Hello everyone, my first mail here.
I'm having problems with the ALSA surround sound plugins on Gentoo
with alsa-plugins compiled and installed and have the following
content in ~/.asoundrc -
pcm.!default {
type plug
slave.pcm "surround51"
}
# upmixing:
pcm.!stereo {
type route
slave.pcm "su
Hi!
I'm facing problems with my system, so decided to learn ALSA to see
what's the problem.
My first question -- what's a device and a sub device?
A device seems like the output streams the sound card supports. Like
HDMI, DP, analog etc...
So what's a subdevice? Is this related to multichanne
ances in a
week and would
love to be using my new alsa dependent applications.
FIY here attached is my .asoundrc, my alsa.conf, /etc/modutils/aliases.
thankyou very kindly,
de|ire
melbourne australia
http://selectparks.net
--
, variab
thanks for your swift reply,
at approximately Fri, 01 Nov 2002 17:51:31 +0100, the X-mailer used by
Adam Jones <[EMAIL PROTECTED]> suggested that
"yam9070.1590.120533200@proserpine" is somehow related to:
> | #O#n 31-Oct-02, de|ire wrote:
> |
> | &
normally. only, i
have to do this after each restart.
thanks for assisting unravelling this mystery!!
de/
at approximately Fri, 01 Nov 2002 10:06:47 -0600, the X-mailer used by
"j.c.w." <[EMAIL PROTECTED]> suggested that "[EMAIL PROTECTED]" is somehow
related
reply.
yet funnily enough ps ax reveals no 'alsasound'? is this because it is not a *process*
so much as a driver?
de/
--
, variables aren't ___ ___ ___
/ / /
/
audio.
this however has proved to be unreliable, and makes for a slow start
into my xsession.
no one seems to be able to solve this problem, here or in any
lists, so it seems it is not the right time for me to be trying alsa.
thanks for your help,
de/
melbourne,
australia
http://selectparks.ne
hi there,
no i am not using devfs, though i was.
my fs is static only.
cheers,
de/
at approximately Sun, 3 Nov 2002 20:30:19 -0500, the X-mailer used by
James Drabb <[EMAIL PROTECTED]> suggested that
"[EMAIL PROTECTED]" is
somehow related to:
> | On Sun, Nov 03,
re is still no explanation why alsa won't 'stick' on my machine.
FYI i know a very experienced linux audio developer who uses the same
card [ maestro3 ]. while he has had different symptoms, they have been
severe enough that he cannot use alsa.
thanks for your time,
de/
at appro
und with execing all applications su, but this doesn't change the
outcome either.
thanks for your assistance,
de/
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hi list,
is there a way to import stereo samples
straight into the multitracker without having to split them up into two
mono tracks first? this would be laborious given the number of
pre-existing stero samples i like to use in my compositions.
!excuse my ardour naivette,
having fun,
julian oli
Hi,
I'm experiencing a little problem with the S/PDIF output of my SBLive
(I use ALSA 0.9.0rc6).
Sometimes when I send my S/PDIF data to the device I can hear some
chunks twice but according to the status the buffer isn't empty:
snd_pcm_status_get_delay() always returns something around 1 fra
r 4 in libasound.so.2.0.0[f4bad000+ab000]
You can find the output of alsa-info.sh here: http://pastebin.ca/1177336
Thanks for help.
De Ganseman Amaury
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Hi,
I have a 24-bit/96KHz file.
When I play it, it says that it plays in s16_le and samplerate ouput
is 48KHz (I see that on my hardware equaliser).
I have an ESI Juli@ and use the alsa driver of the 2.6.26.
output of mplayer:
Playing Radiohead - 06 - Karma Police.flac.
Audio only file format de
20 PM, Vladimir Mosgalin
<[EMAIL PROTECTED]> wrote:
> Hi Amaury De Ganseman!
>
> On 2008.09.03 at 13:16:42 +0200, Amaury De Ganseman wrote next:
>
>> I have a 24-bit/96KHz file.
>> When I play it, it says that it plays in s16_le and samplerate ouput
>> is 48KHz (I
Hi,
Is there anyone with a similar problem? I have read several
similar discussions on the web, but i failed to arrive at a
solution.
Here's the complete details after running the script from
http://git.alsa-project.org/?p=alsa-driver.git;a=blob_plain;f=utils/alsa-info.sh
http://www.alsa-projec
I give up, I have tried everything that I can find on this topic without any
success.
Getting audio out of HDMI was relatively simple but getting anything more
than stereo is proving challenging. I have upgraded my Alsa
dirvers/libs/utils to the latest release v1.0.21 so I have the latest NVidia
H
On Thu, Oct 1, 2009 at 10:58 AM, VDR User wrote:
> On Wed, Sep 30, 2009 at 2:46 PM, Mark de Reeper
> wrote:
> > I give up, I have tried everything that I can find on this topic without
> any
> > success.
> >
> > Getting audio out of HDMI was relatively simple b
On Thu, Oct 1, 2009 at 7:09 PM, VDR User wrote:
> On Wed, Sep 30, 2009 at 4:10 PM, Mark de Reeper
> wrote:
> > OK so it's not just me, that makes me feel a little bit better. Has this
> > been logged as a bug, happy to do so if it hasn't.
>
> I don't think
On Fri, Oct 2, 2009 at 9:03 PM, Marco Gusy wrote:
> In data mercoledì 30 settembre 2009 23:46:31, Mark de Reeper ha scritto:
> : > speaker-test -c6 -Dhdmi
>
> what happens if you use
> speaker-test -c6 -f S16_LE -r 48000 -Dhw:0,0
> ?
>
> In my MCP78 i managed to have 7
Hello,
An updated version of alsamixergui is available from
http://www.iua.upf.es/~mdeboer/projects/
alsamixergui is a FLTK based frontend for alsamixer. It is written
directly on top of the alsamixer source, leaving the original source
intact, only adding a couple of ifdefs, and some calls to
"aconnectgui is a FLTK based frontend for aconnect. It is written
directly on top of the aconnect source, leaving the original source
intact, only adding a couple of ifdefs, and some calls to the gui
part, so it provides exactly the same functionality, but with a
graphical userinterface."
http://
yes, that sounds like a layout problem. i will try to get my hands on
a SBLive, and also use some virtual midi ports. as you might have noticed,
this release is still rather beta: lot's of printf to be removed ;-)
maarten
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Hello,
Sorry for being such a pain, but I really want this to work, and I
really think this is something that has general interest: using a
higher accurate midi event timing. So: HOW DO I USE THE RTC TIMER?
I can find no up-to-date documentation about this.
The latest I find is a mail by Takashi
On Mon, 8 Apr 2002 10:15:49 +0200 (CEST)
Jaroslav Kysela <[EMAIL PROTECTED]> wrote:
> The defines are in asound.h file - look for 'Timer section - /dev/snd/timer'.
>
> Usage (using defaults):
>
> options snd-seq snd_seq_default_timer_device=1
>
> or (full specification):
>
> optio
Hello,
I just updated alsamixergui and aconnectgui. The code is now up to date
with alsa-utils 0.9.0rc1.
alsamixergui and aconnectgui are fltk front-ends for alsamixer
and aconnect. thet are written directly on top of the alsamixer and
aconnect source, leaving the original source intact, only a
Hello,
Is anyone working on a driver for this soundcard ?
It is an "Antex StudioCard", balanced audio in & out etc...
Greetz,
Yves
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Hi, completely new to linux, so I got RH8.0 and used the Planet CCRMA at Home
to get sound support with ALSA and sound applications.
Here a some (probably silly) questions:
- Now I can get some sound out of it, but how do you access all the channels?
The only thing that works is ADAT1 channels
So I got my rme9652 withasla 0.9.0rc7 running. Got Jack on top of that. Can
route around sound with qjackconnect and can see there is activity with
meterbridge.
May I conclude that my setup is OK?
Playback:
I tried alsaplayer with some ripped cd tracks (used cdparanoia). It always
says the f
Hello all,
(first mail on this list, I'm new here!)
I'm trying to do some funky things with ALSA/PulseAudio/Bluetooth on a
raspberry PI, and I already have many things working. However, I would
really like to use some LADSPA plugins as pcm devices/plugs in alsa.
Following the many guides online
does seem that the plugin is correctly applying a gain increase
- bram
On Thu, Jun 13, 2013 at 4:42 PM, Bram de Jong wrote:
> Hello all,
>
>
> (first mail on this list, I'm new here!)
>
> I'm trying to do some funky things with ALSA/PulseAudio/Bluetooth on a
> r
/.asoundrc"
in /usr/share/alsa/alsa.conf
fixed the problem.
- bram
On Thu, Jun 13, 2013 at 4:59 PM, Bram de Jong wrote:
> As a follow-up on this I wanted to make sure that the plugins really
> work (i.e. work properly on the raspberry) and it does seem so:
>
> $ applyplugin
Hi all,
If you have a "type ladspa" pcm device, is there a way to control the
parameters of this plugin via the command line?
- bram
--
http://www.samplesumo.com
http://www.freesound.org
http://www.smartelectronix.com
http://www.musicdsp.org
office: +32 (0) 9 335 59 25
mobile: +32 (0) 484 154
On Fri, Jun 14, 2013 at 12:28 PM, Clemens Ladisch
wrote:
> Bram de Jong wrote:
>> If you have a "type ladspa" pcm device, is there a way to control the
>> parameters of this plugin via the command line?
>
> The ladspa plugin does not allow changing its control val
ri, Jun 14, 2013 at 12:31 PM, Clemens Ladisch
wrote:
> Bram de Jong wrote:
>> I know I can change the values using alsamixer, but I need command
>> line instead...
>
> man amixer
>
>
> Regards,
> Clemens
--
http://www.samplesumo.com
http://www.freesound.org
ht
cs:
Intel i3
C210 chipset
Fedora 19 64 bit, new installation since yesterday
Kernel 3.11
Do not hesitate to ask me further about the specs!
Thank you for reading my mail,
Ruben De Smet
-BEGIN PGP SIGNATURE-
Version: GnuPG v2.0.22 (GNU/Linux)
Comment: Using GnuPG with Thunderb
on)?
What is my best guess?
Thanks a lot!
R
On 10/22/2013 08:45 PM, Ruben De Smet wrote:
> Hi,
>
> I've got a Lenovo U310 with an intel i3 and C210 chipset.
> Everything worked out of the box, except for the internal
> microphone. Not running Winblows, so no chance to te
I tried on Ubuntu 13.10, doesn't work either. Any other suggestion?
R
On 10/27/2013 09:11 PM, Ruben De Smet wrote:
> Anybody an idea? (aka. bump)
>
> I'm sorry to bother you with this, but I'm not sure if it's hardware
> or software related. As I said, can&#x
ollow.
What situations should I test to make it easy for you guys to figure out
a conclusion? And what could be a solution?
R
On 11/13/2013 11:14 PM, Bill Unruh wrote:
> On Wed, 13 Nov 2013, Ruben De Smet wrote:
>
>> I tried on Ubuntu 13.10, doesn't work either. Any other suggestion?
Okay, it now appears that when unplugging en replugging the external
mic, the internal just works.
Any clue how to debug this?
What logs can I supply you?
R
On 11/24/2013 11:49 PM, Ruben De Smet wrote:
> Okay, I tried everything you said. As a last resort, I plugged in an
> extern
compatible.
When I connect USB plug, the device is recognized, but gnome becomes
unresponsive. After a tens of seconds, the system log shows that an crash
message.
I am runging under ubuntu 15.10.
Bellow is a copy of system log.
Any idea? Should I recompile the last alsa driver?
Thank you
Julien de
same USB 2.0 port.
Regards.
Julien
Le Mardi 1 mars 2016 13h59, Clemens Ladisch a
écrit :
julien de ROSNY wrote:
> [ 1201.257672] INFO: task pulseaudio:1992 blocked for more than 120 seconds.
> [ 1201.257727] Call Trace:
> [ 1201.257746] [] schedule+0x29/0x70
> [
n I know that the value of bandwidth that is asked?
Thanks,
M8hpw
Le Mercredi 2 mars 2016 17h42, julien de ROSNY a écrit :
Clement you are right, I try on a usb 2 port instead of a usb 3 one, an this
error is removed.But the usb 32 channel sound card is not really recognized.
But now
.) MxPS= 320 Ivl=4ms
So it is maybe required unbind the HID interfaces?
But i am no at all an expert in USB
Julien
Le Vendredi 4 mars 2016 14h14, julien de ROSNY a écrit :
Ok now on USB 2, the Antelop Orion 32 Channels.I succeed in playing back and
recording some sounds but it is very
Dear ALSA-users,
I have a VIA Epia (type CN1EG) motherboard with a VT8237 sound chip
and a VT1618 mixer.
ALSA plays fine stereo sound, and also the command "aplay -D surround51
chan-id.wav" (which is a 6-channel WAVE file) plays without errors.
Unfortunately I cannot find the mixer control whi
I'm running Slackware 11.0 with my own 2.6.18.1 kernel. When booting
my pc and trying to play music with xmms, I get the message "the sound
device is busy". I have to manually load the modules snd-mixer-oss,
snd-seq-oss and snd-pcm-oss to make my sound work.
When running alsaconf I get "No support
On Friday 19 January 2007 16:28, ronan mcallister wrote:
> Hi Klangmeister,
>
> Thanks for the advice, sorry for the *long* reply.
>
> What Linux/distro do you use?
Using Gentoo with an DIGI5296 and ADI-8 AE. The DIGI96/8 is in another pc at
the moment. No problems unless running "emerge --rsyn
You can use alsaplayer, it is jack-aware. Try "alsaplayer -i gtk -d jack".
Otherwise MhWaveEdit and ReZound are also jack-aware and there is a plug-in
for Xmms.
Regards,
Johan
On Monday 22 January 2007 22:18, ronan mcallister wrote:
> Dominique,
>
> I'm looking into the kernels/distro's you s
After reading your mail again, I noticed a few things more, see in the text.
On Monday 22 January 2007 22:18, ronan mcallister wrote:
> Dominique,
>
> I'm looking into the kernels/distro's you suggested -- in the
> meantime, I did startup meterbridge, when using Ardour I to play a WAV
> file, it'
analysis, etc). Most of the folks I've discussed this
> with use Linux audio tools for creating/modifying/mastering studio music not
> "live" sound.
Never tried to do live sound with it. I'm one of the recording/messing around
types :)
>
> Thank you again,
> Rona
ed to know alot more about
> > digital audio and Linux before even approaching an evaluation of Linux
> > distros, but if you have any thoughts on my current plan (to continue to
use
> > OpenSuse 10.2 + JAD and experiment) -- I'd appreciate it. Until I figure
> > out w
Ronan,
Like Sergei already said, split the topic because it is getting confusing and
has (almost) nothing to do with RME anymore.
Regards,
Johan
On Saturday 27 January 2007 18:32, ronan mcallister wrote:
> Sergei,
>
> For the moment forgetting about the Xover's, how would I use ecasound or
> a
This is becoming a bit off-topic and too technical to be of general interest I
fear.
But I do agree with Sergei that the lower the frequency you want to process,
then more taps you need on your filter. There is no way out. Just the same
for analog filters you need a certain time for the respons
Dear ALSA-users,
I have a VIA Epia (type CN1EG) motherboard with a VT8237 sound chip
and a VT1618 mixer.
ALSA plays fine stereo sound, and also the command "aplay -D surround51
chan-id.wav" (which is a 6-channel WAVE file) plays without errors.
Unfortunately I cannot find the mixer control wh
Doesn't the RME HDSP range support hardware mixing? I don't know how well this
is supported in linux. They would provide everything except your points 3 and
7.
Regards,
Johan
On Saturday 17 February 2007 21:45, Peter wrote:
> Hi,
>
> since the years I put together my last computer, it is easie
2 help me?
Thanks
I was talking about this on alsa-devel and Takeshi invited me to use
alsa-user list. I include here the thread on alsa-devel.:
Xavi de Blas wrote:
>
> 2007/7/9, Takashi Iwai <[EMAIL PROTECTED]>:
> > At Mon, 9 Jul 2007 15:58:30 +0200,
> > Xavi de
It worked!!!
See this:
http://mailman.alsa-project.org/pipermail/alsa-devel/2007-July/001939.html
Thanks to all!
2007/7/9, Xavi de Blas <[EMAIL PROTECTED]>:
> Hello, I installed Wine on my Ubuntu Feisty on IBM-lenovo X60s, and
> after reboot, sound never worked again (sound work
le doesn't turn up much either.
Any idea what is needed?
Thanks,
Johan De Groote
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If you can bring me the light ;-)
Thanks
De Ganseman Amaury
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Ok thanks
Now another question: Which chipset is well supported in alsa and have
a good quality (good SNR: my actual sound card is very noisy...).
I see 2 chipset:: irc17244, CMI8788 (or another good chipset if you know one).
Thanks for infos.
De Ganseman Amaury
On Thu, Feb 21, 2008 at 8:38
Hi,
I have an ESI juli@ and I'm using optical s/pdif. I've had a very bad
sound, to correct it I setup period_time period_size and buffer_size
to correct the problem (as seen in forums).
But I don't undertsand well what is it. If someone could explain to me
or give me a link to documentation abou
With the latest alse-lib release, several audio player (mpg123-alsa,
mpg321, ogg123) started using a huge amount of high cpu to play
anything. As an illustration, with mpg321 I get the numbers like the
following:
before after
real 2m32.188s 2m32.276s
user 0m2.212s0m25.310s
sys
Dear list,
I'm having trouble passing pure 44.1 kHz audio through SPDIF properly.
I'm able to pass it through. But the sound gets sped up to 48 kHz making
it sound as if the pitch is turned up. I know alsa enables me to upmix
all sound to 48 kHz. But that causes too much quality loss. I want it
Bill Unruh wrote:
>>
>> According to Realtek that chip is capable of producing 32, 44.1 and 48
>> kHz sample rates through spdif. So I guess it should be possible? Is
>> ALSA
>
> That may be lie. It may upmix internally-- just as badly as alsa does.
>
>> capable of switching those rates on the f
Mark Knecht wrote:
On Thu, May 1, 2008 at 8:42 AM, Bart de Boer <[EMAIL PROTECTED]> wrote:
Complete output of /proc/asound/card0/codec97#0/ac97#0-0:
0-0/0: Realtek ALC650F
PCI Subsys Vendor: 0x1509
PCI Subsys Device: 0x9202
Capabilities :
DAC reso
Hi all,
I got it to work! :D I was telling MythTV to send everything to my
digital output directly. This was wrong. I needed to send everything to
my analog output and let the chip do the work. I'm now playing music at
44.1 kHz and movies at 48 kHz. :)
Thanx anyway,
Bart
Bart de
Matt Garman wrote:
On Fri, May 02, 2008 at 11:20:28AM +0200, Bart de Boer wrote:
I got it to work! :D I was telling MythTV to send everything to my
digital output directly. This was wrong. I needed to send
everything to my analog output and let the chip do the work. I'm
now playing
Dominique Dumont wrote:
Bart de Boer <[EMAIL PROTECTED]> writes:
It seems I may have cheered too soon. Yes, I am able to get
bitperfect 44.1 kHz sound and bitperfect 48 kHz sound (with and
without AC3). Mplayer is somehow able to trigger the system into 48
kHz after listening to 44
Lee Revell wrote:
On Fri, May 2, 2008 at 5:20 AM, Bart de Boer <[EMAIL PROTECTED]> wrote:
Hi all,
I got it to work! :D I was telling MythTV to send everything to my digital
output directly. This was wrong. I needed to send everything to my analog
output and let the chip do the wor
Dominique Dumont wrote:
Bart de Boer <[EMAIL PROTECTED]> writes:
That sounds plausible. :) I use the via82xx driver. Should I post my
case in the alsa-devel list?
I think so. Try to prove your case by playing with alsamixer.
If anyone knows any workarounds I'd l
Hi all,
Here is the description and workaround to have sound.
It's related to mixer..but I don't know why it saves wrong mixer settings.
Thanks for help
#I have this problem (no sound when I start the system) since I
upgrade to kernel 2.6.26 (driver: ice1724)
#rm asound.state helps but after sto
Hi,
There's a problem when I start to play a file the sound is bad
during 1sec or less (clicking,cracking,...) or when I switch between
tracks I hear clicks.
Any idea ?
Note: It works very well with OSS driver (last mercurial)
Thanks for help.
Here my asoundrc:
defaults.pcm.rate_converter
Hi,
There's a problem, when I start to play a file the sound is bad
during 1sec or less (clicking,cracking,...) or when I switch between
tracks I hear clicks.
Any idea ?
Note: It works very well with OSS driver (last mercurial)
Thanks for help.
Here my asoundrc:
defaults.pcm.rate_converter
What's your problem exactly ?
Because I also have your chipset ALS 650 AC'97, but I can't record when
other people seems to be able to do such things. If you just want to
hear some sound it's easy using via82xx module. If you want to record
... Good luck !
-Message d&#x
Sorry guys, first time it didn't work, but this time I would really like
to unsubscribe.
Your help was really useful, but I don't need more thanks.
'unsubscribe [EMAIL PROTECTED]'
Adrian
_____
Envie de discute
Now that alsa is included in the 2.6.x kernel, how about updates?
Last months, there has been quite some activity with new releases following
each other briskly. How will this happen for the new kernel? Will there be
kernel patches coming out with every new alsa release? Who will make them
avai
Has anybody experience with soundcards with an ADAT I/O that are reported to
work well with Ardour besides the RME range?
I went through the alsa list, but the only models easy found are the RME range
and the Terratec EWS88D (not on terratecs site anymore, but still in shops).
Am I missing some
AFAIK the alsa drivers are included in the kernel distribution, there is no
need to install them afterward. Just enable them during config.
For some other problems with 2.6.0 I was advised to update the modprobe etc,
as the old ones do not work with 2.6.0.
Regards.
PS, glad someone at least go
These are rebadged rme cards. Anyone any experience running them with the rme
alsa drivers?
Regards
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Hello All!
I am using the built in sound of an Asus A7V8X motherboard which
features the ALC650 chip. This generally gets detected as a Via 8235
both in Windows as well as Linux and the good news is that this does not
seem to cause any problems. If I use the ALC driver instead of the VIA
driver in
I have been working with this RME Hammerfall (not the HDSP version) for years
now without trouble. But I would like to try recording 88.2/96 kHz.
I tried to set it using qjackctl, but this fails. Reading on the alsa-poject
site, I saw that in the .asoundrc entry for the card is marked that 88.2
'.>> so when the processor is busy, a
little >> sound is heard and this stays the whole time till the whole
sound has been >> played (might take about 2 mins), so I do hear the
entire sound, but only get >> bits and pieces at a time.>>
>> This sounds :)
exactly
as in Windows.
Can anybody give a hint what to do??
Eeltje de Vries
Further information:
==
/proc/asound/sndstat:
Sound Driver:3.8.2-980706 (ALSA v0.5.12a emulation code)
Kernel: Linux PhatLinux.PhatBox 2.2.15-4mdk #3 Sat Dec 2 07:04:38 EST 2000
i686
nd should
work.
Suse support has only confirmed my request for help
dr. Eeltje de Vries
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>On Sun, 30 Dec 2001 16:11:21 +0100
>"Eeltje de Vries" <[EMAIL PROTECTED]> wrote:
> > In Alsa-users digest, Vol 1 #331 I wrote:
> >
> > >I use Phat Linux 2.2.15-4mdk. My motherboard is AOpen AK33
> > >The sound is integrated on the mothe
Ciao,
i'm still trynig to use an AcerAspire 5715z laptop ( os: Hardy-ubuntu
8.04 - RT )
with a new M-Audio usb soundcard: Fast
Track Ultra -
High-speed 8 x 8 USB 2.0 Interface with MX Core DSP Technology;
http://www.m-audio.com/products/en_us/FastTrackUltra-main.html
but they wouldn't wor
James Shatto ha scritto:
>On Tue, 19 Aug 2008 01:28:46 +0200
>V Gabriele De Palo <[EMAIL PROTECTED]> wrote:
>
>
>
>>Has someone got the Fast Track Ultra from M-Audio working with linux?
>>
>>
>
>I have an M-Audio Mobile Pre working. I
Arthur Marsh ha scritto:
V Gabriele De Palo wrote, on 19/08/08 11:08:
James Shatto ha scritto:
On Tue, 19 Aug 2008 01:28:46 +0200
V Gabriele De Palo <[EMAIL PROTECTED]> wrote:
Has someone got the Fast Track Ultra from M-Audio w
, since usb probably isn't the primary unless all other soundcards are disabled somehow, or you re-indexed it in your alsa configuration. Or it's just not class compliant / supported)
HTH,
James
On Fri, 22 Aug 2008 20:01:56 +0200
V Gabriele De Palo <[EMAIL PROTECTED]> wrote:
James Shatto ha scritto:
It looks like this might be your issue:
La periferica di riproduzione è hw:0
I parametri dello stream sono 48000Hz, S16_LE, 2 canali
Using 16 octaves of pink noise
ALSA lib confmisc.c:1286:(snd_func_refer) Unable to find definition 'defaults.namehint.exten
Maxence Laurent ha scritto:
>Bonjour,
>
>The Fast Track Ultra is not class compliant :-(
>
>A topic on m-audio official forum asks if somebody want m-audio develop
>(or help alsa-devel team to) fast-track-* drivers for linux:
>http://forums.m-audio.com/showthread.php?t=714
>
>I suggest that ever
not tell me that the name of the
> card
> was proprietary.
>
> On Sun, 31 Aug 2008, V Gabriele De Palo wrote:
>
>> Maxence Laurent ha scritto:
>>
>>> Bonjour,
>>>
>>> The Fast Track Ultra is not class compliant :-(
>>>
>>> A topic on
James Shatto ha scritto:
>>Hello everyone! I now resigned to not being able to use
>>the m-audio fast track ULTRA usb soundcard with my "LINUX-DAWs",
>>someone can recommend another card usb I can afford
>>multitrack audio recordings of quality, which is working with Linux?
>>
>>
>
>As said be
James Shatto ha scritto:
>>I need to make multitrack recordings; I' m looking for a sound card
>>usb2 model
>>of at least 4/6/8 balanced inputs, XLR with phantom power to 48V and
>>audio resolution 24-bit/96kHz
>>and with many analog audio outputs maybe XLR balanced, SPDIF in / out
>>and MIDI
Ivan Hernandez ha scritto:
>Hello . I have a notebook i use with a usb midi keyboard for doing
>music. The
>computer has a crappy audio device:
>
>00:10.1 Audio device: nVidia Corporation MCP51 High Definition Audio
>(rev a2)
>
>
>That i managed to play sound with jack without xruns. The problem
Ari Moisio ha scritto:
>Hi
>
> I have Fastrack Pro, are we talking about same card. If not ignore the
>following:
>
>Afaik there are two recording devices at te card, 0 for digitan and 1 for
>analog. I have not managed to get the digital input work bu t the analog
>works as hw:?,1 device.
>
V Gabriele De Palo ha scritto:
> Bill Unruh ha scritto:
>
>> It helps if you give information about the sound card, not about
>> yourself.
>>
>>
>> On Sat, 30 May 2009, Michael B Allen wrote:
>>
>>
>>
>>> Hello,
>>>
&g
Bill Unruh ha scritto:
>It helps if you give information about the sound card, not about yourself.
>
>
>On Sat, 30 May 2009, Michael B Allen wrote:
>
>
>
>>Hello,
>>
>>Does capture work with the M-Audio Fast Track USB?
>>
>>My Fedora 9 laptop recognises it when I plug it in and I can get
>>outpu
Michael B Allen ha scritto:
>On Sat, May 30, 2009 at 2:36 AM, Bill Unruh wrote:
>
>
>>It helps if you give information about the sound card, not about yourself.
>>
>>
>
>It's not a sound "card". It's a little external box connected by USB.
>When I plug it in, it is recognised by the USB lay
Ciao,
i've got maudio ftu too.
i still can't use it on gnu/linux,
maybe we need maudio's firmware for use it with madfload,
but maudio doesn't work for support linux.
i still hope for the future.
ciao
Preston C. ha scritto:
>Can this card work with ALSA? I have not been able to get it too.
>
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