Tomahawk Player - Sporadic "Problem Accessing Audio Device" Errors

Hi, I have recently started using Tomahawk Player in GNOME and some songs from external sources can play and others receive an error that states:
"Sorry, there is a problem accessing your audio device. Make sure you have a suitable Phonon backend and required plugins installed."
However, I have phonon, phonon-gstreamer, and phonon-vlc installed.
How can I further diagnose this?
Last edited by bdamos (2014-11-30 19:48:16)

Hello,
That is unfortunately a not-very-helpful error message as it could mask a variety of problems. One of those problems is that it might happen if Tomahawk is unable to play the url for the track---what source has it resolved to? (rightmost column)?
Feel free to come ask us on Freenode (#tomahawk) and our forums on forum.tomahawk-player.org. You'll probably get a better response there.

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MPD: ERROR: problems opening audio device

Since a day, maybe a little more, mpd is troubling me. When I use mpc, I get the error
ERROR: problems opening audio device
This is my mpd.conf
# An example configuration file for MPD
# See the mpd.conf man page for a more detailed description of each parameter.
# 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 "/home/adriaan/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 "/var/lib/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 "/var/lib/mpd/mpd.db"
# 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 "/var/log/mpd/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 "/var/run/mpd/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 "/var/lib/mpd/mpdstate"
# The location of the sticker database. This is a database which
# manages dynamic information attached to songs.
#sticker_file "~/.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 "adriaan"
# 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 "users"
# 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 "any"
# And for Unix Socket
#bind_to_address "~/.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"
# 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. Optionally, 'comment' can be added to this
# list.
#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"
# 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.
# See <http://mpd.wikia.com/wiki/Configuration#Audio_Outputs> for examples of
# other audio outputs.
# An example of an ALSA output:
audio_output {
type "alsa"
name "My ALSA Device"
device "hw:0,0" # optional
# format "44100:16:2" # optional
# mixer_type "hardware" # optional
# mixer_device "default" # optional
# mixer_control "PCM" # optional
# mixer_index "0" # optional
#audio_output {
# type "alsa"
# name "Sound Card"
# options "dev=dmixer"
# device "plug:dmix"
# An example of an OSS output:
#audio_output {
# type "oss"
# name "My OSS Device"
## device "/dev/dsp" # optional
## format "44100:16:2" # 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
## 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
## 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
# This setting will change all decoded audio to be converted to the specified
# format before being passed to the audio outputs. By default, this setting is
# disabled.
#audio_output_format "44100:16:2"
# 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" or "track". See <http://www.replaygain.org>
# for more details. 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 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 "2048"
# 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"

milso wrote:
type "alsa"
name "equal"
device "default"
format "44100:16:2" # optional
mixer_device "default" # optional
mixer_control "PCM" # optional
mixer_index "0" # optional
I have mine set to plug:plugequal as I use an equalizer with alsa, but when I set it to default it works fine, try that.
That didn't work out.
Mr.Elendig wrote:Tell mpd to be more verbose. And check that mplayer -ao alsa whatever works.
Here's the output of mpd -v
listen: bind to '0.0.0.0:6600' failed: Address already in use (continuing anyway, because binding to '[::]:6600' succeeded)
path: path_set_fs_charset: fs charset is: UTF-8
database: reading DB
disabling the last.fm playlist plugin because account is not configured
daemon: opening pid file
daemon: daemonized!
daemon: writing pid file
Doesn't contain anything unusual.
EDIT: oh, and mplayer -ao alsa works fine.
Last edited by arthd (2011-02-09 15:14:09)

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P/N:MN564PA#UUF
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This question was solved.
View Solution.

Hi
You have a couple of options as I see it.
First disconnect the Battery & Ac Adapter then press and hold the power button on the unit for 30 second then check by powering on the unit.
Go to device manager and uninstall the driver and restart and unit will install the driver automatically.
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Try to update the driver from hp.com website for Audio & Bios.
Let us know how it goes!
*Although I am an HP employee, I am speaking for myself and not for HP.
****Click the White Kudos star to say thanks****
****Please mark Accept As Solution if it solves your problem****
Regards
Manjunath
Let us know how it goes!
"I work for HP."
****Click the (purple thumbs up icon in the lower right corner of a post) to say thanks****
****Please mark Accept As Solution if it solves your problem****
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i have already posted my problem as a reply in this topic: http://bbs.archlinux.org/viewtopic.php?id=47723, howver i'm not suer what section it would really fit in so and the solution i believe the solution is trivial so i sort of duplicate it here...I hope there is no problem.
Well, the problem is that i cannot start jackd -d firewire as a normal user, as root tho, there is no problem and i have tried to play audio files through jack using mplayer w/o any problem... I own a Focusrite Saffire and have installed FFADO libraries.
Error output:
$ jackd -v -d firewire
getting driver descriptor from /usr/lib64/jack/jack_oss.so
getting driver descriptor from /usr/lib64/jack/jack_dummy.so
getting driver descriptor from /usr/lib64/jack/jack_net.so
getting driver descriptor from /usr/lib64/jack/jack_alsa.so
getting driver descriptor from /usr/lib64/jack/jack_firewire.so
no message buffer overruns
jackd 0.116.2
Copyright[...]
JACK compiled with System V SHM support.
server `default' registered
registered builtin port type 32 bit float mono audio
registered builtin port type 8 bit raw midi
clock source = system clock via clock_gettime
loading driver ..
start poll on 3 fd's
can't load "/usr/lib64/jack/jack_firewire.so": libasound.so.2: failed to map segment from shared object: Resource temporarily unavailable
cannot load driver module firewire
starting server engine shutdown
freeing shared port segments
stopping server thread
last xrun delay: 0.000 usecs
max delay reported by backend: 0.000 usecs
freeing engine shared memory
max usecs: 0.000, engine deleted
cleaning up shared memory
cleaning up files
unregistering server `default'
As root:
$ sudo jackd -v -d firewire
getting driver descriptor from /usr/lib64/jack/jack_oss.so
getting driver descriptor from /usr/lib64/jack/jack_dummy.so
getting driver descriptor from /usr/lib64/jack/jack_net.so
getting driver descriptor from /usr/lib64/jack/jack_alsa.so
getting driver descriptor from /usr/lib64/jack/jack_firewire.so
no message buffer overruns
jackd 0.116.2
Copyright ...
JACK compiled with System V SHM support.
server `default' registered
registered builtin port type 32 bit float mono audio
registered builtin port type 8 bit raw midi
clock source = system clock via clock_gettime
loading driver ..
start poll on 3 fd's
new client: firewire_pcm, id = 1 type 1 @ 0x2380fd0 fd = -1
new buffer size 1024
libffado 1.999.43-1582 built Jul 11 2009 17:45:36
firewire MSG: Streaming thread running with Realtime scheduling, priority 10
firewire MSG: Registering audio capture port C0_dev0_Rec 1
firewire MSG: Registering audio capture port C1_dev0_Rec 2
firewire MSG: Registering audio capture port C2_dev0_Rec 3
firewire MSG: Registering audio capture port C3_dev0_Rec 4
firewire MSG: Registering midi capture port C4_dev0_Midi In
firewire MSG: Registering audio playback port P0_dev0_Play 1
firewire MSG: Registering audio playback port P1_dev0_Play 2
firewire MSG: Registering audio playback port P2_dev0_Play 3
firewire MSG: Registering audio playback port P3_dev0_Play 4
firewire MSG: Registering audio playback port P4_dev0_Play 5
firewire MSG: Registering audio playback port P5_dev0_Play 6
firewire MSG: Registering audio playback port P6_dev0_Play 7
firewire MSG: Registering audio playback port P7_dev0_Play 8
firewire MSG: Registering audio playback port P8_dev0_Play 9
firewire MSG: Registering audio playback port P9_dev0_Play 10
firewire MSG: Registering midi playback port P10_dev0_Midi Out
registered port system:capture_1, offset = 4096
registered port system:capture_2, offset = 8192
registered port system:capture_3, offset = 12288
registered port system:capture_4, offset = 16384
registered port firewire_pcm:C4_dev0_Midi In, offset = 4096
registered port system:playback_1, offset = 0
registered port system:playback_2, offset = 0
registered port system:playback_3, offset = 0
registered port system:playback_4, offset = 0
registered port system:playback_5, offset = 0
registered port system:playback_6, offset = 0
registered port system:playback_7, offset = 0
registered port system:playback_8, offset = 0
registered port system:playback_9, offset = 0
registered port system:playback_10, offset = 0
registered port firewire_pcm:P10_dev0_Midi Out, offset = 0
++ jack_sort_graph
++ jack_rechain_graph():
+++ client is now firewire_pcm active ? 1
client firewire_pcm: internal client, execution_order=0.
-- jack_rechain_graph()
-- jack_sort_graph
4239 waiting for signals
load = 0.2531 max usecs: 108.000, spare = 21225.000
some info:
$cat /etc/fstab
# /etc/fstab: static file system information
# <file system> <dir> <type> <options> <dump> <pass>
none /dev/pts devpts defaults 0 0
####none /dev/shm tmpfs defaults 0 0
###ADDED
shm /dev/shm tmpfs nodev,nosuid,noexec 0 0
#/dev/cdrom /media/cd auto ro,user,noauto,unhide 0 0
#/dev/dvd /media/dvd auto ro,user,noauto,unhide 0 0
#/dev/fd0 /media/fl auto user,noauto 0 0
UUID=512fbb4a-7c37-4a91-94fd-c2f75bc3ad1c swap swap defaults 0 0
UUID=ce95308e-a444-4d12-a3bf-a1abdcc82eed / ext4 defaults 0 1
lsmod | grep 1394
dv1394 21832 0
ohci1394 34052 1 dv1394
raw1394 29000 0
ieee1394 98912 3 dv1394,ohci1394,raw1394
Should I change udev rules group to audio? How do I do that in Archlinux? :$
$ ls -l /dev/raw1394
crwxrw-rw- 1 root video 171, 0 2009-07-12 14:31 /dev/raw1394
$ groups
network video audio optical storage users
I think it's some sort of permission problem....Any suggestion is welcome! thank you!
Last edited by 00l0 (2009-07-11 20:52:36)

djszapi wrote:"I hope there is no problem" -> If it was the last i think not
good!
Mr.Elendig wrote:Try adding your user to the video group, since that's the owner of the device.
I already did, forgot to show it
$ groups
network video audio optical storage users

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Hi
Jimi is right. You have to update the Yamaha sound driver!
After a short investigation I found this site:
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Best regards

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provided by the same coldfusion CFC.
I've verified that I can successfully access the web services
via HTTP, and one of the web services actually does work when
called. Both return structures that contain a string var, a numeric
var, and one or more query vars.
Here is my web service declaration:
quote:
<mx:WebService id="ws"
wsdl="https://www.it.dev.duke.edu/components/dukemagsearch/checkMailing2.cfc?wsdl"
useProxy="false">
<mx:operation name="queryDB" result="queryDBResult()"
fault="queryDBFault(event)">
<mx:request>
<RUNDATE>{cboRunDate.selectedItem.XDATE}</RUNDATE>
<ENTITYID>{txtEntityID.text}</ENTITYID>
<LASTNAME>{txtLastName.text}</LASTNAME>
<FIRSTNAME>{txtFirstName.text}</FIRSTNAME>
<MINITIAL>{txtMiddleInitial.text}</MINITIAL>
<PRFSCHCD>{cboPreferredSchool.selectedItem.TABLKEY}</PRFSCHCD>
<PRRECTYP>{cboRecordType.selectedItem.TABLKEY}</PRRECTYP>
<PRFCLASS>{txtPreferredClass.text}</PRFCLASS>
</mx:request>
</mx:operation>
<mx:operation name="getListData"
result="getListDataResult()" fault="getListDataFault()"/>
</mx:WebService>
Sorry if the formatting sucks, I don't know how to post
"code" here.
Anyway, the getListData() method works fine and populates my
list boxes. But when I call the queryDB() method, I get the
following fault event:
quote:
[FaultEvent fault=[RPC Fault faultString="HTTP request error"
faultCode="Server.Error.Request" faultDetail="Error: [IOErrorEvent
type="ioError" bubbles=false cancelable=false eventPhase=2
text="Error #2032: Stream Error. URL:
https://www.it.dev.duke.edu/components/dukemagsearch/checkMailing2.cfc"].
URL:
https://www.it.dev.duke.edu/components/dukemagsearch/checkMailing2.cfc"]
messageId="1E553B99-DF28-ED72-62B8-B84AA3919F9A" type="fault"
bubbles=false cancelable=true eventPhase=2]
I've looked all over the place and I can't seem to find what
this error means. I've tried all kinds of different ways of doing
things. Right now I'm browsing the flex app via a file URL (C:\...)
but I tried putting it up on the server too and that didn't work
either.
The method call *DOES* work when called via HTTP... ie
https://www.it.dev.duke.edu/components/dukemagsearch/checkMailing2.cfc?method=queryDB&RUND ATE=2006-05-19&FIRSTNAME=&LASTNAME=SMITH&MINITIAL=&PRFSCHCD=&PRRECTYP=AL&PRFCLASS=&ENTITYI D=
(You have to be logged in for that to work so you'll just
have to trust me, it returns no records if you're not logged into
the web site already).
I'm totally stressing out about this because I've essentially
spent the entire day since 8am trying to solve this. The
application should've taken 15 minutes total.
HELP!
Thanks for any suggestions y'all have.
Rick

It means your output was not formed correctly and could not
be parsed. set up a server side script or something to check that
the output is indeed what you think it should be, 99% of the time i
get thie error its due to malformed output from my webservice or
db. Also try making an xml model of your target data to test the
application internaly, look up model in the docs, it easy to use
and if the model works then you know the data is faulty and you
need to check your output and queries.

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Picture link (hosted on photobucket.com)
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I tried to uninstall this and let windows update install it again => not worked, same problem.
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Hi
I am trying to get flash player to play nice with averMedia Dark Crystal HD Capture SDK II (C729). But Flash player does not find any audio devices on the system. IE: Microphones.Names is 0 in length. Flash player settings window also dont show anything.
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both drivers marked with purple colour!
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hi luis,
thank you for your reply. we've checked the smoothwall configuration, but couldn't discover anything which could cause this problem. we even tried replacing the sa520 with a draytek vigor router to set up an lan-to-lan vpn with the smoothwall. with the draytek in place we have no problems accessing the aforementioned servers, so it seems the issue is with the SA520.
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here's how i set up the rule:
from zone: UNSECURE (WAN/optional WAN)
to zone: LAN
service: ANY
action: ALLOW always
schedule: (not set)
source hosts: Single address
from: public ip of one of the aforementioned servers
source NAT settings > external IP address: WAN interface address (cannot change this setting)
source NAT settings >WAN interface: dedicated WAN (cannot change this setting)
destination NAT settings > internal ip address: 192.168.11.123 (ip address of my machine)
enable port forwarding: unchecked
translate port number: empty
external IP address: dedicated WAN

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Hi..
How to select the Audio Device in Windows using ActiveX from LabVIEW....
Im using Windows media player Version9..and LV8....
Regards,
Sodlb

Hi Sodlb-
According to the Windows Media - Audio Outputs section on MSDN (see below), unfortunately you will not be able to choose a different output device via ActiveX. This is obviously a limitation outside of LabVIEW.
Windows Media Player ActiveX control
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