[Alsa-user] alsa-lib so files?

[Radivoje Jovanovic <radivojejovanovic@gm...>]

Hi,
I am trying to cross-compile alsa-lib. As a result of building alsa-lib I
get libasound.so file. However, if I try using aplay or anything that uses
alsa I get:

ALSA lib conf.c:2700:(snd_config_hooks_call) Cannot open shared library
libasound_module_conf_pulse.so
ALSA lib control.c:909:(snd_ctl_open_noupdate) Invalid CTL hw:0

I am not sure how to build libasound_module_conf_pulse.so. I see that the
code for this library is in the alsa-lib folder.

Thnak you
Ogi

[Alsa-user] How to cross-compile alsa-lib?

[Radivoje Jovanovic <radivojejovanovic@gm...>]

Hi,
I am trying to cross compile alsa-lib using the tool chain from my chip
vendor. This has  been done before (not sure by whom) so I know that tool
chain works. However, when I run ./configure script:
CC=MY_TOOL_CHAIN ./configure --target=arm-linux --host=arm --enable-shared
-disable-static

as output I get that:

configure:8812: checking whether the MY_TOOL_CHAIN linker supports shared
libraries
configure:9803: result: yes
...

configure:11392: checking if libtool supports shared libraries
configure:11394: result: no
configure:11397: checking whether to build shared libraries
configure:11418: result: no.

I am not sure why alsa needs special libtool and not use the libtool
installed on the system?

Thank you
Ogi

Re: [Alsa-user] US-122L only MIDI?

[Kilian Sprotte <kilian.sprotte@gm...>]

> I am giving up with the Tascam US-122L, hopefully I will have more
> success with the edirol UA-25 EX.

Hi,

I just wanted to report that I am quite happy now with the UA-25 EX.
Plug and play... :)

There is just one thing about the .asoundrc that I did not succeed to
find out and for which I would like to ask for some help.

This works for me:

speaker-test -c2 -r44100 -D plughw:UA25EX

In my case plughw:UA25EX = plughw:1,0.

What do I have to put into .asoundrc to make this my default device?

I tried

pcm.!default {
	type hw
	card 1
}

ctl.!default {
	type hw
	card 1
}

which seems to be not quite right, because I would like to use the
plughw interface.

speaker-test -c2 -r44100 -D default

fails with this, because of wrong Sample format.

Thanks in advance,
  Kilian

[Alsa-user] Basic PCM Recording

[Paul Braman <bramankp@gm...>]

It seems like I must be missing something when it comes to recording
audio from a device. What I expect to happen is for me to set some
simple parameters (much like I do using OSS) and then have the API
tell me what the best buffer size is for the device so that I can read
reliably.  For example ...

#include <alsa/asoundlib.h>
#include <assert.h>
#include <stdint.h>
#include <stdio.h>
int main(int argc, char *argv[])
{
    unsigned int channels = 2;
    snd_pcm_t *pcm;
    assert(snd_pcm_open(&pcm, "hw:0,0", SND_PCM_STREAM_CAPTURE, 0) == 0);
    snd_pcm_hw_params_t *hwp;
    assert(snd_pcm_hw_params_malloc(&hwp) == 0);
    assert(snd_pcm_hw_params_any(pcm, hwp) == 0);
    assert(snd_pcm_hw_params_set_access(pcm, hwp,
SND_PCM_ACCESS_RW_INTERLEAVED) == 0);
    assert(snd_pcm_hw_params_set_format(pcm, hwp, SND_PCM_FORMAT_S16) == 0);
    assert(snd_pcm_hw_params_set_channels(pcm, hwp, channels) == 0);
    assert(snd_pcm_hw_params_set_rate(pcm, hwp, 32000, 0) == 0);
    assert(snd_pcm_hw_params(pcm, hwp) == 0);
    snd_pcm_uframes_t frames;
    assert(snd_pcm_hw_params_get_buffer_size(hwp, &frames) == 0);
    fprintf(stderr, "target=%lu\n", (unsigned long)frames);
    snd_pcm_hw_params_free(hwp);
    int16_t buffer[frames*channels];
    snd_pcm_sframes_t actual;
again:
    while ((actual = snd_pcm_readi(pcm, buffer, frames)) ==
(snd_pcm_sframes_t){ frames })
    {
        assert(fwrite(buffer, channels*sizeof(int16_t), actual,
stdout) == (size_t){ actual });
    }
    fprintf(stderr, "actual=%ld\n", (long)actual);
    if (actual > 0)
    {
        goto again;
    }
    snd_pcm_close(pcm);
    return 0;
}

I expect this code to run, without interruption, forever dumping audio
to standard output. (Assuming no hardware faults, etc.) However, after
a few seconds it fails when snd_pcm_mmap_read() returns a short read
and then an error of
-32 (-EPIPE).

Are there some calls I'm not making in this startup sequence that are
important? (Yes, I know, I'm not dealing with the mixer here ... one
small example at a time.) Why shouldn't this work? I've seen code
around that sets buffer times and periods and all that crap and if
that is what I must do, what are the basic settings that will apply to
any hardware I'm working with? (Onboard audio card, PCI tuner card,
USB sound card, etc.)

Once I figure this out I need to move on to S/PDIF recording and there
are some real challenges I see there.


Paul Braman

Re: [Alsa-user] Nvidia ION HDMI soundcard

[Robert Parker <theparkers.mailbox@gm...>]

>
> I don't have any .asoundrc on my $HOME directory. What would be a
> correct asoundrc file for me? Can it be done to system level instead
> user level?
>

Yes I think so, have a look at http://alsa.opensrc.org/index.php/.asoundrc

I know the hdmi is working for me when "speaker-test -D default" plays
a hissing noise

Re: [Alsa-user] How to investigate the problem with MIC

[Mark Goldstein <goldstein.mark@gm...>]

On Wed, Sep 22, 2010 at 5:26 PM, Bill Unruh <unruh@...> wrote:
> On Wed, 22 Sep 2010, Mark Goldstein wrote:
>
>> On Wed, Sep 22, 2010 at 9:23 AM, Mark Goldstein
>> <goldstein.mark@...> wrote:
>>>
>>> On Wed, Sep 22, 2010 at 8:54 AM, Axel Braun <axel.braun@...> wrote:
>>>>
>>>> Mark Goldstein wrote:
>>>>
>> was able to record the audio from Mic using audacity! After some more
>> playing I found out that when I select "mute" for Capture1" it is
>> actually unmuted and vise-versa. somehow Capture1 control is broken.
>> Still I have very distorted Mic audio in Skype (and it also worked
>> fine before). But it is a bit of progress!
>
> skype by default has automatic volume control So it may be that there is a
> very weak signal coming in and skype is amplifying the hell our of it,
> giving
> a lot of noise (and distortion).
> Make sure that the automatic volume control in skype is switched OFF. It is
> pretty useless at the best of times.
>

If you mean this "allow Skype to automatically adjust my mixer levels"
check-box, then yes, I know about it. I always disable it.

Regards,
-- 
Mark Goldstein

[Alsa-user] Crash with LADSPA + problem reporting bug

[Gennady Kupava <gb@bs...>]

Hi, list,

I am trying to setup LADSPA here with following config:

pcm.pulseplug {
    type plug
    slave {
        pcm {
            type pulse
            server 192.168.0.1
        }
    }
}

pcm.ladspaplug {
    type ladspa
    slave.pcm pulseplug
    path "/usr/lib/ladspa"
        plugins {
            0 {
                label delay_n
                policy duplicate
                input {
                    controls [ 1 1 ]
                }
            }
        }
}

pcm.!default {
    type plug
    slave.pcm ladspaplug
}

But libasound2 (1.0.23-1, as in debian/amd64/testing) crashing somewhere
in ladspa plugin lib (according to core). plugin itself works fine in
audacity.

So, i tried to file bug report at
https://bugtrack.alsa-project.org/alsa-bug/bug_report_page.php

but it has no category 'PCM - ladspa' or something like 'other'.

Also, i can't find way to report this problem with BTS.

So, two things:
1. Is my config OK?
2. Can someone please add categories for poor reporters. And something
like 'other' may be also useful?

thanks, Gennady.

Re: [Alsa-user] How to investigate the problem with MIC

[Bill Unruh <unruh@ph...>]

On Wed, 22 Sep 2010, Mark Goldstein wrote:

> On Wed, Sep 22, 2010 at 9:23 AM, Mark Goldstein
> <goldstein.mark@...> wrote:
>> On Wed, Sep 22, 2010 at 8:54 AM, Axel Braun <axel.braun@...> wrote:
>>> Mark Goldstein wrote:
>>>
> was able to record the audio from Mic using audacity! After some more
> playing I found out that when I select "mute" for Capture1" it is
> actually unmuted and vise-versa. somehow Capture1 control is broken.
> Still I have very distorted Mic audio in Skype (and it also worked
> fine before). But it is a bit of progress!

skype by default has automatic volume control So it may be that there is a
very weak signal coming in and skype is amplifying the hell our of it, giving
a lot of noise (and distortion).
Make sure that the automatic volume control in skype is switched OFF. It is
pretty useless at the best of times.

>
>
> Regards
>

-- 
William G. Unruh   |  Canadian Institute for|     Tel: +1(604)822-3273
Physics&Astronomy  |     Advanced Research  |     Fax: +1(604)822-5324
UBC, Vancouver,BC  |   Program in Cosmology |     unruh@...
Canada V6T 1Z1     |      and Gravity       |  http://www.theory.physics.ubc.ca/

Re: [Alsa-user] How to investigate the problem with MIC

[Mark Goldstein <goldstein.mark@gm...>]

On Wed, Sep 22, 2010 at 9:23 AM, Mark Goldstein
<goldstein.mark@...> wrote:
> On Wed, Sep 22, 2010 at 8:54 AM, Axel Braun <axel.braun@...> wrote:
>> Mark Goldstein wrote:
>>
>>> I started thinking of buying some normal sound card :-(...
>>
>> Yep :-)
>>
>>> What holds me still is the fact that it did work perfectly till the
>>> end of August.
>>> I've got some advices on alsa bugzilla, but I still am too
>>> "illiterate" to understand their slang :-). Started reading HDA spec
>>> to at least understand the terminology.
>>
>> can you post the link to alsa-bugzilla?
>>
>
> 11.3:
> https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5130
> (I only added comments here and the reporter issue was that he had
> Smart5.1 unmuted)
>
> 11.1:
> https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5133
>

Interesting! I started playing with hda-analyzer.py and came to
conclusion that all connections are OK and the problem might be with
the volume of "Capture1" control. I set it high in hda-analyzer and
was able to record the audio from Mic using audacity! After some more
playing I found out that when I select "mute" for Capture1" it is
actually unmuted and vise-versa. somehow Capture1 control is broken.
Still I have very distorted Mic audio in Skype (and it also worked
fine before). But it is a bit of progress!


Regards
-- 
Mark Goldstein

Re: [Alsa-user] Edirol UA-101 broken driver

[John Haxby <jch@th...>]

  On 22/09/10 09:46, Arxontis Politis wrote:
>
> ---------------------------------------------------------------
> kane@...:~/Music$ dmesg | grep snd
> [   34.098749] snd_ua101: Unknown parameter `vid'
> [   34.107709] snd_ua101: Unknown parameter `vid'
> [   34.164993] snd_ua101: Unknown parameter `vid'
> [   34.368494] usbcore: registered new interface driver snd-usb-audio
> --------------------------------------------------------------
>
>

I think you have a bad modprobe config file somewhere.  I forget where 
Ubuntu keeps its modprobe options, but I suspect that somewhere you have 
a file that says something like

    options snd_ua101 vid=xyzzy

instead of

    oprions snd_ua101 id=xyzzy

(I don't know what the id should be, but the parameter name is "id" not 
"vid")

jch

[Alsa-user] Edirol UA-101 broken driver

[Arxontis Politis <deadflagblue@gm...>]

Hello,

after upgrading to the latest Ubuntu (10.10), there seems to be some problem
with the edirol ua-101 driver. The card is detected but the module cannot
load.
Here is the lsmod output:
-------------------------------------------------------------
kane@...:~/Music$ lsmod | grep snd
snd_hda_codec_si3054     3440  1
snd_hda_codec_realtek   217980  1
snd_hda_intel          22107  2
snd_hda_codec          87552  3
snd_hda_codec_si3054,snd_hda_codec_realtek,snd_hda_intel
snd_usb_audio          86704  1
snd_pcm                71475  4
snd_hda_codec_si3054,snd_hda_intel,snd_hda_codec,snd_usb_audio
snd_hwdep               5040  2 snd_hda_codec,snd_usb_audio
snd_usbmidi_lib        17413  1 snd_usb_audio
snd_seq_midi            4588  0
snd_rawmidi            17783  2 snd_usbmidi_lib,snd_seq_midi
snd_seq_midi_event      6047  1 snd_seq_midi
snd_seq                47174  2 snd_seq_midi,snd_seq_midi_event
snd_timer              19067  2 snd_pcm,snd_seq
snd_seq_device          5744  3 snd_seq_midi,snd_rawmidi,snd_seq
snd                    49006  18
snd_hda_codec_si3054,snd_hda_codec_realtek,snd_hda_intel,snd_hda_codec,snd_usb_audio,snd_pcm,snd_hwdep,snd_usbmidi_lib,snd_rawmidi,snd_seq,snd_timer,snd_seq_device
soundcore                880  1 snd
snd_page_alloc          7120  2 snd_hda_intel,snd_pcm

---------------------------------------------------------------

and the dmesg output:

---------------------------------------------------------------
kane@...:~/Music$ dmesg | grep snd
[   34.098749] snd_ua101: Unknown parameter `vid'
[   34.107709] snd_ua101: Unknown parameter `vid'
[   34.164993] snd_ua101: Unknown parameter `vid'
[   34.368494] usbcore: registered new interface driver snd-usb-audio
--------------------------------------------------------------

The snd-usb-audio driver is loaded due to a usb-webcam with microphone, I
don't think it has to do with the UA-101.

Any help much appreciated!

Re: [Alsa-user] How to investigate the problem with MIC

[Mark Goldstein <goldstein.mark@gm...>]

On Wed, Sep 22, 2010 at 8:54 AM, Axel Braun <axel.braun@...> wrote:
> Mark Goldstein wrote:
>
>> I started thinking of buying some normal sound card :-(...
>
> Yep :-)
>
>> What holds me still is the fact that it did work perfectly till the
>> end of August.
>> I've got some advices on alsa bugzilla, but I still am too
>> "illiterate" to understand their slang :-). Started reading HDA spec
>> to at least understand the terminology.
>
> can you post the link to alsa-bugzilla?
>

11.3:
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5130
(I only added comments here and the reporter issue was that he had
Smart5.1 unmuted)

11.1:
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5133

Regards,
-- 
Mark Goldstein

Re: [Alsa-user] How to investigate the problem with MIC

[Axel Braun <axel.braun@gm...>]

Mark Goldstein wrote:

> I started thinking of buying some normal sound card :-(...

Yep :-)
 
> What holds me still is the fact that it did work perfectly till the
> end of August.
> I've got some advices on alsa bugzilla, but I still am too
> "illiterate" to understand their slang :-). Started reading HDA spec
> to at least understand the terminology.

can you post the link to alsa-bugzilla?
 
Cheers
Axel
-- 
Linux on a ThinkPad: http://www.axxite.com/brn

Re: [Alsa-user] Nvidia ION HDMI soundcard

[José Luis Segura Lucas <josel.segura@gm...>]

El Tue, 21 Sep 2010 13:37:23 +0100
Robert Parker <theparkers.mailbox@...> escribió:
> Have you tried unmuting devices using alsamixer ?

Yes, of course. I tried using the Gnome-Volume-Manager and alsamixer.
There is one thing that I haven't test: unmutting only one of the 4
switches... 

> In fedora 10 I found that I had to unmute an item called "IEC958" in
> alsamixer to get HDMI working. At least I think it was called that, in
> fc12 I now have to unmute "S/PDIF 1",  even though it is the same h/w
> (Asrock ION).

My box is a Zotac Zbox HD-ID11

> In fc12 I also have to run "alsactl store" to remember the settings
> made in alsamixer. I think I didnt have to do this step with fc10.

Alsamixer keeps its settings, I checked it right now
 
> My  $HOME/.asoundrc looks like this
> 
>  pcm.!default {
>  type plug
>  slave.pcm "hdmi"
> }
> 

I don't have any .asoundrc on my $HOME directory. What would be a
correct asoundrc file for me? Can it be done to system level instead
user level?

Thanks a lot for your answer

Re: [Alsa-user] Nvidia ION HDMI soundcard

[Robert Parker <theparkers.mailbox@gm...>]

Have you tried unmuting devices using alsamixer ?

In fedora 10 I found that I had to unmute an item called "IEC958" in
alsamixer to get HDMI working. At least I think it was called that, in
fc12 I now have to unmute "S/PDIF 1",  even though it is the same h/w
(Asrock ION).

In fc12 I also have to run "alsactl store" to remember the settings made in
alsamixer. I think I didnt have to do this step with fc10.

My  $HOME/.asoundrc looks like this

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

[Alsa-user] Nvidia ION HDMI soundcard

[José Luis Segura Lucas <josel.segura@gm...>]

Good afternoon!

I'm a "Debian unstable" user, using the latest Alsa packaged version.
With that version, I was able to play sounds through my integrated
Intel sound card, but I wasn't to use the HDMI sound capabilities.

My "aplay -l" output could help:

$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: Intel [HDA Intel], device 0: ALC888 Analog [ALC888 Analog]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 0: Intel [HDA Intel], device 1: ALC888 Digital [ALC888 Digital]
  Subdevices: 1/1
  Subdevice #0: subdevice #0

I'm trying to activate the audio support of my Nvidia Ion graphic card,
and I compile alsa-driver from the latest snapshot. I compile and
install it using:

$ ./configure.sh; make;
$ sudo make install

After the installation and reboot, my aplay -l output changes to that:

$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: Intel [HDA Intel], device 0: ALC888 Analog [ALC888 Analog]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 0: Intel [HDA Intel], device 1: ALC888 Digital [ALC888 Digital]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 1: NVidia [HDA NVidia], device 3: NVIDIA HDMI [NVIDIA HDMI]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 1: NVidia [HDA NVidia], device 7: NVIDIA HDMI [NVIDIA HDMI]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 1: NVidia [HDA NVidia], device 8: NVIDIA HDMI [NVIDIA HDMI]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 1: NVidia [HDA NVidia], device 9: NVIDIA HDMI [NVIDIA HDMI]
  Subdevices: 1/1
  Subdevice #0: subdevice #0


As you can see, the HDMI audio devices are working. On alsamixer I can
see 4 IEC958 switches, but I tried enabling them all and I can't get
audio from the HDMI.

Can anybody help me? Thanks a lot.

P.S. If you need another command output, dmesg or any other stuff, let
me know.

Re: [Alsa-user] How to investigate the problem with MIC

[Mark Goldstein <goldstein.mark@gm...>]

On Tue, Sep 21, 2010 at 8:45 AM, Axel Braun <axel.braun@...> wrote:
> Hi Mark,
>
> Mark Goldstein wrote:
>
>> In 11.3 I worked with 2 (now 3) different kernel versions, all with
>> alsa 1.0.23 and the behavior is the same.
>> All my sound worked on 11.3 out of the box. Now the Mic seems working
>> (I hear Mic audio in speaker and headphones) but can't record from Mic
>> (level is 0) and can't use it in Skype - no audio.
>> Yesterday I also noticed that if I'm trying alsaconfig on 11.3,
>> alsaconfig does not find any sound card (even when I use run level 3).
>
>
> ...it seems to go deeper:
> https://bugzilla.novell.com/show_bug.cgi?id=639592
>
> --- Comment #2 from Takashi Iwai <@novell.com> 2010-09-20 15:43:43 UTC ---
> It's likely a known problem of VT1708S.  It seems broken, unfortunately.
>
> No idea if someone going to fix it.
>

I started thinking of buying some normal sound card :-(...

What holds me still is the fact that it did work perfectly till the
end of August.
I've got some advices on alsa bugzilla, but I still am too
"illiterate" to understand their slang :-). Started reading HDA spec
to at least understand the terminology.

But if Takashi Iwai is saying that... too bad for us.

Regards,
-- 
Mark Goldstein

Re: [Alsa-user] How to investigate the problem with MIC

[Axel Braun <axel.braun@gm...>]

Hi Mark,

Mark Goldstein wrote:

> In 11.3 I worked with 2 (now 3) different kernel versions, all with
> alsa 1.0.23 and the behavior is the same.
> All my sound worked on 11.3 out of the box. Now the Mic seems working
> (I hear Mic audio in speaker and headphones) but can't record from Mic
> (level is 0) and can't use it in Skype - no audio.
> Yesterday I also noticed that if I'm trying alsaconfig on 11.3,
> alsaconfig does not find any sound card (even when I use run level 3).


...it seems to go deeper:
https://bugzilla.novell.com/show_bug.cgi?id=639592

--- Comment #2 from Takashi Iwai <@novell.com> 2010-09-20 15:43:43 UTC ---
It's likely a known problem of VT1708S.  It seems broken, unfortunately.

No idea if someone going to fix it.

Cheers
Axel

-- 
Linux on a ThinkPad: http://www.axxite.com/brn

[Alsa-user] All mute after upgrading ALSA

[Luc GMail <lucmove@gm...>]

I didn't have ALSA loopback so I could record streaming, so I upgraded
ALSA following these instructions:

http://alsa.opensrc.org/index.php/Jack_and_Loopback_device_as_Alsa-to-Jack_bridge#Compiling_snd-aloop_if_needed

First thing: that page is WRONG. It instructs me to backup the
kernel/sound directory:

# p -a /lib/modules/`uname -r`/kernel/sound  .

"...
In case anything went wrong and you wish to go back to your previous
ALSA installation, no problem:
sudo rm /lib/modules/`uname -r`/kernel/sound
sudo cp -a ~/backup/sound /lib/modules/`uname -r`/kernel/
sudo alsa force-reload
... "

But that is not enough. As soon as I ran 'sudo make install' I saw
that another directory should have been backed up:

if [ -L /usr/include/sound ]; then \
                rm -f /usr/include/sound; \

I didn't make a backup of /usr/include/sound and now I am screwed. The
backup /lib/modules/`uname -r`/kernel/sound directory doesn't work at
all. My sound card is not even recognized when I boot with it.


What about the new, upgraded ALSA modules? Well, they *look* OK, but I
don't get any sound through the speakers or headphone jack.

The only part I didn't follow verbatim in those instructions was the final step:

"...
Allrighty, time to load it. But before that, shut down all audio apps
(including firefox). Once done, do this:
sudo alsa force-unload
sudo modprobe snd-whatever-module-you-need
sudo modprobe snd-aloop
..."

I was unsure about "snd-whatever-module-you-need" so I decided to just reboot.

When KDE was just about to kick in, I saw a lot of new text I had
never seen before scroll past very, very fast. Then it stopped and I
could see it had something to with ALSA, but I couldn't really read
anything. KDE started and... now I have no sound.

I checked kmixer, it looks normal. It doesn't seem to be mute.

I ran alsamixer in the command line, it doesn't seem to be mute either.

I ran Kcontrol and checked the audio section, couldn't figure anything out.

Here is the output of lsmod BEFORE the upgrade:

$luc> lsmod | grep snd
snd_hda_intel         434100  6
snd_pcm_oss            46336  0
snd_mixer_oss          22656  2 snd_pcm_oss
snd_pcm                83076  3 snd_hda_intel,snd_pcm_oss
snd_seq_dummy          10756  0
snd_seq_oss            37760  0
snd_seq_midi           14336  0
snd_rawmidi            29696  1 snd_seq_midi
snd_seq_midi_event     15104  2 snd_seq_oss,snd_seq_midi
snd_seq                56880  7
snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event
snd_timer              29704  3 snd_pcm,snd_seq
snd_seq_device         14988  5
snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq
snd                    62756  18
snd_hda_intel,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_seq_oss,snd_rawmidi,snd_seq,snd_timer,snd_seq_device
soundcore              15200  2 snd
snd_page_alloc         16904  2 snd_hda_intel,snd_pcm

Here is the output of lsmod AFTER the upgrade:

$luc> lsmod | grep snd
snd_hda_codec_realtek   226308  1
snd_hda_intel          35840  3
snd_hda_codec         100992  2 snd_hda_codec_realtek,snd_hda_intel
snd_hwdep              15364  1 snd_hda_codec
snd_pcm_oss            46112  0
snd_mixer_oss          23040  3 snd_pcm_oss
snd_pcm                85636  3 snd_hda_intel,snd_hda_codec,snd_pcm_oss
snd_seq_oss            36224  0
snd_seq_midi_event     15232  1 snd_seq_oss
snd_seq                57584  5 snd_seq_oss,snd_seq_midi_event
snd_timer              29320  2 snd_pcm,snd_seq
snd_seq_device         15372  2 snd_seq_oss,snd_seq
snd                    69412  14
snd_hda_codec_realtek,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_seq_oss,snd_seq,snd_timer,snd_seq_device
soundcore              15200  3 snd
snd_page_alloc         17032  2 snd_hda_intel,snd_pcm

Looks like I have lost snd_seq_dummy, snd_seq_midi and snd_rawmidi. I
don't need midi, so I just tried to load snd_seq_dummy with modprobe.
The module is not found.

I rebooted another two times hoping to be able to read something in
that new boot sequence text, but I don't see it anymore. It's gone.

I don't know what else to do. I can't go back to the backup modules,
they are useless without /usr/include/sound which the installation
method erased and nobody warned me it would. Please advise.

[Alsa-user] Soundcards are not detected

[vaibhav wasnik <wasnik.vibe@gm...>]

Hi friends,
could someone help me with this.

I dont see no soundcards on lshw, aplay-l or lspci

Attached is my aadebug file


ALSA Audio Debug v0.1.0 - Mon Sep 20 05:02:29 EDT 2010
http://alsa.opensrc.org/aadebug
http://www.gnu.org/licenses/gpl.txt

Kernel ----------------------------------------------------
Linux wasnik-laptop 2.6.32-24-generic #43-Ubuntu SMP Thu Sep 16 14:17:33 UTC
2010 i686 GNU/Linux

Loaded Modules --------------------------------------------
snd_hda_intel          21470  0
snd_hda_codec          87540  1 snd_hda_intel
snd_hwdep               5604  1 snd_hda_codec
snd_pcm_oss            34539  0
snd_mixer_oss          13865  1 snd_pcm_oss
snd_pcm                71582  3 snd_hda_intel,snd_hda_codec,snd_pcm_oss
snd_seq_dummy           1434  0
snd_seq_oss            27210  0
snd_seq_midi            4557  0
snd_rawmidi            19077  1 snd_seq_midi
snd_seq_midi_event      6003  2 snd_seq_oss,snd_seq_midi
snd_seq                47498  6
snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event
snd_timer              18646  2 snd_pcm,snd_seq
snd_seq_device          5988  5
snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq
snd                    56271  13
snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq,snd_timer,snd_seq_device
snd_page_alloc          7076  2 snd_hda_intel,snd_pcm

Modprobe Conf ---------------------------------------------
Warning: module config file does not exist
This means any kernel modules will not be auto loaded
See your linux distro docs on how to create this file

Proc Asound -----------------------------------------------
Advanced Linux Sound Architecture Driver Version 1.0.23.
Compiled on Sep 19 2010 for kernel 2.6.32-24-generic (SMP).
--- no soundcards ---
  1:        : sequencer
 33:        : timer
Client info
  cur  clients : 3
  peak clients : 3
  max  clients : 192

Client   0 : "System" [Kernel]
  Port   0 : "Timer" (Rwe-)
  Port   1 : "Announce" (R-e-)
    Connecting To: 15:0
Client  14 : "Midi Through" [Kernel]
  Port   0 : "Midi Through Port-0" (RWe-)
Client  15 : "OSS sequencer" [Kernel]
  Port   0 : "Receiver" (-we-)
    Connected From: 0:1

Dev Snd ---------------------------------------------------
seq  timer

CPU -------------------------------------------------------
model name    : AMD Turion(tm) 64 X2 Mobile Technology TL-58
cpu MHz        : 800.000
model name    : AMD Turion(tm) 64 X2 Mobile Technology TL-58
cpu MHz        : 800.000

RAM -------------------------------------------------------
MemTotal:        1996316 kB
SwapTotal:       5847032 kB

Hardware --------------------------------------------------
00:18.0 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron]
HyperTransport Technology Configuration
00:18.1 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron]
Address Map
00:18.2 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] DRAM
Controller
00:18.3 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron]
Miscellaneous Control

my output for alsa-info.sh is below
WARNING: All config files need .conf: /etc/modprobe.d/options, it will be
ignored in a future release.
alsa-info.sh: 381: [[: not found
ALSA Information Script v 0.4.59
--------------------------------

This script visits the following commands/files to collect diagnostic
information about your ALSA installation and sound related hardware.

  dmesg
  lspci
  lsmod
  aplay
  amixer
  alsactl
  /proc/asound/
  /sys/class/sound/
  ~/.asoundrc (etc.)

See 'alsa-info.sh --help' for command line options.

alsa-info.sh: 621: [[: not found
alsa-info.sh: 621: [[: not found
alsa-info.sh: 621: [[: not found
alsa-info.sh: 621: [[: not found
alsa-info.sh: 621: [[: not found
alsa-info.sh: 621: [[: not found
alsa-info.sh: 621: [[: not found
alsa-info.sh: 621: [[: not found
WARNING: All config files need .conf: /etc/modprobe.d/options, it will be
ignored in a future release.
alsa-info.sh: 621: [[: not found
alsa-info.sh: 748: [[: not found
alsa-info.sh: 773: [[: not found
Automatically upload ALSA information to http://www.alsa-project.org? [y/N] : read:
773: Illegal option -e
alsa-info.sh: 809: [[: not found

alsa-info.sh: 809: [[: not found

Your ALSA information is in /tmp/alsa-info.txt.JKuRTjZP0q

wasnik@...:~$ pico /tmp/alsa-info.txt.JKuRTjZP0q

[Alsa-user] AES parameter ignored unless using iecset?

[Stephen Kirkby <bugs@st...>]

 Hi,

I'm trying to get hdmi digital pass through sound working. I've got it
working by using iecset first, this works:

saruman ~ # iecset -c0 -n1 audio false
Mode: consumer
Data: non-audio
Rate: 44100 Hz
Copyright: permitted
Emphasis: none
Category: general
Original: 1st generation
Clock: 1000 ppm
saruman ~ # iecset -c0 -n1 -x
AES0=0x06,AES1=0x00,AES2=0x00,AES3=0x00
saruman ~ # aplay -D hdmi:CARD=Intel,DEV=0 SURROUNDTEST_DD_640.wav

However, I need to be able to set to use audio mode with some content
and not others (for mythtv). So next I tried to pass the AES flag to
aplay which I believe should set the audio mode:
saruman ~ # iecset -c0 -n1 audio true
Mode: consumer
Data: audio
Rate: 44100 Hz
Copyright: permitted
Emphasis: none
Category: general
Original: 1st generation
Clock: 1000 ppm
saruman ~ # iecset -c0 -n1 -x
AES0=0x04,AES1=0x00,AES2=0x00,AES3=0x00
saruman ~ # aplay -D
hdmi:CARD=Intel,DEV=0,AES0=0x06,AES1=0x00,AES2=0x00,AES3=0x00
SURROUNDTEST_DD_640.wav

however this doesn't seem to override the current AES0=0x04 settings!

I'm running on ALSA 1.0.23 with a HDA-Intel card - full details of my
system setup (generated by alsa-info.sh) are here:
http://www.alsa-project.org/db/?f=8f0f4cd6e869b602ee37f70a3d64c9aa6b6ccffe

How do I override the AES0 flag? Any help is appreciated!

Thanks
Steve

[Alsa-user] EMU 0404 Midi Input problem

[art3c <bluesbyte@gm...>]

Hello,

Is there any way to get Midi input working in Emu 0404? I tried with latest
Alsa + Alsa firmware on Ubuntu 10.04 and i got some Midi messages, but they
were sensless (for me and my application ;) ).

Thanks for help.

-- 
art3c

Re: [Alsa-user] US-122L only MIDI?

[David Kastrup <dak@gn...>]

Kilian Sprotte <kilian.sprotte@...> writes:

> I am giving up with the Tascam US-122L, hopefully I will have more
> success with the edirol UA-25 EX.
>
> Is there any other USB interface with good sound quality that you
> favor?

Well, I am using a Hammerfall DSP Multiface with a PC-card interface.
As opposed to USB buses, you have good latencies/response on PC-card
(data transfer does not need the CPU at all, so you just need reasonable
interrupt response times).  I still don't get xrun-free operation
without using an rt kernel and the rtirq-init package.  And pulseaudio
has hickups in Flash and wants tsched=0 in Ubuntu 10.04, but jack runs
fine.  On the plus side, quality is good, you have 8 balanced analog ins
and outs, 8-channel ADAT in/out, word clock in/out (so you can use more
than one unit if really necessary), Midi in/out, SPDIF in/out (optical
if you don't use ADAT, and coax anyway), and can go 96/24 if you want,
and the analog circuits don't make a mockery of that mode.  And you have
crossbar mixing capabilities.  On the down side, you still need mic
preamps.

If you can manage to get to your memory bus without going through USB at
all (Cardbus or so), you have somewhat more reassuring prospects to get
low-latency jobs (like real-time effects) and dropless operation going.
The Hammerfall sounded like a good proposition to me in that respect.

Also because I don't think ASIO generally works with Linux USB, but then
I might be mistaken here.

-- 
David Kastrup

Re: [Alsa-user] US-122L only MIDI?

[Kilian Sprotte <kilian.sprotte@gm...>]

Hi,

I am giving up with the Tascam US-122L, hopefully I will have more
success with the edirol UA-25 EX.

Is there any other USB interface with good sound quality that you
favor?

Best, 
  Kilian

Kilian Sprotte <kilian.sprotte@...> writes:

> Hi,
>
> I have a got a new US-122L and I am trying to get it to work with Ubuntu
> and also Gentoo. 
>
> Linux 2.6.32-24-generic
> Advanced Linux Sound Architecture Driver Version 1.0.20.
>
> My success goes as far that it shows up under cards:
>
> $ cat /proc/asound/cards 
>  0 [Intel          ]: HDA-Intel - HDA Intel
>                       HDA Intel at 0xee240000 irq 17
>  1 [US122L         ]: USB US-122L - TASCAM US-122L
>                       TASCAM US-122L (644:800e if 0 at 001/002)
>
> But does not really qualify as an audio device (?):
>
> $ cat /proc/asound/devices 
>   2:        : timer
>   3:        : sequencer
>   4: [ 0- 1]: digital audio playback
>   5: [ 0- 0]: digital audio playback
>   6: [ 0- 0]: digital audio capture
>   7: [ 0- 0]: hardware dependent
>   8: [ 0]   : control
>   9: [ 1- 0]: hardware dependent
>  10: [ 1- 0]: raw midi
>  11: [ 1]   : control
>
> Is the US-122L supposed to be supported as an audio device?
>
> Thanks in advance,
>   Kilian

-- 
Kilian Sprotte
sprotte.org

[Alsa-user] Programming ALSA sound device.

[Bankim Bhavsar <bankim.bhavsar@gm...>]

Producer thread writes sound bytes worth 10 ms every 10 ms to a ring buffer.
Consumer thread is responsible for writing sound bytes to ALSA sound
device in a strictly non-blocking fashion.
Consumer thread checks snd_pcm_avail/snd_pcm_avail_update, fetches
minimum of sound bytes available in ring buffer and
availFrames(returned by snd_pcm_avail_update) and writes to sound
device using snd_pcm_writei().
Consumer thread polls every 20 ms. However on consumer thread
snd_pcm_avail/snd_pcm_avail_update() returns 0 at least 3-4 times
consecutively (total 60-80 ms) and on every 5th-6th poll
snd_pcm_avail/snd_pcm_avail_update returns full buffer size
occasionally reporting an under-run. On an average, under-run is
reported at least 10-15 times a minute which is not desirable.

Following are the hardware and software parameter settings.

hardware params:
   16-bit 44100 stereo
   buffer time 200 ms
   period time 25 ms

software params:
   start_threshold: full buffer size
   no setting for avail_min

Alternatively I've tried using wait_for_poll() technique as mentioned
in the example.
http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html
However poll() unblocks very rapidly and producer ring buffer doesn't
have enough bytes.

What's the best technique for PCM playback for such use cases? Is it
okay to skip calling snd_pcm_avail/snd_pcm_avail_update and directly
attempt writing to ALSA sound device instead?
Let me know if more information is required.

Thanks,
Bankim.