Linux-audio-dev August 2004

linux-audio-dev@lists.linuxaudio.org

86 participants
67 discussions

by Dave Robillard

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.. -DR-

21 years

[linux-audio-dev] Which audio lib?

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

[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

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

[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

[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, 1 month

[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, 1 month

[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, 1 month

[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, 1 month

[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, 1 month

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Linux-audio-dev August 2004