method
The full output of alsa-info.sh is available here:
http://www.alsa-project.org/db/?f=5a236e8805b9a7d125e877e9cb235023814af813
Does anyone please a tip of what to do next?
Thanks,
Florian
postscript: relevant output of lspci:
00:1b.0 Audio device [0403]: Intel Corporation 7 Series/C210
.c', line: 4608
The full output of alsa-info.sh is available here:
http://www.alsa-project.org/db/?f=5a236e8805b9a7d125e877e9cb235023814af813
Any ideas how to proceed?
Thanks in advance!
Florian
--
Go from Idea to M
On 05/22/13 03:13, Ben Briedis wrote:
>
>> Subject: [off-list] [Alsa-user] Is RME HDSPe AIO supported?
>> From: ralf.mard...@alice-dsl.net
>> To: benbrie...@hotmail.com
>> Date: Tue, 21 May 2013 12:25:41 +0200
>>
>> Off-list, since the mail at least seems to be delayed. It didn't came
>> through t
Ben,
> How well does AIO work? Is the omission on the matrix deliberate - i.e. does
> it
> reflect the fact that there are unresolved issues? Is it safer to buy the
> 9632?
All of the AIO's features work as does the matrix. The card is
configured through standard ALSA controls.
Flo
--
Mach
sorry for >< please >> <<
Hi all,
The Linux Audio Conference submission deadline has been extended! It is now
February 17th, 2013 (23:59 HAST).
So, if you were considering to submit a paper but couldn't make up your mind
yet, here is your chance to become active! Never forget that this confer
finitely not correct.
Moreover, not all the 8 in- and 8 out-channels were displayed correctly.
But audio output was also working for me and I could access all
channels through jack at sample rates up to 96 kHz.
Best,
Florian
Quoting Matthew Robbetts :
> Hi Florian,
>
> On 21/12/12 2
Hi Matthew,
did you try to use alsa-mixer ? It was not working when I was trying
the fireface ucx with the modified mixer.c some weeks ago. It would be
interesting to know whether it is different on your system / with the
new patch.
Thanks,
Florian
Quoting Matthew Robbetts :
> On 18
Quoting Matthew Robbetts :
> On 18/12/12 13:19, Clemens Ladisch wrote:
>> Matthew Robbetts wrote:
>>> Is this bug fixed upstream at this point?
>>
>> This patch is still untested:
>>
>
> Hi Clemens,
>
> I can confirm that I get the original error with a Babyface, and that
> this patch appears to f
without any problems and 8/8 channels are available. I tested the 8
output channels using SuperCollider through JACK, they seem to work
fine. Great!
I did not check the input channels and latency so far.
Best,
Florian
Quoting Clemens Ladisch :
> Florian Hanisch wrote:
>> I am runni
than 2/2
channels, how do I activate these extra channels ? There should be 8/8
channels available in usb-class-compliance mode, these are displayed
on a mac-osx without using the RME native driver.
Thanks a lot,
Florian
these 8/8-channels are displayed!
Best,
Florian
Quoting Florian Hanisch :
> Hi,
>
> thanks for the information. I just commented the following part of mixer.c
>
> if (hdr->bLength < 7 || !csize || hdr->bLength < 7 + csize) {
> snd_pri
rg.jackaudio.Error.InvalidArgs)
The same error occurs for the outchannels and I tried the values 1,2,8
for the nummer of channels but I always get the same error.
Best,
Florian
Quoting Daniel Mack :
> On 02.10.2012 09:27, Clemens Ladisch wrote:
>> Florian Han
swer.
In the post quoted above, the mixer is disabled by modifying the
source code and recompiling afterwards. I am not very familiar with
the details of alsa, so I would like to ask whether this is the only
way to do this or whether there is some easier way to get there ?
Thanks,
Flo
Hi,
thanks a lot.
Quoting Clemens Ladisch :
> Florian Hanisch wrote:
>> I have been trying to connect a friend's soundcard (RME fireface UCX)
>> to my linux computer. [...] RME claims that the device has a class
>> compliance mode (which is working with a mac compute
l options ? Any other ideas ?
Thanks a lot for any help,
Florian
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2012/3/18 Nikos Chantziaras :
> Delete the /etc/asound.conf and ~/.asoundrc files so ALSA can choose the
> default settings.
Ah, that seems to work!
Thx a lot!
> On 18/03/12 14:36, Florian Lindner wrote:
>>
>> Well, I have blacklisted the module for my onboard sound:
>&g
Repost to the list, accidentally postet to Nikos only.
2012/3/18 Nikos Chantziaras :
> On 17/03/12 23:53, Florian Lindner wrote:
>>
>> My system is a Linux 3.2.11 (Archlinux) / KDE 4.8.1.
>>
>> The system used to work perfectly. Today I have installed a new
>>
Hello!
My system is a Linux 3.2.11 (Archlinux) / KDE 4.8.1.
The system used to work perfectly. Today I have installed a new
soundcard (Asus Xonar Essence STX) in addition to my onboard
soundcard.
florian@horus ~ % cat /proc/asound/cards
0 [SB ]: HDA-Intel - HDA ATI SB
does it make a difference for debugging/ working out how to get the sound right
if I use 32bit or 64bit Operating system? Which should I rather use? so far I
got same results on both!
thanks, Florian
--
Create and
On 12/31/10 14:07, Michael Gerdau wrote:
> Could it be that the ExpressCard is not supported by the driver ?
No. The express card behaves exactly like the cardbus card, except for
the host interface.
> Would I need a different firmware that the one provided by ALSA firmware ?
No.
> What else c
On 12/31/10 13:56, David Lam wrote:
> http://wiki.linuxproaudio.org/index.php/Driver:hdspe
>
> I have followed the instructions mentioned in that site still cannot get
> the driver works.
This is not an official alsa driver, so you are asking in the wrong place.
> When I make install it has err
Martin,
> i would like to run a rme hdspe aio on linux,
> is the Card supported ?
Not fully yet, only 1024 period operation for the moment.
> & if true
> how to get it to work ??
http://wiki.linuxproaudio.org/index.php/Driver:hdspm
http://wiki.linuxproaudio.org/index.php/App:hdspmixer_64
> an
Hello,
I already tried to find help in IRC but I had no luck there.
What I want to do is the following:
I have 5.1 sound system which I want to use. Most of the files I want to
play just are stereo. So I need upmix to 5.1. I've already tried a lot
of .asoundrc configurations and I think I underst
Hello,
I already posted my problem at the PulseaAudio mailing list but they sent me
here.
Here we go:
I try to run my system (ArchLinux)with pulseaudio. More or less successful.
After some minutes of playback pulseaudio will just crash.
My soundcards are:
0 [CA0106 ]: CA0106 - CA0106
Tobi,
> I could only found an ALSA driver for the PCI version HDSP MADI. Is there
> also a
> driver available to use with the PCI Express Bus version HDSPe MADI?
The same driver. All the PCIe versions are handled the same way as their
PCI equivalents.
Flo
--
Machines can do the work, so peopl
Paulo,
> I would like that all the resampling is made by the DAC because I can trust
> on its quality.
Bwahahahaha!
Flo
--
Machines can do the work, so people have time to think.
public key 6C002249 x-hkp://wwwkeys.eu.pgp.net
--
> How can I set up capture Volume on RME Hammerfall DSP MADI?
You cannot.
> The capturing volume is low
Well, turn up the gain on whatever you connected to the MADI bus.
Flo
--
Machines can do the work, so people have time to think.
public key 6C002249 x-hkp://wwwkeys.eu.pgp.net
--
Clemens Ladisch wrote:
> Florian Winter wrote:
>
>> Suppose, an ALSA playback device is opened in blocking mode, and one
>> thread calls snd_pcm_writei. If the snd_pcm_writei call blocks, because
>> the internal buffer of the ALSA device is full, is there a way by whi
any data to the device?
Best regards,
Florian
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Lars,
> I have two sound cards; one RME9652, and one built-in VIA thing. The
> RME is connected to the stereo via S/PDIF, and the VIA thing is
> connected to a wireless transmitter.
Besides the ALSA configuration problem - it won't work unless both sound
devices are in sync. All RME devices can
Ales,
> > Yes: What card is it? If it is the interface card, then there is no
> > IO Box connected/recognized. Please post the output
> > of /proc/asound/card0/hdsp and if it is an interface card, the
> > kernel messages when you load the module or boot (with dmesg).
> it is multiface, i think pre
Hi Ales,
> i configure hdsp with alsaconfig, then hdsp appears in my application
> (pure data), but no sound.
>
> i get from alsamixer : "no elems found"
> which should mean that card 1 has no mixer interface,
> i think all volumes on alsamixer are down,
> but i can't open ANY mixer.
The mixer yo
Martin,
> I want to use a RME Multiface soundcard with PCMCIA Cardbus Card on a
> Dell Notebook with Ubuntu. [..]
> Now the card has 8 in and 8 output channels,
Well, actually, you have 18 input and 20 output channels with the
multiface :)
> I get only one digital playback device which is the
Grant,
> Is there any way to test the digital contents of a FLAC file against
> the digital stream sent to the sound card for a match?
Play it out a digital line and loop it back, if you want to be
absolutely sure.
Flo
--
Machines can do the work, so people have time to think.
public key 6C00
Chris,
> > On IEEE 32 bit floats the mantissa is 23 bit, so there might be
> > situations where you loose the LSB.
> And that was the only point - a "pro audio chain" should be able to
> support "digital wire" capability.
This has nothing to do with the original poster's issues, so I changed
the
On Thursday 12 June 2008 20:10:04 Chris Smith wrote:
> On Thursday 12 June 2008, Florian Faber wrote:
> > What makes you think converting a 16 bit unsigned integer to a IEEE
> > 32 bit float and back would change the value?
> Should have used a 24 bit example. I'm of the opi
Chris,
> A little peeve with some so called "pro" audio servers is their
> inability to act as a 'digital wire', ie: what goes in comes out,
> totally unchanged. As an example, there are times you may want the
> same exact 16 bits you send out of app to arrive at the audio device
> unmolested. Jac
ng ALSA drivers)? Is
it safe to assume that ALSA will not use dmix if the sound card (and the
ALSA driver for it) supports hardware mixing?
Best regards,
Florian
Clemens Ladisch wrote:
> Florian Winter wrote:
>
>> - What is the dmix plugin and what are the benefits of using it?
?
- What consequences does disabling the dmix plugin have? What essential
features of ALSA will be missing without it?
Best regards,
Florian
Clemens Ladisch wrote:
> Florian Winter wrote:
>
>> Is there another way to determine whether a certain hardware supports
>> snd
the hardware?
Best regards,
Florian
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Alex,
> The idea is the following :
>
> 1.) Of course there has to be an input double buffer which generates
> the desired block of samples.
You want hardware monitoring - there are sound cards that support
hardware mixing. With good converters you have latencies down to 5
samples at 192kHz, t
Greg,
> I had the vision that the option was in the configuration dialog, but
> that is not the case for hammerfall 9636, though it was for 9632.
Now I understand what you meant. But setting the input level wouldn't
help you connecting microphones to the AEB8 :)
I am rewriting the tools at the
Greg,
> >> I have a hammerfall 9636 card on my linux 64 studio machine. got
> >> it working today but have a question regarding recording volume on
> >> the card. The thing is that it seems too low. When connecting the
> >> microphone directly to the AEB-8 extension board for analog input,
> >> I
Greg,
> I have a hammerfall 9636 card on my linux 64 studio machine. got it
> working today but have a question regarding recording volume on the
> card. The thing is that it seems too low. When connecting the
> microphone directly to the AEB-8 extension board for analog input, I
> only have vague
Mark,
> Any chance you might be working on hdspmixer at the same time.
Yes, another main goal right now is bringing the hdsp user space
applications up-to-date.
> I worked on testing it when Thomas Charbonell first wrote the program.
> One disappointing limitation of the whole HDSP Linux suppo
On 2/22/2008 3:47 PM, Jonathan Stowe wrote:
> On Fri, 2008-02-22 at 13:59 +0100, Florian wrote:
>
>> we manage to get "down" to 8 milliseconds buffer size at CD
>> quality without glitches with the onboard soundcard (Intel HDA).
>> However, we would like t
> Because laptops often use SMM traps to poll battery and fan status
> which can tie up the CPU for several milliseconds.
>
> The vast majority of laptops are simply not designed for low latency work.
yeah, that might be a problem we'll have.
Thanks,
Florian
>
&
im, cups, proftp, cron, atd,
> portmap, nfs, running while you're making said demo. And
> that audio has been given realtime permissions at the user
> level. Plus a low latency kernel.
pretty much all of the above :)
Thanks,
Florian
>
> ---
ze to arbitrary values.
And, as you suggest, the signal's frequency spectrum will contain
very high frequencies, so we've created a tool to automatically
detect underruns from the recorded audio output of the soundcard,
(and to correlate that with the underruns reported by ALSA).
Later
On 2/22/2008 11:55 AM, Florian wrote:
>> But the laptop is not running realtime linux is it? It has
> sure it is...
to clarify: this is not a "full" realtime or embedded linux, it's
Redhat's RHEL 5 with their realtime kernel.
Florian
>> loads of potentia
27;t it work on a laptop? Actually, my collegues here
"think" in guaranteed time slices in the microsecond or even
nanosecond range. For them, 1 millisecond is an eternity where A
LOT can be done on modern processors :)
Later,
Florian
>>
>> Thanks, Florian
>>
>>
ns we need to show that on a laptop... And, btw,
our software synthesizer is running on realtime Java :)
Thanks,
Florian
On 2/22/2008 1:48 AM, Bill Unruh wrote:
> On Fri, 22 Feb 2008, Florian wrote:
>
>> Hi,
>>
>> on our IBM/Lenovo T60 laptop, we want higher audio qua
think You should
> give it a try. At least take a look at it's specs at
> http://www.echoaudio.com/Products/CardBus/IndigoIO/index.php
yes, that's exactly what we needed: a positive assertion that
this card works fine... :)
Thanks,
Florian
>
>
>
> --
>
> R
o T60 laptop (Intel HD-Audio) -- we cannot go lower
than approx. 8 milliseconds period size without glitches.
Thanks for any pointers.
Florian
--
Florian Bomers
Bome Software
---
Music Software, Development Tools: http://www.bome.com
Java
tency?
Thanks,
Florian
--
Florian Bomers
Bome Software
---
Music Software, Development Tools: http://www.bome.com
Java Sound extensions, plugins: http://www.tritonus.org
The Java Sound Resources:http://www.jsresource
Hi,
i use ubuntu gutsy betas and i try to get the alsa pcm jack plugin to work. I
copy here from the bug report i commited to ubuntu:
Binary package hint: libasound2-dev
Install libasound2-dev and libasound2-plugins. Note that the previous ubuntu
release removed the alsa jack plugin from the
On Thursday 27 September 2007, Mark Constable wrote:
> As a baseline, say I started with this, where could I go
>
> >from here to get an app (ie; Amarok) to play back out two
>
> soundcards at once ?
Well, first of all you'd probably like to slap a plug plugin around it:
pcm.pshared {
typ
On Sunday 05 August 2007, europeen wrote:
> Hi List,
>
> I would know if it's possible to route PCM sound to the Outputs 3 & 4 of
> the M-Audio Delta44 soundcard. By default, the outputs 1&2 are used and
> works fine.
> I don't find a tips/tutorial about this. Should I configure
> the .asoundrc wit
soundcard via alsaconf. After a reboot all
information is lost.
How can I make LinuX to store these information and to use them always?
(Debian Etch)
Regards,
Florian
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On Friday 29 June 2007, Mohan Kashyap wrote:
> I waould want to capture audio data from the sound card as and when an alsa
> application starts playing. I could not figure out what lines must be added
> in .asound.rc for this.
Whether this is possible trivially depends on your soundcard. Some soun
system's running, sound is confirgured right
(audigy). But after a reboot the information is lost. I'd love to have
th driver loaded on boot and playing a sound on login. Does anyone have
a solution?
Regards, Florian
--
es my audigy ls. But after a
reboot the system recognizes HDA-Intel-chip only.
That's annoying!!
What can I do??
System: Debian 4.0 Etch r0
Alsa: what's on etch...
Regards and thx for help,
Florian
-
This
Hi,
it would be great if someone could point me to informations on how-to
setup a toHost-Connection with alsa.
i want to connect a KORG ag-10 to the serial port of my computer.
I use a toHost cable.
I attached my aadebug-ouput.
Maybe this helps.
greetings
Florian Over
ALSA Audio Debug v0.1.0
?
What else can I try?
Thanks Florian
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Clemens Ladisch wrote:
Florian Ladstaedter wrote:
I am struggling with setting up the alsa dmixer with the M-Audio
Audiophile USB sound card.
the card still works with this, but no sound mixing. Playing a song and
using another sound appl at the same time gives:
WARNING **: alsa_setup
d appl at the same time gives:
WARNING **: alsa_setup(): Failed to open pcm device (hw:2,0): Device or
resource busy
2. Can jackd and simple alsa apps use the soundcard at the same time
with dmix (theoretically, if it would work)?
any hope?
thanks, any help appreciated
Florian
-
On Mon, 10 Apr 2006 02:11:25 -0700 (PDT)
Mo <[EMAIL PROTECTED]> wrote:
> Greetings.
>
> I would like to construct a dedicated system for recording five
> discrete audio feeds (voice-quality) via the line-in jacks each of five
> sound cards. Spent the weekend tinkering, but was never able to get
On Wed, 12 Apr 2006 23:53:44 +0200
Esben Stien <[EMAIL PROTECTED]> wrote:
> Lee Revell <[EMAIL PROTECTED]> writes:
>
> > I'm not sure that this is supposed to work.
>
> A report from Florian Schmidt on #lad said that it worked for him.
Just for the record:
On Wed, 08 Feb 2006 19:57:43 +0100
Joerg Kampmann <[EMAIL PROTECTED]> wrote:
> thanks Fons, this works. However:
> jaaa does not show any input. Although: when I activate the mike I hear
> something in the speaker. And when I activate "sine" (any level) I hear
> a tone, the frequency of which I
On Mon, 5 Apr 2004 14:36:52 +0200 (CEST)
Jaroslav Kysela <[EMAIL PROTECTED]> wrote:
> On Mon, 5 Apr 2004, Florian Schmidt wrote:
>
> > I have one little question to the alsa devs though:
> >
> > The kernel level OSS emu already does stuff like samplerate
> &g
Therefore 3] really look like the best solution for me..
I have one little question to the alsa devs though:
The kernel level OSS emu already does stuff like samplerate conversion,
etc.. why can't it be extened to do channel mixing like the commercial
oss drivers? I suppose it is unclean design, but
On Fri, 26 Mar 2004 13:06:49 +
Darrell Blake <[EMAIL PROTECTED]> wrote:
> Is there any way to make a device that you made yourself in .asoundrc
> the default playing device? I have a lot of games I want to play that
> only use OSS so I'm having to use OSS emulation which only uses the
> def
On Sat, 20 Mar 2004 10:30:54 -0800 (PST)
x m <[EMAIL PROTECTED]> wrote:
> emu10k1-gp _ _ _ _ _ _ _1352 _ 0 _(unused)
this looks like an OSS module.. See this:
http://alsa.opensrc.org/index.php?page=FAQ010
--
kT
---
This SF.Net email is spo
ernel level oss emyu can be told which devices to
simulate.. So you be able to get it to run with with kernel oss emu...
http://alsa-project.org/~iwai/OSS-Emulation.html
read the part on pcm mapping..
Florian Schmidt
--
kT
---
This SF.Net emai
On Fri, 19 Mar 2004 02:32:50 -0800 (PST)
sachin sharma <[EMAIL PROTECTED]> wrote:
>
>
>
> __
> Do you Yahoo!?
> Yahoo! Mail - More reliable, more storage, less spam
> http://mail.yahoo.com
>
>
> ---
> This S
On Tue, 09 Mar 2004 23:36:31 +
Davide <[EMAIL PROTECTED]> wrote:
> After trying to resolve my problem whit my onboard soundcard(Below)
> I try to install one of my old PCI soundcards,
> A SB PCI Md:8***
> but the efect is the same,, :((( (realy realy sad)
>
> Teamspeak works , quake3 or et wo
On Tue, 02 Mar 2004 00:00:07 +
Davide <[EMAIL PROTECTED]> wrote:
> Can't run TeamSpeak and Quake3 at the same time... :(
> Can someone help me???
>
> i'm using Debian SID, kernel 2.6.3, I have the correct snd_intel8x0
> module installed and the "Alsa Conpt. OSS modules"
>
> If anyone need mo
On Mon, 01 Mar 2004 00:00:35 +0100
Stuart Pook <[EMAIL PROTECTED]> wrote:
> on 29 Feb 2004. Jaroslav Kysela wrote
> > Yes, the dmix plugin might fail with mplayer, too. The mplayer tries
> > to set it's own values for buffer sizes, but dmix has these values
> > fixed. Unfortunately, mplayer fails
Hi,
i wonder if there's a pcm plugin available, that is writeable and also
readable in a sense that everything that is written to it, will be
readable again. this could be useful for recording alsa app output if
the hw does not support capturing the line-out or mix.
Let's call this plugin "redir
On Sun, 29 Feb 2004 13:18:43 +0100 (CET)
Jaroslav Kysela <[EMAIL PROTECTED]> wrote:
> > pcm.foo {
> > type dmix
> > slave.pcm "spdif"
> > }
> >
> > pcm.!default {
> > type plug
> > slave.pcm "foo"
> > }
> >
> > or something..
>
> It won't work, because dmix (dsnoop and dshare) p
On Sun, 29 Feb 2004 11:36:47 +0100 (CET)
Jaroslav Kysela <[EMAIL PROTECTED]> wrote:
> The mmap() access might work with latest aoss (1.0.3 in CVS), but I
> don't have any test machine with working OpenGL and I'm too much lazy
> to study how to enable software rendering, so I don't test these
> app
On Sun, 29 Feb 2004 12:01:33 +0100
Anders Bruun Olsen <[EMAIL PROTECTED]> wrote:
> I have been reading a bit about using dmix but have a slight problem.
> It is easy to setup things so that dmix is the default place to send
> sound, but how can I make dmix use plug:spdif for the actual output?
pc
On Sun, 29 Feb 2004 10:13:15 +0100 (CET)
Jaroslav Kysela <[EMAIL PROTECTED]> wrote:
> On Sun, 29 Feb 2004, cyberpro wrote:
>
> > It doesn't work.
> >
> > "It seems that aoss does not support libc's fopen() function
> > calls. So all OSS apps that use the sound devices by calling fopen()
> >
On Sat, 28 Feb 2004 03:00:07 -0300
cyberpro <[EMAIL PROTECTED]> wrote:
> It's possible to reproduce a simultaneous sound (like xmms with a mp3
> and a game, for example, at the same time)?
>
> I have seen that sb-live 5.1 supports this feature (without any extra
> options in modules.conf or any
On Wed, 25 Feb 2004 10:25:02 -0800
Ben Ford <[EMAIL PROTECTED]> wrote:
> Sorry for the delay, school is kicking my ass right now.
>
> I've upgraded to kernel 2.6.3, which includes ALSA 1.0.2c. I still
> have the same problem. Is dmix in some optional package that I don't
> have installed or wha
On Mon, 23 Feb 2004 10:28:07 +0100 (MET)
Clemens Ladisch <[EMAIL PROTECTED]> wrote:
> Your hardware doesn't support this.
>
> It is possible to emulate three stereo devices with the dshare plugin,
> but this works only for applications using ALSA (not OSS), and it
> isn't documented. Speak up it
On Tue, 24 Feb 2004 09:04:33 -0700 (MST)
Kirk Bauer <[EMAIL PROTECTED]> wrote:
> On Tue, 24 Feb 2004, Jaroslav Kysela wrote:
>
> > On Tue, 24 Feb 2004, Kirk Bauer wrote:
> >
> > > What if I want to run the same program through 'aoss' numerous
> > > times simultaneously with the output going to d
On Tue, 24 Feb 2004 08:12:15 -0700 (MST)
Kirk Bauer <[EMAIL PROTECTED]> wrote:
> On Tue, 24 Feb 2004, Jaroslav Kysela wrote:
>
> > On Tue, 24 Feb 2004, Kirk Bauer wrote:
> >
> > > I think it would help if somebody could explain to me (or point me
> > > to a web resource) about how you select an
On Tue, 24 Feb 2004 15:49:02 +0100 (CET)
Jaroslav Kysela <[EMAIL PROTECTED]> wrote:
> On Tue, 24 Feb 2004, Kirk Bauer wrote:
>
> > I think it would help if somebody could explain to me (or point me
> > to a web resource) about how you select an output in ALSA. I know
> > how to do it with 'aplay
On Mon, 23 Feb 2004 08:52:52 -0500
Preston <[EMAIL PROTECTED]> wrote:
> Well, after running the script, many missing dirs were created.
> The good news is that it didnt broke anything, in fact, now I can use
> OSS emulation. The not so good news is that still, alsa drivers
> doesnt works. i downl
On Sat, 21 Feb 2004 06:07:40 +0100
Florian Schmidt <[EMAIL PROTECTED]> wrote:
>
> subject says it all.. i installed it and suddenly my .asoundrc
> definition of an asym device stopped working. I looked into the
> src/pcm sir and couldn't find it either..
>
> Any
On Sat, 21 Feb 2004 11:42:55 +0200 (EET)
Tommi Sakari Uimonen <[EMAIL PROTECTED]> wrote:
> > > i have a delta/66, which has four output channels. i'm currently
> > > able to play a stereo audio file on the first two channels. is it
> > > possible to play a second file independently on the second
subject says it all.. i installed it and suddenly my .asoundrc
definition of an asym device stopped working. I looked into the src/pcm
sir and couldn't find it either..
Anyone know what happened?
Flo
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On Fri, 20 Feb 2004 13:17:08 +0100
"kluu te" <[EMAIL PROTECTED]> wrote:
> It's not nessessary for 1.x drivers
How do you come to this conclusion. If you didn't have alsa installed
before on that machine and your distribution did not already create the
device files, you need to create them.. eithe
On Thu, 19 Feb 2004 17:03:01 -0500
Preston <[EMAIL PROTECTED]> wrote:
>
> Is required or not to run this script for version 1.x of the drivers?
Well, it is required to be run when the device files in /dev/snd don't
yet exist.. I don't know about devfs though..
FLo
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On Wed, 18 Feb 2004 12:03:22 +0100 (CET)
MvH <[EMAIL PROTECTED]> wrote:
> Hello, I am a bit new to alsa since my applications worked better with
> oss before. I own a emu10k1 card.
>
> I know, the driver has everything unmuted by default, so I use the
> amix program to set the sound in startup, I
On Thu, 12 Feb 2004 20:38:28 +0100
Alt Zerone <[EMAIL PROTECTED]> wrote:
> Hello all.
>
> Can anybody RTFM me to any doc about the gazillion SBLive mixer
> settings I see in alsamixer?
http://alsa.opensrc.org/index.php?page=emu10k1
and from that page:
http://alsa.opensrc.org/index.php?page=Sb
On Wed, 11 Feb 2004 16:51:19 +0100
Florian Schmidt <[EMAIL PROTECTED]> wrote:
> >
> > it probably won't be possible to mix the sound of the second device
> > seperatly of the first device?
>
> sure, start alsamixer with the appropriate command line switch [i&#
On Wed, 11 Feb 2004 15:50:26 +0100
Alien9 <[EMAIL PROTECTED]> wrote:
> at home I have 2 cards, a via82xx and a ens1371.
>
> the via82xx onboard has 2 devices, I'd like to know if it is possible
> to split that card in 2 seperated cards, each stereo instead of one
> surround card.
http://
On Tue, 3 Feb 2004 09:25:38 -0800 (PST)
Bill Unruh <[EMAIL PROTECTED]> wrote:
> On Tue, 3 Feb 2004, Jaroslav Kysela wrote:
>
> >
> > Get alsa-driver 1.0.2b.
> >
> > Jaroslav
>
> On www.alsa-project.org, there is only 1.0.2 listed as the last
> stabl
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