Linux-audio-dev

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3 participants
7520 discussions

by Doru-Catalin Togea

Hi! I want to programme my mic and my speakers on my FC2 box. I am running KDE. Is there a library which I should look at or is the ALSA API the way to go? Thanks, Catalin -- <<<< ================================== >>>> << We are what we repeatedly do. >> << Excellence, therefore, is not an act >> << but a habit. >> <<<< ================================== >>>>

21 years, 4 months

[linux-audio-dev] RE: MIDI groove theory

by philicorda

Dave Robillard wrote: >Hi all, > >Does anyone know of a page somewhere that explains just what (on a >developer level) MIDI "groove" is? I want to implement it in a >sequencer, but all I can find is user documentation pages with useless >information. > >Is it as simple as each note having a time offset (ie snare is early .5 >ms, hi-hate late 1ms, etc.) or something more? > >Thanks.. I think that 'groove quantise' does have quite a specific meaning : To take the timing from one midi part, and apply it to another. So, you get the midi timing from a real drummer grooving away and apply it to your beat. Then it pulls your beats to the nearest beat in the groove, rather than to the normal 16th or whatever. I seem to remember having some floppys of grooves back in the atari days and doing this with midi shaker patterns and the like, but it's been a long time....

21 years, 4 months

Re: [linux-audio-dev] Which audio lib?

by Kjetil Svalastog Matheussen

Hans Fugal: > > ALSA. If you want to tie yourself to KDE and endless misery, try artsd, > but I warned you. > Hmmm, I think the alsa api is a bit huge/complicated. I would never reccomend doing alsa directly, and I think it was a very bad advice actually. Check out portaudio, sndlib or jack instead, which provides easier interface to the soundcard than alsa, and works on top of alsa (and others). --

21 years, 4 months

[linux-audio-dev] SC3 - some help required

by Alfons Adriaensen

Hello, I trying my first steps with Supercollider. Build and install went without any probles, and I learned just enough elisp to get scel running in emacs. Now the problem seems to be that sclang and scsynth do not talk to each other. How is the sclang -> scsynth UDP link configured ? There is no option in sclang to set the port number it sends to so probably this is fixed. OTOH, scsynth requires a port number for the -u option, but what should this be ? TIA, -- Fons

21 years, 4 months

[linux-audio-dev] [ANN] jamin-0.9.0 release

by Jack O'Quin

The second stable release (0.9.0) of JAMin - the JACK Audio Mastering interface is now available for download. JAMin is a GPL-licensed, realtime mastering processor designed to bring out the detail in recorded music and provide a final layer of polish. Every effort has been made to ensure a clean, distortion-free signal path. All processing elements use linear-phase filtering to eliminate phase distortion. JAMin runs on Linux using the JACK Audio Connection Kit, a low-latency audio server. JACK connects multiple applications to a single sound device, and also share audio among themselves. We rely on other JACK applications (like ardour, ecasound, or rezound) for playback and recording. Homepage http://jamin.sourceforge.net/ Download http://prdownloads.sourceforge.net/jamin/jamin-0.9.0.tar.gz?download New dependencies (since jamin-0.8.0) swh-plugins >= 0.4.6 <http://www.plugin.org.uk> liblo >= 0.5 <http://plugin.org.uk/liblo> (optional, for OSC scene change control) All other dependencies remain unchanged, the README file has a complete list. Usage instructions http://jamin.sourceforge.net/Using_JAMin.html Changes since jamin-0.8.0 * Limiter improvements: -- uses new fast lookahead limiter (LADSPA:1913) from the swh-plugins >= 0.4.6. The old limiter had some undesirable sonic artifacts. -- changed order of final gain and limiter stages. * OSC control for scene changes: -- accepts "osc.udp://localhost:4444/jamin/scene" messages -- new `jamin-scene' command sends them -- new jamin-cont plugin (LADSPA:1912) sends scene change messages to the JAMin process via OSC. (This plugin works with ardour, ecamegapedal and applyplugin, but segfaults in jack-rack when removed.) * Increase number of scenes from 6 to 20 * GUI improvements -- extensive context-sensitive help -- color editor for highlights and text -- options pulldown menu for spectrum, crossfade, and EQ * Better bypass control -- specific bypass controls for EQ and limiter -- separate Active/Solo/Bypass for each crossover band * Global state settings saved in XML file. * Expanded Russian translation * Many bug fixes. -- joq

21 years, 4 months

[linux-audio-dev] liblo moved to sf.net

by Steve Harris

http://sourceforge.net/projects/liblo/ Theres a mailing list, liblo-devel. For those who dont know, liblo is a GPL'd implementation of the OSC protocol, which is a Remote Procedure Call language, designed for audio software. http://www.cnmat.berkeley.edu/OpenSoundControl/ I feel that liblo is now mature enough to stand on its own, so I'm moved it to sourceforge. I welcome help and advice from anyone who feels like giving it, though I do want to stop the API getting too sprawly. The current state is that I've just added UNIX domain and TCP transport support, but they are a bit rough around the edges. The current contents of the TODO list are: * Bundle support [needs NTP, argh] * Add source address to received messages * More/better regression tests * More/better example code * Normalise URI handling [needs discussion] * GTK/Qt/etc. tie code [maybe belongs in another library] * OSC pattern matching [still not sure about this one] * Rendevous/OpenWhatever (eg. howl) service discovery [low priority] * Add a JACK transport layer [maybe, low priority] * Make TCP work to spec (currently the TCP server only reads the 1st packet from every conneciton) [low priority] Cheers, Steve

21 years, 4 months

[linux-audio-dev] ardour 0.9beta19 released

by Paul Davis

(not the "feature complete" release I was planning, but the list of changes is enough to warrant a new beta number) Changes since 0.9beta18.4: * optimization flags patch from rob holland * drag-n-drop onto the editor track area where there is no track creates an appropriate track (or multiple, if appropriate) * URL-decode drag-n-dropped filenames * numlock can no longer mess up region dragging - if numlock generates whatever "Meta" is supposed to be then Ardour changes its notion of what Meta is. - if you see a warning about this in the terminal at startup, you should fix your keyboard map using xmodmap or equivalent * remove debugging output from process() tree * removed VST audiomaster callback debugging output * UI for VST plugins now offers presets * VST plugins now save and restore their state - "chunk"-using plugins are not yet supported * initial tempo and meter markers cannot be moved, even after editing * when cleaning up, prompt user about each unused playlist (options are delete, keep, cancel cleanup) * never decrease the number of panning buffers * Can't create a new session on top of an old one anymore. It didn't overwrite the old session, but you would get 2 master tracks and whatever was in the template, if any. * freezing tracks - implementation of freeze completely redone - unfreeze now works across invocations of ardour, so you can freeze, exit, restart and unfreeze. - freeze hides any insert editors, makes rec-enable and redirect display widgets insensitive and locks the new region. - freeze can be used over and over again * reworked relationship between bounce + freeze * removed "Loop Region", renamed "Loop Selected Region" as "Loop Region" because of click-selects selection model * finally made solo button in mixer strip have the same font size as other buttons * removed edit history button and associated code from editor track * removed "automation style" buttons since they did nothing * more automation line + point patches from gerard * fix a couple of valgrind complaints * region trimming now obeys ignore-snap modifier * search unused playlists in Session::playlist_by_name() * added 2-phase unserialization mechanism (Session::StateReady signal) * empty playlists assigned to correct diskstream at startup * removed some debug printfs and some dead code * make Shift_R work like Shift_L * fix crashing bug with zero-length fade ins * internationalize all editor operation strings * transport control is insensitive when there is no session * fix problems with "region->fill track" where region is at or beyond current session end * new mixer strips use the current default width setting * always save to snapshot * snapshotting changes the current snapshot to whatever was saved * delete click_io object at session close, thus allowing new sessions to load correctly * fix problem with code ordering in normalize-region operation (undo after normalize-then-cut will now do the right thing) * force selection.clear() in file selection browsers when widget selections are cleared * committed patches so that ardour now compiles with gcc-3.4. There were several patches from multiple people. So please let me know if I missed one. * patch from nick_m that fixes a problem with the editor-mixer button.

21 years, 4 months

[linux-audio-dev] Re: [linux-audio-user] jack_fst

by Dave Phillips

Greetings: CC'd to the LAD list. Some developers there are not subscribed to LAU. It's nothing personal, they appear to like us users just fine, but the added mail traffic turns into a time sink for them. :) Best, dp davidrclark(a)earthlink.net wrote: >Greetings all, > >Can anyone tell me what the REAL process for getting jack_fst installed >is? > >For fst-1.6: > >fixheaders needs to be run from the $HOME/fst-x.x/vst directory as >"../fixheaders" not from the $HOME/fst-x.x directory as advised by Dave >Phillips (http://www.djcj.org/LAU/quicktoots/toots/vst-plugins/). >Small problem... Like Dave said, this stuff isn't for rank beginners. > >mkinstalldirs, called from Makefile, was missing. I copied the one >supplied with Hydrogen. Annoying problem. > > >For jack_fst-1.2: > >fixheaders, used above, does not actually fix the headers. gcc doesn't >understand the initializer. This is what I'm not sure about: How to emulate >C++ for struct VSTFileTypes initialization. Although I've written a bunch >of code for C++ classes, I've written virtually nothing for structs in C. >Lines near 918 in aeffectx.h need to be further modified. This appears >to be yet another "audioMaster" -related problem. > >Thanks for your attention and any info. > >Regards, >Dave. > > > > >

21 years, 4 months

[linux-audio-dev] [ANN] AudioScience HPK version 2.88 released

by Eliot Blennerhassett

P..S to the previous message: Many thanks to Fred Gleason of Salem Radio Labs http://www.salemradiolabs.com/ for his assistance with this release. regards Eliot Blennerhassett AudioScience Inc.

21 years, 4 months

[linux-audio-dev] [ANN] AudioScience HPK version 2.88 released

by Eliot Blennerhassett

See the release notes and get the tarball from here http://www.audioscience.com/internet/download/linux.htm Many updates to specific card features, plus support for Linux kernel 2.6.x. Also on the same page can be found the patch for alsa 1.0.5a (I know, I'm lagging behind) regards Eliot Blennerhassett AudioScience Inc.

21 years, 4 months

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Linux-audio-dev