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keenblade
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Posts: 1087

PostPosted: Mon Oct 03, 2011 1:55 pm    Post subject: Reply with quote

Try this in your mpd.conf. Lets see if we can get 5.1 audio.
Code:

mixer_type "software"

audio_output {
          type    "alsa"
          device  "plug:upmix_20to51"
          name    "upmix_20to51"
        auto_resample "no"
        use_mmap      "yes"
        }
audio_output {
          type    "alsa"
          device  "plug:ch51dup"
          name    "ch51dup"
        auto_resample "no"
        use_mmap      "yes"
        }
audio_output {
          type    "alsa"
          device  "plug:lowpass_21to21"
          name    "lowpass_21to21"
        auto_resample "no"
        use_mmap      "yes"
        }

Actually, in gmpc selecting "upmix_20to51" must work. But for some reason mine works with "lowpass_21to21" or "ch51dup". Strange. Try all these outputs one by one in gmpc "server" menu.
For this to work we need to change your .asoundrc. I will post one for you very soon. I am adjusting for you.

Also in the first line of your .asoundrc there is a typo. You forgot to type "lt" at the end of first line.
pcm.!defau
must be
pcm.!default
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Anyway it's all the same at the end...
Need help to get it working: "x-fi surround 5.1"
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dr-peppa
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Joined: 28 Apr 2011
Posts: 25

PostPosted: Mon Oct 03, 2011 2:39 pm    Post subject: Reply with quote

This is what I get:
mpd.log
Code:


ALSA lib pcm.c:2212:(snd_pcm_open_noupdate) Unknown PCM upmix_20to51
Oct 03 16:39 : output: Failed to open "upmix_20to51" [alsa]: Failed to open ALSA device "plug:upmix_20to51": No such file or directory
ALSA lib pcm.c:2212:(snd_pcm_open_noupdate) Unknown PCM ch51dup
Oct 03 16:39 : output: Failed to open "ch51dup" [alsa]: Failed to open ALSA device "plug:ch51dup": No such file or directory
ALSA lib pcm.c:2212:(snd_pcm_open_noupdate) Unknown PCM lowpass_21to21
Oct 03 16:39 : output: Failed to open "lowpass_21to21" [alsa]: Failed to open ALSA device "plug:lowpass_21to21": No such file or directory


There's no typo, but copying was messed up. sry. ;-)

When I select upmix from within gmpc I get:
Code:

Oct 03 16:40 : output: Failed to open "upmix_20to51" [alsa]: Failed to open ALSA device "plug:upmix_20to51": No such file or directory


None of them seems to be recognized by alsa...
In mplayer however, it's working fine.
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keenblade
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PostPosted: Mon Oct 03, 2011 3:00 pm    Post subject: Reply with quote

Ok. No typo but still .asoundrc is wrong because "upmix_20to51" is not defined right. Try this .asoundrc
Code:

pcm.dmixed {
    type   asym
    playback.pcm {
           type   dmix
           ipc_key 5678293
       ipc_perm 0660
       ipc_gid audio
       slave {
               channels 6
        pcm {
                   format S16_LE
              rate 48000
              type hw
                  card 0
                  device 0
                  subdevice 0
              }
          }
 
      }
 }

pcm.!default {
    type            plug
    slave.pcm       "dmixed"
}

ctl.!default {
    type hw
    card 0
}

ctl.softvol
{
type hw
card 0
}

pcm.softvol {
type softvol
slave {
pcm "upmix_20to51" #redirect the output to dmix (instead of "hw:0,0")
}
control {
name "Master" #override the PCM slider to set the softvol volume level globally
card 0
}
}

pcm.!surround20 {
   type plug
   slave.pcm "dmixed"
}

pcm.!surround40 {
   type plug
   slave.pcm "dmixed"
}

pcm.!surround51 {
   type plug
   slave.pcm "dmixed"
}

pcm.ch51dup {

        slave.pcm "dmixed"
        slave.channels 6
        type route
 
        # Front
        ttable.0.0 0.5
        ttable.1.1 0.5

        # Rear
        ttable.2.4 2
        ttable.3.5 2
        # Front left/right to rear
        ttable.0.4 1
        ttable.1.5 1

        # Center
        ttable.4.2 0.5
        # Front left/right to center
        ttable.0.2 0.5
        ttable.1.2 0.5
 
        # LFE
        ttable.5.3 1
        # Front to LFE
        ttable.0.3 0.5
        ttable.1.3 0.5
}

pcm.lowpass_21to21 {
   type ladspa
   slave.pcm upmix_21to51
   # Set the path to ladspa, to fix this error:
   # Playback open error: -2,No such file or directory
   path "/usr/lib/ladspa"
   channels 3
   plugins {
      0 {
         id 1098  # Identity (Audio) (1098/identity_audio)
         policy duplicate
         input.bindings.0 "Input";
         output.bindings.0 "Output";
      }

      1 {
         id 1052  # High-pass filter front left
         policy none
         input.bindings.0 "Input";
         output.bindings.0 "Output";
         input {
            controls [ 200 ]
         }
      }
      2 {
         id 1052  # High-pass filter front right
         policy none
         input.bindings.1 "Input";
         output.bindings.1 "Output";
         input {
            controls [ 200 ]
         }
      }

      3 {
         id 1672  # Low-pass filter.
         policy none
         input.bindings.2 "Input";
         output.bindings.2 "Output";
         input {
            controls [ 300 2 ]
         }
      }

   }
}


# speaker-test -D upmix_20to51 -c 2 -t wav
pcm.upmix_20to51 {
   type plug
   slave.pcm "lowpass_21to21"
   slave.channels 3
   ttable {
      0.0     0.5       # left channel
      1.1     0.5       # right channel
      0.3     2     # mix left and right ...
      1.3     2     # ... channel for subwoofer
   }
}

# Debugging:  speaker-test -D upmix_21to51 -c 3 -t wav
pcm.upmix_21to51 {
   type plug
#   slave.pcm "dmixed"
   slave.pcm "ch51dup"
#   slave.pcm "softvol"
   slave.channels 6
   ttable {
      0.0     1       # front left
      1.1     1       # front right
      0.4     2       # rear left
      1.5     2       # rear right

      # Front left/right to center.
      # Imbalanced because is to the left of the monitor!
      # Would normally be 0.5 each.
      0.2     0.5
      1.2     0.5

      # Subwoofer, more powerful to compensate for bass-removal from other speakers.
      2.3     2
    }
}


And check gmpc, if you get audio.
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Need help to get it working: "x-fi surround 5.1"
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dr-peppa
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Joined: 28 Apr 2011
Posts: 25

PostPosted: Mon Oct 03, 2011 3:19 pm    Post subject: Reply with quote

No, doesn't work.
"...problems opening audio-device"
log:
Code:


ALSA lib pcm.c:2212:(snd_pcm_open_noupdate) Unknown PCM upmix_20to51
Oct 03 17:14 : output: Failed to open "upmix_20to51" [alsa]: Failed to open ALSA device "plug:upmix_20to51": No such file or directory


This is my /etc/mpd.conf

Code:


# 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                 "/var/lib/mpd/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/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                        "/var/lib/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                       "~/.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/state"
#
# 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                            "mpd"
#
# 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         "localhost"
#
# And for Unix Socket
bind_to_address         "/var/lib/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      "software"      # optional
#       mixer_device    "default"       # optional
#       mixer_control   "PCM"           # optional
#       mixer_index     "0"             # optional
#}
#
# 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"
#}
#
###############################################################################
 

audio_output {
          type    "alsa"
          device  "plug:upmix_20to51"
          name    "upmix_20to51"
        auto_resample "no"
        use_mmap      "yes"
        }
audio_output {
          type    "alsa"
          device  "plug:ch51dup"
          name    "ch51dup"
        auto_resample "no"
        use_mmap      "yes"
        }
audio_output {
          type    "alsa"
          device  "plug:lowpass_21to21"
          name    "lowpass_21to21"
        auto_resample "no"
        use_mmap      "yes"
        }
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dr-peppa
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PostPosted: Mon Oct 03, 2011 3:38 pm    Post subject: Reply with quote

even (s)mplayer doesn't work anymore.
it gives:
Code:

[AO_ALSA] alsa-lib: pcm_ladspa.c:1504:(snd_pcm_ladspa_add_plugin) Unable to find or load plugin '(null)' ID 1672, path '/usr/lib/ladspa'
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keenblade
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PostPosted: Mon Oct 03, 2011 3:59 pm    Post subject: Reply with quote

dr-peppa wrote:
even (s)mplayer doesn't work anymore.
it gives:
Code:

[AO_ALSA] alsa-lib: pcm_ladspa.c:1504:(snd_pcm_ladspa_add_plugin) Unable to find or load plugin '(null)' ID 1672, path '/usr/lib/ladspa'

That's good. We are getting to somewhere. The error just says you don't have the ladspa plugin with the ID 1672. Yo can try another lowpass filter plugin or choose the one that is installed on your pc. It is same for me, too. Now let's find out a working plugin.
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Anyway it's all the same at the end...
Need help to get it working: "x-fi surround 5.1"
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dr-peppa
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PostPosted: Mon Oct 03, 2011 4:18 pm    Post subject: Reply with quote

I uncommented the "path-line" in .asoundrc as sugested in a forum, which I found with google.
Then I set the correct ladspa-path in /etc/env.d/60-ladspa to:
Code:

LADSPA_PATH=/usr/lib64/ladspa:/usr/lib32/ladspa

The thing is, that mplayer and not mpd +gmpc gave me this ladspa-error. so I think, that there must
be something wrong with the mpd.conf part.
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dr-peppa
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PostPosted: Mon Oct 03, 2011 4:20 pm    Post subject: Reply with quote

now, playing a mp3-file as root with mplayer works but as normal user doesn't.
mpd still gives:
Code:
Oct 03 18:16 : output: Failed to open "upmix_20to51" [alsa]: Failed to open ALSA device "plug:upmix_20to51": No such file or directory
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dr-peppa
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PostPosted: Mon Oct 03, 2011 4:45 pm    Post subject: Reply with quote

I just played around a little bit.
put "ch51dup" in mpd.conf as single plugin.
now mpd.log returns
Code:
output: All audio outputs are disabled


mpd.conf looks now like this:
Code:


audio_output {
          type    "alsa"
          device  "plug:ch51dup"
          name    "ch51dup"
        auto_resample "no"
        use_mmap      "yes"
        }
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keenblade
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PostPosted: Mon Oct 03, 2011 5:32 pm    Post subject: Reply with quote

dr-peppa wrote:
now, playing a mp3-file as root with mplayer works but as normal user doesn't.
mpd still gives:
Code:
Oct 03 18:16 : output: Failed to open "upmix_20to51" [alsa]: Failed to open ALSA device "plug:upmix_20to51": No such file or directory

Since you edited the .asoundrc in your user home dir, root does not effected of this. So, as root it seems to work because, probably, your root account does not have a .asoundrc file in its home dir: /root
Anyway, the real problem is with .asoundrc file. It is very hard to understand. In my previously posted mpd.conf uses all the outputs available to sound. But actually the only right one is the output "upmix_20to51". So we need to find a way to solve the alsa language.
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Anyway it's all the same at the end...
Need help to get it working: "x-fi surround 5.1"
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dr-peppa
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PostPosted: Mon Oct 03, 2011 6:47 pm    Post subject: Reply with quote

No, I copied the user .asoundrc over to /root, anytime I edited it.

If I pass -ao alsa:device=ch51dup to mplayer, it works with both user and root.

mpd simply refuses to work.
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PostPosted: Thu Dec 15, 2011 7:55 am    Post subject: did you try this? Reply with quote

Um, did you try this already?

1. click volume control/mixer item in task bar
2. click Mixer button on the dropdown
3. Settings --> Audio Settings --> Speaker setup
4. select correct Sound Card and Profile, and test....

This is using Phonon.

Hope that helps, sound is excessively annoying to get to behave properly forever.

At the moment my volume is adjusting itself to max, which is very annoying.
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mohiva
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PostPosted: Fri Mar 09, 2012 10:29 pm    Post subject: Reply with quote

Hi,

I try to connect my 2.1 system(two active studio monitors and one active subwoofer) to a X-FI sound card(Model SB0460). The monitors are connected to Line Out 1 and the subwoofer is connected to Line Out 3. Without configuring anything I get sound from my two monitors. But I get no sound from the subwoofer.

I have no idea how I can start to get the system working. Can anyone point me to the right direction.

An other question is, how can I restart the sound system without rebooting the whole system.

Christian
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PostPosted: Mon Mar 12, 2012 7:15 pm    Post subject: Reply with quote

mohiva wrote:
Hi,

I try to connect my 2.1 system(two active studio monitors and one active subwoofer) to a X-FI sound card(Model SB0460). The monitors are connected to Line Out 1 and the subwoofer is connected to Line Out 3. Without configuring anything I get sound from my two monitors. But I get no sound from the subwoofer.

I have no idea how I can start to get the system working. Can anyone point me to the right direction.



Did you try the four steps above?

cheers, Ian
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mohiva
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PostPosted: Wed Mar 14, 2012 10:13 pm    Post subject: Reply with quote

Hi,

now I have sound with the following asound.conf.

Code:

# Virtual 2.1 device (softvol):
# for volume control
pcm.softvol {
    type            softvol
    slave.pcm       "four"
    control {
        name        "UpMix21"
        card        0
    }
}

# software device with four channels
pcm.four {
        type route
        slave.pcm "multi"
        ttable.0.0 1
        ttable.1.1 1
        ttable.2.2 1
        ttable.3.3 1
}

# customize this, if your xfi hast not the index=0
# Splitting the channels into three different devices:
pcm.multi {
    type            multi
    slaves {
        a.pcm        "hw:0,0"
        a.channels    2

        b.pcm        "hw:0,1"
        b.channels    2
    }
    bindings {
        0.slave        a
        0.channel    0
        1.slave        a
        1.channel    1
        2.slave        b
        2.channel    0
        3.slave        b
        3.channel    1
    }
}

# Low/Highpass for channel speration
# controls[ x ] specifies the crossover frequency x for the subwoofer.
# 80-120 for most common systems
pcm.lowpass {
    type ladspa
    slave.pcm lowpass_to_upmix
    path "/usr/lib64/ladspa"
    channels 3
    plugins {
      0 {
         id 1098  # Identity (Audio) (1098/identity_audio)
         policy duplicate
         input.bindings.0 "Input";
         output.bindings.0 "Output";
      }

      1 {
         id 1052  # High-pass filter

         policy none
         input.bindings.0 "Input";
         output.bindings.0 "Output";
         input {
            controls [ 120 ]
         }
      }

      2 {
         id 1052  # High-pass filter

         policy none
         input.bindings.1 "Input";
         output.bindings.1 "Output";
         input {
            controls [ 120 ]
         }
      }

      3 {
         id 1051  # Low-pass filter

         policy none
         input.bindings.2 "Input";
         output.bindings.2 "Output";
         input {
            controls [ 120 ]
         }
      }
    }
}

# lowpass to upmix
pcm.lowpass_to_upmix {
   type plug
   slave.pcm softvol
   slave.channels 4
   ttable {
      0.0     1       # front left
      1.1     1       # front right

      # Power of the subwoofer, normally 1 (or if you like, more)
      2.2     1
   }
}

# upmix 2.0 to 2.1
pcm.upmix_20to21 {
   type plug
   slave.pcm "lowpass"
   slave.channels 3
   ttable {
        0.0     1       # left channel
        1.1     1       # right channel
        0.2     0.5     # mix left and right ...
        1.2     0.5     # as subwoofer channel
   }
}

pcm.!default {
    type            asym
    playback.pcm    "upmix_20to21"
}



But there exists a problem with alsa, ladspa and pulseaudio. The problem is described here:

https://bugs.launchpad.net/ubuntu/+source/alsa-lib/+bug/514902
http://ubuntuforums.org/showthread.php?t=1250483

Has anyone this configuration running?

Greetings
Christian
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