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Hello,
I am trying to use audacity. Can someone help me? I keep getting this error:
ryanvade@ryan-linuux-desktop:~$ audacity -v
ALSA lib pcm_dsnoop.c:618:(snd_pcm_dsnoop_open) unable to open slave
ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave
ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear
ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe
ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side
ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM bs2b
ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM bs2b
ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM bs2b
ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM bs2b
ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM lowpass_21to21
ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM lowpass_21to21
ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM lowpass_21to21
ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM lowpass_21to21
ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM lowpass_21to21
ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM lowpass_21to21
ALSA lib pcm_dmix.c:961:(snd_pcm_dmix_open) The dmix plugin supports only playback stream
ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave
ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave
ALSA lib pcm_mmap.c:427:(snd_pcm_mmap) malloc failed: Cannot allocate memory
Expression 'ret' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 1809
audacity: pcm_null.c:143: snd_pcm_null_drop: Assertion `null->state != SND_PCM_STATE_OPEN' failed.
Aborted (core dumped)I am wondering if the issue is my ~/.asoundrc..
# dmix - plug:dmix supports 1-8 channels, and does use dmix!
# Whereas surround51 doesn't use dmix
# http://bbs.archlinux.org/viewtopic.php?pid=745946#p745946
# cat /proc/asound/card0/pcm0p/sub0/hw_params
# Output to hw:0,0 to keep at 44.1k rather than dmix's 48k
# 44.1k stops dmix from working, though.
# From https://bugs.launchpad.net/debian/+source/sdl-mixer1.2/+bug/66483
# Not needed.
#defaults.pcm.dmix_max_periods -1
#defaults.pcm.rate_converter "samplerate_best"
# See /usr/share/alsa/pcm/dmix.conf
#defaults.dmix.period_time 0
#defaults.dmix.periods 4
#defaults.pcm.surround51.device "0"
# From https://bugtrack.alsa-project.org/alsa-bug/view.php?id=1853
# Posted at http://bbs.archlinux.org/viewtopic.php?id=95582
# Is a dmix that actually works!
pcm.dmixed {
type asym
playback.pcm {
# See plugin:dmix at http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html
type dmix
ipc_key 5678293
ipc_perm 0660
ipc_gid audio
#rate 48000 # Don't put the rate here! Otherwise it resets the rate & channels set below, as shown by: cat /proc/asound/card0/pcm0p/sub0/hw_params
slave {
channels 6
pcm {
format S16_LE
rate 48000
type hw
card 1
device 0
subdevice 0
}
# Play with this value, if you get errors "unable to set buffer size" or "underrun occured"
# 4320 is effective minimum with hda-intel, but flash in firefox needs at least 5000.
buffer_size 5000
period_time 0
#period_size 512
#periods 2
}
}
capture.pcm {
# Dummy, but present to stop wine from moaning: ALSA lib pcm_asym.c:106:(_snd_pcm_asym_open) capture slave is not defined
type null
}
}
# Playing
#pcm.!default {
# type asym
# playback.pcm "upmix_20to51_resample"
#}
# Check that e.g. Thief2 still works, if default is redefined.
pcm.!default {
type plug
# Always output to all 6 channels, so the dmixer actually works if e.g. 6-channel is attempted to be started, while 2-channel is playing.
slave.pcm "dmixed"
}
pcm.!surround20 {
type plug
slave.pcm "dmixed"
}
pcm.!surround40 {
type plug
slave.pcm "dmixed"
}
pcm.!surround51 {
type plug
slave.pcm "dmixed"
}
# If get error "Slave PCM not usable", then need to use plug:
# If get error "Cannot find rate converter", then install libsamplerate and alsa-plugins
# Lunar Linux: lin ladspa-bs2b
# listplugins
# analyseplugin bs2b
pcm.bs2b {
type ladspa
path "/usr/lib/ladspa"
plugins {
0 {
id 4221 # Bauer stereophonic-to-binaural (4221/bs2b)
input {
controls [ 700 6 ]
}
}
}
# http://bbs.archlinux.org/viewtopic.php?id=95582
slave.pcm "plug:dmixed"
}
# speaker-test -D headphones -c 2 -t wav
# audacious uses less CPU when running bs2b as a listed plugin, probably because of samplerate_best
# Posted at http://bbs.archlinux.org/viewtopic.php?pid=626573#p626573
pcm.headphones {
type rate
slave {
pcm "plug:bs2b"
rate 48000
}
# Choices: samplerate_best samplerate_medium samplerate samplerate_order samplerate_linear
converter "samplerate_best"
hint {
show on
description "Headphones"
}
}
pcm.ch51dup {
slave.pcm "dmixed"
slave.channels 6
type route
# Front and rear
ttable.0.0 0.5
ttable.1.1 0.5
ttable.2.2 0.5
ttable.3.3 0.5
# Center and LFE
ttable.4.4 1
ttable.5.5 1
# Front left/right to center
ttable.0.4 0.5
ttable.1.4 0.5
# Front left/right to rear
ttable.0.2 0.5
ttable.1.3 0.5
}
# http://alsa.opensrc.org/SurroundSound
# http://alsa.opensrc.org/index.php/Low-pass_filter_for_subwoofer_channel_(HOWTO)
# Lunar: lin ladspa tap-plugins swh-plugins cmt-plugins libsamplerate
# Fedora: yum install ladspa ladspa-blop-plugins ladspa-caps-plugins ladspa-cmt-plugins ladspa-swh-plugins ladspa-tap-plugins libsamplerate
# Arch Linux: pacman -S ladspa blop swh-plugins libsamplerate tap-plugins cmt
# For id 1672 - 4 Pole Low-Pass Filter with Resonance (FCRCIA) (1672/lp4pole_fcrcia_oa), install blop-plugins
# speaker-test -D upmix_20to51 -c 2 -t wav
# Debugging: speaker-test -D plug:lowpass_21to21 -c 3 -t wav
# listplugins
# analyseplugin cmt
# http://plugin.org.uk/ladspa-swh/docs/ladspa-swh.html
# http://forums.gentoo.org/viewtopic-p-4528619.html#4528619
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
# policy none
# input.bindings.0 "Input";
# output.bindings.0 "Output";
# input {
# controls [ 300 ]
# }
#}
#2 {
# id 1052 # High-pass filter
# policy none
# input.bindings.1 "Input";
# output.bindings.1 "Output";
# input {
# controls [ 300 ]
# }
#}
#3 {
# id 1051 # Low-pass filter.
# policy none
# input.bindings.2 "Input";
# output.bindings.2 "Output";
# input {
# controls [ 300 ]
# }
#}
# From http://alsa.opensrc.org/index.php/Low-pass_filter_for_subwoofer_channel_(HOWTO)
# Can be used instead of 1-3 above.
1 {
id 1672 # 4 Pole Low-Pass Filter with Resonance (FCRCIA) (1672/lp4pole_fcrcia_oa)
policy none
input.bindings.2 "Input";
output.bindings.2 "Output";
input {
controls [ 300 2 ]
}
}
}
}
# speaker-test -D upmix_20to51 -c 2 -t wav
# In audacious: upmix_20to51
pcm.upmix_20to51 {
type plug
slave.pcm "lowpass_21to21"
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 # ... channel for subwoofer
}
# slave.rate 48000 makes CPU utilization 20% instead of 3%
# Can't hear the difference with Audigy4 anyway.
# slave.rate 44100 is 3%, so that proves audacious outputs 44100
#slave.rate 48000
#converter "samplerate"
#slave.rate_converter "samplerate_best"
}
# In audacious: upmix_20to51_resample
# aplay -D upmix_20to51_resample ~/alsa/samplerate-test/udial.wav
pcm.upmix_20to51_resample {
type rate
slave {
pcm upmix_20to51
#format S32_LE
# Audigy4 upmixes to 48000 itself, and seems to use low-quality linear interpolation
rate 48000
}
# Choices: samplerate_best samplerate_medium samplerate samplerate_order samplerate_linear
# 8% CPU with samplerate_medium - good choice
converter "samplerate_medium"
#converter "samplerate_linear"
hint {
show on
description "20to51"
}
}
# Debugging: speaker-test -D upmix_21to51 -c 3 -t wav
pcm.upmix_21to51 {
type plug
# For ice1724:
#slave.pcm surround51-ice
# For Audigy:
slave.pcm "dmixed"
# http://bbs.archlinux.org/viewtopic.php?pid=745946#p745946
#slave.pcm dmixed:6
# For P5K ADI:
#slave.pcm surround51-adi
# Trying to pipe through Pulse Audio, to stop the clicks between songs.
# Can't get Pulse Audio to work like this.
#slave.pcm pulse
# Don't need to specify the number of channels.
slave.channels 6
ttable {
0.0 1 # front left
1.1 1 # front right
0.2 1 # rear left
1.3 1 # rear right
# Front left/right to center.
# Imbalanced because is to the left of the monitor!
# Would normally be 0.5 each.
0.4 0.5
1.4 0.5
# Subwoofer, more powerful to compensate for bass-removal from other speakers.
2.5 2
}
}
# Channels are wrong way around in doom! This fixes them.
# http://www.linuxforen.de/forums/archive/index.php/t-206470.html
# http://forums.seriouszone.com/showthread.php?t=49869&page=10
# http://forums.gentoo.org/viewtopic-p-4173170.html#4173170
# For Audigy 4
# Weird, doom3 has crappy sound if I add an alsa rate converter.
# Posted at http://ubuntuforums.org/showthread.php?t=1304228
pcm.doom-surround51 {
slave.pcm "dmixed"
slave.channels 6
type route
ttable.0.0 1
ttable.1.1 1
ttable.2.4 1
ttable.3.5 1
ttable.4.2 1
ttable.5.3 1
}
pcm.doom3-8930g {
type plug
slave.pcm {
type dmix
ipc_key 1093 # Must be unique
ipc_key_add_uid false
ipc_perm 0660
slave {
pcm "hw:0,0"
rate 44100
channels 2
period_time 0
period_size 1024
buffer_time 0
# Doom 3 wants buffer_size 8192
# In ~/.doom3/base/autoexec.cfg
# And ~/.quake4/q4base/autoexec.cfg
# seta s_alsa_pcm "doom3-8930g"
buffer_size 8192
}
}
}Offline
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