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20713 discussions

Re: [linux-audio-user] Re: Recent LKML discussion on preempt/latency in 2.6 kernels

by Sampo Savolainen

Fernando Pablo Lopez-Lezcano <nando(a)ccrma.stanford.edu>: > Caveats: FC1 qt + dri + 2.6.4 + SCHED_FIFO = hanging machine, as > reported a while back. Yuck. Turning off dri solves this one at the cost > of no acceleration. > > With all the above I'm able to run Jack with SCHED_FIFO and no xruns in > a short test run at 64x2 (with a background task tar'ing /usr to a > file). But with DRI off, Qjackctl and some real Jack applications I do > get xruns even at 128x2, I guess my cpu is maxing out... I'm confused. I probably have misunderstood something, but: First you say, that you get a "hanging machine" with SCEHD_FIFO enabled applications, but later you say that you can run Jack with SCHED_FIFO with all the different configurations/kernels you mention. What do you mean with a hanging machine? Do you get 'random' lockups, or does your system hang always when some specific software is run with SCHED_FIFO? Do you get lockups with this kernel & setup (option d in your email)? Could you give a list of specific patches (and urls) you have applied to 2.6.4-1.286.2.ll (option d), or maybe a link to the source package? Sampo

21 years, 11 months

[linux-audio-user] couple of muse questions

by antoine rivoire

Hi all. a simple stupid question: in muse the tempo is in percentage? or is 100 really 100 bpm? what is the rationale here? Secondly when starting muse as a normal user I get: cannot open rtc clock /dev/rtc: Permission denied how can I change the permission permanently? kernel 2.6 with realtime.0.04 Regards -- antoine rivoire <antoine.rivoire(a)ntlworld.com>

21 years, 11 months

[linux-audio-user] Poor SN ratio with high quality sound cards

by John Ouzts

Gentlemen: I follow this list from the digest, so I could have missed a critical post. If so, please point me to the thread. I have two PCs I use almost exclusively for sound editing (with ReZound --- highly recommended for its crossfade ability). I record classical music on an Alesis MasterLink at 96kHz/24 bits, so I have quite a SN ratio to begin with. I need to edit the gaps between tracks where I want to preserve the hall ambience. Whether I use an RME Digi96 PAD or an M-Audio Delta 44, the signal I get out of the sound editor is so low that I hear a lot of hum and other extraneous noise by the time I crank my amplifier's volume up enough to hear what is between the pieces. This noise is _not_ there when I burn a CD-R, however. I assume this means I need to send a digital signal (preferably 96/24 and lower) to an external DAC before sending the signal to my monitoring system. Would someone be kind enough to explain how to get a signal out of a PC that is fully equivalent in S/N to the CD-Rs the computer can burn? Many thanks, John

21 years, 11 months

[linux-audio-user] [simon@arrowtheory.com: [Portaudio] ANN: dsptools-0.4.0]

by Eric Dantan Rzewnicki

This looks interesting. -Eric Rz. ----- Forwarded message from Simon Burton <simon(a)arrowtheory.com> ----- Date: Thu, 25 Mar 2004 07:18:59 +1100 From: Simon Burton <simon(a)arrowtheory.com> Subject: [Portaudio] ANN: dsptools-0.4.0 To: portaudio(a)techweb.rfa.org X-Mailer: Sylpheed version 0.9.3 (GTK+ 1.2.10; i686-pc-linux-gnu) First public release. Comments sought. dsptools ======== Here be Python wrappers for portaudio, ladspa and libsndfile. There are three modules: ladspa, sndfile, and portaudio. They are independant of each other, ie. they should compile/run individualy. Data interchange is type sensitive and uses numarray arrays to store sound data. It's possible to segfault by doing stupid things, eg. from within the portaudio callback only some portaudio calls are allowed. See the python scripts for example usage, *.pyx files for implementation, and *.pxi for underlying c library info. http://sourceforge.net/projects/dsptools/ It's under tested, under documented, but it's pretty cool. Install ------- Use any of the setup scripts: $ python setup*.py build $ python setup*.py install Dependancies ------------ python 2.2.3 or better: http://www.python.org/ numarray 0.8.1 (or better): http://sourceforge.net/projects/numpy/ For portaudio module: portaudio v18.1 (not v19): http://www.portaudio.com/ For sndfile module: libsndfile 1.0.5 or better: http://www.mega-nerd.com/libsndfile/ For ladspa module: ladspa.h v1.1 : http://www.ladspa.org/ladspa_sdk/ and any plugins (optional): http://www.ladspa.org/ Todo ---- * Enable the writing of ladspa plugins in python (!) * portaudio v19 Related Projects ---------------- fastaudio, for portaudio & libsndfile : http://www.freenet.org.nz/python/pyPortAudio/ libsndfile-python : http://arcsin.org/archive/20030520025359.shtml Thanks to --------- Ross Bencina (portaudio) Erik de Castro Lopo (libsndfile) Greg Ewing and all the pyrex people ------------------------------------------------------------ Simon Burton March 23, 2004 -- Simon Burton, B.Sc. Licensed PO Box 8066 ANU Canberra 2601 Australia Ph. 61 02 6249 6940 http://arrowtheory.com _______________________________________________ Portaudio mailing list Portaudio(a)techweb.rfa.org http://techweb.rfa.org/mailman/listinfo/portaudio ----- End forwarded message -----

21 years, 11 months

[linux-audio-user] Re: [ANN] vsti v0.0.2

by Kjetil Svalastog Matheussen

Daniel James: > > > I agree it's pointless -- you'd just be getting a screenshot of > > someone else's VST plugin. > > The point as I see it is to prove that the VSTis are rendered properly > in a Linux environment, since they are highly visual tools. To see a > couple of them floating around a KDE desktop makes the point for > people who might not appreciate the implication of your code. > Actually, this is not a very revolutionary program. You have had the possibility to run vst instruments in linux since november/december 2002. This program only makes it a bit less work to set up. There are many of screen-shots of vst- plugins running in linux floating around on the net. --

21 years, 11 months

Re: [linux-audio-user] [ANN] vsti v0.0.2

by Kjetil Svalastog Matheussen

Daniel James: > Hi Kjetil, > > > SHORT DESCRIPTION > > Run a windows vsti plugin as a jack and alsa-seq client. > > Are there any screenshots of VSTi's running on Linux available? Do > they look the same as on Windows? > This is a very simple program. It sets up an internal jack client that starts a vst instruement, sets up an alsa client, and opens the gui for the vst instrument. No need for a screen-shot; Same look as in windows. --

21 years, 11 months

[linux-audio-user] Re: Recent LKML discussion on preempt/latency in 2.6 kernels

by Kjetil Svalastog Matheussen

Chris Metzler: > > Hi, > > Some folks here may be interested in a recent discussion of pre-emption > in the 2.6 kernels and potential issues with it. > > See: > > http://kerneltrap.org/node/view/2702 > This is very interesting. It says that preempt have no influence on the worst-case latency. In other words, this means that preempt in theory is useless regarding use of realtime audio. --

21 years, 11 months

Re: [linux-audio-user] Poor SN ratio with high quality sound card s

by Stephen Hassard

I had big problems with the Zalman 400W PSU unit. Not really a problem with the power supply itself, but more a matter of it not sucking enough heat out of my case. Having a few high heat devices (Athlon XP 3000+, Radeon 9800) and not the greatest case for airflow to start, the system was running at extremely high temperatures, causing crashes at regular intervals. You'd probably want to augment the cooling in your case if you go for this supply; I'm not sure if I'd really consider it a "silent" PSU for this reason. :P later, Steve John Anderson wrote: > On Wed, 2004-03-24 at 08:20, Mark Williams (MWP) wrote: > >>Really the only thing you can do is invest in a good quality external DAC and >>run a SPDIF (optical preferably for ground isolation) signal to it. >> >>It seems a lot of high quality soundcards now are quoting the S/N and other >>stats of those of the DACs used on the card, not by measurment of what actually >>comes out of the back of the card. >> >>The main problem is the power supply in the PC. >>Its just far too noisy to be able to produce good quality audio. > > > Would using a good-quality power supply make a difference? Something > like this: > > http://www.zalman.co.kr/english/product/zm400b-aps.htm > > bye > John > >

21 years, 11 months

Re: [linux-audio-user] couple of muse questions

by Andreas Andersson

>From: James Stone <stone1(a)btinternet.com> > > > Secondly when starting muse as a normal user I get: > > > cannot open rtc clock /dev/rtc: Permission denied > > > how can I change the permission permanently? > > > kernel 2.6 with realtime.0.04 > > > > Don't know about 2.6 kernels. Have you tried setting muse suid root? I >assume > > you are launching as muse -R (?) > > > >What does muse -R do? I just run it as muse, and connect to jack. Not >sure what the realtime option does, just hangs on my computer.. unless I >change muse to suid (which is not necessary if it has been compiled with >rtcap). Don't worry about the kernel version. makes no difference. I have the same experience. rtcap enabled, no difference between kernel versions. I don't use the realtime module but the kernel hack. Works well with Jack and everything else. /AA _________________________________________________________________ MSN 8 with e-mail virus protection service: 2 months FREE* http://join.msn.com/?page=features/virus

21 years, 11 months

[linux-audio-user] Re : Poor SN ratio with high quality sound

by Mirouf Linux

>Gentlemen: > >I follow this list from the digest, so I could have >missed a critical >post. If >so, please point me to the thread. > >I have two PCs I use almost exclusively for sound >editing (with ReZound >--- >highly recommended for its crossfade ability). I >record classical music >on an >Alesis MasterLink at 96kHz/24 bits, so I have quite a >SN ratio to begin >with. >I need to edit the gaps between tracks where I want to >preserve the >hall >ambience. Whether I use an RME Digi96 PAD or an >M-Audio Delta 44, the >signal >I get out of the sound editor is so low that I hear a >lot of hum and >other >extraneous noise by the time I crank my amplifier's >volume up enough to >hear >what is between the pieces. This noise is _not_ there >when I burn a >CD-R, >however. > >I assume this means I need to send a digital signal >(preferably 96/24 >and >lower) to an external DAC before sending the signal to >my monitoring >system. >Would someone be kind enough to explain how to get a >signal out of a PC >that >is fully equivalent in S/N to the CD-Rs the computer >can burn? You just have to get out a 16 bit 44,1khz signal, just like on a cd. There is no use for you to have a 24/96 signal for monitoring and wondering all the time if it would be the same on the CD... And I'm wondering if it possible to get a 24/96 signal out with the RME... If you wanna get digital out of the RME, you've got nothing to do, just plug the little cable converter on the sub-d connector ont the board, and play music, the monitoring stream is copy on the digital out by the board. I don't know about the delta 44... Bye François > >Many thanks, > >John Yahoo! Mail : votre e-mail personnel et gratuit qui vous suit partout ! Créez votre Yahoo! Mail sur http://fr.benefits.yahoo.com/ Dialoguez en direct avec vos amis grâce à Yahoo! Messenger !Téléchargez Yahoo! Messenger sur http://fr.messenger.yahoo.com

21 years, 11 months

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