Hi,
I've received some info from Emiliano (thanks again!) which I think may be
useful to other users here (Marc?), so I'm now forwarding it.
Cheers,
Alex
De: Emiliano Grilli <emiliano.grilli(a)poste.it>
Responder a: Emiliano Grilli <emiliano.grilli(a)poste.it>
Enviado el: jueves, 14 de octubre de 2004 20:49:09
Para: Alejandro Lopez <alex_osiris(a)hotmail.com>
Asunto: Re: [linux-audio-user] Demudi looking good, soundcard advice?
giovedì, 14 ottobre 2004 alle 13:42:00, Alejandro Lopez ha scritto:
>I'm now printing your email so I have your references handy when eventually
>browsing or going to the shop. Many thanks Emiliano, that's really useful
>information.
Thanx ;)
And note that these are only my opinions (based on my limited experience, I
have an staudio dsp2000 - the only non-working thing is the MIDI in - it is
a known bug but still unresolved; it has however 2 working MIDI outs and
phantom power on first two audio channels) - I can say that soundcards
based on the envy24 chip (all those I listed are) are probably the best
entry level "pro" interfaces working good on linux, for doing an additional
step you have to go to RME hw, which is more expensive.
Consider "consumer grade" soundcards (like soundblasters) only if you do
not plan to record from analog input, because the most weak part of these
cards is the ADC and the analog circuitry, but they are "honest" for
monitoring purposes (eg doing anything inside the pc). For example, if you
synthesize a sound with csound, the resulting file will be the same
regardless of what soundcard you have, you can even do that *without* any
soundcard ;)
Also, if you plan to buy a "pro level" card, bear in mind that is very
important to feed it with good external audio gear (particularly, a good
mixer and good microphones)
_________________________________________________________________
Moda para esta temporada. Ponte al dÃa de todas las tendencias.
http://www.msn.es/Mujer/moda/default.asp
>> Just a quick question.... Did you compile your kernel with the
CS46XX
>> as a module? This may be a kernel issue rather than a sound issue.
>> Stephen.
>
>> > I get the error:
>> > /lib/modules/2.4.20-8/kernel/sound/pci/cs46xx/snd-cs46xx.o:
init_module:
>> > No such device
>
>How do I find out how cs46xx was compiled?
In this case, it's quite simple: you look at the version of your kernel.
ALSA was not merged into the kernel until early in the 2.5 development
series. You are using a 2.4 kernel, therefore there is no way for it to
be compiled any other way than as a module.
-Reuben
I'd like to try openMosix (openMosix.sf.net), which requires a kernel
patch. Right now, I'm using 2.4.22 with low latency and preempt
patches. Will I be able to use openMosix with the other patches? Has
anyone tried this? Does it matter what order I apply the patches in?
I'm not sure I even need preemption (I read about using it with the
low latency patch once, but I can't remember where), but I'd like to
have at least openMosix + low latency if I can.
Phil Carter
__________________________________
Do you Yahoo!?
Yahoo! Mail - You care about security. So do we.
http://promotions.yahoo.com/new_mail
Hi,
I have this problem when using xmms-jack output plugin in xmms..
If I run it as a normal user, I get usually "Couldn't open audio..." error box,
but when running as root it works without problem. Could this be a
permission issue or something?
jack-rack, alsaplayer and ams don't have this kind of problem at all.
jackd runs in realtime -mode, kernel is 2.6.9-rc2-mm4-VP-S7
xmms 1.2.10, xmms-jack 0.10 (0.9 has same issue, tested it too..)
Here is some debugging output:
xmms (xmms-jack verbose and trace mode enabled):
[...]
JACK_GetVolumeForChannel:deviceID(0), returning volume of 25 for channel 0
JACK_GetVolumeForChannel:deviceID(0), returning volume of 25 for channel 1
JACK_OpenEx:bits_per_channel=16 rate=44100, input_channels=2,
output_channels=2
JACK_OpenEx:jack_port_name = NULL
JACK_ResetFromThis:resetting this->deviceID(0)
JACK_OpenEx:bytes_per_output_frame == 4
JACK_OpenEx:bytes_per_input_frame == 4
JACK_OpenDevice:creating jack client and setting up callbacks
JACK_OpenDevice:client name 'bio2jack_0_7504'
JACK_OpenDevice:setting up jack callbacks
JACK_srate:the sample rate is now 48000/sec
JACK_OpenDevice:engine sample rate: 48000
JACK_OpenDevice:creating output ports
JACK_OpenDevice:port 0 is named 'out_0'
JACK_OpenDevice:port 1 is named 'out_1'
JACK_bufsize:the maximum buffer size is now 256 frames
JACK_bufsize:expanding buffer from this->buffer_size == 0, to 1024
JACK_bufsize:called
JACK_OpenDevice:calling jack_activate()
ERR: JACK_OpenDevice:cannot activate client
JACK_OpenEx:error opening jack device
JACK_GetVolumeForChannel:deviceID(0), returning volume of 25 for channel 0
JACK_GetVolumeForChannel:deviceID(0), returning volume of 25 for channel 1
jackd verbose mode:
[...]
load = 0.3697 max usecs: 13.000, spare = 5320.000
new client: bio2jack_0_7504, id = 3 type 2 @ 0xb6555000 fd = 20
23:07:43.989 Audio connection graph change.
registered port bio2jack_0_7504:out_0, offset = 3072
registered port bio2jack_0_7504:out_1, offset = 4096
gave capabilities to process 7504
23:07:44.152 Audio connection change.
load = 0.3818 max usecs: 21.000, spare = 5312.000
Any help would be appreciated. :)
Thanks.
--jussi
Hi,
I'm new to linux sound (but not to sound or linux)
So I'm starting off on a journey and need some help.
I have a dman pci card with cs46xx chipset.
I'm using Redhat 9, when my machine booted it detected the card and
configured it.
However, all is not right.
there are lots of crackles and pops when the volume is up.
It can play wavs but it seems the volume needs to be quite high.
I have downloaded the alsa tarballs (driver, lib, utils)
what should be my first step in trouble shooting this setup
thanks,
kind regards,
Luke
--
========================
Luke (Terry) Vanderfluit
Mobile: 0421 276 282
========================
Chaps,
Just installed Demudi, sorry I can't remember versions, it's some beta that
came out something like 2-3 months ago which was the first to merge Debian +
music studio on the same CD (requiring one installation only).
Generally, I would recommend the distribution for an easy install (I've not
used it yet), I'd like to make a couple of comments in case anyone is
interested.
-My PC won't boot from CD-ROM. By browsing the directory structure, I kind
of guessed I had at least a couple of choices for DOS: booting from a
rawrite2'd floppy or run loadlin. Fair enough, there's a vmlinuz file there
as well as several .bin floppy images. But: loadlin is not included in the
CD image, not it is rawrite2. Not a big deal for me as I had several other
distros handy but not elegant either. In the same directory where these
images reside, there's a little text file which lists loadlin and rawrite2:
should read "loadlin", reads "lodlin" (in case anyone would like report this
for future versions).
-I was not able to select English language with a Spanish keyboard layout
(I'm Spanish). You are supposed to select both in one go, the 2 different
options in the menu (that you can see if you select back somewhere in the
middle of the process) seem to be related and didn't work as such distinct
options for me.
-At one point, I went back and selected Spanish (for everything, that is),
next step was repartitioning my hard disk which is inherently a dangerous
operation. The translation of the partitioning software to Spanish was so
poor (wrong actually) that I was scared to repartition in Spanish. Went back
to English so I could understand what the software would do with my hard
disk but now my keyboard layout doesn't match. I'm planning installing to a
different machine I imported from Germany, so I'll have the same layout
problem (I'd rather not choose German language to repartition my disk in
order to have the keyboard properly installed).
-From there on, the installation was as smooth as one could have asked for.
All my hardware was detected and installed automagically including graphics
card, monitor, mouse, sound card (ALSA), network card, modem, CD-ROM.
So generally I'd say it's a very easy to install distribution, and
definitely a huge improvement over the classic "2 installation steps"
approach that all the other music distributions have or used to have. I'm
very pleased with regards to that.
The only other thing is I may need to buy a decent soundcard for this PC. It
should have a GM synth (nothing special since I guess I'll be using soft
synths anyway but I don't want to run a synth just to write a couple of
arrangements), one stereo output and one stereo input both with good overall
audio quality. Seemingly, the current trend is towards either USB 2.0,
firewire or PCI. A colleague has just told me that USB 2.0 is supported by
the Linux kernel starting from the latest version (2.6 I think?) only. Since
music distributions are based on other distributions, chances are that they
now run the 2.4 kernel or maybe older. Does that make sense or am I talking
complete b*ll*cks? Also, this friend has heard about drops happening with
audio over USB, but apparently this was on a Mac and the USB device was a
hard disk rather than a soundcard. Still, he seems to think that USB 2.0 is
not as good option as firewire for audio. (Which reminds me of SCSI vs IDE
drives for audio a few years back, yes SCSI was the serious option for a
couple of years but it was an 80% more expensive as well, and shortly
afterwards IDE started to go "fast enough" and cheaper.) Is firewire better?
Is it more expensive? Also, how does it compare to PCI? Lastly, if any of
you has bought a soundcard (recently so the card is still in production)
which is reasonably similar to what I need (wouldn't mind if it's slightly
better, say 4 mono inputs and 4 mono outputs or something) and has succeed
with having it running under a Linux based music studio, I'd be grateful if
you drop a line.
Many thanks!!
_________________________________________________________________
Descarga gratis la Barra de Herramientas de MSN
http://www.msn.es/usuario/busqueda/barra?XAPID=2031&DI=1055&SU=http%3A//www…
It's time to ask for advice. :-)
I'm playing a show next weekend with a rock band. I'm using a MIDI
controller and a Roland Sound Canvas for almost everything, but there
are two songs that need typical "Moog" sounds... highly distorted and
heavy portamento (think Weezer or The Rentals). This the Sound Canvas
cannot do (AFAIK).
Without buying an old Moog, what are my options, and what do you
suggest... looking for cheap, and flawless on-stage?
Ideas:
1. Look for a cheap used MIDI sound module with great portamento. I've
never seen one.
2. Alsa Modular Synth. I really like the portamento effect, but I've
yet to find any great moog sounds, and I'm not good at making my own.
It's quite stable, but gives me "clicky" artifacts sometimes.
3. ZynAddSubFX. Has some great synth sounds, but the portamento is too
"over the top". Over dramatic.
4. VST. I have some *amazing* Moogy VST plugins, but no idea how to
use them with linux. I'm aware of the wine/vst libraries, but though
they were mainly for effects, not instruments.
5. Better ideas are welcome. Remember that simple and reliable is more
important than sound.
Thanks,
Austin
--
Austin <aacton(a)yorku.ca>
After searching and reading nearly everthing about Delta 1010LTs and
Alsa, i have still no clue how to solve my problem. I also posted on the
Alsa mailinglist, but got no answer.
I bought two Delta 1010LTs to do 16 channel multitrack recording with
ardour and jack. And i have no problem doing so with either the first or
the second one alone, despite regular messages of the form:
"load = 0.2046 max usecs: 79.000, spare = 23140.000"
My system is an Athlon TB 900 MHz, 512 MB RAM, both Deltas have a
interrupt on there own (IRQ7 and IRQ5). At the moment it's just plain
Xorg with twm.
I use kernel 2.6.8-r3-gentoo with kernel ALSA driver and preemp and
volutary
preemp patch. ALSA-lib, -utils and -tools have version 1.06.
In the meantime i tried a recent 2.4 kernel with realtime patches and
Alsa driver 1.06 with different problems.
PROBLEM 1: .asoundrc
-------
Since Jack only supports one sound card, i tried to create a virtual 16
channel card through the .asoundrc file (appended at the end of this post).
The Delta has 12 capture and 10 playback devices, so i created two multi
devices, one for capture and one for playback and an asym device to put
them together and some magic for jack. This is the only configuration
that works a little.
Is this the right configuration?
Bevor jack accepts this card, the deltas had to be synced, either
through S/PDIF or Wordclock (i tried both).
Btw. is there a kernel sync like under Windows? Because, when i open two
envy24control windows (one for each card) and change the clock rate in
one envy24control the other changes as well when reset is selected. And
what's the meaning of "reset" and "locked"?
PROBLEM 2: kernel 2.6
-------
Without Syncing, jackd sometimes locks the entire system.
When the cards are synced and i start jack there is an error message:
"Alsa lib pcm.c: 1178: (snd_pcm_link) SNDRV_PCM_IOCTL_LINK failed:
Operation already in progress"
It seems to have no effect, though.
Then i got a lot of XRUN callbacks. With jack Soft Mode i got "load =
0.2046 max usecs: 79.000, spare = 23140.000".
I can work with it, but it shouldn'd be normal?
If this where all, i won't have bothered you.
Using the 16 channel virtual card with ardour works for the first 8
channels.
The second 8 channels of the second card, which where synced to the
first card, produces weird sound, both while recording and playback.
It would be nice if someone can help me. Thanks in advance.
PROBLEM 3: kernel 2.4
-------
Since i get no answer on the above question on the Alsa list, i changed
the kernel back to 2.4 and the external ALSA driver 1.06.
The configuration (hardware and .asoundrc) remained the same.
When i start jackd with playback only, this works great now. The weird
sound is no longer.
With capture too, i get "jackd: pcm.c:5957: snd_pcm_mmap_commit:
Assertion 'frames <= snd_pcm_mmap_avail(pcm)' failed"
Sometime with a few
"load = 2.3689 max usecs: 550.000, spare = 11059.000"
before.
Somewhere i read, that this is about unsyncronisation? With kernel 2.6,
but without syncing, i got the same messages when using capture devices.
Sorry for the big post and the many questions.
Greetings
Thomas Wehrspann
---.asoundrc---
###########################
# M-AUDIO DELTA 1010LT(1) #
###########################
pcm.ice1712_1 {
type hw
card 0
}
ctl.ice1712_1 {
type hw
card 0
}
###########################
# M-AUDIO DELTA 1010LT(2) #
###########################
pcm.ice1712_2 {
type hw
card 1
}
ctl.ice1712_2 {
type hw
card 1
}
###########################
# VIRTUAL 16 CHANNEL #
###########################
pcm.delta8in {
type multi
slaves.a.pcm ice1712_1
slaves.a.channels 12
slaves.b.pcm ice1712_2
slaves.b.channels 12
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 b
bindings.8.channel 0
bindings.9.slave b
bindings.9.channel 1
bindings.10.slave b
bindings.10.channel 2
bindings.11.slave b
bindings.11.channel 3
bindings.12.slave b
bindings.12.channel 4
bindings.13.slave b
bindings.13.channel 5
bindings.14.slave b
bindings.14.channel 6
bindings.15.slave b
bindings.15.channel 7
}
pcm.delta8out {
type multi
slaves.a.pcm ice1712_1
slaves.a.channels 10
slaves.b.pcm ice1712_2
slaves.b.channels 10
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 b
bindings.8.channel 0
bindings.9.slave b
bindings.9.channel 1
bindings.10.slave b
bindings.10.channel 2
bindings.11.slave b
bindings.11.channel 3
bindings.12.slave b
bindings.12.channel 4
bindings.13.slave b
bindings.13.channel 5
bindings.14.slave b
bindings.14.channel 6
bindings.15.slave b
bindings.15.channel 7
}
pcm.delta {
type asym
playback.pcm delta8out
capture.pcm delta8in
}
pcm.jack {
type route
slave.pcm delta
ttable.0.0 1
ttable.1.1 1
ttable.2.2 1
ttable.3.3 1
ttable.4.4 1
ttable.5.5 1
ttable.6.6 1
ttable.7.7 1
ttable.8.8 1
ttable.9.9 1
ttable.10.10 1
ttable.11.11 1
ttable.12.12 1
ttable.13.13 1
ttable.14.14 1
ttable.15.15 1
}
ctl.jack {
type hw
card 0
}
Hi,
I read a lot about having problems with latencies, interrupts on some
motherboards under Linux. I originally asking for use in Asterisk PBX SW
system under Linux, but since it's almost same low latency sensible
application, I thought I'd ask also linux sound gurus - that have sometimes
similar problems.
Could you please be so kind to recomend me brand or type of motherboards
that are working flawlessly for low latency applications under Linux
(Gigabyte, Asus, Intel, some
other...) ?
Is there any site about PC motherboards particularly suitable for low
latency applications under Linux ?
Thanks in advance,
regards,
Robert.
Hi,
I am new to this list and new to music making with linux, so I don't yet
know much about alsa, jack, ladspa etc.
I have a dman pci card that I would like to use. I've searched the net
and found that there should be a driver for this card, the cs46xx.
the card has the crystal cs4614-cm ep chipset.
Does anyone know of a driver for this card?
thanks,
kind regards,
Luke
--
========================
Luke (Terry) Vanderfluit
Mobile: 0421 276 282
========================
