Asymetrical sound cards and jack?

Hello Jack developers

Like most PC sound cards my sound is asymmetrical in that it
has more
outputs than inputs.

To be precise:

IEC958 ouputs - Front (Stereo), Side (Stereo), Rear
(Stereo), Centre/LFE
(Mono)

Analogue outputs - Front (Stereo), Side (Stereo), Rear
(Stereo),
Centre/LFE (Mono)

Analogue inputs - Line in (Stereo), Mic in (Stereo), Phone ,
Aux  (Stereo)

Since ALSA version 1.013 the alsa driver is correct defined
for my sound
card. The KDE and Gnome standard mixers show all the
channels listed
above and provide the necessary faders and switches.
Programs like
Audacity, Xine, Kaffeine and Flash all communicate with the
ALSA drivers
correctly on all channels.

However, Jack only gives 2 playbacks and 2 captures, and
does not
indicate to where the captures and playbacks are connected.

My /etc/asound.conf looks like this:

 pcm.ca0106 {
   type hw
   card 0
}

ctl.ca0106 {
   type hw
   card 0
}

pcm.UM1 {
   type hw
   card 1
}

ctl.UM1 {
   type hw
   card 1
}


"UM1" is a midi-to-USB convertor, which is set to
card 1 by the
following entry in /etc/modprobe.d/alsa-base

options snd-usb-audio index=-2


The PAM settings in /etc/security/limits.conf are

audio - rtprio 100
audio - memlock 4000000
audio - nice -10


The jackd priority is set by the following entry in
/etc/set_rlimits.conf

audio  /usr/bin/jackd  nice=-10  rtprio=100
memlock=4000000


I am using openSUSE 10.2 (without zenworks!).

Jackd starts only in softmode (from user account - not root)
and
indicates a latency of 5.33 msec with 24bit/96kHz.


My questions:

How do I force jack to list _all_ sound card inputs and
outputs by their
proper names?

How can I run jack in realtime without softmode?


Hoping for a positive reply, best wishes, Simon




------------------------------------------------------------
-------------
Take Surveys. Earn Cash. Influence the Future of IT
Join SourceForge.net's Techsay panel and you'll get the
chance to share your
opinions on IT & business topics through brief
surveys-and earn cash
http://www.techsay.com/default.
php?page=join.php&p=sourceforge&CID=DEVDEV
_______________________________________________
Jackit-devel mailing list
Jackit-devellists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/jackit-dev
el

On Thu, 2007-03-01 at 17:07 +0100, Simon Lewis wrote:
> Hello Jack developers
> 
> Like most PC sound cards my sound is asymmetrical in
that it has more
> outputs than inputs.
> 
> To be precise:
> 
> IEC958 ouputs - Front (Stereo), Side (Stereo), Rear
(Stereo), Centre/LFE
> (Mono)
> 
> Analogue outputs - Front (Stereo), Side (Stereo), Rear
(Stereo),
> Centre/LFE (Mono)
> 
> Analogue inputs - Line in (Stereo), Mic in (Stereo),
Phone , Aux  (Stereo)
> 
> Since ALSA version 1.013 the alsa driver is correct
defined for my sound
> card. The KDE and Gnome standard mixers show all the
channels listed
> above and provide the necessary faders and switches.
Programs like
> Audacity, Xine, Kaffeine and Flash all communicate with
the ALSA drivers
> correctly on all channels.
> 
> However, Jack only gives 2 playbacks and 2 captures,
and does not
> indicate to where the captures and playbacks are
connected.

I suspect that your audio interface is represented by ALSA
as several
different devices. please show the contents of
/proc/asound/cards.

> The jackd priority is set by the following entry in
/etc/set_rlimits.conf
> 
> audio  /usr/bin/jackd  nice=-10  rtprio=100
memlock=4000000

absolutely no reason to include this if you are you using
PAM
via /etc/security/limits.conf. if you want to use
set_rtlimit anyway,
keep this but drop the nice part - its irrelevant.

> Jackd starts only in softmode (from user account - not
root) and
> indicates a latency of 5.33 msec with 24bit/96kHz.

i don't think you mean "softmode" in the way we
mean "softmode".

will jackd start with the -R flag when started as a non-root
user?



------------------------------------------------------------
-------------
Take Surveys. Earn Cash. Influence the Future of IT
Join SourceForge.net's Techsay panel and you'll get the
chance to share your
opinions on IT & business topics through brief
surveys-and earn cash
http://www.techsay.com/default.
php?page=join.php&p=sourceforge&CID=DEVDEV
_______________________________________________
Jackit-devel mailing list
Jackit-devellists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/jackit-dev
el

Dear Jack Developers

This is addressed to the Jack developers whose speciality is
interfacing
with ALSA drivers.

As you will see from the correspondence below with Paul
Davis, I have
been unable to find a solution to using jack with audio
cards that have
unequal numbers of inputs and outputs.

The ALSA driver appears to be correct as kmix & audacity
both see all
inputs and ouputs and allow the levels to be adjusted.

The question is how do I force jack to read the alsa driver
correctly
and show all inputs and outputs in the qjackctl connections
dialogue?

Please take this question seriously as I see from various
forums that
this is an ever recurring problem.


Besr reagards, Simon Lewis
Lautensackstrasse 12
80687 München

Tel.: +49 89 57869412
Fax: +49 89 57869411
e-mail: simon.lewismnet-online.de



Previos correspondance:
_______________________


Dear Paul

Thank-you for responding to my e-mail. The contents of
/proc/asound are
as follows:

simonsimons-pc:~> cat -v /proc/asound/modules
 0 snd_ca0106
 1 snd_usb_audio

simonsimons-pc:~> cat -v /proc/asound/cards
 0 [CA0106         ]: CA0106 - CA0106
                      Live! 7.1 24bit [SB0410] at 0xdc00 irq
5
 1 [UM1            ]: USB-Audio - UM-1
                      EDIROL UM-1 at usb-0000:00:07.3-2,
full speed

simonsimons-pc:~> cat -v /proc/asound/devices
  2:        : timer
  3: [ 0- 0]: raw midi
  4: [ 0- 3]: digital audio playback
  5: [ 0- 3]: digital audio capture
  6: [ 0- 2]: digital audio playback
  7: [ 0- 2]: digital audio capture
  8: [ 0- 1]: digital audio playback
  9: [ 0- 1]: digital audio capture
 10: [ 0- 0]: digital audio playback
 11: [ 0- 0]: digital audio capture
 12: [ 0]   : control
 13: [ 1- 0]: raw midi
 14: [ 1]   : control
 15:        : sequencer

simonsimons-pc:~> cat -v /proc/asound/pcm
00-03: ca0106 : CA0106 : playback 1 : capture 1
00-02: ca0106 : CA0106 : playback 1 : capture 1
00-01: ca0106 : CA0106 : playback 1 : capture 1
00-00: ca0106 : CA0106 : playback 1 : capture 1

simonsimons-pc:~> cat -v /proc/asound/timers
G0: system timer : 4000.000us (10000000 ticks)
P0-0-0: PCM playback 0-0-0 : SLAVE
P0-0-1: PCM capture 0-0-1 : SLAVE
P0-1-0: PCM playback 0-1-0 : SLAVE
P0-1-1: PCM capture 0-1-1 : SLAVE
P0-2-0: PCM playback 0-2-0 : SLAVE
P0-2-1: PCM capture 0-2-1 : SLAVE
P0-3-0: PCM playback 0-3-0 : SLAVE
P0-3-1: PCM capture 0-3-1 : SLAVE

simonsimons-pc:~> cat -v /proc/asound/version
Advanced Linux Sound Architecture Driver Version 1.0.12rc1
(Thu Jun 22
13:55:50 2006 UTC).

The timer is shown as 4msec (250Hz). Should this be set to
1msec or even
100µsec and how do I change this perminately?

The alsa version is shown as 1.0.12rc1. The installed rpm
shows the
version 1.0.13-22..!!!

??? It would  be nice to see the 2 inputs and 7 outputs
showing in the
connect window of qjackctl.

_______________________________

I have been using qjackctl to start jack with and without
the softmode
selected. With softmode selected jack starts, without
softmode jack does
not start. See the 2 messages:

With softmode selected...

10:00:55.295 Startup script...
10:00:55.298 artsshell -q terminate
unix_connect: can't connect to server
(unix:/tmp/ksocket-simon/simons-pc.lewis.net-3c75-45ca47fe)
10:00:55.693 Startup script terminated with exit status=256.
10:00:55.696 JACK is starting...
10:00:55.704 /usr/bin/jackd -R -dalsa -dhw:0 -r96000 -p512
-n2 -s -i2 -o2
10:00:55.719 JACK was started with PID=14067 (0x36f3).
jackd 0.102.20
Copyright 2001-2005 Paul Davis and others.
jackd comes with ABSOLUTELY NO WARRANTY
This is free software, and you are welcome to redistribute
it
under certain conditions; see the file COPYING for details
JACK compiled with POSIX SHM support.
loading driver ..
apparent rate = 96000
creating alsa driver ...
hw:0|hw:0|512|2|96000|2|2|nomon|swmeter|soft-mode|32bit
control device hw:0
configuring for 96000Hz, period = 512 frames, buffer = 2
periods
ALSA: final selected sample format for capture: 32bit
little-endian
ALSA: use 2 periods for capture
ALSA: final selected sample format for playback: 32bit
little-endian
ALSA: use 2 periods for playback
10:00:57.834 Server configuration saved to
"/home/simon/.jackdrc".
10:00:57.838 Statistics reset.
10:00:57.853 Client activated.
10:00:57.860 Audio connection change.
10:00:57.886 Audio connection graph change.

With softmode unselected...

09:59:55.829 Startup script...
09:59:55.829 artsshell -q terminate
unix_connect: can't connect to server
(unix:/tmp/ksocket-simon/simons-pc.lewis.net-3c75-45ca47fe)
09:59:56.593 Startup script terminated with exit status=256.
09:59:56.593 JACK is starting...
09:59:56.593 /usr/bin/jackd -R -dalsa -dhw:0 -r96000 -p512
-n2 -i2 -o2
09:59:56.616 JACK was started with PID=14038 (0x36d6).
jackd 0.102.20
Copyright 2001-2005 Paul Davis and others.
jackd comes with ABSOLUTELY NO WARRANTY
This is free software, and you are welcome to redistribute
it
under certain conditions; see the file COPYING for details
JACK compiled with POSIX SHM support.
loading driver ..
apparent rate = 96000
creating alsa driver ...
hw:0|hw:0|512|2|96000|2|2|nomon|swmeter|-|32bit
control device hw:0
configuring for 96000Hz, period = 512 frames, buffer = 2
periods
ALSA: final selected sample format for capture: 32bit
little-endian
ALSA: use 2 periods for capture
ALSA: final selected sample format for playback: 32bit
little-endian
ALSA: use 2 periods for playback
09:59:58.678 Server configuration saved to
"/home/simon/.jackdrc".
09:59:58.702 Statistics reset.
09:59:58.926 Client activated.
09:59:58.927 Audio connection change.
09:59:58.965 Audio connection graph change.
09:59:58.966 XRUN callback (1).
10:00:00.961 XRUN callback (188 skipped).
jackd watchdog: timeout - killing jackd
zombified - calling shutdown handler
10:00:02.059 Shutdown notification.
10:00:02.063 Client deactivated.
10:00:02.068 JACK was stopped successfully.
10:00:02.069 Post-shutdown script...
10:00:02.070 killall jackd
cannot send request type 7 to server
cannot read result for request type 7 from server
(Datenübergabe
unterbrochen (broken pipe))
cannot send request type 7 to server
cannot read result for request type 7 from server
(Datenübergabe
unterbrochen (broken pipe))
jackd: Kein Prozess abgebrochen
10:00:02.490 Post-shutdown script terminated with exit
status=256.

___________________________

I have been corresponding with openSUSE's kernel expert
Takaschi Iwai
who confirmed that the rtlimits must be specified in
openSUSE even when
using PAM. All I know is that jack runs with less xruns when
the
rtlimits are specified

___________________________

Finally, I have been using openSUSE for 3 years. I have
tried other
distributions from time to time including Zenwalk, Ubuntu,
Knoppix, 64
Studio, Arch-Linuc, Foresight, Dyne:bolic and JAD. I always
return to
openSUSE because it is incredably stable, actual, regular
updates, well
supported (e.g.packman  and guru) and programs that are
installed from
sources (when no rpm is availble) always seem to run.

I would like a second distribution tailored to music
recording e.g with
samplers and synthesizes but have not found an optimum
system.
Performance wise dyne:bolic is the best, if only it had a
conventional
disk installation and not docking... Can you recommend a
linux
distrubution for audio work that does not need extensive
configuration?

Once again many thanks for your help, best wishes

Simon Lewis
Lautensackstrasse 12
80687 München

Tel.: +49 89 57869412
Fax: +49 89 57869411
e-mail: simon.lewismnet-online.de


Paul Davis schrieb:
> On Thu, 2007-03-01 at 17:07 +0100, Simon Lewis wrote:
>   
>> Hello Jack developers
>>
>> Like most PC sound cards my sound is asymmetrical
in that it has more
>> outputs than inputs.
>>
>> To be precise:
>>
>> IEC958 ouputs - Front (Stereo), Side (Stereo), Rear
(Stereo), Centre/LFE
>> (Mono)
>>
>> Analogue outputs - Front (Stereo), Side (Stereo),
Rear (Stereo),
>> Centre/LFE (Mono)
>>
>> Analogue inputs - Line in (Stereo), Mic in
(Stereo), Phone , Aux  (Stereo)
>>
>> Since ALSA version 1.013 the alsa driver is correct
defined for my sound
>> card. The KDE and Gnome standard mixers show all
the channels listed
>> above and provide the necessary faders and
switches. Programs like
>> Audacity, Xine, Kaffeine and Flash all communicate
with the ALSA drivers
>> correctly on all channels.
>>
>> However, Jack only gives 2 playbacks and 2
captures, and does not
>> indicate to where the captures and playbacks are
connected.
>>     
>
> I suspect that your audio interface is represented by
ALSA as several
> different devices. please show the contents of
/proc/asound/cards.
>
>   
>> The jackd priority is set by the following entry in
/etc/set_rlimits.conf
>>
>> audio  /usr/bin/jackd  nice=-10  rtprio=100
memlock=4000000
>>     
>
> absolutely no reason to include this if you are you
using PAM
> via /etc/security/limits.conf. if you want to use
set_rtlimit anyway,
> keep this but drop the nice part - its irrelevant.
>
>   
>> Jackd starts only in softmode (from user account -
not root) and
>> indicates a latency of 5.33 msec with 24bit/96kHz.
>>     
>
> i don't think you mean "softmode" in the way
we mean "softmode".
>
> will jackd start with the -R flag when started as a
non-root user?
>
>
>   




------------------------------------------------------------
-------------
Take Surveys. Earn Cash. Influence the Future of IT
Join SourceForge.net's Techsay panel and you'll get the
chance to share your
opinions on IT & business topics through brief
surveys-and earn cash
http://www.techsay.com/default.
php?page=join.php&p=sourceforge&CID=DEVDEV
_______________________________________________
Jackit-devel mailing list
Jackit-devellists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/jackit-dev
el

On Thursday 01 March 2007, Simon Lewis wrote:


> Hello Jack developers

I'm not a jack developer, but i try to answer anyways...

> Like most PC sound cards my sound is asymmetrical in
that it has more
> outputs than inputs.
>
> To be precise:
>
> IEC958 ouputs - Front (Stereo), Side (Stereo), Rear
(Stereo), Centre/LFE
> (Mono)
>
> Analogue outputs - Front (Stereo), Side (Stereo), Rear
(Stereo),
> Centre/LFE (Mono)
>
> Analogue inputs - Line in (Stereo), Mic in (Stereo),
Phone , Aux  (Stereo)
>
> Since ALSA version 1.013 the alsa driver is correct
defined for my sound
> card. The KDE and Gnome standard mixers show all the
channels listed
> above and provide the necessary faders and switches.
Programs like
> Audacity, Xine, Kaffeine and Flash all communicate with
the ALSA drivers
> correctly on all channels.
>
> However, Jack only gives 2 playbacks and 2 captures,
and does not
> indicate to where the captures and playbacks are
connected.

You are severely confused, but that's ok, since ALSA can be
severely 
confusing 

jack uses the device specified via the -d switch of the ALSA
driver options:

jackd -d alsa -d hw:0  -p 512

makes jack use the audio device hw:0,0

which corresponds in my case to this entry in
/proc/asound/devices:

  7: [ 0- 0]: digital audio playback
  8: [ 0- 0]: digital audio capture

which belongs to my CS46XX compatible card (which doesn't
support periodsizes 
> 512 for the record):

~$ cat /proc/asound/cards 
 0 [CS46xx         ]: CS46xx - Sound Fusion CS46xx
                      Sound Fusion CS46xx at
0xe8111000/0xe8000000, irq 18
 1 [M66            ]: ICE1712 - M Audio Delta 66
                      M Audio Delta 66 at 0xd100, irq 19

As you can see from the full /proc/asound/devices output:

~$ cat /proc/asound/devices 
  2:        : timer
  3:        : sequencer
  4: [ 0- 0]: raw midi
  5: [ 0- 2]: digital audio playback
  6: [ 0- 1]: digital audio playback
  7: [ 0- 0]: digital audio playback
  8: [ 0- 0]: digital audio capture
  9: [ 0]   : control
 10: [ 1- 0]: digital audio playback
 11: [ 1- 0]: digital audio capture
 12: [ 1]   : control

card 0 [CS46XX] has two more audio playback devices..

Ok running jackd -d alsa -d hw:0 -p 512

gives me running jack_lsp:
~$ jack_lsp  
alsa_pcm:capture_1
alsa_pcm:capture_2
alsa_pcm:playback_1
alsa_pcm:playback_2

This is to be expected becuase hw:0,0 is a 2 channel full
duplex device..

running jackd -d alsa -d hw:0,1 -p 512:

~$ jack_lsp  
alsa_pcm:playback_1
alsa_pcm:playback_2

This is also ok, since hw:0,1 is a 2 channel playback only
device.. So jackd 
handles channel counts for different devices properly. If i
use my delta66 
[which is an ice1712 device which always has 12 inputs and
10 playback 
channels]:

jackd -d alsa -d hw:1,0

i get:

~$ jack_lsp  
alsa_pcm:capture_1
alsa_pcm:capture_2
alsa_pcm:capture_3
alsa_pcm:capture_4
alsa_pcm:capture_5
alsa_pcm:capture_6
alsa_pcm:capture_7
alsa_pcm:capture_8
alsa_pcm:capture_9
alsa_pcm:capture_10
alsa_pcm:capture_11
alsa_pcm:capture_12
alsa_pcm:playback_1
alsa_pcm:playback_2
alsa_pcm:playback_3
alsa_pcm:playback_4
alsa_pcm:playback_5
alsa_pcm:playback_6
alsa_pcm:playback_7
alsa_pcm:playback_8
alsa_pcm:playback_9
alsa_pcm:playback_10


So the point is: jackd uses the device you specify for
using. In this it is 
completely the same as any other ALSA program, with the
notable difference 
that GUI programs often offer you a dialog to select a
device from. Jackd 
doesn't do this. It just uses what's specified on the
commandline [or the 
default hw:0,0, iirc].

So back to my CS46XX card. From the /proc/asound/devices
output it is clear to 
see that my soundcard isn't really a single alsa device with
2 inputs and 6 
playback channels, but rather it is a collection of devices,
some full duplex 
and some playback only. 

And, this is important, the channels in alsamixer do not
correspond 
nessecarily to hw devices. I.e. my soundcard has a line in
and a mic in. 
These are not two seperate channels on the pcm device layer.
Thee are two 
different channels only at the alsa mixer level. You select
one of them via 
alsamixer and then whatever is on the selected channel is
coming in through 
the hw:0,0 capture channels.

Also some soundcards allow routing the pcm streams to
different physical 
outputs on the cards. I.e. for a typical soundcard pcm:0,0
might caputre from 
one of several inputs and might playback to one of several
outputs depending 
on the particular alsa mixer settings used.

So what you really want to do is to

a] make jack use more than one device (this way you could
i.e. hw:0,0, hw:0,1 
and hw:0,1 (in my case) to use) at a time

jackd has a good reason to disallow this:

It is not nessecarily a given that all the devices on a
soundcard are sample 
synced. Missing sample syncage means the audio streams run
at slightly 
different rates. This is fatal in the long run. Anywhere in
the time range of 
seconds to hours jackd will start giving you XRuns like hell
and might stop 
functioning alltogether. you don't want that..

If you are positive that the different pcm hw devices on
your soundcard are 
sample synced, you can try using the alsa pcm multi plugin
to combine your 
pcm devices into a single one with a higher channel count.

take a look at 

http://alsa.opensrc.org

http://alsa.opensrc.org/F
AQ is a good read, too

especially:

http://alsa.ope
nsrc.org/TwoCardsAsOne

You can probably adjust the example given easily to your
situation. In the 
case with my cs46xx card i would use something like:

pcm.multi {
    type multi;
    slaves.a.pcm "hw:0,0";
    slaves.a.channels 2;
    slaves.b.pcm "hw:0,1";
    slaves.b.channels 2;
    bindings.0.slave a;
    bindings.0.channel 0;
    bindings.1.slave a;
    bindings.1.channel 1;
    bindings.2.slave b;
    bindings.2.channel 0;
    bindings.3.slave b;
    bindings.3.channel 1;
}

which combines the devices hw:0,0 and hw:0,1 into a single
device. Run jackd 
with:

jackd -d alsa -d multi  -p 512 -i 2 -o 4

[i don't know why the -i and -o options are nessecary here
but not with my 
ice1712, oh well]

~$ jack_lsp 
alsa_pcm:capture_1
alsa_pcm:capture_2
alsa_pcm:playback_1
alsa_pcm:playback_2
alsa_pcm:playback_3
alsa_pcm:playback_4

tadaa, there's the channels... I suppose you can cascade
this setup to involve 
a third pcm device.. Keep in mind though that jack works
best directly 
on "hw" devices. Software pcm devices like this
multi pcm sometimes give less 
than optimal performance.. Might have changed with newer
alsa versions 
though..

>
> My /etc/asound.conf looks like this:

Irrelevant..  You  simply define two pcm device names which
map to hw:0,0 and 
hw:1,0 respectively.. You don't really gain anything with
this.

>
> "UM1" is a midi-to-USB convertor, which is
set to card 1 by the
> following entry in /etc/modprobe.d/alsa-base
>
> options snd-usb-audio index=-2

You could alternatively use

options snd-usb-audio index=1

and use index=0 as option for the other soundcard..

> The PAM settings in /etc/security/limits.conf are
>
> audio - rtprio 100
> audio - memlock 4000000
> audio - nice -10
>
>
> The jackd priority is set by the following entry in
/etc/set_rlimits.conf
>
> audio  /usr/bin/jackd  nice=-10  rtprio=100
memlock=4000000

Yuo don't need that if you have the above three lines..

> I am using openSUSE 10.2 (without zenworks!).
>
> Jackd starts only in softmode (from user account - not
root) and
> indicates a latency of 5.33 msec with 24bit/96kHz.

What do you get if you run 

jackd -R 

from the commandline? I don't want to see a descripton of
the output, but 
rather the output itself 

>
>
> My questions:
>
> How do I force jack to list _all_ sound card inputs and
outputs by their
> proper names?

This is a two part questions:

a] how do i force jackd to use all soundcard channels which
are usable as pcm 
devices?

answer: combine the different pcm devices into a bigger one
by way of the alsa 
pcm multi device

b] how do i force jackd to give all the channels the proper
names

answer: you have to modify jackd's ALSA driver for this. All
channels exposed 
by the jackd ALSA driver are called alsa_pcm:playback or
alsa_pcm:capture 
with an index. Optionally the jack alsa driver also accepts
optins to give 
base names to capture and playback channels. This is the
same basename plus 
an index though for each channel.. [see man jackd]

>
> How can I run jack in realtime without softmode?

Use the -R option

> Hoping for a positive reply, best wishes, Simon

One doesn't always get what one wants 

Regards,
Flo


-- 
Palimm Palimm!
http://tapas.affenbande.o
rg

------------------------------------------------------------
-------------
Take Surveys. Earn Cash. Influence the Future of IT
Join SourceForge.net's Techsay panel and you'll get the
chance to share your
opinions on IT & business topics through brief
surveys-and earn cash
http://www.techsay.com/default.
php?page=join.php&p=sourceforge&CID=DEVDEV
_______________________________________________
Jackit-devel mailing list
Jackit-devellists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/jackit-dev
el

On 3/7/07, Simon Lewis <simon.lewismnet-online.de> wrote:
> Using the terminal command jackd -d alsa -d multi -p
512 -i 2 -o 8 I get
> the following message:
>

Because the multi device you defined is just a playback
device.  Try:

jackd -d alsa -P multi -C hw:0,0 -p 512 -i 2 -o 8

> Also, in QJackCtl "Setup" window -
"Settings" tab there is no interfaces
> options for Multi or ttable.
>

qjackctl needs to be updated to use ALSA's new device
enumeration API.

> I just wish that the developers in jack, alsa, pam and
the other related
> projects would put their heads together and design a
simple GUI that
> would guide the average user though the jack audio
configuration process.

It's being worked on but it's not easy.

Many drivers expose a single multichannel hw PCM for example
RME,
ICE1712/1724, and emu10k1 which makes things easier.  CA0106
does not
seem to have this.

Lee

------------------------------------------------------------
-------------
Take Surveys. Earn Cash. Influence the Future of IT
Join SourceForge.net's Techsay panel and you'll get the
chance to share your
opinions on IT & business topics through brief
surveys-and earn cash
http://www.techsay.com/default.
php?page=join.php&p=sourceforge&CID=DEVDEV
_______________________________________________
Jackit-devel mailing list
Jackit-devellists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/jackit-dev
el

On Wed, Mar 07, 2007 at 04:16:04PM +0100, Simon Lewis
wrote:
> Using the terminal command jackd -d alsa -d multi -p
512 -i 2 -o 8 I get
> the following message:
> 
> simonsimons-pc:/> jackd -d alsa -d multi -p 512 -i 2 -o
8
>
> ALSA: cannot set channel count to 2 for capture
> ALSA: cannot configure capture channel
> cannot load driver module alsa

Specifying -i and -o options doesn't work unless the
numbers
correspond to what your card's chipset has (and in the case
of the
multi device this could even be impossible). Just leave them
out
and let jackd work it out for itself. The -i and -o options
are
actually redundant in any case that I can think of.

You will also run into another problem - pcm_multi doesn't
work
in duplex mode with jackd without this patch:

http://www.sound-man.co.uk/linuxaudio/pcm_multi-patch

It got broken in alsa-lib 1.0.9rc1 and never got fixed.
I've raised the issue a couple of times on alsa-devel but
nobody has had the time and/or test hardware to do anything
about it.

John

------------------------------------------------------------
-------------
Take Surveys. Earn Cash. Influence the Future of IT
Join SourceForge.net's Techsay panel and you'll get the
chance to share your
opinions on IT & business topics through brief
surveys-and earn cash
http://www.techsay.com/default.
php?page=join.php&p=sourceforge&CID=DEVDEV
_______________________________________________
Jackit-devel mailing list
Jackit-devellists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/jackit-dev
el

On 3/7/07, John Rigg <jsound-man.co.uk> wrote:
> You will also run into another problem - pcm_multi
doesn't work
> in duplex mode with jackd without this patch:
>
> http://www.sound-man.co.uk/linuxaudio/pcm_multi-patch
>
> It got broken in alsa-lib 1.0.9rc1 and never got
fixed.
> I've raised the issue a couple of times on alsa-devel
but
> nobody has had the time and/or test hardware to do
anything
> about it.

Can you please submit the patch to alsa-devel for inclusion
(with
description and Signed-Off-By)?  If no one has time to fix
this
regression then the next best thing is to back out the
problem patch.

Lee

------------------------------------------------------------
-------------
Take Surveys. Earn Cash. Influence the Future of IT
Join SourceForge.net's Techsay panel and you'll get the
chance to share your
opinions on IT & business topics through brief
surveys-and earn cash
http://www.techsay.com/default.
php?page=join.php&p=sourceforge&CID=DEVDEV
_______________________________________________
Jackit-devel mailing list
Jackit-devellists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/jackit-dev
el

On Wednesday 07 March 2007, Lee Revell wrote:
> On 3/7/07, Simon Lewis <simon.lewismnet-online.de> wrote:
> > Using the terminal command jackd -d alsa -d multi
-p 512 -i 2 -o 8 I get
> > the following message:
>
> Because the multi device you defined is just a playback
device.  Try:
>
> jackd -d alsa -P multi -C hw:0,0 -p 512 -i 2 -o 8

Interesting. Why does jack show some nonfunctioning ports
then though?

pcm.multi {
    type multi;
    slaves.a.pcm "hw:0,0";
    slaves.a.channels 2;
    slaves.b.pcm "hw:0,1";
    slaves.b.channels 2;
    bindings.0.slave a;
    bindings.0.channel 0;
    bindings.1.slave a;
    bindings.1.channel 1;
    bindings.2.slave b;
    bindings.2.channel 0;
    bindings.3.slave b;
    bindings.3.channel 1;
}

jackd -R -d alsa -d multi -i 2 -o 4 -p 512

c~$ jack_lsp 
alsa_pcm:capture_1
alsa_pcm:capture_2
alsa_pcm:playback_1
alsa_pcm:playback_2
alsa_pcm:playback_3
alsa_pcm:playback_4

It even says:

"ALSA: Cannot open PCM device alsa_pcm for capture.
Falling back to 
playback-only mode"

but shows two nonexisting ports..

Flo


-- 
Palimm Palimm!
http://tapas.affenbande.o
rg

------------------------------------------------------------
-------------
Take Surveys. Earn Cash. Influence the Future of IT
Join SourceForge.net's Techsay panel and you'll get the
chance to share your
opinions on IT & business topics through brief
surveys-and earn cash
http://www.techsay.com/default.
php?page=join.php&p=sourceforge&CID=DEVDEV
_______________________________________________
Jackit-devel mailing list
Jackit-devellists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/jackit-dev
el

Hello Lee

Thanks for your reply. I tried the terminal command as you
suggested,
but I don't understand the error message regarding the
DRIVER-NT:

simonsimons-pc:~> jackd -d alsa -P multi -C hw:0,0 -p
512 -i 2 -o 8
jackd 0.102.20
Copyright 2001-2005 Paul Davis and others.
jackd comes with ABSOLUTELY NO WARRANTY
This is free software, and you are welcome to redistribute
it
under certain conditions; see the file COPYING for details

JACK compiled with POSIX SHM support.
loading driver ..
creating alsa driver ...
multi|hw:0,0|512|2|48000|2|8|nomon|swmeter|-|32bit
configuring for 48000Hz, period = 512 frames, buffer = 2
periods
ALSA: final selected sample format for capture: 32bit
little-endian
ALSA: use 2 periods for capture
ALSA: final selected sample format for playback: 32bit
little-endian
ALSA: use 2 periods for playback
ALSA: poll time out, polled for 15998184 usecs
DRIVER NT: could not run driver cycle
jack main caught signal 12
no message buffer overruns

Please can you tell me how I should modify asound.conf to
make multi
both capture and playback?

Best wishes, Simon


Lee Revell schrieb:
> On 3/7/07, Simon Lewis <simon.lewismnet-online.de> wrote:
>> Using the terminal command jackd -d alsa -d multi
-p 512 -i 2 -o 8 I get
>> the following message:
>>
>
> Because the multi device you defined is just a playback
device.  Try:
>
> jackd -d alsa -P multi -C hw:0,0 -p 512 -i 2 -o 8
>
>> Also, in QJackCtl "Setup" window -
"Settings" tab there is no interfaces
>> options for Multi or ttable.
>>
>
> qjackctl needs to be updated to use ALSA's new device
enumeration API.
>
>> I just wish that the developers in jack, alsa, pam
and the other related
>> projects would put their heads together and design
a simple GUI that
>> would guide the average user though the jack audio
configuration
>> process.
>
> It's being worked on but it's not easy.
>
> Many drivers expose a single multichannel hw PCM for
example RME,
> ICE1712/1724, and emu10k1 which makes things easier. 
CA0106 does not
> seem to have this.
>
> Lee
>
>


------------------------------------------------------------
-------------
Take Surveys. Earn Cash. Influence the Future of IT
Join SourceForge.net's Techsay panel and you'll get the
chance to share your
opinions on IT & business topics through brief
surveys-and earn cash
http://www.techsay.com/default.
php?page=join.php&p=sourceforge&CID=DEVDEV
_______________________________________________
Jackit-devel mailing list
Jackit-devellists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/jackit-dev
el

On 3/7/07, Daniel James <daniel64studio.com> wrote:
> Presumably any pair of soundcards would do to test
this? Or need they be
> identical?

I believe any pair of soundcards or even a multi device set
up to
combine 2 PCMs on the same soundcard.  IIRC all that's
needed to hit
the bug is the combination of ALSA multi plugin and poll().

It would be great if someone had the time to really bug the
ALSA devs
about this.  AFAICT the patch that introduced the problem
was supposed
to be a code cleanup not a bug fix.

Lee

------------------------------------------------------------
-------------
Take Surveys. Earn Cash. Influence the Future of IT
Join SourceForge.net's Techsay panel and you'll get the
chance to share your
opinions on IT & business topics through brief
surveys-and earn cash
http://www.techsay.com/default.
php?page=join.php&p=sourceforge&CID=DEVDEV
_______________________________________________
Jackit-devel mailing list
Jackit-devellists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/jackit-dev
el