On Wed, 17 Apr 2024 at 04:32, G S wrote:
>
> Hello ALSA community.
>
> I have a newer Lenovo laptop that has a built-in AMD audio co-processor which
> requires adding an entry in the lookup table for the acp6x kernel module
> (acp6x-mach.c) so the the digital mic is properly detected.
>
> I foun
On Wed, 8 May 2024 at 19:53, Franco Martelli wrote:
>
> Hello everyone,
>
> Basically I've the same issue described here:
> https://askubuntu.com/questions/1180389/speaker-test-returns-all-6-channels-to-front-speakers
>
What does this do for you:
speaker-test -c6 -twav
It should play some voice
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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seem fatal.
nonetheless, any ideas on how to fix that?
-JimC
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I've connected 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
e outlined 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.
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
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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
:
speaker-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 se
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 (yet). But one way
, 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
n
mono file. Which 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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A
1. In alsa speak that's 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 wrote:
> On Sun, 4 Dec 2016 20:18:24 +, zcx wrote:
>>I have a Delta 44 sound card here that
trap. It's faster for me on
my slow 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
--
_
just
--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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it, but the device 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 do
et a value of -65.00
For the API, the dB values are
(integer_value_returned_from_the_API_call / 100) in order to avoid
floating point.
Does this help?
Kind Regards
James
--
Th
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 wrote:
> Problem persists with 3.6.6 ... any hel
Problem persists with 3.6.6 ... any help please ?
On Mon, Oct 29, 2012 at 1:26 PM, James Pic 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
a-utils-1.0.25-r2).
-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
-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 wrote:
> Hello Xenia!
>the o
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 1.
channel capture device in USB, I think
you'll be disappointed. But there's a few dedicated devices that have
USB options like the Zoom R16 which might suit those needs.
- James
On 6/22/11, oskoff lovich wrote:
> Hi all of the list, i am thinking to buy a usb soundcard with 8
>
're OT for this list. You might check the alsa-devel mailing
list for guidance. Although I'm not on that list. alsa-project.org
should have info on that one.
- James
On 6/21/11, David Henderson wrote:
> Hi James, if I use "configure --prefix=/opt/staging/alsa" then I&
rc that's getting in the way.
- James
On 6/21/11, Pierre Habraken 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 Mosgalin
>> Subject: Re: [Alsa-user] plughw versus hw
ns (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 wrote:
> I think your statement here &
our
own and not part of team. But it's almost all open source so if you
can read the source, everything that you need to know is there in one
form or another.
- James
On 6/20/11, David Henderson wrote:
> On 06/20/2011 11:52 AM, Pierre Lorenzon wrote:
>> Hi,
>>
>>
>&g
tible with their package manager(s). Plus the typical
development role of 1001 ways to do one thing. Fortunately alsa is
still a bit old school. Or unfortunately depending on your POV.
- James
--
EditLive Enterprise is
in a way that you want to
use them. i.e. Timidity with sequencer 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 copio
d
other things that just need --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 wrote:
> Hi James, thanks for your help too. :) I'll provid
I'd recommend going 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 wrote:
> Thanks for the reply Pierre. I checked into the blfs book, but it
> merely says "these five chapt
n the lowest 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 wrote:
> I got a new computer (Zotac HD-ID40) ION2
>
> I have installed a
defaulting 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 wrote:
> Hello,
>
> Thank you.
>
> These are the differences of the two logs: http://pas
a common medium (flash drive) of course.
$ diff -a -U 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
gh it's still the
only way to tell some soundcards to record from PCM out 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 wrote:
>
pressing record. Beyond that, your guess is as
good as mine.
- James
On 5/20/11, Peter Hoffmann 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
> se
But it'd be nice to know how to
fix it, if I do run into it.
HTH,
- James
On 5/19/11, Y P 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 upgra
I also have a Delta 44 and it
works fine, with a little extra configuration in some cases.
HTH,
- James
On 5/1/11, Graham Dicker wrote:
> Dominique Michel wrote:
>
>> Le Fri, 29 Apr 2011 16:28:47 +0100,
>> Graham Dicker a écrit :
>>
>>> I have been recordi
d of odd for a 6 second capture don't you think? Or is
the result and the example unrelated?
- James
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w:0,2,0 -c 2 -r 96000 -d 6 -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
--
t it is on is maxed... Not
that I'd know how or IF you can choose CPU per task for non SMP aware
applications.
- James
On 4/8/11, Sergei Steshenko wrote:
> Hello,
>
> I've tried to run 'arecord' as part of simultaneous playback + capture "rig"
&
with that card. 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 wrote:
> Gentlemen, I cranked everything up but still not an ounce of sound.
> My ALSA information is lo
bably not your issue, so I'll stop rambling.
- James
On 4/2/11, James P. Early 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
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
stuff can be a little off. 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 wrote:
> Thanks, James. From a quick google search it looks like one can sti
ll. I may upgrade to 6.0 as soon as some of my
current projects are wrapped up. But for $80 off of craigslist, I'm
not complaining.
- James
On 3/15/11, Jonathan Wilkes wrote:
> Hi,
> I'm looking for a relatively inexpensive USB audio interface that will
> work (pa
I'm not sure of that locations default naming
convention as it probably varies between distros. alsa.conf?
asound.conf? +/- an /etc/ or /etc/alsa/ or /etc/sound/ or ??? And
various tricks of old to delete the asound.state file to force new
defaults. Located at /var/lib/alsa/asound.state o
ray matter only. Aften to create 5.1 ac3 audio. ffmpeg is limited
to creating 5.0 iirc. And other quirks for pretty much all of the
options.
- James
On 2/21/11, Sergei Steshenko wrote:
> On Mon, 21 Feb 2011 17:37:24 +0100
> Peter Hoffmann wrote:
>
> wa
>>
>> (BTW can an
cl/selinux) it's entirely possible that the
process is not that simple. But it could be.
- James
On 2/15/11, Marcin Szyniszewski wrote:
> Hello,
>
> I checked if alsa and stuff is working on other kernels - it seems it is
> working brilliantly! Mic and sound works fine!
> So t
r card is supported in the first place, which it appears to be or
it would have never worked. Otherwise we could troubleshoot for days
without more information about how you got to your current state of
affairs. Not that you'd have that standard M$ answer. I installed
AOL and now XXX doesn't
ource alsa-tools alsa-tools-gui alsa-utils alsamixergui
(removes the packages)
$ sudo apt-get install alsa alsa-base alsa-firmware-loaders alsa-oss
alsa-source alsa-tools alsa-tools-gui alsa-utils alsamixergui
(puts them back)
HTH,
- James
-
Be sure to reboot OR try to use the soundcard to
get the modules to auto magically load. 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 Szyniszew
. By all means nitpick that I used grep -i
"audio", versus awk '{ print $4 " " $9 }' | grep -i audio or
something.
- James
On 2/11/11, Bill Unruh wrote:
> On Fri, 11 Feb 2011, James Shatto wrote:
>
>> Note that * is a wildcard. So /dev/dsp* is any d
grep -i "snd"
$ cat /proc/asound/cards
- James
On 2/11/11, Torsten Schenk 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
hese days.
# /etc/init.d/alsa-utils restart
# /etc/init.d/udev restart
# groups
# 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 wrote:
> In article
> ,
>
On 24 November 2010 13:38, Grega Fajdiga 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.
can
figure 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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you get when compressed data is mistaken
for PCM.
When you try to record from S/PDIF with arecord, it only allows S16_LE
and S32_LE and I'm pretty sure I tried both.
James
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On Mon, 25 Oct 2010 15:18:03 +0100
James Le Cuirot 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
>
le to determine whether any 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
-
n do this. There are probably other solutions but I'm not
that familiar with OSS.
James
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i can't get 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-li
. If you don't see anything in /dev/snd
then the drivers are failing to load in the first place. The sound
drivers you have loaded are all for modems. What sound card do you
actually have? lspci output would help.
James
---
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configuration options in the GUI for stuff that the end user really
shouldn't need to know about.
James
--
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referably
> less than EUR250.)
Sounds like the ASUS Xonar D2X could work for you. It doesn't have
hardware mixing though.
James
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and start using them to si
of buffer or something un-useably small
like that.
I therefore only present the 4 stereo channels option in the driver.
Kind Regards
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 switc
eakers. Surely this should work?
Cheers,
James
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ttings being wider than what would fit in a 1024*768
> display,
> synth output being quieter than I expected,
>
> it is ideal for what I wanted.
>
It might be useful to use "alsamixer" for the mixer. It is a text based gui.
One of the controls
On 14 July 2010 09:24, Arthur Marsh wrote:
> Hi, I picked up a Soundblaster Audigy SB0570 card, hoping that it would
> have a wavetable synth like some of the EMU10K1/EMU10K2 based cards,
> however it identifies as:
>
> CA0106 :00:09.0: found PCI INT A -> IRQ 5
> snd-ca0106: Model 100a Rev 000
On 4 July 2010 16:51, Manuel Reimer 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.
On 4 July 2010 13:45, Manuel Reimer 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 player
Kind Regards
James
ry
the speakers in each socket until one works.
Kind Regards
James
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On 11 June 2010 20:09, Pandu Rakimanputra
wrote:
> Dear all alsa user
>
> It has been 4 month since my first email to this mailing-list and I still
> searching to make my Focusrite Saffire 6 USB works. I've got nothing. So I
> have a few question:
> 1) Is there any spesific requirement/criteria
* * *
29 * * *
30 * * *
- James
On 5/30/10, Jaroslav Kysela 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
actually work) 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
hich sound card 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 probl
priority to explore for me. 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 wrote:
> On Wed, Dec 9, 2009 at 5:16 AM, James Shatto 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 your
> &
On Tue, 8 Dec 2009 21:26:14 -0200
Kazuo Teramoto 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 way). Although
alsa apps reference them by -c # where the # matches
their designation 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
. Configure accordingly. I have a USB M-Audio Mobile 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 aud
nt. Bear in mind that if input is a
mic and the speaker is loud enough you'll get feedback. 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 nu
#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
> wrote:
>
>> James Gadsby wrote:
>> > Simply put, MIDI devices are not detected in t
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. N
nnels, I guess you want -Dsurround71.
Regards,
James
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think this happened with Windows but I don't have it installed anymore
so I couldn't swear to it.
Cheers,
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 no
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 (
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 Analo
g 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
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h
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 wrote:
>
>> hda_codec: Unknown model
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 NVidia,
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.
>
> ┌───
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
quit)]
odify the samples. 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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