Linux-audio-user February 2006

linux-audio-user@lists.linuxaudio.org

207 participants
268 discussions

[linux-audio-user] LAC2006: JackLabMusic

by Michael Bohle

Hi there, This first answer from the LAC jury about our music submission just reached me. They decided : We (me and some friends) can play there on the "Musikbalkon". JackLabMusic "A live experiment with some JackLab musicans" will performing on the »Linux Sound Night« Linux Audio Conference 2006 Saturday Night, 29.April ZKM Karlsruhe http://lac.zkm.de/2006/ More information about this project with some audiosamples. http://jacklab.net/Wikka/wikka.php?wakka=JacklabMusic Michael www.jacklab.net - proAudio for openSUSE

19 years, 12 months

[linux-audio-user] Re: capturing through plugins

by Kjetil S. Matheussen

conrad berh?rster: > > Hello, > here is the next , hmmm, challenge. > i want to capture the audo output of xine, the Video player. I have read > somewhere, that it is possible through the alsa-jack plugin, since xine has > alsa. > so a possible way, i'm thinking of is > xine -> alsa-lib -> alsa-plugin -> jack -> myapp > > the first step is to jackify xine. is there any experience with this topic. > Sure, I do that all the time with libjackasyn. In my experience, xine with libjackasyn works much better than mplayer with jack output because mplayer doesn't do proper resampling. $ jackstart xine movie.avi (it goes thru esd which works just excellent with libjackasyn) http://gige.xdv.org/libjackasyn/

19 years, 12 months

[linux-audio-user] capturing through plugins

by conrad berhörster

Hello, here is the next , hmmm, challenge. i want to capture the audo output of xine, the Video player. I have read somewhere, that it is possible through the alsa-jack plugin, since xine has alsa. so a possible way, i'm thinking of is xine -> alsa-lib -> alsa-plugin -> jack -> myapp the first step is to jackify xine. is there any experience with this topic. thanks c~

19 years, 12 months

Re: [linux-audio-user] Re: asoundrc and aplay

by Sampo Savolainen

Quoting conrad berhörster <conrad.berhoerster(a)gmx.de>: > i need that inputs not in a harddisk recording / pro audio way. all i > need is > getting the signal into the app, use some effects and put them out again. > no > sample sync, no need for "realtime". all inputs are independed. > Than i have read http://alsa.opensrc.org/TwoCardsAsOne. > is this page wrong! > i thought, one big advantage of jack is, to handle more than one card. Jack works only with one soundcard, or an array of sample-synced soundcards. Nothing else. You do need sample sync to make your idea work properly. Sample sync, or you dump the idea of using jackd and do resampling in the code. Here is the reason: Soundcard 1 -> Application -> Soundcard 2 Looks simple, eh? It's simple because it's wrong. The truth is this: Soundcard 1 -> Application -> Soundcard 2 @44102hz @44092hz (This is due to the fact that no two crystals are the same) What happens in this scenario? When you are running the application for a period of say, 10 minutes, the output you hear from soundcard 2 lags about 0.136 seconds. After 20 minutes, it lags about 0.272 seconds, etc. The reason for this? While soundcard 1 takes in 44102 samples per real-world-second, soundcard two outputs only 44092 samples in that time. If soundcard 2 would have a faster crystal than soundcard 1, you'd be in a heap of trouble. For every buffer you would write to soundcard2 you would get an underrun where soundcard 2 isn't fed data fast enough (altough, exact error frequency depends on the buffering scheme used). > yes, i know. i have read them too. maybe you can explain, what you mean > exactly with the notion "multichannel device". Is it just a couple of > IOs, > without sync, or are the requirements for pro audios (with sync). With multichannel, we mean "audio devices running in parallel or serial in the same signal chain". > if all this stuff isn't possible, do you have any idea to fix the > problem. Either use only one soundcard (if you are recording from one and playing back on the other one, i don't see why you couldn't use one soundcard), or don't use jack and resample between the two soundcards. Altough, getting a software resampler to perfectly sync two non-synced cards is a mean feat. If you want to use two soundcards as one, the best option is to: wordclock them. That is, to make them sample-sync. Sorry. But this is how it is. Sampo

20 years

[linux-audio-user] realtime-preempt without PAM?

by ross＠lug.udel.edu

Hello everyone, I've been using the Realtime LSM with 2.6 for some time now, but recent issues with apps getting dropped from Jack makes me think I should explore realtime preempt. I've built myself 2.6.15.1-rt16 and have it running. Everything I've read has information on configuring PAM to setup limits. My box is running Slackware, which doesn't use PAM. Is there some software I can use instead of PAM, for this purpose? -- Ross Vandegrift ross(a)lug.udel.edu "The good Christian should beware of mathematicians, and all those who make empty prophecies. The danger already exists that the mathematicians have made a covenant with the devil to darken the spirit and to confine man in the bonds of Hell." --St. Augustine, De Genesi ad Litteram, Book II, xviii, 37

20 years

[linux-audio-user] Fwd: OT -- USB History

by Christoph Eckert

Hi all, sorry for cross posting, but I found the following info very interesting when discussing about USB 1.1 vs. USB 2.0 stuff. Some of us (including me) would like to see a USB 2.0 breakout box which grants more than 2x2 channels while using a standard USB 2.0 protocol. Bruce Wahler politely agreed to spread his info to our lists, so all credits go to Bruce. ce ---------- ---------- Subject: OT -- USB History Date: Donnerstag, 16. Februar 2006 18:12 From: Bruce Wahler <desp(a)mm.ed> To: access-list(a)ampfea.org Hi All, [Warning: This post has little to do with the Virus TI per se. It might be of interest to some of you, though.] I was involved in the early USB efforts, working for a major PC manufacturer. The 3-tier speed approach of USB is a confusing -- and necessary -- part of the design. Early USB appealed to two groups: 1) manufacturers who wanted to simple, cheap way to untangle the rat's nest of wires that were growing behind computers; and 2) developers who wanted a better, more flexible connection than serial and parallel ports provided. In addition, the creators of USB had this grand vision of "USB everything": kitchen appliances, phones, televisions, you name it. USB attempts to satisfy all of these needs, but the goals of different markets are sometimes at odds with each other. Devices like mice can't afford to add even $1.00-2.00USD of product cost, because their customer base won't accept the price increase. On the other end, there's no such a thing as "too fast" for disk drives and networks. USB 1.0 (and 1.1) came out with Low- and Full-Speed specifications to try to bridge the needs of these two camps. Same connectors, same (or similar) cables, same hardware at the host (computer) end; all of the higher-speed functions had to be a layer on top of the basic ones. At the time of USB 1.0 (1995), the practical limit for cables and such was considered to be somewhere in the range of 10-15Mbit/sec. This wasn't a PHYSICAL limitation; it was governed by the cost of hardware (cables, connectors, ICs, etc.) compared to the amount of data being sent (<1GB). Unfortunately, USB 1.x took several years to gain acceptance. (PCs had the USB ports back in 1995, but there were no real peripherals nor OS support for 3-4 more years. One of my bosses used to refer to it as the "Useless Serial Bus.") By the USB really took hold (2000? 2004?), there were enough advances in technology and manufacturing to up the speed a great deal. Add to that the need to transfer more data, and the fear that FireWire would eclipse USB, and "Hi-Speed USB" was born. Hi-Speed USB follows the same rules as USB 1.0: faster protocols must work around the limits of slower ones, so nothing becomes truly obsolete. This is why a 12Mbit/sec Virus TI is still "USB 2 compliant." Some important things to know about Hi-Speed USB: 1. The GUARANTEED cable length is shorter (5m vs. 2m). With a quality cable, you might run further, but there's no whining if it doesn't work. This certainly limits the ability to use the Virus TI as both a recording platform and a performance synth at the same time. 2. Raw bandwidth numbers of Hi-Speed USB are deceptive. (This is also true of FireWire.) While the cable and ICs can support 480Mbit/sec., it takes great drivers, proper interrupt selection, and a relatively unused computer to use that bandwidth. Otherwise, it's a game of "hurry-up-and-wait." Sharing USB with slower devices also clouds the picture. 3. USB 2.0 enhancements focused on data storage. There weren't any high-speed audio extensions added. If Access had wanted to use 480Mbit USB audio, they would have had to develop and support it from scratch -- on both the Mac and PC. So, it's not just a case of adding a little product cost; it's a large development and testing challenge, too. Why weren't there audio extensions? Probably because the two "official" audio uses for USB -- Internet phones and digital USB audio -- didn't need them. The first one is fine with 12Mbit/sec, and the second one never really caught on. 4. The USB specifications were mostly written by big companies like Microsoft, IBM, Intel, and Compaq. They sunk a lot of resources into USB, and so their needs took top priority. None of those companies is known for professional audio gear -- they're computer companies, and USB audio was and still is a bit of an afterthought. (Quick: Name me one 'major' US PC manufacturer who sells a true MPC in their standard line? Anyone?) So, why not add the hardware (ICs) now, and write the OS support later? The approach rarely works, IMHO. Even in the computer industry, known for technology advances, hardware that is unused at product launch often remains forever unused. Why? Remember the old adage, "If it ain't broke, don't fix it" ? Well, updating software or firmware requires breaking that rule. And anyone who's written software will tell you that bugs crop up in the strangest places. While the updates are cool, there's often very little evidence that the efforts resulted in big sales increases. Thus, a small company like Access must be choosy when planning product updates. Regards, -BW -- Bruce Wahler Design Consultant Ashby Solutions http://consult.ashbysolutions.com 978.386.7389 voice/fax bruce(a)ashbysolutions.com _______________________________________________ access-list mailing list access-list(a)ampfea.org http://www.ampfea.org/mailman/listinfo/access-list <--- SUBSCRIBE/UNSUBSCRIBE DETAILS HERE Patches: http://www.ampfea.org/cool/stuff/access-list -------------------------------------------------------

20 years

[linux-audio-user] CCRMA and yum

by thewade

Hello list! I was hoping to install ardour on one of my FC4 machines. I haven't been using this machine for realtime audio so I am running the FC4 vanilla kernel so that the livna.org's ati-graphics and the ndiswrapper kernel modules will work. I want to investigate using the CCRMA packages though. I have questions: (1)Can CCRMA be accessed through yum instead of apt? If I knew where the rmps lived (and if they are signed with a gpg key where that is too) then I think could get yum to work nicely with CCRMA. (2)If I used the lsm patched CCRMA kernel (the CCRMA kernel is lsm-module patched so that I can allow realtime priorities to a gid, right?) will this kernel work with the livna kernel modules? If not I guess I can custom build the modules using rpmbuild. I have done that in the past, and it is inconvienent but I can do it if need be. Thanks all! -thewade PS: I am trying to stay in the bay area but I am having trouble finding work. If anyone needs a lab-rat/art-geek please let me know! http://aproximation.org/resume.php

20 years

[linux-audio-user] More music

by Steve D

A few weeks ago I posted a recording of a piece for piano I was composing: Arabesque 1, for piano (work in progress) http://www.xscd.com/pub/music/audio/ogg/arabesque1.ogg At that time there was little but one primary melodic/harmonic motive. I have since fleshed-out and completed the piece: Arabesque 1, for piano (complete/final) OGG: http://www.xscd.com/pub/music/audio/ogg/steve-doonan_arabesque-1.ogg MP3: http://www.xscd.com/pub/music/audio/mp3/steve-doonan_arabesque-1.mp3 Recorded in one uninterrupted (no punch-ins/punch-outs) "take" (out of many failed attempts with one or two mistakes ;-) into Ardour at 48,000 sample rate, mastered through Jamin the output of which was recorded back into Ardour, then exported as a .wav at 41,000 sample rate with "shaped noise" dither, encoded with lame (mp3) and oggenc (ogg). Ardour and Jamin--what great software. Steve D New Mexico, US -- ---------------------------------------------------------------- The three things every person needs, gay or straight, are to be a realist about themselves, an optimist about what life can bring them, and willing to take what they're handed and make it work. -Krikit ----------------------------------------------------------------

20 years

Re: [linux-audio-user] 2.6.15.4, realtime-lsm, jackd -R,

by Kjetil S. Matheussen

Frank Barknecht: > > Is it true that realtime-lsm is deprecated? How do I enable realtime > > preemption on a 2.6.15? > > It's true. The easiest alternative IMO is to use set_rtlimits. A quick > explanation is at http://tapas.affenbande.org/?page_id=22 > > Basically you download and install this: > http://www.physics.adelaide.edu.au/~jwoithe/set_rtlimits-1.1.0.tgz > then add something like this to /etc/set_rtlimits.conf: > > @audio /usr/bin/jackd -1 90 > > assuming you have a user group "audio" and start jackd like: > > $ set_rtlimits -r /usr/bin/jackd --realtime -d alsa -d hw:0 > > Alternatively you can use a patched PAM-module, but that is a bit more > work, if your distribution doesn't ship it. That is rediculous. Do you really have to start program that way to get realtime priority? In case, I guess (and _really_ hope) that realtime-lsm (or something similar) will continue to exist for a long time still.

20 years

[linux-audio-user] 2.6.15.4, realtime-lsm, jackd -R, system performance down!

by Ismael Valladolid Torres

I just compiled a vanilla kernel 2.6.15.4 in my G3 powerpc. Also I compiled, installed and loaded the realtime-lsm module and I started to play with jackd -R and some jack enabled applications. If I start jackd without realtime, everything runs nicely, of course with some xruns when I switch windows, manipulate GUIs and such. This is more or less what I expected. But then I run jackd realtime using qjackctl. I start amsynth and it takes a bit longer to load but then runs without a mess. Sound is clean, no artifacts, no xruns. Then I try horgand. It takes a lot for it to load, then it sounds nicely, I get no xruns, but my X Window system becomes unresponsive. Later I try zynaddsubfx. It takes half an hour to open, and then it kills qjackctl, leaving the jack server running. Meanwhile top shows that zynaddsubfx is taking 35% of memory, and qjackctl and jackd 25% each. Could this be an issue with the applications, with the kernel, with the realtime-lsm module, with priority of my X Window session? I must insist this only happens when jackd is run with the -R option. Is it true that realtime-lsm is deprecated? How do I enable realtime preemption on a 2.6.15? Any ideas welcome. Cordially, Ismael -- Dropping science like when Galileo dropped his orange http://lamediahostia.blogspot.com/ http://www.flickr.com/photos/ivalladt/

20 years

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Linux-audio-user February 2006