Hi,
Thanks for this useful guide
Though I still have some issues:
- I couldn't get mbeq_1197 ; is it only available from Intrepide Ubuntu ?
- I have very bad sound capture with teamspeak when using pulseaudio (was ok with aoss and is fine but unsharable with no wrapper).
I'm on Hardy 64 bits. Using X-Fi Extreme Audio creative card (CA0106)
I tried to managed with this :
default-fragments = 8
default-fragment-size-msec = 5
but no success
I run "padsp teamspeak" as if I don't, I can't share the sound with wine programs (ran with "padsp wine wineprg")
Running Teamspeak without padsp, I have good sound capture from the microphone but cannot share sound with wine programs (even or not via padsp).
If I increase the sound quality in Teamspeak options, I have too big latency. So I set it on low (don't have to do this using aoss and not pulseaudio).
The "bad" sound capture is saccaded, my voice's heard as if I was a robot.
So, I disabled load-module module-alsa-sink device=equalized
Here is my .asoundrc
Code:
#pcm.card0 {
# type hw
# card 0
#}
pcm.dmixer {
type dmix
ipc_key 1025
slave {
pcm "hw:0,0"
period_time 0
#period_size 1024
#buffer_size 8192
period_size 2048
buffer_size 32768
rate 48000
}
# bindings {
# 0 0
# 1 1
# }
}
pcm.dsp0 {
type plug
slave.pcm "dmixer"
}
# This following device can fool some applications into using pulseaudio
pcm.dsp1 {
type plug
slave.pcm "pulse"
}
ctl.mixer0 {
type hw
card 0
}
#pcm.css {
# type asym
# playback.pcm "hw:0"
#}
pcm.pulse {
type pulse
}
ctl.pulse {
type pulse
}
# Optional, set defaults
pcm.!default {
type pulse
}
ctl.!default {
type pulse
}
pcm.equalized {
type plug
slave.pcm "equalizer";
}
pcm.equalizer {
type ladspa
# The output from the EQ can either go direct to a hardware device
# (if you have a hardware mixer, e.g. SBLive/Audigy) or it can go
# to the software mixer shown here.
#slave.pcm "plughw:0,0"
slave.pcm "plug:dmix"
# Sometimes you may need to specify the path to the plugins,
# especially if you've just installed them. Once you've logged
# out/restarted this shouldn't be necessary, but if you get errors
# about being unable to find plugins, try uncommenting this.
path "/usr/lib/ladspa:/usr/lib64/ladspa:/usr/lib32/ladspa"
plugins [
{
label mbeq
id 1197
input {
#this setting is here by example, edit to your own taste
#bands: 50hz, 100hz, 156hz, 220hz, 311hz, 440hz, 622hz, 880hz,
# 1250hz, 1750hz, 25000hz, 50000hz, 10000hz, 20000hz
#range: -70 to 30
controls [ -1 -1 -1 -1 -5 -10 -20 -17 -12 -7 -6 -5 -5 0 0 ]
}
}
]
}
pulseaudio starting output :
Code:
I: main.c: We're in the group 'pulse-rt', allowing real-time and high-priority scheduling.
I: core-util.c: Successfully gained nice level -11.
I: main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Opération non permise
I: main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Opération non permise
I: main.c: This is PulseAudio 0.9.10
I: main.c: Page size is 4096 bytes
I: main.c: Fresh high-resolution timers available! Bon appetit!
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9/modules/module-hal-detect.so': success
I: module-hal-detect.c: Trying capability alsa
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/computer_alsa_timer
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/computer_alsa_sequencer
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_7_sound_card_0_alsa_playback_3
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_7_sound_card_0_alsa_capture_3
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_7_sound_card_0_alsa_playback_2
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_7_sound_card_0_alsa_capture_2
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_7_sound_card_0_alsa_playback_1
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_7_sound_card_0_alsa_capture_1
D: module-hal-detect.c: Loading module-alsa-sink with arguments 'device_id=0 sink_name=alsa_output.pci_1102_7_sound_card_0_alsa_playback_0'
D: alsa-util.c: Trying front:0...
W: alsa-util.c: Device front:0 doesn't support 44100 Hz, changed to 48000 Hz.
I: module-alsa-sink.c: Successfully opened device front:0.
I: module-alsa-sink.c: Successfully enabled mmap() mode.
ALSA lib control.c:909:(snd_ctl_open_noupdate) Invalid CTL front:0
I: alsa-util.c: Unable to attach to mixer front:0: Aucun fichier ou dossier de ce type
I: alsa-util.c: Successfully attached to mixer 'hw:0'
I: alsa-util.c: Cannot find mixer control "Master".
W: alsa-util.c: Cannot find fallback mixer control "PCM".
I: sink.c: Created sink 0 "alsa_output.pci_1102_7_sound_card_0_alsa_playback_0" with sample spec "s16le 2ch 48000Hz"
I: source.c: Created source 0 "alsa_output.pci_1102_7_sound_card_0_alsa_playback_0.monitor" with sample spec "s16le 2ch 48000Hz"
I: module-alsa-sink.c: Using 4 fragments of size 4416 bytes.
D: module-alsa-sink.c: Thread starting up
D: rtpoll.c: Acquired POSIX realtime signal SIGRTMIN+29
I: module-alsa-sink.c: Starting playback.
I: module.c: Loaded "module-alsa-sink" (index: #0; argument: "device_id=0 sink_name=alsa_output.pci_1102_7_sound_card_0_alsa_playback_0").
D: module-hal-detect.c: Loading module-alsa-source with arguments 'device_id=0 source_name=alsa_input.pci_1102_7_sound_card_0_alsa_capture_0'
D: alsa-util.c: Trying front:0...
I: module-alsa-source.c: Successfully opened device front:0.
I: module-alsa-source.c: Successfully enabled mmap() mode.
ALSA lib control.c:909:(snd_ctl_open_noupdate) Invalid CTL front:0
I: alsa-util.c: Unable to attach to mixer front:0: Aucun fichier ou dossier de ce type
I: alsa-util.c: Successfully attached to mixer 'hw:0'
I: alsa-util.c: Cannot find mixer control "Capture".
I: alsa-util.c: Using mixer control "Mic".
I: source.c: Created source 1 "alsa_input.pci_1102_7_sound_card_0_alsa_capture_0" with sample spec "s16le 2ch 44100Hz"
I: module-alsa-source.c: Using 2 fragments of size 4416 bytes.
I: alsa-util.c: All 2 channels can be mapped to mixer channels. Using hardware volume control.
D: module-alsa-source.c: Thread starting up
D: rtpoll.c: Acquired POSIX realtime signal SIGRTMIN+28
I: module.c: Loaded "module-alsa-source" (index: #1; argument: "device_id=0 source_name=alsa_input.pci_1102_7_sound_card_0_alsa_capture_0").
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_7_sound_card_0_alsa_midi_0
D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_7_sound_card_0_alsa_control__1
I: module-hal-detect.c: Loaded 2 modules.
I: module.c: Loaded "module-hal-detect" (index: #2; argument: "").
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9/modules/module-esound-protocol-unix.so': success
I: module.c: Loaded "module-esound-protocol-unix" (index: #3; argument: "").
I: protocol-native.c: loading cookie from disk.
I: module.c: Loaded "module-native-protocol-unix" (index: #4; argument: "").
I: module.c: Loaded "module-volume-restore" (index: #5; argument: "").
D: module-default-device-restore.c: Restored default sink 'alsa_output.pci_1102_7_sound_card_0_alsa_playback_0'.
D: core-subscribe.c: dropped redundant event.
D: module-default-device-restore.c: Restored default source 'alsa_input.pci_1102_7_sound_card_0_alsa_capture_0'.
I: module.c: Loaded "module-default-device-restore" (index: #6; argument: "").
I: module.c: Loaded "module-rescue-streams" (index: #7; argument: "").
D: module-suspend-on-idle.c: Sink alsa_output.pci_1102_7_sound_card_0_alsa_playback_0 becomes idle.
D: module-suspend-on-idle.c: Source alsa_output.pci_1102_7_sound_card_0_alsa_playback_0.monitor becomes idle.
D: module-suspend-on-idle.c: Source alsa_input.pci_1102_7_sound_card_0_alsa_capture_0 becomes idle.
I: module.c: Loaded "module-suspend-on-idle" (index: #8; argument: "").
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9/modules/module-gconf.so': success
D: module-gconf.c: Loading module 'module-native-protocol-tcp' with args '' due to GConf configuration.
I: protocol-native.c: using already loaded auth cookie.
I: protocol-native.c: using already loaded auth cookie.
I: module.c: Loaded "module-native-protocol-tcp" (index: #9; argument: "").
D: module-gconf.c: Loading module 'module-esound-protocol-tcp' with args '' due to GConf configuration.
I: module.c: Loaded "module-esound-protocol-tcp" (index: #10; argument: "").
D: module-gconf.c: Loading module 'module-zeroconf-discover' with args '' due to GConf configuration.
I: module.c: Loaded "module-zeroconf-discover" (index: #11; argument: "").
I: module.c: Loaded "module-gconf" (index: #12; argument: "").
D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9/modules/module-x11-publish.so': success
D: module-x11-publish.c: using already loaded auth cookie.
I: module.c: Loaded "module-x11-publish" (index: #13; argument: "").
I: main.c: Daemon startup complete.
D: module-hal-detect.c: dbus: interface=org.freedesktop.DBus, path=/org/freedesktop/DBus, member=NameAcquired
I: module-suspend-on-idle.c: Source alsa_input.pci_1102_7_sound_card_0_alsa_capture_0 idle for too long, suspending ...
I: module-alsa-source.c: Device suspended...
I: module-suspend-on-idle.c: Source alsa_output.pci_1102_7_sound_card_0_alsa_playback_0.monitor idle for too long, suspending ...
I: module-suspend-on-idle.c: Sink alsa_output.pci_1102_7_sound_card_0_alsa_playback_0 idle for too long, suspending ...
I: module-alsa-sink.c: Device suspended...
Running teamspeak (see memory message at the end)
with high latency and default fragments options in pulseaudio daemon.conf
Code:
I: client.c: Created 0 "Native client (UNIX socket client)"
I: protocol-native.c: Got credentials: uid=1000 gid=1000 success=1
I: protocol-native.c: Enabled SHM for new connection
I: client.c: Client 0 changed name from "Native client (UNIX socket client)" to "OSS Emulation[teamspeak.real]"
I: module-volume-restore.c: Restoring source for <pulsecore/protocol-native.c$OSS Emulation[teamspeak.real]>
I: module-alsa-source.c: Trying resume...
I: module-alsa-source.c: Resumed successfully...
D: module-suspend-on-idle.c: Source alsa_input.pci_1102_7_sound_card_0_alsa_capture_0 becomes idle.
D: module-suspend-on-idle.c: Source alsa_input.pci_1102_7_sound_card_0_alsa_capture_0 becomes busy.
D: resampler.c: Channel matrix:
D: resampler.c: I00 I01
D: resampler.c: +------------
D: resampler.c: O00 | 1,000 1,000
I: resampler.c: Using resampler 'speex-float-3'
I: resampler.c: Using float32le as working format.
I: resampler.c: Choosing speex quality setting 3.
I: source-output.c: Created output 0 "Audio Stream" on alsa_input.pci_1102_7_sound_card_0_alsa_capture_0 with sample spec u8 1ch 8000Hz and channel map mono
D: memblockq.c: memblockq requested: maxlength=13312, tlength=0, base=1, prebuf=1, minreq=0
D: memblockq.c: memblockq sanitized: maxlength=13312, tlength=13312, base=1, prebuf=1, minreq=1
D: module-suspend-on-idle.c: Source alsa_input.pci_1102_7_sound_card_0_alsa_capture_0 becomes idle.
D: module-suspend-on-idle.c: Source alsa_input.pci_1102_7_sound_card_0_alsa_capture_0 becomes idle.
I: source-output.c: Freeing output 0 "Audio Stream"
I: module-volume-restore.c: Restoring source for <pulsecore/protocol-native.c$OSS Emulation[teamspeak.real]>
D: module-suspend-on-idle.c: Source alsa_input.pci_1102_7_sound_card_0_alsa_capture_0 becomes busy.
D: resampler.c: Channel matrix:
D: resampler.c: I00 I01
D: resampler.c: +------------
D: resampler.c: O00 | 1,000 1,000
I: resampler.c: Using resampler 'speex-float-3'
I: resampler.c: Using float32le as working format.
I: resampler.c: Choosing speex quality setting 3.
I: source-output.c: Created output 1 "Audio Stream" on alsa_input.pci_1102_7_sound_card_0_alsa_capture_0 with sample spec u8 1ch 8000Hz and channel map mono
D: memblockq.c: memblockq requested: maxlength=13312, tlength=0, base=1, prebuf=1, minreq=0
D: memblockq.c: memblockq sanitized: maxlength=13312, tlength=13312, base=1, prebuf=1, minreq=1
D: module-suspend-on-idle.c: Source alsa_input.pci_1102_7_sound_card_0_alsa_capture_0 becomes idle.
D: module-suspend-on-idle.c: Source alsa_input.pci_1102_7_sound_card_0_alsa_capture_0 becomes idle.
I: source-output.c: Freeing output 1 "Audio Stream"
I: module-volume-restore.c: Restoring source for <pulsecore/protocol-native.c$OSS Emulation[teamspeak.real]>
D: module-suspend-on-idle.c: Source alsa_input.pci_1102_7_sound_card_0_alsa_capture_0 becomes busy.
D: resampler.c: Channel matrix:
D: resampler.c: I00 I01
D: resampler.c: +------------
D: resampler.c: O00 | 1,000 1,000
I: resampler.c: Using resampler 'speex-float-3'
I: resampler.c: Using float32le as working format.
I: resampler.c: Choosing speex quality setting 3.
I: source-output.c: Created output 2 "Audio Stream" on alsa_input.pci_1102_7_sound_card_0_alsa_capture_0 with sample spec u8 1ch 8000Hz and channel map mono
D: memblockq.c: memblockq requested: maxlength=16384, tlength=0, base=1, prebuf=1, minreq=0
D: memblockq.c: memblockq sanitized: maxlength=16384, tlength=16384, base=1, prebuf=1, minreq=1
D: module-suspend-on-idle.c: Source alsa_input.pci_1102_7_sound_card_0_alsa_capture_0 becomes idle.
D: module-suspend-on-idle.c: Source alsa_input.pci_1102_7_sound_card_0_alsa_capture_0 becomes idle.
I: source-output.c: Freeing output 2 "Audio Stream"
I: module-volume-restore.c: Restoring source for <pulsecore/protocol-native.c$OSS Emulation[teamspeak.real]>
D: module-suspend-on-idle.c: Source alsa_input.pci_1102_7_sound_card_0_alsa_capture_0 becomes busy.
D: resampler.c: Channel matrix:
D: resampler.c: I00 I01
D: resampler.c: +------------
D: resampler.c: O00 | 1,000 1,000
I: resampler.c: Using resampler 'speex-float-3'
I: resampler.c: Using float32le as working format.
I: resampler.c: Choosing speex quality setting 3.
I: source-output.c: Created output 3 "Audio Stream" on alsa_input.pci_1102_7_sound_card_0_alsa_capture_0 with sample spec s16le 1ch 8000Hz and channel map mono
D: memblockq.c: memblockq requested: maxlength=16384, tlength=0, base=2, prebuf=1, minreq=0
D: memblockq.c: memblockq sanitized: maxlength=16384, tlength=16384, base=2, prebuf=2, minreq=2
D: module-suspend-on-idle.c: Source alsa_input.pci_1102_7_sound_card_0_alsa_capture_0 becomes idle.
D: module-suspend-on-idle.c: Source alsa_input.pci_1102_7_sound_card_0_alsa_capture_0 becomes idle.
I: source-output.c: Freeing output 3 "Audio Stream"
I: module-volume-restore.c: Restoring source for <pulsecore/protocol-native.c$OSS Emulation[teamspeak.real]>
D: module-suspend-on-idle.c: Source alsa_input.pci_1102_7_sound_card_0_alsa_capture_0 becomes busy.
D: resampler.c: Channel matrix:
D: resampler.c: I00 I01
D: resampler.c: +------------
D: resampler.c: O00 | 1,000 1,000
I: resampler.c: Using resampler 'speex-float-3'
I: resampler.c: Using float32le as working format.
I: resampler.c: Choosing speex quality setting 3.
I: source-output.c: Created output 4 "Audio Stream" on alsa_input.pci_1102_7_sound_card_0_alsa_capture_0 with sample spec s16le 1ch 8000Hz and channel map mono
D: memblockq.c: memblockq requested: maxlength=16384, tlength=0, base=2, prebuf=1, minreq=0
D: memblockq.c: memblockq sanitized: maxlength=16384, tlength=16384, base=2, prebuf=2, minreq=2
D: module-suspend-on-idle.c: Source alsa_input.pci_1102_7_sound_card_0_alsa_capture_0 becomes idle.
D: module-suspend-on-idle.c: Source alsa_input.pci_1102_7_sound_card_0_alsa_capture_0 becomes idle.
I: source-output.c: Freeing output 4 "Audio Stream"
I: module-volume-restore.c: Restoring source for <pulsecore/protocol-native.c$OSS Emulation[teamspeak.real]>
D: module-suspend-on-idle.c: Source alsa_input.pci_1102_7_sound_card_0_alsa_capture_0 becomes busy.
D: resampler.c: Channel matrix:
D: resampler.c: I00 I01
D: resampler.c: +------------
D: resampler.c: O00 | 1,000 1,000
I: resampler.c: Using resampler 'speex-float-3'
I: resampler.c: Using float32le as working format.
I: resampler.c: Choosing speex quality setting 3.
I: source-output.c: Created output 5 "Audio Stream" on alsa_input.pci_1102_7_sound_card_0_alsa_capture_0 with sample spec s16le 1ch 22050Hz and channel map mono
D: memblockq.c: memblockq requested: maxlength=16384, tlength=0, base=2, prebuf=1, minreq=0
D: memblockq.c: memblockq sanitized: maxlength=16384, tlength=16384, base=2, prebuf=2, minreq=2
I: module-volume-restore.c: Restoring sink for <pulsecore/protocol-native.c$OSS Emulation[teamspeak.real]>
I: module-volume-restore.c: Restoring volume for <pulsecore/protocol-native.c$OSS Emulation[teamspeak.real]>
I: module-alsa-sink.c: Trying resume...
I: module-alsa-sink.c: Resumed successfully...
I: module-alsa-sink.c: Starting playback.
D: module-suspend-on-idle.c: Sink alsa_output.pci_1102_7_sound_card_0_alsa_playback_0 becomes idle.
D: module-suspend-on-idle.c: Sink alsa_output.pci_1102_7_sound_card_0_alsa_playback_0 becomes busy.
D: resampler.c: Channel matrix:
D: resampler.c: I00
D: resampler.c: +------
D: resampler.c: O00 | 1,000
D: resampler.c: O01 | 1,000
I: resampler.c: Using resampler 'speex-float-3'
I: resampler.c: Using float32le as working format.
I: resampler.c: Choosing speex quality setting 3.
I: sink-input.c: Created input 0 "Audio Stream" on alsa_output.pci_1102_7_sound_card_0_alsa_playback_0 with sample spec s16le 1ch 22050Hz and channel map mono
D: memblockq.c: memblockq requested: maxlength=16384, tlength=12288, base=2, prebuf=4096, minreq=4096
D: memblockq.c: memblockq sanitized: maxlength=16384, tlength=12288, base=2, prebuf=4096, minreq=4096
D: memblock.c: Memory block too large for pool: 18344 > 16368
D: memblock.c: Memory block too large for pool: 16896 > 16368
D: memblock.c: Memory block too large for pool: 16960 > 16368
D: memblock.c: Memory block too large for pool: 17024 > 16368
D: memblock.c: Memory block too large for pool: 17088 > 16368
D: memblock.c: Memory block too large for pool: 17536 > 16368
I also got unlimited:
Code:
D: memblock.c: Pool full
How to configure this in order not to have memory filled ? How to improve my sound capture quality ?
Setting the output to max makes the sound crashing. But setting it at the middle and increasing the output on the speakers, sound is fine.
Input has to be set to max to be quite audible. Don't know if that helps.
What can I do ?
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