge libflashsupport flashplugin-nonfree-extrasound
Leyendo lista de paquetes... Hecho
Creando árbol de dependencias
Leyendo la información de estado... Hecho
Leyendo la información de estado extendido
Inicializando el estado de los paquetes... Hecho
Construir la base de datos de etiquetas... Hecho
No se instalará, actualizará o eliminará ningún paquete.
0 paquetes actualizados, 0 nuevos instalados, 0 para eliminar y 0 sin actualizar.
Necesito descargar 0B de ficheros. Después de desempaquetar se usarán 0B.
Escribiendo información de estado extendido... Hecho
Leyendo lista de paquetes... Hecho
Creando árbol de dependencias
Leyendo la información de estado... Hecho
Leyendo la información de estado extendido
Inicializando el estado de los paquetes... Hecho
Construir la base de datos de etiquetas... Hecho Is for that reason for no submit you an error report following your instructios in Appendix A - General Troubleshooting, thanks for read this large post an help me please

everah
February 12th, 2010, 05:52 PM
I landed in this thread because I kept experiencing sound stoppage on my computer. I'd have sound for a little while after boot but after a short while (a few minutes to a few days) sound would just disappear. If I rebooted, sound would come back.

I followed the instructions on the first page of this thread the other day and now I have no sound at all. Even after rebooting. I read through Appendix A opening up the PulseAudio Volume Control Playback tab and what I got was the app does NOT play audio but there is an entry in the playback tab.

I am on Ubuntu 8.10, i386. This problem affects all sound on my system, not just Flash. I literally have no sound at all anymore.

Results of aplay -l:

:~$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: ICH5 [Intel ICH5], device 0: Intel ICH [Intel ICH5]
Subdevices: 0/1
Subdevice #0: subdevice #0
card 0: ICH5 [Intel ICH5], device 4: Intel ICH - IEC958 [Intel ICH5 - IEC958]
Subdevices: 1/1
Subdevice #0: subdevice #0

When I ran the verbose output of pulseaudio it actually hung there. The following are the results of pkill pulseaudio; sleep 2; pulseaudio -vv, as far as they went before hanging:

:~$ pkill pulseaudio; sleep 2; pulseaudio -vv
I: main.c: PolicyKit refuses acquire-high-priority privilige.
I: main.c: Called SUID root and real-time/high-priority scheduling was requested in the configuration. However, we lack the necessary priviliges:
I: main.c: We are not in group 'pulse-rt' and PolicyKit refuse to grant us priviliges. Dropping SUID again.
I: main.c: For enabling real-time scheduling please acquire the appropriate PolicyKit priviliges, or become a member of 'pulse-rt', or increase the RLIMIT_NICE/RLIMIT_RTPRIO resource limits for this user.
I: main.c: Note that real-time/high-priority scheduling is NOT normally required. If you experience crackling or other sound anomalies, consider one or more of the above solutions.
I: main.c: High-priority scheduling enabled in configuration but now allowed by policy. Disabling forcibly.
W: ltdl-bind-now.c: Failed to find original dlopen loader.
W: main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted
W: main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation not permitted
I: main.c: This is PulseAudio 0.9.10
I: main.c: Page size is 4096 bytes
I: main.c: Fresh high-resolution timers available! Bon appetit!
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9/modules/module-hal-detect.so': success
I: module-hal-detect.c: Trying capability alsa
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/computer_alsa_timer
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/computer_alsa_sequencer
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_8086_24d5_sound_card_0_alsa_playback_4
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_8086_24d5_sound_card_0_alsa_capture_3
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_8086_24d5_sound_card_0_alsa_capture_2
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_8086_24d5_sound_card_0_alsa_capture_1
D: module-hal-detect.c: Loading module-alsa-sink with arguments 'device_id=0 sink_name=alsa_output.pci_8086_24d5_sound_card_0_a lsa_playback_0'
D: alsa-util.c: Trying front:0...
I: module-alsa-sink.c: Successfully opened device front:0.
I: module-alsa-sink.c: Successfully enabled mmap() mode.
ALSA lib control.c:909:(snd_ctl_open_noupdate) Invalid CTL front:0
I: alsa-util.c: Unable to attach to mixer front:0: No such file or directory
I: alsa-util.c: Successfully attached to mixer 'hw:0'
I: alsa-util.c: Using mixer control "Master".
I: sink.c: Created sink 0 "alsa_output.pci_8086_24d5_sound_card_0_alsa_playba ck_0" with sample spec "s16le 2ch 44100Hz"
I: source.c: Created source 0 "alsa_output.pci_8086_24d5_sound_card_0_alsa_playba ck_0.monitor" with sample spec "s16le 2ch 44100Hz"
I: module-alsa-sink.c: Using 8 fragments of size 1760 bytes.
I: alsa-util.c: All 2 channels can be mapped to mixer channels. Using hardware volume control.
D: module-alsa-sink.c: Thread starting up
D: rtpoll.c: Acquired POSIX realtime signal SIGRTMIN+29
I: module-alsa-sink.c: Starting playback.
I: module.c: Loaded "module-alsa-sink" (index: #0; argument: "device_id=0 sink_name=alsa_output.pci_8086_24d5_sound_card_0_a lsa_playback_0").
D: module-hal-detect.c: Loading module-alsa-source with arguments 'device_id=0 source_name=alsa_input.pci_8086_24d5_sound_card_0_ alsa_capture_0'
D: alsa-util.c: Trying front:0...
I: module-alsa-source.c: Successfully opened device front:0.
I: module-alsa-source.c: Successfully enabled mmap() mode.
ALSA lib control.c:909:(snd_ctl_open_noupdate) Invalid CTL front:0
I: alsa-util.c: Unable to attach to mixer front:0: No such file or directory
I: alsa-util.c: Successfully attached to mixer 'hw:0'
I: alsa-util.c: Using mixer control "Capture".
I: source.c: Created source 1 "alsa_input.pci_8086_24d5_sound_card_0_alsa_capture _0" with sample spec "s16le 2ch 44100Hz"
I: module-alsa-source.c: Using 8 fragments of size 1760 bytes.
I: alsa-util.c: All 2 channels can be mapped to mixer channels. Using hardware volume control.
D: module-alsa-source.c: Thread starting up
D: rtpoll.c: Acquired POSIX realtime signal SIGRTMIN+28
I: module.c: Loaded "module-alsa-source" (index: #1; argument: "device_id=0 source_name=alsa_input.pci_8086_24d5_sound_card_0_ alsa_capture_0").
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_8086_24d5_sound_card_0_alsa_control__1
I: module-hal-detect.c: Loaded 2 modules.
I: module.c: Loaded "module-hal-detect" (index: #2; argument: "").
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9/modules/module-esound-protocol-unix.so': success
I: module.c: Loaded "module-esound-protocol-unix" (index: #3; argument: "").
I: protocol-native.c: loading cookie from disk.
I: module.c: Loaded "module-native-protocol-unix" (index: #4; argument: "").
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9/modules/module-gconf.so': success
I: module.c: Loaded "module-gconf" (index: #5; argument: "").
I: module.c: Loaded "module-volume-restore" (index: #6; argument: "").
D: module-default-device-restore.c: Restored default sink 'alsa_output.pci_8086_24d5_sound_card_0_alsa_playb ack_0'.
D: core-subscribe.c: dropped redundant event.
D: module-default-device-restore.c: Restored default source 'alsa_input.pci_8086_24d5_sound_card_0_alsa_captur e_0'.
I: module.c: Loaded "module-default-device-restore" (index: #7; argument: "").
I: module.c: Loaded "module-rescue-streams" (index: #8; argument: "").
D: module-suspend-on-idle.c: Sink alsa_output.pci_8086_24d5_sound_card_0_alsa_playba ck_0 becomes idle.
D: module-suspend-on-idle.c: Source alsa_output.pci_8086_24d5_sound_card_0_alsa_playba ck_0.monitor becomes idle.
D: module-suspend-on-idle.c: Source alsa_input.pci_8086_24d5_sound_card_0_alsa_capture _0 becomes idle.
I: module.c: Loaded "module-suspend-on-idle" (index: #9; argument: "").
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9/modules/module-x11-publish.so': success
D: module-x11-publish.c: using already loaded auth cookie.
I: module.c: Loaded "module-x11-publish" (index: #10; argument: "").
I: main.c: Daemon startup complete.
D: module-hal-detect.c: dbus: interface=org.freedesktop.DBus, path=/org/freedesktop/DBus, member=NameAcquired
I: module-suspend-on-idle.c: Source alsa_input.pci_8086_24d5_sound_card_0_alsa_capture _0 idle for too long, suspending ...
I: module-alsa-source.c: Device suspended...
I: module-suspend-on-idle.c: Source alsa_output.pci_8086_24d5_sound_card_0_alsa_playba ck_0.monitor idle for too long, suspending ...
I: module-suspend-on-idle.c: Sink alsa_output.pci_8086_24d5_sound_card_0_alsa_playba ck_0 idle for too long, suspending ...
I: module-alsa-sink.c: Device suspended...


EDIT:
After waiting for a very long time I decided to CTRL+C the terminal. This was output when I did that:

^CI: main.c: Got signal SIGINT.
I: main.c: Exiting.
I: main.c: Daemon shutdown initiated.
I: module.c: Unloading "module-alsa-sink" (index: #0).
D: module-rescue-streams.c: No sink inputs to move away.
D: module-rescue-streams.c: No source outputs to move away.
D: module-alsa-sink.c: Thread shutting down
I: sink.c: Freeing sink 0 "alsa_output.pci_8086_24d5_sound_card_0_alsa_playba ck_0"
I: source.c: Freeing source 0 "alsa_output.pci_8086_24d5_sound_card_0_alsa_playba ck_0.monitor"
I: module.c: Unloaded "module-alsa-sink" (index: #0).
I: module.c: Unloading "module-alsa-source" (index: #1).
D: module-rescue-streams.c: No source outputs to move away.
D: module-alsa-source.c: Thread shutting down
I: source.c: Freeing source 1 "alsa_input.pci_8086_24d5_sound_card_0_alsa_capture _0"
I: module.c: Unloaded "module-alsa-source" (index: #1).
I: module.c: Unloading "module-hal-detect" (index: #2).
D: module-hal-detect.c: dbus: interface=org.freedesktop.DBus.Local, path=/org/freedesktop/DBus/Local, member=Disconnected
I: module.c: Unloaded "module-hal-detect" (index: #2).
I: module.c: Unloading "module-esound-protocol-unix" (index: #3).
I: module.c: Unloaded "module-esound-protocol-unix" (index: #3).
I: module.c: Unloading "module-native-protocol-unix" (index: #4).
I: module.c: Unloaded "module-native-protocol-unix" (index: #4).
I: module.c: Unloading "module-gconf" (index: #5).
I: module.c: Unloaded "module-gconf" (index: #5).
I: module.c: Unloading "module-volume-restore" (index: #6).
I: module.c: Unloaded "module-volume-restore" (index: #6).
I: module.c: Unloading "module-default-device-restore" (index: #7).
I: module.c: Unloaded "module-default-device-restore" (index: #7).
I: module.c: Unloading "module-rescue-streams" (index: #8).
I: module.c: Unloaded "module-rescue-streams" (index: #8).
I: module.c: Unloading "module-suspend-on-idle" (index: #9).
I: module.c: Unloaded "module-suspend-on-idle" (index: #9).
I: module.c: Unloading "module-x11-publish" (index: #10).
I: module.c: Unloaded "module-x11-publish" (index: #10).
I: main.c: Daemon terminated.

Lupusceleri
February 13th, 2010, 11:41 AM
/salute

Saved the day for me. Following part A on my Karmic install managed to repair Amarok for me. Kudos!

LearningPerson
February 14th, 2010, 07:02 PM
Hello, I seem to have the same problem as Everah:

I have an Averatec D1133 All-in-One with builtin speakers which play out of the back. The sound is fine on the Vista side of the partition but doesn't work anywhere on the Ubuntu side.

The Pulse Audio Volume Meter shows nothing in the 2 audio bars but it does note: “Showing signal levels of Alsa PCM on front: 0 (ALC662 Analog) via DMA".

Hardy Heron 8.04.4 (2.6.24-27) i686 (32 bit)

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

card 0: SB [HDA ATI SB], device 0: ALC662 Analog [ALC662 Analog]

Subdevices: 0/1

Subdevice #0: subdevice #0

card 1: HDMI [HDA ATI HDMI], device 3: ATI HDMI [ATI HDMI]

Subdevices: 1/1

Subdevice #0: subdevice #0



sharon@sharon-desktop:~$ pkill pulseaudio; sleep 2; pulseaudio -vv

I: main.c: Called SUID root and real-time/high-priority scheduling was requested in the configuration. However, we lack the necessary priviliges:

I: main.c: We are not in group 'pulse-rt' and PolicyKit refuse to grant us priviliges. Dropping SUID again.

I: main.c: For enabling real-time scheduling please acquire the appropriate PolicyKit priviliges, or become a member of 'pulse-rt', or increase the RLIMIT_NICE/RLIMIT_RTPRIO resource limits for this user.

I: main.c: Note that real-time/high-priority scheduling is NOT normally required. If you experience crackling or other sound anomalies, consider one or more of the above solutions.

I: main.c: High-priority scheduling enabled in configuration but now allowed by policy. Disabling forcibly.

I: main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted

I: main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation not permitted

I: main.c: This is PulseAudio 0.9.10

I: main.c: Page size is 4096 bytes

I: main.c: Fresh high-resolution timers available! Bon appetit!

D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9/modules/module-hal-detect.so': success

I: module-hal-detect.c: Trying capability alsa

D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/computer_alsa_timer

D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/computer_alsa_sequencer

D: module-hal-detect.c: Loading module-alsa-sink with arguments 'device_id=0 sink_name=alsa_output.pci_1002_4383_sound_card_0_a lsa_playback_0'

D: alsa-util.c: Trying front:0...

I: module-alsa-sink.c: Successfully opened device front:0.

I: module-alsa-sink.c: Successfully enabled mmap() mode.

ALSA lib control.c:909:(snd_ctl_open_noupdate) Invalid CTL front:0

I: alsa-util.c: Unable to attach to mixer front:0: No such file or directory

I: alsa-util.c: Successfully attached to mixer 'hw:0'

I: alsa-util.c: Using mixer control "Master".

I: sink.c: Created sink 0 "alsa_output.pci_1002_4383_sound_card_0_alsa_playba ck_0" with sample spec "s16le 2ch 44100Hz"

I: source.c: Created source 0 "alsa_output.pci_1002_4383_sound_card_0_alsa_playba ck_0.monitor" with sample spec "s16le 2ch 44100Hz"

I: module-alsa-sink.c: Using 4 fragments of size 4352 bytes.

I: alsa-util.c: ALSA device lacks independant volume controls for each channel, falling back to software volume control.

D: module-alsa-sink.c: Thread starting up

D: rtpoll.c: Acquired POSIX realtime signal SIGRTMIN+29

I: module-alsa-sink.c: Starting playback.

I: module.c: Loaded "module-alsa-sink" (index: #0; argument: "device_id=0 sink_name=alsa_output.pci_1002_4383_sound_card_0_a lsa_playback_0").

D: module-hal-detect.c: Loading module-alsa-source with arguments 'device_id=0 source_name=alsa_input.pci_1002_4383_sound_card_0_ alsa_capture_0'

D: alsa-util.c: Trying front:0...

I: module-alsa-source.c: Successfully opened device front:0.

I: module-alsa-source.c: Successfully enabled mmap() mode.

ALSA lib control.c:909:(snd_ctl_open_noupdate) Invalid CTL front:0

I: alsa-util.c: Unable to attach to mixer front:0: No such file or directory

I: alsa-util.c: Successfully attached to mixer 'hw:0'

I: alsa-util.c: Using mixer control "Capture".

I: source.c: Created source 1 "alsa_input.pci_1002_4383_sound_card_0_alsa_capture _0" with sample spec "s16le 2ch 44100Hz"

I: module-alsa-source.c: Using 4 fragments of size 4352 bytes.

I: alsa-util.c: All 2 channels can be mapped to mixer channels. Using hardware volume control.

D: module-alsa-source.c: Thread starting up

D: rtpoll.c: Acquired POSIX realtime signal SIGRTMIN+28

I: module.c: Loaded "module-alsa-source" (index: #1; argument: "device_id=0 source_name=alsa_input.pci_1002_4383_sound_card_0_ alsa_capture_0").

D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1002_4383_sound_card_0_alsa_hw_specific_0

D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1002_4383_sound_card_0_alsa_control__1

D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1002_960f_sound_card_0_alsa_playback_3

D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1002_960f_sound_card_0_alsa_hw_specific_0

D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1002_960f_sound_card_0_alsa_control__1

I: module-hal-detect.c: Loaded 2 modules.

I: module.c: Loaded "module-hal-detect" (index: #2; argument: "").

D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9/modules/module-esound-protocol-unix.so': success

I: module.c: Loaded "module-esound-protocol-unix" (index: #3; argument: "").

I: protocol-native.c: loading cookie from disk.

I: module.c: Loaded "module-native-protocol-unix" (index: #4; argument: "").

I: module.c: Loaded "module-volume-restore" (index: #5; argument: "").

D: module-default-device-restore.c: Restored default sink 'alsa_output.pci_1002_4383_sound_card_0_alsa_playb ack_0'.

D: core-subscribe.c: dropped redundant event.

D: module-default-device-restore.c: Restored default source 'alsa_input.pci_1002_4383_sound_card_0_alsa_captur e_0'.

I: module.c: Loaded "module-default-device-restore" (index: #6; argument: "").

I: module.c: Loaded "module-rescue-streams" (index: #7; argument: "").

D: module-suspend-on-idle.c: Sink alsa_output.pci_1002_4383_sound_card_0_alsa_playba ck_0 becomes idle.

D: module-suspend-on-idle.c: Source alsa_output.pci_1002_4383_sound_card_0_alsa_playba ck_0.monitor becomes idle.

D: module-suspend-on-idle.c: Source alsa_input.pci_1002_4383_sound_card_0_alsa_capture _0 becomes idle.

I: module.c: Loaded "module-suspend-on-idle" (index: #8; argument: "").

D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9/modules/module-gconf.so': success

D: module-gconf.c: Loading module 'module-native-protocol-tcp' with args '' due to GConf configuration.

I: protocol-native.c: using already loaded auth cookie.

I: protocol-native.c: using already loaded auth cookie.

I: module.c: Loaded "module-native-protocol-tcp" (index: #9; argument: "").

D: module-gconf.c: Loading module 'module-esound-protocol-tcp' with args '' due to GConf configuration.

I: module.c: Loaded "module-esound-protocol-tcp" (index: #10; argument: "").

I: module.c: Loaded "module-gconf" (index: #11; argument: "").

D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9/modules/module-x11-publish.so': success

D: module-x11-publish.c: using already loaded auth cookie.

I: module.c: Loaded "module-x11-publish" (index: #12; argument: "").

I: main.c: Daemon startup complete.

D: module-hal-detect.c: dbus: interface=org.freedesktop.DBus, path=/org/freedesktop/DBus, member=NameAcquired

I: module-suspend-on-idle.c: Sink alsa_output.pci_1002_4383_sound_card_0_alsa_playba ck_0 idle for too long, suspending ...

I: module-alsa-sink.c: Device suspended...

I: module-suspend-on-idle.c: Source alsa_input.pci_1002_4383_sound_card_0_alsa_capture _0 idle for too long, suspending ...

I: module-alsa-source.c: Device suspended...

I: module-suspend-on-idle.c: Source alsa_output.pci_1002_4383_sound_card_0_alsa_playba ck_0.monitor idle for too long, suspending ...


then it just stopped for 15 minutes. So I did Ctrl-C also and got this:
I: main.c: Got signal SIGINT.
I: main.c: Exiting.
I: main.c: Daemon shutdown initiated.
I: module.c: Unloading "module-alsa-sink" (index: #0).
D: module-rescue-streams.c: No sink inputs to move away.
D: module-rescue-streams.c: No source outputs to move away.
D: module-alsa-sink.c: Thread shutting down
I: sink.c: Freeing sink 0 "alsa_output.pci_1002_4383_sound_card_0_alsa_playba ck_0"
I: source.c: Freeing source 0 "alsa_output.pci_1002_4383_sound_card_0_alsa_playba ck_0.monitor"
I: module.c: Unloaded "module-alsa-sink" (index: #0).
I: module.c: Unloading "module-alsa-source" (index: #1).
D: module-rescue-streams.c: No source outputs to move away.
D: module-alsa-source.c: Thread shutting down
I: source.c: Freeing source 1 "alsa_input.pci_1002_4383_sound_card_0_alsa_capture _0"
I: module.c: Unloaded "module-alsa-source" (index: #1).
I: module.c: Unloading "module-hal-detect" (index: #2).
D: module-hal-detect.c: dbus: interface=org.freedesktop.DBus.Local, path=/org/freedesktop/DBus/Local, member=Disconnected
I: module.c: Unloaded "module-hal-detect" (index: #2).
I: module.c: Unloading "module-esound-protocol-unix" (index: #3).
I: module.c: Unloaded "module-esound-protocol-unix" (index: #3).
I: module.c: Unloading "module-native-protocol-unix" (index: #4).
I: module.c: Unloaded "module-native-protocol-unix" (index: #4).
I: module.c: Unloading "module-volume-restore" (index: #5).
I: module.c: Unloaded "module-volume-restore" (index: #5).
I: module.c: Unloading "module-default-device-restore" (index: #6).
I: module.c: Unloaded "module-default-device-restore" (index: #6).
I: module.c: Unloading "module-rescue-streams" (index: #7).
I: module.c: Unloaded "module-rescue-streams" (index: #7).
I: module.c: Unloading "module-suspend-on-idle" (index: #8).
I: module.c: Unloaded "module-suspend-on-idle" (index: #8).
I: module.c: Unloading "module-native-protocol-tcp" (index: #9).
I: module.c: Unloaded "module-native-protocol-tcp" (index: #9).
I: module.c: Unloading "module-esound-protocol-tcp" (index: #10).
I: module.c: Unloaded "module-esound-protocol-tcp" (index: #10).
I: module.c: Unloading "module-gconf" (index: #11).
D: module-gconf.c: Unloading module #9
D: module-gconf.c: Unloading module #10
I: module.c: Unloaded "module-gconf" (index: #11).
I: module.c: Unloading "module-x11-publish" (index: #12).
I: module.c: Unloaded "module-x11-publish" (index: #12).
I: main.c: Daemon terminated.


Any help would be appreciated,

LearningPerson

amalgamas
March 7th, 2010, 11:25 AM
Thanks! Part A fixed my issue with no Flash video sound in Ubuntu Studio 9.10 :p

luh3417
March 8th, 2010, 03:01 PM
I am running Karmic Koala 2.6.31-20-generic. alsa-lib-1.0.22 alsa-driver-1.0.22.1 pulseaudio 0.9.19.
AMD X64 quadcore Opteron, Athlon64, Sempron
Audio is onboard ATI HDA INTEL drivers

I use my digital audio via optic cable.

Everything works except for browser sound. I get no browser audio in digital and I used to once upon a time. If I switch to analogue I can get browser audio

I followed your howto for Karmic and still could not get digital audio in browser everything else was fine. In fact installing pavaudio crashed my spdif iec drivers. I had to remove pavaudio and reinstall and reload drivers to get all my usual audio digital drivers back.

Seems that pulse sits on Alsa a little bit like the scorpion and the frog at times.

My outputs as requested are below. More than happy to send any more info. Hope this can be resolved.

Thanks for all your work and time and effort here. It is truly appreciated.

aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: SB [HDA ATI SB], device 0: VT1708S Analog [VT1708S Analog]
Subdevices: 2/2
Subdevice #0: subdevice #0
Subdevice #1: subdevice #1
card 0: SB [HDA ATI SB], device 1: VT1708S Digital [VT1708S Digital]
Subdevices: 0/1
Subdevice #0: subdevice #0
card 1: HDMI [HDA ATI HDMI], device 3: ATI HDMI [ATI HDMI]
Subdevices: 1/1
Subdevice #0: subdevice #0

If you want the pulseaudio -vv I can email it Both v and vv were too large.

lavezarez
March 11th, 2010, 10:58 PM
Thank you for this post!

Running on Karmic, after following Part A of your guide, still got no sound so I followed your Appendix A, using VLC. Result was -
The application does not play audio and does list an entry in the Playback tab I'm pretty sure the PCM isn't muted at this point.

Here is the output:
$ aplay -l

**** List of PLAYBACK Hardware Devices ****
card 0: Intel [HDA Intel], device 0: ALC861-VD Analog [ALC861-VD Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: Intel [HDA Intel], device 6: Si3054 Modem [Si3054 Modem]
Subdevices: 0/1
Subdevice #0: subdevice #0

$ pkill pulseaudio; sleep 2; pulseaudio -vv

I: main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted
I: main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation not permitted
D: core-rtclock.c: Timer slack is set to 50 us.
I: core-util.c: Failed to acquire high-priority scheduling: No such file or directory
I: main.c: This is PulseAudio 0.9.19
D: main.c: Compilation host: i486-pc-linux-gnu
D: main.c: Compilation CFLAGS: -g -O2 -g -Wall -O3 -Wall -W -Wextra -pipe -Wno-long-long -Winline -Wvla -Wno-overlength-strings -Wunsafe-loop-optimizations -Wundef -Wformat=2 -Wlogical-op -Wsign-compare -Wformat-security -Wmissing-include-dirs -Wformat-nonliteral -Wold-style-definition -Wpointer-arith -Winit-self -Wdeclaration-after-statement -Wfloat-equal -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wmissing-declarations -Wmissing-noreturn -Wshadow -Wendif-labels -Wcast-align -Wstrict-aliasing=2 -Wwrite-strings -Wno-unused-parameter -ffast-math -Wp,-D_FORTIFY_SOURCE=2 -fno-common -fdiagnostics-show-option
D: main.c: Running on host: Linux i686 2.6.31-14-generic #48-Ubuntu SMP Fri Oct 16 14:04:26 UTC 2009
D: main.c: Found 2 CPUs.
I: main.c: Page size is 4096 bytes
D: main.c: Compiled with Valgrind support: no
D: main.c: Running in valgrind mode: no
D: main.c: Optimized build: yes
D: main.c: All asserts enabled.
I: main.c: Machine ID is eaee919c9eaef9635fb339eb4b7f45de.
I: main.c: Session ID is eaee919c9eaef9635fb339eb4b7f45de-1268342704.935924-1244048761.
I: main.c: Using runtime directory /home/dan/.pulse/eaee919c9eaef9635fb339eb4b7f45de-runtime.
I: main.c: Using state directory /home/dan/.pulse.
I: main.c: Using modules directory /usr/lib/pulse-0.9.19/modules.
I: main.c: Running in system mode: no
I: main.c: Fresh high-resolution timers available! Bon appetit!
I: cpu-x86.c: CPU flags: MMX SSE SSE2 SSE3
I: svolume_mmx.c: Initialising MMX optimized functions.
I: remap_mmx.c: Initialising MMX optimized remappers.
I: svolume_sse.c: Initialising SSE2 optimized functions.
I: remap_sse.c: Initialising SSE2 optimized remappers.
I: sconv_sse.c: Initialising SSE2 optimized conversions.
D: memblock.c: Using shared memory pool with 1024 slots of size 64.0 KiB each, total size is 64.0 MiB, maximum usable slot size is 65496
D: database-tdb.c: Opened TDB database '/home/dan/.pulse/eaee919c9eaef9635fb339eb4b7f45de-device-volumes.tdb'
I: module-device-restore.c: Sucessfully opened database file '/home/dan/.pulse/eaee919c9eaef9635fb339eb4b7f45de-device-volumes'.
I: module.c: Loaded "module-device-restore" (index: #0; argument: "").
D: database-tdb.c: Opened TDB database '/home/dan/.pulse/eaee919c9eaef9635fb339eb4b7f45de-stream-volumes.tdb'
I: module-stream-restore.c: Sucessfully opened database file '/home/dan/.pulse/eaee919c9eaef9635fb339eb4b7f45de-stream-volumes'.
I: module.c: Loaded "module-stream-restore" (index: #1; argument: "").
D: database-tdb.c: Opened TDB database '/home/dan/.pulse/eaee919c9eaef9635fb339eb4b7f45de-card-database.tdb'
I: module-card-restore.c: Sucessfully opened database file '/home/dan/.pulse/eaee919c9eaef9635fb339eb4b7f45de-card-database'.
I: module.c: Loaded "module-card-restore" (index: #2; argument: "").
I: module.c: Loaded "module-augment-properties" (index: #3; argument: "").
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.19/modules/module-udev-detect.so': success
D: module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
D: module-udev-detect.c: /devices/pci0000:00/0000:00:1b.0/sound/card0 is busy: yes
I: module-udev-detect.c: Found 1 cards.
I: module.c: Loaded "module-udev-detect" (index: #4; argument: "").
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.19/modules/module-bluetooth-discover.so': failure
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.19/modules/module-esound-protocol-unix.so': success
I: module.c: Loaded "module-esound-protocol-unix" (index: #5; argument: "").
I: module.c: Loaded "module-native-protocol-unix" (index: #6; argument: "").
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.19/modules/module-gconf.so': success
D: module-gconf.c: Loading module 'module-combine' with args '' due to GConf configuration.
I: module-device-restore.c: Restoring volume for sink combined.
I: module-device-restore.c: Restoring mute state for sink combined.
I: sink.c: Created sink 0 "combined" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: sink.c: device.class = "filter"
I: sink.c: device.description = "Simultaneous Output"
I: sink.c: device.icon_name = "audio-card"
D: core-subscribe.c: Dropped redundant event due to change event.
I: source.c: Created source 0 "combined.monitor" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: source.c: device.description = "Monitor of Simultaneous Output"
I: source.c: device.class = "monitor"
I: source.c: device.icon_name = "audio-input-microphone"
D: module-combine.c: Thread starting up
I: core-util.c: Successfully enabled SCHED_RR scheduling for thread, with priority 5, which is lower than the requested 6.
I: module.c: Loaded "module-combine" (index: #7; argument: "").
D: module-gconf.c: Loading module 'module-null-sink' with args 'sink_name=rtp format=s16be channels=2 rate=44100 sink_properties="device.description='RTP Multicast' device.bus='network' device.icon_name='network-server'"' due to GConf configuration.
D: core-subscribe.c: Dropped redundant event due to change event.
I: sink.c: Created sink 1 "rtp" with sample spec s16be 2ch 44100Hz and channel map front-left,front-right
I: sink.c: device.description = "RTP Multicast"
I: sink.c: device.class = "abstract"
I: sink.c: device.bus = "network"
I: sink.c: device.icon_name = "network-server"
D: core-subscribe.c: Dropped redundant event due to change event.
I: source.c: Created source 1 "rtp.monitor" with sample spec s16be 2ch 44100Hz and channel map front-left,front-right
I: source.c: device.description = "Monitor of RTP Multicast"
I: source.c: device.class = "monitor"
I: source.c: device.icon_name = "audio-input-microphone"
D: module-null-sink.c: Thread starting up
I: module.c: Loaded "module-null-sink" (index: #8; argument: "sink_name=rtp format=s16be channels=2 rate=44100 sink_properties="device.description='RTP Multicast' device.bus='network' device.icon_name='network-server'"").
D: module-gconf.c: Loading module 'module-rtp-send' with args 'source=rtp.monitor loop=0' due to GConf configuration.
D: module-stream-restore.c: Not restoring device for stream source-output-by-media-name:RTP Monitor Stream, because already set
D: memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0
D: memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0
I: source-output.c: Created output 0 "RTP Monitor Stream" on rtp.monitor with sample spec s16be 2ch 44100Hz and channel map front-left,front-right
I: source-output.c: media.name = "RTP Monitor Stream"
I: source-output.c: rtp.destination = "224.0.0.56"
I: source-output.c: rtp.mtu = "1280"
I: source-output.c: rtp.port = "46552"
I: source-output.c: rtp.ttl = "1"
I: source-output.c: module-stream-restore.id = "source-output-by-media-name:RTP Monitor Stream"
I: module-rtp-send.c: Configured source latency of 7 ms.
D: memblockq.c: memblockq requested: maxlength=174080, tlength=174080, base=4, prebuf=1, minreq=0 maxrewind=0
D: memblockq.c: memblockq sanitized: maxlength=174080, tlength=174080, base=4, prebuf=4, minreq=4 maxrewind=0
I: module-rtp-send.c: RTP stream initialized with mtu 1280 on 224.0.0.56:46552 ttl=1, SSRC=0xaa95cb91, payload=10, initial sequence #51620
I: module-rtp-send.c: SDP-Data:
I: module-rtp-send.c: v=0
I: module-rtp-send.c: o=dan 3477332199 0 IN IP4 10.42.43.20
I: module-rtp-send.c: s=PulseAudio RTP Stream on dan-laptop
I: module-rtp-send.c: c=IN IP4 224.0.0.56
I: module-rtp-send.c: t=3477332199 0
I: module-rtp-send.c: a=recvonly
I: module-rtp-send.c: m=audio 46552 RTP/AVP 10
I: module-rtp-send.c: a=rtpmap:10 L16/44100/2
I: module-rtp-send.c: a=type:broadcast
I: module-rtp-send.c: EOF
D: core-subscribe.c: Dropped redundant event due to change event.
I: module.c: Loaded "module-rtp-send" (index: #9; argument: "source=rtp.monitor loop=0").
D: module-gconf.c: Loading module 'module-raop-discover' with args '' due to GConf configuration.
E: module.c: Failed to open module "module-raop-discover": file not found
E: module-gconf.c: pa_module_load() failed
D: module-gconf.c: Loading module 'module-native-protocol-tcp' with args '' due to GConf configuration.
I: module.c: Loaded "module-native-protocol-tcp" (index: #10; argument: "").
D: module-gconf.c: Loading module 'module-esound-protocol-tcp' with args '' due to GConf configuration.
I: module.c: Loaded "module-esound-protocol-tcp" (index: #11; argument: "").
D: module-gconf.c: Loading module 'module-rtp-recv' with args '' due to GConf configuration.
I: module.c: Loaded "module-rtp-recv" (index: #12; argument: "").
D: module-gconf.c: Loading module 'module-zeroconf-discover' with args '' due to GConf configuration.
I: module.c: Loaded "module-zeroconf-discover" (index: #13; argument: "").
D: module-gconf.c: Loading module 'module-rygel-media-server' with args '' due to GConf configuration.
E: module.c: Failed to open module "module-rygel-media-server": file not found
E: module-gconf.c: pa_module_load() failed
I: module.c: Loaded "module-gconf" (index: #14; argument: "").
D: core-subscribe.c: Dropped redundant event due to change event.
I: module-default-device-restore.c: Restored default sink 'combined'.
D: core-subscribe.c: Dropped redundant event due to change event.
I: module-default-device-restore.c: Restored default source 'combined.monitor'.
I: module.c: Loaded "module-default-device-restore" (index: #15; argument: "").
I: module.c: Loaded "module-rescue-streams" (index: #16; argument: "").
I: module.c: Loaded "module-always-sink" (index: #17; argument: "").
I: module.c: Loaded "module-intended-roles" (index: #18; argument: "").
D: module-suspend-on-idle.c: Sink combined becomes idle, timeout in 5 seconds.
D: module-suspend-on-idle.c: Sink rtp becomes idle, timeout in 5 seconds.
I: module.c: Loaded "module-suspend-on-idle" (index: #19; argument: "").
D: dbus-util.c: Successfully connected to D-Bus system bus adfe4c971ad0354b6ce782f44b995f92 as :1.63
I: client.c: Created 0 "ConsoleKit Session /org/freedesktop/ConsoleKit/Session2"
D: module-console-kit.c: Added new session /org/freedesktop/ConsoleKit/Session2
I: module.c: Loaded "module-console-kit" (index: #20; argument: "").
I: module.c: Loaded "module-position-event-sounds" (index: #21; argument: "").
D: dbus-util.c: Successfully connected to D-Bus session bus d24aef866da1f33699c3853f4b995fb1 as :1.92
D: main.c: Got org.pulseaudio.Server!
I: main.c: Daemon startup complete.
D: module-console-kit.c: dbus: interface=org.freedesktop.DBus, path=/org/freedesktop/DBus, member=NameAcquired
I: client.c: Created 1 "Native client (UNIX socket client)"
D: protocol-native.c: Protocol version: remote 16, local 16
I: protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
D: protocol-native.c: SHM possible: yes
D: protocol-native.c: Negotiated SHM: yes
D: module-augment-properties.c: Looking for .desktop file for gnome-settings-daemon
I: client.c: Created 2 "Native client (UNIX socket client)"
D: protocol-native.c: Protocol version: remote 16, local 16
I: protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
D: protocol-native.c: SHM possible: yes
D: protocol-native.c: Negotiated SHM: yes
D: module-augment-properties.c: Looking for .desktop file for gnome-volume-control-applet
I: module-suspend-on-idle.c: Sink rtp idle for too long, suspending ...
D: sink.c: Suspend cause of sink rtp is 0x0004, suspending
I: module-suspend-on-idle.c: Sink combined idle for too long, suspending ...
D: sink.c: Suspend cause of sink combined is 0x0004, suspending
I: module-combine.c: Device suspended...
D: module-rtp-recv.c: Checking for dead streams ...
D: module-rtp-recv.c: Checking for dead streams ...
D: module-rtp-recv.c: Checking for dead streams ...
D: module-rtp-recv.c: Checking for dead streams ...
D: module-rtp-recv.c: Checking for dead streams ...
D: module-rtp-recv.c: Checking for dead streams ...

Pressed CTRL-C and got these messages:

D: module-rtp-recv.c: Checking for dead streams ...
^CI: main.c: Got signal SIGINT.
I: main.c: Exiting.
I: main.c: Daemon shutdown initiated.
I: module.c: Unloading "module-device-restore" (index: #0).
I: module.c: Unloaded "module-device-restore" (index: #0).
I: module.c: Unloading "module-stream-restore" (index: #1).
I: module.c: Unloaded "module-stream-restore" (index: #1).
I: module.c: Unloading "module-card-restore" (index: #2).
I: module.c: Unloaded "module-card-restore" (index: #2).
I: module.c: Unloading "module-augment-properties" (index: #3).
I: module.c: Unloaded "module-augment-properties" (index: #3).
I: module.c: Unloading "module-udev-detect" (index: #4).
I: module.c: Unloaded "module-udev-detect" (index: #4).
I: module.c: Unloading "module-esound-protocol-unix" (index: #5).
I: module.c: Unloaded "module-esound-protocol-unix" (index: #5).
I: module.c: Unloading "module-native-protocol-unix" (index: #6).
I: client.c: Freed 1 "GNOME Volume Control Media Keys"
I: client.c: Freed 2 "GNOME Volume Control Applet"
I: module.c: Unloaded "module-native-protocol-unix" (index: #6).
I: module.c: Unloading "module-combine" (index: #7).
D: core-subscribe.c: Dropped redundant event due to change event.
D: module-combine.c: Thread shutting down
I: sink.c: Freeing sink 0 "combined"
I: source.c: Freeing source 0 "combined.monitor"
I: module.c: Unloaded "module-combine" (index: #7).
I: module.c: Unloading "module-null-sink" (index: #8).
D: module-suspend-on-idle.c: Sink rtp becomes idle, timeout in 5 seconds.
D: core.c: Hmm, no streams around, trying to vacuum.
I: source-output.c: Freeing output 0 "RTP Monitor Stream"
D: module-null-sink.c: Thread shutting down
I: sink.c: Freeing sink 1 "rtp"
I: source.c: Freeing source 1 "rtp.monitor"
I: module.c: Unloaded "module-null-sink" (index: #8).
I: module.c: Unloading "module-rtp-send" (index: #9).
I: module.c: Unloaded "module-rtp-send" (index: #9).
I: module.c: Unloading "module-native-protocol-tcp" (index: #10).
I: module.c: Unloaded "module-native-protocol-tcp" (index: #10).
I: module.c: Unloading "module-esound-protocol-tcp" (index: #11).
I: module.c: Unloaded "module-esound-protocol-tcp" (index: #11).
I: module.c: Unloading "module-rtp-recv" (index: #12).
I: module.c: Unloaded "module-rtp-recv" (index: #12).
I: module.c: Unloading "module-zeroconf-discover" (index: #13).
I: module.c: Unloaded "module-zeroconf-discover" (index: #13).
I: module.c: Unloading "module-gconf" (index: #14).
D: module-gconf.c: Unloading module #7
D: module-gconf.c: Unloading module #8
D: module-gconf.c: Unloading module #9
D: module-gconf.c: Unloading module #10
D: module-gconf.c: Unloading module #11
D: module-gconf.c: Unloading module #12
D: module-gconf.c: Unloading module #13
I: module.c: Unloaded "module-gconf" (index: #14).
I: module.c: Unloading "module-default-device-restore" (index: #15).
I: module.c: Unloaded "module-default-device-restore" (index: #15).
I: module.c: Unloading "module-rescue-streams" (index: #16).
I: module.c: Unloaded "module-rescue-streams" (index: #16).
I: module.c: Unloading "module-always-sink" (index: #17).
I: module.c: Unloaded "module-always-sink" (index: #17).
I: module.c: Unloading "module-intended-roles" (index: #18).
I: module.c: Unloaded "module-intended-roles" (index: #18).
I: module.c: Unloading "module-suspend-on-idle" (index: #19).
I: module.c: Unloaded "module-suspend-on-idle" (index: #19).
I: module.c: Unloading "module-console-kit" (index: #20).
D: module-console-kit.c: Removing session /org/freedesktop/ConsoleKit/Session2
I: client.c: Freed 0 "ConsoleKit Session /org/freedesktop/ConsoleKit/Session2"
I: module.c: Unloaded "module-console-kit" (index: #20).
I: module.c: Unloading "module-position-event-sounds" (index: #21).
I: module.c: Unloaded "module-position-event-sounds" (index: #21).
I: main.c: Daemon terminated.

MobiusJedi
March 13th, 2010, 12:10 AM
Hi. My system is karmic amd (athlon xp, upgraded from intrepid) and my soundcard is a turtle beach santa cruz (if that matters). Following the guide and messing with pulse applet settings, I have sound, but it distorts.


**** List of PLAYBACK Hardware Devices ****
card 0: CS46xx [Sound Fusion CS46xx], device 0: CS46xx [CS46xx]
Subdevices: 31/31
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
card 0: CS46xx [Sound Fusion CS46xx], device 1: CS46xx - Rear [CS46xx - Rear]
Subdevices: 31/31
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
card 0: CS46xx [Sound Fusion CS46xx], device 2: CS46xx - IEC958 [CS46xx - IEC958]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: CS46xx [Sound Fusion CS46xx], device 3: CS46xx - Center LFE [CS46xx - Center LFE]
Subdevices: 31/31
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

verbosity:


I: main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted
I: main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation not permitted
D: core-rtclock.c: Timer slack is set to 50 us.
I: core-util.c: Failed to acquire high-priority scheduling: No such file or directory
I: main.c: This is PulseAudio 0.9.19
D: main.c: Compilation host: i486-pc-linux-gnu
D: main.c: Compilation CFLAGS: -g -O2 -g -Wall -O3 -Wall -W -Wextra -pipe -Wno-long-long -Winline -Wvla -Wno-overlength-strings -Wunsafe-loop-optimizations -Wundef -Wformat=2 -Wlogical-op -Wsign-compare -Wformat-security -Wmissing-include-dirs -Wformat-nonliteral -Wold-style-definition -Wpointer-arith -Winit-self -Wdeclaration-after-statement -Wfloat-equal -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wmissing-declarations -Wmissing-noreturn -Wshadow -Wendif-labels -Wcast-align -Wstrict-aliasing=2 -Wwrite-strings -Wno-unused-parameter -ffast-math -Wp,-D_FORTIFY_SOURCE=2 -fno-common -fdiagnostics-show-option
D: main.c: Running on host: Linux i686 2.6.31-20-generic #57-Ubuntu SMP Mon Feb 8 09:05:19 UTC 2010
D: main.c: Found 1 CPUs.
I: main.c: Page size is 4096 bytes
D: main.c: Compiled with Valgrind support: no
D: main.c: Running in valgrind mode: no
D: main.c: Optimized build: yes
D: main.c: All asserts enabled.
I: main.c: Machine ID is 5484f7715b59f843458ce34e4b9612cb.
I: main.c: Session ID is 5484f7715b59f843458ce34e4b9612cb-1268434545.71064-955087267.
I: main.c: Using runtime directory /home/mobius/.pulse/5484f7715b59f843458ce34e4b9612cb-runtime.
I: main.c: Using state directory /home/mobius/.pulse.
I: main.c: Using modules directory /usr/lib/pulse-0.9.19/modules.
I: main.c: Running in system mode: no
E: pid.c: Daemon already running.
E: main.c: pa_pid_file_create() failed.

buntunub
March 13th, 2010, 07:13 AM
I too am having some issues with pulse on Kubuntu Karmic after upgrading to the latest and greatest KDE4.4.1 ppa. I discovered that it had something to do with the upgrade that broke the existing previously perfectly working config files which are in ~./pulse. I simply moved all the files contained therein into a backup folder and logged out then back in and voila! Perfectly working pulseaudio once again. The system simply recreated all the appropriate config files once more (default.pa, etc.).

andy_spoo
March 16th, 2010, 07:40 PM
Since using Karmic (worked in every other version of Ubuntu I've used) my sound has become very 'grainy' / 'bitty'.

The weird thing is, if I use 'Audacity' the sound is the same, until I decrease the volume a little (in Audacity). If I put the volume back to there it was it still works perfectly. If I run another sample in Audacity it goes back to being grainy, until I move the volume down and back up again!!

Obviously Audacity is changing some parameter that my Karmic really likes and needs.

Any ideas???

Of course it goes without saying that if I adjust the volume in Gnome, that changes absolutely nothing; still grainy as hell.

Like I said, weird!

doobiest
March 23rd, 2010, 12:43 AM
I have no sound on 9.10 or 10.04. I do through live CD. Sound used to work on 9.10 and then just stopped one day.

I believe this is a system level issue not user level. I can log in as a new user with no luck.

Most importantly my issues is that pulseaudio is outputting sound but nothing to the speaker.

If I am to open padevchooser and pick Volume meter (playback) it is showing sound.

What could be keeping PA from outputting to my soundcard. All the GUI based settings are correct and wont output to either card.

How can I restore the default PA settings? If I remove the pa related files under /etc/pulse or /etc/default it just causes errors. I would like to know how to remove those configs and have ubuntu recreate the defaults.


doobiest@LinuxBox:/var/log$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: NVidia [HDA NVidia], device 0: ALC880 Analog [ALC880 Analog]
Subdevices: 0/1
Subdevice #0: subdevice #0
card 0: NVidia [HDA NVidia], device 1: ALC880 Digital [ALC880 Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: HDMI [HDA ATI HDMI], device 3: ATI HDMI [ATI HDMI]
Subdevices: 1/1
Subdevice #0: subdevice #0

doobiest
March 23rd, 2010, 01:41 AM
pkill -9 pulse; /usr/bin/pulseaudio --start --log-target=syslog -vvv

-Mar 22 20:39:42 LinuxBox pulseaudio[5147]: main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted
Mar 22 20:39:42 LinuxBox pulseaudio[5147]: core-rtclock.c: Timer slack is set to 50 us.
Mar 22 20:39:42 LinuxBox pulseaudio[5147]: core-util.c: RealtimeKit worked.
Mar 22 20:39:42 LinuxBox pulseaudio[5147]: core-util.c: Successfully gained nice level -11.
Mar 22 20:39:42 LinuxBox pulseaudio[5147]: main.c: This is PulseAudio 0.9.21-63-gd3efa-dirty
Mar 22 20:39:42 LinuxBox pulseaudio[5147]: main.c: Compilation host: x86_64-pc-linux-gnu
Mar 22 20:39:42 LinuxBox pulseaudio[5147]: main.c: Compilation CFLAGS: -g -O2 -g -Wall -O3 -Wall -W -Wextra -pipe -Wno-long-long -Winline -Wvla -Wno-overlength-strings -Wunsafe-loop-optimizations -Wundef -Wformat=2 -Wlogical-op -Wsign-compare -Wformat-security -Wmissing-include-dirs -Wformat-nonliteral -Wold-style-definition -Wpointer-arith -Winit-self -Wdeclaration-after-statement -Wfloat-equal -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wmissing-declarations -Wmissing-noreturn -Wshadow -Wendif-labels -Wcast-align -Wstrict-aliasing=2 -Wwrite-strings -Wno-unused-parameter -ffast-math -Wp,-D_FORTIFY_SOURCE=2 -fno-common -fdiagnostics-show-option
Mar 22 20:39:42 LinuxBox pulseaudio[5147]: main.c: Running on host: Linux x86_64 2.6.32-16-generic #25-Ubuntu SMP Tue Mar 9 16:33:12 UTC 2010
Mar 22 20:39:42 LinuxBox pulseaudio[5147]: main.c: Found 2 CPUs.
Mar 22 20:39:42 LinuxBox pulseaudio[5147]: main.c: Page size is 4096 bytes
Mar 22 20:39:42 LinuxBox pulseaudio[5147]: main.c: Compiled with Valgrind support: no
Mar 22 20:39:42 LinuxBox pulseaudio[5147]: main.c: Running in valgrind mode: no
Mar 22 20:39:42 LinuxBox pulseaudio[5147]: main.c: Running in VM: no
Mar 22 20:39:42 LinuxBox pulseaudio[5147]: main.c: Optimized build: yes
Mar 22 20:39:42 LinuxBox pulseaudio[5147]: main.c: All asserts enabled.
Mar 22 20:39:42 LinuxBox pulseaudio[5147]: main.c: Machine ID is 89b904f58f9eee542bf8db00481fdf98.
Mar 22 20:39:42 LinuxBox pulseaudio[5147]: main.c: Session ID is 89b904f58f9eee542bf8db00481fdf98-1269303979.118591-324768525.
Mar 22 20:39:42 LinuxBox pulseaudio[5147]: main.c: Using runtime directory /home/doobiest/.pulse/89b904f58f9eee542bf8db00481fdf98-runtime.
Mar 22 20:39:42 LinuxBox pulseaudio[5147]: main.c: Using state directory /home/doobiest/.pulse.
Mar 22 20:39:42 LinuxBox pulseaudio[5147]: main.c: Using modules directory /usr/lib/pulse-0.9.21/modules.
Mar 22 20:39:42 LinuxBox pulseaudio[5147]: main.c: Running in system mode: no
Mar 22 20:39:42 LinuxBox pulseaudio[5147]: pid.c: Daemon already running.
Mar 22 20:39:42 LinuxBox pulseaudio[5145]: main.c: Daemon startup successful.
Mar 22 20:39:47 LinuxBox pulseaudio[5150]: pid.c: Stale PID file, overwriting.


I've run a diff against a working computer with 9.10 for /etc/pulse* and /etc/default/pulseaudio, no difference.

I'm wondering if something isnt loading correctly and its not getting logged somewhere that I'm seeing

dealcorn
March 24th, 2010, 01:41 AM
I can not complete HOWTO Part A: Step 5 after a clean install of karmic- AMD64 followed by updates. The issue relates to saa7134 configuration?


~$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: Intel [HDA Intel], device 0: VT1708B Analog [VT1708B Analog]
Subdevices: 2/2
Subdevice #0: subdevice #0
Subdevice #1: subdevice #1
card 0: Intel [HDA Intel], device 1: VT1708B Digital [VT1708B Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0





pulseaudio & pavucontrol

~$ pulseaudio & pavucontrol
[1] 6961
W: alsa-mixer.c: Unable to load mixer: Invalid argument
W: alsa-mixer.c: Unable to load mixer: Invalid argument
E: socket-server.c: bind(): Address already in use
E: module.c: Failed to load module "module-esound-protocol-unix" (argument: ""): initialization failed.
E: main.c: Module load failed.
E: main.c: Failed to initialize daemon.

(pavucontrol:6962): Gtk-CRITICAL **: gtk_main_quit: assertion `main_loops != NULL' failed

The attached file contains pkill output.

I think the issue relates to saa7134 because it was first noted in tvtime and I am troubled by the IRQF_DISABED in the following output"


~$ dmesg | grep saa7134
[ 11.175781] saa7134 0000:03:04.0: PCI INT A -> GSI 16 (level, low) -> IRQ 16
[ 11.175789] saa7134[0]: found at 0000:03:04.0, rev: 1, irq: 16, latency: 64, mmio: 0xfebffc00
[ 11.175795] saa7134[0]: subsystem: 1a7f:2208, board: Encore ENLTV-FM v5.3 [card=148,insmod option]
[ 11.175839] saa7134[0]: board init: gpio is 41000
[ 11.175915] input: saa7134 IR (Encore ENLTV-FM v5. as /devices/pci0000:00/0000:00:1e.0/0000:03:04.0/input/input4
[ 11.175966] IRQ 16/saa7134[0]: IRQF_DISABLED is not guaranteed on shared IRQs
[ 11.350249] saa7134[0]: i2c eeprom 00: 7f 1a 08 22 54 20 1c 00 43 43 a9 1c 55 d2 b2 92
[ 11.350257] saa7134[0]: i2c eeprom 10: 00 ff ff 0f ff 20 ff ff ff ff ff ff ff ff ff ff
[ 11.350264] saa7134[0]: i2c eeprom 20: 01 40 01 02 03 ff 01 03 08 ff 00 29 ff ff ff ff
[ 11.350271] saa7134[0]: i2c eeprom 30: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff
[ 11.350277] saa7134[0]: i2c eeprom 40: 50 90 00 c0 84 ff 03 30 00 05 ff ff ff ff ff ff
[ 11.350284] saa7134[0]: i2c eeprom 50: ff ff ff ff ff ff ff 8c 84 ff 31 30 4d 4f 4f 4e
[ 11.350291] saa7134[0]: i2c eeprom 60: 53 50 44 41 31 30 30 ff 41 ff ff ff ff ff ff ff
[ 11.350562] saa7134[0]: i2c eeprom 70: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff
[ 11.350569] saa7134[0]: i2c eeprom 80: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff
[ 11.350575] saa7134[0]: i2c eeprom 90: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff
[ 11.350582] saa7134[0]: i2c eeprom a0: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff
[ 11.350589] saa7134[0]: i2c eeprom b0: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff
[ 11.350595] saa7134[0]: i2c eeprom c0: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff
[ 11.350602] saa7134[0]: i2c eeprom d0: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff
[ 11.350608] saa7134[0]: i2c eeprom e0: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff
[ 11.350615] saa7134[0]: i2c eeprom f0: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff
[ 11.370181] tuner 1-0061: chip found @ 0xc2 (saa7134[0])
[ 11.451357] saa7134[0]: registered device video0 [v4l2]
[ 11.451375] saa7134[0]: registered device vbi0
[ 11.451609] saa7134[0]: registered device radio0
[ 11.454983] saa7134 ALSA driver for DMA sound loaded
[ 11.454996] IRQ 16/saa7134[0]: IRQF_DISABLED is not guaranteed on shared IRQs
[ 11.455015] saa7134[0]/alsa: saa7134[0] at 0xfebffc00 irq 16 registered as card -2

sendhil76
March 29th, 2010, 10:17 AM
Thanks for the excellent HOWTO on sound configuration.

For quite some time i was having issues with sound under 8.04 LTS...now everything is working as it should :)

Cheers!

movil
April 6th, 2010, 06:09 PM
Good guide thanks!

I'm on hardy 64 bits.

Do you guys know if this getlibs-all is anywhere else ?
> wget http://www.boundlesssupremacy.com/Cappy/getlibs/getlibs-all.deb

It seems the webmaster removed entire directory:
http://www.boundlesssupremacy.com/Cappy/getlibs

Mart_L
April 7th, 2010, 03:03 PM
Hi, and first of all thanks for a great post.

I've got serious problems here, I've been trying a lot of recommendations out and I still haven't got any solution.

Now my problem is this, I cannot hear anything when playing videos in firefox, ie. youtube videos, etc. nonetheless I can playback music in any other environment, rhytmbox works perfectly!
This is a new thing, and I'm worried that this happened as I upgraded two weeks ago.

Now as I've said, I've seen there are a lot of other persons having similar problems, but the solutions working for them, haven't worken for me.

Similarly, looking through the settings, I've seen that adobe flashplayer 9 plugin is installed, I've tried to uninstall it and installed the newest, which already should be included and no difference, if I right-click on the video, it still appears as if it was the 9th.. really strange!

I don't know if this has anything to do with it, but after trying a lot of the solutions mentioned, I'm really running out of patience, please somebody help me out!!

Mart_L
April 7th, 2010, 03:39 PM
Hi, and first of all thanks for a great post.

I've got serious problems here, I've been trying a lot of recommendations out and I still haven't got any solution.

Now my problem is this, I cannot hear anything when playing videos in firefox, ie. youtube videos, etc. nonetheless I can playback music in any other environment, rhytmbox works perfectly!
This is a new thing, and I'm worried that this happened as I upgraded two weeks ago.

Now as I've said, I've seen there are a lot of other persons having similar problems, but the solutions working for them, haven't worken for me.

Similarly, looking through the settings, I've seen that adobe flashplayer 9 plugin is installed, I've tried to uninstall it and installed the newest, which already should be included and no difference, if I right-click on the video, it still appears as if it was the 9th.. really strange!

I don't know if this has anything to do with it, but after trying a lot of the solutions mentioned, I'm really running out of patience, please somebody help me out!!

Sorry I was a bit to fast there,
here you've got the information needed;

1. Karmic Koala, amd 64 x2 4600+
2. **** List of PLAYBACK Hardware Devices ****
card 1: default [PnP Audio Device ], device 0: USB Audio [USB Audio]
Subdevices: 1/1
Subdevice #0: subdevice #0
3. I: main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted
I: main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation not permitted
D: core-rtclock.c: Timer slack is set to 50 us.
I: core-util.c: Failed to acquire high-priority scheduling: No such file or directory
I: main.c: This is PulseAudio 0.9.19
D: main.c: Compilation host: i486-pc-linux-gnu
D: main.c: Compilation CFLAGS: -g -O2 -g -Wall -O3 -Wall -W -Wextra -pipe -Wno-long-long -Winline -Wvla -Wno-overlength-strings -Wunsafe-loop-optimizations -Wundef -Wformat=2 -Wlogical-op -Wsign-compare -Wformat-security -Wmissing-include-dirs -Wformat-nonliteral -Wold-style-definition -Wpointer-arith -Winit-self -Wdeclaration-after-statement -Wfloat-equal -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wmissing-declarations -Wmissing-noreturn -Wshadow -Wendif-labels -Wcast-align -Wstrict-aliasing=2 -Wwrite-strings -Wno-unused-parameter -ffast-math -Wp,-D_FORTIFY_SOURCE=2 -fno-common -fdiagnostics-show-option
D: main.c: Running on host: Linux i686 2.6.31-20-generic #58-Ubuntu SMP Fri Mar 12 05:23:09 UTC 2010
D: main.c: Found 2 CPUs.
I: main.c: Page size is 4096 bytes
D: main.c: Compiled with Valgrind support: no
D: main.c: Running in valgrind mode: no
D: main.c: Optimized build: yes
D: main.c: All asserts enabled.
I: main.c: Machine ID is 6f70a65bf167209eae778fc548bdb831.
I: main.c: Session ID is 6f70a65bf167209eae778fc548bdb831-1270647963.204714-1042700397.
I: main.c: Using runtime directory /home/nesimartin/.pulse/6f70a65bf167209eae778fc548bdb831-runtime.
I: main.c: Using state directory /home/nesimartin/.pulse.
I: main.c: Using modules directory /usr/lib/pulse-0.9.19/modules.
I: main.c: Running in system mode: no
E: pid.c: Daemon already running.
E: main.c: pa_pid_file_create() failed.

4. I mainly have problems with video in firefox (Namoroka 3.6.2)

NOTE: There's a clicking sound when doing step 3

Thanks

progre55_icky
April 12th, 2010, 02:31 PM
Hi people! Thanks for the great guide. But I'm still having problems.

B. The application does plays audio and does not list an entry in the Playback tab;
- the application is either accessing your sound card directly or playing sound via ALSA's dmix device. This will prevent PulseAudio from working correctly & cause audio mixing errors.While firefox is on, and playing sound with no problem, and I open the PA volume control, I dont see it on the "playback" tab. But then I kill firefox and start the PA volume control first, firefox doesnt play any sound when I start it, and nor appears on the playback tab.

Moreover, when I run a track on mplayer (command line), sometimes it just freezes, sometimes seems to be playing, but there is no any sound.
Actually, mplayer appears on the playback tab, but it is using the "R700 Audio Device (Radeon HD 4000 Series, HDMI)", and the bar is moving as it's producing sound, but no output sound can be heard. When I change it to "Internal Audio Analaog Stereo" (I believe that's the HDA Intel card), it just freezes.

BTW, some info about my system:
ubuntu karmic 64bit,


progre55@progre55:~$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: Intel [HDA Intel], device 0: ALC262 Analog [ALC262 Analog]
Subdevices: 0/1
Subdevice #0: subdevice #0
card 1: HDMI [HDA ATI HDMI], device 3: ATI HDMI [ATI HDMI]
Subdevices: 1/1
Subdevice #0: subdevice #0

progre55@progre55:~$ lspci | grep Audio
00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03)
01:00.1 Audio device: ATI Technologies Inc R700 Audio Device [Radeon HD 4000 Series]


progre55@progre55:~$ cat /proc/asound/version
Advanced Linux Sound Architecture Driver Version 1.0.22.
Compiled on Apr 7 2010 for kernel 2.6.31-20-generic (SMP).


progre55@progre55:~$ pulseaudio --version
pulseaudio 0.9.21-63-gd3efa-dirty
and currently Intel [HDA Intel] is set as the default sound device, but when I choose HDMI, it seems to be producing sound, but cant hear anything.

This all actually happened after I tried to manually re-build alsa, as I was having problems with my internal mic. But now the mic is working fine, but having troubles with PA.

Appreciate any help!

progre55_icky
April 12th, 2010, 06:27 PM
Well, got it working. Apparently the firefox flash player uses alsa, and alsa was not using PA but was directly accessing the card. So this guide helped me and fixed my alsa vs PA problems: http://ubuntuforums.org/showthread.php?t=1412153

Mart_L
April 19th, 2010, 07:14 PM
Well I got it working as well, though not by using the above mentioned.
It seemed that the migration from 9.04 to 9.10 had made the firefox version installed from before obsolete, so I uninstalled the firefox package in synaptic and then afterwards I installed it again.
It worked, now firefox recognise the flash player v. 10 and I can hear once again!

Thanks

Crystufer
April 20th, 2010, 03:11 PM
Motherboard: Gigabyte EP43-DS3L
Audio Card: Onboard HDA Intel (ALC888 Analog Audio)
Problem: No audio in flash and (sometimes) Amarok. Ran fine with VLC though.
Pulseaudio Skype Flash No Sound ALC888

---------Just wanted to post that for google's benefit.----------
Thank you so much. Ran section A and boom. done. Sound back in amarok, skype, and flash. Finally an audio output that works for everything.

Talion86
April 27th, 2010, 12:59 PM
I have a problem with pulseaudio on Ubuntu 9.10 amd64.


talion@talion:~$ pulseaudio -vv
pulseaudio: symbol lookup error: pulseaudio: undefined symbol: pa_set_env
talion@talion:~$ pulseaudio -D
pulseaudio: symbol lookup error: pulseaudio: undefined symbol: pa_set_env
talion@talion:~$ aplay -l
**** List of PLAYBACK Hardware Devices ****
/usr/local/bin/pulseaudio: symbol lookup error: /usr/local/bin/pulseaudio: undefined symbol: pa_set_env
card 0: CMI8738 [C-Media CMI8738], device 0: CMI8738-MC6 [C-Media PCI DAC/ADC]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: CMI8738 [C-Media CMI8738], device 1: CMI8738-MC6 [C-Media PCI 2nd DAC]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: CMI8738 [C-Media CMI8738], device 2: CMI8738-MC6 [C-Media PCI IEC958]
Subdevices: 1/1
Subdevice #0: subdevice #0
I tried google and the forum search but did not found anything that helps. Using pulseaudio from ppa (https://code.launchpad.net/%7Eubuntu-audio-dev/pulseaudio/ppa) yields the same problems. Compiling the sources from http://pulseaudio.org/ fails:


modules/module-udev-detect.c: In function ‘is_card_busy’:
modules/module-udev-detect.c:127: warning: implicit declaration of function ‘offsetof’ [-Wimplicit-function-declaration]
modules/module-udev-detect.c:127: error: expected expression before ‘struct’
make[3]: *** [module_udev_detect_la-module-udev-detect.lo] Error 1
make[3]: Leaving directory `/home/talion/.local/share/Trash/files/pulseaudio-0.2.9.21/src'
make[2]: *** [all] Error 2
make[2]: Leaving directory `/home/talion/.local/share/Trash/files/pulseaudio-0.2.9.21/src'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/home/talion/.local/share/Trash/files/pulseaudio-0.2.9.21'
make: *** [all] Error 2
Sound works without problems without Pulseaudio, but I have a program (GuitarPro) that requires pulseaudio. Any help greatly appreciated.

stib
May 7th, 2010, 03:29 PM
I had a really annoying problem with a mythbuntu system, that was constantly clicking and popping whenever the TV wsn't actually playing. I looked in /var/log/messages, and there was about 20Mb worth of this message (it was logging it every couple of seconds, in synch with the pops):

May 8 00:22:02 tellybox pulseaudio[20594]: authkey.c: Failed to open cookie file '/home/telly/.esd_auth': Permission denied
fixing the permissions on /home/telly/.esd_auth worked instantly (Iin a terminal type

sudo chmod a+rw ~/.esd_auth
Don't know what that file does or why the permissions were wrong, but I thought it might help someone out to post how I fixed it.

dannyboy79
May 7th, 2010, 05:33 PM
I had a really annoying problem with a mythbuntu system, that was constantly clicking and popping whenever the TV wsn't actually playing. I looked in /var/log/messages, and there was about 20Mb worth of this message (it was logging it every couple of seconds, in synch with the pops):

May 8 00:22:02 tellybox pulseaudio[20594]: authkey.c: Failed to open cookie file '/home/telly/.esd_auth': Permission denied
fixing the permissions on /home/telly/.esd_auth worked instantly (Iin a terminal type

sudo chmod a+rw ~/.esd_auth
Don't know what that file does or why the permissions were wrong, but I thought it might help someone out to post how I fixed it.
only your user should have rw access to that file.
sudo chmod 0600 /home/telly/.esd_auth

El1iP3S01D
May 12th, 2010, 08:59 PM
folks, can you tell me if this is the cause for Pulseaudio not working on my system?

el1ip3s01d@Iehovah-07:~$ pkill pulseaudio; sleep 2; pulseaudio -vv
I: main.c: We're in the group 'pulse-rt', allowing real-time and high-priority scheduling.
I: core-util.c: Successfully gained nice level -11.
W: main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted
W: main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation not permitted
I: main.c: This is PulseAudio 0.9.10
I: main.c: Page size is 4096 bytes
I: main.c: Fresh high-resolution timers available! Bon appetit!
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9/modules/module-hal-detect.so': success
I: module-hal-detect.c: Trying capability alsa
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/computer_alsa_timer
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_10de_26c_alsa_capture_2
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_10de_26c_alsa_playback_1
D: module-hal-detect.c: Loading module-alsa-sink with arguments 'device_id=0 sink_name=alsa_output.pci_10de_26c_alsa_playback_0 '
D: alsa-util.c: Trying front:0...
I: alsa-util.c: Couldn't open PCM device front:0: Device or resource busy
D: alsa-util.c: Trying surround40:0...
I: alsa-util.c: Couldn't open PCM device surround40:0: Device or resource busy
D: alsa-util.c: Trying surround41:0...
I: alsa-util.c: Couldn't open PCM device surround41:0: Device or resource busy
D: alsa-util.c: Trying surround50:0...
I: alsa-util.c: Couldn't open PCM device surround50:0: Device or resource busy
D: alsa-util.c: Trying surround51:0...
I: alsa-util.c: Couldn't open PCM device surround51:0: Device or resource busy
D: alsa-util.c: Trying surround71:0...
I: alsa-util.c: Couldn't open PCM device surround71:0: Device or resource busy
D: alsa-util.c: Trying hw:0 as last resort...
E: alsa-util.c: Error opening PCM device hw:0: Device or resource busy
E: module.c: Failed to load module "module-alsa-sink" (argument: "device_id=0 sink_name=alsa_output.pci_10de_26c_alsa_playback_0"): initialization failed.
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_10de_26c_alsa_playback_0
D: module-hal-detect.c: Loading module-alsa-source with arguments 'device_id=0 source_name=alsa_input.pci_10de_26c_alsa_capture_0 '
D: alsa-util.c: Trying front:0...
I: module-alsa-source.c: Successfully opened device front:0.
I: module-alsa-source.c: Successfully enabled mmap() mode.
ALSA lib control.c:909:(snd_ctl_open_noupdate) Invalid CTL front:0
I: alsa-util.c: Unable to attach to mixer front:0: No such file or directory
I: alsa-util.c: Successfully attached to mixer 'hw:0'
I: alsa-util.c: Using mixer control "Capture".
I: source.c: Created source 0 "alsa_input.pci_10de_26c_alsa_capture_0" with sample spec "s16le 2ch 44100Hz"
I: module-alsa-source.c: Using 4 fragments of size 4352 bytes.
I: alsa-util.c: All 2 channels can be mapped to mixer channels. Using hardware volume control.
D: module-alsa-source.c: Thread starting up
D: rtpoll.c: Acquired POSIX realtime signal SIGRTMIN+29
I: module.c: Loaded "module-alsa-source" (index: #0; argument: "device_id=0 source_name=alsa_input.pci_10de_26c_alsa_capture_0").
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_10de_26c_alsa_control__1
I: module-hal-detect.c: Loaded 1 modules.
I: module.c: Loaded "module-hal-detect" (index: #1; argument: "").
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9/modules/module-esound-protocol-unix.so': success
I: module.c: Loaded "module-esound-protocol-unix" (index: #2; argument: "socket="/tmp/.esd/socket"").
I: protocol-native.c: loading cookie from disk.
I: module.c: Loaded "module-native-protocol-unix" (index: #3; argument: "").
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9/modules/module-gconf.so': success
D: module-gconf.c: Loading module 'module-native-protocol-tcp' with args '' due to GConf configuration.
I: protocol-native.c: using already loaded auth cookie.
I: protocol-native.c: using already loaded auth cookie.
I: module.c: Loaded "module-native-protocol-tcp" (index: #4; argument: "").
D: module-gconf.c: Loading module 'module-esound-protocol-tcp' with args '' due to GConf configuration.
I: module.c: Loaded "module-esound-protocol-tcp" (index: #5; argument: "").
D: module-gconf.c: Loading module 'module-zeroconf-discover' with args '' due to GConf configuration.
I: module.c: Loaded "module-zeroconf-discover" (index: #6; argument: "").
I: module.c: Loaded "module-gconf" (index: #7; argument: "").
I: module.c: Loaded "module-volume-restore" (index: #8; argument: "").
I: module-default-device-restore.c: Saved default sink 'combined' not existant, not restoring default sink setting.
D: module-default-device-restore.c: Restored default source 'alsa_input.pci_10de_26c_alsa_capture_0'.
I: module.c: Loaded "module-default-device-restore" (index: #9; argument: "").
I: module.c: Loaded "module-rescue-streams" (index: #10; argument: "").
D: module-suspend-on-idle.c: Source alsa_input.pci_10de_26c_alsa_capture_0 becomes idle.
I: module.c: Loaded "module-suspend-on-idle" (index: #11; argument: "").
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9/modules/module-x11-publish.so': success
D: module-x11-publish.c: using already loaded auth cookie.
I: module.c: Loaded "module-x11-publish" (index: #12; argument: "").
I: main.c: Daemon startup complete.
D: module-hal-detect.c: dbus: interface=org.freedesktop.DBus, path=/org/freedesktop/DBus, member=NameAcquired
I: module-suspend-on-idle.c: Source alsa_input.pci_10de_26c_alsa_capture_0 idle for too long, suspending ...
I: module-alsa-source.c: Device suspended...
^X^Z
[1]+ Stopped pulseaudio -vv

mustard greens
May 14th, 2010, 08:28 AM
there is no such option as "sound" under system>preferences. I have been seeing this all day in the forum posts for this problem. can someone tell me why everyone else seems to have a "sound" option that I dont? I am stuck at this junction.

mustard greens
May 14th, 2010, 04:59 PM
ok, so I just noticed that these instructions are not for the 10.04 release. sorry but the naming convention confuses me (which name belongs to which number) are there any fixes for someone running 10.04?

WitchCraft
May 27th, 2010, 09:18 PM
Go to System->Preferences -> Startup Applications

Make sure you're in the tab 'Startup programs'
-> Click on 'Add'

Name: Pulseaudio daemon
Command:/usr/bin/pulseaudio
Comment: Start the sound daemon

Now logout, then login again

->Fixed

zika
May 27th, 2010, 09:45 PM
Go to System->Preferences -> Startup Applications

Make sure you're in the tab 'Startup programs'
-> Click on 'Add'

Name: Pulseaudio daemon
Command:/usr/bin/pulseaudio
Comment: Start the sound daemon

Now logout, then login again

->FixedOr, IMHO:
Command:/usr/bin/start-pulseaudio-x11

wasabishot
May 28th, 2010, 02:23 PM
Is this thread connected somehow to the bug with flash and pulseaudio?
I can't dual listen to flash and other music/video

I'm using Lucid 10.04 so i cannot really follow the instructions above

anybody knows how to make it work???

thanks.

RgnKjnVA
May 28th, 2010, 02:55 PM
Is this thread connected somehow to the bug with flash and pulseaudio?
I can't dual listen to flash and other music/video

I'm using Lucid 10.04 so i cannot really follow the instructions above

anybody knows how to make it work???

thanks.

Strange, I used to have that problem since pulse audio was introduced but it worked out of the box for me with 10.04

wasabishot
May 28th, 2010, 06:56 PM
well it did work out of the box for me. and also with karmic 9.10.
but since opening the box many water flew in the ocean and many changes have been made

It is such an old thread and an old bug... and no answer on how to fix this?? nothing????

some support here please!

DJonsson2008
May 30th, 2010, 10:39 AM
Maybe this was answered further up the topic

I was very relieved to find that 10.04 allows one
to simply disable PulseAudio by unchecking it as a
startup program. Somebody out there is listening.

zika
May 30th, 2010, 11:44 AM
Maybe this was answered further up the topic

I was very relieved to find that 10.04 allows one
to simply disable PulseAudio by unchecking it as a
startup program. Somebody out there is listening.You, just, turned off /usr/bin/start-pulseaudio-x11. Check ps -e|grep pulse...
(If anyone knows how to uninstall pulseaudio (or just turn it off with ALSA still working) from Maverick (or Lucid) I would like to see... Avid ALSA user...
I've tried lot of stuff... pulseaudio --kill was first ... :)
If I try to uninstall it it wants to drag ubuntu-desktop with it...)

wavesound
June 6th, 2010, 03:29 PM
Hey anyone out there having difficulty trying to get Pulseaudio working with a hardware playback device that uses ICE1712 (as displayed when you run the command "aplay -l", there is a bug I have found which seems to affect all cards using this chipset.

For example my card is card 0: H71 [Hoontech STA DSP24 Media 7.1], device 0: ICE1712 multi [ICE1712 multi] but according to this bug other cards such as M-Audio type cards are affected too.

There is a manual workaround listed in this bug.
https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/178442
There is a posting about an M-Audio Delta 44 card...
https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/178442/comments/8

I have found that this workaround works for my card too.

If it helps people I have included the steps I made to get my ICE1712 type card working in pulseaudio.

In my example you will need to replace the "H71" with whatever the device says when you run
asoundconf list

By editing this file,

gksudo gedit /etc/pulse/default.pa
and adding the following lines at the end, I now have a working pulseaudio in 8.10 64 bit.


# Workaround for MAudio Audiophile
# https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/178442/comments/7
# Added comment rod40cool - (STA Audio Media 7.1 card uses same ICE1712 driver)
# Added comment rod40cool - Run command "asoundconf list" to determine what to add after "device=hw:????"
# Added comment rod40cool - Make sure after pasting that there are only 2 lines below starting each with "load-module module..."
load-module module-alsa-sink sink_name=H71_out device=hw:H71 format=s32le channels=10 channel_map=left,right,aux0,aux1,aux2,aux3,aux4,au x5,aux6,aux7
load-module module-alsa-source source_name=H71_in device=hw:H71 format=s32le channels=12 channel_map=left,right,aux0,aux1,aux2,aux3,aux4,au x5,aux6,aux7,aux8,aux9

I hope this helps someone who is trying to get pulseaudio running for a ST Audio Media 7.1 or M-Audio Delta card or in fact any card that uses the ICE1712 driver.

Cheers

HI
THis is what I get from Lucid lynx when try aplay -L:


aplay -l
**** List of PLAYBACK Hardware Devices ****
ALSA lib conf.c:1645:(snd_config_load1) _toplevel_:5:10:Unexpected char
ALSA lib conf.c:3425:(snd_config_hook_load) /home/studio/.asoundrc may be old or corrupted: consider to remove or fix it
ALSA lib conf.c:3286:(snd_config_hooks_call) function snd_config_hook_load returned error: Invalid argument
ALSA lib conf.c:3671:(snd_config_update_r) hooks failed, removing configuration
aplay: device_list:232: control open (0): Invalid argument
*******************************

Would seem to be some coding error.
This for the Audiophile 24/96 which is a well used card on Linux as it was well supported in Alsa

Cheers
Bob

andrius7
June 10th, 2010, 07:15 AM
HI,

I seems lost most bars in alsamixer, whilst installing lives program.
It used to be many bars on alsamixer and now only one - master and even this does not adjust volume.
At the same time I lost sound on skype.
Could anyone help pls.
I tried do remove pulseaudio, install esd, but no luck, still no sound on skype and master does work.

mclizardman24
June 22nd, 2010, 02:12 AM
would a symptom of pulseaudio not working be screen flickering and sounds lagging with actions while using flash content? That is what is happening to me. =(

joeoshawa
June 25th, 2010, 08:05 PM
just thought i would put my 2 cents in. since i started with Ubuntu i have been fixing the sound over and over and it was really getting annoying. Finally i erased it and installed esound and guess what... works perfect no fixing needed event the Ubuntu intro works no fixing needed so buh bye PulseAudio keep putting it on and i will keep deleting it. PulseAudio is garbage and unless they figure out how to get it working properly I am not using it.
P.S. I would remove the I don't want to use it well too bad cause that is a microsoft slogan not a linux one.

ypestis
July 12th, 2010, 04:50 PM
Hi I have a problem with my PulseAudio my volume control stopped working out of the blue and I get a " waiting for sound system to respond" message..
.
My thread is here: http://ubuntuforums.org/showthread.php?t=1529023

hope somebody can help me out!

Thanks alot!

reef2dive
July 20th, 2010, 05:40 PM
I wonder if anyone could help me with this thread "No Sound - Pulseaudio does not show device"
http://ubuntuforums.org/showthread.php?t=1517726

Pulseaudio does not show the analogue device only the HDMI device. I wonder how I can force pulseaudio to show the analogue one.

kakyoism
July 22nd, 2010, 06:52 AM
I have a weird "no sound" problem.

I use a HP dv-4 laptop and run Lucid on it. While watching video or flash, initially there will be sound, but as I move my mouse a little it just becomes silent until I shut down the media player and restart it.

This drives me crazy!

HELP!

duanedesign
July 24th, 2010, 02:56 PM
Has anyone tried the Pulse Audio Equalizer on Lucid?

psyke83
July 24th, 2010, 03:34 PM
Has anyone tried the Pulse Audio Equalizer on Lucid?

The equalizer instructions in this guide are obsolete. See my signature for something more up-to-date...

disastrophe
July 25th, 2010, 12:37 AM
I have it working in Lucid beautifully. All the libraries were already in place, I only had to add the ppa and install the EQ from Synaptic then edit the default.pa and .soundrc files as shown here in appendix D.
One little hazard to watch out for, if you change hardware devices (i.e. from analog stereo to analog 4.0 etc.) from the sound properties dialog make sure to also revert /etc/pulse/default.pa and ~/.asoundrc or your sound system will be very angry at you and quit after which polkit will go nuts eating about 60% of your cpu cycles lol. Editing the files to their preinstall state and restarting Ubuntu settles everything down again. Then I went back, re-edited the two files, re-enabled the EQ and restarted. Works great now, I have global kick **** sound!
Finally, I would like to say thank you very much psyke83 for what is obviously a lot of work and a job very well done.

psyke83
July 25th, 2010, 12:57 AM
I have it working in Lucid beautifully. All the libraries were already in place, I only had to add the ppa and install the EQ from Synaptic then edit the default.pa and .soundrc files as shown here in appendix D.
One little hazard to watch out for, if you change hardware devices (i.e. from analog stereo to analog 4.0 etc.) from the sound properties dialog make sure to also revert /etc/pulse/default.pa and ~/.asoundrc or your sound system will be very angry at you and quit after which polkit will go nuts eating about 60% of your cpu cycles lol. Editing the files to their preinstall state and restarting Ubuntu settles everything down again. Then I went back, re-edited the two files, re-enabled the EQ and restarted. Works great now, I have global kick **** sound!
Finally, I would like to say thank you very much psyke83 for what is obviously a lot of work and a job very well done.

If you've installed the equalizer from the separate thread (here (http://ubuntuforums.org/showthread.php?t=1308838)), you shouldn't follow any part of Appendix D in this guide - they will conflict. Appendix D should be considered as obsolete due to the script providing the same functionality. It's mentioned in the "disclaimer" in this guide.

I recommend that you undo any changes you've made from Appendix D, delete the contents of your "~/.pulse/" folder, restart Pulseaudio (by logging out and in, or killing the process), and then use the "PulseAudio Equalizer" interface on its own.

disastrophe
July 25th, 2010, 03:02 AM
Okay, nevermind lol.

psyke83
July 25th, 2010, 03:45 AM
Okay, nevermind lol.

No worries ;). I just noticed that I only mentioned Karmic in Appendix D; I'll update it to clear up any confusion that others may also have.

disastrophe
July 25th, 2010, 04:20 AM
It's a great add on. I am a noob obviously with way too much time in m$ behind me. Sometimes I relapse:oops:. Just say "it's a script for pa you silly goose" - I mean I was sitting here worrying about how I'm going to autostart it lol.

kainalu
August 7th, 2010, 09:22 AM
psyke83, I logged in just to say that, for a very complicated guide that supports so many configurations and versions, it was one of the best laid out, informative, easy to follow guides Ive ever found on this forum. Kudos!

KJ55
September 8th, 2010, 08:03 PM
My current Ubuntu installation is Hardy Heron 8.04.4 LTS. I previously used this outstanding guide to fix sound problems related to SDL type games. I recently tried upgrading to Lucid 10.04.1 LTS using an Alternate CD, and received the following error:


Could not calculate the upgrade

An unresolveable problem occurred while calculating the upgrade:
E:Error, pkgProblemResolver::Resolve generated breaks, this may be caused by held packages.

This can be caused by:
-Upgrading to a pre-release version of Ubuntu
-Running the current pre-release version of Ubuntu
-Unofficial software packages not provided by Ubuntu

If none of this applies, then please report this bug against the 'update-manger' package and include files in /varlog/dist-upgrade/ in the bug report.

Is the above error caused by the 13 separate sound related packages installed in my installation of Hardy Heron 8.04.4 from psyke83's PPA (e.g.; 1.0.16-1ubuntu3~ppa1, 1.0.16-2ubuntu2~ppa1, etc.)?

Even if these 13 separate sound related packages are not the reason for preventing my upgrade to Lucid 10.04.1 LTS is there a way of removing these installed packages from the PPA and re-installing the default Hardy ones without breaking my Hardy Heron 8.04.4 LTS installation. I have read here (https://help.ubuntu.com/community/CleanUpgrade) that it may be possible, but I'm unsure. The reason I'm unsure is because the referenced guide states to install the "Standard Ubuntu Components," you use the forced version option to install the default Hardy packages.

And, the guide furthermore states there should only be two possible versions of each package (the PPA and the default Hardy package). I have more than two for some packages. For example, for the libpulse0 package I have the following:


0.9.10-2ubuntu4~ppa3 (now)
0.9.10-1ubuntu1.1 (hardy-updates)
0.9.10-1ubuntu1 (hardy)

I believe, in the example above, the version I need to force is 0.9.10-1ubuntu1.1 (hardy-updates).

But, that word forced installation worries me.

Any advice or recommendation for this confused Ubuntu newbie on how to proceed is greatly appreciated.

christhegoth
September 8th, 2010, 08:40 PM
Use paconfig to select your main onboard soundcard. In Ubuntu 10.04, running on a Hardy Install, that sorts it now.

Including flash, and music as well.

What I've done here is used the onboard sound ( cheapo C-Media ) for the speakers and flash etc etc, and also added a USB Soundblaster. Nothing comes out of the USB Card normally, but if you use Audacious for your music you can select an output card on the Alsa System, and have that as a different card to your Pulseaudio normal card. The C-Media is configured for Pulseaudio, and the sound settings are all set to the C-Media card in desktop config. So it's just Audacious that accesses the USB Card.

So I can watch porn in flash, hear the silly noises, and have music on the surround sound via the USB at the same time ;)

Or Youtube obviously *ahem*.

The good thing is Alsa runs alongside Pulse Audio. It's the same with the Last.fm application. It locks a Soundcard so that nothing else comes out sure, but you get to choose which card to lock up. Which is pretty handy.

pdv99
September 17th, 2010, 09:31 PM
Hi
I'm trying to configure pulse to use a usb sound card with Mixx so that I can output he main playback through the usb card and use interval audio for headphone monitor out, to cue tracks. aplay gives
****List of PLAYBACK Hardware Devices ****
card 0: Intel [HDA Intel], device 0:ALC268 Analog [ALC268 Analog]
Subdevices: 0/1
Subdevice #0: subdevice #0
card 1: default [USB AUDIO ], device 0: USB Audio [USB Audio]
Subdevices: 0/1
Subdevice #0: subdevice #0

The USB card shows up under Devices in Pulse Audio Manager

However, only the built in sound card shows up in Pulse Volume control.

I'm running Lucid. How do I configure Pulse to use both cards, - or should I just disable Pulse and try to get Mixx to access hardware directly.

Regards
Peter

Quarterpipesmoker
October 13th, 2010, 12:21 AM
"The latest release of Skype has native PulseAudio support, and does not require special configuration."

I have problems with this, so that doesnt seem to be true. Maybe its true for 10.10

This aint true either:
"Note 4: Kubuntu users: Don't follow this guide - PulseAudio isn't used in your distribution."

Since 10.04 or 10.10 or whatever version pulse audio is used with kubuntu.

mocha
November 3rd, 2010, 06:30 PM
Just some general information for anyone that might be getting static or crackling sounds from their analog line-in or microphone input recordings on Maverick 10.10. After I upgraded to 10.10 this is the first version of Ubuntu or Pulseaudio that gave me problems when using the Pulseaudio equalizer LADSPA effects. I finally figured out to disable the equalizer effects and now my recordings don't have static in them. I didn't think the playback effects would have anything to do with recording, but they do for some reason. Anyway, it's something to be aware of.

Enigmapond
November 5th, 2010, 02:06 PM
I have a Dell Insipron 1764 Laptop and I installed the equalizer and all works well except that it seems to have broken my internal mic. I have check the setting for the mic and it is fine according to alsamixer. Is there anyone else having this problem and if so, is there a fix? I pull up the sound preferences and is doesn't even show the mic in the inputs and the slider is greyed out. Thank you...

khatarnak
December 17th, 2010, 07:55 PM
Hello,

I'm trying to run a command-line ubuntu system (10.10) as a music-player. I want to use pulseaudio and it's network capailities and mpd as well.

unfortunately, I don't get it running. Sound in general works (using live-CD for example).

My system is configured as are as follows:


cat /proc/asound/cards
0 [I82801DBICH4 ]: ICH4 - Intel 82801DB-ICH4
Intel 82801DB-ICH4 with ALC650F at irq 11


aplay -l
**** Liste der Hardware-Geräte (PLAYBACK) ****
Karte 0: I82801DBICH4 [Intel 82801DB-ICH4], Gerät 0: Intel ICH [Intel 82801DB-ICH4]
Sub-Geräte: 1/1
Sub-Gerät #0: subdevice #0
Karte 0: I82801DBICH4 [Intel 82801DB-ICH4], Gerät 4: Intel ICH - IEC958 [Intel 82801DB-ICH4 - IEC958]
Sub-Geräte: 1/1
Sub-Gerät #0: subdevice #0



pkill pulseaudio; sleep 2; pulseaudio -vv
I: main.c: setrlimit(RLIMIT_NICE, (31, 31)) fehlgeschlagen: Operation not permitted
I: main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) fehlgeschlagen: Operation not permitted
D: core-rtclock.c: Timer slack is set to 50 us.
I: core-util.c: Failed to acquire high-priority scheduling: Input/output error
I: main.c: Dies ist PulseAudio 0.9.21-63-gd3efa-dirty
D: main.c: Kompilier-Host: i686-pc-linux-gnu
D: main.c: Kompilier-CFLAGS: -g -O2 -g -Wall -O3 -Wall -W -Wextra -pipe -Wno-long-long -Winline -Wvla -Wno-overlength-strings -Wunsafe-loop-optimizations -Wundef -Wformat=2 -Wlogical-op -Wsign-compare -Wformat-security -Wmissing-include-dirs -Wformat-nonliteral -Wold-style-definition -Wpointer-arith -Winit-self -Wdeclaration-after-statement -Wfloat-equal -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wmissing-declarations -Wmissing-noreturn -Wshadow -Wendif-labels -Wcast-align -Wstrict-aliasing=2 -Wwrite-strings -Wno-unused-parameter -ffast-math -Wp,-D_FORTIFY_SOURCE=2 -fno-common -fdiagnostics-show-option
D: main.c: Laufe auf Host: Linux i686 2.6.35-22-generic #35-Ubuntu SMP Sat Oct 16 20:36:48 UTC 2010
D: main.c: 1 CPUs gefunden.
I: main.c: Seitengröße ist 4096 Bytes.
D: main.c: Kompiliere mit Valgrind-Unterstützung: nein
D: main.c: Läuft im Valgrind-Modus: no
D: main.c: Running in VM: no
D: main.c: Optimiertes Build: ja
D: main.c: Alle Ansprüche aktiviert.
I: main.c: System- ID ist 49dcdd0b374327837e7d58d200000756.
I: main.c: System- ID ist 49dcdd0b374327837e7d58d200000756-1292608677.215415-1465211344.
I: main.c: Nutze Laufzeit-Verzeichnis /home/mucke/.pulse/49dcdd0b374327837e7d58d200000756-runtime.
I: main.c: Nutze Zustands-Verzeichnis /home/mucke/.pulse.
I: main.c: Modul-Verzeichnis /usr/lib/pulse-0.9.21/modules benutzen.
I: main.c: Laufe im System-Modus: no
I: main.c: Neue hochauslösende Timer verfügbar! Guten Appetit!
I: cpu-x86.c: CPU flags: CMOV MMX SSE
I: svolume_mmx.c: Initialising MMX optimized functions.
I: remap_mmx.c: Initialising MMX optimized remappers.
I: sconv_sse.c: Initialising SSE optimized conversions.
D: memblock.c: Using shared memory pool with 1024 slots of size 64,0 KB each, total size is 64,0 MB, maximum usable slot size is 65496
D: database-tdb.c: Opened TDB database '/home/mucke/.pulse/49dcdd0b374327837e7d58d200000756-device-volumes.tdb'
I: module-device-restore.c: Sucessfully opened database file '/home/mucke/.pulse/49dcdd0b374327837e7d58d200000756-device-volumes'.
I: module.c: Loaded "module-device-restore" (index: #0; argument: "").
D: database-tdb.c: Opened TDB database '/home/mucke/.pulse/49dcdd0b374327837e7d58d200000756-stream-volumes.tdb'
I: module-stream-restore.c: Sucessfully opened database file '/home/mucke/.pulse/49dcdd0b374327837e7d58d200000756-stream-volumes'.
I: module.c: Loaded "module-stream-restore" (index: #1; argument: "").
D: database-tdb.c: Opened TDB database '/home/mucke/.pulse/49dcdd0b374327837e7d58d200000756-card-database.tdb'
I: module-card-restore.c: Sucessfully opened database file '/home/mucke/.pulse/49dcdd0b374327837e7d58d200000756-card-database'.
I: module.c: Loaded "module-card-restore" (index: #2; argument: "").
I: module.c: Loaded "module-augment-properties" (index: #3; argument: "").
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.21/modules/module-udev-detect.so': success
D: module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
D: module-udev-detect.c: /devices/pci0000:00/0000:00:1f.5/sound/card0 is busy: no
D: module-udev-detect.c: Loading module-alsa-card with arguments 'device_id="0" name="pci-0000_00_1f.5" card_name="alsa_card.pci-0000_00_1f.5" tsched=yes ignore_dB=no card_properties="module-udev-detect.discovered=1"'
D: reserve-wrap.c: Unable to contact D-Bus session bus: org.freedesktop.DBus.Error.Spawn.ExecFailed: /bin/dbus-launch terminated abnormally with the following error: Autolaunch error: X11 initialization failed.
D: alsa-mixer.c: Looking at profile output:analog-mono
D: alsa-mixer.c: Checking for playback on Analog Mono (analog-mono)
D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:analog-mono+input:analog-mono
D: alsa-mixer.c: Checking for playback on Analog Mono (analog-mono)
D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:analog-mono+input:analog-stereo
D: alsa-mixer.c: Checking for playback on Analog Mono (analog-mono)
D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:analog-mono+input:iec958-stereo
D: alsa-mixer.c: Checking for playback on Analog Mono (analog-mono)
D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:analog-mono+input:iec958-surround-40
D: alsa-mixer.c: Checking for playback on Analog Mono (analog-mono)
D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:analog-stereo
D: alsa-mixer.c: Checking for playback on Analog Stereo (analog-stereo)
D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open front:0
D: alsa-util.c: Maximum hw buffer size is 371 ms
D: alsa-util.c: Set buffer size first (to 3528 samples), period size second (to 441 samples).
D: alsa-mixer.c: Profile output:analog-stereo supported.
D: alsa-mixer.c: Looking at profile output:analog-stereo+input:analog-mono
D: alsa-mixer.c: Checking for recording on Analog Mono (analog-mono)
D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:analog-stereo+input:analog-stereo
D: alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo)
D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open front:0
D: alsa-util.c: Maximum hw buffer size is 371 ms
D: alsa-util.c: Set buffer size first (to 3528 samples), period size second (to 441 samples).
D: alsa-mixer.c: Profile output:analog-stereo+input:analog-stereo supported.
D: alsa-mixer.c: Looking at profile output:analog-stereo+input:iec958-stereo
D: alsa-mixer.c: Checking for recording on Digital Stereo (IEC958) (iec958-stereo)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D4c' failed (-2)
I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:analog-stereo+input:iec958-surround-40
D: alsa-mixer.c: Checking for recording on Digital Surround 4.0 (IEC958) (iec958-surround-40)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D4c' failed (-2)
I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:analog-surround-40
D: alsa-mixer.c: Checking for playback on Analog Surround 4.0 (analog-surround-40)
D: alsa-util.c: Trying surround40:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open surround40:0
D: alsa-util.c: Maximum hw buffer size is 185 ms
D: alsa-util.c: Set buffer size first (to 3528 samples), period size second (to 441 samples).
D: alsa-mixer.c: Profile output:analog-surround-40 supported.
D: alsa-mixer.c: Looking at profile output:analog-surround-40+input:analog-mono
D: alsa-mixer.c: Checking for recording on Analog Mono (analog-mono)
D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:analog-surround-40+input:analog-stereo
D: alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo)
D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open front:0
D: alsa-util.c: Maximum hw buffer size is 371 ms
D: alsa-util.c: Set buffer size first (to 3528 samples), period size second (to 441 samples).
D: alsa-mixer.c: Profile output:analog-surround-40+input:analog-stereo supported.
D: alsa-mixer.c: Looking at profile output:analog-surround-40+input:iec958-stereo
D: alsa-mixer.c: Checking for recording on Digital Stereo (IEC958) (iec958-stereo)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D4c' failed (-2)
I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:analog-surround-40+input:iec958-surround-40
D: alsa-mixer.c: Checking for recording on Digital Surround 4.0 (IEC958) (iec958-surround-40)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D4c' failed (-2)
I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:analog-surround-41
D: alsa-mixer.c: Checking for playback on Analog Surround 4.1 (analog-surround-41)
D: alsa-util.c: Trying surround41:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open surround41:0
D: alsa-util.c: Maximum hw buffer size is 123 ms
D: alsa-util.c: Set buffer size first (to 3528 samples), period size second (to 441 samples).
D: alsa-mixer.c: Profile output:analog-surround-41 supported.
D: alsa-mixer.c: Looking at profile output:analog-surround-41+input:analog-mono
D: alsa-mixer.c: Checking for recording on Analog Mono (analog-mono)
D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:analog-surround-41+input:analog-stereo
D: alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo)
D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open front:0
D: alsa-util.c: Maximum hw buffer size is 371 ms
D: alsa-util.c: Set buffer size first (to 3528 samples), period size second (to 441 samples).
D: alsa-mixer.c: Profile output:analog-surround-41+input:analog-stereo supported.
D: alsa-mixer.c: Looking at profile output:analog-surround-41+input:iec958-stereo
D: alsa-mixer.c: Checking for recording on Digital Stereo (IEC958) (iec958-stereo)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D4c' failed (-2)
I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:analog-surround-41+input:iec958-surround-40
D: alsa-mixer.c: Checking for recording on Digital Surround 4.0 (IEC958) (iec958-surround-40)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D4c' failed (-2)
I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:analog-surround-50
D: alsa-mixer.c: Checking for playback on Analog Surround 5.0 (analog-surround-50)
D: alsa-util.c: Trying surround50:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open surround50:0
D: alsa-util.c: Maximum hw buffer size is 123 ms
D: alsa-util.c: Set buffer size first (to 3528 samples), period size second (to 441 samples).
D: alsa-mixer.c: Profile output:analog-surround-50 supported.
D: alsa-mixer.c: Looking at profile output:analog-surround-50+input:analog-mono
D: alsa-mixer.c: Checking for recording on Analog Mono (analog-mono)
D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:analog-surround-50+input:analog-stereo
D: alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo)
D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open front:0
D: alsa-util.c: Maximum hw buffer size is 371 ms
D: alsa-util.c: Set buffer size first (to 3528 samples), period size second (to 441 samples).
D: alsa-mixer.c: Profile output:analog-surround-50+input:analog-stereo supported.
D: alsa-mixer.c: Looking at profile output:analog-surround-50+input:iec958-stereo
D: alsa-mixer.c: Checking for recording on Digital Stereo (IEC958) (iec958-stereo)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D4c' failed (-2)
I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:analog-surround-50+input:iec958-surround-40
D: alsa-mixer.c: Checking for recording on Digital Surround 4.0 (IEC958) (iec958-surround-40)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D4c' failed (-2)
I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:analog-surround-51
D: alsa-mixer.c: Checking for playback on Analog Surround 5.1 (analog-surround-51)
D: alsa-util.c: Trying surround51:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open surround51:0
D: alsa-util.c: Maximum hw buffer size is 123 ms
D: alsa-util.c: Set buffer size first (to 3528 samples), period size second (to 441 samples).
D: alsa-mixer.c: Profile output:analog-surround-51 supported.
D: alsa-mixer.c: Looking at profile output:analog-surround-51+input:analog-mono
D: alsa-mixer.c: Checking for recording on Analog Mono (analog-mono)
D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:analog-surround-51+input:analog-stereo
D: alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo)
D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open front:0
D: alsa-util.c: Maximum hw buffer size is 371 ms
D: alsa-util.c: Set buffer size first (to 3528 samples), period size second (to 441 samples).
D: alsa-mixer.c: Profile output:analog-surround-51+input:analog-stereo supported.
D: alsa-mixer.c: Looking at profile output:analog-surround-51+input:iec958-stereo
D: alsa-mixer.c: Checking for recording on Digital Stereo (IEC958) (iec958-stereo)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D4c' failed (-2)
I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:analog-surround-51+input:iec958-surround-40
D: alsa-mixer.c: Checking for recording on Digital Surround 4.0 (IEC958) (iec958-surround-40)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D4c' failed (-2)
I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:analog-surround-71
D: alsa-mixer.c: Checking for playback on Analog Surround 7.1 (analog-surround-71)
D: alsa-util.c: Trying surround71:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM surround71:0
I: alsa-util.c: Error opening PCM device surround71:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:analog-surround-71+input:analog-mono
D: alsa-mixer.c: Checking for playback on Analog Surround 7.1 (analog-surround-71)
D: alsa-util.c: Trying surround71:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM surround71:0
I: alsa-util.c: Error opening PCM device surround71:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:analog-surround-71+input:analog-stereo
D: alsa-mixer.c: Checking for playback on Analog Surround 7.1 (analog-surround-71)
D: alsa-util.c: Trying surround71:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM surround71:0
I: alsa-util.c: Error opening PCM device surround71:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:analog-surround-71+input:iec958-stereo
D: alsa-mixer.c: Checking for playback on Analog Surround 7.1 (analog-surround-71)
D: alsa-util.c: Trying surround71:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM surround71:0
I: alsa-util.c: Error opening PCM device surround71:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:analog-surround-71+input:iec958-surround-40
D: alsa-mixer.c: Checking for playback on Analog Surround 7.1 (analog-surround-71)
D: alsa-util.c: Trying surround71:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM surround71:0
I: alsa-util.c: Error opening PCM device surround71:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:iec958-stereo
D: alsa-mixer.c: Checking for playback on Digital Stereo (IEC958) (iec958-stereo)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open iec958:0
D: alsa-util.c: Maximum hw buffer size is 371 ms
D: alsa-util.c: Set buffer size first (to 3528 samples), period size second (to 441 samples).
D: alsa-mixer.c: Profile output:iec958-stereo supported.
D: alsa-mixer.c: Looking at profile output:iec958-stereo+input:analog-mono
D: alsa-mixer.c: Checking for recording on Analog Mono (analog-mono)
D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:iec958-stereo+input:analog-stereo
D: alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo)
D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open front:0
D: alsa-util.c: Maximum hw buffer size is 371 ms
D: alsa-util.c: Set buffer size first (to 3528 samples), period size second (to 441 samples).
D: alsa-mixer.c: Profile output:iec958-stereo+input:analog-stereo supported.
D: alsa-mixer.c: Looking at profile output:iec958-stereo+input:iec958-stereo
D: alsa-mixer.c: Checking for recording on Digital Stereo (IEC958) (iec958-stereo)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D4c' failed (-2)
I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-stereo+input:iec958-surround-40
D: alsa-mixer.c: Checking for recording on Digital Surround 4.0 (IEC958) (iec958-surround-40)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D4c' failed (-2)
I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-surround-40
D: alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC958) (iec958-surround-40)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open iec958:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
D: alsa-util.c: Trying iec958:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open iec958:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
D: alsa-util.c: Trying plug:iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:iec958:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
D: alsa-util.c: Trying plug:iec958:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:iec958:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
I: alsa-util.c: Failed to set hardware parameters on plug:iec958:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:iec958-surround-40+input:analog-mono
D: alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC958) (iec958-surround-40)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open iec958:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
D: alsa-util.c: Trying iec958:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open iec958:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
D: alsa-util.c: Trying plug:iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:iec958:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
D: alsa-util.c: Trying plug:iec958:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:iec958:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
I: alsa-util.c: Failed to set hardware parameters on plug:iec958:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:iec958-surround-40+input:analog-stereo
D: alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC958) (iec958-surround-40)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open iec958:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
D: alsa-util.c: Trying iec958:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open iec958:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
D: alsa-util.c: Trying plug:iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:iec958:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
D: alsa-util.c: Trying plug:iec958:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:iec958:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
I: alsa-util.c: Failed to set hardware parameters on plug:iec958:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:iec958-surround-40+input:iec958-stereo
D: alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC958) (iec958-surround-40)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open iec958:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
D: alsa-util.c: Trying iec958:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open iec958:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
D: alsa-util.c: Trying plug:iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:iec958:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
D: alsa-util.c: Trying plug:iec958:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:iec958:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
I: alsa-util.c: Failed to set hardware parameters on plug:iec958:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:iec958-surround-40+input:iec958-surround-40
D: alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC958) (iec958-surround-40)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open iec958:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
D: alsa-util.c: Trying iec958:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open iec958:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
D: alsa-util.c: Trying plug:iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:iec958:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
D: alsa-util.c: Trying plug:iec958:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:iec958:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(4) failed: Invalid argument
I: alsa-util.c: Failed to set hardware parameters on plug:iec958:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-40
D: alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC958/AC3) (iec958-ac3-surround-40)
D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-40+input:analog-mono
D: alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC958/AC3) (iec958-ac3-surround-40)
D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-40+input:analog-stereo
D: alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC958/AC3) (iec958-ac3-surround-40)
D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-40+input:iec958-stereo
D: alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC958/AC3) (iec958-ac3-surround-40)
D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-40+input:iec958-surround-40
D: alsa-mixer.c: Checking for playback on Digital Surround 4.0 (IEC958/AC3) (iec958-ac3-surround-40)
D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-51
D: alsa-mixer.c: Checking for playback on Digital Surround 5.1 (IEC958/AC3) (iec958-ac3-surround-51)
D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-51+input:analog-mono
D: alsa-mixer.c: Checking for playback on Digital Surround 5.1 (IEC958/AC3) (iec958-ac3-surround-51)
D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-51+input:analog-stereo
D: alsa-mixer.c: Checking for playback on Digital Surround 5.1 (IEC958/AC3) (iec958-ac3-surround-51)
D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-51+input:iec958-stereo
D: alsa-mixer.c: Checking for playback on Digital Surround 5.1 (IEC958/AC3) (iec958-ac3-surround-51)
D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:iec958-ac3-surround-51+input:iec958-surround-40
D: alsa-mixer.c: Checking for playback on Digital Surround 5.1 (IEC958/AC3) (iec958-ac3-surround-51)
D: alsa-util.c: Trying a52:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: No such file or directory
D: alsa-mixer.c: Looking at profile output:hdmi-stereo
D: alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo)
D: alsa-util.c: Trying hdmi:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM hdmi:0
I: alsa-util.c: Error opening PCM device hdmi:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:hdmi-stereo+input:analog-mono
D: alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo)
D: alsa-util.c: Trying hdmi:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM hdmi:0
I: alsa-util.c: Error opening PCM device hdmi:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:hdmi-stereo+input:analog-stereo
D: alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo)
D: alsa-util.c: Trying hdmi:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM hdmi:0
I: alsa-util.c: Error opening PCM device hdmi:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:hdmi-stereo+input:iec958-stereo
D: alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo)
D: alsa-util.c: Trying hdmi:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM hdmi:0
I: alsa-util.c: Error opening PCM device hdmi:0: Invalid argument
D: alsa-mixer.c: Looking at profile output:hdmi-stereo+input:iec958-surround-40
D: alsa-mixer.c: Checking for playback on Digital Stereo (HDMI) (hdmi-stereo)
D: alsa-util.c: Trying hdmi:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)conf.c: Unknown parameters 0
I: (alsa-lib)pcm.c: Unknown PCM hdmi:0
I: alsa-util.c: Error opening PCM device hdmi:0: Invalid argument
D: alsa-mixer.c: Looking at profile input:analog-mono
D: alsa-mixer.c: Checking for recording on Analog Mono (analog-mono)
D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
D: alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open plug:hw:0
D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument
I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid argument
D: alsa-mixer.c: Looking at profile input:analog-stereo
D: alsa-mixer.c: Checking for recording on Analog Stereo (analog-stereo)
D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open front:0
D: alsa-util.c: Maximum hw buffer size is 371 ms
D: alsa-util.c: Set buffer size first (to 3528 samples), period size second (to 441 samples).
D: alsa-mixer.c: Profile input:analog-stereo supported.
D: alsa-mixer.c: Looking at profile input:iec958-stereo
D: alsa-mixer.c: Checking for recording on Digital Stereo (IEC958) (iec958-stereo)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D4c' failed (-2)
I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
D: alsa-mixer.c: Looking at profile input:iec958-surround-40
D: alsa-mixer.c: Checking for recording on Digital Surround 4.0 (IEC958) (iec958-surround-40)
D: alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ...
I: (alsa-lib)pcm_hw.c: open '/dev/snd/pcmC0D4c' failed (-2)
I: alsa-util.c: Error opening PCM device iec958:0: No such file or directory
I: card.c: Created 0 "alsa_card.pci-0000_00_1f.5"
D: reserve-wrap.c: Unable to contact D-Bus session bus: org.freedesktop.DBus.Error.Spawn.ExecFailed: /bin/dbus-launch terminated abnormally with the following error: Autolaunch error: X11 initialization failed.
D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open front:0
D: alsa-util.c: Maximum hw buffer size is 371 ms
D: alsa-util.c: Set buffer size first (to 88200 samples), period size second (to 88200 samples).
I: alsa-sink.c: Successfully opened device front:0.
I: alsa-sink.c: Selected mapping 'Analog Stereo' (analog-stereo).
I: alsa-sink.c: Successfully enabled mmap() mode.
I: alsa-sink.c: Successfully enabled timer-based scheduling mode.
I: (alsa-lib)control.c: Invalid CTL front:0
I: alsa-mixer.c: Unable to attach to mixer front:0: No such file or directory
I: alsa-mixer.c: Successfully attached to mixer 'hw:0'
D: alsa-mixer.c: Probing path 'analog-output'
D: alsa-mixer.c: Probing path 'analog-output-speaker'
D: alsa-mixer.c: Probe of element 'Speaker' failed.
D: alsa-mixer.c: Probing path 'analog-output-speaker'
D: alsa-mixer.c: Probe of element 'Desktop Speaker' failed.
D: alsa-mixer.c: Probing path 'analog-output-headphones'
D: alsa-mixer.c: Probe of element 'Headphone' failed.
D: alsa-mixer.c: Probing path 'analog-output-headphones'
D: alsa-mixer.c: Probe of element 'Headphone2' failed.
D: alsa-mixer.c: Probing path 'analog-output-mono'
D: alsa-mixer.c: Probing path 'analog-output-lfe-on-mono'
D: alsa-sink.c: Probed mixer paths:
D: alsa-mixer.c: Path Set 0xa0488f8, direction=1, probed=yes
D: alsa-mixer.c: Path analog-output (Analog Output), direction=1, priority=99, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=31, min_dB=-127,5, max_dB=12
D: alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes
D: alsa-mixer.c: Element Master Mono, direction=1, switch=2, volume=2, enumeration=0, required=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=no
D: alsa-mixer.c: Element Surround, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x60, n_channels=2, override_map=yes
D: alsa-mixer.c: Element Center, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x4900000000018, n_channels=1, override_map=yes
D: alsa-mixer.c: Element LFE, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x80, n_channels=1, override_map=yes
D: alsa-mixer.c: Element PCM, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x402b600000000f66, n_channels=2, override_map=yes
D: alsa-mixer.c: Element External Amplifier, direction=1, switch=4, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Option on (output-amplifier-on/Amplifier) index=1, priority=10
D: alsa-mixer.c: Option off (output-amplifier-off/No Amplifier) index=0, priority=0
D: alsa-mixer.c: Setting output-amplifier-on (Amplifier) priority=10
D: alsa-mixer.c: Setting output-amplifier-off (No Amplifier) priority=0
D: alsa-mixer.c: Path analog-output-mono (Analog Mono Output), direction=1, priority=50, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=31, min_dB=-81, max_dB=12
D: alsa-mixer.c: Element Master, direction=1, switch=2, volume=2, enumeration=0, required=0, required_absent=0, mask=0x6, n_channels=2, override_map=no
D: alsa-mixer.c: Element Master Mono, direction=1, switch=1, volume=1, enumeration=0, required=4, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes
D: alsa-mixer.c: Element Surround, direction=1, switch=2, volume=2, enumeration=0, required=0, required_absent=0, mask=0x6, n_channels=2, override_map=no
D: alsa-mixer.c: Element Center, direction=1, switch=2, volume=2, enumeration=0, required=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=no
D: alsa-mixer.c: Element LFE, direction=1, switch=2, volume=2, enumeration=0, required=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=no
D: alsa-mixer.c: Element PCM, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x402b600000000f66, n_channels=2, override_map=yes
D: alsa-mixer.c: Element External Amplifier, direction=1, switch=4, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Option on (output-amplifier-on/Amplifier) index=1, priority=10
D: alsa-mixer.c: Option off (output-amplifier-off/No Amplifier) index=0, priority=0
D: alsa-mixer.c: Setting output-amplifier-on (Amplifier) priority=10
D: alsa-mixer.c: Setting output-amplifier-off (No Amplifier) priority=0
D: alsa-mixer.c: Path analog-output-lfe-on-mono (Analog Output (LFE)), direction=1, priority=40, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=31, min_dB=-81, max_dB=12
D: alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes
D: alsa-mixer.c: Element Master Mono, direction=1, switch=1, volume=1, enumeration=0, required=4, required_absent=0, mask=0x80, n_channels=1, override_map=yes
D: alsa-mixer.c: Element Surround, direction=1, switch=2, volume=2, enumeration=0, required=0, required_absent=0, mask=0x6, n_channels=2, override_map=no
D: alsa-mixer.c: Element Center, direction=1, switch=2, volume=2, enumeration=0, required=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=no
D: alsa-mixer.c: Element LFE, direction=1, switch=2, volume=2, enumeration=0, required=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=no
D: alsa-mixer.c: Element PCM, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x402b600000000f66, n_channels=2, override_map=yes
D: alsa-mixer.c: Element External Amplifier, direction=1, switch=4, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Option on (output-amplifier-on/Amplifier) index=1, priority=10
D: alsa-mixer.c: Option off (output-amplifier-off/No Amplifier) index=0, priority=0
D: alsa-mixer.c: Setting output-amplifier-on (Amplifier) priority=10
D: alsa-mixer.c: Setting output-amplifier-off (No Amplifier) priority=0
D: alsa-mixer.c: Added 6 ports
I: sink.c: Created sink 0 "alsa_output.pci-0000_00_1f.5.analog-stereo" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: sink.c: alsa.resolution_bits = "16"
I: sink.c: device.api = "alsa"
I: sink.c: device.class = "sound"
I: sink.c: alsa.class = "generic"
I: sink.c: alsa.subclass = "generic-mix"
I: sink.c: alsa.name = "Intel 82801DB-ICH4"
I: sink.c: alsa.id = "Intel ICH"
I: sink.c: alsa.subdevice = "0"
I: sink.c: alsa.subdevice_name = "subdevice #0"
I: sink.c: alsa.device = "0"
I: sink.c: alsa.card = "0"
I: sink.c: alsa.card_name = "Intel 82801DB-ICH4"
I: sink.c: alsa.long_card_name = "Intel 82801DB-ICH4 with ALC650F at irq 11"
I: sink.c: alsa.driver_name = "snd_intel8x0"
I: sink.c: device.bus_path = "pci-0000:00:1f.5"
I: sink.c: sysfs.path = "/devices/pci0000:00/0000:00:1f.5/sound/card0"
I: sink.c: device.bus = "pci"
I: sink.c: device.vendor.id = "8086"
I: sink.c: device.vendor.name = "Intel Corporation"
I: sink.c: device.product.id = "24c5"
I: sink.c: device.product.name = "82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) AC'97 Audio Controller"
I: sink.c: device.form_factor = "internal"
I: sink.c: device.string = "front:0"
I: sink.c: device.buffering.buffer_size = "65536"
I: sink.c: device.buffering.fragment_size = "65536"
I: sink.c: device.access_mode = "mmap+timer"
I: sink.c: device.profile.name = "analog-stereo"
I: sink.c: device.profile.description = "Analog Stereo"
I: sink.c: device.description = "Internes Audio Analog Stereo"
I: sink.c: alsa.mixer_name = "Realtek ALC650F"
I: sink.c: alsa.components = "AC97a:414c4723"
I: sink.c: module-udev-detect.discovered = "1"
I: sink.c: device.icon_name = "audio-card-pci"
D: core-subscribe.c: Dropped redundant event due to change event.
I: source.c: Created source 0 "alsa_output.pci-0000_00_1f.5.analog-stereo.monitor" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: source.c: device.description = "Monitor of Internes Audio Analog Stereo"
I: source.c: device.class = "monitor"
I: source.c: alsa.card = "0"
I: source.c: alsa.card_name = "Intel 82801DB-ICH4"
I: source.c: alsa.long_card_name = "Intel 82801DB-ICH4 with ALC650F at irq 11"
I: source.c: alsa.driver_name = "snd_intel8x0"
I: source.c: device.bus_path = "pci-0000:00:1f.5"
I: source.c: sysfs.path = "/devices/pci0000:00/0000:00:1f.5/sound/card0"
I: source.c: device.bus = "pci"
I: source.c: device.vendor.id = "8086"
I: source.c: device.vendor.name = "Intel Corporation"
I: source.c: device.product.id = "24c5"
I: source.c: device.product.name = "82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) AC'97 Audio Controller"
I: source.c: device.form_factor = "internal"
I: source.c: device.string = "0"
I: source.c: module-udev-detect.discovered = "1"
I: source.c: device.icon_name = "audio-card-pci"
I: alsa-sink.c: Using 1,0 fragments of size 65536 bytes (371,52ms), buffer size is 65536 bytes (371,52ms)
I: alsa-sink.c: Time scheduling watermark is 20,00ms
D: alsa-sink.c: hwbuf_unused=0
D: alsa-sink.c: setting avail_min=15502
D: alsa-mixer.c: Activating path analog-output
D: alsa-mixer.c: Path analog-output (Analog Output), direction=1, priority=99, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=31, min_dB=-127,5, max_dB=12
D: alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes
D: alsa-mixer.c: Element Master Mono, direction=1, switch=2, volume=2, enumeration=0, required=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=no
D: alsa-mixer.c: Element Surround, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x60, n_channels=2, override_map=yes
D: alsa-mixer.c: Element Center, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x4900000000018, n_channels=1, override_map=yes
D: alsa-mixer.c: Element LFE, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x80, n_channels=1, override_map=yes
D: alsa-mixer.c: Element PCM, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x402b600000000f66, n_channels=2, override_map=yes
D: alsa-mixer.c: Element External Amplifier, direction=1, switch=4, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Option on (output-amplifier-on/Amplifier) index=1, priority=10
D: alsa-mixer.c: Option off (output-amplifier-off/No Amplifier) index=0, priority=0
D: alsa-mixer.c: Setting output-amplifier-on (Amplifier) priority=10
D: alsa-mixer.c: Setting output-amplifier-off (No Amplifier) priority=0
I: alsa-sink.c: Hardware volume ranges from -127,50 dB to 12,00 dB.
I: alsa-sink.c: Fixing base volume to -12,00 dB
I: alsa-sink.c: Using hardware volume control. Hardware dB scale supported.
I: alsa-sink.c: Using hardware mute control.
D: alsa-util.c: snd_pcm_dump():
D: alsa-util.c: Hardware PCM card 0 'Intel 82801DB-ICH4' device 0 subdevice 0
D: alsa-util.c: Its setup is:
D: alsa-util.c: stream : PLAYBACK
D: alsa-util.c: access : MMAP_INTERLEAVED
D: alsa-util.c: format : S16_LE
D: alsa-util.c: subformat : STD
D: alsa-util.c: channels : 2
D: alsa-util.c: rate : 44100
D: alsa-util.c: exact rate : 44100 (44100/1)
D: alsa-util.c: msbits : 16
D: alsa-util.c: buffer_size : 16384
D: alsa-util.c: period_size : 16384
D: alsa-util.c: period_time : 371519
D: alsa-util.c: tstamp_mode : ENABLE
D: alsa-util.c: period_step : 1
D: alsa-util.c: avail_min : 16384
D: alsa-util.c: period_event : 0
D: alsa-util.c: start_threshold : -1
D: alsa-util.c: stop_threshold : 1073741824
D: alsa-util.c: silence_threshold: 0
D: alsa-util.c: silence_size : 0
D: alsa-util.c: boundary : 1073741824
D: alsa-util.c: appl_ptr : 0
D: alsa-util.c: hw_ptr : 0
D: alsa-sink.c: Read hardware volume: 0: 42% 1: 42%
D: alsa-sink.c: Thread starting up
I: core-util.c: Successfully enabled SCHED_RR scheduling for thread, with priority 4, which is lower than the requested 5.
I: alsa-sink.c: Starting playback.
D: alsa-sink.c: Cutting sleep time for the initial iterations by half.
D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ...
D: alsa-util.c: Managed to open front:0
D: alsa-util.c: Maximum hw buffer size is 371 ms
D: alsa-util.c: Set buffer size first (to 88200 samples), period size second (to 88200 samples).
I: alsa-source.c: Successfully opened device front:0.
I: alsa-source.c: Selected mapping 'Analog Stereo' (analog-stereo).
I: alsa-source.c: Successfully enabled mmap() mode.
I: alsa-source.c: Successfully enabled timer-based scheduling mode.
I: (alsa-lib)control.c: Invalid CTL front:0
I: alsa-mixer.c: Unable to attach to mixer front:0: No such file or directory
D: alsa-sink.c: Cutting sleep time for the initial iterations by half.
I: alsa-mixer.c: Successfully attached to mixer 'hw:0'
D: alsa-mixer.c: Probing path 'analog-input'
D: alsa-mixer.c: Probe of element 'Mic' failed.
D: alsa-mixer.c: Probing path 'analog-input-microphone'
D: alsa-mixer.c: Probing path 'analog-input-linein'
D: alsa-mixer.c: Probing path 'analog-input'
D: alsa-mixer.c: Probing path 'analog-input-video'
D: alsa-mixer.c: Probing path 'analog-input-video'
D: alsa-mixer.c: Probe of element 'TV Tuner' failed.
D: alsa-mixer.c: Probing path 'analog-input-radio'
D: alsa-mixer.c: Probe of element 'FM' failed.
D: alsa-mixer.c: Probing path 'analog-input'
D: alsa-mixer.c: Probe of element 'Mic/Line' failed.
D: alsa-source.c: Probed mixer paths:
D: alsa-mixer.c: Path Set 0xa0743c8, direction=2, probed=yes
D: alsa-mixer.c: Path analog-input-microphone (Analog Microphone), direction=2, priority=100, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=15, min_dB=0, max_dB=22,5
D: alsa-mixer.c: Element Capture, direction=2, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x4037e00000000f66, n_channels=2, override_map=yes
D: alsa-mixer.c: Element Mic, direction=2, switch=1, volume=0, enumeration=0, required=4, required_absent=0, mask=0x0, n_channels=0, override_map=yes
D: alsa-mixer.c: Element Line, direction=2, switch=2, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Element Aux, direction=2, switch=2, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Element Video, direction=2, switch=2, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Element Mic Select, direction=2, switch=0, volume=0, enumeration=1, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Option Mic1 (input-microphone-1/Microphone 1) index=0, priority=20
D: alsa-mixer.c: Option Mic2 (input-microphone-2/Microphone 2) index=1, priority=19
D: alsa-mixer.c: Setting input-microphone-1 (Microphone 1) priority=20
D: alsa-mixer.c: Setting input-microphone-2 (Microphone 2) priority=19
D: alsa-mixer.c: Path analog-input-linein (Analog Line-In), direction=2, priority=90, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=15, min_dB=0, max_dB=22,5
D: alsa-mixer.c: Element Capture, direction=2, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x4037e00000000f66, n_channels=2, override_map=yes
D: alsa-mixer.c: Element Mic, direction=2, switch=2, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Element Line, direction=2, switch=1, volume=0, enumeration=0, required=4, required_absent=0, mask=0x0, n_channels=0, override_map=yes
D: alsa-mixer.c: Element Aux, direction=2, switch=2, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Element Video, direction=2, switch=2, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Path analog-input (Analog Input), direction=2, priority=90, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=15, min_dB=0, max_dB=22,5
D: alsa-mixer.c: Element Capture, direction=2, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x4037e00000000f66, n_channels=2, override_map=yes
D: alsa-mixer.c: Element Mic, direction=2, switch=2, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Element Line, direction=2, switch=2, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Element Aux, direction=2, switch=1, volume=0, enumeration=0, required=4, required_absent=0, mask=0x0, n_channels=0, override_map=yes
D: alsa-mixer.c: Element Video, direction=2, switch=2, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Path analog-input-video (Analog Video), direction=2, priority=70, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=15, min_dB=0, max_dB=22,5
D: alsa-mixer.c: Element Capture, direction=2, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x4037e00000000f66, n_channels=2, override_map=yes
D: alsa-mixer.c: Element Mic, direction=2, switch=2, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Element Line, direction=2, switch=2, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Element Aux, direction=2, switch=2, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Element Video, direction=2, switch=1, volume=0, enumeration=0, required=4, required_absent=0, mask=0x0, n_channels=0, override_map=yes
D: alsa-mixer.c: Added 5 ports
D: core-subscribe.c: Dropped redundant event due to change event.
I: source.c: Created source 1 "alsa_input.pci-0000_00_1f.5.analog-stereo" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: source.c: alsa.resolution_bits = "16"
I: source.c: device.api = "alsa"
I: source.c: device.class = "sound"
I: source.c: alsa.class = "generic"
I: source.c: alsa.subclass = "generic-mix"
I: source.c: alsa.name = "Intel 82801DB-ICH4"
I: source.c: alsa.id = "Intel ICH"
I: source.c: alsa.subdevice = "0"
I: source.c: alsa.subdevice_name = "subdevice #0"
I: source.c: alsa.device = "0"
I: source.c: alsa.card = "0"
I: source.c: alsa.card_name = "Intel 82801DB-ICH4"
I: source.c: alsa.long_card_name = "Intel 82801DB-ICH4 with ALC650F at irq 11"
I: source.c: alsa.driver_name = "snd_intel8x0"
I: source.c: device.bus_path = "pci-0000:00:1f.5"
I: source.c: sysfs.path = "/devices/pci0000:00/0000:00:1f.5/sound/card0"
I: source.c: device.bus = "pci"
I: source.c: device.vendor.id = "8086"
I: source.c: device.vendor.name = "Intel Corporation"
I: source.c: device.product.id = "24c5"
I: source.c: device.product.name = "82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) AC'97 Audio Controller"
I: source.c: device.form_factor = "internal"
I: source.c: device.string = "front:0"
I: source.c: device.buffering.buffer_size = "65536"
I: source.c: device.buffering.fragment_size = "65536"
I: source.c: device.access_mode = "mmap+timer"
I: source.c: device.profile.name = "analog-stereo"
I: source.c: device.profile.description = "Analog Stereo"
I: source.c: device.description = "Internes Audio Analog Stereo"
I: source.c: alsa.mixer_name = "Realtek ALC650F"
I: source.c: alsa.components = "AC97a:414c4723"
I: source.c: module-udev-detect.discovered = "1"
I: source.c: device.icon_name = "audio-card-pci"
I: alsa-source.c: Using 1,0 fragments of size 65536 bytes (371,52ms), buffer size is 65536 bytes (371,52ms)
I: alsa-source.c: Time scheduling watermark is 20,00ms
D: alsa-source.c: hwbuf_unused=0
D: alsa-source.c: setting avail_min=15502
D: alsa-mixer.c: Activating path analog-input-microphone
D: alsa-mixer.c: Path analog-input-microphone (Analog Microphone), direction=2, priority=100, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=15, min_dB=0, max_dB=22,5
D: alsa-mixer.c: Element Capture, direction=2, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x4037e00000000f66, n_channels=2, override_map=yes
D: alsa-mixer.c: Element Mic, direction=2, switch=1, volume=0, enumeration=0, required=4, required_absent=0, mask=0x0, n_channels=0, override_map=yes
D: alsa-mixer.c: Element Line, direction=2, switch=2, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Element Aux, direction=2, switch=2, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Element Video, direction=2, switch=2, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Element Mic Select, direction=2, switch=0, volume=0, enumeration=1, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no
D: alsa-mixer.c: Option Mic1 (input-microphone-1/Microphone 1) index=0, priority=20
D: alsa-mixer.c: Option Mic2 (input-microphone-2/Microphone 2) index=1, priority=19
D: alsa-mixer.c: Setting input-microphone-1 (Microphone 1) priority=20
D: alsa-mixer.c: Setting input-microphone-2 (Microphone 2) priority=19
I: alsa-source.c: Hardware volume ranges from 0,00 dB to 22,50 dB.
I: alsa-source.c: Fixing base volume to -22,50 dB
I: alsa-source.c: Using hardware volume control. Hardware dB scale supported.
I: alsa-source.c: Using hardware mute control.
D: alsa-util.c: snd_pcm_dump():
D: alsa-util.c: Hardware PCM card 0 'Intel 82801DB-ICH4' device 0 subdevice 0
D: alsa-util.c: Its setup is:
D: alsa-util.c: stream : CAPTURE
D: alsa-util.c: access : MMAP_INTERLEAVED
D: alsa-util.c: format : S16_LE
D: alsa-util.c: subformat : STD
D: alsa-util.c: channels : 2
D: alsa-util.c: rate : 44100
D: alsa-util.c: exact rate : 44100 (44100/1)
D: alsa-util.c: msbits : 16
D: alsa-util.c: buffer_size : 16384
D: alsa-util.c: period_size : 16384
D: alsa-util.c: period_time : 371519
D: alsa-util.c: tstamp_mode : ENABLE
D: alsa-util.c: period_step : 1
D: alsa-util.c: avail_min : 16384
D: alsa-util.c: period_event : 0
D: alsa-util.c: start_threshold : -1
D: alsa-util.c: stop_threshold : 1073741824
D: alsa-util.c: silence_threshold: 0
D: alsa-util.c: silence_size : 0
D: alsa-util.c: boundary : 1073741824
D: alsa-util.c: appl_ptr : 0
D: alsa-util.c: hw_ptr : 0
D: alsa-source.c: Read hardware volume: 0: 42% 1: 42%
D: alsa-source.c: Thread starting up
I: core-util.c: Successfully enabled SCHED_RR scheduling for thread, with priority 4, which is lower than the requested 5.
I: module.c: Loaded "module-alsa-card" (index: #4; argument: "device_id="0" name="pci-0000_00_1f.5" card_name="alsa_card.pci-0000_00_1f.5" tsched=yes ignore_dB=no card_properties="module-udev-detect.discovered=1"").
I: module-udev-detect.c: Card /devices/pci0000:00/0000:00:1f.5/sound/card0 (alsa_card.pci-0000_00_1f.5) module loaded.
I: module-udev-detect.c: Found 1 cards.
I: module.c: Loaded "module-udev-detect" (index: #5; argument: "").
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.21/modules/module-bluetooth-discover.so': failure
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.21/modules/module-esound-protocol-unix.so': success
I: alsa-source.c: Starting capture.
I: module.c: Loaded "module-esound-protocol-unix" (index: #6; argument: "").
I: module.c: Loaded "module-native-protocol-unix" (index: #7; argument: "").
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.21/modules/module-gconf.so': success
I: module.c: Loaded "module-gconf" (index: #8; argument: "").
D: core-subscribe.c: Dropped redundant event due to change event.
I: module-default-device-restore.c: Restored default sink 'alsa_output.pci-0000_00_1f.5.analog-stereo'.
D: core-subscribe.c: Dropped redundant event due to change event.
I: module-default-device-restore.c: Restored default source 'alsa_input.pci-0000_00_1f.5.analog-stereo'.
I: module.c: Loaded "module-default-device-restore" (index: #9; argument: "").
I: module.c: Loaded "module-rescue-streams" (index: #10; argument: "").
I: module.c: Loaded "module-always-sink" (index: #11; argument: "").
I: module.c: Loaded "module-intended-roles" (index: #12; argument: "").
D: module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_1f.5.analog-stereo becomes idle, timeout in 5 seconds.
D: module-suspend-on-idle.c: Source alsa_input.pci-0000_00_1f.5.analog-stereo becomes idle, timeout in 5 seconds.
I: module.c: Loaded "module-suspend-on-idle" (index: #13; argument: "").
D: dbus-util.c: Successfully connected to D-Bus system bus 50739e1c56883ceef7ea115e0000000f as :1.13
I: client.c: Created 0 "ConsoleKit Session /org/freedesktop/ConsoleKit/Session1"
D: module-console-kit.c: Added new session /org/freedesktop/ConsoleKit/Session1
I: client.c: Created 1 "ConsoleKit Session /org/freedesktop/ConsoleKit/Session2"
D: module-console-kit.c: Added new session /org/freedesktop/ConsoleKit/Session2
I: module.c: Loaded "module-console-kit" (index: #14; argument: "").
I: module.c: Loaded "module-position-event-sounds" (index: #15; argument: "").
W: main.c: Unable to contact D-Bus: org.freedesktop.DBus.Error.Spawn.ExecFailed: /bin/dbus-launch terminated abnormally with the following error: Autolaunch error: X11 initialization failed.
I: main.c: Start des Daemons abgeschlossen.
D: module-console-kit.c: dbus: interface=org.freedesktop.DBus, path=/org/freedesktop/DBus, member=NameAcquired
D: module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
D: alsa-sink.c: Wakeup from ALSA!
I: alsa-sink.c: Underrun!
I: alsa-sink.c: Increasing wakeup watermark to 30,00 ms
D: alsa-sink.c: Wakeup from ALSA!
I: alsa-sink.c: Underrun!
I: alsa-sink.c: Increasing wakeup watermark to 40,00 ms
D: alsa-sink.c: Wakeup from ALSA!
I: alsa-sink.c: Underrun!
I: alsa-sink.c: Increasing wakeup watermark to 50,00 ms
D: alsa-source.c: Wakeup from ALSA!
D: alsa-sink.c: Wakeup from ALSA!
I: alsa-sink.c: Underrun!
I: alsa-sink.c: Increasing wakeup watermark to 60,00 ms
I: module-suspend-on-idle.c: Source alsa_input.pci-0000_00_1f.5.analog-stereo idle for too long, suspending ...
D: source.c: Suspend cause of source alsa_input.pci-0000_00_1f.5.analog-stereo is 0x0004, suspending
I: alsa-source.c: Device suspended...
I: module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_1f.5.analog-stereo idle for too long, suspending ...
D: sink.c: Suspend cause of sink alsa_output.pci-0000_00_1f.5.analog-stereo is 0x0004, suspending
I: alsa-sink.c: Device suspended...
D: module-udev-detect.c: /dev/snd/controlC0 is accessible: yes

when I try to playback a mp3-file with aplay, it says:


aplay test.mp3
ALSA lib pulse.c:229:(pulse_connect) PulseAudio: Unable to connect: Zugriff abgelehnt

aplay: main:654: Fehler beim Öffnen des Gerätes: Connection refused


paplay test.mp3
Öffnen der Audio-Datei fehlgeschlagen.

sorry for the german, the last error means "open audio file failed"

What might be the problem? What information can i additionally provide?

schorsch
December 29th, 2010, 08:21 PM
Hi,


Just some general information for anyone that might be getting static or crackling sounds from their analog line-in or microphone input recordings on Maverick 10.10. After I upgraded to 10.10 this is the first version of Ubuntu or Pulseaudio that gave me problems when using the Pulseaudio equalizer LADSPA effects. I finally figured out to disable the equalizer effects and now my recordings don't have static in them. I didn't think the playback effects would have anything to do with recording, but they do for some reason. Anyway, it's something to be aware of.

could you please explain how you disabled the equalizer LADSPA effects as I have exactly the same problem?

Best regards

schorsch

mocha
January 17th, 2011, 08:52 AM
Hi,



could you please explain how you disabled the equalizer LADSPA effects as I have exactly the same problem?

Best regards

schorsch

Sorry, I don't check this thread much anymore. Anyway, the problem I had previously described turned out not to be due to LADSPA or this equalizer script.

JustJ
February 9th, 2011, 07:37 AM
Skype v2.1.0.81 uses pulse audio but the mic problem persists.

The trick is that the 'input' to configure only appears when it is in use!

Start the pulse audio manager paman (or run padevchooser system tray app and select it from there).
If you don't have it it's a short 'sudo apt-get install paman' away.

Switch to the 'Devices' tab.

Now make a skype call. Use the options/soundDevices/TestCall if necessary - you'll have 20 seconds... might want to read ahead.

As the phone call is connected watch the Pulse Audio Manager's device area. You'll see an:
alsa_input.pci-000xxxxxxx
or something similar appear somewhere in there.

Select alsa_input.pci-000xxxxxxx and then press the Properties button before it disappears.

If there's a "go to source" button click it ( it means you selected one of the sub-items as I did in the attached screenshots )

Now change the volume to 100% or even 200% ( mine was on %50 initially)!

That's it!

I suspect there's a way to change things further in ( in alsa?) as if you choose 200% it will fall back to a still usable 100% if you use another program to change volume.


oh, and if you don't have the ring, blip, plop and other sounds then go to your installed skype directory and type:
sudo mkdir /usr/share/skype
sudo cp -r sounds /usr/share/skype/

coolbeans777
February 12th, 2011, 09:10 PM
Will this work on Maverick Meerkat?

bartman2589
February 19th, 2011, 08:52 PM
Might be about time to update this since PulseAudio is used by default now in Kubuntu 10.10 (which really really sucks since the default configuration provides NO access to all of the input channels of the sound hardware used, making it impossible to configure various things like Mic 2 selection, Mic boost, mute or unmute aux/line in/phone or any other input).

domino1241
March 4th, 2011, 07:39 PM
Thank you for this.

kamelyan
April 11th, 2011, 07:53 AM
damo@damo-HP-Pavilion-dv7-Notebook-PC:~$ pkill pulseaudio; sleep 2; pulseaudio -vv
I: main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted
I: main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation not permitted
D: core-rtclock.c: Timer slack is set to 50 us.
D: core-util.c: RealtimeKit worked.
I: core-util.c: Successfully gained nice level -11.
I: main.c: This is PulseAudio 0.9.21-63-gd3efa-dirty
D: main.c: Compilation host: i686-pc-linux-gnu
D: main.c: Compilation CFLAGS: -g -O2 -g -Wall -O3 -Wall -W -Wextra -pipe -Wno-long-long -Winline -Wvla -Wno-overlength-strings -Wunsafe-loop-optimizations -Wundef -Wformat=2 -Wlogical-op -Wsign-compare -Wformat-security -Wmissing-include-dirs -Wformat-nonliteral -Wold-style-definition -Wpointer-arith -Winit-self -Wdeclaration-after-statement -Wfloat-equal -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wmissing-declarations -Wmissing-noreturn -Wshadow -Wendif-labels -Wcast-align -Wstrict-aliasing=2 -Wwrite-strings -Wno-unused-parameter -ffast-math -Wp,-D_FORTIFY_SOURCE=2 -fno-common -fdiagnostics-show-option
D: main.c: Running on host: Linux i686 2.6.35-28-generic-pae #49-Ubuntu SMP Tue Mar 1 14:58:06 UTC 2011
D: main.c: Found 2 CPUs.
I: main.c: Page size is 4096 bytes
D: main.c: Compiled with Valgrind support: no
D: main.c: Running in valgrind mode: no
D: main.c: Running in VM: no
D: main.c: Optimized build: yes
D: main.c: All asserts enabled.
I: main.c: Machine ID is ba7716074992340f68c5140000000007.
I: main.c: Session ID is ba7716074992340f68c5140000000007-1302485562.759307-1573744398.
I: main.c: Using runtime directory /home/damo/.pulse/ba7716074992340f68c5140000000007-runtime.
I: main.c: Using state directory /home/damo/.pulse.
I: main.c: Using modules directory /usr/lib/pulse-0.9.21/modules.
I: main.c: Running in system mode: no
E: pid.c: Daemon already running.
E: main.c: pa_pid_file_create() failed.
damo@damo-HP-Pavilion-dv7-Notebook-PC:~$


damo@damo-HP-Pavilion-dv7-Notebook-PC:~$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: SB [HDA ATI SB], device 0: STAC92xx Analog [STAC92xx Analog]
Subdevices: 0/1
Subdevice #0: subdevice #0
card 0: SB [HDA ATI SB], device 1: STAC92xx Digital [STAC92xx Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: HDMI [HDA ATI HDMI], device 3: ATI HDMI [ATI HDMI]
Subdevices: 1/1
Subdevice #0: subdevice #0
damo@damo-HP-Pavilion-dv7-Notebook-PC:~$

lucid lynx 11.04 1386

all aps appear in apps box even skype but no sound still get connection failed

damianogrady@ hotmail.com

linuxyogi
April 29th, 2011, 11:53 AM
As you know, Pulse audio defaults to two channels.

Why ?

Almost all mainboards these days comes with at least 6 channel integrated audio.

But because pulse audio defaults to 2 channels people can't enjoy surround sound (without doing some serious tweaking) despite having the hardware.

jono_tt
May 10th, 2011, 09:38 PM
I had this issue on my Dell XPS M1530. I resolved it by following the instructions in Appendix A point 1 above.

My specific actions were as follows:
rm -rf ~./pulse
sudo reboot 0


RESOLVED
I have commented on this so people searching for this issue on Dell XPS M1530 will find help

oklinuxyo
May 22nd, 2011, 08:34 PM
pleas update for current version 11.04 thank i can give cooki if fast uipdate ned eqaulizer

madtom1999
June 3rd, 2011, 09:57 AM
pkill pulseaudio; sleep 2; pulseaudio -vv
I: main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted
I: main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation not permitted
D: core-rtclock.c: Timer slack is set to 50 us.
D: core-util.c: RealtimeKit worked.
I: core-util.c: Successfully gained nice level -11.
I: main.c: This is PulseAudio 0.9.21-63-gd3efa-dirty
D: main.c: Compilation host: i686-pc-linux-gnu
D: main.c: Compilation CFLAGS: -g -O2 -g -Wall -O3 -Wall -W -Wextra -pipe -Wno-long-long -Winline -Wvla -Wno-overlength-strings -Wunsafe-loop-optimizations -Wundef -Wformat=2 -Wlogical-op -Wsign-compare -Wformat-security -Wmissing-include-dirs -Wformat-nonliteral -Wold-style-definition -Wpointer-arith -Winit-self -Wdeclaration-after-statement -Wfloat-equal -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wmissing-declarations -Wmissing-noreturn -Wshadow -Wendif-labels -Wcast-align -Wstrict-aliasing=2 -Wwrite-strings -Wno-unused-parameter -ffast-math -Wp,-D_FORTIFY_SOURCE=2 -fno-common -fdiagnostics-show-option
D: main.c: Running on host: Linux i686 2.6.35-28-generic #50-Ubuntu SMP Fri Mar 18 19:00:26 UTC 2011
D: main.c: Found 1 CPUs.
I: main.c: Page size is 4096 bytes
D: main.c: Compiled with Valgrind support: no
D: main.c: Running in valgrind mode: no
D: main.c: Running in VM: no
D: main.c: Optimised build: yes
D: main.c: All asserts enabled.
I: main.c: Machine ID is ea7ca271934b0b9b38bdb31d0000000b.
I: main.c: Session ID is ea7ca271934b0b9b38bdb31d0000000b-1307091093.672770-1645104095.
I: main.c: Using runtime directory /home/holefarm/.pulse/ea7ca271934b0b9b38bdb31d0000000b-runtime.
I: main.c: Using state directory /home/holefarm/.pulse.
I: main.c: Using modules directory /usr/lib/pulse-0.9.21/modules.
I: main.c: Running in system mode: no
E: pid.c: Daemon already running.
E: main.c: pa_pid_file_create() failed.
holefarm@dbell:~$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: CMI8738 [C-Media CMI8738], device 0: CMI8738-MC6 [C-Media PCI DAC/ADC]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: CMI8738 [C-Media CMI8738], device 1: CMI8738-MC6 [C-Media PCI 2nd DAC]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: CMI8738 [C-Media CMI8738], device 2: CMI8738-MC6 [C-Media PCI IEC958]
Subdevices: 1/1
Subdevice #0: subdevice #0
holefarm@dbell:~$
worked until a recent kernel upgrade
Any clues on how to fix?

nightfever
July 7th, 2011, 09:53 AM
bogdan@bogdan-945PL-S3P:~$ pulseaudio-equalizer
/usr/bin/pulseaudio-equalizer: line 68: [: 0: unary operator expected
/usr/bin/pulseaudio-equalizer: line 78: 10+: syntax error: operand expected (error token is "+")
(standard_in) 1: syntax error

This happened after reinstalling (completely removing with synaptic and installing again).
Any ideas?

EDIT: deleted .pulse in home folder and works now

I was wrong about editing preset file to add custom frequencies.
This being, PLEASE tell us how we can add them.

partyk1d24
October 7th, 2011, 08:21 PM
I am having some issues. I can hear sound fine but when I open up the Pulse VolumeManager the vu-meter isn't registering anything for the ALSA plugin. When I make a loopback adapter and try using that instead I still don't get anything. I tried to record the monitor but no audio was recorded. This seems to work fine for the internal mic, just not from the soundcard itself.

My aplay...

**** List of PLAYBACK Hardware Devices ****
card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog]
Subdevices: 0/1
Subdevice #0: subdevice #0
card 0: Intel [HDA Intel], device 3: HDMI 0 [HDMI 0]
Subdevices: 1/1
Subdevice #0: subdevice #0

The other command...
http://pastebin.com/d3NUnBGP

Thanks

rulet
October 8th, 2011, 12:31 PM
Hello.
I have 5.1 system. How to make redirection of bass with pulseaudio or with alsa from all channells to subwoofer?

oldsoundguy
October 8th, 2011, 06:04 PM
Hello.
I have 5.1 system. How to make redirection of bass with pulseaudio or with alsa from all channells to subwoofer?

If your 5.1 system is digital input only .. you will be hard pressed to get it functioning. I have never been able to use the digital out of computer to digital input on my surround amplifier system .. but the analog to analog works fine.
And have no issues with trying to re-direct the sub signal .. since the speaker system does that automatically.

rulet
October 8th, 2011, 06:15 PM
I have 5.1 with usual analog input.
You don't understand, the basses louds too much and I want manually to redirect all basses to subwoofer(with pulseaudio or with alsa) to make sound more clear(compared to that which gives win7 driver). The soundcard is Asus Xonar DX.
Just wonder how it will affect on the quality of a sound.

Medo42
January 17th, 2012, 10:11 AM
Thank you so much psyke83, I didn't have sound in my browsers for ages and rebuilding the config (which I had dragged from one Ubuntu to the next and the next...) finally made if work as it should.

skeedo
May 20th, 2012, 05:30 PM
I am absolutely stumped as to what is going on with my sound. I was listening to some music one day with a flash based player in Firefox, and my sound just completely died.

Thing is, everything seems to indicate my sound is working, I just don't hear anything. Volume meter bounces up and down when I play a Youtube video or use an application to play a sound file, and the application is displayed under the Playback tab in the Pulseaudio Volume Control.

I've tried wiping out ~/.pulse* and made sure I have necessary PulseAudio libraries and configuration utilities installed by doing apt-get command shown at beginning of this thread. Libflashsupport not installed according to Ubuntu Software Center. I've rebooted countless numbers of times as well.

I've messed with Sound Preferences and Pulseaudio Applet to no end, I'm certain that sound is not muted. Using Analog Stereo output profile, and I've tried both Analog Ouput and Analog Speaker ports under Output Devices.

I have a usb headphone jack that plugs directly into a USB port on my motherboard. I've checked my connections and everything is fine.


According to Appendix A, my problem is listed as:
The application does not play audio and does list an entry in the Playback tab;
- the application is using PulseAudio but there is a problem, such as: a bug in PulseAudio, a problem with your ALSA kernel module or libraries, or your PCM/Master volume is muted.
But as stated I killed all .pulse* dirs and made sure I had proper libraries and sound is not muted.

Not sure what's up with alsamixer:
pit87:~> alsamixer -Dhw
cannot open mixer: No such file or directory


Here is system information, any input to my problem will be greatly appreciated:

Ubuntu 10.04.4 LTS
Linux 2.6.32-41-generic-pae #89-Ubuntu SMP Fri Apr 27 23:59:24 UTC 2012 i686 GNU/Linux

pit87:~> aplay -l
**** List of PLAYBACK Hardware Devices ****
card 1: Set [USB Headphone Set], device 0: USB Audio [USB Audio]
Subdevices: 0/1
Subdevice #0: subdevice #0
pit87:~> pkill pulseaudio ; sleep 2 ; pulseaudio -vv
I: main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted
I: main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation not permitted
D: core-rtclock.c: Timer slack is set to 50 us.
D: core-util.c: RealtimeKit worked.
I: core-util.c: Successfully gained nice level -11.
I: main.c: This is PulseAudio 0.9.21-63-gd3efa-dirty
D: main.c: Compilation host: i486-pc-linux-gnu
D: main.c: Compilation CFLAGS: -g -O2 -g -Wall -O3 -Wall -W -Wextra -pipe -Wno-long-long -Winline -Wvla -Wno-overlength-strings -Wunsafe-loop-optimizations -Wundef -Wformat=2 -Wlogical-op -Wsign-compare -Wformat-security -Wmissing-include-dirs -Wformat-nonliteral -Wold-style-definition -Wpointer-arith -Winit-self -Wdeclaration-after-statement -Wfloat-equal -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wmissing-declarations -Wmissing-noreturn -Wshadow -Wendif-labels -Wcast-align -Wstrict-aliasing=2 -Wwrite-strings -Wno-unused-parameter -ffast-math -Wp,-D_FORTIFY_SOURCE=2 -fno-common -fdiagnostics-show-option
D: main.c: Running on host: Linux i686 2.6.32-41-generic-pae #89-Ubuntu SMP Fri Apr 27 23:59:24 UTC 2012
D: main.c: Found 2 CPUs.
I: main.c: Page size is 4096 bytes
D: main.c: Compiled with Valgrind support: no
D: main.c: Running in valgrind mode: no
D: main.c: Running in VM: no
D: main.c: Optimized build: yes
D: main.c: All asserts enabled.
I: main.c: Machine ID is 95b0cef255d8df3352d71fe14ce31cdb.
I: main.c: Session ID is 95b0cef255d8df3352d71fe14ce31cdb-1337470509.703044-589715843.
I: main.c: Using runtime directory /usr/home/staff/skeedo/.pulse/95b0cef255d8df3352d71fe14ce31cdb-runtime.
I: main.c: Using state directory /usr/home/staff/skeedo/.pulse.
I: main.c: Using modules directory /usr/lib/pulse-0.9.21/modules.
I: main.c: Running in system mode: no
E: pid.c: Daemon already running.
E: main.c: pa_pid_file_create() failed.

jcoles
July 1st, 2012, 11:43 AM
Are these instructions still valid for 12.04?
Pulseaudio is still a horror show. Volume control adjusts Speaker, not master, for example. Sometimes Sound Settings cannot connect to PulseAudio. No app available to ease configuration.

12.04 is a LTS release but is really not ready for prime time. It reports frequent application crashes that don't get resolved with updates.

Emopunk
August 19th, 2012, 01:20 PM
Hello to everyone. I tried to install Pulseaudio equalizer on Ubuntu 12.04 and it didn't work as expected. Moreover, I am now missing the volume control icon from the upper panel. Can someone please explain to me how to restore the control icon?

Thanks.