http://blog.gmane.org/gmane.linux.alsa.user hourly 1 1901-01-01T00:00+00:00 http://gmane.org/img/gmane-25t.png http://gmane.org http://comments.gmane.org/gmane.linux.alsa.user/36355 <pre>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? ------------------------------------------------------------------------------ Live Security Virtual Conference Exclusive live event will cover all the ways today's security and threat landscape has changed and how IT managers can respond. Discussions will include endpoint security, mobile security and the latest in malware threats. http://www.accelacomm.com/jaw/sfrnl04242012/114/50122263/_______________________________________________ Alsa-user mailing list Alsa-user&lt; at &gt;lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user </pre> Stefan Thomas 2012-05-26T08:50:01 http://comments.gmane.org/gmane.linux.alsa.user/36354 <pre>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 kicking in...however other cards remain 1/1) http://ubuntuforums.org/archive/index.php/t-1126128.html when i do mendred&lt; at &gt;dantooine:~$ alsamixer -c3 cannot load mixer controls: Invalid argument So I tweaked around with my /etc/modprobe.d/alsa-base.conf and added ignore_ctl_error=1 to the snd-usb-audio module options snd-usb-audio index=-2 ignore_ctl_error=1 nrpacks=1 and then after restarting (Strangely sudo alsa force-reload didnt get it to work, i needed to restart the comp) alsamixer -c3 brought up the XStation mixer and it was visible on kmix as well However, the volumes could either be set at 0 or at 100 and nothing in between and it didnt seem that the XStation responded to the software mixer (e.g. muting it didnt mute anything for e.g.). Also the system became unstable and froze repeatedly, so I have turned back ignore_ctl_error=0 The wierd thing is that XStation works fine under both pulseaudio using pavu control and jack. I am assuming that this is some issue with the driver that deals with the mixer part? Any help on working around this to get a working alsa mixer would be appreciated! Thanks! Bharat ------------------------------------------------------------------------------ Live Security Virtual Conference Exclusive live event will cover all the ways today's security and threat landscape has changed and how IT managers can respond. Discussions will include endpoint security, mobile security and the latest in malware threats. http://www.accelacomm.com/jaw/sfrnl04242012/114/50122263/_______________________________________________ Alsa-user mailing list Alsa-user&lt; at &gt;lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user </pre> Bharat Rajagopalan 2012-05-25T12:26:49 http://comments.gmane.org/gmane.linux.alsa.user/36352 <pre>hi my .asoundrc I have: ------------8&lt;---------------------------- pcm.filter { &lt; at &gt;args [ ENNOISE ENBIQUAD FLAT ] &lt; at &gt;args.ENNOISE { type integer default 0 } &lt; at &gt;args.ENBIQUAD { type integer default 0 } &lt; at &gt;args.FLAT { type integer default 0 } type test slave { pcm "play0" } noise $ENNOISE biquad $ENBIQUAD flat $FLAT } pcm.filter_mono { &lt; at &gt;args [ _ENNOISE _ENBIQUAD _FLAT ] &lt; at &gt;args._ENNOISE { type integer default 0 } &lt; at &gt;args._ENBIQUAD { type integer default 0 } &lt; at &gt;args._FLAT { type integer default 0 } type plug slave { pcm "filter:$_ENNOISE,$_ENBIQUAD,$_FLAT" channels 1 rate 44100 } } ------------8&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 file but I have to use filter_mono device, and I get an error: # mplayer -vo null -ao alsa:device=filter_mono=1,0,0 sweep_100-5000.wav MPlayer UNKNOWN-4.3.3 (C) 2000-2010 MPlayer Team Playing sweep_100-5000.wav. Audio only file format detected. ========================================================================== Forced audio codec: mad Opening audio decoder: [pcm] Uncompressed PCM audio decoder AUDIO: 44100 Hz, 1 ch, s16le, 705.6 kbit/100.00% (ratio: 88200-&gt;88200) Selected audio codec: [pcm] afm: pcm (Uncompressed PCM) ========================================================================== [AO_ALSA] alsa-lib: conf.c:1685:(snd_config_load1) _toplevel_:171:33:Unexpected char [AO_ALSA] alsa-lib: conf.c:3467:(snd_config_hook_load) /home/root/.asoundrc may be old or corrupted: consider to remove or fix it [AO_ALSA] alsa-lib: conf.c:3326:(snd_config_hooks_call) function snd_config_hook_load returned error: Invalid argument [AO_ALSA] alsa-lib: conf.c:3713:(snd_config_update_r) hooks failed, removing configuration [AO_ALSA] Playback open error: Invalid argument how can I pass arguments from filter_mono to filter device? thanks ------------------------------------------------------------------------------ Live Security Virtual Conference Exclusive live event will cover all the ways today's security and threat landscape has changed and how IT managers can respond. Discussions will include endpoint security, mobile security and the latest in malware threats. http://www.accelacomm.com/jaw/sfrnl04242012/114/50122263/ </pre> Massimiliano Cialdi 2012-05-24T15:46:17 http://comments.gmane.org/gmane.linux.alsa.user/36349 <pre>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&lt; at &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. ------------------------------------------------------------------------------ Live Security Virtual Conference Exclusive live event will cover all the ways today's security and threat landscape has changed and how IT managers can respond. Discussions will include endpoint security, mobile security and the latest in malware threats. http://www.accelacomm.com/jaw/sfrnl04242012/114/50122263/ </pre> mattia.martinello< at >gmail.com 2012-05-23T16:27:16 http://comments.gmane.org/gmane.linux.alsa.user/36346 <pre>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 ------------------------------------------------------------------------------ Live Security Virtual Conference Exclusive live event will cover all the ways today's security and threat landscape has changed and how IT managers can respond. Discussions will include endpoint security, mobile security and the latest in malware threats. http://www.accelacomm.com/jaw/sfrnl04242012/114/50122263/ </pre> Vincent Tschanz 2012-05-22T09:56:00 http://comments.gmane.org/gmane.linux.alsa.user/36342 <pre>In this link i will try to get help with aforementioned card . ------------------------------------------------------------------------------ Live Security Virtual Conference Exclusive live event will cover all the ways today's security and threat landscape has changed and how IT managers can respond. Discussions will include endpoint security, mobile security and the latest in malware threats. http://www.accelacomm.com/jaw/sfrnl04242012/114/50122263/ </pre> paul424 2012-05-21T09:17:13 http://comments.gmane.org/gmane.linux.alsa.user/36341 <pre>As in title in this post I will try to get help with the aforementioned card. ------------------------------------------------------------------------------ Live Security Virtual Conference Exclusive live event will cover all the ways today's security and threat landscape has changed and how IT managers can respond. Discussions will include endpoint security, mobile security and the latest in malware threats. http://www.accelacomm.com/jaw/sfrnl04242012/114/50122263/ </pre> paul424 2012-05-21T09:14:47 http://comments.gmane.org/gmane.linux.alsa.user/36340 <pre>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 -&gt; FXBus out 16&amp;17 -&gt; FXBus in 0&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 ------------------------------------------------------------------------------ Live Security Virtual Conference Exclusive live event will cover all the ways today's security and threat landscape has changed and how IT managers can respond. Discussions will include endpoint security, mobile security and the latest in malware threats. http://www.accelacomm.com/jaw/sfrnl04242012/114/50122263/_______________________________________________ Alsa-user mailing list Alsa-user&lt; at &gt;lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user </pre> Dan Swain 2012-05-20T21:43:56 http://comments.gmane.org/gmane.linux.alsa.user/36338 <pre>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 ------------------------------------------------------------------------------ Live Security Virtual Conference Exclusive live event will cover all the ways today's security and threat landscape has changed and how IT managers can respond. Discussions will include endpoint security, mobile security and the latest in malware threats. http://www.accelacomm.com/jaw/sfrnl04242012/114/50122263/ </pre> Dieter Knopf 2012-05-19T04:25:26 http://comments.gmane.org/gmane.linux.alsa.user/36333 <pre>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://kmuto.jp/debian/hcl/): 8086:27d8 Intel Corporation N10/ICH 7 Family High Definition Audio Controller 10de:0a64 nVidia Corporation GT218 [ION] 10de:0be3 nVidia Corporation High Definition Audio Controller The problem is that when the audio stream starts, the first one or two seconds are not audible, it's trimmed from the stream. I have found a few available test samples that show the problem at http://www.audiocheck.net/ The longer mono sample at http://www.audiocheck.net/Audio/audiocheck.net_welcome2.wav need to be converted to stereo before being playable on the HDMI output: sox audiocheck.net_welcome2.wav -c 2 audiocheck.net_welcome-stereo.wav When played with aplay -D hw:1,7 audiocheck.net_welcome-stereo.wav the "Welcome to" words at the very beginning are not audible. The next two samples at http://www.audiocheck.net/Audio/audiocheck.net_L.wav http://www.audiocheck.net/Audio/audiocheck.net_R.wav are shorter than 2 seconds and are not audible at all when played. When played with "aplay -D hw:0,0" over the mainboard device, all samples are OK. Does anyone know whether it's a known problem or maybe an already fixed one? I don't think it's TV-dependent as this problem can be reproduced on two or three different TV models. Thanks in advance, Zoltán Böszörményi </pre> Boszormenyi Zoltan 2012-05-14T12:02:06 http://comments.gmane.org/gmane.linux.alsa.user/36331 <pre> 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? ------------------------------------------------------------------------------ Live Security Virtual Conference Exclusive live event will cover all the ways today's security and threat landscape has changed and how IT managers can respond. Discussions will include endpoint security, mobile security and the latest in malware threats. http://www.accelacomm.com/jaw/sfrnl04242012/114/50122263/ </pre> daniel< at >benoy.name 2012-05-14T03:38:35 http://comments.gmane.org/gmane.linux.alsa.user/36330 <pre>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------------------------------------------------------------------------------ Live Security Virtual Conference Exclusive live event will cover all the ways today's security and threat landscape has changed and how IT managers can respond. Discussions will include endpoint security, mobile security and the latest in malware threats. http://www.accelacomm.com/jaw/sfrnl04242012/114/50122263/_______________________________________________ Alsa-user mailing list Alsa-user&lt; at &gt;lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user </pre> EJS 2012-05-10T01:32:09 http://comments.gmane.org/gmane.linux.alsa.user/36329 <pre>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; 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; pulseaudio. It all worked fine until recently. No idea what triggered the change. Any help much appreciated. Thank you! Jeremy PS I have the same problem when I logged in using Gnome, instead of KDE4. So I doubt it is because of KDE... but I am not certain. ------------------------------------------------------------------------------ Live Security Virtual Conference Exclusive live event will cover all the ways today's security and threat landscape has changed and how IT managers can respond. Discussions will include endpoint security, mobile security and the latest in malware threats. http://www.accelacomm.com/jaw/sfrnl04242012/114/50122263/_______________________________________________ Alsa-user mailing list Alsa-user&lt; at &gt;lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user </pre> Jeremy Kasha 2012-05-07T14:13:41 http://comments.gmane.org/gmane.linux.alsa.user/36328 <pre>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 file.wav`, but then I must run `amixer sset 'IEC958',1 unmute` (or, alternatively, unmute this channel manually using alsamixer, where it is marked "S/PDIF 1"). I want to play to the TV by HDMI by default. I find that I can select this output with the following /etc/asound.conf: # cat /etc/asound.conf pcm.!default { type hw card 0 device 3 } # Now next time I reboot my computer `aplay -vvD file.wav` plays to the TV by default. However I must still manually unmute the "S/PDIF 1" channel before the music is actually audible. Is it possible to configure a muting / unmuting default for a channel in /etc/asound.conf, please? I understand that I can store the current state - including muting - with `alsactl -f file store` and then `alsactl -f file restore` on next boot-up. My disto provides an init.d script for this purpose, but I would prefer not to do things that way. I would prefer to set the default by /etc/asound.conf or similar, I guess), rather than saving a state. I hope this distinction makes sense, and I am not alone in perceiving it. Thanks for reading, and thanks in advance for any help you can offer, Stroller. ------------------------------------------------------------------------------ Live Security Virtual Conference Exclusive live event will cover all the ways today's security and threat landscape has changed and how IT managers can respond. Discussions will include endpoint security, mobile security and the latest in malware threats. http://www.accelacomm.com/jaw/sfrnl04242012/114/50122263/ </pre> Stroller 2012-05-04T08:41:09 http://comments.gmane.org/gmane.linux.alsa.user/36310 <pre>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;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;hash=item27c0441a1c ; http://www.ebay.com/itm/USB-6-Channel-5-1-External-Audio-Sound-Card-S-PDIF-/190490619104?pt=LH_DefaultDomain_0&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;hash=item5ae3d5b217 ; http://www.ebay.com/itm/New-USB-6-Channel-External-Audio-Sound-Card-S-PDIF-PC-/270748963523?pt=LH_DefaultDomain_0&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;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;hash=item2a1cd6bfa5 ; http://www.ebay.com/itm/Multi-sound-effects-USB-6-Channel-5-1-External-Audio-Sound-Card-Laptop-PC-/110860124104?pt=LH_DefaultDomain_0&amp;hash=item19cfc73fc8 - in this listing CMedia is mentioned ;http://www.ebay.com/itm/6-Channel-6CH-USB-2-0-5-1-External-USB-Video-Audio-Adapter-Audio-Sound-Card-NEW-/270955321001?pt=LH_DefaultDomain_0&amp;hash=item3f163216a9 - in this listing CMedia is mentioned ; http://www.ebay.com/itm/NEW-USB-2-0-5-1-Channel-Speaker-System-Sound-Adapter-/250666394781?pt=LH_DefaultDomain_0&amp;hash=item3a5ce1a49d - in this listing "USB Audio Device Class Specification 1.0" is mentioned and similar. Does anybody know whether these are supported by ALSA, what chip(s) are used, etc. ? I interested in "Line In" (capture) support. Thanks, Sergei. ------------------------------------------------------------------------------ Live Security Virtual Conference Exclusive live event will cover all the ways today's security and threat landscape has changed and how IT managers can respond. Discussions will include endpoint security, mobile security and the latest in malware threats. http://www.accelacomm.com/jaw/sfrnl04242012/114/50122263/ </pre> Sergei Steshenko 2012-04-29T20:51:16 http://comments.gmane.org/gmane.linux.alsa.user/36308 <pre>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. ------------------------------------------------------------------------------ Live Security Virtual Conference Exclusive live event will cover all the ways today's security and threat landscape has changed and how IT managers can respond. Discussions will include endpoint security, mobile security and the latest in malware threats. http://www.accelacomm.com/jaw/sfrnl04242012/114/50122263/_______________________________________________ Alsa-user mailing list Alsa-user&lt; at &gt;lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user </pre> Johnny Izzo 2012-04-28T20:48:50 http://comments.gmane.org/gmane.linux.alsa.user/36303 <pre>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 01:01.00: [irq 5] kernel: [ 28.250769] sbawe 01:01.00: [dma 1] kernel: [ 28.250791] sbawe 01:01.00: [dma 5] kernel: [ 28.250832] sbawe 01:01.00: [io 0x0220-0x022f] kernel: [ 28.250865] sbawe 01:01.00: [io 0x0330-0x0331] kernel: [ 28.250898] sbawe 01:01.00: [io 0x0388-0x038b] kernel: [ 28.253273] sbawe 01:01.00: activated kernel: [ 28.253314] sbawe 01:01.02: [io 0x0620-0x0623] kernel: [ 28.260152] sbawe 01:01.02: [io 0x0a20-0x0a23] kernel: [ 28.260204] sbawe 01:01.02: [io 0x0e20-0x0e23] kernel: [ 28.262745] sbawe 01:01.02: activated "reboot" command had solve this problem. I wonder. Why this problem may happen? Could anyone tell what mean "sbawe 01:01.00" and "sbawe 01:01.02" ? </pre> Блогер 2012-04-20T08:26:30 http://comments.gmane.org/gmane.linux.alsa.user/36297 <pre>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 0x014d bcdDevice 0.00 iManufacturer 1 BOSS iProduct 2 GT-100 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 176 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 255 Vendor Specific Subclass bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 06 24 f1 01 00 00 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 07 24 01 01 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 02 04 18 01 44 ac 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x0d EP 13 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0038 1x 56 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 ** UNRECOGNIZED: 07 24 01 07 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 02 04 18 01 44 ac 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x8e EP 14 IN bmAttributes 37 Transfer Type Isochronous Synch Type Asynchronous Usage Type Implicit feedback Data wMaxPacketSize 0x0038 1x 56 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 06 24 f1 02 01 01 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x85 EP 5 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 0 bDeviceProtocol 255 bMaxPacketSize0 64 bNumConfigurations 1 Device Status: 0x0001 Self Powered ------------------------------------------------------------------------------ For Developers, A Lot Can Happen In A Second. Boundary is the first to Know...and Tell You. Monitor Your Applications in Ultra-Fine Resolution. Try it FREE! http://p.sf.net/sfu/Boundary-d2dvs2_______________________________________________ Alsa-user mailing list Alsa-user&lt; at &gt;lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user </pre> John McFarland 2012-04-15T16:37:25 http://comments.gmane.org/gmane.linux.alsa.user/36294 <pre>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 Second. Boundary is the first to Know...and Tell You. Monitor Your Applications in Ultra-Fine Resolution. Try it FREE! http://p.sf.net/sfu/Boundary-d2dvs2 _______________________________________________ Alsa-user mailing list Alsa-user&lt; at &gt;lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user </pre> Uroš Vampl 2012-04-13T21:49:47 http://comments.gmane.org/gmane.linux.alsa.user/36282 <pre>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. ------------------------------------------------------------------------------ Better than sec? Nothing is better than sec when it comes to monitoring Big Data applications. Try Boundary one-second resolution app monitoring today. Free. http://p.sf.net/sfu/Boundary-dev2dev_______________________________________________ Alsa-user mailing list Alsa-user&lt; at &gt;lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user </pre> Fabio Gonzalez 2012-04-11T19:24:00 http://comments.gmane.org/gmane.linux.alsa.user/36281 <pre>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? ------------------------------------------------------------------------------ Better than sec? Nothing is better than sec when it comes to monitoring Big Data applications. Try Boundary one-second resolution app monitoring today. Free. http://p.sf.net/sfu/Boundary-dev2dev </pre> Mark Bishop 2012-04-10T21:39:17 Search the mailing list at Gmane query http://search.gmane.org/?group=$group=gmane.linux.alsa.user