ap',device=8
numid=24,iface=PCM,name='ELD',device=9
numid=34,iface=PCM,name='Playback Channel Map',device=9
numid=30,iface=PCM,name='ELD',device=10
numid=35,iface=PCM,name='Playback Channel Map',device=10
Thanks,
Robert
Original Message
> From: Alan Corey
> Date: Sun, 29 M
obably have to plug
in a set of separate speakers and go back to the motherboard's
audio device.)
Thanks,
Robert
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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
d 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" this device:
w \
-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,
>
> I want to record t
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
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
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 r
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
t
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
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
ating 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 usefully greater range of volume con
> 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
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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
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
' : 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
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Check out the vibrant tech community
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 <alsa-user@lists.
> 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"
ented WHAT "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
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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
plugin do the same job as dmix, i.e. mix together
applications using the device ?
Regards
/Robert
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> > 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
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
workin
> 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
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 <robert.bie...@dirac.com>; 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 Stretc
, 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
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.
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 Mess
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
&
d_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 every ti
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
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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...@
gin, thus I only need to route ALSA input to output in the application
Regards
/Robert
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ble ?
Regards
/Robert
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Alsa-u
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
so,
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 toward Fir
, inform us your distribution and release so
someone who knows that distribution can give a more precise
answer.
HTH
Robert
> From: Kristoffer Gustafsson <kg.kristof...@gmail.com>
> Date: Tue, 1 Nov 2016 02:39:04 +0100
> To: alsa-user@lists.sourceforge.net
>
> hi.
> I need 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,
Robert Ri
> From: José Luis Artuch <art...@speedy.com.ar>
> To: "Robert M. Riches Jr." <rm.ric...@jacob21819.net>
> 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. escri
.
HTH
Robert
> From: José Luis Artuch <art...@speedy.com.ar>
> 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 al
specially for a one-time event,
you can use a 1K resister instead of the choke and drop the MOV,
Zener diodes, and R1.
HTH
Robert
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; }
ctl.NVidia { type hw; card NVidia; }
-pcm.!default pcm.Intel
+pcm.!default {
+ type plug
+ slave.pcm "Intel"
+}
ctl.!default ctl.Intel
HTH
Robert
--
___
Alsa-user mailing
that. An alternative
to saving the (noisy) stereo file would be to output RAW samples
and do the conversion using Linux pipes--but that would probably
introduce considerable latency, depending on buffer sizes.
HTH
Robert
, 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 f.j.h.d...@gmail.com
To: Robert M. Riches Jr. rm.ric...@jacob21819.net,
alsa-user@lists.sourceforge.net
Hi,
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 f.j.h.d...@gmail.com
To: alsa-user@lists.sourceforge.net
Hi,
I need some
Robert
From: daggs da...@gmx.com
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.
I wonder, I can define in ~/.asoundrc
the playback client get too far
ahead, which can cause difficulties in some situations.
HTH
Robert
From: daggs da...@gmx.com
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 via a network
using alsa
, is there a any
less drastic measures to prevent the kernel panics?
Thanks,
Robert
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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
Date: Mon, 03 Nov 2014 03:19:19 +0100
To: Advanced
.)
Robert
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Comply
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 clherman...@gmail.com
To: Robert M. Riches Jr. rm.ric
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
Output of aplay -l:
List of PLAYBACK Hardware Devices
card 1
) libahci(F)
drm r8169 rtsx_pci mii(F) wmi video(F)
[ 1622.353034] CPU: 0 PID: 2666 Comm: kworker/0:1 Tainted: GF O
3.11.0-12-generic #19-Ubuntu
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anders...@fastmail.fm
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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
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
file to match what they set the card
to.
HTH
Robert
--
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Get your
directory: Permission denied', and so forth with PyAudio
during program initialization. With AlsaAudio, there is no such
complaining.
HTH
Robert
--
Android apps run on BlackBerry 10
Introducing the new BlackBerry 10.2.1
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
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: Thu, 30 Jan
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
rm.ric...@jacob21819.net
--
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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 once it has its firmware.)
Another long shot: Might it be flaky physical connectors?
HTH
Robert Riches (just another lurker
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 paolo.bolzoni.br
Date: Fri, 20 Sep 2013 09:19:48 +0200
From: Clemens Ladisch cladi...@googlemail.com
To: Robert M. Riches Jr. rm.ric...@jacob21819.net,
alsa-user@lists.sourceforge.net
Robert M. Riches Jr. wrote:
I'm seeking suggestions for a low latency usb (or PCIE) sound
interface (or card
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
--
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module accepts samples too rapidly 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
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 some errors in pointer
arithmetic or something like that.
HTH
Robert
and its argument and tee
both stdout and stderr from the aplay command to a file. Or,
run the strace aplay ... command from inside 'script'.
HTH
Robert Riches
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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
Robert
To: alsa-user@lists.sourceforge.net
From: li
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
Robert
Date: Fri, 01 Mar 2013 18:13
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
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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.
[...]
aplay -c 2 -D hw:0,0 test24.wav
Wiedergabe: WAVE 'test24
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
--
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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
--
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I am 99% sure it is Tenor 8802 chip. Does this help with anything?
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/showthread.php?94512-
Announce-Enhanced-Digital-Output-app-USB-Dac-and-192k-Digital-
Ouputp=718870viewfull=1#post718870
Thank you for your input!
BR, Robert
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Clemens Ladisch cladisch at 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/asound/cards so it can't
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:
On Aug 12, 2012 8:34 PM, Robert M. Albrecht li
: [ 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:
[romal@chessur ~]$ uname --all
Linux chessur.vorlon.lan
://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 for testing.
Robert, are you really sure you were using a stock 3.5 kernel? I need to
dig a little
at 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,
after
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
geschrieben: 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.
http://www.rodemic.com/mics/podcaster
Rode has a a Windows-Vista
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
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:
Data
wMaxPacketSize 0x0040 1x 64 bytes
bInterval 16
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.
Audacity actually records from
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
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 functional
(below
/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: /dev/input/event6
[23.844] (WW) HDA NVidia HDMI/DP,pcm=8: Don't know how to use device
On Tue, Jul 10, 2012 at 2:36 PM, Robert Krakora
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
this is no alternative.
Greetings from Robert
from Cottbus, Germany !
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Alsa
Hello,
ok, I just figured it out. The card needs an extra power supply, I did not know
that - how embarassing.
Thanks anyway, Clemens!
Best,
Robert
On Thursday, 1. March 2012 at 09:25, Clemens Ladisch wrote:
Robert Scherer wrote:
I noticed that outputs 1-2 are significantly louder than
and adjusts volumes automatically? If so, how can I avoid
that?
Thanks,
Robert
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Hello,
I have a 7.1 Xonar DX surround sound card installed (uses the CMI8788 driver)
and I want to split the outputs to 4 stereo outputs.
This works fine, my config looks like this:
pcm_slave.eightchannels {
pcm surround71:DX,0
channels 8
}
pcm.stereo1 {
type plughw
slave.pcm {
type dshare
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