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    <title>gmane.linux.alsa.user</title>
    <link>http://blog.gmane.org/gmane.linux.alsa.user</link>
    <description/>
    <syn:updatePeriod>hourly</syn:updatePeriod>
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    <syn:updateBase>1901-01-01T00:00+00:00</syn:updateBase>
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    <link>http://gmane.org</link>
  </image>
  <item rdf:about="http://comments.gmane.org/gmane.linux.alsa.user/36355">
    <title>Can not open /dev/sequencer</title>
    <link>http://comments.gmane.org/gmane.linux.alsa.user/36355</link>
    <description>&lt;pre&gt;Dear community,
I have the following problem:
I want to use the software rumor (which can be used do create midi-code via
midi-keyboard).
When I type

I get

Can not open /dev/sequencer

When I just type rumor
I get

and it doesn't react von midi-input.
What can I do now to get it working?
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Alsa-user mailing list
Alsa-user&amp;lt; at &amp;gt;lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/alsa-user
&lt;/pre&gt;</description>
    <dc:creator>Stefan Thomas</dc:creator>
    <dc:date>2012-05-26T08:50:01</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.linux.alsa.user/36354">
    <title>Novation XStation 25 - alsamixer error - cannot load mixer controls: Invalid argument - ignore_ctl_error causes instability</title>
    <link>http://comments.gmane.org/gmane.linux.alsa.user/36354</link>
    <description>&lt;pre&gt;Hi all,

I am running Kubuntu 12.04 with the 3.2.0.23 lowlatency kernel. I have
installed Kxstudio and am using my XStation 25 rather successfully with
jackdbus. I have disabled pulseaudio on kde and am routing everything via
jack and using the pulseaudio bridge for applications like skype which seem
to need it.

Now I am trying to use a master mixer for jack to control my hardware
channels, so as a result settled on Kmix (started as env
KMIX_PULSEAUDIO_DISABLE=1  kmix to disable pulse and to show alsa channels).

However I noted that while my onboard sound card showed up, the Xstation
didn't. So i did

sudo lsusb -v

Output here:
http://pastebin.com/3TCxYz8d

aplay -L - Output here:

http://pastebin.com/afgf9LtR


Key thing to note in the output of aplay was:



card 3: XStation [XStation], device 0: USB Audio [USB Audio]


  Subdevices: 0/1


  Subdevice #0: subdevice #0

The Subdevices show 0/1 AFTER I boot into kde (it shows 1/1 immediately on
boot prior to logging into kde prior to jackdbus/pulseaudio k&lt;/pre&gt;</description>
    <dc:creator>Bharat Rajagopalan</dc:creator>
    <dc:date>2012-05-25T12:26:49</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.linux.alsa.user/36352">
    <title>passing arguments</title>
    <link>http://comments.gmane.org/gmane.linux.alsa.user/36352</link>
    <description>&lt;pre&gt;hi
my .asoundrc I have:

------------8&amp;lt;----------------------------
pcm.filter {
     &amp;lt; at &amp;gt;args [ ENNOISE ENBIQUAD FLAT ]
     &amp;lt; at &amp;gt;args.ENNOISE {
         type integer
         default 0
     }
     &amp;lt; at &amp;gt;args.ENBIQUAD {
         type integer
         default 0
     }
     &amp;lt; at &amp;gt;args.FLAT {
         type integer
         default 0
     }
     type test
     slave {
         pcm "play0"
     }
     noise $ENNOISE
     biquad $ENBIQUAD
     flat $FLAT
}
pcm.filter_mono {
     &amp;lt; at &amp;gt;args [ _ENNOISE _ENBIQUAD _FLAT ]
     &amp;lt; at &amp;gt;args._ENNOISE {
         type integer
         default 0
     }
     &amp;lt; at &amp;gt;args._ENBIQUAD {
         type integer
         default 0
     }
     &amp;lt; at &amp;gt;args._FLAT {
         type integer
         default 0
     }
     type plug
     slave {
         pcm "filter:$_ENNOISE,$_ENBIQUAD,$_FLAT"
         channels 1
         rate 44100
     }
}
------------8&amp;lt;----------------------------

If I use filter device there is no problems, for example:
# mplayer -vo null -ao alsa:device=filter=1,0,0 sweep_100-5000.wav
it plays correctly wav&lt;/pre&gt;</description>
    <dc:creator>Massimiliano Cialdi</dc:creator>
    <dc:date>2012-05-24T15:46:17</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.linux.alsa.user/36349">
    <title>Input/Output error</title>
    <link>http://comments.gmane.org/gmane.linux.alsa.user/36349</link>
    <description>&lt;pre&gt;Hi.

I'm using Ubuntu 11.04 i386 with kernel 2.6.38-15-generic-pae to
record some audio with arecord.

I'm getting this strange error when I try to record from my FM tuner
card from device hw1,0:

root&amp;lt; at &amp;gt;dvr-test:/home/dvr/dvr# arecord -f cd -D hw:1,0 -d 20 test.wav
Registrazione in corso WAVE 'test.wav' : Signed 16 bit Little Endian,
Frequenza 44100 Hz, Stereo
Attenzione: il campionamento non è accurato (richiesto=44100Hz,
ottenuto=32000Hz)
         provare il plugin plug
arecord: pcm_read:1773: read error: Input/output error

And the test.wav file is empty, of course.

I tried to upgrade to Ubuntu 12.04 also, but I get the same error.
I tried to change my FM tuner PCI card, but I didn't solve the problem.

I googled a lot but I didn't find any solution for my problem.

Here you can find my ALSA info:
http://www.alsa-project.org/db/?f=cfc2632071342f2b0373c3a484fa04b19acf0958

Could you help me, please?

Thank you very much!
Bye.

------------------------------------------------------------------------------
&lt;/pre&gt;</description>
    <dc:creator>mattia.martinello&lt; at &gt;gmail.com</dc:creator>
    <dc:date>2012-05-23T16:27:16</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.linux.alsa.user/36346">
    <title>Macbook pro 1.2 Headphone and speakers</title>
    <link>http://comments.gmane.org/gmane.linux.alsa.user/36346</link>
    <description>&lt;pre&gt;Hi,

I have a Macbook Pro 1.2 .With ubuntu 12.04, the sound works, but when I
plug in headphones, I have sound both in headphones and internal speakers.

I tried to add "options snd-hda-intel model=macbook-pro-v1" at the end
of /etc/modprobe.d/alsa-base.conf

But it is not working better.
The soundcard on that laptop is a SigmaTel STAC9221 A1

Is there something else I can try to get the sound working correctly?

Thank you


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&lt;/pre&gt;</description>
    <dc:creator>Vincent Tschanz</dc:creator>
    <dc:date>2012-05-22T09:56:00</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.linux.alsa.user/36342">
    <title>Speedlink SL-8871 sound card , chipset cmi8768</title>
    <link>http://comments.gmane.org/gmane.linux.alsa.user/36342</link>
    <description>&lt;pre&gt;In this link i will try to get help with aforementioned card . 

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&lt;/pre&gt;</description>
    <dc:creator>paul424</dc:creator>
    <dc:date>2012-05-21T09:17:13</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.linux.alsa.user/36341">
    <title>Speedlink SL8871 sound card , chipset cmi8768</title>
    <link>http://comments.gmane.org/gmane.linux.alsa.user/36341</link>
    <description>&lt;pre&gt;As in title in this post I will try to get help with the aforementioned card. 

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&lt;/pre&gt;</description>
    <dc:creator>paul424</dc:creator>
    <dc:date>2012-05-21T09:14:47</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.linux.alsa.user/36340">
    <title>emu10k1 (Audigy) FXBus routing query</title>
    <link>http://comments.gmane.org/gmane.linux.alsa.user/36340</link>
    <description>&lt;pre&gt;Dear all,

I am writing to ask how to route certain DSP 'outputs' from my Audigy 2 ZS
back into the equivalent 'ASIO inputs' for recording purposes.

Specifically, I'd like to route outputs from the Wavetable Synths on my
card to inputs that would go straight into a DAW (such as Ardour), so that
I can apply effects, record and mix-down as if they were external synth
signals.

I know that with the kX project drivers in Windows this is possible, by
routing the Synth outputs to FXBus outputs 16-63, leaving the ASIO inputs
as 0-15 for the audio input, and then patching which channels I require
from the Synth to those inputs (e.g. Synth Channel 1 -&amp;gt; FXBus out 16&amp;amp;17 -&amp;gt;
FXBus in 0&amp;amp;1).

After reading a thread that contained responses from Jaroslav, I believe
the answer is in emufx.c, but I'm unsure as to where to start as I need to
assign the outputs from the Synths correctly first.

Thanks in advance! Any help would be greatly appreciated!

Kind Regards,

Dan
--------------------------------------------------------&lt;/pre&gt;</description>
    <dc:creator>Dan Swain</dc:creator>
    <dc:date>2012-05-20T21:43:56</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.linux.alsa.user/36338">
    <title>Permanent audio device with snd-usb-audio</title>
    <link>http://comments.gmane.org/gmane.linux.alsa.user/36338</link>
    <description>&lt;pre&gt;Hello,

i'm searching for a way to create a permanent audio device with the
snd-usb-audio and a USB microphone.

The device should be already there and working and after plugging in
the USB device it should be avaialble through that device.

Example: /dev/pcm is already there and i can stream from it and after
i plugged in the USB device this should be available through that
device. After unplugging it it should be still there. Like a mixer.

Thanks

Dieter

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&lt;/pre&gt;</description>
    <dc:creator>Dieter Knopf</dc:creator>
    <dc:date>2012-05-19T04:25:26</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.linux.alsa.user/36333">
    <title>HDMI audio on NVIDIA trims the first second off thestream</title>
    <link>http://comments.gmane.org/gmane.linux.alsa.user/36333</link>
    <description>&lt;pre&gt;Hi Linux sound driver maintainers and authors,

we have a little(?) problem that we discovered while experimenting
with HDMI audio output on a Zotac ZBOX with NVIDIA ION GT218.
We are using the embedded Angstrom distribution with kernel
version 2.6.38.4 and NVIDIA proprietary driver version 260.19.21.

Both the mainboard audio controller and the HDMI audio are
driven by the in-kernel snd-hda-intel. lspci output is:

00:00.0 Class 0600: 8086:a000
00:1b.0 Class 0403: 8086:27d8
00:1c.0 Class 0604: 8086:27d0
00:1c.1 Class 0604: 8086:27d2
00:1c.2 Class 0604: 8086:27d4
00:1d.0 Class 0c03: 8086:27c8
00:1d.1 Class 0c03: 8086:27c9
00:1d.2 Class 0c03: 8086:27ca
00:1d.3 Class 0c03: 8086:27cb
00:1d.7 Class 0c03: 8086:27cc
00:1e.0 Class 0604: 8086:2448
00:1f.0 Class 0601: 8086:27bc
00:1f.2 Class 0101: 8086:27c0
00:1f.3 Class 0c05: 8086:27da
01:00.0 Class 0c03: 1106:3432
02:00.0 Class 0280: 168c:002b
03:00.0 Class 0300: 10de:0a64
03:00.1 Class 0403: 10de:0be3
04:0b.0 Class 0200: 10ec:8167

Relevant parts (decoded by http:&lt;/pre&gt;</description>
    <dc:creator>Boszormenyi Zoltan</dc:creator>
    <dc:date>2012-05-14T12:02:06</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.linux.alsa.user/36331">
    <title>Combining multiple inputs</title>
    <link>http://comments.gmane.org/gmane.linux.alsa.user/36331</link>
    <description>&lt;pre&gt; I need some help configuring my asound.conf.  I want to combine 
 multiple capture devices into one.  I saw 'dsnoop' but that seems to 
 make it so that multiple programs can read from the same sound device.  
 I want to do the opposite.  I want one program to read from multiple 
 sound devices.  It seems that the 'multi' plugin can be used to output 
 from one program to multiple sound cards, so is it possible to do that 
 for capture?  Is it the same plugin?

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&lt;/pre&gt;</description>
    <dc:creator>daniel&lt; at &gt;benoy.name</dc:creator>
    <dc:date>2012-05-14T03:38:35</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.linux.alsa.user/36330">
    <title>asoundrc to reduce output volume</title>
    <link>http://comments.gmane.org/gmane.linux.alsa.user/36330</link>
    <description>&lt;pre&gt;Hi all,
 
I am having a problem that whenever I startup skype, it is much much louder than my other applications and it almost deafens me. Skype on linux does not have its own volume control; so I was wondering if there were a simple way to use my asoundrc to create a virtual output device or plugin where the volume is reduced by like 50%, and then configure skype to play through that device?
 
I have played with my asoundrc a decent amount in the past, but I never feel like I understand exactly what I am doing.

Thanks in advance,
J------------------------------------------------------------------------------
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Alsa-user mailing list
Alsa-user&amp;lt; at &amp;gt;lists.&lt;/pre&gt;</description>
    <dc:creator>EJS</dc:creator>
    <dc:date>2012-05-10T01:32:09</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.linux.alsa.user/36329">
    <title>Mystery screeching signal in all 8 inputs in envy24control</title>
    <link>http://comments.gmane.org/gmane.linux.alsa.user/36329</link>
    <description>&lt;pre&gt;Please help!

There is a loud, high-pitched constant signal peaking out all 8 inputs of
my m-audio interface, even when the interface is turned off. (In other
words, seems to be a software problem). I can see the signals peaking in
red at top of meters in envy24control. If I use envy24control to route
those inputs to my stereo output, or if I try to record using ardour/jack,
All I get is that annoying shrieking signal, which is extremely loud (ie,
not just a little noise). It seems like there's a logical feedback loop or
something is incorrectly routed to those inputs in alsa, pulseaudio etc.
It's made my studio completely unusuable [image: :(]

I have checked alsa &amp;amp; pulseaudio settings using alsa mixer, kmixer,
paprefs, etc, but don't see anything wrong.... But I am not an expert and
may easily have missed something.

alsa-info posted here:

http://www.alsa-project.org/db/?f=ebddc1d574175bb3529e5411f70d1969d4c6437a

I am using Fedora15, KDE4 (with phonon), alsa &amp;amp; pulseaudio. It all worked
fine until recentl&lt;/pre&gt;</description>
    <dc:creator>Jeremy Kasha</dc:creator>
    <dc:date>2012-05-07T14:13:41</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.linux.alsa.user/36328">
    <title>Muting or unmuting by default?</title>
    <link>http://comments.gmane.org/gmane.linux.alsa.user/36328</link>
    <description>&lt;pre&gt;Hi there,

I have an eMachines 1401 nettop mini-desktop PC. It has a low-power AMD64 dual-core CPU and an onboard nVidia Geforce 9200 GPU. In order for the sound to work the nVidia binary drivers must be loaded. My distro is Gentoo; I have versions c 1.0.25 of alsa-lib, -headers and -tools installed.

   # aplay -l                                                               
   **** List of PLAYBACK Hardware Devices ****
   card 0: NVidia [HDA NVidia], device 0: ALC662 rev1 Analog [ALC662 rev1 Analog]
     Subdevices: 1/1
     Subdevice #0: subdevice #0
   card 0: NVidia [HDA NVidia], device 1: ALC662 rev1 Digital [ALC662 rev1 Digital]
     Subdevices: 1/1
     Subdevice #0: subdevice #0
   card 0: NVidia [HDA NVidia], device 3: HDMI 0 [HDMI 0]
     Subdevices: 1/1
     Subdevice #0: subdevice #0
   # 

The analogue headphone socket is selected by default. If I run `aplay -vvD file.wav` then the audio plays to the headphone socket. 

To play to the TV via HDMI, not only must I run `aplay -vvD plug:hdmi fil&lt;/pre&gt;</description>
    <dc:creator>Stroller</dc:creator>
    <dc:date>2012-05-04T08:41:09</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.linux.alsa.user/36310">
    <title>are cheap USB cards with analog line input supported by ALSA ?</title>
    <link>http://comments.gmane.org/gmane.linux.alsa.user/36310</link>
    <description>&lt;pre&gt;Hello,

I see a number of relatively cheap soundcards on eBay:

http://www.ebay.com/itm/W449-USB-6-Channel-5-1-Audio-Sound-Card-S-PDIF-Exter-/250754613678?pt=LH_DefaultDomain_0&amp;amp;hash=item3a6223c1ae ;
http://www.ebay.com/itm/C449-USB-6-Channel-5-1-Audio-Sound-Card-S-PDIF-Exter-/170729413148?pt=LH_DefaultDomain_0&amp;amp;hash=item27c0441a1c ;
http://www.ebay.com/itm/USB-6-Channel-5-1-External-Audio-Sound-Card-S-PDIF-/190490619104?pt=LH_DefaultDomain_0&amp;amp;hash=item2c5a202ce0 ;
http://www.ebay.com/itm/USB-6-Channel-5-1-External-Audio-Sound-Card-S-PDIF-SC6-/390369489431?pt=LH_DefaultDomain_0&amp;amp;hash=item5ae3d5b217 ;
http://www.ebay.com/itm/New-USB-6-Channel-External-Audio-Sound-Card-S-PDIF-PC-/270748963523?pt=LH_DefaultDomain_0&amp;amp;hash=item3f09e552c3 ;
http://www.ebay.com/itm/External-USB-6-Channel-7-1-Audio-Sound-Card-S-PDIF-877-/180660714139?pt=LH_DefaultDomain_0&amp;amp;hash=item2a1037ba9b ;
http://www.ebay.com/itm/USB-6-Channel-5-1-External-Audio-Sound-Card-S-PDIF-Consumer-interface-SC6-/180872462245?pt=LH_DefaultDomain_0&amp;amp;hash=item2a1&lt;/pre&gt;</description>
    <dc:creator>Sergei Steshenko</dc:creator>
    <dc:date>2012-04-29T20:51:16</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.linux.alsa.user/36308">
    <title>no input source, HDA nvidia.</title>
    <link>http://comments.gmane.org/gmane.linux.alsa.user/36308</link>
    <description>&lt;pre&gt;Hi all!

In the previous distro in AlsaMixer (v.1.0.22), under recording i had
"Capture" and "Input Source". From input source i selected front mic
(external audio jack) or ATAPI mic (computer internal mic).
Now installed latest Ubuntu, in Alsamixer (v.1.0.25) under recording i
have: "Mic Boost", "Capture" and "Internal Mic Boost".
I think this change is from Alsa not Ubuntu?

External jack works, but how do i select the mic from computer?

Card: HDA NVidia
Chip: Realtek ALC861-VD
Comp: HP Pavilion tx1000

Regards, J.
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h&lt;/pre&gt;</description>
    <dc:creator>Johnny Izzo</dc:creator>
    <dc:date>2012-04-28T20:48:50</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.linux.alsa.user/36303">
    <title>No soundcards found</title>
    <link>http://comments.gmane.org/gmane.linux.alsa.user/36303</link>
    <description>&lt;pre&gt;Today I turn on my PC and got no sound, this message durring boot:
Restoring ALSA Levels [BUSY]
/usr/sbin/alsactl: load_state:1686: No soundcards found... [FAIL] 

"No sound device" or similar content in /proc/asound/cards file. Some
difference I have found in kernel.log file. Today:

kernel: [   30.484912] sbawe 01:01.00: [irq 5]
kernel: [   30.484946] sbawe 01:01.00: [dma 1]
kernel: [   30.484968] sbawe 01:01.00: [dma 5]
kernel: [   30.485009] sbawe 01:01.00: [io  0x0220-0x022f]
kernel: [   30.485042] sbawe 01:01.00: [io  0x0330-0x0331]
kernel: [   30.485075] sbawe 01:01.00: [io  0x0388-0x038b]
kernel: [   30.490713] sbawe 01:01.00: activated
kernel: [   30.490776] sbawe 01:01.02: [io  0x0620-0x0623]
kernel: [   30.490811] sbawe 01:01.02: [io  0x0a20-0x0a23]
kernel: [   30.490845] sbawe 01:01.02: [io  0x0e20-0x0e23]
kernel: [   30.499201] sbawe 01:01.02: activated
kernel: [   30.504322] sbawe 01:01.00: disabled
kernel: [   30.509833] sbawe 01:01.02: disabled

Yesterday:
kernel: [   28.250735] sbawe 01:01.0&lt;/pre&gt;</description>
    <dc:creator>Блогер</dc:creator>
    <dc:date>2012-04-20T08:26:30</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.linux.alsa.user/36297">
    <title>Unrecognized audio interface: Boss GT-100</title>
    <link>http://comments.gmane.org/gmane.linux.alsa.user/36297</link>
    <description>&lt;pre&gt;Hello,

Just purchased the Boss (Roland) GT-100 guitar processor.  This is a
new model released in the last few weeks, and it can function as a USB
audio interface (it is the successor to the Boss GT-10, which I think
support has been added for).  I am running Linux Mint 12.  When I
connect the device, it is not recognized as an audio interface (not
seen in the output of aplay -l).

I have never done any alsa tweaking before, but maybe it is just a
matter of adding a new section to the quirks file and recompiling
alsa?  I would like to try and get this device to work, if someone
could provide some guidance.  I am attaching the output from sudo
lsusb -v.

Thanks,
John
Bus 002 Device 014: ID 0582:014d Roland Corp. 
Device Descriptor:
  bLength                18
  bDescriptorType         1
  bcdUSB               2.00
  bDeviceClass          255 Vendor Specific Class
  bDeviceSubClass         0 
  bDeviceProtocol       255 
  bMaxPacketSize0        64
  idVendor           0x0582 Roland Corp.
  idProduct         &lt;/pre&gt;</description>
    <dc:creator>John McFarland</dc:creator>
    <dc:date>2012-04-15T16:37:25</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.linux.alsa.user/36294">
    <title>No sound on Quanta KN1 starting with kernel 3.4</title>
    <link>http://comments.gmane.org/gmane.linux.alsa.user/36294</link>
    <description>&lt;pre&gt;Hi,

codec is Realtek ALC260, lscpi -vnn shows:

00:1b.0 Audio device [0403]: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6
Family) High Definition Audio Controller [8086:2668] (rev 04)
Subsystem: QUANTA Computer Inc Device [152d:0729]
Flags: bus master, fast devsel, latency 0, IRQ 44
Memory at c0000000 (64-bit, non-prefetchable) [size=16K]
Capabilities: [50] Power Management version 2
Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit+
Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00
Capabilities: [100] Virtual Channel
Capabilities: [130] Root Complex Link
Kernel driver in use: snd_hda_intel


Previous kernels had a static quirk

SND_PCI_QUIRK(0x152d, 0x0729, "CTL U553W", ALC260_BASIC),

which did the trick, but static quirks were removed in kernel 3.4. Just tell
what other info I should provide that'll help solve this and I'll provide it.

Regards,
Uroš


------------------------------------------------------------------------------
For Developers, A Lot Can Happen In A&lt;/pre&gt;</description>
    <dc:creator>Uroš Vampl</dc:creator>
    <dc:date>2012-04-13T21:49:47</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.linux.alsa.user/36282">
    <title>Playing audio with ALSA api(newbie)</title>
    <link>http://comments.gmane.org/gmane.linux.alsa.user/36282</link>
    <description>&lt;pre&gt;I'm new to ALSA. I made a program that plays a file, pure sound. The
problem is that every time I write, apparently repeats a part of the previous
sound. I know that the program gives segmentation fault, that the moment is
not a problem. This occurs after the song finishes playing. I send the
source code as an attachment. When the amount of sound that I play every
moment is smaller, the sound is very bad.
------------------------------------------------------------------------------
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&lt;/pre&gt;</description>
    <dc:creator>Fabio Gonzalez</dc:creator>
    <dc:date>2012-04-11T19:24:00</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.linux.alsa.user/36281">
    <title>copy from one hw to another hw</title>
    <link>http://comments.gmane.org/gmane.linux.alsa.user/36281</link>
    <description>&lt;pre&gt;I have multiple codecs attached to my processor and I want to copy the  
data from user space through ALSA into both a hw0 and a hw1.  Possible?

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&lt;/pre&gt;</description>
    <dc:creator>Mark Bishop</dc:creator>
    <dc:date>2012-04-10T21:39:17</dc:date>
  </item>
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