10.0-29-amd64 from Devuan Chimaera,
snd-seq-oss is not even listed.
HTH
--
Robert Riches
spamtra...@jacob21819.net
(Yes, that is one of my email addresses.)
Original Message
> Message-ID:
> Date: Sun, 19 May 2024 20:51:34 +0200
> To: alsa-user@lists.sourceforge.net
&g
7;,index=2
numid=23,iface=MIXER,name='IEC958 Playback Switch',index=3
numid=29,iface=MIXER,name='IEC958 Playback Switch',index=4
numid=6,iface=PCM,name='ELD',device=3
numid=31,iface=PCM,name='Playback Channel Map',device=3
numid=12,iface=PCM,name='ELD',
#x27;ll probably have to plug
in a set of separate speakers and go back to the motherboard's
audio device.)
Thanks,
Robert
___
Alsa-user mailing list
Alsa-user@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/alsa-user
Hi,
If I have a USB soundcard with say 8 channels, how do I route channels 7+8 to a
stereo capture device?
Regards
/R
The information in this email (including any attachments) may contain
confidential and/or proprietary material. Any review, retransmission or use of
this information by person
that should tell you whether that is _ALL_
of the problem.
HTH
Robert
> From: Fernando Carello
> Date: Sun, 31 Jan 2021 13:20:19 +0100
> To: Clemens Ladisch
> Cc: alsa-user@lists.sourceforge.net
>
> So, I've seen that a working MIDI keyboard "creates&quo
S16_LE-r48000-c_2.raw \
-r 44100 something.wav
Crop it as needed.
Optional: Convert it to MP3:
ffmpeg -i something.wav something.mp3
HTH
Robert
> Date: Sun, 27 Dec 2020 10:25:14 +0100
> From: tu...@posteo.de
> To: alsa-user@lists.sourceforge.net
>
> Hi,
>
>
Setting
options snd-usb-audio index=5
in alsa-base.conf seems to do the trick, thanks!
Regards
/R
-Original Message-
From: Ralf Mardorf
Sent: Sunday, 22 November 2020 16:10
To: alsa-user@lists.sourceforge.net
Subject: Re: [Alsa-user] USB ALSA card number
Hi,
I'm using the below /etc/
I have a system where I need USB attached audio devices to start numbering from
ALSA card5 and upwards (i.e. card0 to card4 are reserved). Is this possible?
Regards
/Robert
The information in this email (including any attachments) may contain
confidential and/or proprietary material. Any
HTH
Robert
> Date: Sun, 26 Jul 2020 17:43:12 +1000
> From: Philip Rhoades
> To: ALSA user
> Reply-To: p...@pricom.com.au
>
> People,
>
> I am not sure what is going on - I seem to have had increased sound
> problems on recent versions of Fedora (30-31). I have been
be a better
> idea.
Interesting though. Can this loopback routing be done entirely within a
.asoundrc configuration?
Regards,
/Robert
The information in this email (including any attachments) may contain
confidential and/or proprietary material. Any review, retransmission or use of
this
I’d like to know if there is any way to get the following sound chain through
ALSA:
Mediaplayer -> dmix -> LADSPA -> plughw:0,0
For my application it is crucial that the LADSPA plugin be applied AFTER dmix.
Regards
/Robert
The information in this email (including any attachments) ma
On a similar note, the sample rate seems to be specified when instantiating the
PCM plugin, but I’m not aware if frames per buffer is ?
From: Robert Bielik
Sent: Monday, 18 May 2020 10:39
To: alsa-user@lists.sourceforge.net
Subject: ALSA PCM plugin lifetime
Hi all,
I have a system setup where
, whether or not there is any audio running? (there will be an
external application that communicates with the plugin through unix sockets)
Regards
/Robert
The information in this email (including any attachments) may contain
confidential and/or proprietary material. Any review, retransmission or
What is the path or URL to that list? On sourceforge, I see
alsa-announce, alsa-cvslog, and alsa-user, but no alsa-dev*.
Thanks,
Robert
> Date: Mon, 31 Dec 2018 22:48:50 -0800
> From: frede...@ofb.net
> To: "Robert M. Riches Jr."
> Cc: alsa-user@lists.sourceforge
. With both those operating systems, the USB
sound devices had a very adequate volume control range, certainly a
whole lot more than 1dB.
Being as the USB sound devices are the ones my wife uses, I'd be
extremely grateful if somebody could point me a to way (if one
exists) to a useful
> So I guess its a ALSA version issue ☹
ALSA version where hints work is 1.1.3 and where they don't version is 1.0.29.
/R
--
Check out the vibrant tech community on one of the world's most
engaging tech sites, Slashdot.or
+.
Whereas on the target platform:
Advanced Linux Sound Architecture Driver Version k4.1.25.
So I guess its a ALSA version issue ☹
Regards
/R
>
> Regards
> /Robert
>
> > -Original Message-
> > From: Robert Bielik
> > Sent: den 6 september 2018 14:46
&g
SA is used on a particular platform ?
Regards
/Robert
> -Original Message-----
> From: Robert Bielik
> Sent: den 6 september 2018 14:46
> To: Alsa User
> Subject: Re: [Alsa-user] List devices
>
> Hi Clemens,
>
> Thanks for hint 😊 I added hint descript
just basic name hints
defaults.namehint.basic on
# show extended name hints
defaults.namehint.extended off
Anything else I need to do ?
Regards
/Robert
> -Original Message-
> From: Clemens Ladisch via Alsa-user
> Sent: den 6 september 2018 11:34
> To: alsa-user@lists.sourcef
ev/urandom' : Signed 32 bit Little Endian, Rate 96000 Hz,
Stereo
I'm working on an ALSA C++ backend so in order to use the device I need to be
able to list it, so how to ?
Regards
/Robert
--
Check out the vibra
Hmm... I was a bit too fast there...
> aplay -L
> null
> Discard all samples (playback) or generate zero samples (capture)
> pulse
> PulseAudio Sound Server
Can you try playing through pulseaudio with:
aplay -D pulse test.wav ?
Regards
/R
---
Yó napot kivánok! 😊
Take a look at https://alsa.opensrc.org/Dmix , dmix is the ALSA plugin you
should use for this.
Regards
/Robert
> -Original Message-
> From: Csányi Pál [mailto:csanyi...@gmail.com]
> Sent: den 10 februari 2018 12:02
> To: Alsa User
> Subject: [Als
> I'm trying to reorder my soundcards on a RPi so that the I2S based cards
> always is index zero. I looked at the docs
> (https://alsa.opensrc.org/MultipleCards), which just says, "easy peasy, just
> use options snd slots=this, that". Problem is that nowhere is it documented
> WHAT "this, that" is
"this, that" is! Is it card name ? Is it
module name ? If latter, how do I get driver name ?
Help would be appreciated.
Regards
/Robert
--
Check out the vibrant tech community on one of the world&#
Would the dshare plugin do this for you ?
http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html
Regards
/Robert
> -Original Message-
> From: Samuel Nicholas [mailto:nicholas.sam...@gmail.com]
> Sent: den 1 februari 2018 22:09
> To: alsa-user@lists.sourceforge.n
gin do the same job as dmix, i.e. mix together
applications using the device ?
Regards
/Robert
--
Check out the vibrant tech community on one of the world's most
engaging tech sites, Slashdot.org! http://sd
> > is there some ALSA plugin that can coalesce buffering ? Meaning that
> > the plugin can take f.i. larger period_size than what the dmix device
> > is working with ?
>
> What problem would that solve?
Not sure. It would allow clients connecting to that device to have a more
relaxed callback s
se the dmix rate. (If you wanted to, it would
> be possible to put a "plug" or "rate" plugin on top of it.)
On that note, is there some ALSA plugin that can coalesce buffering ? Meaning
that the plugin can take f.i. larger period_size than what the dmix device is
w
> The reason is that for my project I need to have as low a latency as possible
> in
> the dmix chain. Is there any other plugin doing the same thing as dmix... but
> working ? 😊
More specifically, I'd need a mixing plugin that does not do sample rate
conversion, i.e. each client connecting to i
Hi, I'm using the audioinjector octocard on a R Pi 3, and I have a problem
where the system default dmix (dmix:0,0) plays just fine (via aplay), but my
own defined dmix device occasionally stops streaming with a xrun condition:
Status(R/W):
state : RUNNING
trigger_time: 13953.124684
);
playback.putBuffer(out_buffer);
}
}
Regards
/R
> -Original Message-
> From: Robert Bielik
> Sent: den 15 januari 2018 17:59
> To: alsa-user@lists.sourceforge.net
> Subject: RE: Strange i/o problem
>
> Yet more info, the output of snd_pcm_hw_params_dump and
>
age-
> From: Robert Bielik
> Sent: den 15 januari 2018 17:47
> To: Robert Bielik ; alsa-user@lists.sourceforge.net
> Subject: RE: Strange i/o problem
>
> Ah, forgot to mention a couple of things, this is on a Raspberry Pi 3 with
> Raspbian Stretch, the rendering thread is
, 666, 722, 2.2253
min, mean, max, stddev: 656, 666, 680, 1.57805
min, mean, max, stddev: 643, 666, 683, 1.54424
(which to me looks more than OK)
> -Original Message-
> From: Robert Bielik [mailto:robert.bie...@dirac.com]
> Sent: den 15 januari 2018 17:41
> To: alsa-user@lists.sou
I have a strange problem: I'm trying to pipe audio input -> output using a I2S
device @48000 Hz and 32 frames buffer size and 2 periods, to get as low a
latency as possible.
It works nicely if I either:
1. Use capture + playback and record capture to a wav file (sounds fine).
2. Use playback onl
Ok, hehe... found the problem, I was running gdbserver as root so it was the
wrong .asoundrc I changed...
> -Original Message-
> From: Robert Bielik [mailto:robert.bie...@dirac.com]
> Sent: den 8 januari 2018 10:30
> To: alsa-user@lists.sourceforge.net
> Subject: [Alsa-
I've come across an odd behavior: If I add a dummy pcm in .asoundrc :
pcm.dummy {
type plug
slave.pcm "plughw:0,0"
}
I can see it listed with aplay -L.
However, my own code, which uses the same exact mechanism as aplay does
(snd_device_name_hint) does NOT list the dummy device.
Ideas?
Hi Clemens,
Hah, you're quite correct, I handle error conditions by throwing exceptions,
and I think those cases are indeed induced by opening the device, but not
releasing it properly. Using exception safe coding, it now seems to work a lot
better 😊
Thanks!
/Robert
> -Original
Mind you, this works nicely:
> aplay -D default:CARD=MOXF6MOXF8 test.wav
So I guess I must be doing something wrong ☹
/R
> -Original Message-
> From: Robert Bielik [mailto:robert.bie...@dirac.com]
> Sent: den 7 januari 2018 10:16
> To: alsa-user@lists.sourceforge.net
&
ry snd_pcm_open on the IDs, most of which I get -EBUSY.
Why is this ? There are no other applications using the devices. F.i. I cannot
open the "default:CARD=MOXF6MOXF8" device, using that exact string as id to
snd_pcm_open, I get -EBUSY eve
I've used the ALSA LADSPA PCM plugin, and it works nicely. However, I'd like to
use LV2 plugins aswell. Is there such a project active somewhere ?
Rgrds
/R
--
Check out the vibrant tech community on one of the world's m
> -Original Message-
> From: Robert Bielik [mailto:robert.bie...@dirac.com]
> Sent: den 21 juli 2017 11:46
> To: Clemens Ladisch ; alsa-user@lists.sourceforge.net
> Subject: Re: [Alsa-user] Route input to output with minimal latency
>
> Dear Clemens,
>
> Thank you so much f
Dear Clemens,
Thank you so much for the alsaloop tip, I just ran it with:
> chrt -rr 70 alsaloop -f S32_LE -C plughw:0 -P plug:ladspa -l 48
And it works perfectly, exactly what I needed! 😃
Regards
/Robert
> -Original Message-
> From: Robert Bielik [mailto:robert.bie...@
plugin, thus I only need to route ALSA input to output in the application 😊
Regards
/Robert
--
Check out the vibrant tech community on one of the world's most
engaging tech sites, Slashdo
ble ?
Regards
/Robert
--
Check out the vibrant tech community on one of the world's most
engaging tech sites, Slashdot.org! http://sdm.link/slashdot
___
Alsa-user mailing list
A
Hi all,
Started experimenting using LADSPA plugins with the ALSA LADSPA PCM plugin, and
it works nicely. However, it seems only the PCM interface is exposed from the
plugin.
Question is if there is a way to expose the LADSPA plugin parameters directly
to alsamixer ?
Regards
/Rob
wanted. Also,
I have a thin/zero-client setup at home, with .asoundrc tailored
to push sound to the appropriate audio device for each session.
Quite often, Firefox misdirects the sound from my wife's browser
session to my monitor. That and a few other things are raising
my level of unhappiness
f
If all else fails, inform us your distribution and release so
someone who knows that distribution can give a more precise
answer.
HTH
Robert
> From: Kristoffer Gustafsson
> Date: Tue, 1 Nov 2016 02:39:04 +0100
> To: alsa-user@lists.sourceforge.net
>
> hi.
> I need to add thing
on't get fixed--ever.)
One alternative that IIUC _should_ work is to use ~/.asoundrc for
a per-user workaround rather than /etc/asound.conf for an all-user
workaround.
Sorry, but I don't know anything that would likely help system
sounds for a "desktop environment".
Thanks
> From: José Luis Artuch
> To: "Robert M. Riches Jr."
> Cc: alsa-user@lists.sourceforge.net
> Date: Fri, 29 Apr 2016 23:55:56 -0300
>
> El vie, 29-04-2016 a las 17:26 -0700, Robert M. Riches Jr. escribió:
> > The module's options are likely at least a st
sulted in xruns.
HTH
Robert
> From: José Luis Artuch
> To: alsa-user@lists.sourceforge.net
> Date: Fri, 29 Apr 2016 16:49:07 -0300
>
> Hi,
> I fixed the names of each real sound card (card_0, card_1, card_2, ...).
> Now, loading the snd-aloop module for all real sound c
apacitor solution, you'll also
need something to provide a DC path across the phone line or the
phone company switch will hang up the call. Here's a schematic I
posted to another forum:
https://tech.lds.org/forum/download/file.php?id=2636
If maximum economy is required, e
.NVidia { type hw; card NVidia; }
ctl.NVidia { type hw; card NVidia; }
-pcm.!default pcm.Intel
+pcm.!default {
+ type plug
+ slave.pcm "Intel"
+}
ctl.!default ctl.Intel
HTH
Robert
--
___
the difference. Sox might have a filter for that. Otherwise, a
C program wouldn't take long to write to do that. An alternative
to saving the (noisy) stereo file would be to output RAW samples
and do the conversion usin
I have not used.
Second, the module needs to be loaded into the running kernel.
That is done by the modprobe command.
HTH
Robert
> Date: Mon, 17 Aug 2015 20:41:08 +0200
> From: "F. Dols"
> To: "Robert M. Riches Jr." ,
> alsa-user@lists.sourceforge.net
>
&g
dule:
sudo modinfo indigodjx
IME, sound cards normally default to enabled, but perhaps this
driver is different. You might need to manually force it to be
enabled.
HTH
Robert
> Date: Sun, 16 Aug 2015 20:14:10 +0200
> From: "F. Dols"
> To: alsa-user@lists.sourceforge.ne
p NetJACK.
HTH
Robert
> From: daggs
> To: li...@lazygranch.com
> Date: Thu, 26 Mar 2015 08:01:40 +0100
> Cc: alsa-user@lists.sourceforge.net
>
> thanks for the info guys, but it isn't quite what I need. I need to stream
> any sound from one computer to another.
>
&
t lets the playback client get too far
ahead, which can cause difficulties in some situations.
HTH
Robert
> From: daggs
> To: alsa-user@lists.sourceforge.net
> Date: Wed, 25 Mar 2015 20:25:54 +0100
>
> Greetings,
>
> is there a way to stream sound from one machine to another vi
type of thing? If so, is there a any
less drastic measures to prevent the kernel panics?
Thanks,
Robert
--
Download BIRT iHub F-Type - The Free Enterprise-Grade BIRT Server
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o see whether the OPL3 drivers were enabled.
Regarding compiling from source, is there a file with a name
similar to 'bootstrap'? Sometimes that is a precursor to
./configure. If there's an INSTALL or README* file, that often
contains compilation instructions.
HTH
Robert
might need resyncing.)
Robert
--
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lay. It's not that it plays some modes
but not others; it doesn't play properly with any mode. I reran
some tests, based on your advice. Info interspersed below.
Thanks,
Robert
> Date: Fri, 29 Aug 2014 22:46:27 -0700
> From: chris hermansen
> To: "Robe
3.8.13 kernel? It will be feeding a church PA
system but studio-quality sound is not necessary.
Thanks in advance,
Robert
Details:
Output of "uname -a":
Linux box 3.8.13-tinycore64 #777 SMP Fri Oct 18 15:13:45 UTC 2013 x86_64
x86_64 x86_64 GNU/Linux
Ou
had no reply. Things I
have tried from other post reading on various forums likewise have had
no effect. alsamixer levels are all unmuted and (except for Master) at
maximum levels. Sound output is very good and both headphones and HDMI
sound is also good.
thanks in advance for any help
name=robert
Hello,
I hope posting here is appropriate. A recent install of mint 16 petra
(cinamon) on an ASUS F550L laptop with intel HDA and HDMI sound cards.
Two problems (that I can see).
Pulse audio Volume control Inputs shows 2 microphone options - 1
labelled Internal Microphone and 2 labelled Microphon
Like I suspect others, I joined this list in hopes of getting some advice
with a problem that has thus far eluded me. I'm not an ALSA expert by
any means, nor a programmer, but I consider myself a decent
troubleshooter, and thought that by sharing what I've done it may
trigger some ideas from tho
alized. Perhaps playing a short piece of
silence via PyAudio before playing your real content via AlsaAudio
would be the most practical solution for that situation.
One other thing I remembered about PyAudio with my program is I
see a lot of whining about 'Unable to find definition ...',
ther possible issue if if PyAudio vs. AlsaAudio set up the
card to different sample rates, one of them using software to
resample from the input WAV file to match what they set the card
to.
HTH
Robert
--
Android apps ru
I have a HP ProDesk 600 G1 machine which seems to have some strange
behavior with the front headphone jack(s) running under Fedora 20
(running 3.13.2 kernel):
This machine has two front jacks: a headphone jack and a
microphone/headphone jack. When the headphones are plugged into the
headphone jack
ChaosEsque Team,
Congratulations on being first person from a mailing list that I
have ever added to an email deny list. If you can't accept
reasonable advice without foul-mouthed reviling and threats of
violence against a benefactor, you aren't allowed in my inbox.
Robert
> Date
captured from a USB DTV tuner via a Raspberry Pi's USB
port. (During video recording, the stream was sent back out the
network port while it was coming in via USB.) If the Raspberry
Pi can handle HD video, it should be able to handle the vast
majority of audio tasks.
HTH
Robert Riches
ed on
a wild guess that there might be firmware that gets downloaded to
the device that enables 13.10 to see it. (I have a USB-attached
TV tuner that enumerates as one USB device, installs firmware,
disconnects as a USB device, then re-enumerates as a different
USB device
I probably don't know any answers, but would like to make sure I
at least understand the question. Are you trying do AGC on a
pair-wise basis? Or, is it something else you're trying to do?
Thanks,
Robert Riches
> Date: Sun, 13 Oct 2013 15:02:42 +0200
> From: Paolo Bolzoni
> Date: Fri, 20 Sep 2013 09:19:48 +0200
> From: Clemens Ladisch
> To: "Robert M. Riches Jr." ,
> alsa-user@lists.sourceforge.net
>
> Robert M. Riches Jr. wrote:
> > I'm seeking suggestions for a low latency usb (or PCIE) sound
> > interface (or ca
rnelspace.
Anyway, are there USB sound interfaces with latencies no more
than 25-50msec? (Or, are they around 500msec latencies like I
hear at work on Doze-based laptops over Lync?)
Thanks,
Robert Riches
--
LIMITED TI
dly from the sending
client, which overruns the buffer of the receiving client.
I asked here a while ago if there were solutions. Yes, I ought
to file a bug report if I could scrape together a few minutes to
do so.
Robert
--
G
haven't already tried this, if it were
my program I would instrument the program to print out buffer
pointers for each packet of samples sent to Alsa (or wherever
you're sending the data). If it were my program, it would
likely turn out that I had made som
e directory is not accessible.
> Is your home directory nfs mounted or something?
> The audio open error seems to be from "snd_pcm_open"
> Also try
> -Dhw:0
> or
> -Dplughw:0
> as an argument
ient with absolutely no way to specify a sound
unit to use, you could use separate user accounts, each with a
.soundrc file that directs ALSA to use a given device as the
default audio device. There is documentation on the details to
do that, probably better than I could remember to tell you.
HTH
Rob
am gets overruns. I have yet to hear an
answer or other reply to that problem/question. I'm about to
file a bug report about the problem.
Fortunately for you, in your case, if your SDR sends at exactly
the right rate, you might be okay once the buffers all find their
pace and stride.
HTH
Ro
pback device control the frame rate to
be what the applications opened it for?
Is there a solution to keep the above sender and receiver
command chain from having overruns?
Thanks,
Robert
--
Everyone hates slow websites
Hi,
this problem is fixed by upgrading to kernel 3.6.
cu romal
Am 15.08.12 10:15, schrieb Daniel Mack:
> On 13.08.2012 19:46, Felix Homann wrote:
>> Am 13.08.2012 16:59, schrieb Robert M. Albrecht:
>>> I can't play audio on the FTU.
>>>
>>> [...
ntu CD, that
seems to be working.
I've read somewhere before, that the Tenor TE8802 corrupts the feedback
messages
it sends to the linux host. If that is the case, how can it be fixed?
Thanks again for your help
Robert
--
>
> There is no information about that in the alsa-info output, and the
> snd-usb-audio driver is not loaded.
Can someone please explain how I can load the snd-usb-audio driver?
Thank you
--
Got visibility?
Most devs h
I am 99% sure it is Tenor 8802 chip. Does this help with anything?
--
Got visibility?
Most devs has no idea what their production app looks like.
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Clemens Ladisch googlemail.com> writes:
>
> Robert wrote:
> > I have tried the DAC on two distros, Ubuntu 12.04.1 and Mageia 2 (both use
> > Alsa 1.0.25) but it only works in Ubuntu. Mageia displays the name of the
DAC
> > in dmesg, but it is not visible in /proc/
ased on Linux): http://forums.slimdevices.com/showthread.php?94512-
Announce-Enhanced-Digital-Output-app-USB-Dac-and-192k-Digital-
Ouput&p=718870&viewfull=1#post718870
Thank you for your input!
BR, Robert
--
Every
rb 0, error -2: endpoint not enabled
Aug 13 16:45:44 chessur kernel: [ 811.391616] ALSA
sound/usb/endpoint.c:867 -- ep_num 81 pipe 34432
Aug 13 16:45:44 chessur kernel: [ 811.391618] ALSA
sound/usb/endpoint.c:868 -- type 0 flags 2
cu romal
Am 12.08.2012 20:37, schrieb Daniel Mack:
>
>
ernel: [ 172.846017] ALSA
sound/usb/mixer.c:786 usb-audio: set quirks for FTU Effect Feedback/Volume
cu romal
Am 12.08.2012 13:47, schrieb Daniel Mack:
> Hi Robert,
>
> On 11.08.2012 17:45, Robert M. Albrecht wrote:
>> today came a new kernel, still broken:
>>
&
t; I've just tested with a fresh kernel from
>> git://git.kernel.org/pub/scm/linux/kernel/git/stable/linux-stable.git
>> and don't see any problems. I've also tested with Fedora's packaged
>> 3.5.0-2.fc17 kernel. No problems here either.
>
> Thanks fo
Interface level)
bDeviceSubClass 0
bDeviceProtocol 0
bMaxPacketSize064
bNumConfigurations 1
Device Status: 0x0001
Self Powered
[root@chessur ~]#
Am 05.08.2012 20:24, schrieb Daniel Mack:
> On 04.08.2012 13:03, Robert M. Albrecht wrote:
>> Hi,
&g
Hi,
after upgrading to Kernel 3.5 I get this when switching on the M-Audio FTU
Aug 4 12:53:06 chessur kernel: [ 211.714699] usb 1-1.2.1.1: new
high-speed USB device number 8 using ehci_hcd
Aug 4 12:53:06 chessur kernel: [ 211.801149] usb 1-1.2.1.1: config 1
interface 3 altsetting 0 bulk end
ieben:> On
22.07.2012 15:28, Robert M. Albrecht wrote:
> > Hi Dave,
> >
> > the Rode Podcaster is a microphone, optimized for recording voice for
> > podcasting.
> >
> > It has an integrated audio interface and connects via usb to the computer.
> >
> &g
audio interface.
The firmware seems to fix some compatibility issues Without this
firmware update Windows XP could record, but Windows Vista could not.
Alsa can also playback via the integrated headphone connector does also
work.
Fantastic.
Regards,
Robert
Am 21.07.2012 15:45, schrieb Daniel
Data
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Am 20.07.2012 17:37, schrieb Daniel Mack:
> On 20.07.2012 17:31, Robert M. Albrecht wrote:
>>
>> Hi,
>>
>> I got a Rode podcaster and tried to connect it to my Linux system.
>
Hi,
I got a Rode podcaster and tried to connect it to my Linux system.
Audacity actually records from the device, but it's only digital zero.
What does "ALSA sound/usb/clock.c:236 8:2:1: cannot get freq at ep 0x82"
mean ?
cu romal
Jul 20 16:34:04 chessur kernel: [ 1710.985946] usb 1-1.1.3:
DP,pcm=8
[23.838] (II) Using input driver 'evdev' for 'HDA NVidia HDMI/DP,pcm=8'
[23.838] (II) Loading /usr/lib64/xorg/modules/input/evdev_drv.so
[23.838] (**) HDA NVidia HDMI/DP,pcm=8: always reports core events
[23.838] (**) HDA NVidia HDMI/DP,pcm=8: Device: &quo
BTW ALSA version is 1.0.25
On Tue, Jul 10, 2012 at 2:36 PM, Robert Krakora <
rob.krak...@messagenetsystems.com> wrote:
> Hi All,
>
> I have a Zotac HD80 with a NVIDIA GeForce 520M. There is only one HDMI
> output. However, ALSA shows three HDMI outputs but only one is f
Hi All,
I have a Zotac HD80 with a NVIDIA GeForce 520M. There is only one HDMI
output. However, ALSA shows three HDMI outputs but only one is functional
(below). With debug enabled dmesg trace (further below) indicates that the
snd-hda-codec-realtek kernel module is being used instead of the
sn
Thanks Torsten for your very good and exact answer
i now use parallel to the preshutdownscript of mythtv your script in an
light edited style
if silence is detected your script kills vlc, mythtvfrontend and
rhytmbox instances. Then my existing preshutdown script returns 0 so the
mythbacke
alternative.
Greetings from Robert
from Cottbus, Germany !
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