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
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
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
---
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Java
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
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
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
>>
>>
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
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
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
>
> ---
> 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
>
&
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
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
any data to the device?
Best regards,
Florian
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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
> 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
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--
Paulo,
> I would like that all the resampling is made by the DAC because I can trust
> on its quality.
Bwahahahaha!
Flo
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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
--
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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
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
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
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
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
2 chipset, and that should be coverd by the alsa
> driver whitout any problems.So if there is anybody
> outthere who has installed alsa-driver-5.0.12a and
> alsa-lib-5.0.10 on a redhat 7.2 system, please give me
> a note.
You can try to run automake and autocon
xists, that plughw:0,0
is the same card/device ?
Thanks!
Florian
--
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Java Sound
Java Software/Sun Microsystems, Inc.
http://java.sun.com/products/java-media/sound/
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installing a new kernel.
Thanks for any hints or pointers,
Florian
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Java Sound
Java Software/Sun Microsystems, Inc.
http://java.sun.com/products/java-media/sound/
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Hi Paul,
nice to see you here, too :)
I could eventually solve my problems by updating to a stock SuSE kernel 2.4.16.
I do have missing symbols in some strange modules now, but alsa works :)
Good luck...
Florian
Paul Sorenson wrote:
>
> Alsa people,
>
> I pinged the list
I have Mandrake 9.
I downloaded the 5.12a drivers.
I join a dump of what happens when I ./cvscompile.
It seems like it doesn't like the genksym -k 0.0.0.0
Regards,
++
Florian
___
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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
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
>>
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
l options ? Any other ideas ?
Thanks a lot for any help,
Florian
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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
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
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
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
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
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
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
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
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
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
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
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
.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
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
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
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
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
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
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
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
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 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
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
the hardware?
Best regards,
Florian
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?
- 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
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?
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
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,
> > 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
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
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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
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
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
On Sat, 19 Apr 2003 15:45:25 +1000
Mark Constable <[EMAIL PROTECTED]> wrote:
> On Sat, 19 Apr 2003 12:35 pm, Florian Schmidt wrote:
> > Let me once again reply to my own question. If i get this solved, i
> > promise to make an addition to the wikki so everyone gets to kno
On Thu, 3 Jul 2003 13:05:26 -0400
"Robert Liguori" <[EMAIL PROTECTED]> wrote:
> How do I
>
> U N S U B S C R I B E
>
> from this list?
http://www.alsa-project.org/mailing-lists.php3
we find this link:
http://lists.sourceforge.net/lists/listinfo/alsa-user
navigate down to the entry fie
hi!
I'm running alsa 0.9.4 with linux 2.4.21 + acpi-patch on my
Medion/Fujitsu-Siemens Amillo D notebook the distribution is debian
unstable).
This notebook has the following audio controller wtih an optical spdif
output:
00:1f.5 Multimedia audio controller: Intel Corp. 82801BA/BAM AC'97 Audio
(
another midi port that was routed directly to
the daughterboard. Is this possible with the cd-46xx driver, too?
Regards,
Florian Schmidt
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Hi,
is there a simple vu-meter available to help setting up my recordings?
Or will i have to write my own?
Florian Schmidt
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On Wed, 29 Oct 2003 15:54:28 -0600
Ryan Mayer <[EMAIL PROTECTED]> wrote:
> I am using ALSA 9.6 right now and I can't seem to get my sound card to
> play more than one sound at a time. Before, in windows, my card could
> play multiple sounds at once (play a game while listening to mp3's).
> I loo
ook that one up in the alsa sound card matrix to see, if the driver
suports multiple open. If your driver is reported to have no
hardwaremixing capabilities, you can give the dmix plugin a shot. Look
it up in the alsa-docs.
Florian Schmidt
---
e for your driver. as far
as i can tell your driver does not support hw mixing. Read in the alsa
docs and the wiki about the dmix plugin. Also search the mailinglist
archives. This has been discussed before..
Florian Schmidt
---
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On Wed, 29 Oct 2003 20:23:42 -0600
Ryan Mayer <[EMAIL PROTECTED]> wrote:
> How did you tell if my card supports hw mixing or not? I don't see my
> card any where.
> > > The module I use is the snd-intel8x0. I'll check it out. Would
intel810
intel810i
intel820
one of these?
---
On Mon, 3 Nov 2003 00:11:13 -0500
Marcel <[EMAIL PROTECTED]> wrote:
> I installed jack for evaluation.
>
> Whenever jackd is running, no other application can use alsa. Is this
> normal?
can other alsa-audio programs run at the same time? it depends on if
your soundcard supports hw-mixing.
-
docs [there's one called alsa-lib doc
or something but it really leads to a description of the plugins]..
Florian Schmidt
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On Thu, 13 Nov 2003 18:28:26 -0600
Matthew Landry <[EMAIL PROTECTED]> wrote:
test for mathew..
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On Fri, 14 Nov 2003 15:53:21 +0100
audio2 <[EMAIL PROTECTED]> wrote:
> Hi
>
> Is there anybody who knows a good midi solution for via82xx, alsa
> and .asoundrc (asound.conf).
>
> Please let me know.
Please be more precise in what you want to achieve, what you have
already tried and why it didn
On Fri, 14 Nov 2003 16:43:30 -0600
Matthew Landry <[EMAIL PROTECTED]> wrote:
> Forgive my lack of experience, but when I do a configure, make and
> install it doesn't seem to make the alsamixer file. That is, when I
> type alsamixer, I just get an error.
Hi, make sure you have all the modules l
On Sat, 15 Nov 2003 11:22:35 -0700
Gianmarco Di Loreto <[EMAIL PROTECTED]> wrote:
> Hi guys! I have this nightmare: I managed to get the spdif output to
> work with xine, and now it works great! But I'mm' still going crazy
> trying to enable the spdif output by default for all my sound
> applic
On Sat, 15 Nov 2003 11:44:36 -0700
Gianmarco Di Loreto <[EMAIL PROTECTED]> wrote:
> Thanks Florian, but whatever and wherever i write h"¨hw:0,0¨ xmms
> gives me an error (mixer error)...and second, I don have a .asoundrc
> file!! Only asound.state...
Ok, create a file call
On Sat, 15 Nov 2003 12:16:31 -0700
Gianmarco Di Loreto <[EMAIL PROTECTED]> wrote:
> Ok...I tried what ou said...I created the file in /root (since Im'
> root), the output for the devices is this
>
>0: [0- 0]: ctl
> 16: [0- 0]: digital audio playback
> 24: [0- 0]: digital audio capture
>
On Sat, 15 Nov 2003 18:58:28 +0100
Florian Schmidt <[EMAIL PROTECTED]> wrote:
> On Sat, 15 Nov 2003 11:44:36 -0700
> Gianmarco Di Loreto <[EMAIL PROTECTED]> wrote:
>
> > Thanks Florian, but whatever and wherever i write h"¨hw:0,0¨ xmms
> > gives me an er
On Sat, 22 Nov 2003 10:09:53 +0100
Friedrich Ewaldt <[EMAIL PROTECTED]> wrote:
> how about:
> "try -ao alsa9 -abs [1234]. 1 through 4 sets different
> buffer/periodsizes." (don't know where I found this, maybe in the
> manpages or some mailing list).
> I don't use SPDIF output, but get the same
On Sun, 23 Nov 2003 13:50:05 +0100
Marco Tommaso Coiatelli <[EMAIL PROTECTED]> wrote:
> Hi All! I'm wrtiting to ask something about audio quality. The output
>
> of my soundcard goes directly to the hi-fi but I feel the quality of
> the sound really low.. With the same hi-fi I listen to music wi
to know: Is this plugin interface available
for ctl's too? can i fake a pcm-control that really adjusts a pcm
plugin's volume [so that xmms doesn't alter my cards pcm volume but only
its own stream]...
userspace progs: which purpose does each of them have? example of usege,
et
to the questions i sent. :-)
I will be glad to read it then. I would also like, if my aruments
convinced you that at least a basic treatment of the architecture of
alsa is necessary, assist you in writing something up.
It would be great to have a document that covers all aspects a (new)
user need
On Mon, 24 Nov 2003 04:46:13 +0100
David Garcia Garzon <[EMAIL PROTECTED]> wrote:
> Thanks, any help wil be wellcome. And, yes, i am convinced: a basic
> treatment of the vertical view of the architecture will be present,
> but not as the key structure of the document, i prefer an horizontal
> div
On Mon, 24 Nov 2003 18:05:40 +0100
Maarten Vanraes <[EMAIL PROTECTED]> wrote:
> hi,
>
> the /dev/sound/dsp uses 3 and the /dev/sound/adsp uses 12 . I want to
> switch the char minor of those devices, because I can't record using
> dsp, but I can record through adsp...
>
> how can I accomplish th
]. I must admit though that i
have no interest in 3D-sound, etc, therefore i haven't played aorund
with those sliders..
Florian Schmidt
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On Fri, 28 Nov 2003 14:52:55 +0100 (CET)
Jaroslav Kysela <[EMAIL PROTECTED]> wrote:
>
> Look for IEC958.
Hi, i heard, these IEC thingies are tandardized in some way? Is there a
list available on what each of these mean? and if there were, wouldn't
it be feasable to translate them to human reada
On Mon, 01 Dec 2003 22:06:50 +0900
Patrick Shirkey <[EMAIL PROTECTED]> wrote:
> As some of you already know I am the occasional webmaster/docs
> maintainer for ALSA.
>
> I have been planning some additions to the site to make it more
> accessible but before I jump into anything that could be co
On Mon, 1 Dec 2003 20:10:03 +0100
Florian Schmidt <[EMAIL PROTECTED]> wrote:
[snip]
> Feel free to add meat, modify, or flame me to death :)
I must add, that for many of these things it is possible to reuse
existing docs. There just needs to be a clearly structured place from
whic
On Fri, 5 Dec 2003 09:23:27 -0800
Andrew Burgess <[EMAIL PROTECTED]> wrote:
> >> Have you selected the microphone as capture source and raised the
> >> capture level in alsamixer?
> >
> >Alsamixer shows capture set to max. Although the "mic boost"
> >indicates +20db, but I'm unable to adjust it t
Hello,
did anyone manage to get ac3-passthrough working on the envy24ht chipset?
I get the following error when running "mplayer dvd:// -ao alsa9:spdif -ac
hwac3"
Checking audio filter chain for 48000Hz/2ch/16bit -> 48000Hz/2ch/8bit...
AF_pre: af format: 2 bps, 2 ch, 48000 hz, big endian AC3
AF
Hi,
i played around with seq24 and saw that it is one of those old apps that
insist on using hw midi ports and won't polay nice with alsa-patch-bay.
Well then, i thought, the solution to this is to use the virmidi module
which provides a virtual hw midi port [or better: several of them].
Now my
On Tue, 6 Jan 2004 16:23:41 +0100 (CET)
Jaroslav Kysela <[EMAIL PROTECTED]> wrote:
> On Tue, 6 Jan 2004, André Heßling wrote:
>
> > Hi!
> >
> > I am using the latest alsa rc and want to try out dmix with an
> > intel8x0. XMMS is working just fine and I am able to play a sound
> > with aplay at t
On Wed, 07 Jan 2004 10:49:21 +0100
Alex Haan <[EMAIL PROTECTED]> wrote:
> Hi, this seems to be the same error I get.
> see '[Alsa-user] "modprobe snd-ymfpci" results in "init_module: No
> such device"' just a few messages before yours. At least I'm not alone
> on this.
>
> I didn't change the /et
On Wed, 07 Jan 2004 01:49:51 +0100
Berend Veldkamp <[EMAIL PROTECTED]> wrote:
> During the bootprocess, or if I modprobe snd-ens1371, I get the
> following errors
> /lib/modules/2.4.22-1.2115.nptl/kernel/sound/snd.o: unresolved symbol
> proc_symlink
check the new FAQ on the alsa wiki:
http://a
On Wed, 7 Jan 2004 13:08:18 +0100
Petr Kopecký <[EMAIL PROTECTED]> wrote:
> Hi,
>
> somebody send me a working patch for emu10k1 to enable routing from
> line-in to front and rear but my HDD is dead and I lost all my data
> including this patch.
>
> Could you send it to me once more, please.
I
Hi,
i wanted to let you guys know that there's a new [user maintained] FAQ
in the alsa-wiki. Check it out and fill it with some meat :)
http://alsa.opensrc.org/index.php?page=FAQ
Discussions on the format are welcome, either in this thread or on the
"ThisWiki" page of th wiki:
http://alsa.ope
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