HOWTO: PulseAudio Fixes & System-Wide Equalizer Support

[dansan]

Thanks for the help getting me started.

...

Open the PulseAudio Device Chooser (from Applications/Sound & Video), then click on the applet in the notification tray and choose "Manager..."

Look on the Server Information tab and see what is defined for the "Default Sink" (probably the CS-60).

As you see from the screenshot, two sinks are shown, both USB devices. The Server tab shows the top one as Default. My speakers plug into the MB, so they aren't a USB device.

Try this: play some content in e.g. Totem, and launch the PulseAudio Volume Control applet (again from the PulseAudio Device Chooser). On the Playback tab, you should see an entry for Totem. Right click on the Totem entry, select "Move stream" and try to change the stream from the CS-60 to your desired output device.

Yes, this all checks, except there was only one choice.

ALSA PCM on front: 2 (USB Audio) via DCM (the Default) or
ALSA PCM on front: 0 (USB Audio) via DCM

Switching the stream to the latter did not get sound to the speakers.

I hope the screenshot gives you useful information.

[psyke83]

Thanks for the idea. I am a noob so it might take me a day to be able to test this out. Do the connections have to be all at once to do this or is it a max number over time thing? I had a hard time understanding the bug report page. When i first start the machine an check the "Clients" tab in the PA Manager i have 8 or 9 EsoundD clients attached. After it crashes only 1 shows up after i restart the service. PA will still crash after that anyways though

The connection limit may be reached due to bugs in ALSA, so you don't need to do anything more than update the offending ALSA components.


Assuming you're using 32bit Ubuntu, download and install the Intrepid packages, i.e.:

$ wget -c http://archive.ubuntu.com/ubuntu/pool/main/a/alsa-lib/libasound2_1.0.16-2ubuntu1_i386.deb http://archive.ubuntu.com/ubuntu/pool/main/a/alsa-plugins/libasound2-plugins_1.0.16-1ubuntu1_i386.deb http://archive.ubuntu.com/ubuntu/pool/main/a/alsa-lib/libasound2-dev_1.0.16-2ubuntu1_i386.deb
$ sudo dpkg -i libasound2_1.0.16-2ubuntu1_i386.deb libasound2-plugins_1.0.16-1ubuntu1_i386.deb libasound2-dev_1.0.16-2ubuntu1_i386.deb

Please be careful with these commands and do not allow packages to be uninstalled. If in doubt, post a log here.

[psyke83]

As you see from the screenshot, two sinks are shown, both USB devices. The Server tab shows the top one as Default. My speakers plug into the MB, so they aren't a USB device.

Ok, so judging from that screenshot it appears that your built-in sound card has "disappeared", right? Please post the verbose log of PulseAudio (see Appendix B) and also the listing of /proc/asound/pcm (and any other sound logs you can think may be helpful). This may be an ALSA bug.

[manlypain]

The connection limit may be reached due to bugs in ALSA, so you don't need to do anything more than update the offending ALSA components.


Assuming you're using 32bit Ubuntu, download and install the Intrepid packages, i.e.:

$ wget -c http://archive.ubuntu.com/ubuntu/pool/main/a/alsa-lib/libasound2_1.0.16-2ubuntu1_i386.deb http://archive.ubuntu.com/ubuntu/pool/main/a/alsa-plugins/libasound2-plugins_1.0.16-1ubuntu1_i386.deb http://archive.ubuntu.com/ubuntu/pool/main/a/alsa-lib/libasound2-dev_1.0.16-2ubuntu1_i386.deb
$ sudo dpkg -i libasound2_1.0.16-2ubuntu1_i386.deb libasound2-plugins_1.0.16-1ubuntu1_i386.deb libasound2-dev_1.0.16-2ubuntu1_i386.deb

Please be careful with these commands and do not allow packages to be uninstalled. If in doubt, post a log here.

I am using 64bit. i already found the packages. I downloaded them individually. I also downloaded the ones i currently have installed to i can force them back if it dies. Thanks for all the help. I will try this tonight.

[psyke83]

I am using 64bit. i already found the packages. I downloaded them individually. I also downloaded the ones i currently have installed to i can force them back if it dies. Thanks for all the help. I will try this tonight.

Alright. Just remember that you may need the package lib32asound2 if you're on 64bit. Make sure you're not using the equalizer in this guide (to help isolate your problem).

Also, you can revert packages using apt-get this way:

$ sudo apt-get install libasound2/hardy libasound2-plugins/hardy lib32asound2/hardy libasound2-dev/hardy

[Toshibawarrior]

Ok, this fixes have worked great for me until today. Audacity doesn't seem to work with any audio configuration...it worked before following this tutorial.

So if anyone can help me, I'll appreciate it! ...

Apparently Audacity is not friendly with Pulseaudio...I've tried almost every driver on audacity and none of them work...

Please help!

[Toshibawarrior]

Ok, apparently Hydrogen doesn't work either! I didn't try these programs after doing the fixes, and now I know they don't work...Hydrogen sounds trembly and choppy, and I don't know what else to do to configure it with PulseAudio......

Someone please help!

[manlypain]

Alright. Just remember that you may need the package lib32asound2 if you're on 64bit. Make sure you're not using the equalizer in this guide (to help isolate your problem).

Also, you can revert packages using apt-get this way:

$ sudo apt-get install libasound2/hardy libasound2-plugins/hardy lib32asound2/hardy libasound2-dev/hardy

I went to install libasound2 using the deb file and it went ok then errored out after the install and game me this "Processing triggers for libc6 failed". Your uninstall command fixed everything. Can you give me the command to install the 64bit packages like you did for the 32. i will try to run it that way and see if i have better luck.

[Julianito]

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

#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 :

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

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:

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 ?

[ethos_dacapo]

My little laptop speakers are still being shredded by the lack of eq lol
After i set everything up i didn't notice any difference so i tried adjusting the eq killed pulse and restarted this is what i get:

$ pulseaudio
W: main.c: RLIMIT_RTPRIO failed: Operation not permitted
ALSA lib pcm_ladspa.c:1275:(snd_pcm_ladspa_parse_controls) Control port 2 has not an float or integer value
ALSA lib control.c:909:(snd_ctl_open_noupdate) Invalid CTL equalized
ALSA lib conf.c:3952:(snd_config_expand) Unknown parameters 0
ALSA lib pcm.c:2145:(snd_pcm_open_noupdate) Unknown PCM surround71:0
E: alsa-util.c: Error opening PCM device hw:0: Device or resource busy
E: module.c: Failed to load  module "module-alsa-sink" (argument: "device_id=0 sink_name=alsa_output.pci_8086_2485_sound_card_0_alsa_playback_0"): initialization failed.
ALSA lib control.c:909:(snd_ctl_open_noupdate) Invalid CTL front:0
ALSA lib pcm_ladspa.c:1275:(snd_pcm_ladspa_parse_controls) Control port 2 has not an float or integer value
ALSA lib pcm_ladspa.c:1275:(snd_pcm_ladspa_parse_controls) Control port 2 has not an float or integer value
ALSA lib pcm_ladspa.c:1275:(snd_pcm_ladspa_parse_controls) Control port 2 has not an float or integer value

now what?

helps if you type this in the right box: btw i just noticed flash is no longer working!!! What went wrong??

edit: oops! i already had flash installed and when i added the nonfree from synaptic it confused the heck out of firefox. Doh!

Still wondering whats up with these error messages from pulse....