20 Apr
2013
20 Apr
'13
5:06 p.m.
Hello everyone!
I've still got problems joining my twosoundcards (Delta 1010LT and EMU
1212m). I've created the multi device for capture and playback, taken care of
the bit width conversion and still it doesn't work. See the .asoundrc below.
With Delta and EMU's analog I/O (device 0) JACK will start, but gives a lot
of ALSA_PCM xruns. With the S/PDIF and/or ADAT/DSP part of the EMU card
included JACK won't start or will shutdown immediately. Also ecasound/arecord
won't work.
Having tried the EMU S/PDIF on its own with arecord, I found, that it would
only allow 8kHz samplerate, instead of the requested 48kHz. The EMU card is
however setup to take the clocksource from the delta card, which works
perfectly with 48kHz. OK, even it takes its clock from the outside (Lindy A/D
converter).
To test the S/PDIF in on the EMU I tried playing back something to the
Delta's S/PDIF out, but only got silence.
So it looks, like there is more than one problem. But I can't spot it. I've
looked at the .asoundrc documentation again and the list of ALSA plugins. OK,
there is the route plugin for creating interleaved audio, using the ttable,
but for a start, I've read, that this will cause great latency. Anyway, I've
built my JACK version myself and I don't see, why it shouldn't include complex
MMAP. But that is a little further along the way.
So here's my .asoundrc:
# The 8 ADAT ins of EMU 1212m, conversion to S16_LE
pcm.emu_adat {
type linear
slave {
pcm "hw:4,2"
format S32_LE
}
}
# The 2 Ins, 2 Outs of EMU 1212m, they have S16_LE
pcm.emu_analog {
type hw
card 4
device 0
}
# The 2 S/PDIF Ins and 2 S/PDIF Outs of EMU 1212m,they have S16_LE
pcm.emu_spdif {
type hw
card 4
device 1
}
# The 16 DSP out channels of EMU 1212m, they have S16_LE
pcm.emu_dsp {
type hw
card 4
device 3
}
# The M-Audio Delta 1010 LT, conversion from S32_LE to S16_LE
pcm.delta {
type linear
slave {
pcm "hw:1"
format S32_LE
}
}
# Multi capture device including EMU ADAT, Analog and S/PDIF channels and
# Delta analog and S/PDIF in channels
pcm.multi_capture {
type multi
slaves.a {
pcm "delta"
channels 12
}
slaves.b {
pcm "emu_analog"
channels 2
}
#slaves.c {
# pcm "emu_spdif"
# channels 2
#}
#slaves.d {
# pcm "emu_adat"
# channels 8
#}
bindings.0.slave a
bindings.0.channel 0
bindings.1.slave a
bindings.1.channel 1
bindings.2.slave a
bindings.2.channel 2
bindings.3.slave a
bindings.3.channel 3
bindings.4.slave a
bindings.4.channel 4
bindings.5.slave a
bindings.5.channel 5
bindings.6.slave a
bindings.6.channel 6
bindings.7.slave a
bindings.7.channel 7
bindings.8.slave a
bindings.8.channel 8
bindings.9.slave a
bindings.9.channel 9
bindings.10.slave b
bindings.10.channel 0
bindings.11.slave b
bindings.11.channel 1
#bindings.12.slave c
#bindings.12.channel 0
#bindings.13.slave c
#bindings.13.channel 1
#bindings.14.slave d
#bindings.14.channel 0
#bindings.15.slave d
#bindings.15.channel 1
#bindings.16.slave d
#bindings.16.channel 2
#bindings.17.slave d
#bindings.17.channel 3
#bindings.18.slave d
#bindings.18.channel 4
#bindings.19.slave d
#bindings.19.channel 5
#bindings.20.slave d
#bindings.20.channel 6
#bindings.21.slave d
#bindings.21.channel 7
}
# Control device for multi_capture, I chose card 1 (Dela), because it gives
# the clock signal.
ctl.multi_capture {
type hw
card 1
}
# Multi playback for EMU 1212m Analog, S/PDIF and DSP out channels as well as
# Delta 1010 LT's analog and S/PDIF out channels
pcm.multi_playback {
type multi
slaves.a {
pcm "delta"
channels 10
}
slaves.b {
pcm "emu_analog"
channels 2
}
slaves.c {
pcm "emu_spdif"
channels 2
}
#slaves.d {
# pcm "emu_dsp"
# channels 16
#}
bindings.0.slave a
bindings.0.channel 0
bindings.1.slave a
bindings.1.channel 1
bindings.2.slave a
bindings.2.channel 2
bindings.3.slave a
bindings.3.channel 3
bindings.4.slave a
bindings.4.channel 4
bindings.5.slave a
bindings.5.channel 5
bindings.6.slave a
bindings.6.channel 6
bindings.7.slave a
bindings.7.channel 7
bindings.8.slave a
bindings.8.channel 8
bindings.9.slave a
bindings.9.channel 9
bindings.10.slave b
bindings.10.channel 0
bindings.11.slave b
bindings.11.channel 1
bindings.12.slave c
bindings.12.channel 0
bindings.13.slave c
bindings.13.channel 1
#bindings.14.slave d
#bindings.14.channel 0
#bindings.15.slave d
#bindings.15.channel 1
#bindings.16.slave d
#bindings.16.channel 2
#bindings.17.slave d
#bindings.17.channel 3
#bindings.18.slave d
#bindings.18.channel 4
#bindings.19.slave d
#bindings.19.channel 5
#bindings.20.slave d
#bindings.20.channel 6
#bindings.21.slave d
#bindings.21.channel 7
#bindings.22.slave d
#bindings.22.channel 8
#bindings.23.slave d
#bindings.23.channel 9
#bindings.24.slave d
#bindings.24.channel 10
#bindings.25.slave d
#bindings.25.channel 11
#bindings.26.slave d
#bindings.26.channel 12
#bindings.27.slave d
#bindings.27.channel 13
#bindings.28.slave d
#bindings.28.channel 14
#bindings.29.slave d
#bindings.29.channel 15
}
# Control device for multi_playback, I chose card 1 (Delta) again as it is
# the clock source
ctl.multi_playback {
type hw
card 1
}
Any idea anyone? Could it be a specific configuration issue of the EMU 1212m
or is there a special kernel option - not directly related to sound -, which
might help?
Warm regards and thanks for any hints
Julien
----------------------------------------
http://juliencoder.de/nama/music.html
20 Apr
20 Apr
6:01 p.m.
On Sat, April 20, 2013 10:06 am, Julien Claassen wrote:
Hello everyone!
I've still got problems joining my twosoundcards (Delta 1010LT and EMU
1212m). I've created the multi device for capture and playback, taken care
of
the bit width conversion and still it doesn't work. See the .asoundrc
below.
With Delta and EMU's analog I/O (device 0) JACK will start, but gives a
lot
of ALSA_PCM xruns. With the S/PDIF and/or ADAT/DSP part of the EMU card
included JACK won't start or will shutdown immediately. Also
ecasound/arecord
won't work.
Having tried the EMU S/PDIF on its own with arecord, I found, that it
would
only allow 8kHz samplerate, instead of the requested 48kHz. The EMU card
is
however setup to take the clocksource from the delta card, which works
perfectly with 48kHz. OK, even it takes its clock from the outside (Lindy
A/D
converter).
To test the S/PDIF in on the EMU I tried playing back something to the
Delta's S/PDIF out, but only got silence.
So it looks, like there is more than one problem. But I can't spot it.
I've
looked at the .asoundrc documentation again and the list of ALSA plugins.
OK,
there is the route plugin for creating interleaved audio, using the
ttable,
but for a start, I've read, that this will cause great latency. Anyway,
I've
built my JACK version myself and I don't see, why it shouldn't include
complex
The first thing is that the problems with the 1212 should be looked at
separately. Also, there have been threads about pcm-multi on this list
before even with two "identical cards". Also, as far as I know, pcm-multi
and RT don't play well and even with a more relaxed kernel there are xruns
though they may not affect sound. Using alsa-in/out or zita-a2j and j2a
seem to have worked better in that case and may help in yours as well. Set
jack up as normal with the 1010 and add in the 1212 with alsa-in/out or
zita-a2j/j2a. (the zita package seems to use less CPU and is higher
quality... I am told, I haven't noticed one being bad) In this way it
should be easier to trouble shoot the 1212 problems because pcm-multi is
not involved and syncing is not involved either. So you should be able to
use the 1010 as a spdif input without interaction of the two kernel
drivers.
I'll think more... maybe look back in the LAU archives for other
pcm-multi/dual soundcard threads.
--
Len Ovens
www.OvenWerks.net
6:10 p.m.
Hello Hartmut and Len!
Thanks both of you for suggesting this. I've tried alsa_in/alsa_out. It
partly works. I could include the analog, s/pdif and adat in and the analog
out. but no s/pdif and dsp out with alsa_io. I'll give zita-a-j2/j2a a try.
Warm regards
Julien
----------------------------------------
http://juliencoder.de/nama/music.html
6:50 p.m.
Hello again!
So zita-a2j isi working with all but emU's S/PDIF. I can spare that for the
time being.
But one more problem: The analog I/O is on one device - in my case hw:4,0.
Of course I need this in both directions. what's the trick to get bot in and
out in JACK?
Warm regards
Julien
----------------------------------------
http://juliencoder.de/nama/music.html
7:12 p.m.
On Sat, April 20, 2013 11:50 am, Julien Claassen wrote:
Hello again!
So zita-a2j isi working with all but emU's S/PDIF. I can spare that for
the
time being.
But one more problem: The analog I/O is on one device - in my case
hw:4,0.
Of course I need this in both directions. what's the trick to get bot in
and
out in JACK?
I thought jack had the 1010 on it... However if you mean using zita-i/o,
you can run more than one instance of zita and you can use the same device
for two of them. For example zita-a2j with hw:1,0 and zita-j2a with
hw:1,0. One for inputs and one for outputs (at least it worked for me with
my ensoniq 1370 and my internal ac97 IF). Jack sees one card for inputs
and one for outputs or one for both. You can run as many zita-a2j
instances as you have audio IFs. You can tell each instance what you want
it to call itself on the jack graph too to keep from confusion.
So, if you have hw:4,0 with both input and outputs, that is two zita
instances. Then if you also have hw:4,1 that is only inputs, that can be
another instance and hw:4,2 could be yet another.
Easier to do than explain.
--
Len Ovens
www.OvenWerks.net
7:39 p.m.
Hello Len!
Thanks for the info. I knew about the setup, but it seems, that oe piece of
information got lost in all those messages. Sorry!
It was just, tht for capture, the period size had to be a little bigger. So
now, I just have to test it in practise.
Warm regards
Julien
----------------------------------------
http://juliencoder.de/nama/music.html
6:37 p.m.
On Sat, Apr 20, 2013 at 11:01 AM, Len Ovens <len(a)ovenwerks.net> wrote:
On Sat, April 20, 2013 10:06 am, Julien Claassen wrote:
The main issue with the multi device is that it doesn't actually sync
up the polling of the two sound cards. The multi device's polling
file descriptor is that of the master sound card registered with the
multi device. When poll() returns, indicating that data is available,
it's because there's data available on the master sound card; however,
there is no guarantee that data is also available on the slave
devices. The barrage of xruns that is seen when using the multi
device is caused by this issue. A period's worth of data will be
available on the master sound card, but one or more slave devices will
have less than a period's worth of data. JACK interprets this
situation as an xrun.
Unfortunately, the problem goes a bit deeper, because there
(currently) isn't any call in the ALSA API to make the polling of one
sound card depend on data being available on sound cards that are
linked as slaves. Theoretically, snd_pcm_link(), which links the
start, stop, drain, pause, suspend, resume, reset, and prepare
operations of two cards, could be modified to also link the polling of
data.
Hello everyone!
I've still got problems joining my twosoundcards (Delta 1010LT and EMU
1212m). I've created the multi device for capture and playback, taken care
of
the bit width conversion and still it doesn't work. See the .asoundrc
below.
Also, there have been threads about pcm-multi on this list
before even with two "identical cards". Also, as far as I know, pcm-multi
and RT don't play well and even with a more relaxed kernel there are xruns
though they may not affect sound. Using alsa-in/out or zita-a2j and j2a
seem to have worked better in that case and may help in yours as well.
I've found that the alsa-in/out clients are terrible. The zita
clients perform very well.
--
Devin Anderson
surfacepatterns (at) gmail (dot) com
blog - http://surfacepatterns.blogspot.com/
midisnoop - http://midisnoop.googlecode.com/
psinsights - http://psinsights.googlecode.com/
synthclone - http://synthclone.googlecode.com/
4620
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6 comments
3 participants
participants (3)
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Devin Anderson -
Julien Claassen -
Len Ovens