Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

7513 discussions

by Steve Harris

OK, I've read the more thoroughly now. On page 23 you have a list of properties (ie. metadata), some of these should be per-port, and the generic ones (name, authors etc.) should probably follow the qualified dublin core standard. http://dublincore.org/ It makes sense to me that versions should follow the library versioning convention, so that hosts could do substitution where appropriate. The denormal numbers stuff seems a bit low level, I'm not away of any operations you can do on denormal numbers without incurring penalty (maybe delay with no gain, but thats stetching it a bit). Also DC offset, shouldn't the plugin be expected to kill its own DC offset if its desired? There is a standard for describing this kind of structural metadata as well as classifications, but I've gone on about it a lot before (and implemnented support for LADSPA), so I'm not going to mention it here again or the LAD regulars will lynch me :) I wont talk about the VVID voice allocation scheme, because I think David can describe it better. Overall PTAF is supprisingly similar to XAP (which is encouraging), there are just some differences in emphasis. Us LAD people tend to be simplecity freaks. I think its one of the really good things about LADSPA. - Steve

22 years, 7 months

[linux-audio-dev] [ANN] Bandlimited LADSPA Oscillator Plugins ('BLOP') v0.2.6

by Mike Rawes

BLOP is a set of LADSPA plugins - after way too long, it's up to v0.2.6 Website: http://blop.sf.net This release includes: * Full RDF metadata, for use with liblrdf * Bandlimited oscillators (no aliasing noise) Sawtooth Square Variable width pulse Variable slope triangle Improved performance and quality (fixed some stupid mistakes) over v0.2.5 4 Pole Moog-type resonant filter Tuned, stable LADSPA version of this filter: http://musicdsp.org/archive.php?classid=3#26 Two ADSRs Single gate with variable threshold Gate + Trigger variant (New since v0.2.5) Analogue-Style Step Sequencer (New since v0.2.5) Random wave generator, amplifier, 1V/Octave->Hz convertor ('fmod') and a few other useful things. Enjoy, Mike __________________________________________________ Do You Yahoo!? Everything you'll ever need on one web page from News and Sport to Email and Music Charts http://uk.my.yahoo.com

22 years, 7 months

[linux-audio-dev] watchdog timeout on realtime operation

by Son of Zev

Hi On loading MusE with the -R option I get this error after a few seconds allan@littlewolf2 allan]$ muse -R Loading required GL library /usr/X11R6/lib/libGL.so.1.2 no locale <muse_en_GB>/</usr/share/muse/locale> ALSA port add: <MIDI 0-0>, 64:0 flags 3 0x7f Track: unknown tag <record> at line 236 ReadEvent: warning: event not in part: 0 - 12288 -12288, discarded WatchDog: fatal error, realtime task timeout (12326,0-3) - stopping all services Any suggestions? DO I just need to recompile a kernel without the software watchdog or is it something else? cheers Allan Klinbail

22 years, 7 months

[linux-audio-dev] New Tkeca website!

by Luis Pablo Gasparotto

Hi, I've uploaded the tkeca website. Take a look of this: http://tkeca.sourceforge.net I know, I know ... I'm not a graphic designer. Regards, Luis Pablo Gasparotto Ahora podés usar Yahoo! Messenger desde tu celular. Aprendé cómo hacerlo en Yahoo! Móvil: http://ar.mobile.yahoo.com/sms.html

22 years, 7 months

[linux-audio-dev] [ANN] JACK Rack 1.2.0

by Bob Ham

Hi all, No response to the beta testing request, so I'll have to subject you all to a likely hairy release :) Arbitrary channels are the biggest thing. Also, previous save files will no longer work as the save files use XML now. Sorry, but it's better to change it now instead of later as changing the xml format will create less of a disturbance. * can now have arbitrary channel numbers. you can use any plugin that has equal numbers of input and output channels where they divide exactly into the number of rack channels. eg, you could have a 4 channel rack with 2 stereo plugins in one slot and 4 mono plugins in another. obviously, you can also have mono racks now. use the -c command line option to specify the number of channels on startup. * now, when you lock a group of controls, only one widget will be shown. click on a control while holding down the CTRL key to make it the one that remains visible. * the PID is now used in the jack client name by default. you can change it to the previous behaviour (of using just "jack_rack") with the -n option. * save files now use an XML format (which is incompatible with the previous format, sorry) * port connecting works now, -i to connect inputs, -o to connect outputs * quite a bit of code factoring and cleanups and stuff http://pkl.net/~node/jack-rack.html Bob -- If you're happy and you know it, bomb iraq

22 years, 7 months

[linux-audio-dev] Question regarding Usb-midi and Midisport 2x2

by Ivica Bukvic

Hi all, a small question regarding getting this thing to work. I'd appreciate any help I can get. I am running MDK 9.0 and am trying to get Midisport 2x2 to hotplug. The problem is that if I use the ezusbmidi script (obviously coupled with the right firmware and hotplug stuff) the /var/log/messages definitely notes its successful recognition of the device and announces the last line echo that the firmware has been installed onto the device /proc/bus/usb/002/021 (or whatever number it gets assigned), yet the midisport remains dead. I understand that the ezusbmidi script uses fxload to upload the firmware, but according to the syntax of the ezusbmidi script it sends the command in the following format: Fxload -I /path/to/firmware But this is not sufficient when you try just fxload by itself, it complains that it is missing the destination device (-D flag). So subsequently I am guessing that this is the stepping stone but then how is everyone else getting this thing to work. OTOH, when I try ezusb2131 it works flawlessly (manually) but there is no automation script for it, so I know that everything loads properly, it is just that my fxload for some reason is not working properly. I tested my midi after using ezusb2131 and it does work (/dev/midi1). Now, I hacked the script for the ezusbmidi and it does hotplug, but it is an ugly hack and I am wondering what is wrong with this picture? Any help is greatly appreciated! Sincerely, Ivica Ico Bukvic

22 years, 7 months

[linux-audio-dev] Addition to the Usb-midi problem (Midisport 2x2)

by Ivica Bukvic

Forgot to mention that I blacklisted usb-midi and audio drivers in the /etc/hotplug/blacklist. The computer tried also to load snd-usb-audio (why?), so I blacklisted that one as well. Btw, using Alsa 9.0.rc5. Finally, even though the /var/log/messages mentions creating two pairs I/O ports (since it is Midisport 2x2) I only have one /dev/midi1 device hooked up I guess to the first 16 channels. Where are the other 16? Any help is appreciated! Ico

22 years, 7 months

[linux-audio-dev] [admin] LAD website broken...

by Joern Nettingsmeier

hello * ! after what looks like a server upgrade (if you can call 2.0.40 an upgrade), the LAD homepage is broken. (yet again.) i hope root@localhost will be able to fix it, and maybe he will set the server admin mail address properly while he's at it... sorry for any inconvenience. the archives can be reached directly at http://eca.cx/lad, http://eca.cx/lau, and http://eca.cx.laa . the mailman web interface is available at http://music.columbia.edu/mailman/listinfo . best, jörn -- Jörn Nettingsmeier Kurfürstenstr 49, 45138 Essen, Germany http://spunk.dnsalias.org (my server) http://www.linuxdj.com/audio/lad/ (Linux Audio Developers)

22 years, 7 months

[linux-audio-dev] Remove from list

by David Seymour

David Seymour Benchmark Media Systems 800-262-4675 www.benchmarkmedia.com -----Original Message----- From: David Olofson [mailto:david@olofson.net] Sent: Monday, February 03, 2003 10:26 AM To: linux-audio-dev(a)music.columbia.edu Subject: Re: [linux-audio-dev] Re: linux-audio-dev digest, Vol 1 #312 - 8 msgs On Monday 03 February 2003 15.36, Ben Loftis wrote: > >> > * The "save state chunk" call seems cool, but what's > >> > the point, really? > >> > >> This could be useful for state storing in hosts, but how many > >> would use it? I'd certainly be reluctant to implemnt it. > > > >Right... I'm not even sure what state we're talking about here, > > but I *assume* it's basically all you need to continue playing > > from where you saved the state. Freezing reverb tails and stuff? > > Sounds fun but pretty useless... Might help some very special > > apps - but then, will it be sufficient? > > If you want to restart playback at an arbitrary point and be > sample-accurate, you must have this, right? Nope. Sample accuracy has nothing to do with it. You'd just start playing and make sure there are events to set the transport at the right position before the first sample frame. (Or if you have the plugins running all the time, you just start the sequencer, period.) > I think it's a good > idea (I suggested it myself a while back). The host can cache > these at regular intervals (and at the user's locate points) so you > can start playback and the host only has to preroll from the most > recent state chunk. Implementation is easy if you design for it... > just stream all your internal variables into a buffer. So, we're basically talking about a brute-force solution to the classic "make MIDI behave as recorded audio" problem? It could work in theory, but what about the bandwidth required for some plugins to do this correctly? You'll have to drop delay buffers and the like, and then it's not a complete solution anyway. Could work pretty well for synths though, and that's probably the interesting part. //David Olofson - Programmer, Composer, Open Source Advocate .- The Return of Audiality! --------------------------------. | Free/Open Source Audio Engine for use in Games or Studio. | | RT and off-line synth. Scripting. Sample accurate timing. | `---------------------------> http://olofson.net/audiality -' --- http://olofson.net --- http://www.reologica.se ---

22 years, 7 months

[linux-audio-dev] Re: linux-audio-dev digest, Vol 1 #312 - 8 msgs

by Ben Loftis

> >> > * The "save state chunk" call seems cool, but what's >> > the point, really? >> >> This could be useful for state storing in hosts, but how many would >> use it? I'd certainly be reluctant to implemnt it. > >Right... I'm not even sure what state we're talking about here, but I >*assume* it's basically all you need to continue playing from where >you saved the state. Freezing reverb tails and stuff? Sounds fun but >pretty useless... Might help some very special apps - but then, will >it be sufficient? If you want to restart playback at an arbitrary point and be sample-accurate, you must have this, right? I think it's a good idea (I suggested it myself a while back). The host can cache these at regular intervals (and at the user's locate points) so you can start playback and the host only has to preroll from the most recent state chunk. Implementation is easy if you design for it... just stream all your internal variables into a buffer. -Ben

22 years, 7 months

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

Linux-audio-dev