OSS4 vs ALSA vs Pusleaudio vs Jack
I wonder which sound system is better? I used ALSA, resently switched to OSS4 and I seam to like it better.
I've also tried to get jack to work, but without any success... I haven't tried pulseaudio... So what are the user's impressions from each of these systems?
Last year, I used alsa with my snd-hda-intel (nvidia nforce chipset)
After spending hours (well coming from gentoo I learned patience, but it's still really frustrating),
I found an asound.conf working with :
alsa apps (yes some users can't ..)
my mic
enabling oss emulation without monopolizing the device ( can't play UT2004 with sound otherwise)
glest (don't know why, it didn't work out of the box)
java ( if you read java faq, they don't want to support software mixing, /dev/dsp, that's all)
and a working spdif
now I'm in China and I bought a new laptop, snd-hda-intel too
I install Ubuntu because I wanted something working out of the box, (it's not like I can borrow my friends internet connection 'till 6am)
guess what, alsa didn't work, and I had tried my lisp alsa config (why should we learn lisp to make alsa works anyway?) , no luck either, I tested again and again ...
well, I came across articles about how great OSSv4 was, particularly, the development of HDAudio driver, so I gave it a go
nothing to configure except blacklisting the alsa drivers. it works out of the box!!
The sound is crystal clear ( Alsa with HDA was not very loud I recall), I can even distinguish the cracking sound of bad mp3 encode on my laptop speakers with no effort (actually maybe this is not nice)
now, sadly enough, kmix doesn't yet support OSS
and plug in an head set doesn't mute automatically the rest.
I don't now about the spdif, I'll test it later
Anyway I'm totally converted to OSS
Similar Messages
-
2 sound cards one using OSS4, other ALSA
So, is it possible to have 2 sound cards where one will use OSS4 and other will use ALSA?
When you use oss4, one of the things it does is remove/disable all the alsa drivers that come with the kernel.
Both oss4 and alsa try to take full control of all audio devices so without changes to oss4 and/or driver cherry picking/modification I suppose it will not work. -
Hi all,
I'm currently running ALSA as a backend to JACK, and Pulseaudio over everything for my normal audio apps. JACK is used because I do some audio work on the side, and its also useful for redirecting output/input etc. Pulseaudio, contrary to the experiences of most, has been a gem for me, no real problems. But the only real feature I use is the per-app volume control, that and the good support for all applications (as compared with a pure JACK to ALSA setup which gave me quite a bit of trouble).
My question is, is the above setup replace-able with OSS4 and JACK alone? JACK can use OSS as a backend, it seems, and OSS also provides its own per-app mixer. The points I'm looking for is:-
1. per-app volume control (ossmix does this, I believe)
2. per-app redirection of output a la Pulseaudio (not sure whether ossmix handles this, another alternative is directing ossmix output to JACK, is that possible?)
3. app-compatibility, the wiki seems to indicate that all the common apps are supported (only skype needing a separate install).
So, any comments/advise from anyone?litemotiv wrote:
venky80 wrote:i thought oss4 development is dead ..coz the lead developer quit
it's not dead, and he did not quit (yet). he did say he will have less time to work on it as the switch to open source lost him a steady income (yes it's ironic, since going open source was how he actually planned to get more work done on OSS). these are his current plans:
I will probably start in a new job within next couple of weeks. This job will be kernel development but not related with OSS. This is all I can disclose at this moment. However this doesn't mean that I stop developing OSS. Not at all. Getting a full time job outside OSS is just better way than any of the other alternatives. It removes the economical risks and I can focus on doing actual work instead of worrying about the economy side all the time. The other alternative is doing some side business (selling OSS T-shirts or trying to invent some web2.0 service) to fund development of OSS. However this will simply not work. The economical risks in the side business are even higher than in the OSS business itself.
I will continue working on OSS in the following ways:
1. I will continue as the maintainer of the development version of OSS (currently 4.2).
2. I will do OSS related development on contract if some customer wants to pay. This includes drivers for new sound cards, adding new features to OSS and also development of custom applications.
3. There are earlier commitments like our contract with Sun.
4. Hacking "just for fun". I will continue working on OSS features I need myself and features I see interesting in other ways.
OSS is still a long way of being a broadly implemented stable audio subsystem though. More info here:
http://4front-tech.com/hannublog/?m=200903
Thanks , I am happy to see that. I use ALSa though but iam glad there is an alternative -
I'm a new user. Recently I've done an upgrade and had to install OSS for a game and my ALSA driver seems to be completely gone.
# /etc/rc.d/alsa start
:: Restoring ALSA Levels [BUSY] /usr/sbin/alsactl: load_state:1569: Cannot open /etc/asound.state for reading: No such file or directory
[DONE]
so I tried aslactl store to restore the file:
# alsactl store
alsactl: save_state:1502: No soundcards found..
No go...
# ls -l /dev/dsp
lrwxrwxrwx 1 root root 26 2009-10-12 13:30 /dev/dsp -> /dev/oss/oss_hdaudio0/pcm0
# aplay -l
aplay: device_list:223: no soundcards found...
# alsamixer
cannot open mixer: No such file or directory
However, alsa-lib, alsa-utils, alsa-oss, alsa-plugins. I even tried to install a bunch of alsa-related stuff from AUR as well. I don't know why its not being found. I'm really hating how OSS can only manage 1 sound source.
pacman -S alsa-lib alsa-oss alsa-plugins
warning: alsa-lib-1.0.21.a-1 is up to date -- reinstalling
warning: alsa-oss-1.0.17-1 is up to date -- reinstalling
warning: alsa-plugins-1.0.21-1 is up to date -- reinstalling
resolving dependencies...
looking for inter-conflicts...
Targets (3): alsa-lib-1.0.21.a-1 alsa-oss-1.0.17-1 alsa-plugins-1.0.21-1
Total Download Size: 0.00 MB
Total Installed Size: 2.67 MB
Proceed with installation? [Y/n] y
checking package integrity...
(3/3) checking for file conflicts [#####################] 100%
(1/3) upgrading alsa-lib [#####################] 100%
(2/3) upgrading alsa-oss [#####################] 100%
(3/3) upgrading alsa-plugins [#####################] 100%
Install jack-audio-connection-kit, libsamplerate, ffmpeg
or pulseaudio to get their respective plugins working
Also, it shows up in lspci along with many other things
00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 02)
Subsystem: ASUSTeK Computer Inc. P5B
Flags: fast devsel, IRQ 22
Memory at febf8000 (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 <?>
Anyone have any suggestions?
Thanks
edit: added more info
Last edited by BowlSmoker (2009-10-12 18:14:56)I am having the exact same problem, I have no idea how to fix it or what it going wrong. Any help would be appreciated.
There is only one difference with my problem
ls -l /dev/dsp
ls: cannot access /dev/dsp: No such file or directory
/dev/dsp does not have a soft link -
I'm using mpd (Music Player Deamon) at the moment to listen to my music collection, and although ALSA works without a hitch, I wanted to have sound on all my speakers for normal stereo sources.
In my ~/.asoundrc file I added add the following (as per instruction on the ALSA website)
pcm.duplicate {
type plug
slave.pcm "surround51"
slave.channels 6
route_policy duplicate
After that I set the ALSA device used for mpd to duplicate.
For mpd it's in /etc/mpd.conf under the audio_ouput section.
I added
device "duplicate"
Although this works, it seems to mess up the hardware mixing on my sound card (an AC97 variant). If I use the normal "surround51″ device, all is well.
Any ideas how I, or if, I can solve this?I've found the solution. Thought I'd share.
First things first, make sure you have JACK installed. Then, edit your ~/.asoundrc file to look like this
ctl.jack51 {
type hw
card 0
pcm.jack51 {
# "asym" allows for different
# handling of in/out devices
type asym
playback.pcm {
# route for mmap workaround
type plug
slave.pcm "surround51"
slave.channels 6
route_policy duplicate
capture.pcm {
# 2 channels only
type hw
card 0
Then, start JACK with
jackd -d alsa -d jack51
(Loading JACK at startup can be done in a variety of ways, depending on how you do things)
I had trouble starting it up as a regular user, but that can be fixed by editing /etc/security/limits.conf (as described in the wiki).
It's nice to have qjackctl installed to control JACK from your desktop environment.
pacman -S qjackctl
If your app supports JACK output, you're good to go! Simply set it to use JACK.
For MPD, I had to install the SVN 4125 trunk. Download the PKGBUILD and simply edit the version to the newest SVN build (check this on mpd's site). The 4125 release has support for JACK output.
Then, in /etc/mpd.conf, use this
audio_output {
type "jack"
name "mpd"
Et voila! Surround sound with mixing. -
Firewire audio with jack, route audio to card?
Hi
I hooked up my focusrite saffire 24 pro to my p150em and managed to get qjackcontrol to show that it is connected with 48khz sample rate and the green light is lit up on my saffire. So that's good.
I then tried listening to sounds from various programs, mpg123, cmus and google-chrome (youtube) but encountered errors like "connection refused" from audio server. So I installed alsa-plugins and configured jack to use alsa. Now I can hear things again, but of course, only through my intel soundcard and not the fw saffire!
I figured maybe I could open alsamixer and choose saffire as my soundcard? But it's not there.
How to I route the audio through the saffire?
Best regards,
LoveI have a very cheap C-Media PCI CMI8738-MC6 6-channel sound card connected by three jacks to Logitech 5.1 speakers. I also have a headset for Skype connected to the onboard sound.
Here's what I know:
1) To always get the sound card module to load before the onboard sound module, I have added snd_cmipci to my "modules" list in /etc/rc.conf. This means the cmi card is always card 0, the module for the onboard sound gets loaded automatically later.
2) I have ~.asoundrc as follows:
# 6 channel dmix:
pcm.dmix6 {
type dmix
ipc_key 1024
ipc_key_add_uid false
ipc_perm 0660
slave {
pcm "hw:0,1"
rate 48000
channels 6
period_time 0
period_size 1024
buffer_time 0
buffer_size 5120
# upmixing:
pcm.ch51dup {
type route
slave.pcm dmix6
slave.channels 6
ttable.0.0 1
ttable.1.1 1
ttable.0.2 1
ttable.1.3 1
ttable.0.4 0.5
ttable.1.4 0.5
ttable.0.5 0.5
ttable.1.5 0.5
pcm.duplex {
type asym
playback.pcm "ch51dup" # upmix first
capture.pcm "hw:0"
# change default device:
pcm.!default {
type softvol
slave.pcm "duplex"
control {
name "Software Master"
card 0
# for aoss
pcm.dsp "duplex"
pcm.dsp1 "duplex"
3) To allow amarok to play each speaker individually using a 6 channel DTS .wav file I added the following to my ~.xine/config
audio.device.alsa_surround51_device:plughw:0,1
4) You can download a 60MB 6-channel DTS test file here. It is in swedish but tests each speaker individually.
My info came from browsing the Alsa wiki -
[RESOLVED] MPD can't add files to db, says ignoring unrecognized file
Sorry to bother you guys with this, but I have searched a while on google and the forums and come up empty.
I'm trying to add some music to my database with mpd and ncmpcpp. I start mpd and it starts fine, and start ncmpcpp and it starts fine, and hit the 'u' key to update. Then I look at my library. It has all the folders that are in my music folder, but none of the .mp3 files are listed. So I started mpd with
mpd --verbose --stdout --no-daemon
And hit the 'u' key in ncmpcpp again, and for every file in my music folder, mpd says such things as
update: ignoring unrecognized file Snow Patrol/Chocolate.mp3
I am running mpd as a user (not as root or as a daemon) and even then I checked all the permissions on all the relevant files and directories and they all seem to be good.
I searched a lot and didn't find this error. Has anyone else had this problem and fixed it?
Last edited by kens (2011-11-16 22:25:47)Here's the output of mpd --version:
mpd (MPD: Music Player Daemon) 0.16.5
Copyright (C) 2003-2007 Warren Dukes <[email protected]>
Copyright (C) 2008-2010 Max Kellermann <[email protected]>
This is free software; see the source for copying conditions. There is NO
warranty; not even MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
Supported decoders:
[mad] mp3 mp2
[vorbis] ogg oga
[oggflac] ogg oga
[flac] flac
[sndfile] wav aiff aif au snd paf iff svx sf voc w64 pvf xi htk caf sd2
[audiofile] wav au aiff aif
[faad] aac
[mp4ff] m4a m4b mp4
[mpcdec] mpc
[wavpack] wv
[modplug] 669 amf ams dbm dfm dsm far it med mdl mod mtm mt2 okt s3m stm ult umx xm
[ffmpeg] 16sv 3g2 3gp 4xm 8svx aa3 aac ac3 afc aif aifc aiff al alaw amr anim apc ape asf atrac au aud avi avm2 avs bap bfi c93 cak cin cmv cpk daud dct divx dts dv dvd dxa eac3 film flac flc fli fll flx flv g726 gsm gxf iss m1v m2v m2t m2ts m4a m4b m4v mad mj2 mjpeg mjpg mka mkv mlp mm mmf mov mp+ mp1 mp2 mp3 mp4 mpc mpeg mpg mpga mpp mpu mve mvi mxf nc nsv nut nuv oga ogm ogv ogx oma ogg omg psp pva qcp qt r3d ra ram rl2 rm rmvb roq rpl rvc shn smk snd sol son spx str swf tgi tgq tgv thp ts tsp tta xa xvid uv uv2 vb vid vob voc vp6 vmd wav wma wmv wsaud wsvga wv wve
Supported outputs:
shout null fifo alsa ao oss pulse jack httpd recorder
Supported encoders:
null vorbis lame wave flac
Supported protocols:
file:// http:// mms:// mmsh:// mmst:// mmsu:// gopher:// rtp:// rtsp:// rtmp:// rtmpt:// rtmps://
It seems to support mp3 but it's still giving the "ignoring unrecognized file" thing. -
Audio - Link Microphone and playback into one device?
Is it possible to merge two audiosources togeather in any nifty way?
Mainly interested in the default recording device (or otherwise specified) with the playback device.
I'm using JACK which i thought was able to do so but never got it working in any practical way al tho i enjoy playing around with it.
Why i want to do it is because i want ffmpeg to easily pick up two sources from one channel, and being able to switch around in say JACK would be awesome in real time not affecting the ffmpeg stream, considering i might be streaming something for a bit and want to switch audio source mid-stream to erm i dunno say a radiostation while i go off doing something else.
Linux is versatile so i bet it's possible, i just don't know how yetI've got this partially solved:
#!/bin/bash
KEY=`cat key.db`
ffmpeg \
-i http://pub1.di.fm:80/di_vocaltrance \
-f alsa -ac 2 -i hw:0 -acodec mp3 -aq 1 -ab 128k \
-f x11grab -s 1920x1080 -r 25 -i :0.0 \
-c:v libx264 -preset fast -pix_fmt yuv420p -s 1280x800 \
-c:a mp3 -ab 96k -ar 22050 \
-threads 0 \
-map 0:0 -map 1:0 -map 2:0 \
-f flv "rtmp://live.twitch.tv/app/$KEY"
But now for the tricky part..
Everything sounds like i'm inside a tin-can for some reason (probably to do with the mixing, ffmpeg probably don't like that for the format?).
So i thought i'd give ffmpeg-full a go with jack support.
Giving me the option to link several inputs into one single channel making it a single-audio track in the video, also giving me the option to patch this by muxers and what not.. But it gives me this:
-f alsa -ac 2 -i jack \
Produces:
[alsa @ 0x1399280] cannot set sample format 0x10000 2 (Invalid argument) jack: Input/output error -
Aqualung + skins pkgbuild. new music player for linux.
Aqualung is a new linux music player. More info and screenshots are available here http://aqualung.sourceforge.net/
It supports alsa, oss and jack and uses gtk2.
aqualung pkgbuild
pkgname=aqualung
pkgver=0.9beta1
pkgrel=1
pkgdesc="Skinable xmms/bmp like music player"
url="http://aqualung.sourceforge.net"
depends=('gtk2' 'libxml2' 'liblrdf'
'jack-audio-connection-kit' 'alsa-lib'
'libsndfile' 'flac' 'libvorbis'
'libmad' 'libsamplerate')
source=(http://dl.sourceforge.net/sourceforge/$pkgname/$pkgname-$pkgver.tar.gz)
md5sums=('85d9dbb91052df9f755f0a985ee7fa48')
build() {
cd $startdir/src/$pkgname-$pkgver
./configure --prefix=/usr
make || return 1
make prefix=$startdir/pkg/usr install
aqualung-skins pkgbuild
pkgname=aqualung-skins
pkgver=1.0
pkgrel=1
pkgdesc="Additional skins for Aqualung music player"
url="http://aqualung.sourceforge.net/download.html"
depends=('aqualung')
source=(
http://aqualung.sf.net/skins/aqualung-dark.tar.gz
http://aqualung.sf.net/skins/aqualung-metal.tar.gz
http://aqualung.sf.net/skins/aqualung-woody.tar.gz
md5sums=(
'4bdfb815875db5bfefaa138c3b93833b'
'193ec183c7050cf8f844287953c3d6e8'
'd13546c75e5712d9fed880d0e3a7e792'
build() {
cd $startdir/src
install -d -m755
$startdir/pkg/usr/share/aqualung/skin/dark
install -m644 $startdir/src/dark/*
$startdir/pkg/usr/share/aqualung/skin/dark
install -d -m755
$startdir/pkg/usr/share/aqualung/skin/metal
install -m644 $startdir/src/metal/*
$startdir/pkg/usr/share/aqualung/skin/metal
install -d -m755
$startdir/pkg/usr/share/aqualung/skin/woody
install -m644 $startdir/src/woody/*
$startdir/pkg/usr/share/aqualung/skin/woody
Special thanks to all the developers, package maintainers, trusted users and other archers - every little lib was already available
I know that a few dependencies are already covered by others (as namcap reports them) but I left them all just to be "compatibile" with aqualung required and optional libraries list (check aqualung homepage for more info).
In this version gui skin chooser doesn't work (at least for me) - it can't see other skins. You can change the default skin to other (build and install aqualung-skins first) manually by changing it in the config file in your ~/.aqualung.
For fast start just run aqualung --output alsa (or oss or jack if you prefer). Jack is required always but it doesn't need to run if you are using only alsa/oss. More info on aqualung homepage.
...and for the future releases a cvs version pkgubild (no difference for now - the project is too fresh :-) but you can use it as a template)
aqualung-cvs pkgbuild
pkgname=aqualung-cvs
pkgver=0.9beta1
pkgrel=1
pkgdesc="Skinable xmms/bmp like music player"
url="http://aqualung.sourceforge.net"
makedepends=('cvs')
depends=('gtk2' 'libxml2' 'liblrdf'
'jack-audio-connection-kit' 'alsa-lib'
'libsndfile' 'flac' 'libvorbis'
'libmad' 'libsamplerate')
conflicts=('aqualung')
provides=('aqualung')
cvsroot=":pserver:anonymous:@cvs.sf.net:/cvsroot/aqualung"
cvsmod="aqualung"
build() {
cd $startdir/src
msg "Logging in to anonymous CVS and fetching sources..."
cvs -Q -d $cvsroot login
cvs -Q -z3 -d $cvsroot co $cvsmod
cd $cvsmod
./autogen.sh
./configure --prefix=/usr
make || return 1
make prefix=$startdir/pkg/usr installagoogle search will lead you to sites that have skins
when you connect your phone to software updater you can check if there is an update available whenever your connected to pc suite a message should come through that there is an update availalble when it arrives
If i have helped at all a click on the white star below would be nice thanks.
Now using the Lumia 1520 -
I installed iTunes, the sound quality dropped considerably. I removed my Terratec Aureon 5.1 Fun soundcard and now the sound is fine. I have searched these forums for help under 'terratec' and 'aureon' and found nothing. I'm using two speakers now instead of the 5.1 setup which worked fine with Windows Media Player and Realplayer. Any ideas anyone?
ooo wrote:
How about if you use alsa as audio output with vlc? If you got the alsa-jack-plug working, it should handle the downmixing to stereo just fine. I've been using mplayer and its variants this way and never had problems with 5.1 audio.
the html5 video issue is strange though. especially if flash videos are working fine. I'm using chromium myself and never had any sound issues on my arch install with it. However I just tried html5 videos with firefox and they doesn't seem to work for me either (video seems to be loading fine but it just keeps buffering and doesn't start playback). I have no ideas how to solve this unfortunately unless firefox has its own internal sound settings somewhere..
If I set alsa as the audio uotput in vlc and jackd is already running I get:
Audio output failed:
The audio device "sysdefault:CARD=Intel" could not be used:
Device or resource busy.
Maybe I have something misconfigured - but I have exactly what you posted in .asoundrc (there is no /etc/asound.conf on my system.) and I can;lt see what I can possibly have mis-configured.
Is it right that config you gave for .asoundrc means that I set my apps to use ALSA and ALSA takes care of routing the audio through JACK.
If that's true - why does vlc complain about not having exclusive access to the audio hardware?
I understand jackd grabs the audio exclusively but why isn't vlc sending its output to ALSA and ALSA forwading it to JACK?
I appreciate your help and suggestions, but it's so enormously frustrating. I've been struggling with this problem for so long now. -
Hi All,
I'm running oss4 but my 8 year old son is becoming interested in softsynths and sequencers (I'm mostly interested in non-sequencer, qtractor and amsynth). The standard qjackctl doesn't produce any sound on my system. I guess that I would have to rebuild jack with oss enabled although the standard qjackctl has an oss option. Is this the old oss or the new oss4? Is it worth it? I read that oss4 has superior sound but I can't verify that because I can't truly do an AB comparison, by the time I switch from alsa to oss4, my beer may have changed from English to German (cited as one of the major causes of listening bias by a major audiophile magazine).
http://www.roger-russell.com/truth/truth.htm
http://www.roger-russell.com/wire/wire.htm
Question, can I get qjackctl, non-sequencer, qtractor and amsynth to play nicely together with oss4? Is it worth it in 2014 or should I just switch back to alsa? Please chime in, all opinions welcome!
Thanks!
Oh yes, I have some fairly high end equipment, nad, hafler and klipsch.
Last edited by Frabato (2014-08-02 04:26:33)sekret wrote:That was one of my main reasons as well. But then I found out that the main issue with alsa is dmix, the software mixer for alsa, which is said to have low quality. Pulseaudio for example replaces dmix and provides better sound quality and more features. The same with oss4, which has higher quality software mixing.
Dmix has maximum 24 bit resolution for mixing. So that is one aspect. Or do you also mean the mixing algorithm? Can you pinpoint to the troubling part?
sekret wrote:But if you want high quality audio playback, you can simply not use dmix. For example I use mpv for videos and moc for audio. In mpv you can do it with mpv --ao=alsa:device=[plughw:UA25]
No offence, but using plughw is futile, since it will resample if necessary*, therefore degrading/modifying original signal. Set the appropriate parameters either through custom pcm devices of type hw or create a new hifihw device that will accept runtime parameters (channels, format, frequency etc.) and change them on between playback. This should be feasible through creative scripting, maybe even without much help outside of alsa. Simply because of courtesy and comfort of changing parameters through device addressing or some anomalies, otherwise using type hw should handle changes on its own.
As for the definition, in case of wanted resampling, a resampler may be specified, either for each pcm definition or through 'defaults.pcm.rate_converter' node. See the wiki and the conversation page, there was a change recently involving speexrate resampler.
sekret wrote:
All you lose is, that those applications run like this will block the audio device for other applications. E.g. if I play music in moc and want to play a video with mpv, I get this output
[ao/alsa] Playback open error: Device or resource busy
[ao] Failed to initialize audio driver 'alsa'
Could not open/initialize audio device -> no sound.
You can avoid those errors by specifying 'nonblock 0' into the type hw pcm definition or 'defaults.pcm.nonblock' node. Depending on the sound source, it might play afterwards.
I would recommend using best possible parameters for the jack server with hw:cardname-reference instead of plughw.
Edit: * Unless it is necessary for your hardware, which could be the case. Again I am not trying to flame. Just make sure you understand what plughw actually does, otherwise you might be unknowingly thinking, you are using the best settings. Check 'less /usr/share/alsa/alsa.conf' for more information and our alsa wiki. At some point I will probably add advanced configuration with custom runtime parameters, so it will be explained how they work.
Edit2: Typos, but there are probably more.
The plug pcm in between pcm.default and pcm.jack could be expanded, since it will also resample "willy nilly", unless specified. The best way is of course to have a signal path without conversions and use jack directly.
Last edited by emeres (2014-08-03 15:59:57) -
[SOLVED] Disabling "dry" signal (ALSA or JACK)
I'm using VirtualBox to run FL Studio in a virtual Windows box. The problem (ok, not so much of a problem, but an annoyance) is, that I can't seem to find a way to disable the "dry" microphone signal. Probably you don't understand what I mean, so I'll give an example: when using a vocoder type effect, you can still hear the original, unaltered voice "under" the übercool Kraftwerk robot voice, and that is just lame. Still trying to make it clearer: what I'm trying to achieve is the microphone not going straight to the speakers. So, are there any Linux audio gurus out there who can help me?
P.S. Yes, it's a Linux problem, not Windows or VirtualBox one, since I can hear the voice even when VB isn't running.
Last edited by kamiheku (2009-05-24 07:54:35)Oh, dang, man... Now I'm embarrassed! That truly was quite obvious, maybe it was too obvious! Thanks. This one is now [SOLVED]
-
Beginners Guide sound installation -- OSS and ALSA objective features
Beginners Guide sound installation -- OSS and ALSA objective features needed
Hi,
In most wiki pages i just edit the things that i think need editing. Just look at the oss history
In all those cases i didn't really saw a need to ask for permission.. that would kinda destroy the wiki idea.
However i want to change the sound instructions in the beginners guide so i made a copy of the entire guide and the part that i changed there is: http://wiki.archlinux.org/index.php/Use … ling_Sound now i have a slight issue there. As you can see i'm in favor of OSS and because you can spot that it's not objective. Now i would ask 2 things.
1. Could you all post your features of OSS and ALSA
2. Once that's done can i have permission to place that section in the beginners guide? that will also include removing the sound installation from the beginners guide which i already did in my version
My personal reason to do this. i've read this and am since then pro OSS and against the ALSA/PulseAudio combo. i think that combo needs to get out of linux (most notably Fedora and Ubuntu) ASAP. and oss needs to go back into the linux kernel
Thank you for your time,
Markngoonee wrote:
I'm an alsa/pulse user, so I'll give a bit of the 'other side'.
Alsa/pulse
Pros:-
network sound
advanced connection of sink/source (including merging sinks)
bluetooth support!
highly supported by existing apps, either through directly supporting Pulse or through its alsa plugin (you should not need to recompile properly-written apps which do not assume they should write audio data directly to hardware, I think I only needed to recompile mpd on my system)
Cons:-
Setup isn't the easiest. Can't comment vs OSS4 because I haven't tried it
OSS4
Pros:-
Everyone seems to say sound quality is better. I guess that's because they're comparing it with dmix alsa. Use pulse with alsa and you should not notice any difference in sound quality though.
Cons:-
Most apps nowadays default output to alsa. Meaning OSS plays them using an alsa plugin.
USB support is admittedly skimpy.
EDIT: Having read the sound article you referred to, my only comment is that the writer really has it in for Pulse... 3 seconds latency, where'd he get that from? I use pulse for audio recording (when I'm lazy to fire up JACK) and while there IS latency, its definitely in the ms range.
Thanx for the input
Gen2ly wrote:
Gen2ly wrote:...As a side note, do you need libflashsupport here???...
markg85, libflashsupport isn't needed. [1]
pacman -Ql oss | grep flash
If you don't know, please don't put in wiki, this could cause unnecessary problems. As for the mms section:
If your stream sounds ugly in totem like it did with me then you could try to play it with another codec like ffmpeg (mplayer). That "fixed" the issue for me. This will not fix the issue that somehow pops up in gstreamer when playing MMS streams but it will give you the option to play it with good sound quality. Playing it in mplayer is simple:
# mplayer mmsh://yourstreamurl
Could you fix this? ffmpeg is not a codec . Also define ugly, and what is somehow?
markg85 wrote:Thanx a lot for your feedback. i will certainly use it when i make more edits.
As for the things you didn't know. As soon as i fully understand how i can get a microphone working in OSS i will add that to the wiki as well. Unless you already know it.. in that case, feel free to add it.
For the mic, I did get mine going. Can't remember just how I did mine (sorry, think I had to disable one of the inputs), but do remember to prevent it from passing through the speakers had to disable "Misc Microphone".
# ossmix
Selected mixer 0/High Definition Audio ALC888
Known controls are:
jack.green.mode <front|rear|center/LFE|side|pcm4|input> (currently front)
jack.green [<leftvol>:<rightvol>] (currently 29.9:29.9 dB)
jack.green.mute ON|OFF (currently OFF)
jack.black.mode <front|rear|center/LFE|side|pcm4|input> (currently center/LFE)
jack.black [<leftvol>:<rightvol>] (currently 29.9:29.9 dB)
jack.black.mute ON|OFF (currently OFF)
jack.orange.mode <front|rear|center/LFE|side|pcm4|input> (currently rear)
jack.orange [<leftvol>:<rightvol>] (currently 29.9:29.9 dB)
jack.orange.mute ON|OFF (currently OFF)
jack.gray.mode <front|rear|center/LFE|side|pcm4|input> (currently pcm4)
jack.gray [<leftvol>:<rightvol>] (currently 29.9:29.9 dB)
jack.gray.mute ON|OFF (currently OFF)
jack.pink.mode <front|rear|center/LFE|side|pcm4|input> (currently input)
jack.pink [<leftvol>:<rightvol>] (currently 19.9:19.9 dB)
jack.pink.mute ON|OFF (currently OFF)
jack.fp-pink.mode <front|rear|center/LFE|side|pcm4|input> (currently front)
jack.fp-pink [<leftvol>:<rightvol>] (currently 29.9:29.9 dB)
jack.fp-pink.mute ON|OFF (currently OFF)
jack.blue.mode <front|rear|center/LFE|side|pcm4|input> (currently input)
jack.blue [<leftvol>:<rightvol>] (currently 29.9:29.9 dB)
jack.blue.mute ON|OFF (currently OFF)
jack.fp-green.mode <front|rear|center/LFE|side|pcm4|input> (currently front)
jack.fp-green [<leftvol>:<rightvol>] (currently 29.9:29.9 dB)
jack.fp-green.mute ON|OFF (currently OFF)
record.mix.mute.mic1 ON|OFF (currently OFF)
record.mix.mute.fp-mic1 ON|OFF (currently OFF)
record.mix.mute.linein1 ON|OFF (currently OFF)
record.mix.mute.fp-headphone1 ON|OFF (currently OFF)
record.mix.mute.green1 ON|OFF (currently OFF)
record.mix.mute.black1 ON|OFF (currently OFF)
record.mix.mute.orange1 ON|OFF (currently OFF)
record.mix.mute.gray1 ON|OFF (currently OFF)
record.mix.mute.input-mix1 ON|OFF (currently OFF)
record.mix1 [<leftvol>:<rightvol>] (currently 38.9:38.9 dB)
record.mix.mute.mic2 ON|OFF (currently OFF)
record.mix.mute.fp-mic2 ON|OFF (currently OFF)
record.mix.mute.linein2 ON|OFF (currently OFF)
record.mix.mute.fp-headphone2 ON|OFF (currently OFF)
record.mix.mute.green2 ON|OFF (currently OFF)
record.mix.mute.black2 ON|OFF (currently OFF)
record.mix.mute.orange2 ON|OFF (currently OFF)
record.mix.mute.gray2 ON|OFF (currently OFF)
record.mix.mute.input-mix2 ON|OFF (currently OFF)
record.mix2 [<leftvol>:<rightvol>] (currently 4.4:2.9 dB)
misc.mic [<leftvol>:<rightvol>] (currently 0.0:0.0 dB)
misc.fp-mic [<leftvol>:<rightvol>] (currently 46.4:37.4 dB)
misc.linein [<leftvol>:<rightvol>] (currently 38.9:38.9 dB)
misc.fp-headphone [<leftvol>:<rightvol>] (currently 34.4:34.4 dB)
misc.green [<leftvol>:<rightvol>] (currently 34.4:38.9 dB)
misc.black [<leftvol>:<rightvol>] (currently 38.9:38.9 dB)
misc.orange [<leftvol>:<rightvol>] (currently 38.9:38.9 dB)
misc.gray [<leftvol>:<rightvol>] (currently 40.4:41.9 dB)
misc.input-mix <mic|fp-mic|linein> (currently mic)
misc.front-mute ON|OFF (currently OFF)
misc.input-mix-mute1 ON|OFF (currently OFF)
misc.front1 [<leftvol>:<rightvol>] (currently 43.4:43.4 dB)
misc.front2 <front|input-mix> (currently front)
misc.rear-mute ON|OFF (currently OFF)
misc.input-mix-mute2 ON|OFF (currently OFF)
misc.rear1 [<leftvol>:<rightvol>] (currently 4.4:4.4 dB)
misc.rear2 <rear|input-mix> (currently rear)
misc.center/lfe-mute ON|OFF (currently OFF)
misc.input-mix-mute3 ON|OFF (currently OFF)
misc.center/lfe1 [<leftvol>:<rightvol>] (currently 41.9:41.9 dB)
misc.center/lfe2 <center/LFE|input-mix> (currently center/LFE)
misc.side-mute ON|OFF (currently OFF)
misc.input-mix-mute4 ON|OFF (currently OFF)
misc.side1 [<leftvol>:<rightvol>] (currently 35.9:35.9 dB)
misc.side2 <side|input-mix> (currently side)
misc.pcm4-mute ON|OFF (currently OFF)
misc.input-mix-mute5 ON|OFF (currently OFF)
misc.pcm41 [<leftvol>:<rightvol>] (currently 25.4:25.4 dB)
misc.pcm42 <pcm4|input-mix> (currently pcm4)
vmix0-enable ON|OFF (currently ON)
vmix0-rate <decimal value> (currently 48000) (Read-only)
vmix0-channels <Stereo|Multich> (currently Stereo)
vmix0-src <Fast|Low|Medium|High|High+|Production|OFF> (currently Medium)
vmix0-outvol <monovol> (currently 25.0 dB)
vmix0-invol <monovol> (currently 25.0 dB)
vmix0.pcm8 [<leftvol>:<rightvol>] (currently 19.9:19.9 dB) ("knotify4")
vmix0.pcm9 [<leftvol>:<rightvol>] (currently 25.0:25.0 dB)
vmix0.pcm10 [<leftvol>:<rightvol>] (currently 25.0:25.0 dB)
vmix0.pcm11 [<leftvol>:<rightvol>] (currently 25.0:25.0 dB)
For libflashsupport on the same page you linked it clearly states:
# Flash V9 and V10 require libflashsupport to output sound via OSS. Typically a 32-bit version of the library is required.
# Flash V10 has a 64-bit version which requires a 64 bit libflashsupport.
Also i tested it with and without libflashsupport. On archlinux (x64 running here) there most certainly is a need for libflashsupport when you want to have sound in your flash. And yes i tested the archlinux OSS version and the mercurial version (running now) bith need it installed manually! On my pc sound in flash didn't work without it but did with it. So, no not removing from the wiki as it's needed. But i see you removed it for me! please do NOT do that if you didn't even verified it. I use flash 10 x64 and i need it!
As for the ffmpeg "codec" changed it to backend.
And i did get the microphone working near perfect: http://www.4front-tech.com/forum/viewtopic.php?p=13192
Now for some news you all might like.
On my school i need to do an investigation to whatever i want and i'm heavily thinking about investigating the pros/cons of alsa compared to oss (or oss compared to alsa). That investigation will take from monday next week till next mondey till friday 23 of oktober. In that investigation i'm going to do some in depth look of alsa and oss and that will include the usability as well.
Following up on that investigation i will spend another 8 weeks on my school making a volume control application that can be used with alsa and oss and easily expandable with other sound systems. The goal of this is to make one sound application that can manage (in the first place) alsa and oss. oss is going to be implemented and alsa is probably going to be dummy implemented because it's likely way to much for me to implement both.
Before you get to exited, both projects (investigating and making the application) are just made up today and i just don't know if both will get accepted by my school. I asked one teacher and he liked the idea a lot and could potentially have a value for the sound management under linux. Once i do get this started i will involve the community (YOU!) with this since this project can't be done without the community specially the investigation.
And once i start and have something to tell/ask i will blog about it on http://blog.mageprojects.com
edit::
And this idea already got dumped. read more a few posts down or click: http://bbs.archlinux.org/viewtopic.php? … 34#p612634
Last edited by markg85 (2009-09-03 17:51:45) -
Different program's sound sent to different jack ports with Pulseaudio
Hello,
I just installed Pulseaudio so I could send the output of different programs to different jacks on my computer, both the jacks on my computer are provided by the same motherboard sound card, one on the front for my headphone and one on the back. So far I haven't been able to see them in pavucontrol I only see a single output.
Here's the output of "pacmd list"
Welcome to PulseAudio! Use "help" for usage information.
>>> Memory blocks currently allocated: 530, size: 1.2 MiB.
Memory blocks allocated during the whole lifetime: 4140720, size: 2.1 GiB.
Memory blocks imported from other processes: 125, size: 550.6 KiB.
Memory blocks exported to other processes: 1, size: 512 B.
Total sample cache size: 0 B.
Default sample spec: s16le 2ch 44100Hz
Default channel map: front-left,front-right
Default sink name: alsa_output.pci-0000_00_1b.0.analog-stereo
Default source name: alsa_input.pci-0000_00_1b.0.analog-stereo
Memory blocks of type POOL: 404 allocated/2532098 accumulated.
Memory blocks of type POOL_EXTERNAL: 1 allocated/651424 accumulated.
Memory blocks of type APPENDED: 0 allocated/0 accumulated.
Memory blocks of type USER: 0 allocated/0 accumulated.
Memory blocks of type FIXED: 0 allocated/1302521 accumulated.
Memory blocks of type IMPORTED: 125 allocated/306101 accumulated.
27 module(s) loaded.
index: 0
name: <module-device-restore>
argument: <>
used: -1
load once: yes
properties:
module.author = "Lennart Poettering"
module.description = "Automatically restore the volume/mute state of devices"
module.version = "2.1"
index: 1
name: <module-stream-restore>
argument: <>
used: -1
load once: yes
properties:
module.author = "Lennart Poettering"
module.description = "Automatically restore the volume/mute/device state of streams"
module.version = "2.1"
index: 2
name: <module-card-restore>
argument: <>
used: -1
load once: yes
properties:
module.author = "Lennart Poettering"
module.description = "Automatically restore profile of cards"
module.version = "2.1"
index: 3
name: <module-augment-properties>
argument: <>
used: -1
load once: yes
properties:
module.author = "Lennart Poettering"
module.description = "Augment the property sets of streams with additional static information"
module.version = "2.1"
index: 4
name: <module-alsa-card>
argument: <device_id="0" name="pci-0000_00_1b.0" card_name="alsa_card.pci-0000_00_1b.0" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes card_properties="module-udev-detect.discovered=1">
used: 6
load once: no
properties:
module.author = "Lennart Poettering"
module.description = "ALSA Card"
module.version = "2.1"
index: 5
name: <module-udev-detect>
argument: <>
used: -1
load once: yes
properties:
module.author = "Lennart Poettering"
module.description = "Detect available audio hardware and load matching drivers"
module.version = "2.1"
index: 6
name: <module-jackdbus-detect>
argument: <>
used: -1
load once: yes
properties:
module.author = "David Henningsson"
module.description = "Adds JACK sink/source ports when JACK is started"
module.version = "2.1"
index: 7
name: <module-bluetooth-discover>
argument: <>
used: -1
load once: yes
properties:
module.author = "Joao Paulo Rechi Vita"
module.description = "Detect available bluetooth audio devices and load bluetooth audio drivers"
module.version = "2.1"
index: 8
name: <module-esound-protocol-unix>
argument: <>
used: -1
load once: no
properties:
module.author = "Lennart Poettering"
module.description = "ESOUND protocol (UNIX sockets)"
module.version = "2.1"
index: 9
name: <module-native-protocol-unix>
argument: <>
used: -1
load once: no
properties:
module.author = "Lennart Poettering"
module.description = "Native protocol (UNIX sockets)"
module.version = "2.1"
index: 10
name: <module-gconf>
argument: <>
used: -1
load once: yes
properties:
module.author = "Lennart Poettering"
module.description = "GConf Adapter"
module.version = "2.1"
index: 11
name: <module-default-device-restore>
argument: <>
used: -1
load once: yes
properties:
module.author = "Lennart Poettering"
module.description = "Automatically restore the default sink and source"
module.version = "2.1"
index: 12
name: <module-rescue-streams>
argument: <>
used: -1
load once: yes
properties:
module.author = "Lennart Poettering"
module.description = "When a sink/source is removed, try to move their streams to the default sink/source"
module.version = "2.1"
index: 13
name: <module-always-sink>
argument: <>
used: -1
load once: yes
properties:
module.author = "Colin Guthrie"
module.description = "Always keeps at least one sink loaded even if it's a null one"
module.version = "2.1"
index: 14
name: <module-intended-roles>
argument: <>
used: -1
load once: yes
properties:
module.author = "Lennart Poettering"
module.description = "Automatically set device of streams based of intended roles of devices"
module.version = "2.1"
index: 15
name: <module-suspend-on-idle>
argument: <>
used: -1
load once: yes
properties:
module.author = "Lennart Poettering"
module.description = "When a sink/source is idle for too long, suspend it"
module.version = "2.1"
index: 16
name: <module-console-kit>
argument: <>
used: -1
load once: yes
properties:
module.author = "Lennart Poettering"
module.description = "Create a client for each ConsoleKit session of this user"
module.version = "2.1"
index: 17
name: <module-systemd-login>
argument: <>
used: -1
load once: yes
properties:
module.author = "Lennart Poettering"
module.description = "Create a client for each login session of this user"
module.version = "2.1"
index: 18
name: <module-position-event-sounds>
argument: <>
used: -1
load once: yes
properties:
module.author = "Lennart Poettering"
module.description = "Position event sounds between L and R depending on the position on screen of the widget triggering them."
module.version = "2.1"
index: 19
name: <module-role-cork>
argument: <>
used: -1
load once: yes
properties:
module.author = "Lennart Poettering"
module.description = "Mute & cork streams with certain roles while others exist"
module.version = "2.1"
index: 20
name: <module-filter-heuristics>
argument: <>
used: -1
load once: yes
properties:
module.author = "Colin Guthrie"
module.description = "Detect when various filters are desirable"
module.version = "2.1"
index: 21
name: <module-filter-apply>
argument: <>
used: -1
load once: yes
properties:
module.author = "Colin Guthrie"
module.description = "Load filter sinks automatically when needed"
module.version = "2.1"
index: 22
name: <module-dbus-protocol>
argument: <>
used: -1
load once: yes
properties:
module.author = "Tanu Kaskinen"
module.description = "D-Bus interface"
module.version = "2.1"
index: 23
name: <module-switch-on-port-available>
argument: <>
used: -1
load once: no
properties:
index: 24
name: <module-x11-publish>
argument: <display=:0.0>
used: -1
load once: no
properties:
module.author = "Lennart Poettering"
module.description = "X11 credential publisher"
module.version = "2.1"
index: 25
name: <module-x11-xsmp>
argument: <display=:0.0 session_manager=local/archpc:@/tmp/.ICE-unix/15374,unix/archpc:/tmp/.ICE-unix/15374>
used: -1
load once: no
properties:
module.author = "Lennart Poettering"
module.description = "X11 session management"
module.version = "2.1"
index: 26
name: <module-cli-protocol-unix>
argument: <>
used: -1
load once: no
properties:
module.author = "Lennart Poettering"
module.description = "Command line interface protocol (UNIX sockets)"
module.version = "2.1"
1 sink(s) available.
index: 7
name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
driver: <module-alsa-card.c>
flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
state: RUNNING
suspend cause:
priority: 9959
volume: 0: 100% 1: 100%
0: 0.00 dB 1: 0.00 dB
balance 0.00
base volume: 100%
0.00 dB
volume steps: 65537
muted: yes
current latency: 19.31 ms
max request: 3 KiB
max rewind: 344 KiB
monitor source: 8
sample spec: s16le 2ch 44100Hz
channel map: front-left,front-right
Stereo
used by: 1
linked by: 3
configured latency: 20.00 ms; range is 0.50 .. 2000.00 ms
card: 0 <alsa_card.pci-0000_00_1b.0>
module: 4
properties:
alsa.resolution_bits = "16"
device.api = "alsa"
device.class = "sound"
alsa.class = "generic"
alsa.subclass = "generic-mix"
alsa.name = "ALC888 Analog"
alsa.id = "ALC888 Analog"
alsa.subdevice = "0"
alsa.subdevice_name = "subdevice #0"
alsa.device = "0"
alsa.card = "0"
alsa.card_name = "HDA Intel"
alsa.long_card_name = "HDA Intel at 0xfbff8000 irq 45"
alsa.driver_name = "snd_hda_intel"
device.bus_path = "pci-0000:00:1b.0"
sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0"
device.bus = "pci"
device.vendor.id = "8086"
device.vendor.name = "Intel Corporation"
device.product.name = "5 Series/3400 Series Chipset High Definition Audio"
device.form_factor = "internal"
device.string = "front:0"
device.buffering.buffer_size = "352800"
device.buffering.fragment_size = "176400"
device.access_mode = "mmap+timer"
device.profile.name = "analog-stereo"
device.profile.description = "Analog Stereo"
device.description = "Built-in Audio Analog Stereo"
alsa.mixer_name = "Realtek ALC888"
alsa.components = "HDA:10ec0888,1458a002,00100001"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-pci"
ports:
analog-output: Analog Output (priority 9900, available: unknown)
properties:
analog-output-headphones: Headphones (priority 9000, available: yes)
properties:
active port: <analog-output-headphones>
2 source(s) available.
index: 8
name: <alsa_output.pci-0000_00_1b.0.analog-stereo.monitor>
driver: <module-alsa-card.c>
flags: DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
state: RUNNING
suspend cause:
priority: 1950
volume: 0: 100% 1: 100%
0: 0.00 dB 1: 0.00 dB
balance 0.00
base volume: 100%
0.00 dB
volume steps: 65537
muted: no
current latency: 0.00 ms
max rewind: 344 KiB
sample spec: s16le 2ch 44100Hz
channel map: front-left,front-right
Stereo
used by: 2
linked by: 2
configured latency: 20.00 ms; range is 0.50 .. 2000.00 ms
monitor_of: 7
card: 0 <alsa_card.pci-0000_00_1b.0>
module: 4
properties:
device.description = "Monitor of Built-in Audio Analog Stereo"
device.class = "monitor"
alsa.card = "0"
alsa.card_name = "HDA Intel"
alsa.long_card_name = "HDA Intel at 0xfbff8000 irq 45"
alsa.driver_name = "snd_hda_intel"
device.bus_path = "pci-0000:00:1b.0"
sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0"
device.bus = "pci"
device.vendor.id = "8086"
device.vendor.name = "Intel Corporation"
device.product.name = "5 Series/3400 Series Chipset High Definition Audio"
device.form_factor = "internal"
device.string = "0"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-pci"
index: 10
name: <alsa_input.pci-0000_00_1b.0.analog-stereo>
driver: <module-alsa-card.c>
flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
state: RUNNING
suspend cause:
priority: 9959
volume: 0: 10% 1: 10%
0: -60.00 dB 1: -60.00 dB
balance 0.00
base volume: 10%
-60.00 dB
volume steps: 65537
muted: no
current latency: 0.09 ms
max rewind: 0 KiB
sample spec: s16le 2ch 44100Hz
channel map: front-left,front-right
Stereo
used by: 1
linked by: 1
configured latency: 20.00 ms; range is 1.00 .. 2000.00 ms
card: 0 <alsa_card.pci-0000_00_1b.0>
module: 4
properties:
alsa.resolution_bits = "16"
device.api = "alsa"
device.class = "sound"
alsa.class = "generic"
alsa.subclass = "generic-mix"
alsa.name = "ALC888 Analog"
alsa.id = "ALC888 Analog"
alsa.subdevice = "0"
alsa.subdevice_name = "subdevice #0"
alsa.device = "0"
alsa.card = "0"
alsa.card_name = "HDA Intel"
alsa.long_card_name = "HDA Intel at 0xfbff8000 irq 45"
alsa.driver_name = "snd_hda_intel"
device.bus_path = "pci-0000:00:1b.0"
sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0"
device.bus = "pci"
device.vendor.id = "8086"
device.vendor.name = "Intel Corporation"
device.product.name = "5 Series/3400 Series Chipset High Definition Audio"
device.form_factor = "internal"
device.string = "front:0"
device.buffering.buffer_size = "352800"
device.buffering.fragment_size = "176400"
device.access_mode = "mmap+timer"
device.profile.name = "analog-stereo"
device.profile.description = "Analog Stereo"
device.description = "Built-in Audio Analog Stereo"
alsa.mixer_name = "Realtek ALC888"
alsa.components = "HDA:10ec0888,1458a002,00100001"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-pci"
ports:
analog-input-microphone-front: Front Microphone (priority 8500, available: no)
properties:
analog-input-microphone-rear: Rear Microphone (priority 8200, available: no)
properties:
analog-input-linein: Line In (priority 8100, available: no)
properties:
active port: <analog-input-microphone-front>
9 client(s) logged in.
index: 0
driver: <module-systemd-login.c>
owner module: 17
properties:
application.name = "Login Session 2"
systemd-login.session = "2"
index: 3
driver: <module-x11-xsmp.c>
owner module: 25
properties:
application.name = "XSMP Session on xfce4-session as 276d8dfaf-2e31-425c-a448-6e95e23c73a0"
xsmp.vendor = "xfce4-session"
xsmp.client.id = "276d8dfaf-2e31-425c-a448-6e95e23c73a0"
index: 13
driver: <protocol-native.c>
owner module: 9
properties:
application.name = "wrapper"
native-protocol.peer = "UNIX socket client"
native-protocol.version = "26"
application.process.id = "15869"
application.process.user = "jordan"
application.process.host = "archpc"
application.process.binary = "wrapper"
window.x11.display = ":0.0"
application.language = "en_US.UTF-8"
application.process.machine_id = "3cb559c10be41c9a438615f400000947"
index: 14
driver: <protocol-native.c>
owner module: 9
properties:
application.name = "wrapper"
native-protocol.peer = "UNIX socket client"
native-protocol.version = "26"
application.process.id = "15869"
application.process.user = "jordan"
application.process.host = "archpc"
application.process.binary = "wrapper"
window.x11.display = ":0.0"
application.language = "en_US.UTF-8"
application.process.machine_id = "3cb559c10be41c9a438615f400000947"
index: 15
driver: <protocol-native.c>
owner module: 9
properties:
application.name = "wrapper"
native-protocol.peer = "UNIX socket client"
native-protocol.version = "26"
application.process.id = "15869"
application.process.user = "jordan"
application.process.host = "archpc"
application.process.binary = "wrapper"
window.x11.display = ":0.0"
application.language = "en_US.UTF-8"
application.process.machine_id = "3cb559c10be41c9a438615f400000947"
index: 32
driver: <protocol-native.c>
owner module: 9
properties:
application.name = "PulseAudio Volume Control"
native-protocol.peer = "UNIX socket client"
native-protocol.version = "26"
application.id = "org.PulseAudio.pavucontrol"
application.icon_name = "audio-card"
application.version = "1.0"
application.process.id = "20106"
application.process.user = "jordan"
application.process.host = "archpc"
application.process.binary = "pavucontrol"
application.language = "en_US.UTF-8"
window.x11.display = ":0.0"
application.process.machine_id = "3cb559c10be41c9a438615f400000947"
index: 40
driver: <protocol-native.c>
owner module: 9
properties:
application.name = "PulseAudio Volume Control"
native-protocol.peer = "UNIX socket client"
native-protocol.version = "26"
window.x11.display = ":0.0"
window.x11.screen = "0"
application.process.id = "20106"
application.process.user = "jordan"
application.process.host = "archpc"
application.process.binary = "pavucontrol"
application.language = "en_US.UTF-8"
application.process.machine_id = "3cb559c10be41c9a438615f400000947"
application.icon_name = "multimedia-volume-control"
index: 57
driver: <protocol-native.c>
owner module: 9
properties:
application.name = "Clementine"
native-protocol.peer = "UNIX socket client"
native-protocol.version = "26"
application.process.id = "15910"
application.process.user = "jordan"
application.process.host = "archpc"
application.process.binary = "clementine"
application.language = "en_US.UTF-8"
window.x11.display = ":0.0"
application.process.machine_id = "3cb559c10be41c9a438615f400000947"
application.icon_name = "application-x-clementine"
index: 58
driver: <cli.c>
owner module: 26
properties:
application.name = "UNIX socket client"
1 card(s) available.
index: 0
name: <alsa_card.pci-0000_00_1b.0>
driver: <module-alsa-card.c>
owner module: 4
properties:
alsa.card = "0"
alsa.card_name = "HDA Intel"
alsa.long_card_name = "HDA Intel at 0xfbff8000 irq 45"
alsa.driver_name = "snd_hda_intel"
device.bus_path = "pci-0000:00:1b.0"
sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0"
device.bus = "pci"
device.vendor.id = "8086"
device.vendor.name = "Intel Corporation"
device.product.name = "5 Series/3400 Series Chipset High Definition Audio"
device.form_factor = "internal"
device.string = "0"
device.description = "Built-in Audio"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-pci"
profiles:
output:analog-stereo: Analog Stereo Output (priority 6000)
output:analog-stereo+input:analog-stereo: Analog Stereo Duplex (priority 6060)
output:analog-stereo+input:iec958-stereo: Analog Stereo Output + Digital Stereo (IEC958) Input (priority 6055)
output:analog-surround-40: Analog Surround 4.0 Output (priority 700)
output:analog-surround-40+input:analog-stereo: Analog Surround 4.0 Output + Analog Stereo Input (priority 760)
output:analog-surround-40+input:iec958-stereo: Analog Surround 4.0 Output + Digital Stereo (IEC958) Input (priority 755)
output:analog-surround-41: Analog Surround 4.1 Output (priority 800)
output:analog-surround-41+input:analog-stereo: Analog Surround 4.1 Output + Analog Stereo Input (priority 860)
output:analog-surround-41+input:iec958-stereo: Analog Surround 4.1 Output + Digital Stereo (IEC958) Input (priority 855)
output:analog-surround-50: Analog Surround 5.0 Output (priority 700)
output:analog-surround-50+input:analog-stereo: Analog Surround 5.0 Output + Analog Stereo Input (priority 760)
output:analog-surround-50+input:iec958-stereo: Analog Surround 5.0 Output + Digital Stereo (IEC958) Input (priority 755)
output:analog-surround-51: Analog Surround 5.1 Output (priority 800)
output:analog-surround-51+input:analog-stereo: Analog Surround 5.1 Output + Analog Stereo Input (priority 860)
output:analog-surround-51+input:iec958-stereo: Analog Surround 5.1 Output + Digital Stereo (IEC958) Input (priority 855)
output:analog-surround-71: Analog Surround 7.1 Output (priority 700)
output:analog-surround-71+input:analog-stereo: Analog Surround 7.1 Output + Analog Stereo Input (priority 760)
output:analog-surround-71+input:iec958-stereo: Analog Surround 7.1 Output + Digital Stereo (IEC958) Input (priority 755)
output:iec958-stereo: Digital Stereo (IEC958) Output (priority 5500)
output:iec958-stereo+input:analog-stereo: Digital Stereo (IEC958) Output + Analog Stereo Input (priority 5560)
output:iec958-stereo+input:iec958-stereo: Digital Stereo Duplex (IEC958) (priority 5555)
input:analog-stereo: Analog Stereo Input (priority 60)
input:iec958-stereo: Digital Stereo (IEC958) Input (priority 55)
off: Off (priority 0)
active profile: <output:analog-stereo+input:analog-stereo>
sinks:
alsa_output.pci-0000_00_1b.0.analog-stereo/#7: Built-in Audio Analog Stereo
sources:
alsa_output.pci-0000_00_1b.0.analog-stereo.monitor/#8: Monitor of Built-in Audio Analog Stereo
alsa_input.pci-0000_00_1b.0.analog-stereo/#10: Built-in Audio Analog Stereo
ports:
analog-output: Analog Output (priority 9900, available: unknown)
properties:
analog-output-headphones: Headphones (priority 9000, available: yes)
properties:
analog-input-microphone-front: Front Microphone (priority 8500, available: no)
properties:
analog-input-microphone-rear: Rear Microphone (priority 8200, available: no)
properties:
analog-input-linein: Line In (priority 8100, available: no)
properties:
iec958-stereo-input: iec958-stereo-input (priority 0, available: unknown)
properties:
iec958-stereo-output: Digital Output (S/PDIF) (priority 0, available: unknown)
properties:
1 sink input(s) available.
index: 90
driver: <protocol-native.c>
flags: START_CORKED
state: RUNNING
sink: 7 <alsa_output.pci-0000_00_1b.0.analog-stereo>
volume: 0: 100% 1: 100%
0: 0.00 dB 1: 0.00 dB
balance 0.00
muted: no
current latency: 89.86 ms
requested latency: 90.00 ms
sample spec: float32le 2ch 44100Hz
channel map: front-left,front-right
Stereo
resample method: copy
module: 9
client: 57 <Clementine>
properties:
media.name = "'Summertime' by 'Sex Bob-omb'"
application.name = "Clementine"
native-protocol.peer = "UNIX socket client"
native-protocol.version = "26"
application.process.id = "15910"
application.process.user = "jordan"
application.process.host = "archpc"
application.process.binary = "clementine"
application.language = "en_US.UTF-8"
window.x11.display = ":0.0"
application.process.machine_id = "3cb559c10be41c9a438615f400000947"
application.icon_name = "application-x-clementine"
module-stream-restore.id = "sink-input-by-application-name:Clementine"
media.title = "Summertime"
media.artist = "Sex Bob-omb"
3 source outputs(s) available.
index: 48
driver: <protocol-native.c>
flags: DONT_MOVE
state: RUNNING
source: 8 <alsa_output.pci-0000_00_1b.0.analog-stereo.monitor>
volume: 0: 100%
0: 0.00 dB
balance 0.00
muted: no
current latency: 9.68 ms
requested latency: 20.00 ms
sample spec: float32le 1ch 25Hz
channel map: mono
Mono
resample method: peaks
owner module: 9
client: 32 <PulseAudio Volume Control>
properties:
media.name = "Peak detect"
application.name = "PulseAudio Volume Control"
native-protocol.peer = "UNIX socket client"
native-protocol.version = "26"
application.id = "org.PulseAudio.pavucontrol"
application.icon_name = "audio-card"
application.version = "1.0"
application.process.id = "20106"
application.process.user = "jordan"
application.process.host = "archpc"
application.process.binary = "pavucontrol"
application.language = "en_US.UTF-8"
window.x11.display = ":0.0"
application.process.machine_id = "3cb559c10be41c9a438615f400000947"
module-stream-restore.id = "source-output-by-application-id:org.PulseAudio.pavucontrol"
index: 51
driver: <protocol-native.c>
flags: DONT_MOVE
state: RUNNING
source: 10 <alsa_input.pci-0000_00_1b.0.analog-stereo>
volume: 0: 10%
0: -60.00 dB
balance 0.00
muted: no
current latency: 0.00 ms
requested latency: 20.00 ms
sample spec: float32le 1ch 25Hz
channel map: mono
Mono
resample method: peaks
owner module: 9
client: 32 <PulseAudio Volume Control>
properties:
media.name = "Peak detect"
application.name = "PulseAudio Volume Control"
native-protocol.peer = "UNIX socket client"
native-protocol.version = "26"
application.id = "org.PulseAudio.pavucontrol"
application.icon_name = "audio-card"
application.version = "1.0"
application.process.id = "20106"
application.process.user = "jordan"
application.process.host = "archpc"
application.process.binary = "pavucontrol"
application.language = "en_US.UTF-8"
window.x11.display = ":0.0"
application.process.machine_id = "3cb559c10be41c9a438615f400000947"
module-stream-restore.id = "source-output-by-application-id:org.PulseAudio.pavucontrol"
index: 58
driver: <protocol-native.c>
flags: DONT_MOVE
state: RUNNING
source: 8 <alsa_output.pci-0000_00_1b.0.analog-stereo.monitor>
volume: 0: 100%
0: 0.00 dB
balance 0.00
muted: no
current latency: 9.71 ms
requested latency: 20.00 ms
sample spec: float32le 1ch 25Hz
channel map: mono
Mono
resample method: peaks
owner module: 9
client: 32 <PulseAudio Volume Control>
direct on input: 90
properties:
media.name = "Peak detect"
application.name = "PulseAudio Volume Control"
native-protocol.peer = "UNIX socket client"
native-protocol.version = "26"
application.id = "org.PulseAudio.pavucontrol"
application.icon_name = "audio-card"
application.version = "1.0"
application.process.id = "20106"
application.process.user = "jordan"
application.process.host = "archpc"
application.process.binary = "pavucontrol"
application.language = "en_US.UTF-8"
window.x11.display = ":0.0"
application.process.machine_id = "3cb559c10be41c9a438615f400000947"
module-stream-restore.id = "source-output-by-application-id:org.PulseAudio.pavucontrol"
0 cache entrie(s) available.
Thanks!Did you ever figure this out? I am in the same situation and cannot figure out how to refer to specific ports on the same motherboard.
https://wiki.archlinux.org/index.php/Pu … ont.2Frear looks promising. However I forget whether one of my cords is plugged in to the rear jack or the center jack, and moving my desktop to look is a little involved. Oh well, next time the need arises I will experiment further.
Last edited by rosshadden (2014-03-06 03:22:47) -
[SOLVED]Jack doesn't work anymore
Good day people,
It seems that Jack stopped working all of a sudden. Well, it's not really all of a sudden, it hasn't been working for 3 months or so, but before that it worked just fine and I did not change any setting since then. I didn't post before because I don't really use JACK too often but now it seems like I'll be going to need it.
The package I'm using is jack2 from the community repository, not jack2-dbus or jack or any other package, although I've tried them with the same results.
First, let's try to start jack the old fashioned way
$jack_control start
it results in
DBus exception: org.jackaudio.Error.Generic: Failed to open server
...ook. Let's try an even older fashion way.
$jackd -d alsa
The output is:
jackdmp 1.9.8
Copyright 2001-2005 Paul Davis and others.
Copyright 2004-2011 Grame.
jackdmp comes with ABSOLUTELY NO WARRANTY
This is free software, and you are welcome to redistribute it
under certain conditions; see the file COPYING for details
could not open component .so '/usr/lib/jack/jack_firewire.so': libffado.so.2: cannot open shared object file: No such file or directory
could not open driver .so '/usr/lib/jack/jack_firewire.so': libffado.so.2: cannot open shared object file: No such file or directory
jack_get_descriptor returns null for 'jack_firewire.so'
could not open component .so '/usr/lib/jack/jack_firewire.so': libffado.so.2: cannot open shared object file: No such file or directory
JACK server starting in realtime mode with priority 10
Jack: Create non RT thread
Jack: ThreadHandler: start
Jack: JackDriver::Open capture_driver_name = hw:0
Jack: JackDriver::Open playback_driver_name = hw:0
Jack: Check protocol client = 8 server = 8
Jack: JackEngine::ClientInternalOpen: name = system
Jack: JackEngine::AllocateRefNum ref = 0
Jack: JackPosixSemaphore::Allocate name = jack_sem.1000_default_system val = 0
Jack: JackEngine::NotifyAddClient: name = system
Jack: JackGraphManager::SetBufferSize size = 1024
Jack: JackConnectionManager::DirectConnect first: ref1 = 0 ref2 = 0
Jack: JackGraphManager::ConnectRefNum cur_index = 0 ref1 = 0 ref2 = 0
Jack: JackDriver::SetupDriverSync driver sem in flush mode
control device hw:0
control device hw:0
audio_reservation_init
Failed to acquire device name : Audio0 error : Method "RequestRelease" with signature "i" on interface "org.freedesktop.ReserveDevice1" doesn't exist
Audio device hw:0 cannot be acquired...
Jack: ~JackDriver
Cannot initialize driver
Jack: no message buffer overruns
Jack: JackPosixThread::Stop
Jack: ThreadHandler: exit
JackServer::Open() failed with -1
Jack: Succeeded in unlocking 82245916 byte memory area
Jack: JackShmMem::delete size = 0 index = 0
Jack: ~JackDriver
Jack: Succeeded in unlocking 1012 byte memory area
Jack: JackShmMem::delete size = 0 index = 1
Jack: cleaning up shared memory
Jack: cleaning up files
Jack: unregistering server `default'
Failed to open server
No luck this time, either. And before you ask, I tried giving jackd other parameters but it didn't change anything.
I now try Qjackctl. As soon as I open it, it says:
D-BUS: JACK server could not be started
Sorry
This is the error log:
11:31:23.797 Patchbay deactivated.
11:31:23.800 Statistics reset.
11:31:23.802 ALSA connection change.
11:31:23.819 D-BUS: Service is available (org.jackaudio.service aka jackdbus).
11:31:23.994 D-BUS: JACK server could not be started. Sorry
Cannot connect to server socket err = No such file or directory
Cannot connect to server socket
jack server is not running or cannot be started
Cannot connect to server socket err = No such file or directory
Cannot connect to server socket
jack server is not running or cannot be started
11:31:24.016 ALSA connection graph change.
Thu Aug 23 11:31:23 2012: Starting jack server...
Thu Aug 23 11:31:23 2012: JACK server starting in realtime mode with priority 10
Thu Aug 23 11:31:23 2012: control device hw:0
Thu Aug 23 11:31:23 2012: control device hw:0
Thu Aug 23 11:31:23 2012: Acquired audio card Audio0
Thu Aug 23 11:31:23 2012: creating alsa driver ... hw:0|hw:0|1024|2|48000|0|0|nomon|swmeter|-|32bit
Thu Aug 23 11:31:23 2012: control device hw:0
Thu Aug 23 11:31:23 2012: configuring for 48000Hz, period = 1024 frames (21.3 ms), buffer = 2 periods
Thu Aug 23 11:31:23 2012: ALSA: final selected sample format for capture: 32bit integer little-endian
Thu Aug 23 11:31:23 2012: ALSA: use 2 periods for capture
Thu Aug 23 11:31:23 2012: ALSA: final selected sample format for playback: 32bit integer little-endian
Thu Aug 23 11:31:23 2012: ALSA: use 2 periods for playback
Thu Aug 23 11:31:23 2012: [1m[31mERROR: ALSA: cannot set hardware parameters for playback[0m
Thu Aug 23 11:31:23 2012: [1m[31mERROR: ALSA: cannot configure playback channel[0m
Thu Aug 23 11:31:23 2012: [1m[31mERROR: JackTemporaryException : now quits...[0m
Thu Aug 23 11:31:23 2012: [1m[31mERROR: Cannot initialize driver[0m
Thu Aug 23 11:31:23 2012: [1m[31mERROR: JackServer::Open() failed with -1[0m
Thu Aug 23 11:31:23 2012: [1m[31mERROR: Failed to open server[0m
11:32:54.690 Could not connect to JACK server as client. - Overall operation failed. - Unable to connect to server. Please check the messages window for more info.
Cannot connect to server socket err = No such file or directory
Cannot connect to server socket
jack server is not running or cannot be started
Ok, let's now uncheck "Enable D-Bus interface" and try again, maybe I'll have more luck. No, it seems I don't.
This is the message log:
11:34:17.340 Patchbay deactivated.
11:34:17.345 Statistics reset.
11:34:17.347 ALSA connection change.
11:34:17.356 JACK is starting...
11:34:17.356 /usr/bin/jackd -dalsa -dhw:0 -r48000 -p1024 -n2
Cannot connect to server socket err = No such file or directory
Cannot connect to server socket
jack server is not running or cannot be started
Cannot connect to server socket err = No such file or directory
Cannot connect to server socket
jack server is not running or cannot be started
11:34:17.361 ALSA connection graph change.
could not open component .so '/usr/lib/jack/jack_firewire.so': libffado.so.2: cannot open shared object file: No such file or directory
could not open driver .so '/usr/lib/jack/jack_firewire.so': libffado.so.2: cannot open shared object file: No such file or directory
jack_get_descriptor returns null for 'jack_firewire.so'
could not open component .so '/usr/lib/jack/jack_firewire.so': libffado.so.2: cannot open shared object file: No such file or directory
jackdmp 1.9.8
Copyright 2001-2005 Paul Davis and others.
Copyright 2004-2011 Grame.
jackdmp comes with ABSOLUTELY NO WARRANTY
This is free software, and you are welcome to redistribute it
under certain conditions; see the file COPYING for details
JACK server starting in realtime mode with priority 10
11:34:17.396 JACK was started with PID=2499.
Jack: Create non RT thread
Jack: ThreadHandler: start
Jack: playback device hw:0
Jack: capture device hw:0
Jack: apparent rate = 48000
Jack: frames per period = 1024
Jack: JackDriver::Open capture_driver_name = hw:0
Jack: JackDriver::Open playback_driver_name = hw:0
Jack: Check protocol client = 8 server = 8
Jack: JackEngine::ClientInternalOpen: name = system
Jack: JackEngine::AllocateRefNum ref = 0
Jack: JackPosixSemaphore::Allocate name = jack_sem.1000_default_system val = 0
Jack: JackEngine::NotifyAddClient: name = system
Jack: JackGraphManager::SetBufferSize size = 1024
Jack: JackConnectionManager::DirectConnect first: ref1 = 0 ref2 = 0
Jack: JackGraphManager::ConnectRefNum cur_index = 0 ref1 = 0 ref2 = 0
Jack: JackDriver::SetupDriverSync driver sem in flush mode
control device hw:0
control device hw:0
audio_reservation_init
Failed to acquire device name : Audio0 error : Device or resource busy
Audio device hw:0 cannot be acquired...
Jack: ~JackDriver
Cannot initialize driver
Jack: no message buffer overruns
Jack: JackPosixThread::Stop
Jack: ThreadHandler: exit
JackServer::Open() failed with -1
Jack: Succeeded in unlocking 82245916 byte memory area
Jack: JackShmMem::delete size = 0 index = 0
Jack: ~JackDriver
Jack: Succeeded in unlocking 1012 byte memory area
Jack: JackShmMem::delete size = 0 index = 1
Jack: cleaning up shared memory
Jack: cleaning up files
Jack: unregistering server `default'
Failed to open server
11:34:17.533 JACK was stopped with exit status=255.
11:34:19.562 Could not connect to JACK server as client. - Overall operation failed. - Unable to connect to server. Please check the messages window for more info.
Cannot connect to server socket err = No such file or directory
Cannot connect to server socket
jack server is not running or cannot be started
I've tried deleting all jack config files, changing settings and before you ask, I've tried with and without pulseaudio (which got along with Jack just fine before everything broke). It seems to be a d-bus problem at first but it gives me issues with d-bus disabled as well. I've read A LOT of forum posts and mailing lists before posting here and I can tell you that none of the solutions that worked for others worked for me, and many just gave up.
I don't have any audio/drive issues and as I said, my computer used to run Jack2 just fine. The output of cat /proc/asound/cards is
0 [Intel ]: HDA-Intel - HDA Intel
HDA Intel at 0xf0400000 irq 45
It seems strange to me that jack2 and jack2-dbus act exactly the same way in regards to dbus but dbus is just one of the several problems, it seems. Right now I'm running my music production thanks to a live Ubuntu Studio stick (the horror, I know) and after extensive research I feel I'm out of ideas. I mean, I uninstalled jack and deleted all config files and it still doesn't go back to normal, what can I do? I suspect my problem may be related to improper configuration of something in /etc/ but I didn't touch anything audio-related in ages and as I said, jack used to work with the same configuration. It seems to have problem accepting that my sound card is hw:0, maybe. I don't know, it makes me crazy. Can someone help me? Thanks.
Edit: The problem is solved, checkout my last reply to this thread for the solution.
Last edited by VisionsOf (2012-08-28 11:30:13)Hi, I'm having a similar issue. I've been using jack2 as the main audio server with pulseaudio routed through it for more than a year with no issues until a couple of weeks ago when after a power failure jack just stopped working.
After double checking my conf (found nothing strange, again, I'm using jack in normal production for over a year and havent touched conf in a while) and downgrading kernels and jack with no success I realized that if I suspended the PC, and tryed to start jack after resuming it, there wasnt any problems at all!!? Jack works!
Something similar happens in my laptop (same setup), but in this case jack fails to start only the first try, on the 2nd try it will start flawlesly.
Could it be related to systemctl?? dbus?? (the same happens with Dbus interface enabled or not)
Pulseaudio? I've read in forums there are some issues between jack and pulseaudio, and in fact I can see pulseaudio “dead” in both situations (jack working and not)
systemctl status pulseaudio
pulseaudio.service
Loaded: error (Reason: No such file or directory)
Active: inactive (dead)
I guess I'm still not getting right the new Systemd method or the pulseaudio method or both
Here's the log of Qjackctl when it fails (same result starting jack from command line)
ERROR: JackProcessSync::LockedTimedWait error usec = 5000000 err = Connection timed out
ERROR: Driver is not running
ERROR: Cannot open client name = dbusapi
ERROR: failed to create dbusapi jack client
17:49:51.668 ALSA connection graph change.
17:49:51.698 ALSA connection graph change.
17:49:58.744 Could not connect to JACK server as client. - Overall operation failed. - Server communication error. Please check the messages window for more info.
Client name = qjackctl conflits with another running client
Any hints will be very welcome!
Cheers!
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Suddenly my mailboxes time out from all computers at my house to my mail server. Also, browsers cannot connect to the server to see my website, and I cannot connect directly to my webmail application to bypass Mail. Nothing has changed, but suddenl
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SP2013 Search - Query documents only in a library with a specific property bag value
Hi, I am hoping someone might be able to help me as I am struggling to figure out how to get the following scenario to work using query variables. I have configured a property bag value against document libraries and made it indexable and turned it