> the big problem is that writing even a simple WM is not
> trivial. still, i would guess that taking twm and hacking it to
> enforce window sizes and position can't be that hard.
twm would be an awful choice -- the code's completely unreadable
and surprisingly huge. There are plenty of ultra-minimal window
managers around these days that have far better code. Practically
anything written since 9wm (a pivotal app in the window manager
world). Hacking on one of those need not be …
[View More]hard at all.
http://www.plig.org/xwinman/others.html
The newer ones are generally nearer the top.
I wrote wm2 and wmx which have far too particular a look for the
sort of thing you want, but the code's not _too_ bad and I'd be
happy to answer questions (though I haven't the time to work on
any actual coding of this sort). They're in C++ with BSD-style
licensing. They're getting on a bit now though and there are
doubtless easier choices even than wm2 these days.
http://www.all-day-breakfast.com/wm2/
Chris
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I use my PC as a multitrack-recorder and the only program I miss from windows
OS is the fruityloops. I tried to ask from fruitydevelopers about linux-support
and the answer I get was:
"(I wrote)do you think that linux users won´t pay for a good program? is
it time, do you have hands full of work in windows versions?
(reflex wrote)These two are probably among the main reasons. Also, you'll
have to admit that Linux isn't as mature a platform as Windows yet."
I´m not a programmer, but If I …
[View More]were, I´d try to do some co-operation with
fruity-developers and do an linux-support to the fruityloops. ´cause it´s
only the question of time&money. Is there anyone who could do something about
this?
or is there any program on linux already which works like fruity?
you can check our conversation on fruityloops-site(click the "linux, why
not?")
http://forum.e-officedirect.com/forum.exe?forumname=Fruityloops_DevelopersA…
I could give my other testicle for the one who does the linuxversion of fruity.
thanks.
,Tomi
_____________________________________________________________
Kuukausimaksuton nettiyhteys: http://www.suomi24.fi/liittyma/
Yli 12000 logoa ja soittoääntä: http://sms.suomi24.fi/
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amSynth - Analogue Modelling SYNTHesizer
****************************************
amSynth 1.0-rc2, code-named "jack", is now available!!
Get the source code at http://amsynthe.sourceforge.net/amSynth
Changes in this release:
* you've been waiting for it -- JACK audio output support!!
* completely revised ./configure & build system - will adapt features to
libraries installed, proper 'make install' target...
* launch a virtual keyboard from the amSynth menu!
* transparent per-user …
[View More]installation for first time users
* build fixes galore, now compiles fine on latest GCC versions
Enjoy!
(and any problems with it please get in touch, details on website)
-Nick d-.-b
__________________________________________________
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
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k_jack is a jack reimplementation, and mammut is a very special sound
transformating sound editor.
Download from http://www.notam02.no/arkiv/src/
New in mammut v0.14->v0.15:
---------------------------
-Removed the synth transform. It was not supposed to be there and had no
function.
-Fixed the wobble transform. Segfaulted with mono-files. (Bug found
by Dave Phillips)
k_jack V0.0.0.5 ALPHA - EXPERIMENTAL
-------------------------------------
ABOUT
k_jack currently consists …
[View More]of k_jackd~, libk_jack and libaipc.
k_jackd~ is a jack server external for pure-data.
libk_jack is (supposed to be) a (somewhat) libjack compatible
library.
Jack applications that want to contact k_jackd~ instead
of jackd must (somehow) be linked with libk_jack and
libaipc instead of libjack.
k_jackd~ does not speak with libjack, and jackd does not
speak with libk_jack.
libaipc is a library for audio interprocess communication,
based on code from the vstserver. A preview version is
included with this version of k_jack. (API is not settled.)
By using libaipc for interprocess communication, and letting
PD take care of various low/mid/high-level audio-stuff,
only a few hundred lines of code is currently used for this
implementation of a simple jack system.
k_jackd~ and libk_jack are not based on the jack sourcecode
found at jackit.sf.net, except for protos in the header files.
COMPILE
1. Go into the aipc/src folder and write make to compile up
libaipc.a
2. Go into the library folder and write make to compile up
libk_jack.a
3. Write make to compile up k_jack~.pd_linux.
4. Relink you jack application(s) somehow.
USAGE
1. Start pd with the "-lib k_jackd~" option.
2. Start a jack application linked with libk_jack.
3. Make an object in pd called "k_jackd~ <clientname>".
Correct number of inlets and outlets will be made
automaticly.
That should be it. Later, when things get more stable, point 3
can do point 2 automaticly.
WHY
use k_jackd~ ?
1. Simple. Only the clientname is used, not the portnames.
2. Easy and powerful interface to control the audioflow.
3. Good performance. Shouldn't be necessary to run as root.
TESTED CLIENTS
simple_client, freqtweak, ceres.
BUGS
Crashes pretty often. Does not clean up. Huge risk of not
freeing shared memory in the current implementation: Client
must exit before server, and client must not crash. And
server must not crash.
CONTACT
Send ideas (especially about the k_jackd~ object syntax),
comments (especially about the libaipc API), questions,
code, food, etc. to k.s.matheussen(a)notam02.no
--
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I just noticed the announcement from RME about their Hammerfall DSP MADI
PCI card. From the announcement:
"MADI, the professionals' multichannel audio interface, offers 64
channels of 24 bit audio at a sample rate of up to 48 kHz and 32
channels at up to 96 kHz. Transmission is done via a single line, either
coaxial with BNC plugs or with fibre cable. In both cases more than 100
m cable length can be achieved. Hammerfall DSP is fully compatible to
all devices with MADI interface."
This will …
[View More]make seriously high end gear available for use with standard
PC's. As ardour nears readiness, I have been contemplating a total
upgrade of my recording gear but I don't like the idea of Toslink in
general (short cable length), or adat optical in particular (high
jitter), but I do like the idea of optical coupling of audio gear to
computer gear (cleaner electricity for the audio gear). I'm glad I
waited. Availability is predicted to be within the next 6 months. Now
if manufacturers of reasonably priced a/d converters can offer something
with a MADI interface within the next 6 months...
Tom
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> >Argh, what's the great thing about standards again. I still prefer mLAN, as
> >it uses generic, consumer i/o cards, and firewire is fitted to almost all
> >laptops without needing expensive audio only hardware.
A single firewire standard would be great, but it hasn't happened yet.
MOTU, Digi, Metric Halo, and Yamaha all have incompatible proprietary
implementations. Plus there is the minor 4 or 6 conductor thing
courtesy of Sony. Starting to look like a mess to me.
&…
[View More]gt; it all still sounds pretty dubious to me,
Me too.
> hopefully, this should echo the fact that i'm with steve: MADI is an
> audio-only system, its expensive, and i don't think it has any
> particular technical benefits over mLAN. its sole advantage at this
> point is that anyone (as i understand it) can implement it without the
> licensing and other uncertainties that surround mLAN at this time.
The gear that uses it is expensive, but it isn't. RME will sell theirs
for the same price as a hammerfall. There is no reason why budget gear
can't use it. Audio-only systems are all that's available right now.
MADI and adat optical exist as open solutions right now. S/MUX support
for adat is spotty. mlan, and 1394 in general, doesn't exist as an open
solution. Sure I would like midi and audio on the same cable, but it's
not available right now. 6 months from now I will go with mlan if it is
openly available. If not then I will go with madi if I can afford it.
Otherwise I will go with adat optical. I have been postponing new gear
purchases for a long time waiting for an open 1394 solution. Eventually
I will have to stop being a cheerleader and actually acquire some gear.
The current favorite, adat optical, simply doesn't impress me. If
development stays on course, and I am ready to start using ardour in 6
months, mlan will be the only solution of the three that will *not* be
available.
Tom
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I realized a few more things about ramping when messing with
Audiality's voice mixers:
1) The STOP event *is* rather handy, as it has no value
argument to be calculated or processed. If you're doing
internal ramping (or coefficients) instead of using the
ramped value directly, STOP just means you stop ramping
your internal controls. SET (which is what you must use
if you don't have STOP) is supposed to *set* a new value,
obviously, and would normally result in the …
[View More]coefficients
being recalculated, regardless of current stat.
2) RAMP events with a <target_value, duration> aim point
do not guarantee *anything* but the value hitting the
aim point. Plugins are not required to perform truly
linear ramping; only something that "sounds similar".
If you would SET halfway to the aim point, you would
get a click, unless the receiver is doing *truly*
linear ramping internally!
3) Ramping with aim points means you don't have to know
where you are, to set up an accurate ramp to a desired
value. You don't even have to know if the current value
is above or below the target value.
Now, a STOP event before the aim point of a ramp in
progress would not always give you *exactly* what you
want, but close enough. (Provided the receiver "fakes"
linear ramping well enough - which it basically *must*
do anyway, for ramping to be usable.) Remember that if
you set up a new ramp, it will take you to the new aim
point regardless of the current value. So, as long as
you don't *SET* before (or after) the aim point, you're
safe.
4) Normally, you'd probably re-aim with RAMP, but in some
cases, you really just want to hold where you are
indefinitely. That's when you would use STOP - or fake
it with a "long" ramp to what should be the current
value. Both should work, but STOP is easier and faster.
5) A more serious issue is that if control events are not
allowed while ramping, except at the time of the aim
point, there is no way to avoid sending one RAMP event
for each block while ramping. You can't aim further
ahead than the first frame of the next block, or you
might have to break in and adjust the aim before you
hit the aim point.
This could potentially be a serious performance issue
with low latency hosts, since it means 1000+ RAMP events
(and the related transformation overhead) per ramping
control.
As a real life example of the alternative, the RT
envelope generator in Audiality uses one event per node.
(DAHDSR currently, which means 7 nodes.) Extra events
will have to be inserted to handle real time modulation
of the envelope (volume and pan controls, at the very
least), but there's never a need for more than one
extra event per linear section added.
Questions:
A) Is it necessary to require that aim points are
within the current block, to avoid "re-aiming"?
B) Is STOP useful enough to be in XAP?
Anyway, I think I have answered my own questions by explaining all
this (A: Yes, or complex effects won't mix with low latency, and B:
no, you'll probably end up using RAMP all the time anyway) - but I'm
still interested in comments. This might not be all there is to it.
//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 ---
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Well, I got the new device subsystem working in Audiality, although
I've only finished the SDL audio driver. ALSA 0.9 rawmidi more or
less finished, OSS audio is a mess, and there's no OSS rawmidi yet.
(Although that's rather trivial; basically a stripped down version of
ALSA rawmidi.)
Anyway, I got sidetracked - as usual. ;-)
First, I started playing with a noise modulated delay line, but didn't
find the results very interesting. So I decided to run a bunch of
fixed feedback delay …
[View More]lines in parallel - and that sounded a lot more
interesting: It became a pretty neat reverb. :-)
With 16 delays and exponential or exponential +/- Golomb ruler, it
sounds great for "normal" sounds, but has some problems with
transients; it's not dense enough. However, with these uneven delay
time distributions, it doesn't sound metallic at all. It rather turns
the transients into some sort of "noise" (sounds like random granular
synthesis, basically), that soon becomes this "shhhh" most of us
desire.
With even/linear delay time distribution, it still sounds pretty good,
but it starts to remind of a gong simulator or something. Plate
reverb, maybe... Interesting, but not what I was looking for -
although I'm throwing all interesting variants in, so they can be
selected through a plugin control.
With 32 delays, it starts to sound *real* interesting - but it also
starts to abuse my PC133 RAM pretty seriously. It won't run real time
on my P-III 933 above some 48 delay lines, although this is with 32
bit buffers of 8192 samples each. This can be improved upon a lot,
since I'm only using a tiny fraction of the maximum delay times with
that dense settings. (Like 300 +/- 100 samples.) Also, it's all
integer, so I could probably use 16 bits.
I have no feedbacks in beetween delays; just one local, LP filtered
feedback per delay. Every other pair of delays is stereo reversed.
Tried some network feedback setups, but I couldn't figure out any
truly useful configurations.
Untested idea: Have multiple feedbacks for each delay line. (The old
"reverb" of Audiality is actually just one stereo delay line with
multiple feedbacks, BTW.) The idea is that, while not as effective as
separate delays, this is a lot cheaper, so I might get more
density/cycle.
Oh, and I'll try the well known delay time modulation trick, of
course. Maybe using filtered noise instead of periodic LFO waveforms?
Thoughts? Any ideas for useful cross feedback configurations?
//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 ---
[View Less]
Changes:
- Bug when .ecasoundrc was not in ~/ directory (Ecasound 2.2.0)
- Bug when no ladspa directory defined in ecasoundrc
http://www.sourceforge.net/projects/tkeca
Regards,
Luis Pablo
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