Linux-audio-dev May 2005

linux-audio-dev@lists.linuxaudio.org

90 participants
70 discussions

by Albert Graef

(Cross-posted to the Faust and Q mailing lists.) Hi all, I thought that some of you might be interested in a Faust [1] interface I created for my functional programming language Q [2]. The interface allows you to load and run Faust DSPs in Q. Those of you who attended Yann Olarey's Faust workshop and my talk about Q at LAC05 should have an idea of what I'm talking about. ;-) I think that Faust and Q really make a great combo, which allows you to do all your multimedia/DSP stuff using nothing but modern FP tools. The Q-Faust interface is currently only available in cvs, see the q-faust module at [3]. A few examples are included, such as a realtime synthesizer playable via MIDI. To build and run this stuff, you'll need the Q core system, Q-SWIG, and the q-midi and q-audio modules, all readily available from [2]. And, of course, you'll also need Faust [1]. Relevant links: [1] Faust homepage: http://faudiostream.sourceforge.net [2] Q homepage: http://q-lang.sf.net [3] Q cvs repository: http://cvs.sourceforge.net/viewcvs.py/q-lang Enjoy! :) Cheers, Albert -- 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.musikwissenschaft.uni-mainz.de/~ag

18 years, 11 months

[linux-audio-dev] Mixing signals

by Viceic Predrag

Hi all, I'm the author of Freecycle, one of the younger FOSS audio projects out there. I have a problem that may astound by it's simplicity, so I barely dare to ask for help... Freecycle provides some LADSPA functionality, and as there are lot of great but mono ladspa effects, I need a very simple way of sending stereo audio to mono input.. As I want to make my software as simple as possible for the end user, I provide some basic routing for the LADSPA giving the possibility to enter LADSPA input audio port either from the left channel, either from the right channel either from the mix of two channels. The signal is then passed through LADSPA and every 1024 frames the LADSPA control port input values are changed according to the desired automation. My problem consists in a correct way of mixing the two channels into one. I have found 4 ways of doing that: 1) sum L and R and divide by 2 : well.. 2) if L>0 and R>0 take the max, if L<0 and R<0, take the min, else add. : current implementation 3) add, and then normalize to the max after summation. The way that "feels" most correct to me is 3), but I don't like the two pass approach, as I mix the channels every 1024 frames and the send those 1024 frames to LADSPA. Of course, I would like to avoid letting the user set the gains manually... Could someone please help with this apparently simple problem? Many thanks, Predrag Viceic http://www.redsteamrecords.com/freecycle

18 years, 11 months

[linux-audio-dev] A Taste of Things to Come

by Pete Bessman

Hello Lists, As some of you may know, I'm the guy who wrote Specimen. And as a tiny fraction of that some may know (or care), Specimen hasn't seen an update in an exceedingly long time. What, exactly, is going on? The answer, dear friends, is simple: music. My introduction to the art of electronic music composition began precisely when Specimen development halted. For a few months, I fumbled around with countless little songs in an attempt to figure out how one goes about the process of coaxing the electrons in such a fashion as to result in sonic euphoria. And once I reached that point where the magnitude of my suckiness became bearable, I began writing songs. This process began in February with the assistance of an old friend, who lent his discerning ear and exceptional bass talent to the effort. And yesterday, we finished step one --- general composition. We now have 11 rough drafts that will be polished into a full length album and released next fall --- 100% Linux and OSS produced. It occurred to us that we might share some of this with all you folks out there in Internet land, both to stoke the flames of anticipation and to remind my adoring public that I have not yet shed this mortal coil. So, I am pleased to present a collection of snippets from the aforementioned tracks above, pieced together in an easy to swallow medley: http://www.gazuga.net/preliminary_beats.ogg http://www.gazuga.net/preliminary_beats.mp3 And for those interested in staying abreast of the latest and greatest, be sure to tune into The State of the Beat: http://www.gazuga.net/blog That's all for now, but don't worry --- I'll be back before you know it. Kinda like herpes, only better. Peace out, -Pete

18 years, 11 months

[linux-audio-dev] OpenGL + Audio = Bad Idea?

by Richard Spindler

Hi, I have a little Problem: I'm using OpenGL in my Program, in combination with OSS via portaudio. The Updates of the OpenGL display are triggered by the audio callback, protected using the available Methods of the GUI-Toolkit that I use (fltk). And I'm experiencing very strange effects, like: without Audio the OpenGL works fine. without the OpenGL, the audio works fine, however when used in combination, the Audio Thread (callback) stops very soon, and I have no Idea why? I used efence to discover wheater I'm screwing up memory somewhere, but I couldn't find anything. I ran valgrind, but it couldn't find something helpful either. I ran it from within gdb, but the program behaves correctly if run from gdb (heisenbug?). It even works better whenever I insert some cout's for debugging purposes. in gdb it also reports a signal like this when initializing the openGL, but I believe that might be propably unreleated. http://music.columbia.edu/pipermail/linux-audio-user/2004-October/016821.ht… BTW. I'm running an unmodified ubuntu warty. Any tips or pointers?? Thanks Richard PS.: (writing this message led me to some more experiments ;) ) I'm fairly sure that something is screwed up within thread syncronisation, because I can "fix" it by inserting a usleep() call at a certain position, but I do have a fairly hard time imagining what could be wrong. I'm considering to switch to blocking IO for Audio because, latency and stuff is not really an issue.

18 years, 11 months

[linux-audio-dev] Please test the RT rlimits patch for audio

by Jody McIntyre

The RT rlimits patch (nice-and-rt-prio-rlimits.patch) has been proposed as a solution to allow audio users to run their applications with realtime priorities. While more complicated to configure, it's a much cleaner patch than realtime-lsm and it's likely to get merged soon _if_ enough audio users test it and confirm that it works. To encourage this, I have created a wiki page containing installation instructions, links to prebuilt PAM packages, etc: http://www.steamballoon.com/wiki/Rlimits If this works for you, I am collecting success reports. Please email rlimits-success(a)steamballoon.com . If you have any problems with it, LAU is probably the best place to ask for help. Unfortunately I don't have large amounts of time to spend helping people with this, so any help requests emailed to me directly may be deleted without a reply. Sorry. Thanks for testing! Jody

18 years, 11 months

[linux-audio-dev] [EMPLOYMENT] contractor sought for linux sample playback engine

by Paul Davis

Through local contacts, I ended up meeting with a person close to Philadelphia who is starting a company to build a sample playback engine that will be used to enhance and extend the sound of an existing large scale instrument. They need a relatively simple sample playback engine: - MIDI in - up to 400 samples being played at once - all samples assumed to live in RAM - up to 50-70 channels of output - no functional GUI (simple one required at install/configuration time) The work needs doing quickly (they have existing customers who are waiting), and well (the system will have no monitor; stuff has to Just Work All The Time). If you think you might be interested in doing this, get in touch with me, estimating time to completion and cost. Available compensation is reasonable, and may be adjustable to include stock in the company. --p ps. you know who you are :)

18 years, 11 months

[linux-audio-dev] Re: [CM] [ANN] Realtime Extension documentation for SND, and snd-ls V0.9.3.0

by shreeswifty

looks nice Kjetil! OS X ?? also (OT) are you still developing/mainting Mammut? cheers~ PAtrick On Fri May 20 08:32:42 EDT 2005, Kjetil Svalastog Matheussen <k.s.matheussen(a)notam02.no> wrote: > > > > The Realtime Extension for the sound editor SND consists of two > parts: > > 1. The RT Engine - An engine for doing realtime signal > processing. > 2. The RT Compiler - A compiler for a scheme-like programming > language > to generate realtime-safe code understood > by the > RT Engine. > > Homepage: > http://www.notam02.no/arkiv/doc/snd-rt/ > > Screenshot: > http://www.notam02.no/arkiv/doc/snd-rt/screenshot.png > > > ******************************************** > > > Snd-ls v0.9.3.0 > --------------- > Released 19.5.2005 > > > About > ----- > Snd-ls is a distribution of the sound editor Snd. Its target is > people that don't know scheme very well, and don't want > to spend too much time configuring Snd. It can also serve > as a quick introduction to Snd and how it can be set up. > > > Changes 0.9.2.0 -> 0.9.3.0 > --------------------------- > -Updated SND to v7.13 from 18.5.2005. Many important changes. > -Fixed a small error in the installation script. > > > http://www.notam02.no/arkiv/src/snd/ > > > > _______________________________________________ > Cmdist mailing list > Cmdist(a)ccrma.stanford.edu > http://ccrma-mail.stanford.edu/mailman/listinfo/cmdist > > Patrick Pagano,M.F.A Digital Media Specialist Digital Worlds Institute University Of Florida (352) 294-2082

18 years, 11 months

[linux-audio-dev] [ANN] Realtime Extension documentation for SND, and snd-ls V0.9.3.0

by Kjetil Svalastog Matheussen

The Realtime Extension for the sound editor SND consists of two parts: 1. The RT Engine - An engine for doing realtime signal processing. 2. The RT Compiler - A compiler for a scheme-like programming language to generate realtime-safe code understood by the RT Engine. Homepage: http://www.notam02.no/arkiv/doc/snd-rt/ Screenshot: http://www.notam02.no/arkiv/doc/snd-rt/screenshot.png ******************************************** Snd-ls v0.9.3.0 --------------- Released 19.5.2005 About ----- Snd-ls is a distribution of the sound editor Snd. Its target is people that don't know scheme very well, and don't want to spend too much time configuring Snd. It can also serve as a quick introduction to Snd and how it can be set up. Changes 0.9.2.0 -> 0.9.3.0 --------------------------- -Updated SND to v7.13 from 18.5.2005. Many important changes. -Fixed a small error in the installation script. http://www.notam02.no/arkiv/src/snd/

18 years, 11 months

Re: [linux-audio-dev] LADSPA Issues

by Mike Rawes

--- Dave Robillard <drobilla(a)connect.carleton.ca> wrote: > Hi all, > > A while ago I started a thread about the proper way to refer to LADSPA > plugins (in save files or whatever) and the consensus was library > filename + label. > > People have been having problems with library name - different packages > seem to make different names for the libraries (prefixing blop_, for > example) so it doesn't always work. Basically I think using shared > library file name is an awful way to reference plugins for numerous > reasons. As the guilty party (author of blop) I admit that when this option was added (--program-prefix for library files) it was done in blissful ignorance of the use of library basename as an identifier, as I had assumed that the Unique ID was as claimed in ladspa.h. The purpose was to avoid name clashes with generic names such as 'sawtooth' - I ended up copying swh, and append the UID (so it does have a use after all :) to the filenames. I'd meant to remove the --program-prefix option from configure, but forgot. > So why wasn't the unique ID the thing to use? There is a unique plugin > ID in LADSPA, if not for this then for what reason? Going by what is said on ladspa.org, I think that it was originally intended to be the way to refer to plugins, and changed as development progressed. IIRC, the UID is still required to lookup metadata with liblrdf, but this may have changed since I last looked. > In a similar vein, I really think the current system for LADSPA > distribution sucks - big tarballs from various devs containing heaps of > completely unrelated plugins. A centralized site where plugins can be > submitted on their own (or in related groups) would be a great thing, > IMO, and would make it easy to verify that unique IDs are in fact unique > to solve the above problem. I don't think you'll get very far arguing the case for UIDs - the arguments against were pretty clear in the previous discussion and summarised by Chris Cannam in his response. I think we're stuck with basename+label until a better scheme can be implemented, possibly for LADSPA 2. Regarding unrelated plugins in one library, I personally don't have a problem with this as liblrdf does a fine job of categorising plugins where it's really needed (in the host). > Right now if a developer wants to make just one random plugin, they > don't really have a sane way of getting it out there. I agree here. The best option short of creating a new library distribution is to get your plugin 'adopted' into an existing library. I'm certainly willing to merge any homeless plugins into blop. > I'm willing to full-time maintain the site, but I don't really have the > hosting/abilities to create it. What do the other plugin authors think > about this? I'm all for it. Maybe liase with Richard Furse to update the ladspa.org site itself? There's already a list of links there so all that is really needed is to add details for maintainers willing to adopt plugins, with appropriate provisions (kind of plugin, language, build system and so on)? - Mike ___________________________________________________________ Yahoo! Messenger - want a free and easy way to contact your friends online? http://uk.messenger.yahoo.com

18 years, 11 months

[linux-audio-dev] Re: Minimum reasonable latency

by Juhana Sadeharju

>From: Alfons Adriaensen <fons.adriaensen(a)alcatel.be> > >IFF JACK could be modified to allow this then probably you wouldn't >need M, its function would be taken care of by jackd. That was the idea. >The problem with such a scheme could be that the load is not spread >evenly, so it would be necessary to give those clients that use >longer buffers a lower priority. There may be other troule hidden >somewhere... Yes. The priority should depend on the buffer size. Longer buffers would run with non-soft-RT schdulers. Juhana -- http://music.columbia.edu/mailman/listinfo/linux-graphics-dev for developers of open source graphics software

18 years, 11 months

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Linux-audio-dev May 2005