Hallo, The Eye hat gesagt: // The Eye wrote: > On Fri, Jan 23, 2004 at 06:41:27PM +0100, Frank Barknecht wrote: > > I have no problems with dropouts on my Delta Audiophile card. Do you > > run a low-latency/preemptible patched kernel? > > nope, since so far I have not found a reliable answer to the question > _which_ one I should run .. i.e. which patches I should use and so forth > .. Is it possible to have a kernel that has stability and security stuff > _as_well_as_ the low latency stuff?
I run 2.4 in various versions with Andrew Morton's Low latency patch and the preemptible patch, which is already included in 2.6. No stability problems here, and certainly not because of these patches. Security-wise I don't see a problem with LL and Preemptible patch, but if you want to use JACK later on, you will need a tiny capabilties patch on 2.4, which *might* have security implications. I wouldn't run it on a public webserver, but I don't run a public webserver on my personal audio machine anyways. > > Do you run ecasound with higher scheduling (-r:sched_priority) ? > > yes, doing that. without it, the situation is even worse. This is good. But without some kind of latency counter measures the kernel itself still can interrupt ecasound any time. That's why the preemptible patch (maybe combined with LL patch) is so crucial for audio on Linux. Just use those two patches in versions matching to your favourite kernel. > well, as far as I understood Takashi, it is. I mean my card is a stereo > card but still apparently internally expects 10 channels, if I > understood him correctly. I'm a little confused. All I want is to be > able to record stereo sound with excellent quality on line in. An listen > to it with excellent quality on line out. I don't want much else, but the Linux kernel 2.4 cannot do this reliably without the patches. You can however tune ecasound to use larger buffers. I have to run an unpatched kernel at work to do unobserved recordings using ecasound from a cronjob. There I use ecasound in double-buffering mode and with a large buffer. I cannot look it up currently, but it involves this option: '-z:db,dbsize' enables double-buffering for audio objects that support it (dbzise=0 for default, otherwise buffer size in sample frames). Try to play with this. Regarding xmms: I stopped using it on my ice1712 card months, maybe almost 2 years ago because it proved to be very unreliable, it doesn't understand the mixer on the Audiophile (OSS-emu or ALSA), it cannot change Volume on mixerless cards like my USB device because it requires a mixer which it doesn't even understand on the other card. But maybe this has changed in the last year. Because of all this I use alsaplayer for daily music delivery to great pleasure: It has a very clean architecture, it uses large enough buffers for skipfree audio even without realtime scheduling, it doesn't hurt the eyes, it can be remote-controlled from Pd, it's generally very cool. Maybe it can make you as happy, too? ciao -- Frank Barknecht _ ______footils.org__ ------------------------------------------------------- The SF.Net email is sponsored by EclipseCon 2004 Premiere Conference on Open Tools Development and Integration See the breadth of Eclipse activity. February 3-5 in Anaheim, CA. http://www.eclipsecon.org/osdn _______________________________________________ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
