0
Revision Id: 0x100202
Kind Regards
James
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or directory
(The /var/tmp/portage/media-libs/alsa-lib-1.2.4/work/ was the build dir)
-JimC
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.
nonetheless, any ideas on how to fix that?
-JimC
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a Focusrite Scarlett 2i2 and it shows up as an input as I
expect. This leads me to believe either the RCP is not class-2 compliant as
advertised or that there's a bug in ALSA somewhere.
If anyone has any tips on how I might fix this error parsing audio format rates
I'd be eternally grateful!
-
above. It might work (I haven't tried
it) and you would need modern hardware to support the pass-through.
I hope this is of some help to you.
James
On 08/09/2019 08:15, Fryziu DeMol wrote:
Audio card Marian Marc 2 24/96
(earlier sold as SekD Prodif Plus)
https://i.imgur.com/fWVE3mk.jpg
er,snd_pcm
soundcore 16384 1 snd
You appear to be missing: snd-hda-intel
Try "modprobe snd-hda-intel" and see if that helps.
It should be automatically detecting it, but the modprobe is a good test.
Kind Regards
James
___
Al
On 22/04/18 13:14, Marc Haber wrote:
On Sun, Apr 08, 2018 at 05:20:13PM +0100, James wrote:
On 31/03/18 20:02, Marc Haber wrote:
In order to get output to the digital output, you need to use the "iec958"
device.
E.g.:
speaker-test -c2 -d iec958:CARD=CMI8738,DEV=0 --rate 48000
Not
aker-test -c2 -d iec958:CARD=CMI8738,DEV=0 --rate 48000
Note that you also appear to have a digital out of the SB card, so you
need to select the card when outputting.
You might also need to mess with AES0 setting
On 5/9/17, remu kelly wrote:
>
> How this can be achieved, seeing that we can't have a plugin after dmix.
Couldn't you use snd-aloop and have a plugin AFTER dmix? It basically
creates a loopback interface who's output channel is the input
channel. Although I've never used
, it can be a little more informative. With a blue "w" when
it's waiting on something from the system. Which might indicate
hardware type issues like bus speed or swap usage. Also check dmesg
for indications of hardware issues (iffy connections).
- James
On 1/15/17, Fabian Keller <
you could tail -f on other terminals to pipe that to
other things. Seems like --separate-channels is that option.
$ arecord --help
- James
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equivalent to -Dhw:1,0 . Although you
might want to omit the ,0 since that's typcially playback, not
capture, so -Dhw:1
- James
On 12/4/16, Ralf Mardorf <ralf.mard...@alice-dsl.net> wrote:
> On Sun, 4 Dec 2016 20:18:24 +, zcx wrote:
>>I have a Delta 44 sound card here that u
w internet, and I can get extras like network drivers while
still on the network with the host linux install. Much like an
arch-chroot install.
- James
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--ao=alsa, or completely omit the option. Things like audacity let
you select the available card under preferences. If you wish to use
something other than the system defaults.
- James
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driver has not implemented
support for it yet.
Have you tried using plug devices or the default
e.g
arecord --device=plughw:0,0 --format S16_LE --rate 44100 -c1 /tmp/test.wav
This will give you 1 channel, and alsa lib will do the down-mixing 2-1 for you.
James
) in order to avoid
floating point.
Does this help?
Kind Regards
James
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Hi all,
Sorry for bumping, but it's been two months without microphone, hardening
my work processes...
Is there anything I can do to help ?
Thanks in advance for any clue you could give me
On Wed, Nov 14, 2012 at 6:01 PM, James Pic james...@gmail.com wrote:
Problem persists with 3.6.6
Problem persists with 3.6.6 ... any help please ?
On Mon, Oct 29, 2012 at 1:26 PM, James Pic james...@gmail.com wrote:
Hello everybody,
Internal microphone does not work on asus zenbook ux31a. Example
recording attached as rec.wav.
uname -a: Linux zen 3.6.3-1-ARCH #2 SMP PREEMPT Mon Oct
).
-JimC
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Audio to the display works, and I don't strictly /need/ the higher bit
depth or sample frequencies, but it seems to be a related symptom. And
32kHz support would be welcome for some tasks.
Where should I look to debug this and get the eld file to show?
-JimC
--
James Cloos cl
-pcsp
My overkill list added to my blacklist items. You might also check
/etc/rc*.d/ for anything that might be playing sound(s) on shutdown.
If that other thing doesn't work.
$ find /etc/rc?.d/ -name 'K*'
- James
On 8/17/11, Julien Claassen jul...@c-lab.de wrote:
Hello Xenia!
the only idea
My computer S/PDIF on the motherboard is connected to an amplifier by a
digital cable.
I used to get 5.1 sound but now I only get left front and right front,
no center, no sub, no rear.
VLC only has stereo; I can't tell with Amarok but I think it only stereo.
$ speaker-test -c 6
speaker-test
in the way.
- James
On 6/21/11, Pierre Habraken pierre.habra...@free.fr wrote:
On 06/20/2011 10:06 PM, alsa-user-requ...@lists.sourceforge.net wrote:
Date: Mon, 20 Jun 2011 22:34:46 +0400
From: Vladimir Mosgalinmosga...@vm10124.spb.edu
Subject: Re: [Alsa-user] plughw versus hw
To: alsa-user
support. Jackd with sequencer
support. Alsa with OSS emulation. And other fine tuning type needs.
Or your distro is on such an ancient kernel, that stuff just doesn't
work at all given the lack of age of your hardware versus the copious
amounts of age in your kernel version.
- James
On 6/20/11
. Fortunately alsa is
still a bit old school. Or unfortunately depending on your POV.
- James
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can read the source, everything that you need to know is there in one
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- James
On 6/20/11, David Henderson dhender...@digital-pipe.com wrote:
On 06/20/2011 11:52 AM, Pierre Lorenzon wrote:
Hi,
From: David Hendersondhender...@digital-pipe.com
Subject: Re
(with a different case / case sensitive) for gcc to
bypass / customize a lot of that. A real PITB IMO. But just my
opinion. i.e. Use what is already there, not re-invent it in your
image. And yes a bit OT at this point.
- James
On 6/20/11, David Henderson dhender...@digital-pipe.com wrote:
I think your
with existing distros. There's enough of them
that one might suit your current needs. www.distrowatch.com
HTH,
- James
On 6/19/11, David Henderson dhender...@digital-pipe.com wrote:
Thanks for the reply Pierre. I checked into the blfs book, but it
merely says these five chapters will cover
--with-oss=yes or things might not work as
expected, if at all. Little things that you'll find out one way or
another as you learn your way around.
HTH,
- James
On 6/19/11, David Henderson dhender...@digital-pipe.com wrote:
Hi James, thanks for your help too. :) I'll provide replies in the same
of
lowest common denominators. At least change the indexing so that your
HDMI card is card 0, that way even the stupid apps try to use it,
versus some other card.
- James
On 6/16/11, Jerry Geis ge...@pagestation.com wrote:
I got a new computer (Zotac HD-ID40) ION2
I have installed alsa 1.0.24
3 alsa_knoppix.log alsa_debian.log
And of course noting /proc/asound/cards plus you might want to modinfo
the module of importance for each, to see what differs there
(versioning / parms). And various other things generally covered in
the alsa-info.sh script.
HTH,
- James
On 5/23/11, s.keup
to a sigmatel codec and NOT failing.
Which is the same module / driver for all intents, so something
configuration is awry. Or I could be wrong.
- James
On 5/23/11, s.keup...@arcor.de s.keup...@arcor.de wrote:
Hello,
Thank you.
These are the differences of the two logs: http://pastebin.com
pressing record. Beyond that, your guess is as
good as mine.
- James
On 5/20/11, Peter Hoffmann p...@peter-hoffmann.com wrote:
Hello,
I'm recording audio 24/7 with a delta 1010 sound card and have a
strange problem:
Every night at 2:30 I get spikes and some inaccurancy within some
seconds
IME. I've
never found a way to affect that setting in any other way, in the
manner needed. Even though I can see the effect of that change in the
output of amixer.
HTH,
- James
On 5/20/11, Y P yellowpeng...@edpnet.be wrote:
Hello James:
On Thu, May 19, 2011 at 09:59:01AM -0500, James
it, if I do run into it.
HTH,
- James
On 5/19/11, Y P yellowpeng...@edpnet.be wrote:
Hello,
escuse me but I'm asking myself if the problem I encounter regarding very
low level of volumes is due to Alsa/Pulse :
a few weeks/maybe a month ago, I upgraded my EEE netbook's Ubuntu O S
, with a little extra configuration in some cases.
HTH,
- James
On 5/1/11, Graham Dicker graham.dic...@antecor.com wrote:
Dominique Michel wrote:
Le Fri, 29 Apr 2011 16:28:47 +0100,
Graham Dicker graham.dic...@antecor.com a écrit :
I have been recording for many years with a Yamaha digital 4 track
or IF you can choose CPU per task for non SMP aware
applications.
- James
On 4/8/11, Sergei Steshenko steshenko_ser...@list.ru wrote:
Hello,
I've tried to run 'arecord' as part of simultaneous playback + capture rig
(for acoustic measurements) and noticed overruns.
So, even plain single
-f S32_LE recorded.wav
So the sox variant you're using is?
rec -s -4 -L -c 2 -r 96000 recorded.wav trim 00:00:00 00:00:06
Have you tried arecord without -D ? And/or with -t wav
- James
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you think? Or is
the result and the example unrelated?
- James
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And it wants your games.
http://p.sf.net
. And google hits are varied. Just a
user as far as alsa goes. And don't have that particular card on
anything of mine.
- James
On 4/5/11, jida...@jidanni.org jida...@jidanni.org wrote:
Gentlemen, I cranked everything up but still not an ounce of sound.
My ALSA information is located at
http
Greetings everyone,
I'm writing to request assistance with getting my M-Audio Delta 44
(ICE1712) functioning under a fresh Gentoo installation. At first, I
thought I had an issue with jackd, because I could not get it to
start. I tried many configuration options, but typically got ALSA
poll time
not your issue, so I'll stop rambling.
- James
On 4/2/11, James P. Early earl...@gmail.com wrote:
Greetings everyone,
I'm writing to request assistance with getting my M-Audio Delta 44
(ICE1712) functioning under a fresh Gentoo installation. At first, I
thought I had an issue with jackd
upgrade to 6.0 as soon as some of my
current projects are wrapped up. But for $80 off of craigslist, I'm
not complaining.
- James
On 3/15/11, Jonathan Wilkes jancs...@yahoo.com wrote:
Hi,
I'm looking for a relatively inexpensive USB audio interface that will
work (painlessly) on a laptop
. But I'm not exactly running the latest and
greatest of everything. Most of my hardware is sufficiently old that
I don't need to in most cases.
- James
On 3/15/11, Jonathan Wilkes jancs...@yahoo.com wrote:
Thanks, James. From a quick google search it looks like one can still get
these. But just
naming
convention as it probably varies between distros. alsa.conf?
asound.conf? +/- an /etc/ or /etc/alsa/ or /etc/sound/ or ??? And
various tricks of old to delete the asound.state file to force new
defaults. Located at /var/lib/alsa/asound.state on my system. YMMV
HTH,
- James
On 2/26/11
only. Aften to create 5.1 ac3 audio. ffmpeg is limited
to creating 5.0 iirc. And other quirks for pretty much all of the
options.
- James
On 2/21/11, Sergei Steshenko steshenko_ser...@list.ru wrote:
On Mon, 21 Feb 2011 17:37:24 +0100
Peter Hoffmann p...@peter-hoffmann.com wrote:
wa
(BTW can
. But it could be.
- James
On 2/15/11, Marcin Szyniszewski mszyn...@gmail.com wrote:
Hello,
I checked if alsa and stuff is working on other kernels - it seems it is
working brilliantly! Mic and sound works fine!
So the problem would be with the latest kernel. I removed it while being on
previous
. They generally load at boot
because your distro will likely try to restore mixer settings. And
therefor try to use your soundcard. (which is or was failing for you)
- James
On 2/12/11, Marcin Szyniszewski mszyn...@gmail.com wrote:
Thank you all for the replies! Very appreciated! :)
$ sudo modprobe
alsa-tools alsa-tools-gui alsa-utils alsamixergui
(puts them back)
HTH,
- James
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to be or
it would have never worked. Otherwise we could troubleshoot for days
without more information about how you got to your current state of
affairs. Not that you'd have that standard M$ answer. I installed
AOL and now XXX doesn't work anymore...
- James
On 2/12/11, Bill Unruh un
-utils restart
# /etc/init.d/udev restart
# groups user
# grep -i audio /etc/group
lsmod, dmesg, and all of the other stuff that's probably covered by
that alsa-info.sh script thing.
- James
On 2/11/11, Jim Lesurf j...@audiomisc.co.uk wrote:
In article
AANLkTikA=hHDEy3pCsamVvgye7u9=_4pqqw_pscjb
/proc/asound/cards
- James
On 2/11/11, Torsten Schenk torsten.sch...@zoho.com wrote:
I also use ubuntu (10.04) and it came to happen that the system didn't load
the modules automatically any more. I don't know why that happened or where
this loading is prohibited. Just try to load the module
audio, versus awk '{ print $4 $9 }' | grep -i audio or
something.
- James
On 2/11/11, Bill Unruh un...@physics.ubc.ca wrote:
On Fri, 11 Feb 2011, James Shatto wrote:
Note that * is a wildcard. So /dev/dsp* is any devices that start
with /dev/dsp.
It looks like you don't have the modules
On 24 November 2010 13:38, Grega Fajdiga gregor.fajd...@guest.arnes.si wrote:
Hello,
I am using Ubuntu 10.10 with a Creative Sound Blaster Audigy SE.
The snd_ca0106 module is loaded.
What does this show?
cat /proc/asound/cards
It will tell me if the driver is recognised or not.
that out, it's probably the best way. Otherwise try setting a
lower buffer/period time/size on aplay. You can use -v to see what
settings it is using now and work from there.
James
--
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sound cards out there have
these at all. I've heard of some Creative cards having on-board AC3
decoders but I think this may have simply been for DVDs being played on
the machine itself, not for external sources.
James
On Mon, 25 Oct 2010 15:18:03 +0100
James Le Cuirot ch...@aura-online.co.uk wrote:
Apparently a PLL is needed to synchronise the clock frequency but I
haven't been able to determine whether any sound cards out there have
these at all. I've heard of some Creative cards having on-board AC3
with arecord, it only allows S16_LE
and S32_LE and I'm pretty sure I tried both.
James
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can do this. There are probably other solutions but I'm not
that familiar with OSS.
James
--
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standards for HTML5, CSS3, SVG 1.1, ECMAScript5, and DOM L2 L3
them to make a sound without jack. Ultimately I would
like to have everything working with jack and making sounds all at
the same time.
What distribution are you using and do you have the
file /usr/lib/alsa-lib/libasound_module_pcm_jack.so?
James
to provide
configuration options in the GUI for stuff that the end user really
shouldn't need to know about.
James
--
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standards for HTML5, CSS3, SVG 1.1, ECMAScript5
than EUR250.)
Sounds like the ASUS Xonar D2X could work for you. It doesn't have
hardware mixing though.
James
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Hello everybody.
This is my first post in here - hopefully this is the right place to
ask this...
I would like to have a software volume control (using softvol plugin),
which can be muted, I know that, if I add resolution 2 at the end of
softvol configuration, I will get the mute/unmute switch -
On 4 July 2010 16:51, Manuel Reimer manuel.s...@nurfuerspam.de wrote:
James Courtier-Dutton wrote:
Why would you need to downmix 5.1 to 2.0 ?
Most Linux applications do the downmix for you.
I.e. You tell it how many speakers you have, and it outputs the sound to
them.
e.g. The xine media
Regards
James
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- James
On 5/30/10, Jaroslav Kysela pe...@perex.cz wrote:
On Sat, 29 May 2010, James Shatto wrote:
My debian distro comes with a 2.6.26-2-686 kernel. Which has version
1.0.17 of alsa. I was hoping to just install the 1.0.23 version from
alsa-project.org. But the links to download
) if
compiled from sources. And my current version of mplayer wont compile
with jack support against my current version of jackd. aka dependency
hell in source mode.
Thanks,
- James
you have.
The audio on the ALSA seems to look to be much louder than the OSS one.
Do you have the volume controls turned up?
What are you measuring this on. There should be components in the
output that remove any offset.
This might instead be a mic problem.
Kind Regards
James
. Which
is kind of ironic since the audio device registers and an hda-intel device.
AKA high definition audio. But the limits are listed in the manual. Not that
I've looked at it in the past year+.
HTH,
James
On Wed, 9 Dec 2009 09:23:44 -0200
Kazuo Teramoto kaz@gmail.com wrote:
On Wed, Dec 9, 2009 at 5:16 AM, James Shatto shado...@earthlink.net wrote:
You can set the record device to PCM (aumix term, never been able to find
the equivalent alsamixer way). Although you'll likely need to adjust
On Tue, 8 Dec 2009 21:26:14 -0200
Kazuo Teramoto kaz@gmail.com wrote:
Hello.
I like to redirect the sound I hear in the speakers to microphone, so
it can be recorded with e.g. arecord.
You can set the record device to PCM (aumix term, never been able to find the
equivalent alsamixer
Pre and it
works fine. Although web browsers don't seem to use it properly even though I
have it configured to card 0. I've never had a problem recording from it
though. Not really the best audio option, but loads better than most stock
soundcards.
HTH,
James
in /proc/asound/cards. Many apps that use alsa use something
like -D hw:2 or -ao alsa:device=hw:2 and that is assuming that you don't want
to just use the defaults.
HTH,
- James
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#0: subdevice #0
Additionally, doing a modprobe of snd-usb-audio returns:
FATAL: Module snd_usb_audio not found.
On Fri, Aug 28, 2009 at 9:44 AM, Clemens Ladisch
cladi...@googlemail.comwrote:
James Gadsby wrote:
Simply put, MIDI devices are not detected in the audio programs I use
. And if you're doing it
for some sort of TV Capture card that audio has less latency than video, so
you'll hear them talk before their lips move (just slightly) which can/will
drive you nuts.
HTH,
James
--
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Hello,
I've been spending quite a long time attempting to get MIDI support for my
music applications with ALSA. I need to say that all these MIDI devices
function fine and as they should do on other hardware. The hardware I'm
trying to work with is a mid-2009 MacBook Pro, running on Ubuntu 9.04.
. For 8 channels, I guess you want -Dsurround71.
Regards,
James
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James
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James wrote:
Clemens Ladisch wrote:
James wrote:
$ speaker-test -D iec958 -c 6
...
Channels count (6) not available for playbacks: Invalid argument
S/PDIF does not support uncompressed surround sound; you have to play
stereo data (use -c 2) or AC-3/DTS-compressed
Clemens Ladisch wrote:
James wrote:
$ speaker-test -D iec958 -c 6
...
Channels count (6) not available for playbacks: Invalid argument
S/PDIF does not support uncompressed surround sound; you have to play
stereo data (use -c 2) or AC-3/DTS-compressed data.
Best regards
My digital sound broke again :-(
I like Gentoo because it updates alot but I hate it because it breaks my
digital sound alot. :-(
I am using the 2.6.28.2 kernel and alsa 1.0.19
$ aplay -l
List of PLAYBACK Hardware Devices
card 0: NVidia [HDA NVidia], device 0: ALC888 Analog [ALC888
.
The driver lets one send samples at any rate 44.1, 48 etc.
The 0404 USB internally only works at a fixed but configurable rate via
a vendor specific processing unit control.
I.e. if the 0404 USB is internally set to 44.1 and one sends 48 to it
things go wrong.
Kind Regards
James
how to I set alsa to use -Dplug:spdif as the default?
$ aplay -L
default:CARD=NVidia
HDA NVidia, ALC888 Analog
Default Audio Device
front:CARD=NVidia,DEV=0
HDA NVidia, ALC888 Analog
Front speakers
surround40:CARD=NVidia,DEV=0
HDA NVidia, ALC888 Analog
4.0 Surround output
David McCloskey wrote:
That's just a warning. Usually it will be working after that warning.
ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/docs/HD-Audio.pdf
Dave
On Sun, Jan 11, 2009 at 11:48 PM, James bjloc...@lockie.ca wrote:
hda_codec: Unknown model for ALC883, trying auto
hda_codec: Unknown model for ALC883, trying auto-probe from BIOS...
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I can't digital output for analog works fine.
Should aplay -L show a digital output?
$ aplay -L
default:CARD=NVidia
HDA NVidia, ALC888 Analog
Default Audio Device
front:CARD=NVidia,DEV=0
HDA NVidia, ALC888 Analog
Front speakers
surround40:CARD=NVidia,DEV=0
HDA
On 12/28/08 23:04, James wrote:
Something happened to my ALSA.
It plays sound through the regular speakers but not through the digital
out of the motherboard.
I can play DVDs through the digital out so I know the output should work.
It used to work.
┌──[AlsaMixer v1.0.18
Something happened to my ALSA.
It plays sound through the regular speakers but not through the digital
out of the motherboard.
I can play DVDs through the digital out so I know the output should work.
It used to work.
┌──[AlsaMixer v1.0.18 (Press Escape to
. If the original sound
file is 24bit 44.1khz, and the usb sound device can only do 24bit 96khz,
ALSA will have to resample them to get 44.1khz into the 96khz pipe.
Kind Regards
James
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Paulo Moura Guedes wrote:
I'm trying to get bit perfect output out of my linux box, but I can't find
much
info on the web. I'm using ALSA.
Some questions:
- does Linux/ALSA features dynamic sample rates?
- is it possible to set the bit-depth? (in my case to 24 bit)
- what other
Zbigniew Baniewski wrote:
Hallo,
Over a year ago I've reported the problem with very low microphone input
sensitivity, while recording using Audigy2, what is making this card unusable
for VoIP.
At that time, I've received a tip (thanks again), to fix it by introducing
a line:
, but I have no idea how to do that.
Any suggestions?
Thanks,
James
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kernel.
With alsa compiled over it. That way there's not multiple versions /
locations. As the path assumptions would match.
HTH,
James
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Build
if you ARE using
udev). But that doesn't appear to be your issue. And other ways to implement
the above custom configuration with alsaconf and other utilities. I just never
got them to work for me back in the day, and never adapted to letting current
tools try to do it for me.
- James
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 in / out / trough.
For those
are you needing? And how many
inputs? TRS / XLR / 3.5mm stereo? There's several out there depending on your
needs. What's your budget?
If you need a lot more inputs at higher rates, then firewire might be better
suited.
freebob.sf.net
HTH,
- James
from source is a no brainer. And a lot less annoying on todays faster cpus.
HTH,
- James
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On Thu, 28 Aug 2008 17:03:44 +0200
brunal [EMAIL PROTECTED] wrote:
I have to precise that I'm using a M-Audio fatst track pro sound
card, which works fine if I only record stereo files.
Is there some reason the application doesn't do mono? Almost all I've seen
allow you to record only one
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