Linux-audio-dev

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7513 discussions

[linux-audio-dev] help with multiple control ports

by Artemio

Hello all! I have a problem with creating a plugin with more than one control port. In the attachment you'll find "booster-simple.c" which has one "Gain" port and my attempt to add a second port in "booster.c". For some reason the latter cannot be used - the sources seem to be correct, it compiles fine, but then if I test with "applyplugin" it segfaults and other hosts complain about a malloc() error and hang. Debugging shows that ports are not properly connected: [connectPortToBooster] setting BOOSTER_INPUT1 [connectPortToBooster] setting BOOSTER_OUTPUT1 [runMonoBooster] applyplugin: booster.c:112: runMonoBooster: Assertion `psBooster->m_pfGainValue != 0' failed. ...but I cannot investigate further as my C is very basic. I would really appreciate any help. I based my plugin on the amp.c from the SDK, and I see that in example plugins like simple delay (where there is more than one control port) memory allocation is done differently, but I just don't know how to properly apply these techniques to my plugin... Thanks very much! Artemiy. P.S. I've sent a message with a 50Kb attachment and it's in moderation queue, but these sources are newer.

20 years, 2 months

[linux-audio-dev] All things LADSPA, your input please

by Tim Goetze

Greetings, you may have seen the recent mail on this list looking for an author doing a LADSPA article in a german Linux magazine; I've decided to step in. Now, while I do have some knowledge of the technical nature of LADSPA I can't truthfully say I know all the plugins and hosts out there as well as they deserve. So, I'd love to hear from you: what do you write or use (hosts and plugins, realtime or offline, a shameless plug for your efforts is welcome too!), any issues you take with it, LADSPA vs. VST, the DSSI expansion etc etc; in short, anything you find worth noting is appreciated. Thanks, Tim

20 years, 2 months

[linux-audio-dev] "Call for an Online-Demonstration against Software Patents"

by Jens Radloff

Hi, Let me inform you about the online demonstration against the patentability of software (=software patens) which the Commission of the European Union wants to introduce. As far as I understand the process of passing the appropriate bill in the institutions of the EU, the patentability of software could be possible from July 2005 on. There is a site, run by among others Attac Germany, where you can protest against this, by uploading a picture of yourself: http://demo.stoppt-softwarepatente.de/index.php?content=demo&lang=GB "So far 3883 people have demonstrated here. The more people participate, the clearer a joint picture of the protest will be shown. Short before the decision in the European Parliament we will print this picture on a huge banner and show it in front of the European Parliament in Strasbourg. This way you can also be a part of the international demonstration." Regards, Jens ___________________________________________________________ Gesendet von Yahoo! Mail - Jetzt mit 1GB Speicher kostenlos - Hier anmelden: http://mail.yahoo.de

20 years, 2 months

Re: [linux-audio-dev] LilyPond version 2.6 available - Music Notationfor Everyone.

by Chris Cannam

Albert Graef writes: > that's great news. Certainly is! > Has Rosegarden been updated to a newer Lily yet? Last > time I looked, they were still generating 2.0 code. :( Rosegarden's been able to generate 2.2 code for ages, and 2.4 for a while - though I can't quite remember whether that was in the 1.0 release or not. We're partly but not completely up to date with 2.6. I might well look at that this week, unless anyone else offers. Our Lilypond output is not terribly well structured for any version, so a bit of an overhaul might be nice too. Chris

20 years, 2 months

[linux-audio-dev] LilyPond version 2.6 available - Music Notation for Everyone.

by Jan Nieuwenhuizen

LilyPond is a free/open source package to create beautiful music notation. With version 2.6, LilyPond is now truly for everyone. - For every platform LilyPond now installs in a snap on Windows, MacOS X, and any version of Linux/x86. Get up and running in minutes! - For every language Pango text formatting lets you print Unicode lyrics in your favorite script and font. - For every application Create SVG files, and edit them in Inkscape. In addition, version 2.6 adds support for - staves starting anywhere on the page - solfa notation - arrowed lines - better auto-beaming - circled text - string-number notation - better ledger line formatting - score separators - cleaner syntax for text markup - pagebreaks around titles - stemlets on beams - easier titles customization - direct PostScript or SVG output - (te)TeX no longer necessary - revised manual - website now translated into Dutch and French Grab it at http://lilypond.org A big thank-you goes out to our contributors, translators, website translators and bug-hunters: CONTRIBUTORS Andreas Scherer, Arno Waschk, Bertalan Fodor, Carl Sorensen, Christian Hitz, David Jedlinsky, Erlend Aasland, Heikki Junes, John Williams, Jonatan Liljedahl, Jürgen Reuter, Mats Bengtsson, Matthias Neeracher, Mathieu Giraud, Nicolas Sceaux, Pal Benko, Sebastiano Vigna, Tatsuya Ono, Vicente Solsona Della, Werner Lemberg, and Yuval Harel. TRANSLATORS Abel Cheung, John Mandereau, Olcay Yıldırım, Roland Stigge and Steven Michael Murphy WEBSITE TRANSLATORS Gauvain Pocentek, Jean-Charles Malahieude, John Mandereau, and Tineke de Munnik. SPONSORS Bertalan Fodor, Chris Sawer, Gunther Strube, Hans Forbrich, Jonathan Walther, Marcus Macauley, and Steve Doonan. BUG HUNTERS/SUGGESTIONS Alexandre Beneteau, Andreas Scherer, Anthony W. Youngman, Antti Kaihola, Arjan Bos, David Bobroff, Bernard Hurley, Bruce Fairchild, Bruce McIntyre, Bruce Fairchild, Daniel Johnson, David Rogers, Dylan Nicholson, Ed Jackson, Erik Ronström, Fernando Pablo Lopez-Lezcano, Gilles Sadowski, Jack O'Quin, Jeff Smith, Johannes Schindelin, John Mandereau, Jose Miguel Pasini, Josiah Boothby, Jürgen Reuter, Karl Hammar, Laura Conrad, Olivier Guéry, Paul Scott, Richard Schoeller, Rob Platt, Roman Stöckl-Schmidt, Russ Jorgensen, Simon Bailey, Stephen McCarthy, Sven Axelsson, Tapio Tuovila, Tom Cato Amundsen, Will Oram, and Wolfgang Hoffmann. Happy music printing, The LilyPond development team, Han-Wen Nienhuys & Jan Nieuwenhuizen Core development Graham Percival Documentation Editor Erik Sandberg Bug Meister Pedro Kroeger Build Meister -- Jan Nieuwenhuizen <janneke(a)gnu.org> | GNU LilyPond - The music typesetter http://www.xs4all.nl/~jantien | http://www.lilypond.org

20 years, 2 months

Re: [linux-audio-dev] FM, Phase Modulation and The Wiki

by julien2

Le Mardi 28 Juin 2005 16:26, Jens M Andreasen a écrit : > On Tue, 2005-06-28 at 16:14 +0200, julien2 wrote: > > there is two major french institute : > > www.grame.fr > > www.ircam.fr > > cheers > > Thanx! > > IRCAM is the intitute that presented phase distortion synthesis wayback > in a universe unoccuppied by patent lawyers. > > There should be a person and/or a group of people having done the > research and discoveries. > > Anybody recall? have a look at http://articles.ircam.fr, you will find the author the group in question is the analysis/synthesis team jul > > > /ja > > > Le Mardi 28 Juin 2005 15:18, Jens M Andreasen a écrit : > > > On Tue, 2005-06-28 at 14:14 +0200, Jens M Andreasen wrote: > > > > I am currently looking [in wikipedia] at the Casio section near the > > > > end (regarding phase distortion.) Wasn't this technique developed way > > > > back in the 60's by french academics? > > > > > > Let me refrase that to: > > > > > > I am almost certain that this technique was developed at the "french > > > national institute of electronic music" (or some such?) > > > > > > Question: > > > What is the actual name of that institute, and can anybody cunning in > > > french help me search in their wayback archives? > > > > > > /ja

20 years, 2 months

[linux-audio-dev] Q-Faust 1.0 and QFSynth 1.0 released

by Albert Graef

Hi all, I've just released my Faust module for the Q programming language. A realtime synth application based on this module, QFSynth, is also available. Faust (http://faudiostream.sf.net) is Grame's functional DSP programming language. Q (http://q-lang.sf.net) is a general-purpose functional programming language with an extensive collection of multimedia-related modules. If you're a developer interested in modern FP tools for multimedia and DSP programming, you should definitely take a look. ;-) 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

20 years, 2 months

[linux-audio-dev] OpenAl and dmix

by Brian Black

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 hi, This might be a a newbie question, but i have openal installed and i am running into problems. i have also posted to openal-dev list about my concerns. I have created an ~/.asoundrc file for help with dmix, because i have had problems playing multiple sound programs at the same time. This .asoundrc file helped me use xmms and amarok at the same time. 1. My problem is when i am trying to run an openal program when i am using one of the other sound programs, the program ran last will block untill the first program that i started had finished. This seems strange to me that some programs will "dmix" and some will not. Any ideas on this? Also, I am thinking about developing and audio app using the openal api, and i had some questions that maybe someone could clear up for me. 1. I am assuming this api wraps around alsa or oss, i am going to be using an RME hammerfall dsp and i need to attach some inputs to some outputs to allow the input from a mic to go to the output which will go to their headset, (low latency Monitoring). How is this done, as i have not found any docs talking about how to send sounds out certain channels? 2. Some examples i have found do not compile because they are including the file #include <strstring.h> , which i do not have. (i am running gentoo 2005.0, kernel-2.6.11)? is this a known issue? thanx in advanced bj -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFCwEfQI9Y/1VIS+jgRAqDRAKCGQNQFa9cJ55qFHO4hYnVfJzsNVgCeJQNb 9aSjBYkHBeLaagkjI9h65BQ= =98ga -----END PGP SIGNATURE-----

20 years, 2 months

[linux-audio-dev] jack_convolve-0.0.10, libconvolve-0.0.3 released

by Florian Schmidt

Hi, i added an experimental feature to libconvolve/jack_convolve which might be able to preserve some of those precious cpu cycles from being burnt. jack_convolve now has commandline switches --min_bin=bin_no and --max_bin=bin_no which can be used to specify which bins of the fourier transformed signal and response to multiply. The range for both is always 0..periodsize+1, but min_bin must always be < than max_bin.. An example: assuming jack periodsize of 2048 and samplerate of 48khz: jack_convolve response_file.wav --max_bin=1200 This leaves out the top 849 bins out of the multiplication and thus reduces cpu load by ca. 2/5. The cost is that the high frequency spectrum is cut off. And it's not even a really clean cutoff which jack_convolve response_file.wav --max_bin=40 would show (you get crackles due to edge effects, etc). This feature is probably most useful to get some preview with degraded quality but less cpu consumption. Although at some samplerates it even makes sense. Assume a samplerate of 96khz, then there's quite a bit of signal which doesn't need to be processed since it's far out of the range of human perception. the min_bin parameter is not really useful and only included for completeness sake :) It cuts out low frequency bins out of the equation. Due to the crackle effects when throwing away bins that are audible one should not use these settings for filtering of the convolution in the audible frequency range.. In one of the previous releases (which went unannounced i think) i also added a gain argument which can be used to boost the level of the convolution output. Example: jack_convolve response_file.wav --gain=4.0 to get a 300% increase in level.. Regards, Flo P.S.: I am also currently working on a qt app which uses libconvolve, but it might still take a while until initial release (due to my time constraints induced by studying). Preliminary screenshot here: http://www.affenbande.org/~tapas/wordpress/?page_id=27 Help would, of course, be appreciated :) Anyone know of waveform or vu meter widgets for qt which are reusable? Flo -- Palimm Palimm! http://affenbande.org/~tapas/

20 years, 2 months

Re: [linux-audio-dev] jack_convolve-0.0.10, libconvolve-0.0.3 released

by Ben Loftis

> (*) Recent experiments by prof. Angelo Farina (Univ. of Parma, Italy) > suggest strongly that when the DA conversion is done properly, there is > no audible difference between a sample rate of 48 kHz and any higher value. > OTOH, he only found one type of DAC that was good enough in order to be > completely free of audible artefacts at 48 kHz (by Apogee, and they are > quite expensive). > Is there an online publication of Mr Farina's findings? Google is not turning up anything definitive for me. -Ben

20 years, 2 months

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Linux-audio-dev