Linux-audio-dev December 2005

linux-audio-dev@lists.linuxaudio.org

61 participants
53 discussions

Re: [linux-audio-dev] dealing with complex numbers

by Taybin Rutkin

Sorry, meant to include this link: http://developer.apple.com/documentation/DeveloperTools/gcc-4.0.1/gcc/Compl… http://www.gnu.org/software/libc/manual/html_node/Complex-Numbers.html 'nando's C++ usage looked a bit cleaner to me, IMHO. Taybin -----Original Message----- From: Artemiy Pavlov <artemio(a)kdemail.net> Sent: Dec 13, 2005 3:22 PM To: The Linux Audio Developers' Mailing List <linux-audio-dev(a)music.columbia.edu> Subject: [linux-audio-dev] dealing with complex numbers Hey everybody! This may be a little bit off-topic, but can anyone suggest me any reading on how to use complex numbers in C or C++? Is there any library or ++ classes for such computations? I'd appreciate any help. Thanks! With respect, Artemiy.

18 years, 9 months

[linux-audio-dev] dealing with complex numbers

by Artemiy Pavlov

Hey everybody! This may be a little bit off-topic, but can anyone suggest me any reading on how to use complex numbers in C or C++? Is there any library or ++ classes for such computations? I'd appreciate any help. Thanks! With respect, Artemiy.

18 years, 9 months

[linux-audio-dev] NPTL jack+ardour: large memlock required (was: Jack and NPTL (again?))

by Jonathan Woithe

Hi On Tues Nov 22 I wrote: > I've recently moved a system over to Slackware 10.2 which utilises NPTL and > I'm aware of the issues NPTL has raised in the past. Based on a comment on > the Jack website though I sort of assumed that things were now in hand and > that Jack had a workaround in place for the issue. > > Despite this I have found that jackd itself (when run using set_rtlimits) > gives an error (-11, EAGAIN) when "creating realtime thread". I have since discovered (thanks to strace) that the problem lies in an mmap2() call which requests a size of around 8MB. It appears to be part of the NPTL pthread_create() function. The error returned by mmap2() (EAGAIN) indicates either a locked file or "too much locked memory" (according to the manpage). Because this is an anonymous map the problem must have been the latter. Increasing the "locked memory" limit from 20MB to 40MB made jackd start without having to resort to the LD_ASSUME_KERNEL hack. Another problem then surfaced: with jackd running under NPTL, jack applications using NPTL (eg: ardour) suddenly started to require all of their memory to be lockable. For ardour with a large session this can get quite large (I saw 130MB at one point last night). So, with jackd running with LD_ASSUME_KERNEL = 2.4.19 and a memlock limit of 20MB everything worked and ardour (with NPTL - ie: without LD_ASSUME_KERNEL) was obviously not trying to lock excessive amounts of memory. With jackd running with NPTL and a large enough memlock limit things work but jack clients appear to require much more memory be locked. One thing I will try next is recompiling ardour; perhaps there's something funny there. In any case though, does any of this ring a bell with anyone? Regards jonathan

18 years, 9 months

Re: Re: [linux-audio-dev] Merging digital sampling sessions

by Jonathan Woithe

David Kastrup writes: > On another note: it would seem like a simple enough task to let the > recording start with the first non-zero sample. ... I'd also like to prune > trailing zeros at some point of time. Although it is not a native ALSA app (due mainly to me not having the time to port it yet), batchrec does this. It still works with ALSA through the OSS emulation layer; adding native ALSA support is on my todo list for the next few months. You can grab it from http://www.physics.adelaide.edu.au/~jwoithe/batchrec-1.2.0.tgz It's pretty raw (no configure script yet) but shouldn't be too hard to compile on most systems. Batchrec can be requested to wait for signal above a certain amplitude before it records. It will also stop recording after the same threshold has been present for a user-specified time (if desired); at that point recording will stop, the file will be closed and the program will then wait for signal above the threshold again. I've used it extensively with analog inputs (hense the use of a threshold rather than exactly 0). The only fly in the ointment will be whether the SPDIF input on your soundcard can be mapped in the OSS emulation layer in such a way that OSS applications can use it. Regards jonathan

18 years, 9 months

[linux-audio-dev] Cross compiling the kernel

by Lee Revell

Hello, All I am trying to do is compile an x86_64 kernel on i386 machine. It dos not work: $ ARCH=x86_64 make oldconfig <no errors> $ ARCH=x86_64 make CHK include/linux/version.h SPLIT include/linux/autoconf.h -> include/config/* CC arch/x86_64/kernel/asm-offsets.s arch/x86_64/kernel/asm-offsets.c:1: error: code model 'kernel' not supported in the 32 bit mode make[1]: *** [arch/x86_64/kernel/asm-offsets.s] Error 1 make: *** [prepare0] Error 2 What gives? Do I need a different compiler to build for x86_64? Shouldn't this Just Work? Lee

18 years, 9 months

[linux-audio-dev] Proffessionals?

by james＠dis-dot-dat.net

Hi everyone, Can anyone give me any examples of Free audio software being used by professionals? Anywhere where it performs better, or simply doesn't cost two or more body parts to use? Quick answers get bonus points. James -- "I'd crawl over an acre of 'Visual This++' and 'Integrated Development That' to get to gcc, Emacs, and gdb. Thank you." (By Vance Petree, Virginia Power)

18 years, 9 months

[linux-audio-dev] Replacement for my M-Audio Audiophile USB

by Richard Smith

After several frustrating weeks playing with my Audiophile USB I'mgoing to punt and get something else. For the archives: ALSA users interested in high-end capture should_not_ buy this device.(FWIW it sucks in windows too) - Its only a USB 1.1 device and 24bit 96k is half-duplex only.- Its (almost) S24_3BE so you have to use plughw or jackd has to bepatched to use it. I say almost because the few times I did get thecard to record something I had to reorder the byte data to get it notto sound like white noise.- As of ALSA 1.0.10 capture is broken. So I'm looking for a replacement. I bought this card because it had areally good recording specs (I just missed the half-duplex part) and Idont' need a pro-multi channel since all I'm tyring to do is recordLPs to CD. Can someone recommend a 24bit 96kHz full-duplex card that they thinkhas good enough recording specs for LP's that not going to cause me topull all my hair out trying to work with ALSA. --Richard A. Smith

18 years, 9 months

[linux-audio-dev] ANN: "Modern Computer Music and DSP Programming Tools" Workshop in Mainz/Germany

by Albert Graef

Dear LAD'ers, our Department of Music-Informatics at the Johannes Gutenberg University of Mainz is organizing a little workshop with introductory courses on Faust, SuperCollider and Q by Yann Orlarey (Grame, Lyon), Stefan Kersten (TU Berlin) and me. The workshop is sponsored by the "Interdisciplinary Working Group of Music and Arts Informatics" of the Johannes Gutenberg University, so it's free, and takes place at the Institute of Musicology (Philosophicum, Jakob-Welder-Weg 18 on the campus) on Tuesday, Dec 20th 2005, from 14:00 h to 18:00 h. If you'd like to participate, then please *please* register (just mail me at Dr.Graef(a)t-online.de or my secretary, Mrs. Felicitas Volke, at volke(a)uni-mainz.de). Registration is free but mandatory, as we can only accommodate a limited number of participants. If you register, please also indicate whether you can bring your own laptop, as the Linux PC and MIDI equipment available for the workshop is even more restricted than the amount of available chairs, coffee and Christmas cookies. ;-) More information can be found in the attached text file. Information on how to find us on the campus is available at http://www.uni-mainz.de/zentral/75.php. You can also contact me at Dr.Graef(a)t-online.de if you have any further questions. Please feel free to pass this around to your friends, colleagues and other mailing lists as you see fit. Cheers, Albert Graef -- Dr. Albert Gr"af Dept. of Music-Informatics, University of Mainz, Germany Email: Dr.Graef(a)t-online.de, ag(a)muwiinfa.geschichte.uni-mainz.de WWW: http://www.musikinformatik.uni-mainz.de/ag Workshop Modern Computer Music and DSP Programming Tools Location: Johannes Gutenberg-Universität Mainz, Philosophicum, Jakob-Welder-Weg 18, Alter Fakultätssaal (Room P 01-185) Date: Tuesday, Dec 20th 2005, from 14:00 h to 18:00 h Workshop language: English The number of participants is limited, (free) registration is required. Registration: Tel.: +49 (0)6131 3925142, Email: volke(a)uni-mainz.de Programme: 14:00 h Albert Gräf (University of Mainz): Functional Multimedia Programming with Q 15:00 h Yann Orlarey (Grame): FAUST Coffee Break 16:00 h Stefan Kersten (Technical University of Berlin): SuperCollider 18:00 h End 14:00 h Albert Gräf: Functional Multimedia Programming with Q This presentation gives a hands-on introduction to the equational programming language "Q", and some of its facilities for multimedia programming. Q can best be described as a kind of modern-style "functional scripting language." Q's multimedia library comprises interfaces to Grame's MidiShare and Faust, as well as an OSC-based SuperCollider interface, and thus provides the necessary tools to create advanced computer music applications in the context of a very-high-level, non-imperative programming language. Albert Gräf is head of the Dept. of Music-Informatics at the Institute of Musicology of the Johannes Gutenberg University Mainz. His research interests include the mathematical theory of music and advanced functional programming tools for computer music and other real-time multimedia applications. 15:00 h Yann Orlarey: FAUST FAUST (Functional AUdio STreams) is a programming language for real-time signal processing and synthesis that targets sample-level high-performance signal processing applications and audio plugins. FAUST proposes an innovative approach to signal processing that combines two programming models: functional programming and block diagram composition, in a highly structured textual syntax that can be compiled into efficient C/C++ code. The presentation will give an overview of the main features of the language and its compiler through several simple and practical examples. Composer and researcher in computer music, Yann Orlarey is currently the Scientific Director of Grame - Centre National de Création Musicale in France. His main research interests are music programming languages, with a particular focus on lambda-calculus and functional programming, and real-time distributed systems. He is the author and co-author of various musical softwares and systems including MidiShare. 16:00 h Stefan Kersten: SuperCollider SuperCollider is a real-time synthesis engine and object oriented composition language. This course introduces the architecture and the working environment on OSX and Linux and provides an introduction to basic synthesis techniques and sequencing strategies. Stefan Kersten (*1978) is currently studying communication and computer science at the Technical University of Berlin. He has ported SuperCollider to Linux and is the author of SCUM, SuperCollider's GUI module for Linux. He uses SuperCollider for most of his projects in research and music.

18 years, 9 months

[linux-audio-dev] 2.4.x patches?

by Tim Goetze

http://www.zip.com.au/~akpm/ (and schedlat.html below it) seems to have gone 404. Is anyone out there mirroring it? Thanks, Tim

18 years, 9 months

[linux-audio-dev] RELEASE jack-audio-connection-kit-0.100.7

by Taybin Rutkin

Minor bug fixes since last release. Notable changes: * Attempt to let client timeout option work again on realtime mode * Let jackd quit gracefully when USB soundcard or power cable is unplugged * Better support for US428 USB soundcard * SSE/E3DNow mixing support. Disabled by default. Enable with -- enable-dynsimd JACK is available at http://jackit.sf.net and our new website http:// jackaudio.org.

18 years, 9 months

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Linux-audio-dev December 2005