Linux-audio-user May 2006

linux-audio-user@lists.linuxaudio.org

184 participants
187 discussions

[linux-audio-user] fst source not available

by Benjamin Fabricius

Hello all, Slowly but surely Im progressing on my vst server project ive got jack, alsa and my hdsp madi card up and running under my embedded linux. next step is wine... before starting on that i wanted to take a look at the current fst source. it seems to be the most popular solution for running vst/vsti plugins at the moment. unfortunately i am constantly running into dead links. do you have any idea where i can get the fst source? tried http://galan.sf.net/fst-1.7.tar.gz linked at http://joebutton.co.uk/fst/. .. no luck there. I can also not reach www.linuxaudiosystems.com where fst is also supposed to be available for download. Is there any central ressource currently up from where i can download fst? is there any ftp available where the releases of wine, jack and fst or dssi-vst that work best together have been located? Id be glad for any help, Cheers, --bf

18 years, 1 month

Re: [linux-audio-user] -rt IRQ handler priorities

by Kjetil S. Matheussen

Lee Revell: > Subject: Re: [linux-audio-user] -rt IRQ handler priorities (was: Re: > molnar patch) > To: Florian Paul Schmidt <mista.tapas(a)gmx.net> > Cc: Ingo Molnar <mingo(a)elte.hu>, linux-audio-user(a)music.columbia.edu > Message-ID: <1147129704.9116.21.camel@mindpipe> > Content-Type: text/plain > > On Tue, 2006-05-09 at 00:47 +0200, Florian Paul Schmidt wrote: >> On Mon, 8 May 2006 21:54:53 +0200 >> Wolfgang Hoffmann <woho(a)woho.de> wrote: >> >>>> I browsed over that page and didn't find any info on setting up the irq >>>> handler priorities which is _the_ essential feature of -rt. >>> >>> Apropos: on your page on -rt setup (excellent page, btw., many thanks! :-), >>> you suggest raising "softirq-timer/0" to prio 99, to make sleep() function >>> right (http://tapas.affenbande.org/?page_id=40 sleep() based/system timer). >>> >>> I did so, and got strange latencies (> 40 ms) exactly once every 10 minutes, >>> caused by some routing-related action (rt_secret_rebuild) being run by the >>> softirq-timer/0 thread. Don't you get bit by that, too? Kernel is 2.6.16-rt16 >> >> I haven't been bitten by that. Do you also get xruns with [i suppose so, >> just asking to make sure]? I haven't had as much time as before to play >> around and test things, so maybe it has crept into the kernel recently >> or maybe i just always had high-res timers enabled. >> >>> My solution is to configure with CONFIG_HIGH_RES_TIMERS=y. Then, sleep() wakes >>> up correctly even with softirq-timer/0 being low-priority (SCHED_FIFO 1 or >>> even SCHED_OTHER). >>> >>> In general I find adjusting priorities of the various softirq kernel threads a >>> bit of secret art. I can't find much documentation about "what kernel thread >>> runs which job" that would help making some proper decisions here. I found my >>> desktop "feels" most responsive when demoting all softirq thread to >>> SCHED_OTHER. I did so after seeing that with a non-rt kernel, bottom-half >>> handler don't run SCHED_FIFO/_RR at all. So -rt now gives me robust low >>> latencies for jackd, and still proper desktop feeling. >>> >>> Well, maybe this is getting off-topic for this list. But it seems to me >>> trimming priorities between kernel and userland threads is a bit like no >>> man's land. >> >> I agree. Maybe Lee Revell knows more [CC'ing him]. Lee, you know >> something about all these softirq threads? What exactly do they do? > > Not really. The problem with making the softirq timer thread high > priority is that lots of random code gets run from softirq timer > context, including rt_secret_rebuild() which is a well known latency > killer. > Then theres a latency problem in the kernel. A sleeping high priorioty SCHED_FIFO thread must be woken up in time even if another lower priority SCHED_FIFO thread is buzy-looping. And currently, unless the softirq timer has priority 99, that condition can not be fullfilled. So, the softirq timer must run with priority 99.

18 years, 1 month

[linux-audio-user] Simple automated audio processing: newbie question

by Andrew Green

Hi, I have a whole bunch of MP3's for which I would like to do the following: 1) detect a few seconds of silence near the end of the recording; 2) cut off the end of the recording from the silence onward; and 3) save the recording without re-encoding, or save it in a manner that neither diminishes quality nor increases file size unreasonably. Any ideas on how to do this automatically? Many thanks in advance, Andrew Green ___________________________________________________________ Do You Yahoo!? La mejor conexión a Internet y <b >2GB</b> extra a tu correo por $100 al mes. http://net.yahoo.com.mx

18 years, 1 month

Re: [linux-audio-user] Re: Apogee Ensemble with Ardour

by Lee A. Azzarello

----- Carlo Capocasa <capocasa(a)gmx.net> wrote: > Hi Lee! Considered that... Well, I would like to have that 192kHz > option > and the Fireface 800 would have to be reverse-engineered by the > driver > people as far as I know. Also, it does approach Apogee pricing so > I'll > probably just wait for 'The Real Thing'... But thanks for the hin I think that firewire audio is only recently supported with the freebob package. I would recommend the RME Hammerfall PCI card, which IIRC has ALSA drivers in the kernel: http://www.rme-audio.com/english/hdsp/cardpci.htm#PCI -lee

18 years, 1 month

Re: [linux-audio-user] Re: Apogee Ensemble with Ardour

by Lee A. Azzarello

----- Carlo Capocasa <capocasa(a)gmx.net> wrote: > Hi Reuben! Thank you very much for your very helpful information... > It > looks right now as if I will continue the Ultra-Low-End for a while > until Apogee has better support. The PCI interfaces by RME are supported in the kernel last time I checked. Their converters and I/O options are excellent, though it's been four years since I used the multiface. -lee

18 years, 1 month

[linux-audio-user] [ANN] Dino 0.2

by Lars Luthman

Dino is a MIDI sequencer for GNU/Linux that uses JACK MIDI and JACK transport to send MIDI events to synths and synchronise with other sequencers or transport aware programs. It uses LASH to save and restore sessions. This is the first release. Get it at http://dinoseq.sf.net . Requirements: * libglademm >= 2.4.1 * libxml++ >= 2.6.1 * JACK >= 0.100 with the MIDI patch available here: http://www.custard.org/~deviant/jack-midi/ * LASH >= 0.5.0 -- Lars Luthman PGP key: http://www.student.nada.kth.se/~d00-llu/pgp_key.php Fingerprint: FCA7 C790 19B9 322D EB7A E1B3 4371 4650 04C7 7E2E

18 years, 1 month

[linux-audio-user] ntfs & smb in ccrma kernel

by BJaY

Hi, I installed kernel-2.6.16-1.2080.13.rdt.rhfc4.ccrma.i686.rpm and support for ntfs and smbfs seems to have dissappeared. Can I put this back in without rebuilding the kernel ? Also I tried out the rrt version of above, worked OK for audio but hard locked when I tried to use tvtime and my webcam (Creative). Does this mean I should'nt use it for audio because at some time it will lock up or, could it just be somthing to do with the camera driver and tvtime. Cheers, Bruce.

18 years, 1 month

[linux-audio-user] My MIDI latency is crap

by Josh Lawrence

Hey all, So I sat down last night to do a little sequencing with my Roland RD-700SX. Since Rosegarden was giving me fits, I jumped over to MusE, got everything setup, and started trying to play. The latency was awful. I'm curious what I'm doing wrong...here's a quick rundown on what I have going: Ubuntu 6.06 Dapper, using the default kernel (I wonder if this is it? Does MIDI latency depend on having a preemptable kernel?) Jack wasn't running, I was doing all of the MIDI connections via MusE. Finally, I was connecting the RD directly to my computer via the USB port for my MIDI interface. aconnect recognized it just fine. My guess is that, somehow, good MIDI latency somehow depends on having a preemptable kernel running, but that is just a guess. Or maybe I need to be making all of my MIDI connections via jack? Normally I wouldn't pester the list with something like this, but I have to have this sequence done tonight. I didn't adequately test my new setup prior to committing to having this done, so I'm kinda in a pinch. Any direction you can give me on troubleshooting this would be appreciated. Thanks, Josh -- Josh Lawrence http://www.hardbop200.com

18 years, 1 month

[linux-audio-user] segmentation fault using miniFMsynth test program

by Daniel

Hi, I've got ALSA, installed and configured. esdplay can playback a wav file. When I compiled and ran miniFMsynth I get a segmentation fault error. What can I do to probe the setup and check the ALSA configuration is alright. Thanks, Daniel

18 years, 1 month

[linux-audio-user] Re: Reel to Reel Tape, Speeds and Software Equalisation (Steve Fosdick)

by Phil Mendelsohn

> 7. Reel to Reel Tape, Speeds and Software Equalisation > (Steve Fosdick) Hi, Steve, > I can record the output of the tape machine running at 7 1/2 inch/sec and > then use software re-sampling (sox, or sndfile-resample) to get a file > that plays at the correct speed but I wonder if I should also be applying > some equalisation in software as multi-speed machines usually have > different equalisation constants for different speeds. The idea thing would be to record it at 88.1kHz, then you'd only have to decimate it (i.e., downsample by 2, with no filtering) to get the result you're after. Reconstruction filters on playback would be giving you free anti-aliasing. :) This would also have some (really mild) noise reduction benefits. > So, does the equalisation compensate for the record process or playback > process at a given speed or a combination of both. If it compensating > only for playback then I should be OK, otherwise I presumably should be > applying extra EQ in software - anyone know how I could calculate what, > assuming I have the usual LADSPA plugins at my disposal and that the > signal has already been through the playback EQ for 7 1/2 inch/sec? You probably don't need to get very uptight. I think what you're referring to are the NAB/IEC/DIN EQ curves, but I don't believe those were ever specified for the speeds you're talking about. If you were worried about the response for 15ips or 30ips, then sure, it would make sense. But the odds are that the EQ circuits at your disposal are fairly funky to begin with, so I'd really recommend doing it by ear. (It always chafed a little, but even doing pro level mastering, that's what it comes down to for a lot of tape decks.) In fact, unless you were given a set of calibration tones for the partcular machine that *made* the recordings, you really have no other choice. This is a really long way of saying, "Yes, EQ compensates for _both_ record and playback." HTH. Phil M P.S.: If these are mono recordings, you will get about 6dB of noise reduction by summing the two channels of a stereo playback machine to mono! -- Dept. of Mathematics, 342 Machray Hall U. of Manitoba, Winnipeg, Manitoba, Canada R3T 2N2 Office: 446 Machray Hall, 204-474-6470 http://www.rephil.org/ phil at rephil dot org

18 years, 1 month

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Linux-audio-user May 2006