search for: artsd

I've noticed that artsd sometimes eats as much as 20% of CPU when idle. This is on my old laptop (where playing MP3s eats up to 50-60% of CPU). On newer faster processors this would probably go unnoticed (if it eats 20% on Pentium MMX, on Pentium 4 it might be around 1-2% or even lower). Just wondering if anybody...

Hi , I'm triying to use kphone 4.02, but when i'm make a call the programs doesn't respond any command, so i can't hear any sound .. in sip.conf that's my codec config: disallow=all allow=gsm allow=ulaw allow=ilbc and the kphone give the follow : SipClient: Sending: 06:46:28.116 -------------------------------- ACK

I need to redirect the sound from a remote Centos 5.2 computer to my local Centos 5.2 computer. Both are i386 OS. Searching the web and Centos web site has indicated that it is possible but I have not found any information about how to do it. I am currently using ssh and/or vnc to display the remote computer locally. At this point, the sound is being played on the remote computer only.

...; on the first computer , i copy the line export ARTS_SERVER=ip of > > second > > machine:<port> to the currently logged in user's .bashrc file. Then > > play > > any mp3 with mplayer with audio output chosen as arts. > > > > on the second machine start artsd with artsd -n -u -p 5001 option ( > > however please consider the security) and set up iptables or firewall > > to > > make the second computer to accept request from first computer on port > > 5001. > > _______________________________________________ > > CentOS...

...-- Executing Dial("OSS/dsp", "OH323/06302|20|tT") in new stack *CLI> WrapH323Connection::WrapH323Connection: WrapH323Connection created. -- Called 06302 WARNING[1272234688]: File chan_oss.c, Line 624 (oss_read): Error reading from sound device (If you're running 'artsd' then kill it): Resource temporarily unavailable ClearCallThread::ClearCallThread: Object initialized. -- Hungup 'H323:7530' == Spawn extension (gatekeeper, 06302, 1) exited non-zero on 'OSS/dsp' << Hangup on console >> 0:09.524 H225 Caller:8118810...

...-- Executing Dial("OSS/dsp", "OH323/06302|20|tT") in new stack *CLI> WrapH323Connection::WrapH323Connection: WrapH323Connection created. -- Called 06302 WARNING[1272234688]: File chan_oss.c, Line 624 (oss_read): Error reading from sound device (If you're running 'artsd' then kill it): Resource temporarily unavailable ClearCallThread::ClearCallThread: Object initialized. -- Hungup 'H323:25363' == Spawn extension (gatekeeper, 06302, 1) exited non-zero on 'OSS/dsp' << Hangup on console >> 7:23.991 H225 Caller:8120258...

hello with the lastest CVS version of wine and with winearts i heard no sound ... arts version 1.0.1 from Kde3 so i try a lot of thing ... it start artsd with log option so, i start mplayer2 and play tada.wav ;o) artsd receive the stream and say job finished as xmms-arts but with xmms-arts i heard the sound ... the log of wine : ---------------- I just see about the sound : mcop warning: user defined signal handler found for SIG_PIPE, overriding...

...Foreign Address State PID/Program name tcp 0 0 0.0.0.0:37 0.0.0.0:* LISTEN 7004/xinetd tcp 0 0 0.0.0.0:901 0.0.0.0:* LISTEN 6812/smbd tcp 0 0 0.0.0.0:1030 0.0.0.0:* LISTEN 7942/artsd tcp 0 0 0.0.0.0:7 0.0.0.0:* LISTEN 7004/xinetd tcp 0 0 0.0.0.0:199 0.0.0.0:* LISTEN 4925/snmpd tcp 0 0 0.0.0.0:137 0.0.0.0:* LISTEN 6812/smbd tcp 0 0 0.0.0.0:873...

...n it about a > > "capture channel". How could I find where it is defined ? > then try alsamixer or amixer and watch out for capture and unmute and apply > this for the according channel (mic, line-in, pcm or master) Hum... the problem is that RH 9 uses classic OSS drivers and artsd, not alsa... so I can't use that alsamixer without installing all alsa packages. Maybe it is the solution, but I thought it would be possible to use ices on a classic RH config without installing all the alsa... Pierre --- >8 ---- List archives: http://www.xiph.org/archives/ icecast projec...

I am experiencing a strange problem and wanted to know if someone has faced any similar issues or could provide me with a way to counter this problem. I am in the process of experimenting with asterisk and trying to setup a basic functional system. I have one TDM400P (single port) and one X100P. I am using one analog phone connected to the TDM400P and I also have a couple of Xlite SIP phones

...at XEN level - right? >> Is sound card virtualization now working in any form (at least in >> "3"-mode, by emulation)? > I''d imagine that''s reasonably easy to make work for VT guests. Paravirt > guests need to use network sound daemons at the moment, (artsd, esoundd, or > use FreeNX if you can make it work). Well, with Linux-as-guest it should be possible - but I think neither of those network sound daemons can help with Windows-as-guest, right? >> distribute Windows files over LVM? > You''d want something like my XenFS for this....

Hi, I installed windows media player 6.2 on my system using winetools. I use KDE and artsd is runnning. The sound works correctly on linux but I do not get sound through wine using mplayer2. The error I get is that no audio hardware is available. What can I do to get the sound working on my system. I am using wine 20050111. I haven't used any other player except mplayer2 though win...

...t; "capture channel". How could I find where it is defined ? > > then try alsamixer or amixer and watch out for capture and unmute and apply > > this for the according channel (mic, line-in, pcm or master) > > Hum... the problem is that RH 9 uses classic OSS drivers and artsd, not > alsa... so I can't use that alsamixer without installing all alsa packages. > Maybe it is the solution, but I thought it would be possible to use ices on > a classic RH config without installing all the alsa... Have you tried aumix from the command line? I have used it in RH 9...

How do I use a voicemodem/soundcard to PSTN-gateway - is it possible ? /HHA

I have just installed the Alsa drivers for my 2.4.18-14 kernel (RH8). I have configured the sound card ok with alsaconf and tested with the aplay , works fine. But when I run asterisk it says.. ------------------------------- [chan_alsa.so] => (ALSA Console Channel Driver) Apr 20 18:28:34 ERROR[8192]: chan_alsa.c:339 alsa_card_init: snd_pcm_open failed: No such device or address Apr 20

Another bizarry: If I run the Echo application from the console, I can hear a very long delay (upward to 1,000 ms). I can run the same application from a GrandStream phone (on the same LAN) and hear little delay. What could possibly be wrong? If it were interrupt overload, I'd hear lots of cracks in my echo, right? I'm not hearing that. Besides, a telephony card is not involved.

* EvilOverlord (eviloverlord@kucs.net) écrivait : > In whatever you use as mixer control for the soundcard (alsamixer, > aumix, etc) what is set as the "capture" channel? Your soundcard may > not support capturing what is being played. If it helps : my soundcard is a ES1988 Allegro-1, the module is "maestro3". I use kmix as mixer, but there is nothing in it about a

Potentially nasty bug in current HEAD when printing the name of the file being played (the "Playing: filename.ogg" bit), any % symbols in the name appear to be printf parsed causing either junk to be printed or a crash or both. I noticed this when trying out http streaming as the url has %20's in it for spaces in the file name. Oh, and it still says it's rc2... --