Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

3 participants
7536 discussions

by Gordon JC Pearce

For a while I've been using the sirlabs.de vocoder plugin: http://www.sirlab.de/linux/descr_vocoder.html But it has some problems, notably expecting two inputs but only offering one output. There was a patch on linuxrockstar which gave it two outputs, but something seemed a bit lacking, still. An evening of hacking and here we have it: http://www.gjcp.net/~gordonjcp/vocoder.tar.bz2 It's hard-set to eight bands, all with the same level. There's a bandspread control to alter the frequency spacing of the partials, and a width control to vary the stereo effect where odd bands go to the left output and even bands go to the right output. Give it a go and let me know how you get on. Gordon

17 years, 7 months

[LAD] [ANN] jack_capture V0.9.17

by Kjetil S. Matheussen

jack_capture ============ jack_capture is a program for recording soundfiles with jack. It's default operation is to capture whatever sound is going out to your speakers into a file, but it can do a number of other operations as well. Normal URL: http://www.notam02.no/arkiv/src/?M=D Note that the above link doesn't seem to currently work because of something which seems to be www upgrade at Notam. But this link does seem to work for now: http://old.notam02.no/arkiv/src/?M=D 0.9.10 -> 0.9.17: ----------------- *Made sure the process thread won't continue sending data when jack_capture is told to quit. This led to a race conditions when recording too many channels at once (which was 256 channels on my machine). *Added the jack_capture_gui script, based on code by Svend-Erik Kjær Madsen's. *Replaced sh by bash to make it work in ubuntu. *Fixed exact port name match and gen_setformat. Thanks to Gabriel J.L. Beckers for reporting the bugs. *Increased the time the console meter is displayed in red from 50ms to 2000ms. (when the recorded jack value is equal to or higher than 1.0.) *Added the -dBr argument to specify reference level when using the console dB meter. *Added the -mr argument to specify reference level when using the meterbridge meter. *Added peak indicators to the console meter. Code to do so taken from meterbridge by Steve Harris. *Updated --help and README with the new options. *Added option "-mt" to change meterbridge type. Current valid options are vu (default), ppm, dpm, jf or sco. It's not necesarry to specify "-mb" if using "-mt". *Added option "-dB" to get a dB meter for the console meter. *Added examples how to record ogg and mp3 files using the -ws option. *Decreased default buffersize from 60 to 20 seconds. (I even had trouble provocing underruns using a minimal 0.05 seconds long buffer, so 60 seconds was obviously overkill) *Fixed message and error printing to stderr when vu meter is running.

17 years, 7 months

Re: [LAD] Host 128.173.232.121 is unreachable via http or smtp ports

by Ivica Ico Bukvic

Thanks Ron for the info, we are already aware of this issue and are working on it as fast as we can. As unlikely as it sounds, it appears we've had a major thunderstorm that has knocked out several buildings on campus, including the building hosting our server. Unfortunately, I've been out of the country and unaware of this occurrence until it was brought to my attention yesterday. Let me say that this is a very unusual and very much unexpected occurrence. That being said, based on this unfortunate development we will look to invest into better UPS asap to minimize any potential downtime in the future. As of 4:58PM the server should be back up and running. Best wishes, Ivica Ico Bukvic, D.M.A. Composition, Music Technology, CCTAD CHCI, CS, and Art (by courtesy) Director, DISIS Interactive Sound & Intermedia Studio Virginia Tech Dept. of Music - 0240 Blacksburg, VA 24061 (540) 231-6139 (540) 231-5034 (fax) ico(a)vt.edu http://www.music.vt.edu/faculty/bukvic/ > -----Original Message----- > From: Ron Fox [mailto:rfox@aloha.com] > Sent: Tuesday, July 01, 2008 9:08 AM > To: ico(a)vt.edu > Subject: Host 128.173.232.121 is unreachable via http or smtp ports > > Aloha Dr. Bukvic, > > I've been trying to reach the www.linuxaudio.org and > lists.linuxaudio.org websites for a few hours without success. I'm > sending this just in case you aren't aware of the situation. > > Mahalo, > Ron Fox

17 years, 7 months

[LAD] libsndfile-1.0.18pre7 patch

by Fons Adriaensen

Libsndfile-1.0.18pre7 has some regression w.r.t. earlier prereleases. - When writing a WAVEX it always writes the Ambisonic GUID, and sets the channel mask to 0. Since it can also read this and you can't get the channel mask value, this will go unnoticed until you use some other lib to read the file. - When reading it always tells you the file is not Ambisionic. The attached patch will fix this AFAICS. Ciao, -- FA Laboratorio di Acustica ed Elettroacustica Parma, Italia Wie der Mond heute Nacht aussieht ! Ist es nicht ein seltsames Bild ?

17 years, 8 months

[LAD] hosts for LV2 synths

by Pau Arumí

Hi all, I want to try LV2 synths and try developing new ones. The plugins API and compilation seems all pretty clear, but I don't know what are the existing host options. I was about to try zynjacku with jack-keyboard. Is it compatible with any LV2 synth? Are there sequencers connectable to zynjacku? (that is: jack-midi enabled?) http://home.gna.org/zynjacku/ http://pin.if.uz.zgora.pl/~trasz/jack-keyboard/ Thanks! Pau

17 years, 8 months

[LAD] Bit OT: cleaning code for gcc4

by Julien Claassen

Hello everyone! I'm just working on some older code, trying to clean it up for GCC4.2.3. I get a lot of warnings like this: CuSE.cxx:321: warning: deprecated conversion from string constant to 'char*' Not this line but still same kind of error: char* choices[] = {" [ ] ", " [*] "}; The line 321 WARNING was using the defined choices and another char* array used in a newCDKSelection call. How can I clean this up? Thanks and kindest regards Julien -------- Music was my first love and it will be my last (John Miles) ======== FIND MY WEB-PROJECT AT: ======== http://ltsb.sourceforge.net the Linux TextBased Studio guide ======= AND MY PERSONAL PAGES AT: ======= http://www.juliencoder.de

17 years, 8 months

[LAD] Linux Plumbers Conference CFP

by Lennart Poettering

Heya! Anyone doing (audio) infrastructure work on Linux? If so, please make sure to join us at the Linux Plumbers Conference in Portland/Oregon in September. I will be doing an Audio track there and would like to make sure that everyone who does audio infrastructure work will submit a paper, or at least attend, or maybe just know about this conference! It's a conference about the kernel-userspace interfacing, about the lower levels in our software stack. For further details, checkout the conference website: http://www.linuxplumbersconf.org/ Or this blog story of mine: http://0pointer.de/blog/projects/plumbersconf.html Spread the word! Thanks, Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

17 years, 8 months

[LAD] Sources of Paul "mda" Kellett's plugins have been released

by Frank Neumann

Hi list, not sure if this was mentioned here before (I hardly find the time to read the LA* lists regularly these days), but someone from IRC let me know that the VST/VSTi plugin suite by Paul Kellett has been released by its author as source, see here: http://sourceforge.net/project/showfiles.php?group_id=230958 I don't have a Windows-based audio system so I cannot judge about the quality of his plugins, but I am pretty confident (from what I know and read about his work) that these could be pretty fine, perhaps an interesting alternative or addendum to, say, the swh-plugins. Should be a cornucopia for interested LADSPA/LV2 porters, anyway :-). As it seems, there is already a binary of a ported package at http://linux-vst.com/download/mda_linux.tar.gz No idea who managed to convince Paul to do this, but big thanks to both of you for arranging this! This should be a quite valuable resource, for DSP coders and finally also for users. Regards, Frank

17 years, 8 months

Re: [LAD] starting out with tux audio

by AlgoMantra

[[[again, meant for the list. Dunno why Gmail is defaulting weirdly....]]] On 6/18/08, AlgoMantra <algomantra(a)gmail.com> wrote: > > Do you have any midi-keyboard around? Because stdio won't tell you that >> a key has been released, only that it has been hit. > > > Interesting!!! It'll be fun to find a way around this. > > > This limits the "toy >> Casio experience" somewhat ... There is of course a workaround - given >> that you have root permission, you could read the keyboard >> from /dev/input/ instead. This is unfortunately a bit like having a >> "keyboard sniffer" installed, which may not be ideal in your >> environment. Next step up the ladder I think is fullblown X ... >> >> A virtual midi-keyboard would solve that, perhaps vkeybd? > > > Just for myself, I would love to play with a lot of different external > HIDs actually..I'm planning to code for playing C code music with > one of those old Nintendo joysticks modded for USB. It's quite > cheap here (around $8 USD). > > But the class I teach coding barely knows how to TYPE!!!! Grrrr.... > > > -- >> ------- -.- >> 1/f ))) --. >> ------- ... >> http://www.algomantra.com > > -- ------- -.- 1/f ))) --. ------- ... http://www.algomantra.com

17 years, 8 months

[LAD] starting out with tux audio

by AlgoMantra

I'm a newbie to audio in linux. I can code in C, and I know that if you $cat /dev/dsp you can see your microphone feeding weird shit into this sound device. I've also read that this device is encoded as 8-bit unsigned, and I know the binary math of that format. I'm currently coding on Ubuntu Hardy. I'm looking for some helpful tips on two simple processes, for which I'll be very grateful: 1. What are these weird "{{PP--"characters in /dev/dsp and how do I convert them to some numerical form that I can understand and play with? Are they ASCII representations of 8-bit unsigned numbers? How do I convert this to samples? (I know a "little" DSP theory, but kindly assume that I'm not very good). 2. I'd like to feed a pure sine wave to /dev/audio, and then another wave and mix them together to hear some beats. Basically, I want to try out raw additive synthesis using C and feeding cool arrays to either /dev/audio or /dev/dsp. Any help would be appreciated, and as for karma, you know you're only going to profit. ta-da, fadereu ------- -.- 1/f ))) --. ------- ... http://www.algomantra.com

17 years, 8 months

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Linux-audio-dev