Screenshot instructions:
Windows
Mac
Red Hat Linux
Ubuntu
Click URL instructions:
Right-click on ad, choose "Copy Link", then paste here →
(This may not be possible with some types of ads)
You can subscribe to this list here.
2001 |
Jan
|
Feb
|
Mar
|
Apr
|
May
(204) |
Jun
(332) |
Jul
(443) |
Aug
(448) |
Sep
(353) |
Oct
(355) |
Nov
(421) |
Dec
(496) |
---|---|---|---|---|---|---|---|---|---|---|---|---|
2002 |
Jan
(421) |
Feb
(430) |
Mar
(430) |
Apr
(529) |
May
(364) |
Jun
(338) |
Jul
(299) |
Aug
(377) |
Sep
(330) |
Oct
(483) |
Nov
(454) |
Dec
(301) |
2003 |
Jan
(536) |
Feb
(547) |
Mar
(557) |
Apr
(467) |
May
(343) |
Jun
(524) |
Jul
(490) |
Aug
(339) |
Sep
(466) |
Oct
(585) |
Nov
(496) |
Dec
(495) |
2004 |
Jan
(775) |
Feb
(668) |
Mar
(574) |
Apr
(463) |
May
(371) |
Jun
(363) |
Jul
(301) |
Aug
(527) |
Sep
(319) |
Oct
(396) |
Nov
(336) |
Dec
(554) |
2005 |
Jan
(423) |
Feb
(589) |
Mar
(634) |
Apr
(366) |
May
(436) |
Jun
(467) |
Jul
(346) |
Aug
(464) |
Sep
(345) |
Oct
(442) |
Nov
(420) |
Dec
(431) |
2006 |
Jan
(662) |
Feb
(424) |
Mar
(454) |
Apr
(387) |
May
(302) |
Jun
(452) |
Jul
(353) |
Aug
(329) |
Sep
(319) |
Oct
(344) |
Nov
(306) |
Dec
(231) |
2007 |
Jan
(313) |
Feb
(316) |
Mar
(203) |
Apr
(211) |
May
(213) |
Jun
(214) |
Jul
(183) |
Aug
(287) |
Sep
(240) |
Oct
(227) |
Nov
(321) |
Dec
(218) |
2008 |
Jan
(303) |
Feb
(194) |
Mar
(289) |
Apr
(193) |
May
(208) |
Jun
(247) |
Jul
(284) |
Aug
(217) |
Sep
(150) |
Oct
(148) |
Nov
(171) |
Dec
(254) |
2009 |
Jan
(258) |
Feb
(144) |
Mar
(123) |
Apr
(239) |
May
(181) |
Jun
(182) |
Jul
(123) |
Aug
(100) |
Sep
(127) |
Oct
(137) |
Nov
(169) |
Dec
(111) |
2010 |
Jan
(162) |
Feb
(112) |
Mar
(145) |
Apr
(96) |
May
(110) |
Jun
(67) |
Jul
(86) |
Aug
(112) |
Sep
(87) |
Oct
(126) |
Nov
(83) |
Dec
(37) |
2011 |
Jan
(102) |
Feb
(147) |
Mar
(71) |
Apr
(68) |
May
(67) |
Jun
(97) |
Jul
(56) |
Aug
(73) |
Sep
(78) |
Oct
(37) |
Nov
(75) |
Dec
(54) |
2012 |
Jan
(137) |
Feb
(103) |
Mar
(70) |
Apr
(67) |
May
(37) |
Jun
(104) |
Jul
(94) |
Aug
(100) |
Sep
(79) |
Oct
(80) |
Nov
(98) |
Dec
(38) |
2013 |
Jan
(84) |
Feb
(137) |
Mar
(107) |
Apr
(106) |
May
(117) |
Jun
(53) |
Jul
(61) |
Aug
(48) |
Sep
(39) |
Oct
(69) |
Nov
(99) |
Dec
(86) |
2014 |
Jan
(125) |
Feb
(87) |
Mar
(49) |
Apr
(105) |
May
(86) |
Jun
(34) |
Jul
(45) |
Aug
(65) |
Sep
(59) |
Oct
(69) |
Nov
(71) |
Dec
(58) |
2015 |
Jan
(26) |
Feb
(73) |
Mar
(49) |
Apr
(38) |
May
(78) |
Jun
(30) |
Jul
(12) |
Aug
(48) |
Sep
(56) |
Oct
(47) |
Nov
(13) |
Dec
(25) |
2016 |
Jan
(48) |
Feb
(49) |
Mar
(41) |
Apr
(46) |
May
(44) |
Jun
(29) |
Jul
(30) |
Aug
(37) |
Sep
(50) |
Oct
(31) |
Nov
(52) |
Dec
(37) |
2017 |
Jan
(8) |
Feb
(29) |
Mar
(35) |
Apr
(19) |
May
(39) |
Jun
(24) |
Jul
(46) |
Aug
(19) |
Sep
(15) |
Oct
(31) |
Nov
(22) |
Dec
(12) |
2018 |
Jan
(42) |
Feb
(17) |
Mar
(7) |
Apr
(29) |
May
|
Jun
|
Jul
|
Aug
|
Sep
|
Oct
|
Nov
|
Dec
|
S | M | T | W | T | F | S |
---|---|---|---|---|---|---|
|
|
|
1
(3) |
2
(5) |
3
(1) |
4
(3) |
5
(2) |
6
(3) |
7
(5) |
8
(2) |
9
(1) |
10
(2) |
11
(4) |
12
|
13
(1) |
14
(2) |
15
(2) |
16
(2) |
17
(2) |
18
|
19
(2) |
20
(2) |
21
(5) |
22
(8) |
23
(6) |
24
(10) |
25
(5) |
26
|
27
(1) |
28
(2) |
29
(2) |
30
(4) |
|
|
From: Radivoje Jovanovic <radivojejovanovic@gm...> - 2010-09-23 17:58:38
|
Hi, I am trying to cross-compile alsa-lib. As a result of building alsa-lib I get libasound.so file. However, if I try using aplay or anything that uses alsa I get: ALSA lib conf.c:2700:(snd_config_hooks_call) Cannot open shared library libasound_module_conf_pulse.so ALSA lib control.c:909:(snd_ctl_open_noupdate) Invalid CTL hw:0 I am not sure how to build libasound_module_conf_pulse.so. I see that the code for this library is in the alsa-lib folder. Thnak you Ogi |
From: Radivoje Jovanovic <radivojejovanovic@gm...> - 2010-09-23 17:53:33
|
Hi, I am trying to cross compile alsa-lib using the tool chain from my chip vendor. This has been done before (not sure by whom) so I know that tool chain works. However, when I run ./configure script: CC=MY_TOOL_CHAIN ./configure --target=arm-linux --host=arm --enable-shared -disable-static as output I get that: configure:8812: checking whether the MY_TOOL_CHAIN linker supports shared libraries configure:9803: result: yes ... configure:11392: checking if libtool supports shared libraries configure:11394: result: no configure:11397: checking whether to build shared libraries configure:11418: result: no. I am not sure why alsa needs special libtool and not use the libtool installed on the system? Thank you Ogi |
From: Kilian Sprotte <kilian.sprotte@gm...> - 2010-09-23 17:37:42
|
> I am giving up with the Tascam US-122L, hopefully I will have more > success with the edirol UA-25 EX. Hi, I just wanted to report that I am quite happy now with the UA-25 EX. Plug and play... :) There is just one thing about the .asoundrc that I did not succeed to find out and for which I would like to ask for some help. This works for me: speaker-test -c2 -r44100 -D plughw:UA25EX In my case plughw:UA25EX = plughw:1,0. What do I have to put into .asoundrc to make this my default device? I tried pcm.!default { type hw card 1 } ctl.!default { type hw card 1 } which seems to be not quite right, because I would like to use the plughw interface. speaker-test -c2 -r44100 -D default fails with this, because of wrong Sample format. Thanks in advance, Kilian |
From: Paul Braman <bramankp@gm...> - 2010-09-23 13:27:44
|
It seems like I must be missing something when it comes to recording audio from a device. What I expect to happen is for me to set some simple parameters (much like I do using OSS) and then have the API tell me what the best buffer size is for the device so that I can read reliably. For example ... #include <alsa/asoundlib.h> #include <assert.h> #include <stdint.h> #include <stdio.h> int main(int argc, char *argv[]) { unsigned int channels = 2; snd_pcm_t *pcm; assert(snd_pcm_open(&pcm, "hw:0,0", SND_PCM_STREAM_CAPTURE, 0) == 0); snd_pcm_hw_params_t *hwp; assert(snd_pcm_hw_params_malloc(&hwp) == 0); assert(snd_pcm_hw_params_any(pcm, hwp) == 0); assert(snd_pcm_hw_params_set_access(pcm, hwp, SND_PCM_ACCESS_RW_INTERLEAVED) == 0); assert(snd_pcm_hw_params_set_format(pcm, hwp, SND_PCM_FORMAT_S16) == 0); assert(snd_pcm_hw_params_set_channels(pcm, hwp, channels) == 0); assert(snd_pcm_hw_params_set_rate(pcm, hwp, 32000, 0) == 0); assert(snd_pcm_hw_params(pcm, hwp) == 0); snd_pcm_uframes_t frames; assert(snd_pcm_hw_params_get_buffer_size(hwp, &frames) == 0); fprintf(stderr, "target=%lu\n", (unsigned long)frames); snd_pcm_hw_params_free(hwp); int16_t buffer[frames*channels]; snd_pcm_sframes_t actual; again: while ((actual = snd_pcm_readi(pcm, buffer, frames)) == (snd_pcm_sframes_t){ frames }) { assert(fwrite(buffer, channels*sizeof(int16_t), actual, stdout) == (size_t){ actual }); } fprintf(stderr, "actual=%ld\n", (long)actual); if (actual > 0) { goto again; } snd_pcm_close(pcm); return 0; } I expect this code to run, without interruption, forever dumping audio to standard output. (Assuming no hardware faults, etc.) However, after a few seconds it fails when snd_pcm_mmap_read() returns a short read and then an error of -32 (-EPIPE). Are there some calls I'm not making in this startup sequence that are important? (Yes, I know, I'm not dealing with the mixer here ... one small example at a time.) Why shouldn't this work? I've seen code around that sets buffer times and periods and all that crap and if that is what I must do, what are the basic settings that will apply to any hardware I'm working with? (Onboard audio card, PCI tuner card, USB sound card, etc.) Once I figure this out I need to move on to S/PDIF recording and there are some real challenges I see there. Paul Braman |
From: Robert Parker <theparkers.mailbox@gm...> - 2010-09-23 11:57:54
|
> > I don't have any .asoundrc on my $HOME directory. What would be a > correct asoundrc file for me? Can it be done to system level instead > user level? > Yes I think so, have a look at http://alsa.opensrc.org/index.php/.asoundrc I know the hdmi is working for me when "speaker-test -D default" plays a hissing noise |
From: Mark Goldstein <goldstein.mark@gm...> - 2010-09-22 15:38:20
|
On Wed, Sep 22, 2010 at 5:26 PM, Bill Unruh <unruh@...> wrote: > On Wed, 22 Sep 2010, Mark Goldstein wrote: > >> On Wed, Sep 22, 2010 at 9:23 AM, Mark Goldstein >> <goldstein.mark@...> wrote: >>> >>> On Wed, Sep 22, 2010 at 8:54 AM, Axel Braun <axel.braun@...> wrote: >>>> >>>> Mark Goldstein wrote: >>>> >> was able to record the audio from Mic using audacity! After some more >> playing I found out that when I select "mute" for Capture1" it is >> actually unmuted and vise-versa. somehow Capture1 control is broken. >> Still I have very distorted Mic audio in Skype (and it also worked >> fine before). But it is a bit of progress! > > skype by default has automatic volume control So it may be that there is a > very weak signal coming in and skype is amplifying the hell our of it, > giving > a lot of noise (and distortion). > Make sure that the automatic volume control in skype is switched OFF. It is > pretty useless at the best of times. > If you mean this "allow Skype to automatically adjust my mixer levels" check-box, then yes, I know about it. I always disable it. Regards, -- Mark Goldstein |
From: Gennady Kupava <gb@bs...> - 2010-09-22 15:36:54
|
Hi, list, I am trying to setup LADSPA here with following config: pcm.pulseplug { type plug slave { pcm { type pulse server 192.168.0.1 } } } pcm.ladspaplug { type ladspa slave.pcm pulseplug path "/usr/lib/ladspa" plugins { 0 { label delay_n policy duplicate input { controls [ 1 1 ] } } } } pcm.!default { type plug slave.pcm ladspaplug } But libasound2 (1.0.23-1, as in debian/amd64/testing) crashing somewhere in ladspa plugin lib (according to core). plugin itself works fine in audacity. So, i tried to file bug report at https://bugtrack.alsa-project.org/alsa-bug/bug_report_page.php but it has no category 'PCM - ladspa' or something like 'other'. Also, i can't find way to report this problem with BTS. So, two things: 1. Is my config OK? 2. Can someone please add categories for poor reporters. And something like 'other' may be also useful? thanks, Gennady. |
From: Bill Unruh <unruh@ph...> - 2010-09-22 15:26:35
|
On Wed, 22 Sep 2010, Mark Goldstein wrote: > On Wed, Sep 22, 2010 at 9:23 AM, Mark Goldstein > <goldstein.mark@...> wrote: >> On Wed, Sep 22, 2010 at 8:54 AM, Axel Braun <axel.braun@...> wrote: >>> Mark Goldstein wrote: >>> > was able to record the audio from Mic using audacity! After some more > playing I found out that when I select "mute" for Capture1" it is > actually unmuted and vise-versa. somehow Capture1 control is broken. > Still I have very distorted Mic audio in Skype (and it also worked > fine before). But it is a bit of progress! skype by default has automatic volume control So it may be that there is a very weak signal coming in and skype is amplifying the hell our of it, giving a lot of noise (and distortion). Make sure that the automatic volume control in skype is switched OFF. It is pretty useless at the best of times. > > > Regards > -- William G. Unruh | Canadian Institute for| Tel: +1(604)822-3273 Physics&Astronomy | Advanced Research | Fax: +1(604)822-5324 UBC, Vancouver,BC | Program in Cosmology | unruh@... Canada V6T 1Z1 | and Gravity | http://www.theory.physics.ubc.ca/ |
From: Mark Goldstein <goldstein.mark@gm...> - 2010-09-22 15:16:10
|
On Wed, Sep 22, 2010 at 9:23 AM, Mark Goldstein <goldstein.mark@...> wrote: > On Wed, Sep 22, 2010 at 8:54 AM, Axel Braun <axel.braun@...> wrote: >> Mark Goldstein wrote: >> >>> I started thinking of buying some normal sound card :-(... >> >> Yep :-) >> >>> What holds me still is the fact that it did work perfectly till the >>> end of August. >>> I've got some advices on alsa bugzilla, but I still am too >>> "illiterate" to understand their slang :-). Started reading HDA spec >>> to at least understand the terminology. >> >> can you post the link to alsa-bugzilla? >> > > 11.3: > https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5130 > (I only added comments here and the reporter issue was that he had > Smart5.1 unmuted) > > 11.1: > https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5133 > Interesting! I started playing with hda-analyzer.py and came to conclusion that all connections are OK and the problem might be with the volume of "Capture1" control. I set it high in hda-analyzer and was able to record the audio from Mic using audacity! After some more playing I found out that when I select "mute" for Capture1" it is actually unmuted and vise-versa. somehow Capture1 control is broken. Still I have very distorted Mic audio in Skype (and it also worked fine before). But it is a bit of progress! Regards -- Mark Goldstein |
From: John Haxby <jch@th...> - 2010-09-22 09:51:44
|
On 22/09/10 09:46, Arxontis Politis wrote: > > --------------------------------------------------------------- > kane@...:~/Music$ dmesg | grep snd > [ 34.098749] snd_ua101: Unknown parameter `vid' > [ 34.107709] snd_ua101: Unknown parameter `vid' > [ 34.164993] snd_ua101: Unknown parameter `vid' > [ 34.368494] usbcore: registered new interface driver snd-usb-audio > -------------------------------------------------------------- > > I think you have a bad modprobe config file somewhere. I forget where Ubuntu keeps its modprobe options, but I suspect that somewhere you have a file that says something like options snd_ua101 vid=xyzzy instead of oprions snd_ua101 id=xyzzy (I don't know what the id should be, but the parameter name is "id" not "vid") jch |
From: Arxontis Politis <deadflagblue@gm...> - 2010-09-22 08:46:53
|
Hello, after upgrading to the latest Ubuntu (10.10), there seems to be some problem with the edirol ua-101 driver. The card is detected but the module cannot load. Here is the lsmod output: ------------------------------------------------------------- kane@...:~/Music$ lsmod | grep snd snd_hda_codec_si3054 3440 1 snd_hda_codec_realtek 217980 1 snd_hda_intel 22107 2 snd_hda_codec 87552 3 snd_hda_codec_si3054,snd_hda_codec_realtek,snd_hda_intel snd_usb_audio 86704 1 snd_pcm 71475 4 snd_hda_codec_si3054,snd_hda_intel,snd_hda_codec,snd_usb_audio snd_hwdep 5040 2 snd_hda_codec,snd_usb_audio snd_usbmidi_lib 17413 1 snd_usb_audio snd_seq_midi 4588 0 snd_rawmidi 17783 2 snd_usbmidi_lib,snd_seq_midi snd_seq_midi_event 6047 1 snd_seq_midi snd_seq 47174 2 snd_seq_midi,snd_seq_midi_event snd_timer 19067 2 snd_pcm,snd_seq snd_seq_device 5744 3 snd_seq_midi,snd_rawmidi,snd_seq snd 49006 18 snd_hda_codec_si3054,snd_hda_codec_realtek,snd_hda_intel,snd_hda_codec,snd_usb_audio,snd_pcm,snd_hwdep,snd_usbmidi_lib,snd_rawmidi,snd_seq,snd_timer,snd_seq_device soundcore 880 1 snd snd_page_alloc 7120 2 snd_hda_intel,snd_pcm --------------------------------------------------------------- and the dmesg output: --------------------------------------------------------------- kane@...:~/Music$ dmesg | grep snd [ 34.098749] snd_ua101: Unknown parameter `vid' [ 34.107709] snd_ua101: Unknown parameter `vid' [ 34.164993] snd_ua101: Unknown parameter `vid' [ 34.368494] usbcore: registered new interface driver snd-usb-audio -------------------------------------------------------------- The snd-usb-audio driver is loaded due to a usb-webcam with microphone, I don't think it has to do with the UA-101. Any help much appreciated! |
From: Mark Goldstein <goldstein.mark@gm...> - 2010-09-22 07:23:34
|
On Wed, Sep 22, 2010 at 8:54 AM, Axel Braun <axel.braun@...> wrote: > Mark Goldstein wrote: > >> I started thinking of buying some normal sound card :-(... > > Yep :-) > >> What holds me still is the fact that it did work perfectly till the >> end of August. >> I've got some advices on alsa bugzilla, but I still am too >> "illiterate" to understand their slang :-). Started reading HDA spec >> to at least understand the terminology. > > can you post the link to alsa-bugzilla? > 11.3: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5130 (I only added comments here and the reporter issue was that he had Smart5.1 unmuted) 11.1: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5133 Regards, -- Mark Goldstein |
From: Axel Braun <axel.braun@gm...> - 2010-09-22 06:54:59
|
Mark Goldstein wrote: > I started thinking of buying some normal sound card :-(... Yep :-) > What holds me still is the fact that it did work perfectly till the > end of August. > I've got some advices on alsa bugzilla, but I still am too > "illiterate" to understand their slang :-). Started reading HDA spec > to at least understand the terminology. can you post the link to alsa-bugzilla? Cheers Axel -- Linux on a ThinkPad: http://www.axxite.com/brn |
From: José Luis Segura Lucas <josel.segura@gm...> - 2010-09-21 14:47:05
|
El Tue, 21 Sep 2010 13:37:23 +0100 Robert Parker <theparkers.mailbox@...> escribió: > Have you tried unmuting devices using alsamixer ? Yes, of course. I tried using the Gnome-Volume-Manager and alsamixer. There is one thing that I haven't test: unmutting only one of the 4 switches... > In fedora 10 I found that I had to unmute an item called "IEC958" in > alsamixer to get HDMI working. At least I think it was called that, in > fc12 I now have to unmute "S/PDIF 1", even though it is the same h/w > (Asrock ION). My box is a Zotac Zbox HD-ID11 > In fc12 I also have to run "alsactl store" to remember the settings > made in alsamixer. I think I didnt have to do this step with fc10. Alsamixer keeps its settings, I checked it right now > My $HOME/.asoundrc looks like this > > pcm.!default { > type plug > slave.pcm "hdmi" > } > I don't have any .asoundrc on my $HOME directory. What would be a correct asoundrc file for me? Can it be done to system level instead user level? Thanks a lot for your answer |
From: Robert Parker <theparkers.mailbox@gm...> - 2010-09-21 12:37:29
|
Have you tried unmuting devices using alsamixer ? In fedora 10 I found that I had to unmute an item called "IEC958" in alsamixer to get HDMI working. At least I think it was called that, in fc12 I now have to unmute "S/PDIF 1", even though it is the same h/w (Asrock ION). In fc12 I also have to run "alsactl store" to remember the settings made in alsamixer. I think I didnt have to do this step with fc10. My $HOME/.asoundrc looks like this pcm.!default { type plug slave.pcm "hdmi" } |
From: José Luis Segura Lucas <josel.segura@gm...> - 2010-09-21 12:27:47
|
Good afternoon! I'm a "Debian unstable" user, using the latest Alsa packaged version. With that version, I was able to play sounds through my integrated Intel sound card, but I wasn't to use the HDMI sound capabilities. My "aplay -l" output could help: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC888 Analog [ALC888 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: ALC888 Digital [ALC888 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 I'm trying to activate the audio support of my Nvidia Ion graphic card, and I compile alsa-driver from the latest snapshot. I compile and install it using: $ ./configure.sh; make; $ sudo make install After the installation and reboot, my aplay -l output changes to that: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC888 Analog [ALC888 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: ALC888 Digital [ALC888 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 3: NVIDIA HDMI [NVIDIA HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 7: NVIDIA HDMI [NVIDIA HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 8: NVIDIA HDMI [NVIDIA HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 9: NVIDIA HDMI [NVIDIA HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 As you can see, the HDMI audio devices are working. On alsamixer I can see 4 IEC958 switches, but I tried enabling them all and I can't get audio from the HDMI. Can anybody help me? Thanks a lot. P.S. If you need another command output, dmesg or any other stuff, let me know. |
From: Mark Goldstein <goldstein.mark@gm...> - 2010-09-21 07:02:01
|
On Tue, Sep 21, 2010 at 8:45 AM, Axel Braun <axel.braun@...> wrote: > Hi Mark, > > Mark Goldstein wrote: > >> In 11.3 I worked with 2 (now 3) different kernel versions, all with >> alsa 1.0.23 and the behavior is the same. >> All my sound worked on 11.3 out of the box. Now the Mic seems working >> (I hear Mic audio in speaker and headphones) but can't record from Mic >> (level is 0) and can't use it in Skype - no audio. >> Yesterday I also noticed that if I'm trying alsaconfig on 11.3, >> alsaconfig does not find any sound card (even when I use run level 3). > > > ...it seems to go deeper: > https://bugzilla.novell.com/show_bug.cgi?id=639592 > > --- Comment #2 from Takashi Iwai <@novell.com> 2010-09-20 15:43:43 UTC --- > It's likely a known problem of VT1708S. It seems broken, unfortunately. > > No idea if someone going to fix it. > I started thinking of buying some normal sound card :-(... What holds me still is the fact that it did work perfectly till the end of August. I've got some advices on alsa bugzilla, but I still am too "illiterate" to understand their slang :-). Started reading HDA spec to at least understand the terminology. But if Takashi Iwai is saying that... too bad for us. Regards, -- Mark Goldstein |
From: Axel Braun <axel.braun@gm...> - 2010-09-21 06:46:26
|
Hi Mark, Mark Goldstein wrote: > In 11.3 I worked with 2 (now 3) different kernel versions, all with > alsa 1.0.23 and the behavior is the same. > All my sound worked on 11.3 out of the box. Now the Mic seems working > (I hear Mic audio in speaker and headphones) but can't record from Mic > (level is 0) and can't use it in Skype - no audio. > Yesterday I also noticed that if I'm trying alsaconfig on 11.3, > alsaconfig does not find any sound card (even when I use run level 3). ...it seems to go deeper: https://bugzilla.novell.com/show_bug.cgi?id=639592 --- Comment #2 from Takashi Iwai <@novell.com> 2010-09-20 15:43:43 UTC --- It's likely a known problem of VT1708S. It seems broken, unfortunately. No idea if someone going to fix it. Cheers Axel -- Linux on a ThinkPad: http://www.axxite.com/brn |
From: Luc GMail <lucmove@gm...> - 2010-09-20 11:42:16
|
I didn't have ALSA loopback so I could record streaming, so I upgraded ALSA following these instructions: http://alsa.opensrc.org/index.php/Jack_and_Loopback_device_as_Alsa-to-Jack_bridge#Compiling_snd-aloop_if_needed First thing: that page is WRONG. It instructs me to backup the kernel/sound directory: # p -a /lib/modules/`uname -r`/kernel/sound . "... In case anything went wrong and you wish to go back to your previous ALSA installation, no problem: sudo rm /lib/modules/`uname -r`/kernel/sound sudo cp -a ~/backup/sound /lib/modules/`uname -r`/kernel/ sudo alsa force-reload ... " But that is not enough. As soon as I ran 'sudo make install' I saw that another directory should have been backed up: if [ -L /usr/include/sound ]; then \ rm -f /usr/include/sound; \ I didn't make a backup of /usr/include/sound and now I am screwed. The backup /lib/modules/`uname -r`/kernel/sound directory doesn't work at all. My sound card is not even recognized when I boot with it. What about the new, upgraded ALSA modules? Well, they *look* OK, but I don't get any sound through the speakers or headphone jack. The only part I didn't follow verbatim in those instructions was the final step: "... Allrighty, time to load it. But before that, shut down all audio apps (including firefox). Once done, do this: sudo alsa force-unload sudo modprobe snd-whatever-module-you-need sudo modprobe snd-aloop ..." I was unsure about "snd-whatever-module-you-need" so I decided to just reboot. When KDE was just about to kick in, I saw a lot of new text I had never seen before scroll past very, very fast. Then it stopped and I could see it had something to with ALSA, but I couldn't really read anything. KDE started and... now I have no sound. I checked kmixer, it looks normal. It doesn't seem to be mute. I ran alsamixer in the command line, it doesn't seem to be mute either. I ran Kcontrol and checked the audio section, couldn't figure anything out. Here is the output of lsmod BEFORE the upgrade: $luc> lsmod | grep snd snd_hda_intel 434100 6 snd_pcm_oss 46336 0 snd_mixer_oss 22656 2 snd_pcm_oss snd_pcm 83076 3 snd_hda_intel,snd_pcm_oss snd_seq_dummy 10756 0 snd_seq_oss 37760 0 snd_seq_midi 14336 0 snd_rawmidi 29696 1 snd_seq_midi snd_seq_midi_event 15104 2 snd_seq_oss,snd_seq_midi snd_seq 56880 7 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_timer 29704 3 snd_pcm,snd_seq snd_seq_device 14988 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq snd 62756 18 snd_hda_intel,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_seq_oss,snd_rawmidi,snd_seq,snd_timer,snd_seq_device soundcore 15200 2 snd snd_page_alloc 16904 2 snd_hda_intel,snd_pcm Here is the output of lsmod AFTER the upgrade: $luc> lsmod | grep snd snd_hda_codec_realtek 226308 1 snd_hda_intel 35840 3 snd_hda_codec 100992 2 snd_hda_codec_realtek,snd_hda_intel snd_hwdep 15364 1 snd_hda_codec snd_pcm_oss 46112 0 snd_mixer_oss 23040 3 snd_pcm_oss snd_pcm 85636 3 snd_hda_intel,snd_hda_codec,snd_pcm_oss snd_seq_oss 36224 0 snd_seq_midi_event 15232 1 snd_seq_oss snd_seq 57584 5 snd_seq_oss,snd_seq_midi_event snd_timer 29320 2 snd_pcm,snd_seq snd_seq_device 15372 2 snd_seq_oss,snd_seq snd 69412 14 snd_hda_codec_realtek,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_seq_oss,snd_seq,snd_timer,snd_seq_device soundcore 15200 3 snd snd_page_alloc 17032 2 snd_hda_intel,snd_pcm Looks like I have lost snd_seq_dummy, snd_seq_midi and snd_rawmidi. I don't need midi, so I just tried to load snd_seq_dummy with modprobe. The module is not found. I rebooted another two times hoping to be able to read something in that new boot sequence text, but I don't see it anymore. It's gone. I don't know what else to do. I can't go back to the backup modules, they are useless without /usr/include/sound which the installation method erased and nobody warned me it would. Please advise. |
From: vaibhav wasnik <wasnik.vibe@gm...> - 2010-09-20 09:11:41
|
Hi friends, could someone help me with this. I dont see no soundcards on lshw, aplay-l or lspci Attached is my aadebug file ALSA Audio Debug v0.1.0 - Mon Sep 20 05:02:29 EDT 2010 http://alsa.opensrc.org/aadebug http://www.gnu.org/licenses/gpl.txt Kernel ---------------------------------------------------- Linux wasnik-laptop 2.6.32-24-generic #43-Ubuntu SMP Thu Sep 16 14:17:33 UTC 2010 i686 GNU/Linux Loaded Modules -------------------------------------------- snd_hda_intel 21470 0 snd_hda_codec 87540 1 snd_hda_intel snd_hwdep 5604 1 snd_hda_codec snd_pcm_oss 34539 0 snd_mixer_oss 13865 1 snd_pcm_oss snd_pcm 71582 3 snd_hda_intel,snd_hda_codec,snd_pcm_oss snd_seq_dummy 1434 0 snd_seq_oss 27210 0 snd_seq_midi 4557 0 snd_rawmidi 19077 1 snd_seq_midi snd_seq_midi_event 6003 2 snd_seq_oss,snd_seq_midi snd_seq 47498 6 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_timer 18646 2 snd_pcm,snd_seq snd_seq_device 5988 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq snd 56271 13 snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq,snd_timer,snd_seq_device snd_page_alloc 7076 2 snd_hda_intel,snd_pcm Modprobe Conf --------------------------------------------- Warning: module config file does not exist This means any kernel modules will not be auto loaded See your linux distro docs on how to create this file Proc Asound ----------------------------------------------- Advanced Linux Sound Architecture Driver Version 1.0.23. Compiled on Sep 19 2010 for kernel 2.6.32-24-generic (SMP). --- no soundcards --- 1: : sequencer 33: : timer Client info cur clients : 3 peak clients : 3 max clients : 192 Client 0 : "System" [Kernel] Port 0 : "Timer" (Rwe-) Port 1 : "Announce" (R-e-) Connecting To: 15:0 Client 14 : "Midi Through" [Kernel] Port 0 : "Midi Through Port-0" (RWe-) Client 15 : "OSS sequencer" [Kernel] Port 0 : "Receiver" (-we-) Connected From: 0:1 Dev Snd --------------------------------------------------- seq timer CPU ------------------------------------------------------- model name : AMD Turion(tm) 64 X2 Mobile Technology TL-58 cpu MHz : 800.000 model name : AMD Turion(tm) 64 X2 Mobile Technology TL-58 cpu MHz : 800.000 RAM ------------------------------------------------------- MemTotal: 1996316 kB SwapTotal: 5847032 kB Hardware -------------------------------------------------- 00:18.0 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] HyperTransport Technology Configuration 00:18.1 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Address Map 00:18.2 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] DRAM Controller 00:18.3 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Miscellaneous Control my output for alsa-info.sh is below WARNING: All config files need .conf: /etc/modprobe.d/options, it will be ignored in a future release. alsa-info.sh: 381: [[: not found ALSA Information Script v 0.4.59 -------------------------------- This script visits the following commands/files to collect diagnostic information about your ALSA installation and sound related hardware. dmesg lspci lsmod aplay amixer alsactl /proc/asound/ /sys/class/sound/ ~/.asoundrc (etc.) See 'alsa-info.sh --help' for command line options. alsa-info.sh: 621: [[: not found alsa-info.sh: 621: [[: not found alsa-info.sh: 621: [[: not found alsa-info.sh: 621: [[: not found alsa-info.sh: 621: [[: not found alsa-info.sh: 621: [[: not found alsa-info.sh: 621: [[: not found alsa-info.sh: 621: [[: not found WARNING: All config files need .conf: /etc/modprobe.d/options, it will be ignored in a future release. alsa-info.sh: 621: [[: not found alsa-info.sh: 748: [[: not found alsa-info.sh: 773: [[: not found Automatically upload ALSA information to http://www.alsa-project.org? [y/N] : read: 773: Illegal option -e alsa-info.sh: 809: [[: not found alsa-info.sh: 809: [[: not found Your ALSA information is in /tmp/alsa-info.txt.JKuRTjZP0q wasnik@...:~$ pico /tmp/alsa-info.txt.JKuRTjZP0q |
From: Stephen Kirkby <bugs@st...> - 2010-09-19 12:08:16
|
Hi, I'm trying to get hdmi digital pass through sound working. I've got it working by using iecset first, this works: saruman ~ # iecset -c0 -n1 audio false Mode: consumer Data: non-audio Rate: 44100 Hz Copyright: permitted Emphasis: none Category: general Original: 1st generation Clock: 1000 ppm saruman ~ # iecset -c0 -n1 -x AES0=0x06,AES1=0x00,AES2=0x00,AES3=0x00 saruman ~ # aplay -D hdmi:CARD=Intel,DEV=0 SURROUNDTEST_DD_640.wav However, I need to be able to set to use audio mode with some content and not others (for mythtv). So next I tried to pass the AES flag to aplay which I believe should set the audio mode: saruman ~ # iecset -c0 -n1 audio true Mode: consumer Data: audio Rate: 44100 Hz Copyright: permitted Emphasis: none Category: general Original: 1st generation Clock: 1000 ppm saruman ~ # iecset -c0 -n1 -x AES0=0x04,AES1=0x00,AES2=0x00,AES3=0x00 saruman ~ # aplay -D hdmi:CARD=Intel,DEV=0,AES0=0x06,AES1=0x00,AES2=0x00,AES3=0x00 SURROUNDTEST_DD_640.wav however this doesn't seem to override the current AES0=0x04 settings! I'm running on ALSA 1.0.23 with a HDA-Intel card - full details of my system setup (generated by alsa-info.sh) are here: http://www.alsa-project.org/db/?f=8f0f4cd6e869b602ee37f70a3d64c9aa6b6ccffe How do I override the AES0 flag? Any help is appreciated! Thanks Steve |
From: art3c <bluesbyte@gm...> - 2010-09-19 11:44:25
|
Hello, Is there any way to get Midi input working in Emu 0404? I tried with latest Alsa + Alsa firmware on Ubuntu 10.04 and i got some Midi messages, but they were sensless (for me and my application ;) ). Thanks for help. -- art3c |
From: David Kastrup <dak@gn...> - 2010-09-17 18:45:23
|
Kilian Sprotte <kilian.sprotte@...> writes: > I am giving up with the Tascam US-122L, hopefully I will have more > success with the edirol UA-25 EX. > > Is there any other USB interface with good sound quality that you > favor? Well, I am using a Hammerfall DSP Multiface with a PC-card interface. As opposed to USB buses, you have good latencies/response on PC-card (data transfer does not need the CPU at all, so you just need reasonable interrupt response times). I still don't get xrun-free operation without using an rt kernel and the rtirq-init package. And pulseaudio has hickups in Flash and wants tsched=0 in Ubuntu 10.04, but jack runs fine. On the plus side, quality is good, you have 8 balanced analog ins and outs, 8-channel ADAT in/out, word clock in/out (so you can use more than one unit if really necessary), Midi in/out, SPDIF in/out (optical if you don't use ADAT, and coax anyway), and can go 96/24 if you want, and the analog circuits don't make a mockery of that mode. And you have crossbar mixing capabilities. On the down side, you still need mic preamps. If you can manage to get to your memory bus without going through USB at all (Cardbus or so), you have somewhat more reassuring prospects to get low-latency jobs (like real-time effects) and dropless operation going. The Hammerfall sounded like a good proposition to me in that respect. Also because I don't think ASIO generally works with Linux USB, but then I might be mistaken here. -- David Kastrup |
From: Kilian Sprotte <kilian.sprotte@gm...> - 2010-09-17 18:23:13
|
Hi, I am giving up with the Tascam US-122L, hopefully I will have more success with the edirol UA-25 EX. Is there any other USB interface with good sound quality that you favor? Best, Kilian Kilian Sprotte <kilian.sprotte@...> writes: > Hi, > > I have a got a new US-122L and I am trying to get it to work with Ubuntu > and also Gentoo. > > Linux 2.6.32-24-generic > Advanced Linux Sound Architecture Driver Version 1.0.20. > > My success goes as far that it shows up under cards: > > $ cat /proc/asound/cards > 0 [Intel ]: HDA-Intel - HDA Intel > HDA Intel at 0xee240000 irq 17 > 1 [US122L ]: USB US-122L - TASCAM US-122L > TASCAM US-122L (644:800e if 0 at 001/002) > > But does not really qualify as an audio device (?): > > $ cat /proc/asound/devices > 2: : timer > 3: : sequencer > 4: [ 0- 1]: digital audio playback > 5: [ 0- 0]: digital audio playback > 6: [ 0- 0]: digital audio capture > 7: [ 0- 0]: hardware dependent > 8: [ 0] : control > 9: [ 1- 0]: hardware dependent > 10: [ 1- 0]: raw midi > 11: [ 1] : control > > Is the US-122L supposed to be supported as an audio device? > > Thanks in advance, > Kilian -- Kilian Sprotte sprotte.org |
From: Bankim Bhavsar <bankim.bhavsar@gm...> - 2010-09-16 17:16:43
|
Producer thread writes sound bytes worth 10 ms every 10 ms to a ring buffer. Consumer thread is responsible for writing sound bytes to ALSA sound device in a strictly non-blocking fashion. Consumer thread checks snd_pcm_avail/snd_pcm_avail_update, fetches minimum of sound bytes available in ring buffer and availFrames(returned by snd_pcm_avail_update) and writes to sound device using snd_pcm_writei(). Consumer thread polls every 20 ms. However on consumer thread snd_pcm_avail/snd_pcm_avail_update() returns 0 at least 3-4 times consecutively (total 60-80 ms) and on every 5th-6th poll snd_pcm_avail/snd_pcm_avail_update returns full buffer size occasionally reporting an under-run. On an average, under-run is reported at least 10-15 times a minute which is not desirable. Following are the hardware and software parameter settings. hardware params: 16-bit 44100 stereo buffer time 200 ms period time 25 ms software params: start_threshold: full buffer size no setting for avail_min Alternatively I've tried using wait_for_poll() technique as mentioned in the example. http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html However poll() unblocks very rapidly and producer ring buffer doesn't have enough bytes. What's the best technique for PCM playback for such use cases? Is it okay to skip calling snd_pcm_avail/snd_pcm_avail_update and directly attempt writing to ALSA sound device instead? Let me know if more information is required. Thanks, Bankim. |