[Solved]Alsa/pulseaudio going mute
Logging out/in with awesome/slim seems to have audio at first, but after awhile ssems to switch to pulseaudio, but then pulseaudio fails to recognize audio hardware as anything but dummy output.
Last edited by nomorewindows (2014-06-07 00:48:57)
emeres wrote:fuser -v /dev/snd/*
USER PID ACCESS COMMAND
/dev/snd/controlC0: <user> xxxx F.... plugin-containe
/dev/snd/pcmC0D0p: <user> xxxx F...m plugin-containe
/dev/snd/timer: <user> xxxx f.... plugin-containe
Looks like to me that plugin-container is the problem.
emeres wrote:
That would be probably flash. What if you kill plugin-container?
fuser -v /dev/snd/*
pkill -9 plugin-container
It goes away with exiting firefox and nothing shows up with
fuser -v /dev/snd/*
when the failure occurs. No applications with sound work, and fuser doesn't show anything.
Last edited by nomorewindows (2014-06-07 00:28:19)
Similar Messages
-
Why can't I listen to music while charging my iPhone 4S? As soon as i plug in the phone goes mute
Why can't I listen to my music while charging my iPhone 4S? As soon as I plug in my phone goes mute!
reset the device (home and sleep/wake button for 5-10sec)
make sure mute isnt turned on or the ringer/silent switch
-double tap home and swipe the bottom window to the left and make sure the mute button hasnt been toggled (if its set up there, might be screen lock depending on settings)
reset all settings (settings >general >reset)
if all else fails try a backup/restore -
My FM Radio app randomly goes mute. I already cleaned the app data, did a software repair, what seemed to be a factory reset. But the problem continues.
There is another thread going on with the same problem.
http://talk.sonymobile.com/t5/Xperia-Z1-Compact/FM-Radio-died/m-p/590479#U590479 -
When I share a video with fcpx and it goes to Quicktime, the last few seconds goes mute. I checked the clip and sent that the same route and all is fine. It is only when I send the whole 3gb video. Any ideas?
paul71054 wrote:
... Any ideas?
add a few seconds black to your project.
Try again.
could be some really weird codec problem, with 'cut' GOPs', .. somethin'... -
Okay, so I don't know if this is just me or not... But Skype has this tendency to go mute. Both during calls and IMs. And I mean entirely mute. No sound notifications for the IMs and during calls neither myself or the person on the other end of the call can hear anything. Now, I have never once touched the sound settings for Skype and my background sounds continue to play without interuption. Sometimes, I can get a temporary fix to this by restarting my computer a ridiculous number of times (most recent amounts to five times) but I can always guarantee that it goes mute once more when I've not touched anything regarding Skype. Is this just me? Or anyone else having this issue? -Note: I have the latest Skype. I've actually attempted to uninstalll and reinstall a number of times.
Okay, so I don't know if this is just me or not... But Skype has this tendency to go mute. Both during calls and IMs. And I mean entirely mute. No sound notifications for the IMs and during calls neither myself or the person on the other end of the call can hear anything. Now, I have never once touched the sound settings for Skype and my background sounds continue to play without interuption. Sometimes, I can get a temporary fix to this by restarting my computer a ridiculous number of times (most recent amounts to five times) but I can always guarantee that it goes mute once more when I've not touched anything regarding Skype. Is this just me? Or anyone else having this issue? -Note: I have the latest Skype. I've actually attempted to uninstalll and reinstall a number of times.
-
Sometimes an app goes mute, and it is not something about the volume, it just doesn´t have any sound any more. What should I do?
If you lose sounds for keyboard clicks, games or other apps, email notifications and other notifications, system sounds may have been muted.
System sounds can be muted and controlled two different ways. The screen lock rotation can be controlled in the same manner as well.
Settings>General>Use Side Switch to: Mute System sounds. If this option is selected, the switch on the side of the iPad above the volume rocker will mute system sounds.
If you choose Lock Screen Rotation, then the switch locks the screen. If the screen is locked, you will see a lock icon in the upper right corner next to the battery indicator gauge.
If you have the side switch set to lock screen rotation then the system sound control is in the task bar. Double tap the home button and in the task bar at the bottom, swipe all the way to the right. The speaker icon is all the way to the left. Tap on it and system sounds will return.
If you have the side switch set to mute system sounds, then the screen lock rotation can be accessed via the task bar in the same manner as described above.
This support article from Apple explains how the side switch works.
http://support.apple.com/kb/HT4085 -
[Solved]No sound anymore (ALSA/PulseAudio)
Ok so I have no idea why I have no sound anymore, I didn't install anything...
Anyway, that's what I've tried:
-Check if mixer are all up in alsamixer
-Try to play a bit with pavucontrol
-Reboot
-Rerun pulseaudio with verbose
-Mute and unmute with physical button (I don't know why but I used to get this problem and it used to be solved just doing this)
-Googgling and searching this forum
Here are some more info
Pulseaudio logs:
└┼─$─┤▶ pulseaudio -k; pulseaudio -vvvvv
E: [pulseaudio] main.c: Failed to kill daemon: No such process
I: [pulseaudio] main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted
I: [pulseaudio] main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation not permitted
D: [pulseaudio] core-rtclock.c: Timer slack is set to 50 us.
D: [pulseaudio] core-util.c: RealtimeKit worked.
I: [pulseaudio] core-util.c: Successfully gained nice level -11.
I: [pulseaudio] main.c: This is PulseAudio 5.0
D: [pulseaudio] main.c: Compilation host: x86_64-unknown-linux-gnu
D: [pulseaudio] main.c: Compilation CFLAGS: -march=x86-64 -mtune=generic -O2 -pipe -fstack-protector --param=ssp-buffer-size=4 -Wall -W -Wextra -Wno-long-long -Wno-overlength-strings -Wunsafe-loop-optimizations -Wundef -Wformat=2 -Wlogical-op -Wsign-compare -Wformat-security -Wmissing-include-dirs -Wformat-nonliteral -Wpointer-arith -Winit-self -Wdeclaration-after-statement -Wfloat-equal -Wmissing-prototypes -Wredundant-decls -Wmissing-declarations -Wmissing-noreturn -Wshadow -Wendif-labels -Wcast-align -Wstrict-aliasing -Wwrite-strings -Wno-unused-parameter -ffast-math -fno-common -fdiagnostics-show-option
D: [pulseaudio] main.c: Running on host: Linux x86_64 3.14.1-1-ARCH #1 SMP PREEMPT Mon Apr 14 20:40:47 CEST 2014
D: [pulseaudio] main.c: Found 4 CPUs.
I: [pulseaudio] main.c: Page size is 4096 bytes
D: [pulseaudio] main.c: Compiled with Valgrind support: no
D: [pulseaudio] main.c: Running in valgrind mode: no
D: [pulseaudio] main.c: Running in VM: no
D: [pulseaudio] main.c: Optimized build: yes
D: [pulseaudio] main.c: FASTPATH defined, only fast path asserts disabled.
I: [pulseaudio] main.c: Machine ID is a3af3988434b49f3b8af2ab66eba685b.
I: [pulseaudio] main.c: Session ID is c1.
I: [pulseaudio] main.c: Using runtime directory /run/user/1001/pulse.
I: [pulseaudio] main.c: Using state directory /home/unkn0wn/.config/pulse.
I: [pulseaudio] main.c: Using modules directory /usr/lib/pulse-5.0/modules.
I: [pulseaudio] main.c: Running in system mode: no
I: [pulseaudio] main.c: Fresh high-resolution timers available! Bon appetit!
D: [pulseaudio] 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
I: [pulseaudio] cpu-x86.c: CPU flags: CMOV MMX SSE SSE2 SSE3 SSSE3 SSE4_1 SSE4_2
I: [pulseaudio] svolume_mmx.c: Initialising MMX optimized volume functions.
I: [pulseaudio] remap_mmx.c: Initialising MMX optimized remappers.
I: [pulseaudio] svolume_sse.c: Initialising SSE2 optimized volume functions.
I: [pulseaudio] remap_sse.c: Initialising SSE2 optimized remappers.
I: [pulseaudio] sconv_sse.c: Initialising SSE2 optimized conversions.
I: [pulseaudio] svolume_orc.c: Initialising ORC optimized volume functions.
D: [pulseaudio] database-tdb.c: Opened TDB database '/home/unkn0wn/.config/pulse/a3af3988434b49f3b8af2ab66eba685b-device-volumes.tdb'
I: [pulseaudio] module-device-restore.c: Successfully opened database file '/home/unkn0wn/.config/pulse/a3af3988434b49f3b8af2ab66eba685b-device-volumes'.
I: [pulseaudio] module.c: Loaded "module-device-restore" (index: #0; argument: "").
D: [pulseaudio] database-tdb.c: Opened TDB database '/home/unkn0wn/.config/pulse/a3af3988434b49f3b8af2ab66eba685b-stream-volumes.tdb'
I: [pulseaudio] module-stream-restore.c: Successfully opened database file '/home/unkn0wn/.config/pulse/a3af3988434b49f3b8af2ab66eba685b-stream-volumes'.
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1 added for object /org/pulseaudio/stream_restore1
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry0
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry1
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry2
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry3
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry4
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry5
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry6
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry7
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry8
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry9
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry10
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry11
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry12
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry13
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry14
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry15
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry16
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry17
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry18
D: [pulseaudio] protocol-dbus.c: Interface org.PulseAudio.Ext.StreamRestore1.RestoreEntry added for object /org/pulseaudio/stream_restore1/entry19
I: [pulseaudio] module.c: Loaded "module-stream-restore" (index: #1; argument: "").
D: [pulseaudio] database-tdb.c: Opened TDB database '/home/unkn0wn/.config/pulse/a3af3988434b49f3b8af2ab66eba685b-card-database.tdb'
I: [pulseaudio] module-card-restore.c: Successfully opened database file '/home/unkn0wn/.config/pulse/a3af3988434b49f3b8af2ab66eba685b-card-database'.
I: [pulseaudio] module.c: Loaded "module-card-restore" (index: #2; argument: "").
I: [pulseaudio] module.c: Loaded "module-augment-properties" (index: #3; argument: "").
I: [pulseaudio] module.c: Loaded "module-switch-on-port-available" (index: #4; argument: "").
D: [pulseaudio] module.c: Checking for existence of '/usr/lib/pulse-5.0/modules/module-udev-detect.so': success
D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes
D: [pulseaudio] module-udev-detect.c: /devices/pci0000:00/0000:00:1b.0/sound/card0 is busy: yes
D: [pulseaudio] module-udev-detect.c: Ignoring /devices/platform/thinkpad_acpi/sound/card29, because marked so.
I: [pulseaudio] module-udev-detect.c: Found 1 cards.
I: [pulseaudio] module.c: Loaded "module-udev-detect" (index: #5; argument: "").
D: [pulseaudio] module.c: Checking for existence of '/usr/lib/pulse-5.0/modules/module-jackdbus-detect.so': success
D: [pulseaudio] dbus-util.c: Successfully connected to D-Bus session bus c9a6328a5bfaef44136d134253502922 as :1.32
D: [pulseaudio] module-jackdbus-detect.c: jackdbus isn't running.
I: [pulseaudio] module.c: Loaded "module-jackdbus-detect" (index: #6; argument: "channels=2").
D: [pulseaudio] module.c: Checking for existence of '/usr/lib/pulse-5.0/modules/module-bluetooth-policy.so': success
I: [pulseaudio] module.c: Loaded "module-bluetooth-policy" (index: #7; argument: "").
D: [pulseaudio] module.c: Checking for existence of '/usr/lib/pulse-5.0/modules/module-bluetooth-discover.so': success
D: [pulseaudio] module.c: Checking for existence of '/usr/lib/pulse-5.0/modules/module-bluez5-discover.so': success
D: [pulseaudio] dbus-util.c: Successfully connected to D-Bus system bus 54d6e91cb1d72b925a540cf453502907 as :1.40
I: [pulseaudio] module.c: Loaded "module-bluez5-discover" (index: #9; argument: "").
D: [pulseaudio] module.c: Checking for existence of '/usr/lib/pulse-5.0/modules/module-bluez4-discover.so': failure
I: [pulseaudio] module.c: Loaded "module-bluetooth-discover" (index: #8; argument: "").
D: [pulseaudio] module.c: Checking for existence of '/usr/lib/pulse-5.0/modules/module-esound-protocol-unix.so': success
I: [pulseaudio] module.c: Loaded "module-esound-protocol-unix" (index: #10; argument: "").
I: [pulseaudio] module.c: Loaded "module-native-protocol-unix" (index: #11; argument: "").
D: [pulseaudio] module.c: Checking for existence of '/usr/lib/pulse-5.0/modules/module-gconf.so': success
I: [pulseaudio] module.c: Loaded "module-gconf" (index: #12; argument: "").
I: [pulseaudio] module-default-device-restore.c: Saved default sink 'auto_null' not existent, not restoring default sink setting.
I: [pulseaudio] module-default-device-restore.c: Saved default source 'auto_null.monitor' not existent, not restoring default source setting.
I: [pulseaudio] module.c: Loaded "module-default-device-restore" (index: #13; argument: "").
I: [pulseaudio] module.c: Loaded "module-rescue-streams" (index: #14; argument: "").
D: [pulseaudio] module-always-sink.c: Autoloading null-sink as no other sinks detected.
I: [pulseaudio] module-device-restore.c: Restoring volume for sink auto_null: front-left: 99957 / 153%, front-right: 99957 / 153%
I: [pulseaudio] module-device-restore.c: Restoring mute state for sink auto_null.
I: [pulseaudio] sink.c: Created sink 0 "auto_null" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: [pulseaudio] sink.c: device.description = "Dummy Output"
I: [pulseaudio] sink.c: device.class = "abstract"
I: [pulseaudio] sink.c: device.icon_name = "audio-card"
D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event.
I: [pulseaudio] module-device-restore.c: Restoring mute state for source auto_null.monitor.
I: [pulseaudio] source.c: Created source 0 "auto_null.monitor" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: [pulseaudio] source.c: device.description = "Monitor of Dummy Output"
I: [pulseaudio] source.c: device.class = "monitor"
I: [pulseaudio] source.c: device.icon_name = "audio-input-microphone"
D: [null-sink] module-null-sink.c: Thread starting up
D: [pulseaudio] module-device-restore.c: Could not set format on sink auto_null
I: [pulseaudio] module.c: Loaded "module-null-sink" (index: #16; argument: "sink_name=auto_null sink_properties='device.description="Dummy Output"'").
I: [pulseaudio] module.c: Loaded "module-always-sink" (index: #15; argument: "").
I: [pulseaudio] module.c: Loaded "module-intended-roles" (index: #17; argument: "").
D: [pulseaudio] module-suspend-on-idle.c: Sink auto_null becomes idle, timeout in 5 seconds.
I: [pulseaudio] module.c: Loaded "module-suspend-on-idle" (index: #18; argument: "").
D: [pulseaudio] module.c: Checking for existence of '/usr/lib/pulse-5.0/modules/module-console-kit.so': success
I: [pulseaudio] module.c: Loaded "module-console-kit" (index: #19; argument: "").
D: [pulseaudio] module.c: Checking for existence of '/usr/lib/pulse-5.0/modules/module-systemd-login.so': success
I: [pulseaudio] client.c: Created 0 "Login Session c1"
D: [pulseaudio] module-systemd-login.c: Added new session c1
I: [pulseaudio] module.c: Loaded "module-systemd-login" (index: #20; argument: "").
I: [pulseaudio] module.c: Loaded "module-position-event-sounds" (index: #21; argument: "").
D: [pulseaudio] module-role-cork.c: Using role 'phone' as trigger role.
D: [pulseaudio] module-role-cork.c: Using roles 'music' and 'video' as cork roles.
I: [pulseaudio] module.c: Loaded "module-role-cork" (index: #22; argument: "").
I: [pulseaudio] module.c: Loaded "module-filter-heuristics" (index: #23; argument: "").
I: [pulseaudio] module.c: Loaded "module-filter-apply" (index: #24; argument: "").
D: [pulseaudio] main.c: Got org.PulseAudio1!
D: [pulseaudio] main.c: Got org.pulseaudio.Server!
I: [pulseaudio] main.c: Daemon startup complete.
E: [pulseaudio] bluez5-util.c: GetManagedObjects() failed: org.freedesktop.DBus.Error.ServiceUnknown: The name org.bluez was not provided by any .service files
I: [pulseaudio] module-suspend-on-idle.c: Sink auto_null idle f
└┼─$─┤▶ loginctl session-status c1
c1 - unkn0wn (1001)
Since: Thu 2014-04-17 21:18:58 CEST; 16min ago
Leader: 1048 (slim)
Seat: seat0; vc7
Display: :0.0
Remote: user root
Service: slim; type x11; class user
State: active
Unit: session-c1.scope
|-1048 /usr/bin/slim -nodaemon
|-2206 /usr/bin/openbox --startup /usr/lib/openbox/openbox-autostart OPENBOX
|-2214 dbus-launch --sh-syntax --exit-with-session
|-2215 /usr/bin/dbus-daemon --fork --print-pid 5 --print-address 7 --session
|-2227 tint2
|-2230 xcompmgr -CfF
|-2231 volumeicon
|-2236 conky -q
|-2238 /usr/lib/at-spi2-core/at-spi-bus-launcher
|-2252 /usr/bin/dbus-daemon --config-file=/etc/at-spi2/accessibility.conf --no...
|-2255 /usr/lib/at-spi2-core/at-spi2-registryd --use-gnome-session
|-2259 /usr/lib/gvfs/gvfsd
|-2263 /usr/lib/gvfs/gvfsd-fuse /run/user/1001/gvfs -f -o big_writes
|-2278 /usr/lib32/skype/skype
|-2320 /usr/lib/xfce4/notifyd/xfce4-notifyd
|-2322 /usr/lib/xfce4/xfconf/xfconfd
|-2376 firefox
|-2416 /usr/lib/virtualbox/VBoxXPCOMIPCD
-I see the mixer going up and down in pavucontrol if I play some music
└┼─$─┤▶ loginctl session-status c1
c1 - unkn0wn (1001)
Since: Thu 2014-04-17 21:18:58 CEST; 16min ago
Leader: 1048 (slim)
Seat: seat0; vc7
Display: :0.0
Remote: user root
Service: slim; type x11; class user
State: active
Unit: session-c1.scope
|-1048 /usr/bin/slim -nodaemon
|-2206 /usr/bin/openbox --startup /usr/lib/openbox/openbox-autostart OPENBOX
|-2214 dbus-launch --sh-syntax --exit-with-session
|-2215 /usr/bin/dbus-daemon --fork --print-pid 5 --print-address 7 --session
|-2227 tint2
|-2230 xcompmgr -CfF
|-2231 volumeicon
|-2236 conky -q
|-2238 /usr/lib/at-spi2-core/at-spi-bus-launcher
|-2252 /usr/bin/dbus-daemon --config-file=/etc/at-spi2/accessibility.conf --no...
|-2255 /usr/lib/at-spi2-core/at-spi2-registryd --use-gnome-session
|-2259 /usr/lib/gvfs/gvfsd
|-2263 /usr/lib/gvfs/gvfsd-fuse /run/user/1001/gvfs -f -o big_writes
|-2278 /usr/lib32/skype/skype
|-2320 /usr/lib/xfce4/notifyd/xfce4-notifyd
|-2322 /usr/lib/xfce4/xfconf/xfconfd
|-2376 firefox
|-2416 /usr/lib/virtualbox/VBoxXPCOMIPCD
└┼─$─┤▶ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: PCH [HDA Intel PCH], device 0: CX20590 Analog [CX20590 Analog]
Subdevices: 0/1
Subdevice #0: subdevice #0
card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2]
Subdevices: 1/1
Subdevice #0: subdevice #0
└┼─$─┤▶ aplay -L
null
Discard all samples (playback) or generate zero samples (capture)
pulse
PulseAudio Sound Server
default
Default ALSA Output (currently PulseAudio Sound Server)
sysdefault:CARD=PCH
HDA Intel PCH, CX20590 Analog
Default Audio Device
front:CARD=PCH,DEV=0
HDA Intel PCH, CX20590 Analog
Front speakers
surround40:CARD=PCH,DEV=0
HDA Intel PCH, CX20590 Analog
4.0 Surround output to Front and Rear speakers
surround41:CARD=PCH,DEV=0
HDA Intel PCH, CX20590 Analog
4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=PCH,DEV=0
HDA Intel PCH, CX20590 Analog
5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=PCH,DEV=0
HDA Intel PCH, CX20590 Analog
5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=PCH,DEV=0
HDA Intel PCH, CX20590 Analog
7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
hdmi:CARD=PCH,DEV=0
HDA Intel PCH, HDMI 0
HDMI Audio Output
hdmi:CARD=PCH,DEV=1
HDA Intel PCH, HDMI 1
HDMI Audio Output
hdmi:CARD=PCH,DEV=2
HDA Intel PCH, HDMI 2
HDMI Audio Output
Don't really know what else to provide...
About my setup it's a thinkpad t420 not plugged in any hdmi screen or simething like that
Thank you
Last edited by Truc (2014-04-19 13:15:28)└┼─$─┤▶ fuser -v /dev/snd/*
USER PID ACCESS COMMAND
/dev/snd/controlC0: unkn0wn 2231 F.... volumeicon
└┼─$─┤▶ speaker-test -c 2 -t wav -D plughw:0,0
speaker-test 1.0.27.2
Playback device is plughw:0,0
Stream parameters are 48000Hz, S16_LE, 2 channels
WAV file(s)
Playback open error: -16,Device or resource busy
It says it's busy but I don't see anything that could use it. I quit skype before trying -
[SOLVED] Alsa: How to disable auto-mute when plugging in headphones?
I have a set of headphones plugged into my workstation permanently (the headphone jack is hard to reach so I don't want to plug those headphones in and out) as well as some speakers. Jack sensing works perfectly - when I unplug the headphones, the speakers are unmuted and when I plug the headphones back in the speakers are muted. Now as I said, my headphones are plugged in permanently and occasionally I want to have sound on the speakers at the same time. So far I have not found a way to have both headphones and speakers unmuted simultaneously.
I originally thought this was a problem with Pulseaudio (I am using Cinnamon as my desktop) but when I directly go to the Alsa hw device I get exactly the same behavior. The sound chip is a built-in VIA VT1818S and aplay displays the correct cards and devices:
$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: SB [HDA ATI SB], device 0: VT1818S Analog [VT1818S Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: SB [HDA ATI SB], device 1: VT1818S Digital [VT1818S Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: SB [HDA ATI SB], device 2: VT1818S HP [VT1818S HP]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: NVidia [HDA NVidia], device 3: HDMI 0 [HDMI 0]
Subdevices: 1/1
Subdevice #0: subdevice #0
I verified with "mpg123 -a hw:0,0" that I get no sound on the speakers when the headphones are plugged in and vice versa. I tried a lot of different settings with alsamixer and amixer to no avail. These are my current settings:
$ amixer -c 0 contents
numid=42,iface=CARD,name='Front Headphone Jack'
; type=BOOLEAN,access=r-------,values=1
: values=on
numid=44,iface=CARD,name='Front Mic Jack'
; type=BOOLEAN,access=r-------,values=1
: values=off
numid=46,iface=CARD,name='HDMI Phantom Jack'
; type=BOOLEAN,access=r-------,values=1
: values=on
numid=45,iface=CARD,name='Line Jack'
; type=BOOLEAN,access=r-------,values=1
: values=off
numid=40,iface=CARD,name='Line Out CLFE Jack'
; type=BOOLEAN,access=r-------,values=1
: values=off
numid=38,iface=CARD,name='Line Out Front Jack'
; type=BOOLEAN,access=r-------,values=1
: values=on
numid=41,iface=CARD,name='Line Out Side Jack'
; type=BOOLEAN,access=r-------,values=1
: values=off
numid=39,iface=CARD,name='Line Out Surround Jack'
; type=BOOLEAN,access=r-------,values=1
: values=off
numid=43,iface=CARD,name='Rear Mic Jack'
; type=BOOLEAN,access=r-------,values=1
: values=off
numid=47,iface=CARD,name='SPDIF Phantom Jack',index=1
; type=BOOLEAN,access=r-------,values=1
: values=on
numid=37,iface=MIXER,name='Master Playback Switch'
; type=BOOLEAN,access=rw------,values=1
: values=on
numid=36,iface=MIXER,name='Master Playback Volume'
; type=INTEGER,access=rw---R--,values=1,min=0,max=42,step=0
: values=23
| dBscale-min=-63.00dB,step=1.50dB,mute=0
numid=13,iface=MIXER,name='Headphone Playback Switch'
; type=BOOLEAN,access=rw------,values=2
: values=on,on
numid=12,iface=MIXER,name='PCM Playback Switch'
; type=BOOLEAN,access=rw------,values=2
: values=on,on
numid=11,iface=MIXER,name='PCM Playback Volume'
; type=INTEGER,access=rw---R--,values=2,min=0,max=31,step=0
: values=31,31
| dBscale-min=-34.50dB,step=1.50dB,mute=0
numid=28,iface=MIXER,name='Front Mic Boost Volume'
; type=INTEGER,access=rw---R--,values=2,min=0,max=3,step=0
: values=0,0
| dBscale-min=0.00dB,step=10.25dB,mute=0
numid=24,iface=MIXER,name='Front Mic Playback Switch'
; type=BOOLEAN,access=rw------,values=2
: values=on,on
numid=23,iface=MIXER,name='Front Mic Playback Volume'
; type=INTEGER,access=rw---R--,values=2,min=0,max=31,step=0
: values=31,31
| dBscale-min=-34.50dB,step=1.50dB,mute=0
numid=2,iface=MIXER,name='Front Playback Switch'
; type=BOOLEAN,access=rw------,values=2
: values=on,on
numid=1,iface=MIXER,name='Front Playback Volume'
; type=INTEGER,access=rw---R--,values=2,min=0,max=42,step=0
: values=42,42
| dBscale-min=-63.00dB,step=1.50dB,mute=0
numid=4,iface=MIXER,name='Surround Playback Switch'
; type=BOOLEAN,access=rw------,values=2
: values=on,on
numid=3,iface=MIXER,name='Surround Playback Volume'
; type=INTEGER,access=rw---R--,values=2,min=0,max=42,step=0
: values=42,42
| dBscale-min=-63.00dB,step=1.50dB,mute=0
numid=6,iface=MIXER,name='Center Playback Switch'
; type=BOOLEAN,access=rw------,values=1
: values=on
numid=5,iface=MIXER,name='Center Playback Volume'
; type=INTEGER,access=rw---R--,values=1,min=0,max=42,step=0
: values=42
| dBscale-min=-63.00dB,step=1.50dB,mute=0
numid=8,iface=MIXER,name='LFE Playback Switch'
; type=BOOLEAN,access=rw------,values=1
: values=on
numid=7,iface=MIXER,name='LFE Playback Volume'
; type=INTEGER,access=rw---R--,values=1,min=0,max=42,step=0
: values=42
| dBscale-min=-63.00dB,step=1.50dB,mute=0
numid=26,iface=MIXER,name='Line Playback Switch'
; type=BOOLEAN,access=rw------,values=2
: values=on,on
numid=25,iface=MIXER,name='Line Playback Volume'
; type=INTEGER,access=rw---R--,values=2,min=0,max=31,step=0
: values=31,31
| dBscale-min=-34.50dB,step=1.50dB,mute=0
numid=16,iface=MIXER,name='Capture Switch'
; type=BOOLEAN,access=rw------,values=2
: values=on,on
numid=18,iface=MIXER,name='Capture Switch',index=1
; type=BOOLEAN,access=rw------,values=2
: values=on,on
numid=15,iface=MIXER,name='Capture Volume'
; type=INTEGER,access=rw---R--,values=2,min=0,max=31,step=0
: values=0,0
| dBscale-min=-16.50dB,step=1.50dB,mute=0
numid=17,iface=MIXER,name='Capture Volume',index=1
; type=INTEGER,access=rw---R--,values=2,min=0,max=31,step=0
: values=0,0
| dBscale-min=-16.50dB,step=1.50dB,mute=0
numid=14,iface=MIXER,name='Loopback Mixing'
; type=ENUMERATED,access=rw------,values=1,items=2
; Item #0 'Disabled'
; Item #1 'Enabled'
: values=0
numid=35,iface=MIXER,name='IEC958 Default PCM Playback Switch'
; type=BOOLEAN,access=rw------,values=1
: values=off
numid=31,iface=MIXER,name='IEC958 Playback Con Mask'
; type=IEC958,access=r-------,values=1
: values=[AES0=0x0f AES1=0xff AES2=0x00 AES3=0x00]
numid=32,iface=MIXER,name='IEC958 Playback Pro Mask'
; type=IEC958,access=r-------,values=1
: values=[AES0=0x0f AES1=0x00 AES2=0x00 AES3=0x00]
numid=33,iface=MIXER,name='IEC958 Playback Default'
; type=IEC958,access=rw------,values=1
: values=[AES0=0x04 AES1=0x00 AES2=0x00 AES3=0x00]
numid=34,iface=MIXER,name='IEC958 Playback Switch'
; type=BOOLEAN,access=rw------,values=1
: values=off
numid=30,iface=MIXER,name='Dynamic Power-Control'
; type=ENUMERATED,access=rw------,values=1,items=2
; Item #0 'Disabled'
; Item #1 'Enabled'
: values=0
numid=29,iface=MIXER,name='Independent HP'
; type=ENUMERATED,access=rw------,values=1,items=2
; Item #0 'OFF'
; Item #1 'ON'
: values=0
numid=19,iface=MIXER,name='Input Source'
; type=ENUMERATED,access=rw------,values=1,items=4
; Item #0 'Rear Mic'
; Item #1 'Front Mic'
; Item #2 'Line'
; Item #3 'Stereo Mixer'
: values=1
numid=20,iface=MIXER,name='Input Source',index=1
; type=ENUMERATED,access=rw------,values=1,items=4
; Item #0 'Rear Mic'
; Item #1 'Front Mic'
; Item #2 'Line'
; Item #3 'Stereo Mixer'
: values=0
numid=27,iface=MIXER,name='Rear Mic Boost Volume'
; type=INTEGER,access=rw---R--,values=2,min=0,max=3,step=0
: values=0,0
| dBscale-min=0.00dB,step=10.25dB,mute=0
numid=22,iface=MIXER,name='Rear Mic Playback Switch'
; type=BOOLEAN,access=rw------,values=2
: values=off,off
numid=21,iface=MIXER,name='Rear Mic Playback Volume'
; type=INTEGER,access=rw---R--,values=2,min=0,max=31,step=0
: values=23,23
| dBscale-min=-34.50dB,step=1.50dB,mute=0
numid=10,iface=MIXER,name='Side Playback Switch'
; type=BOOLEAN,access=rw------,values=2
: values=on,on
numid=9,iface=MIXER,name='Side Playback Volume'
; type=INTEGER,access=rw---R--,values=2,min=0,max=42,step=0
: values=42,42
| dBscale-min=-63.00dB,step=1.50dB,mute=0
numid=49,iface=PCM,name='Capture Channel Map'
; type=INTEGER,access=r----R--,values=2,min=0,max=36,step=0
: values=0,0
| | TLV size error (257, 8, 0)!
numid=50,iface=PCM,name='Capture Channel Map',index=1
; type=INTEGER,access=r----R--,values=2,min=0,max=36,step=0
: values=0,0
| | TLV size error (257, 8, 0)!
numid=48,iface=PCM,name='Playback Channel Map'
; type=INTEGER,access=r----R--,values=8,min=0,max=36,step=0
: values=0,0,0,0,0,0,0,0
| | TLV size error (257, 8, 0)!
numid=51,iface=PCM,name='Playback Channel Map',device=1
; type=INTEGER,access=r----R--,values=2,min=0,max=36,step=0
: values=0,0
| | TLV size error (257, 8, 0)!
numid=52,iface=PCM,name='Playback Channel Map',device=2
; type=INTEGER,access=r----R--,values=2,min=0,max=36,step=0
: values=0,0
| | TLV size error (257, 8, 0)!
If I read the Alsa kernel docs right, the VIA driver simply does not provide any option to disable auto-muting: http://git.alsa-project.org/?p=alsa-ker … xt;hb=HEAD.
I would be glad if I had misread/misunderstood something or there were any other means of achieving what I am after? All I found so far is lots of posts where auto-muting did not work but very few people seem to want the opposite. FWIW, my old installation of Ubuntu 11.10 was working fine in that regard, i.e. it probably did not support auto-muting for my hardware.
Last edited by fax (2013-03-24 20:20:19)Solved, well, good enough for me anyway. In alsamixer, I enabled "Independent HP". Independent HP means that two of the six surround channels get diverted to the headphones. Alsa creates a second device, one for line out and the other one for the headphones. It is a very neat feature that is meant to allow you to e.g. do a voice call with your headphones while you play music through the speakers at the same time. I can now choose in PulseAudio whether I want output to my headphones or the speakers. It required that in /etc/pulse/default.pa, I added the line "load-module module-alsa-sink device=hw:0,2" after this paragraph:
### Automatically load driver modules depending on the hardware available
.ifexists module-udev-detect.so
load-module module-udev-detect
.else
### Use the static hardware detection module (for systems that lack udev support)
load-module module-detect
.endif
# Take Independent HP into use.
load-module module-alsa-sink device=hw:0,2
All this really is a work-around for two bugs:
Call it a bug or a missing feature, but it does not look like the Alsa snd_hda_codec_via driver allows to disable auto-muting.
Before the change to default.pa, PulseAudio already allows me to switch between two output devices - "analog output" and "analog headphones". "Analog headphones" however does the same thing as "analog output" and plays sound on the external speakers and not the headphones. I suspect that the PulseAudio module-udev-detect detects both Alsa devices but wires them wrongly.
The above still doesn't give me what I originally asked, i.e. sound on both speakers and headphones at the same time, but I realized that I don't really need that. I just need a way to switch on speakers occasionally while my headphones are plugged in.
Last edited by fax (2013-03-24 23:06:27) -
[SOLVED] No Sound with ALSA & Pulseaudio
Recently I made the transition from Debian to Arch, installed on a
seperate partition using the chroot install method.
All went smoothly, with the exception that there is no audio whatsoever.
Over the last week I've read through the related topics on the wiki and searched the forum,
but to no avail
Any assistance would be appreciated.
Here are some details;
lspci | grep audio
00:1b.0 Audio device [0403]: Intel Corporation 82801FB/FBM/FR/FW/FRW
(ICH6 Family) High Definition Audio Controller [8086:2668] (rev 03)
lsmod | grep snd
snd_intel8x0m 9736 0
snd_ac97_codec 89404 1 snd_intel8x0m
ac97_bus 910 1 snd_ac97_codec
snd_hda_codec_realtek 35187 1
snd_hda_intel 31255 0
snd_hda_codec 128666 2 snd_hda_codec_realtek,snd_hda_intel
snd_hwdep 4750 1 snd_hda_codec
snd_pcm_oss 33509 0
snd_mixer_oss 12450 1 snd_pcm_oss
snd_pcm 63880 5
snd_pcm_oss,snd_ac97_codec,snd_hda_codec,snd_hda_intel,snd_intel8x0m
snd_page_alloc 5978 3 snd_pcm,snd_hda_intel,snd_intel8x0m
snd_seq_dummy 1131 0
snd_seq_oss 25116 0
snd_seq_midi 4104 0
snd_seq_midi_event 4484 2 snd_seq_oss,snd_seq_midi
snd_rawmidi 14828 1 snd_seq_midi
snd_seq 41044 6
snd_seq_midi_event,snd_seq_oss,snd_seq_dummy,snd_seq_midi
snd_seq_device 4256 5
snd_seq,snd_rawmidi,snd_seq_oss,snd_seq_dummy,snd_seq_midi
snd_timer 14946 2 snd_pcm,snd_seq
snd 44566 14
snd_hda_codec_realtek,snd_pcm_oss,snd_ac97_codec,snd_hwdep,snd_timer,snd_pcm,snd_se$
soundcore 4386 1 snd
ls /dev/snd/*
/dev/snd/controlC0
/dev/snd/hwC0D2
/dev/snd/seq
/dev/snd/timer
/dev/snd/by-path:
pci-0000:00:1b.0
cat /proc/asound/modules
0 snd_hda_intel
aplay -lL
null
Discard all samples (playback) or generate zero samples (capture)
pulse
PulseAudio Sound Server
default
Default ALSA Output (currently PulseAudio Sound Server)
**** List of PLAYBACK Hardware Devices ****
Screenshots of alsamixer
http://www.imgbin.org/index.php?page=image&id=15874
http://www.imgbin.org/index.php?page=image&id=15875
pavucontrol shows no devices for configuration
Last edited by naesk (2013-12-13 08:11:44)a couple of days i had a similar problem, because sound modules for my internal modem were loaded. maybe you can try to blacklist the snd_hda_codec_realtek module, which looks like a network card.
you can achieve that by putting
blacklist snd_hda_codec_realtek
in /etc/modprobe.d/modprobe.conf and reboot.
rgds
hcjl
Last edited by hcjl (2013-12-13 08:39:48) -
[SOLVED] Alsa works, but not pulse (pulseaudio: symbol lookup error)
SOLUTION: Fixed in latest pulseaudio. Build from latest git, or use AUR.
I've just installed Arch following the beginner's guide, managing to get an up to date system, Xorg working with Awesome, and positive results with systemd. My next goal was basic audio and video.
I installed Alsa, added myself to the audio group, tested the playback and recording with arecord and aplay, no problems.
I installed mplayer2, which pulled `libpulse=0.9.22-2` as a dependency.
Mplayer2 did not work. It looked like it had something to do with pulseaudio, but I didn't look too hard before removing mplayer2 and libpulse.
Installed pulseaudio, pavucontrol, paprefs, avahi. I want to use mplayer2, but mostly I want pulseaudio for its own merits, especially its network transparency. I'm also hoping it fixes a skype problem.
Tried running `pulseaudio --start`, but it fails returning this error:
[jake@clyde pkg]$ pulseaudio --start
pulseaudio: symbol lookup error: /usr/lib/libpulsecommon-0.9.22.so: undefined symbol: STRING
Here's also a snippet of strace, I can post the full log if it would help.
[jake@clyde ~]$ strace -f pulseaudio --start
read(3, "\177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\220\r\0\0004\0\0\0"..., 512) = 512
fstat64(3, {st_mode=S_IFREG|0755, st_size=16064, ...}) = 0
mmap2(NULL, 18804, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0xb6e91000
mmap2(0xb6e95000, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x3) = 0xb6e95000
close(3) = 0
mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = 0xb6e90000
mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = 0xb6e8f000
mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = 0xb6e8e000
set_thread_area({entry_number:-1 -> 6, base_addr:0xb6e8e700, limit:1048575, seg_32bit:1, contents:0, read_exec_only:0, limit_in_pages:1, seg_not_present:0, useable:1}) = 0
mprotect(0xb7038000, 311296, PROT_READ|PROT_WRITE) = 0
mprotect(0xb7038000, 311296, PROT_READ|PROT_EXEC) = 0
mprotect(0xb7099000, 4096, PROT_READ) = 0
mprotect(0xb70b2000, 4096, PROT_READ) = 0
mprotect(0xb7242000, 8192, PROT_READ) = 0
mprotect(0xb7270000, 4096, PROT_READ) = 0
mprotect(0xb7275000, 4096, PROT_READ) = 0
mprotect(0xb727f000, 4096, PROT_READ) = 0
mprotect(0xb729d000, 4096, PROT_READ) = 0
mprotect(0xb72e8000, 4096, PROT_READ) = 0
writev(2, [{"pulseaudio", 10}, {": ", 2}, {"symbol lookup error", 19}, {": ", 2}, {"/usr/lib/libpulsecommon-0.9.22.s"..., 33}, {": ", 2}, {"undefined symbol: STRING", 24}, {"", 0}, {"", 0}, {"\n", 1}], 10pulseaudio: symbol lookup error: /usr/lib/libpulsecommon-0.9.22.so: undefined symbol: STRING
) = 93
exit_group(127) = ?
Hope someone can point me in the right direction. I'm liking Arch so far. It feels like Slackware, but without the dependency hell!
Last edited by djeikyb (2011-06-23 20:51:22)rickeyski wrote:
I know there is an issue with pulse and xcb-util 3.8, taken from the awesome-git aur page
http://sources.gentoo.org/cgi-bin/viewv … threv=HEAD
Gentoo's Bugzilla
http://bugs.gentoo.org/show_bug.cgi?id=364965
Thank you for this find! Your links have been mentioned in the awesome-git AUR page:
http://aur.archlinux.org/packages.php?ID=13916 -
Java/OpenJDK problem with OSS/osspd/ALSA/pulseaudio [SOLVED]
I've got problems with sound output of java programs, which usually try to hog /dev/dsp, using pulseaudio and openjdk 7.
Some rare java apps' sound methods surprisingly do work. Others (which the majority of java programs seem to use) do not. In Sun Java I could make those work by using 'padsp', however this method apparently fails on Archlinux with both latest Oracle Java (formerly worked on Ubuntu w/ Sun Java 6) and with OpenJDK too.
So I tried 'osspd' (package is called 'ossp') and it shows the root process "/usr/sbin/osspd --dsp-slave=/usr/sbin/ossp-padsp" been started, but the java applications will still not provide sound which I find pretty odd. I tried adding 'soundcore.preclaim_oss=0' to the 'kernel..' line in my grub's menu.lst, but that didn't help either.
Last edited by Jindur (2012-05-20 02:59:20)It's been quite a while, but I finally found a solution (read: big ugly hack).
Someone literally created a biguglyhack and posted it here:
http://lifein19x19.com/forum/viewtopic. … 243#p98243
(downloadable file there: javadummymixer_biguglyhack.zip [12.97 KiB] -> rename it from .zip to .jar and move it into your java's lib/ext folder -> profit!)
I applied it to OpenJDK 6 (should be same for 7 though) and all my java sound works flawlessly now, with pulseaudio, without any need for padsp stuff or osspd or whatever.
My sound.properties file reads:
javax.sound.sampled.Clip=com.sun.media.sound.DirectAudioDeviceProvider
javax.sound.sampled.Port=com.sun.media.sound.PortMixerProvider
javax.sound.sampled.SourceDataLine=com.sun.media.sound.DirectAudioDeviceProvider
javax.sound.sampled.TargetDataLine=com.sun.media.sound.DirectAudioDeviceProvider
but actually I'm not sure whether that even matters at all.
(Side note: The linked thread also mentions a "-D.sun.." command-line parameter that supposedly forces java to avoid disfunctional audio methods. The above biguglyhack however worked for me without any need of applying that command-line parameter.)
Last edited by Jindur (2012-05-20 02:58:05) -
Hello all!
I'm a new user of this forum, and first of all, i would like to thank thins community for helping me migrating from ubunto to arch linux, which i do like very much because of it's freedom of choice and it's challenging installation.
As a new Arch user, i had and have some issues that i had to overcome.
One of them was to put ALSA to work correctly.
First, i'm going to describe my system:
- hp dv25699ep
- (uname -a) Linux (none) 3.3.2-1-ARCH #1 SMP PREEMPT Sat Apr 14 10:08:43 UTC 2012 i686 Intel(R) Core(TM)2 Duo CPU T7500 @ 2.20GHz GenuineIntel GNU/Linux
- DE -> xfce4
My issue with ALSA was that i had a scale that i was not used to see, i only could get sound from "unamplified" to "100%", and from "100%" to "infinity".
The thing was that the sound did not raised linearly, causing the "boom" effect, it raised from mute to "boom" in a little space of the scale.
After many hours searching, i found this website: https://gist.github.com/1904230
which helped me a lot. For those who can't access the website, here's what is says:
Copy pulseaudio configuration template to home folder.
https://gist.github.com/1904230 wrote:1. cp /etc/pulse/default.pa ~/.pulse/default.pa
2. Append ignore_dB=1 to line with load-module module-udev-detect.
3. Make sure "speaker" volume is at 100% using alsamixer (press F6 and select HDA NVidia to see more volume controls.
4. To make "speaker" volume persist between boots, run alsactl store -f ~/.asoundrc then make alsactl restore -f ~/.asoundrc start when logging into your user. (See Arch Wiki - ALSA)
5. Profit.
I read somewhere that the alsa driver was not sending the right dB's to pulse, and pulse simply improvised, is this strange?
I have now verified (after the upper solution) that at 20-30% i can't almost hear anything(no, i'm not deaf ), have anyone got a solution or this is normal?
Thank you!> aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: Intel [HDA Intel], device 0: CONEXANT Analog [CONEXANT Analog]
Subdevices: 0/1
Subdevice #0: subdevice #0
card 0: Intel [HDA Intel], device 1: Conexant Digital [Conexant Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0 -
Blue Yeti USB microphone not working in ALSA/PulseAudio
Hi all,
I recently bought a Blue Yeti USB microphone. It is supposed to require no drivers for Windows and OS X, and so far, it works well under Windows, but I cannot seem to get it to function on Arch Linux. As fair warning, although I am relatively experienced with Linux, I don't know much about the internals of ALSA/PA --- up until now, it has just "always worked", so this is my first real foray into the area. I've tried for many days, with no success, to get this thing working.
So, let's start off first with my configuration.
I have a few audio devices:
An on-board audio solution that I don't typically use. For testing, I have connected an old, crappy pair of headphones, but normally I keep this disabled. It has one capture device. (Analog stereo [or 5.1/7.1/etc] duplex)
An external DAC that is connected via USB and is output-only. So, no capture devices on that. (Digital/IEC958 output)
The Blue Yeti itself, which is connected via USB and supports output (it has an internal DAC) as well as input. So, one capture device here. (Digital stereo output and analog stereo input???)
The HDMI-out stuff for nvidia, which I never use
Here is the output of "arecord -l":
**** List of CAPTURE Hardware Devices ****
card 1: Microphone [Yeti Stereo Microphone], device 0: USB Audio [USB Audio]
Subdevices: 0/1
Subdevice #0: subdevice #0
card 2: PCH [HDA Intel PCH], device 0: ALC892 Analog [ALC892 Analog]
Subdevices: 0/1
Subdevice #0: subdevice #0
card 2: PCH [HDA Intel PCH], device 2: ALC892 Alt Analog [ALC892 Alt Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
Here is the the output of "arecord -L":
null
Discard all samples (playback) or generate zero samples (capture)
pulse
PulseAudio Sound Server
default
Default ALSA Output (currently PulseAudio Sound Server)
sysdefault:CARD=Microphone
Yeti Stereo Microphone, USB Audio
Default Audio Device
front:CARD=Microphone,DEV=0
Yeti Stereo Microphone, USB Audio
Front speakers
surround40:CARD=Microphone,DEV=0
Yeti Stereo Microphone, USB Audio
4.0 Surround output to Front and Rear speakers
surround41:CARD=Microphone,DEV=0
Yeti Stereo Microphone, USB Audio
4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=Microphone,DEV=0
Yeti Stereo Microphone, USB Audio
5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=Microphone,DEV=0
Yeti Stereo Microphone, USB Audio
5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=Microphone,DEV=0
Yeti Stereo Microphone, USB Audio
7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
iec958:CARD=Microphone,DEV=0
Yeti Stereo Microphone, USB Audio
IEC958 (S/PDIF) Digital Audio Output
sysdefault:CARD=PCH
HDA Intel PCH, ALC892 Analog
Default Audio Device
front:CARD=PCH,DEV=0
HDA Intel PCH, ALC892 Analog
Front speakers
surround40:CARD=PCH,DEV=0
HDA Intel PCH, ALC892 Analog
4.0 Surround output to Front and Rear speakers
surround41:CARD=PCH,DEV=0
HDA Intel PCH, ALC892 Analog
4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=PCH,DEV=0
HDA Intel PCH, ALC892 Analog
5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=PCH,DEV=0
HDA Intel PCH, ALC892 Analog
5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=PCH,DEV=0
HDA Intel PCH, ALC892 Analog
7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
Here is the output of "aplay -l" (for completeness):
**** List of PLAYBACK Hardware Devices ****
card 0: Audio [USB2.0 High-Speed True HD Audio], device 0: USB Audio [USB Audio]
Subdevices: 0/1
Subdevice #0: subdevice #0
card 1: Microphone [Yeti Stereo Microphone], device 0: USB Audio [USB Audio]
Subdevices: 0/1
Subdevice #0: subdevice #0
card 2: PCH [HDA Intel PCH], device 0: ALC892 Analog [ALC892 Analog]
Subdevices: 0/1
Subdevice #0: subdevice #0
card 2: PCH [HDA Intel PCH], device 1: ALC892 Digital [ALC892 Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 3: NVidia [HDA NVidia], device 3: HDMI 0 [HDMI 0]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 3: NVidia [HDA NVidia], device 7: HDMI 1 [HDMI 1]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 3: NVidia [HDA NVidia], device 8: HDMI 2 [HDMI 2]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 3: NVidia [HDA NVidia], device 9: HDMI 3 [HDMI 3]
Subdevices: 1/1
Subdevice #0: subdevice #0
If the output of "aplay -L" is needed, let me know, but it's not all that different from arecord -L.
Here is my /etc/asound.conf:
defaults.pcm.rate_converter "samplerate_best"
pcm.!default {
type pulse
fallback "sysdefault"
hint {
show on
description "Default ALSA Output (currently PulseAudio Sound Server)"
ctl.!default {
type pulse
fallback "sysdefault"
# vim:set ft=alsaconf:
I don't remember where the bulk of that came from; maybe it was installed like that or maybe I put it in there. Here is another useful command showing non-commented-out/non-blank lines in the pulseaudio conf:
$ egrep -v '^;|^#|^$' /etc/pulse/*.conf
/etc/pulse/client.conf:autospawn=no
/etc/pulse/daemon.conf:exit-idle-time=0
/etc/pulse/daemon.conf:default-sample-rate = 192000
So, there's not much I've done there. Here's the contents of /proc/asound/cards:
0 [Audio ]: USB-Audio - USB2.0 High-Speed True HD Audio
CMEDIA USB2.0 High-Speed True HD Audio at usb-0000:00:1d.0-1.7, high speed
1 [Microphone ]: USB-Audio - Yeti Stereo Microphone
Blue Microphones Yeti Stereo Microphone at usb-0000:00:1d.0-1.8, full speed
2 [PCH ]: HDA-Intel - HDA Intel PCH
HDA Intel PCH at 0xf7530000 irq 52
3 [NVidia ]: HDA-Intel - HDA NVidia
HDA NVidia at 0xf7080000 irq 17
And here's the contents of /proc/asound/modules:
0 snd_usb_audio
1 snd_usb_audio
2 snd_hda_intel
3 snd_hda_intel
Now, here is the output of "pactl list sources" (the Blue Yeti is last in the list):
Source #3
State: RUNNING
Name: alsa_input.pci-0000_00_1b.0.analog-stereo
Description: Built-in Audio Analog Stereo
Driver: module-alsa-card.c
Sample Specification: s16le 2ch 192000Hz
Channel Map: front-left,front-right
Owner Module: 7
Mute: yes
Volume: front-left: 17972 / 27% / -33.71 dB, front-right: 17972 / 27% / -33.71 dB
balance 0.00
Base Volume: 6554 / 10% / -60.00 dB
Monitor of Sink: n/a
Latency: 610 usec, configured 100000 usec
Flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY
Properties:
alsa.resolution_bits = "16"
device.api = "alsa"
device.class = "sound"
alsa.class = "generic"
alsa.subclass = "generic-mix"
alsa.name = "ALC892 Analog"
alsa.id = "ALC892 Analog"
alsa.subdevice = "0"
alsa.subdevice_name = "subdevice #0"
alsa.device = "0"
alsa.card = "2"
alsa.card_name = "HDA Intel PCH"
alsa.long_card_name = "HDA Intel PCH at 0xf7530000 irq 52"
alsa.driver_name = "snd_hda_intel"
device.bus_path = "pci-0000:00:1b.0"
sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card2"
device.bus = "pci"
device.vendor.id = "8086"
device.vendor.name = "Intel Corporation"
device.product.id = "1e20"
device.product.name = "7 Series/C210 Series Chipset Family High Definition Audio Controller"
device.form_factor = "internal"
device.string = "front:2"
device.buffering.buffer_size = "76800"
device.buffering.fragment_size = "19200"
device.access_mode = "mmap"
device.profile.name = "analog-stereo"
device.profile.description = "Analog Stereo"
device.description = "Built-in Audio Analog Stereo"
alsa.mixer_name = "Realtek ALC892"
alsa.components = "HDA:10ec0892,10438436,00100302"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-pci"
Ports:
analog-input-front-mic: Front Microphone (priority: 8500, not available)
analog-input-rear-mic: Rear Microphone (priority: 8200, available)
analog-input-linein: Line In (priority: 8100, not available)
Active Port: analog-input-rear-mic
Formats:
pcm
Source #11
State: RUNNING
Name: alsa_output.pci-0000_00_1b.0.analog-stereo.monitor
Description: Monitor of Built-in Audio Analog Stereo
Driver: module-alsa-card.c
Sample Specification: s16le 2ch 192000Hz
Channel Map: front-left,front-right
Owner Module: 7
Mute: no
Volume: front-left: 65536 / 100% / 0.00 dB, front-right: 65536 / 100% / 0.00 dB
balance 0.00
Base Volume: 65536 / 100% / 0.00 dB
Monitor of Sink: alsa_output.pci-0000_00_1b.0.analog-stereo
Latency: 0 usec, configured 100000 usec
Flags: DECIBEL_VOLUME LATENCY
Properties:
device.description = "Monitor of Built-in Audio Analog Stereo"
device.class = "monitor"
alsa.card = "2"
alsa.card_name = "HDA Intel PCH"
alsa.long_card_name = "HDA Intel PCH at 0xf7530000 irq 52"
alsa.driver_name = "snd_hda_intel"
device.bus_path = "pci-0000:00:1b.0"
sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card2"
device.bus = "pci"
device.vendor.id = "8086"
device.vendor.name = "Intel Corporation"
device.product.id = "1e20"
device.product.name = "7 Series/C210 Series Chipset Family High Definition Audio Controller"
device.form_factor = "internal"
device.string = "2"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-pci"
Formats:
pcm
Source #15
State: RUNNING
Name: alsa_output.usb-CMEDIA_USB2.0_High-Speed_True_HD_Audio-00-Audio.iec958-stereo.monitor
Description: Monitor of USB2.0 High-Speed True HD Audio Digital Stereo (IEC958)
Driver: module-alsa-card.c
Sample Specification: s16le 2ch 192000Hz
Channel Map: front-left,front-right
Owner Module: 8
Mute: no
Volume: front-left: 65536 / 100% / 0.00 dB, front-right: 65536 / 100% / 0.00 dB
balance 0.00
Base Volume: 65536 / 100% / 0.00 dB
Monitor of Sink: alsa_output.usb-CMEDIA_USB2.0_High-Speed_True_HD_Audio-00-Audio.iec958-stereo
Latency: 0 usec, configured 100000 usec
Flags: DECIBEL_VOLUME LATENCY
Properties:
device.description = "Monitor of USB2.0 High-Speed True HD Audio Digital Stereo (IEC958)"
device.class = "monitor"
alsa.card = "0"
alsa.card_name = "USB2.0 High-Speed True HD Audio"
alsa.long_card_name = "CMEDIA USB2.0 High-Speed True HD Audio at usb-0000:00:1d.0-1.7, high speed"
alsa.driver_name = "snd_usb_audio"
device.bus_path = "pci-0000:00:1d.0-usb-0:1.7:1.0"
sysfs.path = "/devices/pci0000:00/0000:00:1d.0/usb6/6-1/6-1.7/6-1.7:1.0/sound/card0"
udev.id = "usb-CMEDIA_USB2.0_High-Speed_True_HD_Audio-00-Audio"
device.bus = "usb"
device.vendor.id = "0d8c"
device.vendor.name = "C-Media Electronics, Inc."
device.product.id = "0319"
device.product.name = "USB2.0 High-Speed True HD Audio"
device.serial = "CMEDIA_USB2.0_High-Speed_True_HD_Audio"
device.string = "0"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-usb"
Formats:
pcm
Source #19
State: RUNNING
Name: alsa_output.usb-Blue_Microphones_Yeti_Stereo_Microphone_REV8-00-Microphone.iec958-stereo.monitor
Description: Monitor of Integrated Rate Matching Hub Digital Stereo (IEC958)
Driver: module-alsa-card.c
Sample Specification: s16le 2ch 48000Hz
Channel Map: front-left,front-right
Owner Module: 9
Mute: no
Volume: front-left: 65536 / 100% / 0.00 dB, front-right: 65536 / 100% / 0.00 dB
balance 0.00
Base Volume: 65536 / 100% / 0.00 dB
Monitor of Sink: alsa_output.usb-Blue_Microphones_Yeti_Stereo_Microphone_REV8-00-Microphone.iec958-stereo
Latency: 0 usec, configured 100000 usec
Flags: DECIBEL_VOLUME LATENCY
Properties:
device.description = "Monitor of Integrated Rate Matching Hub Digital Stereo (IEC958)"
device.class = "monitor"
alsa.card = "1"
alsa.card_name = "Yeti Stereo Microphone"
alsa.long_card_name = "Blue Microphones Yeti Stereo Microphone at usb-0000:00:1d.0-1.8, full speed"
alsa.driver_name = "snd_usb_audio"
device.bus_path = "pci-0000:00:1d.0-usb-0:1.8:1.0"
sysfs.path = "/devices/pci0000:00/0000:00:1d.0/usb6/6-1/6-1.8/6-1.8:1.0/sound/card1"
udev.id = "usb-Blue_Microphones_Yeti_Stereo_Microphone_REV8-00-Microphone"
device.bus = "usb"
device.vendor.id = "b58e"
device.vendor.name = "Intel Corp."
device.product.id = "9e84"
device.product.name = "Integrated Rate Matching Hub"
device.serial = "Blue_Microphones_Yeti_Stereo_Microphone_REV8"
device.form_factor = "microphone"
device.string = "1"
module-udev-detect.discovered = "1"
device.icon_name = "audio-input-microphone-usb"
Formats:
pcm
Source #23
State: RUNNING
Name: alsa_input.usb-Blue_Microphones_Yeti_Stereo_Microphone_REV8-00-Microphone.analog-stereo
Description: Integrated Rate Matching Hub Analog Stereo
Driver: module-alsa-card.c
Sample Specification: s16le 2ch 48000Hz
Channel Map: front-left,front-right
Owner Module: 9
Mute: no
Volume: front-left: 65536 / 100% / 0.00 dB, front-right: 65536 / 100% / 0.00 dB
balance 0.00
Base Volume: 17775 / 27% / -34.00 dB
Monitor of Sink: n/a
Latency: 949811526 usec, configured 100000 usec
Flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY
Properties:
alsa.resolution_bits = "16"
device.api = "alsa"
device.class = "sound"
alsa.class = "generic"
alsa.subclass = "generic-mix"
alsa.name = "USB Audio"
alsa.id = "USB Audio"
alsa.subdevice = "0"
alsa.subdevice_name = "subdevice #0"
alsa.device = "0"
alsa.card = "1"
alsa.card_name = "Yeti Stereo Microphone"
alsa.long_card_name = "Blue Microphones Yeti Stereo Microphone at usb-0000:00:1d.0-1.8, full speed"
alsa.driver_name = "snd_usb_audio"
device.bus_path = "pci-0000:00:1d.0-usb-0:1.8:1.0"
sysfs.path = "/devices/pci0000:00/0000:00:1d.0/usb6/6-1/6-1.8/6-1.8:1.0/sound/card1"
udev.id = "usb-Blue_Microphones_Yeti_Stereo_Microphone_REV8-00-Microphone"
device.bus = "usb"
device.vendor.id = "b58e"
device.vendor.name = "Intel Corp."
device.product.id = "9e84"
device.product.name = "Integrated Rate Matching Hub"
device.serial = "Blue_Microphones_Yeti_Stereo_Microphone_REV8"
device.form_factor = "microphone"
device.string = "front:1"
device.buffering.buffer_size = "19200"
device.buffering.fragment_size = "4800"
device.access_mode = "mmap"
device.profile.name = "analog-stereo"
device.profile.description = "Analog Stereo"
device.description = "Integrated Rate Matching Hub Analog Stereo"
alsa.mixer_name = "USB Mixer"
alsa.components = "USBb58e:9e84"
module-udev-detect.discovered = "1"
device.icon_name = "audio-input-microphone-usb"
Ports:
analog-input-mic: Microphone (priority: 8700)
Active Port: analog-input-mic
Formats:
pcm
Honestly, I think that's about all of the configuration info I can give. Here are some screenshots of pavucontrol, though, in case you are not yet sated: http://imgur.com/a/ngyLl
Now, on to the debugging stuff. First: everything except Blue Yeti recording works. The external DAC works, the on-board audio (both input and output) work, using the DAC in the Blue Yeti to play music/monitor the microphone works (so it works as an output). The only thing that doesn't work is actual recording from the Blue Yeti itself.
When I try to record from the Blue Yeti with parecord -v, I get very strange behavior. Essentially, the "latency" of the microphone measured is exactly the same as the time it has been recording. That is, if it has been recording for 5 sec, then the latency is 5000000 usec. Also, if I try to record something in Audacity, I get no result except for a message box that says "Latency Correction has caused the recorded audio to be hidden before zero. Audacity has brought it back to start at zero." Also, pavucontrol doesn't show an input volume bar for the Blue Yeti, as in the above screenshots, whereas it normally will for microphones that are getting input. The Yeti is not muted, either, so it should be picking up ambient fan noise and such.
Does anyone have any ideas? Is there some configuration error I've made? Typically, this sort of stuff simply works out of the box for me, so I am somewhat stuck on what to do. Could it be a driver-level bug?I'm not sure what you mean by "mixed with audio in Alsamixer". Can you clarify?
With that said, my problem did mysteriously fix itself after some update (presumably to alsa). Unfortunately, I noticed it in passing one day while I was changing the volume of something in pavucontrol, so I don't remember the exact update that caused it to work. But yes, my current setup does work now: I can listen to audio through my DAC and use the Yeti for recording. However, I have found that use of the Yeti crashes Google Hangouts, although it works perfectly in every other program I've tried.
As documentation, here is my current working setup (it is very similar to the above): in PulseAudio, the DAC is set to be a Digital Stereo (IEC958) output. The Yeti (called an 'Integrated Rate Matching Hub' in pavucontrol) is set to be an Analog Stereo Input. In the input devices section, I have the Yeti's volume set to around 58% (-14.31 dB), which is past the base volume a fair ways but produces decent levels for recording. The DAC's volume is set to 100% (0dB gain, the DAC takes care of the level). In alsamixer, I have the mic level for the Yeti set to 40.
Edit: This is a long-time-later edit, but let me say that when you have multiple USB audio devices, what USB hub and in what order they are connected seems to matter a great deal. If you can't get your Blue Yeti to work, try moving it to a different set of USB ports (controlled by a different controller) on your motherboard. This magically fixes the issue for me.
Last edited by rwiggins (2014-12-14 00:26:39) -
[SOLVED]MPD, PulseAudio & Systemd/User
Since the advent of skype 4.3 I've had to switch to using pulseaudio. Things seems to be working fine except for the interaction between pulseaudio and mpd. My goal is to attain the same functionality as I had before when I was just using alsa. The main problem seems to be that the mpd daemon starts before pulseaudio. This means that when I reboot, for example, my music doesn't automatically keep playing. However if I toggle mpc (or open ncmpcpp and unpause) then it works fine.
What I have done
1) I installed mpd using the script found on the wiki
1a) copied ~/.config/mpd/mpd.conf to /etc/mpd.conf and uncomment #user line so it runs as my user
1b) enabled the mpd service with systemctl so it starts on boot, as directed to from 1)
2) Applied the workaround detailed here to make pulseaudio play nice with mpd
From what I can understand this results in the mpd daemon using the /etc/mpd.conf file which mirrors my local one (but I think the process is still started as root?)
If I don't do 2) then I get a problem where if I try to open pavucontrol I get an error saying I'm unable to connect and the system feels very unresponsive. Two second delays navigating around in thunar for example.
fuser /dev/snd/* outputs
» sudo fuser /dev/snd/*
/dev/snd/controlC0: 440
/dev/snd/controlC1: 440
/dev/snd/pcmC0D0p: 440m
» ps aux | grep 440
quiv 440 1.2 0.1 495964 13248 ? Sl 15:16 0:05 /usr/bin/pulseaudio --start --log-target=syslog
quiv 2259 0.0 0.0 11668 2296 pts/0 S+ 15:23 0:00 grep --color=auto 440
For now I have added in
if grep -q "state: pause" /home/quiv/.config/mpd/state; then
mpc toggle
fi
to my ~/.xinitrc as a work around to get pulse and mpd to work together properly. However it doesn't seem to work all the time. If I leave my music playing then when I reboot the system with reboot it seems that somtimes the state is being saved as paused, and other times as play. I suppose I could just mpc toggle without checking, but I don't want a situation where if I happen to not be playing music that every time I reboot my music starts playing.
Other things I have tried are;
A) not using pulseaudio with mpd.
B) not using systemd to manage mpd. Instead I manually started it in .xinitrc and pointed it to my ~/.config/mpd/mpd.conf file. I believe this means it runs as my own user instead of as root. When I did this I commented the username line and also undid 2) from above in ~/.config/mpd/mpd.conf
The problem with A) is that if I use alsa with mpd, mpd hogs my soundcard. My card becomes unselectable in pavucontrol and I cannot use it with pulseaudio., meanng no skype. The benefit of doing this (this is how I used to do it before I needed pulseaudio) is I have no problems with mpd's state being saved incorrectly.
The problem with B) is the mpdstate is saved incorrectly when for example I issue reboot. I can reproduce this by changing my currently played song then rebooting. When I boot back up mpd resumes playback from the previous session instead of recognizing the last thing I was playing. I had a search around on the forums and I found this issue which seems to describe what is happening. There doesn't seem to be a solution there instead a suggestion to user systemd (which I would like to do!) instead to manage mpd. Rasi's post details a unit file but I'm unsure what to do with this.
Looking at the wiki it seems like maybe I am supposed to provide this file so that it overwrites the default mpd.service unit. So I followed the steps listed there. First I created the directory /etc/systemd/system/mpd.d/ then made the user.conf file inside. I tried just putting the additional arguments in
[Service]
User=YOUR_USER
PAMName=system-local-login
I also remember trying to overwrite the entire [Service] section of the original by using
[Service]
User=YOUR_USER
PAMName=system-local-login
ExecStart=
ExecStart=/usr/bin/mpd --no-daemon
because I think just adding the addtional parameters might not have worked. But I'm not sure on this point, maybe both ways worked. This seemed to work but I had the same problem as above. The state was still being saved incorrectly sometimes when I rebooted. I would get a previous session instead of the most recent change. So I tried the other method for overwriting systemd unit files and made the file /etc/systemd/system/mpd.service in which I pasted Rasi's entire unit file. This also seemed to work, but still the incorrect state problem happened.
At this point I didn't know what to try. It seemed like the only option left was to set up mpd using a local configuration and setting up a systemd/User session and managing it that way. The wiki indiciated that if I did it this way I would not have to use the method 2) above. However I was/still am hesistant about doing this. I'm new to arch and the systemd/User wiki page seems very intimidating. I worry that I will create problems that I won't understand how to solve using this method just for the chance that mpd will function correclty with pulse. However I seemed to have exhausted all my possible options so I thought whatever I'll try it anyway! I'm certain I probably haven't done it right however.
So once again I turned to the wiki to help me with setting this up. However I can't make sense of the article. I don't know if I should follow ONLY the Setup since systemd 206 or if I should follow both setup sections? I decided to follow only the first section. I use no DE so I commented out the
[[ -z $DISPLAY && $XDG_VTNR -eq 1 ]] && exec startx
line in my ~/.zprofile. When I set up my system I followed the Automatic login to virtual console. I left this file alone, but upon second thoughts maybe I should have disabled this too? I don't understand if using xlogin-git to austoart Xorg, as described in the systemd/User page, requires me to get rid of this too. I think xlogin-git just starts Xorg so automatic login via getty should be fine?
Immediately when I boot in using systemd to manage my session I notice that there are problems. I don't think my environment variables are being sourced from .zshenv because my panel fails to load. If I kill the panel and load it again it works. This makes me think there is something wrong with the order of units started with systemd.
» sudo journalctl --this-boot --no-pager | grep -i panel ~
Jul 05 16:25:39 lorelai bash[494]: /home/quiv/.config/bspwm/panel/panel: line 16: : No such file or directory
Jul 05 16:25:39 lorelai bash[494]: /home/quiv/.config/bspwm/panel/panel: line 17: : No such file or directory
Jul 05 16:25:39 lorelai bash[494]: /home/quiv/.config/bspwm/panel/panel: line 18: : No such file or directory
In each instance panel is trying to do things based on $PANEL_FIFO which is exported from ~/.zshenv but it doesn't appear to be set at the time systemd starts things up. Apart from these problems, which are probably undoubtedly due to the fact I failed to set up systemd/User properly, I still have problems with mpd. The state seems to be remembered correctly now, however when I reboot I still have to toggle mpc for it to start playback. It seems to be the same problem as when I use setup detailed intially. I suspect perhaps once again there is something wrong with the order here. MPD starts before pulse so it doesn't find anything to play through, then once I issue the mpc toggle command pulse is started and all is happy.
A final note is that I was getting this error yesterday when using systemd/User
● mpd.service - Music Player Daemon
Loaded: loaded (/usr/lib/systemd/system/mpd.service; enabled)
Active: inactive (dead)
Jul 05 03:11:41 lorelai pulseaudio[450]: [pulseaudio] module-jackdbus-detect.c: Unable to contact D-Bus session bus: org.freedesktop.DBus.Error.NotSupported: Unable to autolaunch a dbus-daemon without a $DISPLAY for X11
Jul 05 03:11:41 lorelai pulseaudio[450]: [pulseaudio] module.c: Failed to load module "module-jackdbus-detect" (argument: "channels=2"): initialization failed.
Jul 05 03:11:41 lorelai pulseaudio[450]: [pulseaudio] main.c: Module load failed.
Jul 05 03:13:35 lorelai systemd[1]: Stopping Music Player Daemon...
Jul 05 03:13:35 lorelai systemd[1]: Stopped Music Player Daemon.
but it doesn't seem to happen anymore.
Also I have no idea if I'm even supposed to use it like this but
» sudo systemctl --user enable mpd ~
Failed to get D-Bus connection: Connection refused
I tried this because I was wondering if mpd was even using the correct service when using systemd/User. I thought perhaps it just was using the one in /etc/systemd/system/ instead of /usr/lib/systemd/user/. I'm still not sure about this.
In the end I've exhausted whatever I can think of. I'm sure I've done a lot of things wrong but I tried my best. Hopefully somebody knows what I've done wrong and can help me.
~/.config/mpd/mpd.conf
# An example configuration file for MPD.
# Read the user manual for documentation: http://www.musicpd.org/doc/user/
# Files and directories #######################################################
# This setting controls the top directory which MPD will search to discover the
# available audio files and add them to the daemon's online database. This
# setting defaults to the XDG directory, otherwise the music directory will be
# be disabled and audio files will only be accepted over ipc socket (using
# file:// protocol) or streaming files over an accepted protocol.
music_directory "~/Music"
# This setting sets the MPD internal playlist directory. The purpose of this
# directory is storage for playlists created by MPD. The server will use
# playlist files not created by the server but only if they are in the MPD
# format. This setting defaults to playlist saving being disabled.
playlist_directory "~/.config/mpd/playlists"
# This setting sets the location of the MPD database. This file is used to
# load the database at server start up and store the database while the
# server is not up. This setting defaults to disabled which will allow
# MPD to accept files over ipc socket (using file:// protocol) or streaming
# files over an accepted protocol.
db_file "~/.config/mpd/database"
# These settings are the locations for the daemon log files for the daemon.
# These logs are great for troubleshooting, depending on your log_level
# settings.
# The special value "syslog" makes MPD use the local syslog daemon. This
# setting defaults to logging to syslog, otherwise logging is disabled.
log_file "~/.config/mpd/log"
# This setting sets the location of the file which stores the process ID
# for use of mpd --kill and some init scripts. This setting is disabled by
# default and the pid file will not be stored.
pid_file "~/.config/mpd/pid"
# This setting sets the location of the file which contains information about
# most variables to get MPD back into the same general shape it was in before
# it was brought down. This setting is disabled by default and the server
# state will be reset on server start up.
state_file "~/.config/mpd/state"
# The location of the sticker database. This is a database which
# manages dynamic information attached to songs.
sticker_file "~/.config/mpd/sticker.sql"
# General music daemon options ################################################
# This setting specifies the user that MPD will run as. MPD should never run as
# root and you may use this setting to make MPD change its user ID after
# initialization. This setting is disabled by default and MPD is run as the
# current user.
#user "quiv"
# This setting specifies the group that MPD will run as. If not specified
# primary group of user specified with "user" setting will be used (if set).
# This is useful if MPD needs to be a member of group such as "audio" to
# have permission to use sound card.
#group "nogroup"
# This setting sets the address for the daemon to listen on. Careful attention
# should be paid if this is assigned to anything other then the default, any.
# This setting can deny access to control of the daemon.
# For network
bind_to_address "127.0.0.1"
# And for Unix Socket
bind_to_address "~/.config/mpd/socket"
# This setting is the TCP port that is desired for the daemon to get assigned
# to.
port "6600"
# This setting controls the type of information which is logged. Available
# setting arguments are "default", "secure" or "verbose". The "verbose" setting
# argument is recommended for troubleshooting, though can quickly stretch
# available resources on limited hardware storage.
log_level "default"
# If you have a problem with your MP3s ending abruptly it is recommended that
# you set this argument to "no" to attempt to fix the problem. If this solves
# the problem, it is highly recommended to fix the MP3 files with vbrfix
# (available from <http://www.willwap.co.uk/Programs/vbrfix.php>), at which
# point gapless MP3 playback can be enabled.
gapless_mp3_playback "yes"
# Setting "restore_paused" to "yes" puts MPD into pause mode instead
# of starting playback after startup.
#restore_paused "no"
# This setting enables MPD to create playlists in a format usable by other
# music players.
#save_absolute_paths_in_playlists "no"
# This setting defines a list of tag types that will be extracted during the
# audio file discovery process. The complete list of possible values can be
# found in the mpd.conf man page.
#metadata_to_use "artist,album,title,track,name,genre,date,composer,performer,disc"
# This setting enables automatic update of MPD's database when files in
# music_directory are changed.
auto_update "yes"
# Limit the depth of the directories being watched, 0 means only watch
# the music directory itself. There is no limit by default.
auto_update_depth "3"
# Symbolic link behavior ######################################################
# If this setting is set to "yes", MPD will discover audio files by following
# symbolic links outside of the configured music_directory.
#follow_outside_symlinks "yes"
# If this setting is set to "yes", MPD will discover audio files by following
# symbolic links inside of the configured music_directory.
#follow_inside_symlinks "yes"
# Zeroconf / Avahi Service Discovery ##########################################
# If this setting is set to "yes", service information will be published with
# Zeroconf / Avahi.
#zeroconf_enabled "yes"
# The argument to this setting will be the Zeroconf / Avahi unique name for
# this MPD server on the network.
#zeroconf_name "Music Player"
# Permissions #################################################################
# If this setting is set, MPD will require password authorization. The password
# can setting can be specified multiple times for different password profiles.
#password "password@read,add,control,admin"
# This setting specifies the permissions a user has who has not yet logged in.
#default_permissions "read,add,control,admin"
# Database #######################################################################
#database {
# plugin "proxy"
# host "other.mpd.host"
# port "6600"
# Input #######################################################################
input {
plugin "curl"
proxy "proxy.isp.com:8080"
proxy_user "user"
proxy_password "password"
# Audio Output ################################################################
# MPD supports various audio output types, as well as playing through multiple
# audio outputs at the same time, through multiple audio_output settings
# blocks. Setting this block is optional, though the server will only attempt
# autodetection for one sound card.
# An example of an ALSA output:
#audio_output {
# type "alsa"
# name "My ALSA Device"
## device "hw:0,0" # optional
## mixer_type "hardware" # optional
## mixer_device "default" # optional
## mixer_control "PCM" # optional
## mixer_index "0" # optional
audio_output {
type "pulse"
name "pulse audio"
format "48000:16:2"
## server "127.0.0.1"
## sink "remote_server_sink" # optional
# An example of an OSS output:
#audio_output {
# type "oss"
# name "My OSS Device"
## device "/dev/dsp" # optional
## mixer_type "hardware" # optional
## mixer_device "/dev/mixer" # optional
## mixer_control "PCM" # optional
# An example of a shout output (for streaming to Icecast):
#audio_output {
# type "shout"
# encoding "ogg" # optional
# name "My Shout Stream"
# host "localhost"
# port "8000"
# mount "/mpd.ogg"
# password "hackme"
# quality "5.0"
# bitrate "128"
# format "44100:16:1"
## protocol "icecast2" # optional
## user "source" # optional
## description "My Stream Description" # optional
## url "http://example.com" # optional
## genre "jazz" # optional
## public "no" # optional
## timeout "2" # optional
## mixer_type "software" # optional
# An example of a recorder output:
#audio_output {
# type "recorder"
# name "My recorder"
# encoder "vorbis" # optional, vorbis or lame
# path "/var/lib/mpd/recorder/mpd.ogg"
## quality "5.0" # do not define if bitrate is defined
# bitrate "128" # do not define if quality is defined
# format "44100:16:1"
# An example of a httpd output (built-in HTTP streaming server):
#audio_output {
# type "httpd"
# name "My HTTP Stream"
# encoder "vorbis" # optional, vorbis or lame
# port "8000"
# bind_to_address "0.0.0.0" # optional, IPv4 or IPv6
## quality "5.0" # do not define if bitrate is defined
# bitrate "128" # do not define if quality is defined
# format "44100:16:1"
# max_clients "0" # optional 0=no limit
# An example of a pulseaudio output (streaming to a remote pulseaudio server)
#audio_output {
# type "pulse"
# name "My Pulse Output"
## server "remote_server" # optional
## sink "remote_server_sink" # optional
# An example of a winmm output (Windows multimedia API).
#audio_output {
# type "winmm"
# name "My WinMM output"
## device "Digital Audio (S/PDIF) (High Definition Audio Device)" # optional
# or
## device "0" # optional
## mixer_type "hardware" # optional
# An example of an openal output.
#audio_output {
# type "openal"
# name "My OpenAL output"
## device "Digital Audio (S/PDIF) (High Definition Audio Device)" # optional
## Example "pipe" output:
#audio_output {
# type "pipe"
# name "my pipe"
# command "aplay -f cd 2>/dev/null"
## Or if you're want to use AudioCompress
# command "AudioCompress -m | aplay -f cd 2>/dev/null"
## Or to send raw PCM stream through PCM:
# command "nc example.org 8765"
# format "44100:16:2"
## An example of a null output (for no audio output):
#audio_output {
# type "null"
# name "My Null Output"
# mixer_type "none" # optional
# If MPD has been compiled with libsamplerate support, this setting specifies
# the sample rate converter to use. Possible values can be found in the
# mpd.conf man page or the libsamplerate documentation. By default, this is
# setting is disabled.
#samplerate_converter "Fastest Sinc Interpolator"
# Normalization automatic volume adjustments ##################################
# This setting specifies the type of ReplayGain to use. This setting can have
# the argument "off", "album", "track" or "auto". "auto" is a special mode that
# chooses between "track" and "album" depending on the current state of
# random playback. If random playback is enabled then "track" mode is used.
# See <http://www.replaygain.org> for more details about ReplayGain.
# This setting is off by default.
replaygain "album"
# This setting sets the pre-amp used for files that have ReplayGain tags. By
# default this setting is disabled.
replaygain_preamp "0"
# This setting sets the pre-amp used for files that do NOT have ReplayGain tags.
# By default this setting is disabled.
#replaygain_missing_preamp "0"
# This setting enables or disables ReplayGain limiting.
# MPD calculates actual amplification based on the ReplayGain tags
# and replaygain_preamp / replaygain_missing_preamp setting.
# If replaygain_limit is enabled MPD will never amplify audio signal
# above its original level. If replaygain_limit is disabled such amplification
# might occur. By default this setting is enabled.
#replaygain_limit "yes"
# This setting enables on-the-fly normalization volume adjustment. This will
# result in the volume of all playing audio to be adjusted so the output has
# equal "loudness". This setting is disabled by default.
#volume_normalization "no"
# MPD Internal Buffering ######################################################
# This setting adjusts the size of internal decoded audio buffering. Changing
# this may have undesired effects. Don't change this if you don't know what you
# are doing.
#audio_buffer_size "4096"
# This setting controls the percentage of the buffer which is filled before
# beginning to play. Increasing this reduces the chance of audio file skipping,
# at the cost of increased time prior to audio playback.
#buffer_before_play "10%"
# Resource Limitations ########################################################
# These settings are various limitations to prevent MPD from using too many
# resources. Generally, these settings should be minimized to prevent security
# risks, depending on the operating resources.
#connection_timeout "60"
#max_connections "10"
#max_playlist_length "16384"
#max_command_list_size "2048"
#max_output_buffer_size "8192"
# Character Encoding ##########################################################
# If file or directory names do not display correctly for your locale then you
# may need to modify this setting.
#filesystem_charset "UTF-8"
# This setting controls the encoding that ID3v1 tags should be converted from.
#id3v1_encoding "ISO-8859-1"
# SIDPlay decoder #############################################################
# songlength_database:
# Location of your songlengths file, as distributed with the HVSC.
# The sidplay plugin checks this for matching MD5 fingerprints.
# See http://www.c64.org/HVSC/DOCUMENTS/Songlengths.faq
# default_songlength:
# This is the default playing time in seconds for songs not in the
# songlength database, or in case you're not using a database.
# A value of 0 means play indefinitely.
# filter:
# Turns the SID filter emulation on or off.
#decoder {
# plugin "sidplay"
# songlength_database "/media/C64Music/DOCUMENTS/Songlengths.txt"
# default_songlength "120"
# filter "true"
/etc/mpd.config
# An example configuration file for MPD.
# Read the user manual for documentation: http://www.musicpd.org/doc/user/
# Files and directories #######################################################
# This setting controls the top directory which MPD will search to discover the
# available audio files and add them to the daemon's online database. This
# setting defaults to the XDG directory, otherwise the music directory will be
# be disabled and audio files will only be accepted over ipc socket (using
# file:// protocol) or streaming files over an accepted protocol.
music_directory "~/Music"
# This setting sets the MPD internal playlist directory. The purpose of this
# directory is storage for playlists created by MPD. The server will use
# playlist files not created by the server but only if they are in the MPD
# format. This setting defaults to playlist saving being disabled.
playlist_directory "~/.config/mpd/playlists"
# This setting sets the location of the MPD database. This file is used to
# load the database at server start up and store the database while the
# server is not up. This setting defaults to disabled which will allow
# MPD to accept files over ipc socket (using file:// protocol) or streaming
# files over an accepted protocol.
db_file "~/.config/mpd/database"
# These settings are the locations for the daemon log files for the daemon.
# These logs are great for troubleshooting, depending on your log_level
# settings.
# The special value "syslog" makes MPD use the local syslog daemon. This
# setting defaults to logging to syslog, otherwise logging is disabled.
log_file "~/.config/mpd/log"
# This setting sets the location of the file which stores the process ID
# for use of mpd --kill and some init scripts. This setting is disabled by
# default and the pid file will not be stored.
pid_file "~/.config/mpd/pid"
# This setting sets the location of the file which contains information about
# most variables to get MPD back into the same general shape it was in before
# it was brought down. This setting is disabled by default and the server
# state will be reset on server start up.
state_file "~/.config/mpd/state"
# The location of the sticker database. This is a database which
# manages dynamic information attached to songs.
sticker_file "~/.config/mpd/sticker.sql"
# General music daemon options ################################################
# This setting specifies the user that MPD will run as. MPD should never run as
# root and you may use this setting to make MPD change its user ID after
# initialization. This setting is disabled by default and MPD is run as the
# current user.
user "quiv"
# This setting specifies the group that MPD will run as. If not specified
# primary group of user specified with "user" setting will be used (if set).
# This is useful if MPD needs to be a member of group such as "audio" to
# have permission to use sound card.
#group "nogroup"
# This setting sets the address for the daemon to listen on. Careful attention
# should be paid if this is assigned to anything other then the default, any.
# This setting can deny access to control of the daemon.
# For network
bind_to_address "127.0.0.1"
# And for Unix Socket
bind_to_address "~/.config/mpd/socket"
# This setting is the TCP port that is desired for the daemon to get assigned
# to.
port "6600"
# This setting controls the type of information which is logged. Available
# setting arguments are "default", "secure" or "verbose". The "verbose" setting
# argument is recommended for troubleshooting, though can quickly stretch
# available resources on limited hardware storage.
log_level "default"
# If you have a problem with your MP3s ending abruptly it is recommended that
# you set this argument to "no" to attempt to fix the problem. If this solves
# the problem, it is highly recommended to fix the MP3 files with vbrfix
# (available from <http://www.willwap.co.uk/Programs/vbrfix.php>), at which
# point gapless MP3 playback can be enabled.
gapless_mp3_playback "yes"
# Setting "restore_paused" to "yes" puts MPD into pause mode instead
# of starting playback after startup.
#restore_paused "no"
# This setting enables MPD to create playlists in a format usable by other
# music players.
#save_absolute_paths_in_playlists "no"
# This setting defines a list of tag types that will be extracted during the
# audio file discovery process. The complete list of possible values can be
# found in the mpd.conf man page.
#metadata_to_use "artist,album,title,track,name,genre,date,composer,performer,disc"
# This setting enables automatic update of MPD's database when files in
# music_directory are changed.
auto_update "yes"
# Limit the depth of the directories being watched, 0 means only watch
# the music directory itself. There is no limit by default.
auto_update_depth "3"
# Symbolic link behavior ######################################################
# If this setting is set to "yes", MPD will discover audio files by following
# symbolic links outside of the configured music_directory.
#follow_outside_symlinks "yes"
# If this setting is set to "yes", MPD will discover audio files by following
# symbolic links inside of the configured music_directory.
#follow_inside_symlinks "yes"
# Zeroconf / Avahi Service Discovery ##########################################
# If this setting is set to "yes", service information will be published with
# Zeroconf / Avahi.
#zeroconf_enabled "yes"
# The argument to this setting will be the Zeroconf / Avahi unique name for
# this MPD server on the network.
#zeroconf_name "Music Player"
# Permissions #################################################################
# If this setting is set, MPD will require password authorization. The password
# can setting can be specified multiple times for different password profiles.
#password "password@read,add,control,admin"
# This setting specifies the permissions a user has who has not yet logged in.
#default_permissions "read,add,control,admin"
# Database #######################################################################
#database {
# plugin "proxy"
# host "other.mpd.host"
# port "6600"
# Input #######################################################################
input {
plugin "curl"
proxy "proxy.isp.com:8080"
proxy_user "user"
proxy_password "password"
# Audio Output ################################################################
# MPD supports various audio output types, as well as playing through multiple
# audio outputs at the same time, through multiple audio_output settings
# blocks. Setting this block is optional, though the server will only attempt
# autodetection for one sound card.
# An example of an ALSA output:
#audio_output {
# type "alsa"
# name "My ALSA Device"
## device "hw:0,0" # optional
## mixer_type "hardware" # optional
## mixer_device "default" # optional
## mixer_control "PCM" # optional
## mixer_index "0" # optional
audio_output {
type "pulse"
name "pulse audio"
format "48000:16:2"
server "127.0.0.1"
## sink "remote_server_sink" # optional
# An example of an OSS output:
#audio_output {
# type "oss"
# name "My OSS Device"
## device "/dev/dsp" # optional
## mixer_type "hardware" # optional
## mixer_device "/dev/mixer" # optional
## mixer_control "PCM" # optional
# An example of a shout output (for streaming to Icecast):
#audio_output {
# type "shout"
# encoding "ogg" # optional
# name "My Shout Stream"
# host "localhost"
# port "8000"
# mount "/mpd.ogg"
# password "hackme"
# quality "5.0"
# bitrate "128"
# format "44100:16:1"
## protocol "icecast2" # optional
## user "source" # optional
## description "My Stream Description" # optional
## url "http://example.com" # optional
## genre "jazz" # optional
## public "no" # optional
## timeout "2" # optional
## mixer_type "software" # optional
# An example of a recorder output:
#audio_output {
# type "recorder"
# name "My recorder"
# encoder "vorbis" # optional, vorbis or lame
# path "/var/lib/mpd/recorder/mpd.ogg"
## quality "5.0" # do not define if bitrate is defined
# bitrate "128" # do not define if quality is defined
# format "44100:16:1"
# An example of a httpd output (built-in HTTP streaming server):
#audio_output {
# type "httpd"
# name "My HTTP Stream"
# encoder "vorbis" # optional, vorbis or lame
# port "8000"
# bind_to_address "0.0.0.0" # optional, IPv4 or IPv6
## quality "5.0" # do not define if bitrate is defined
# bitrate "128" # do not define if quality is defined
# format "44100:16:1"
# max_clients "0" # optional 0=no limit
# An example of a pulseaudio output (streaming to a remote pulseaudio server)
#audio_output {
# type "pulse"
# name "My Pulse Output"
## server "remote_server" # optional
## sink "remote_server_sink" # optional
# An example of a winmm output (Windows multimedia API).
#audio_output {
# type "winmm"
# name "My WinMM output"
## device "Digital Audio (S/PDIF) (High Definition Audio Device)" # optional
# or
## device "0" # optional
## mixer_type "hardware" # optional
# An example of an openal output.
#audio_output {
# type "openal"
# name "My OpenAL output"
## device "Digital Audio (S/PDIF) (High Definition Audio Device)" # optional
## Example "pipe" output:
#audio_output {
# type "pipe"
# name "my pipe"
# command "aplay -f cd 2>/dev/null"
## Or if you're want to use AudioCompress
# command "AudioCompress -m | aplay -f cd 2>/dev/null"
## Or to send raw PCM stream through PCM:
# command "nc example.org 8765"
# format "44100:16:2"
## An example of a null output (for no audio output):
#audio_output {
# type "null"
# name "My Null Output"
# mixer_type "none" # optional
# If MPD has been compiled with libsamplerate support, this setting specifies
# the sample rate converter to use. Possible values can be found in the
# mpd.conf man page or the libsamplerate documentation. By default, this is
# setting is disabled.
#samplerate_converter "Fastest Sinc Interpolator"
# Normalization automatic volume adjustments ##################################
# This setting specifies the type of ReplayGain to use. This setting can have
# the argument "off", "album", "track" or "auto". "auto" is a special mode that
# chooses between "track" and "album" depending on the current state of
# random playback. If random playback is enabled then "track" mode is used.
# See <http://www.replaygain.org> for more details about ReplayGain.
# This setting is off by default.
replaygain "album"
# This setting sets the pre-amp used for files that have ReplayGain tags. By
# default this setting is disabled.
replaygain_preamp "0"
# This setting sets the pre-amp used for files that do NOT have ReplayGain tags.
# By default this setting is disabled.
#replaygain_missing_preamp "0"
# This setting enables or disables ReplayGain limiting.
# MPD calculates actual amplification based on the ReplayGain tags
# and replaygain_preamp / replaygain_missing_preamp setting.
# If replaygain_limit is enabled MPD will never amplify audio signal
# above its original level. If replaygain_limit is disabled such amplification
# might occur. By default this setting is enabled.
#replaygain_limit "yes"
# This setting enables on-the-fly normalization volume adjustment. This will
# result in the volume of all playing audio to be adjusted so the output has
# equal "loudness". This setting is disabled by default.
#volume_normalization "no"
# MPD Internal Buffering ######################################################
# This setting adjusts the size of internal decoded audio buffering. Changing
# this may have undesired effects. Don't change this if you don't know what you
# are doing.
#audio_buffer_size "4096"
# This setting controls the percentage of the buffer which is filled before
# beginning to play. Increasing this reduces the chance of audio file skipping,
# at the cost of increased time prior to audio playback.
#buffer_before_play "10%"
# Resource Limitations ########################################################
# These settings are various limitations to prevent MPD from using too many
# resources. Generally, these settings should be minimized to prevent security
# risks, depending on the operating resources.
#connection_timeout "60"
#max_connections "10"
#max_playlist_length "16384"
#max_command_list_size "2048"
#max_output_buffer_size "8192"
# Character Encoding ##########################################################
# If file or directory names do not display correctly for your locale then you
# may need to modify this setting.
#filesystem_charset "UTF-8"
# This setting controls the encoding that ID3v1 tags should be converted from.
#id3v1_encoding "ISO-8859-1"
# SIDPlay decoder #############################################################
# songlength_database:
# Location of your songlengths file, as distributed with the HVSC.
# The sidplay plugin checks this for matching MD5 fingerprints.
# See http://www.c64.org/HVSC/DOCUMENTS/Songlengths.faq
# default_songlength:
# This is the default playing time in seconds for songs not in the
# songlength database, or in case you're not using a database.
# A value of 0 means play indefinitely.
# filter:
# Turns the SID filter emulation on or off.
#decoder {
# plugin "sidplay"
# songlength_database "/media/C64Music/DOCUMENTS/Songlengths.txt"
# default_songlength "120"
# filter "true"
/etc/X11/xinit/xinitrc.d/pulseaudio [not sure if I even need this?]
#!/bin/bash
case "$DESKTOP_SESSION" in
gnome|kde*|xfce*) # PulseAudio is started via XDG Autostart
# Extra checks in case DESKTOP_SESSION is not set correctly
if [[ -z $KDE_FULL_SESSION && -z $GNOME_DESKTOP_SESSION_ID ]]; then
/usr/bin/start-pulseaudio-x11
fi
esac
/etc/pulse/daemon.conf
# 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.
## Configuration file for the PulseAudio daemon. See pulse-daemon.conf(5) for
## more information. Default values are commented out. Use either ; or # for
## commenting.
; daemonize = no
; fail = yes
; allow-module-loading = yes
; allow-exit = yes
; use-pid-file = yes
; system-instance = no
; local-server-type = user
; enable-shm = yes
; shm-size-bytes = 0 # setting this 0 will use the system-default, usually 64 MiB
; lock-memory = no
; cpu-limit = no
; high-priority = yes
; nice-level = -11
; realtime-scheduling = yes
; realtime-priority = 5
exit-idle-time=0
; exit-idle-time = 20
; scache-idle-time = 20
; dl-search-path = (depends on architecture)
; load-default-script-file = yes
; default-script-file = /etc/pulse/default.pa
; log-target = auto
log-level = error
; log-meta = no
; log-time = no
; log-backtrace = 0
resample-method = speex-float-7
; enable-remixing = yes
; enable-lfe-remixing = no
flat-volumes = no
; rlimit-fsize = -1
; rlimit-data = -1
; rlimit-stack = -1
; rlimit-core = -1
; rlimit-as = -1
; rlimit-rss = -1
; rlimit-nproc = -1
; rlimit-nofile = 256
; rlimit-memlock = -1
; rlimit-locks = -1
; rlimit-sigpending = -1
; rlimit-msgqueue = -1
; rlimit-nice = 31
; rlimit-rtprio = 9
; rlimit-rttime = 1000000
default-sample-format = s24le
default-sample-rate = 48000
; alternate-sample-rate = 48000
; default-sample-channels = 2
; default-channel-map = front-left,front-right
default-fragments = 2
default-fragment-size-msec = 76
; enable-deferred-volume = yes
; deferred-volume-safety-margin-usec = 8000
; deferred-volume-extra-delay-usec = 0
/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
.fail
### Automatically restore the volume of streams and devices
load-module module-device-restore
load-module module-stream-restore
load-module module-card-restore
### Automatically augment property information from .desktop files
### stored in /usr/share/application
load-module module-augment-properties
### Should be after module-*-restore but before module-*-detect
load-module module-switch-on-port-available
### Load audio drivers statically
### (it's probably better to not load these drivers manually, but instead
### use module-udev-detect -- see below -- for doing this automatically)
#load-module module-alsa-sink
#load-module module-alsa-source device=hw:1,0
#load-module module-oss device="/dev/dsp" sink_name=output source_name=input
#load-module module-oss-mmap device="/dev/dsp" sink_name=output source_name=input
#load-module module-null-sink
#load-module module-pipe-sink
### Automatically load driver modules depending on the hardware available
.ifexists module-udev-detect.so
load-module module-udev-detect
.else
### Use the static hardware detection module (for systems that lack udev support)
load-module module-detect
.endif
### Automatically connect sink and source if JACK server is present
.ifexists module-jackdbus-detect.so
.nofail
load-module module-jackdbus-detect channels=2
.fail
.endif
### Automatically load driver modules for Bluetooth hardware
#.ifexists module-bluetooth-policy.so
#load-module module-bluetooth-policy
#.endif
#.ifexists module-bluetooth-discover.so
#load-module module-bluetooth-discover
#.endif
### Load several protocols
.ifexists module-esound-protocol-unix.so
load-module module-esound-protocol-unix
.endif
load-module module-native-protocol-unix
### Network access (may be configured with paprefs, so leave this commented
### here if you plan to use paprefs)
#load-module module-esound-protocol-tcp
load-module module-native-protocol-tcp auth-ip-acl=127.0.0.1
#load-module module-zeroconf-publish
### Load the RTP receiver module (also configured via paprefs, see above)
#load-module module-rtp-recv
### Load the RTP sender module (also configured via paprefs, see above)
#load-module module-null-sink sink_name=rtp format=s16be channels=2 rate=44100 sink_properties="device.description='RTP Multicast Sink'"
#load-module module-rtp-send source=rtp.monitor
### Load additional modules from GConf settings. This can be configured with the paprefs tool.
### Please keep in mind that the modules configured by paprefs might conflict with manually
### loaded modules.
.ifexists module-gconf.so
.nofail
load-module module-gconf
.fail
.endif
### Automatically restore the default sink/source when changed by the user
### during runtime
### NOTE: This should be loaded as early as possible so that subsequent modules
### that look up the default sink/source get the right value
load-module module-default-device-restore
### Automatically move streams to the default sink if the sink they are
### connected to dies, similar for sources
load-module module-rescue-streams
### Make sure we always have a sink around, even if it is a null sink.
load-module module-always-sink
### Honour intended role device property
load-module module-intended-roles
### Automatically suspend sinks/sources that become idle for too long
load-module module-suspend-on-idle
### If autoexit on idle is enabled we want to make sure we only quit
### when no local session needs us anymore.
.ifexists module-console-kit.so
load-module module-console-kit
.endif
.ifexists module-systemd-login.so
load-module module-systemd-login
.endif
### Enable positioned event sounds
load-module module-position-event-sounds
### Cork music/video streams when a phone stream is active
load-module module-role-cork
### Modules to allow autoloading of filters (such as echo cancellation)
### on demand. module-filter-heuristics tries to determine what filters
### make sense, and module-filter-apply does the heavy-lifting of
### loading modules and rerouting streams.
load-module module-filter-heuristics
load-module module-filter-apply
# X11 modules should not be started from default.pa so that one daemon
# can be shared by multiple sessions.
### Load X11 bell module
#load-module module-x11-bell sample=bell-windowing-system
### Register ourselves in the X11 session manager
#load-module module-x11-xsmp
### Publish connection data in the X11 root window
#.ifexists module-x11-publish.so
#.nofail
#load-module module-x11-publish
#.fail
#.endif
### Make some devices default
#set-default-sink output
#set-default-source input
~/.xinitrc [currently using daemon mpd]
#!/bin/sh
# ~/.xinitrc
# Executed by startx (run your window manager from here)
if [ -d /etc/X11/xinit/xinitrc.d ]; then
for f in /etc/X11/xinit/xinitrc.d/*; do
[ -x "$f" ] && . "$f"
done
unset f
fi
#if grep -q "state: pause" /home/quiv/.config/mpd/state; then
# mpc toggle
#fi
#mpd ~/.config/mpd/mpd.conf
~/.config/bspwm/panel/notify_mpd
xrdb ~/.Xresources
xset +fp ~/.fonts
xset +fp /usr/share/fonts/misc
xset fp rehash
xsetroot -cursor_name left_ptr &
# imlibsetroot -x e -s f /home/quiv/Pictures/bloom_one_desktop.jpg
sh ~/.fehbg &
compton -CGb --backend glx --paint-on-overlay --vsync opengl-swc &
# exec gnome-session
# exec startkde
# exec startxfce4
# ...or the Window Manager of your choice
sxhkd &
exec bspwm
~/.zprofile [currently trying to use systemd/User]
#[[ -z $DISPLAY && $XDG_VTNR -eq 1 ]] && exec startx
EDIT:A small update; it seems I don't need to have /etc/systemd/system/[email protected]/autologin.conf when using systemd to manag my user session as I removed it and I'm both automatically logged on and in X-session.
Last edited by quiv (2014-07-05 13:40:30)o_caino wrote:
Setting mpd as a systemd user service is very simple. This is what I did.
In ~/.config/systemd/user/mpd.service
[Unit]
Description=Music Player Daemon
After=network.target sound.target
[Service]
ExecStart=/usr/bin/mpd --no-daemon
ExecStop=/usr/bin/mpd --kill
[Install]
WantedBy=default.target
To enable
systemctl --user enable mpd
Done.
Well shit. I guess I didn't need to do all that other stuff. I only wish I'd known about this before wasting most of my day. Thank you very much, everything appears to be working flawlessly now. -
(Solved) MPD Pulseaudio client no sound
Hi,
I've installed MPD on my server, configured the music path I can play music on clients, but I don't have sound on client !
I've try audio_output HTTP and Icecast, I'm not interrested in these two solutions.
So I wanted to broadcast sound over network, so I leanrt about Pulseaudio, in my /etc/mpd.conf I uncomment the pulse sound card.
Edit /etc/pulse/default.pa in order to add my client IP
load-module module-native-protocol-tcp auth-ip-acl=192.168.1.5
And start pulseaudio --start.
On the client edit the /etc/pulse/client.conf to add server ip address.
default-server=192.168.1.10
And started as well !
Message error on client :
User-configured server at 192.168.1.10 refusing to start /autospawn.
I simply want to broadcast the audio_outup in the server to the clients
How can I fix it ?
Thanks in advance.
Last edited by NeanderMarcl (2013-07-15 11:49:36)Okay, so it is a back end problem.
In Pavucontrol, go to the Output Devices tab and check the port pull down. Make sure yours is set for something rational. Maybe try all the ones that make sense. Also, go to the configuration tab and look at the various profiles. Make sure that it, too, selects something rational; If you are using built in speakers, make sure it is not trying to use HDMI.
Then, pop over to your favorite mixer and make sure all the controls are un-muted and that the volumes are up.
If none of that helps, post the output of
pacmd dump
pacmd dump-volumes
and of
amixer
ewaller$@$odin ~ 1005 %pacmd dump-volumes
Welcome to PulseAudio! Use "help" for usage information.
>>> Sink 0: reference = 0: 35% 1: 35%, real = 0: 35% 1: 35%, soft = 0: 100% 1: 100%, current_hw = 0: 35% 1: 35%, save = yes
Input 0: volume = 0: 35% 1: 35%, reference_ratio = 0: 100% 1: 100%, real_ratio = 0: 100% 1: 100%, soft = 0: 100% 1: 100%, volume_factor = 0: 100% 1: 100%, volume_factor_sink = 0: 100% 1: 100%, save = yes
Source 0: reference = 0: 153% 1: 153%, real = 0: 153% 1: 153%, soft = 0: 153% 1: 153%, current_hw = 0: 153% 1: 153%, save = yes
Source 1: reference = 0: 100% 1: 100%, real = 0: 100% 1: 100%, soft = 0: 100% 1: 100%, current_hw = 0: 100% 1: 100%, save = yes
ewaller$@$odin ~ 1006 %pacmd dump
Welcome to PulseAudio! Use "help" for usage information.
>>> ### Configuration dump generated at Sun Jul 14 09:05:28 2013
load-module module-device-restore
load-module module-stream-restore
load-module module-card-restore
load-module module-augment-properties
load-module module-switch-on-port-available
load-module module-alsa-card device_id="0" name="pci-0000_00_1b.0" card_name="alsa_card.pci-0000_00_1b.0" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes card_properties="module-udev-detect.discovered=1"
load-module module-udev-detect
load-module module-jackdbus-detect channels=2
load-module module-bluetooth-policy
load-module module-bluetooth-discover
load-module module-esound-protocol-unix
load-module module-native-protocol-unix
load-module module-gconf
load-module module-default-device-restore
load-module module-rescue-streams
load-module module-always-sink
load-module module-intended-roles
load-module module-suspend-on-idle
load-module module-console-kit
load-module module-systemd-login
load-module module-position-event-sounds
load-module module-role-cork
load-module module-filter-heuristics
load-module module-filter-apply
load-module module-dbus-protocol
load-module module-cli-protocol-unix
set-sink-volume alsa_output.pci-0000_00_1b.0.analog-stereo 0x5ad5
set-sink-mute alsa_output.pci-0000_00_1b.0.analog-stereo no
suspend-sink alsa_output.pci-0000_00_1b.0.analog-stereo no
set-source-volume alsa_output.pci-0000_00_1b.0.analog-stereo.monitor 0x18675
set-source-mute alsa_output.pci-0000_00_1b.0.analog-stereo.monitor no
suspend-source alsa_output.pci-0000_00_1b.0.analog-stereo.monitor no
set-source-volume alsa_input.pci-0000_00_1b.0.analog-stereo 0xffff
set-source-mute alsa_input.pci-0000_00_1b.0.analog-stereo yes
suspend-source alsa_input.pci-0000_00_1b.0.analog-stereo yes
set-card-profile alsa_card.pci-0000_00_1b.0 output:analog-stereo+input:analog-stereo
set-default-sink alsa_output.pci-0000_00_1b.0.analog-stereo
set-default-source alsa_input.pci-0000_00_1b.0.analog-stereo
### EOF
>>> %
ewaller$@$odin ~ 1007 %amixer
Simple mixer control 'Master',0
Capabilities: pvolume pswitch pswitch-joined
Playback channels: Front Left - Front Right
Limits: Playback 0 - 65536
Mono:
Front Left: Playback 23253 [35%] [on]
Front Right: Playback 23253 [35%] [on]
Simple mixer control 'Capture',0
Capabilities: cvolume cswitch cswitch-joined
Capture channels: Front Left - Front Right
Limits: Capture 0 - 65536
Front Left: Capture 65535 [100%] [off]
Front Right: Capture 65535 [100%] [off]
ewaller$@$odin ~ 1008 %
Last edited by ewaller (2013-07-14 16:10:08)
Maybe you are looking for
-
I have a person that is called "josée". I need "JOSEE" to be inserted into the database, not "JOSéE". I also have a person called "jürgen". I need "JURGEN" to be inserted into the database. Is there a way to do this ? (replace the non-ascii characte
-
How can I add a latest tweet box over this image?
Hi, I have tried to overlay a latest tweet box onto this picture however I am having no luck could you please help me? *** willlh.com *** This is the website and the tweet box is on the left below the blue twitter logo. Any help appreciated, thanks,
-
Does imessage send to iphones(with imessage) without internet?
im considering getting an ipod and i want to know will i be able to imessage people even if i do not have access to internet?
-
I do not need Mac OS9 Classic so can I just erase everything and start fresh with Mac OS 10.3.9? I have the installation discs for that program and that is what system is running now.
-
Importing music form windows media player
Hey I need help Bad. I have all of my music stored on windows media player and I can't figure out how to get my playlists on to iTunes. I've tried everything I can think of. I tried importing file(s) and folders. none seem to work. Could someone plea