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-
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. >>
<<<< ================================== >>>>
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....
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).
--
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
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
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
(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.
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.
>
>
>
>
>
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.