Sound and Mic Problems on EeePC 1018p

[cspollard]

I have an Asus EeePC 1018p and I'm having trouble debugging an audio problem in Lucid. The internal microphone does not work, nor does the headphone jack or external mic input. I've tried fiddling with alsamixer, pavucontrol (some fixes involve setting turning down the left or right capture channels), etc., but haven't had any success.

It seems that many people with EeePCs (other models) have fixed the problems with what I mentioned above, but it's no use for me. I've also booted into UNR to see if the modified kernel that it ships with fixes any problems, but nothing changed.

Any ideas? I don't even know which logs to look in to find the issue.

Here is some information on my system if it helps.

ALSA Audio Debug v0.1.0 - Tue Jul 13 14:02:40 CEST 2010
http://alsa.opensrc.org/aadebug
http://www.gnu.org/licenses/gpl.txt

Kernel ----------------------------------------------------
Linux ####### 2.6.32-23-generic #37-Ubuntu SMP Fri Jun 11 08:03:28 UTC 2010 x86_64 GNU/Linux

Loaded Modules --------------------------------------------
snd_hda_codec_realtek   279040  1 
snd_hda_intel          25677  6 
snd_hda_codec          85759  2 snd_hda_codec_realtek,snd_hda_intel
snd_hwdep               6924  1 snd_hda_codec
snd_pcm_oss            41394  0 
snd_mixer_oss          16299  1 snd_pcm_oss
snd_pcm                87882  5 snd_hda_intel,snd_hda_codec,snd_pcm_oss
snd_seq_dummy           1782  0 
snd_seq_oss            31219  0 
snd_seq_midi            5829  0 
snd_rawmidi            23420  1 snd_seq_midi
snd_seq_midi_event      7267  2 snd_seq_oss,snd_seq_midi
snd_seq                57481  6 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event
snd_timer              23649  2 snd_pcm,snd_seq
snd_seq_device          6888  5 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq
snd                    71106  22 snd_hda_codec_realtek,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_seq_oss,snd_rawmidi,snd_seq,snd_timer,snd_seq_device
snd_page_alloc          8500  2 snd_hda_intel,snd_pcm

Dev Snd ---------------------------------------------------
by-path  controlC0  hwC0D0  pcmC0D0c  pcmC0D0p	seq  timer

CPU -------------------------------------------------------
model name	: Intel(R) Atom(TM) CPU N450   @ 1.66GHz
cpu MHz		: 1667.000
model name	: Intel(R) Atom(TM) CPU N450   @ 1.66GHz
cpu MHz		: 1667.000

RAM -------------------------------------------------------
MemTotal:        1014136 kB
SwapTotal:       1033208 kB

Hardware --------------------------------------------------
00:00.0 Host bridge: Intel Corporation N10 Family DMI Bridge

[cspollard]

Bump bump...Is this in the right forum?

[cspollard]

OK, well I've submitted this information to the launchpad bug found here:

https://bugs.launchpad.net/ubuntu/+bug/601518

Is there anything else I should do?

[lidex]

asus models seem to require more fiddling than any other. It's probably workable.
Can you post the output of these terminal commands for me please:

uname -a
aplay -l
dpkg -l | grep "alsa"
head -n 1 /proc/asound/card*/codec#*

Terminal="Applications->Accessories->Terminal"
Please also include the make/model of your PC/Laptop.
Please post text output using code tags (the # in toolbar)

[cspollard]

Thanks for the help!

uname -a
------------------
Linux 2.6.32-24-generic #38-Ubuntu SMP Mon Jul 5 09:20:59 UTC 2010 x86_64 GNU/Linux


aplay -l
------------------
**** List of PLAYBACK Hardware Devices ****
card 0: Intel [HDA Intel], device 0: ALC269 Analog [ALC269 Analog]
  Subdevices: 0/1
  Subdevice #0: subdevice #0


dpkg -l | grep "alsa"
------------------
ii  alsa-base                                                       1.0.22.1+dfsg-0ubuntu3                          ALSA driver configuration files
ii  alsa-utils                                                      1.0.22-0ubuntu5                                 ALSA utilities
ii  bluez-alsa                                                      4.60-0ubuntu8                                   Bluetooth audio support
ii  gstreamer0.10-alsa                                              0.10.28-1                                       GStreamer plugin for ALSA


head -n 1 /proc/asound/card*/codec#*
------------------
Codec: Realtek ALC269

[lidex]

Using a Terminal="Applications->Accessories->Terminal"
Open this file for editing:

 gksudo gedit /etc/modprobe.d/alsa-base.conf

Insert this line at the bottom:

 options snd-hda-intel model=auto

Save. Close. Reboot. Check your levels in alsamixer.

 alsamixer

Press <F6> to select the correct soundcard.
Press <F3> to show playback levels. <F4> selects capture levels [or use <Tab>]
Use the left/right arrow keys to select and up/down arrow keys to change levels. <M> to mute/unmute.
Go to "System ->Preferences ->Sound" and make sure the correct soundcard is default and adjust your profile on the hardware tab.

[cspollard]

No change as far as I can tell. Internal speakers still work. Still no output via headphone jack, nor is there any response from the internal microphone.

Any other ideas?

[lidex]

First, for mic problem, check this.
Gnome-alsamixer
Using Alt+F2 run dialog

 gnome-alsamixer

Use 'Edit -> Sound Card Properties' menu to enable elements.
Soundcard tab to select / adjust levels.
To install:

sudo apt-get install gnome-alsamixer

Then use pulseaudio volume control (pavucontrol) or 'Sound Preferences' to select and unmute your mic in 'Input Devices' tab.

From alsa documentation:

Capture Problems
~~~~~~~~~~~~~~~~
The capture problems are often because of missing setups of mixers.
Thus, before submitting a bug report, make sure that you set up the
mixer correctly. For example, both "Capture Volume" and "Capture
Switch" have to be set properly in addition to the right "Capture
Source" or "Input Source" selection. Some devices have "Mic Boost"
volume or switch.

When the PCM device is opened via "default" PCM (without pulse-audio
plugin), you'll likely have "Digital Capture Volume" control as well.
This is provided for the extra gain/attenuation of the signal in
software, especially for the inputs without the hardware volume
control such as digital microphones. Unless really needed, this
should be set to exactly 50%, corresponding to 0dB -- neither extra
gain nor attenuation. When you use "hw" PCM, i.e., a raw access PCM,
this control will have no influence, though.

[cspollard]

hmm. All devices are on in alsamixer, gnome-alsamixer, and pavucontrol. Mic Boost, Capture, and Front Mic are all at 100%, as are Speaker, PCM, and Master.

Something funny though--the input in pavucontrol has a non-zero signal when the output slider level is set to something other than zero--even when output is muted. If the output slider is at zero, there is never any input signal.

Also, pavucontrol seems to think that I have a stereo mic, but it's mono. I've read that setting one slider to ~90 and the other to ~10 will fix the problem, but I haven't had any luck. Every time I restart pavucontrol, the two channels are locked to one another again.

Cheers.

[lidex]

Try removing your pulse settings:

rm -r ~/.pulse ~/.asound* 
sudo rm /etc/asound.conf

Logout/in.
Now try again. The stereo mic is an issue. Turn the right channel all the way down.