On Mon, 2004-02-09 at 18:50, Dave Griffiths wrote:
jack works with intel830 DRI
Hmm.. I only have Radeon's, so I'm a bit out of luck as far as testing
other cards I guess.
Anyone else have a positive/negative experience?
(For clarification, it's only realtime jack (ie jackstart -R) that
I guess I'll have to wait for the day when I can afford two
multi-channel audio interfaces, two multi-channel MIDI interfaces, and a
dedicated machine to handle the visualisation aspect of things.
or ... you could just get a Matrox video interface, since they seem to
be just about the only
On Mon, Feb 09, 2004 at 07:32:35PM -0500, Paul Davis wrote:
its a basic problem with real time software, the POSIX API etc.
JACK tries to lock *all* the process memory.
This is to ensure that nothing gets swapped out, right ?
Else it is very hard to ensure real time performance ?
(sorry
*) is anybody successfully using hdsp-cardbus with really low latencies ?
many people, including myself, use it down to 64 frames (i have a few
problems at 64 frames under heavy load).
*) how important is the firmware version - which firmware version should
i use for my hdsp card?
not
If I remove a synth application in Jack, then what happens to the effect
applications in the path? Does they receive silent input or break up?
The silent input would be better because then I could switch the
synth application with a need to remake the whole system.
silent input, of course.
SoX
The Swiss Army Knife of sound processing utils. It can convert audio
files to other popular audio file types and also apply sound effects and
filters during the conversion.
as much as i love sox, its too bad that it can't read WAV files in
IEEE float format let alone Broadcast WAVE. if it
pleased to announce the initial release of the caps audio plugin
suite under the GNU public license.
quoting http://quitte.de/dsp/caps.html :
% make
make: *** No rule to make target `ladspa.h', needed by `dep'. Stop.
http://liarliar.sourceforge.net
does it know JACK, though?
I still think about this from time to time and I would really like to see a
REAL implementation of the proposal. :-)
ardour basically allows this already, although it is not
automatic. there is no way to know what the user wants, so we leave up
to the user to initiate freezing any particular
It appears to be ethernet, not IP-based.
Ah. Silly me.
http://www.mkpe.com/articles/2001/Networks_2001/networks_2001.htm
so, where are the products? (referring to RTP and RTCP). Silly
author. Expecting people to productize[sic] publically owned protocols.
It was, however, the automatic merits that I wished mainly to explore.
Freezing has it's merits, but it requires that you dedicate some brain cycles
to deciding when and where you wish to freeze/unfreeze something. I could
sure use those cycles for keeping creativity flowing.
remember:
I want my plug-ins frozen the instant I close the parameter editor. ;)
oh, you don't want to do any graphical editing of plugin parameter
automation? :))
Agreed, it's a very definite trade-off - storage space for CPU cycles.
It is my observation, however, that storage space is cheap,
A bit more seriously, offline rendering in a tree graph works for 3d editing,
and has worked for a number of years. I think it's a natural progression to
try the same concepts on audio.
with the greatest respect, this is how audio used to be done and still
is for some purposes. just think of
It is my observation, however, that storage space is cheap, and readilly
available.
not my experience.
OT, pointless observation of the week:
cheap, yes. readily available no. we've been needing some bags
of sand in the back garden for a while too, and i still haven't manage
to get
http://www.peakaudio.com/CobraNet/FAQ.html
The difference is, they have armys of programmers whereas the open source
apps only have a few at most with the skill to work on apps as elaborate as
say Cubase or similar. I think it would be far better to collaborate on
what evidence do you have of the armys of programmers? my impression
is
Yes, but we need to think carefully about the name, _latency or
.latency might be a better choice, with a note that hosts should ignore
_/. prefixed ports that they do not understand.
i would prefer
#define LADSPA_LATENCY_PORT_NAME _wotever
or something similar.
Eliot Blennerhassett [EMAIL PROTECTED] wrote:
Greetings,
I am aware of PortMixer, which is a simple API that abstracts volume
controls.
Is anyone aware of any other crossplatform, cross-API mixer abstraction
layers out there?
i think its going to be pretty hard to do this.
I see Lilypond as one of the top Linux Audio projects.
It addresses the same critical issue for score editing and typesetting
that linux audio applications had to address in order to achieve
low-latency, non-destructive audio editing.
i did too, until i read about a tool called amadeus. it too is
it appears that many people don't know about amadeus.
There is a good reason for that: it's totally unavailable to
practically everyone. It's an extremely expensive closed-source
product that as far as I know is no longer even sold.
i think i worded my comments very poorly yesterday.
I
It seems that there is documentation (and even an implementation) for
A/M, but nothing for Connection Management. Has anyone tried recently
to get documentation? Based on
http://www.yamaha.co.jp/tech/1394mLAN/english/ptt.html , it seems that
documentation is available for a nominal fee, but it
on the right side of the page, scroll down to VIDEO OF MUSIKMESSE,
you will find 4 videos
(under Linux you can play them with xine or mplayer if you have the
win32 codecs installed)
... Do I even need to say it?
no, but you could explain the precise steps you would take to avoid
the win32
P.S.: Who is DJ Lego?
Some ask who, some ask what? :)
LADSPA and Jackit. Before rolling on I would like to ask if there is
already a project planning on something similar.
Gah, use Ardour.
As nice as Ardour may be, I personaly still prefer the interfaces of
modern UI toolkits, in combination with a nice Object Oriented language
(aka C++
The GPL also uses the term ,any third party'.
And the FAQ clarifies exactly what is meant by third party: Under some
circumstances (ie GPL section 3c) Distributees may pass along your written
offer of source code when they pass along your binary. Your offer must
extend to these third parties
Torben Hohn and I are pleased to release an initial version of libfst,
a small GPL'ed C library that provides support for using win32/x86 VST
plugins (FX and VST/i) within native Linux applications, with the
assistance of the Wine project's libwine.
Torben Hohn and I are pleased to announce the initial release of
jack_fst, a small JACK client designed to run VST FX and VST/i's with
connections to the rest of the JACK world, and, for VST/i's the ALSA
sequencer.
Tarball is available at:
Can you elaborate in terms of which version of wine you have used
successfully? (ie: wine = x.y.z or wine = x.y.z?)
i am using wine 20040309. i think torben has a slightly earlier
version than this which has worked for him to the same extent.
--p
Our Wine based VST hosting app is doing much better with very recent
Wine's: we are happy with the April 4th build. Most of the compatibility
issues are with GUIs.
oh, absolutely. i have yet to find a VST that i can't run if i don't
open the editor. admittedly, i am deliberately only trying free
I'm getting the following error with jack_fst
make all-am
make[1]: Entering directory `/home/lawrie/src/rpm/SOURCES/jack_fst-1.2'
if gcc -DHAVE_CONFIG_H -I. -I. -I. -g -O2 -I/usr/include/gtk-2.0
-I/usr/lib/gtk-2.0/include -I/usr/include/atk-1.0
-I/usr/include/pango-1.0
Does this mean that now one can channel ALSA-only aware apps directly to JACK
Yes.
and if so, are there any penalties of doing it this way as oposed to using JAC
K-aware apps (i.e. Sample-sync?)?
Yes, you lose sample-synchronous guarantees. Also, there *might* be glitches
in the audio output
This is the version I'm using with the vstserver now as well. It
seems to work better than the december 2003 version (finally!).
(Vsterver won't compile with it yet though.)
so i have noticed :( any idea when this might get fixed?
btw: torben has tested some of the plugins that are reported to
so are you actually doing the same as libfst ?
I haven't looked into libfst, but it sounds like 'yes'. However our code
i hope not! its hard to think of a reason for you to do this. you guys
are already presumably using wine and i would think it much easier to
use kjetil's approach of running
This is a lovely piece of work. Very impressive.
Thanks. We are still looking into some outstanding issues with
it. Come learn all about it at ZKM :)
Can you explain the licensing? The library appears at first glance to
contain both original code and (LGPL) code from other authors, the
whole
Am I completely misreading the thing? To me, the spirit of the GPL
would seem to support that interpretation, i.e. that your source code
offer must include everything that a normal person would otherwise
lack in order to build your program.
I believe we can work around this by adding an
Besides I see around the LADSPA API for sound processing but nothing si
milar
for input / output.
there are good reasons for that.
However, what about writing an advanced API for I/O plugins, completely
detached from other programs ( as Xmms ), potentially multiplatform, to suit
Has there been any discussion of including this in the mainstream
kernel? It would be wonderful to finally have full POSIX nanosleep()
and clock_nanosleep() support for everyone.
linus nixed it at the 2.5 stage. the POSIX API was added to the
kernel, but not the APIC-based high-res
well, it appears that there is little to no response to the proposal
from the LADSPA meeting at ZKM. just to be sure that the silence is an
accurate reflection of what people think, i want to take a harsh
stance on the proposal and see if it generates any response...
if we follow through with the
I don't mind *IFF* the metadata file has a simple, human readable
syntax (no XML please) that can be parsed line by line.
no XML, and yes, parsable line by line, and yes, human readable. *but*
the plan should be to use the supplied library to get and set
values. nobody should be doing it
Why rule out XML? It's one of the few widely-used language groups
that actually sorta meets both those requirements (*fairly* simple and
*somewhat* human-readable). ;-)
Reactions to XML can be as strong as your feelings about C++ :)
Personally, I am fan of both of them, but I don't think that
On Fri, May 14, 2004 at 12:01:01PM -0400, Paul Davis wrote:
I don't mind *IFF* the metadata file has a simple, human readable
syntax (no XML please) that can be parsed line by line.
no XML, and yes, parsable line by line, and yes, human readable. *but*
the plan should be to use the supplied
stable library interface. I don't know anyone now who *ever* writes X
protocol code, and I've never met anyone (except a few people I once
knew who worked on a commercial X server, and even that was more than
15 years ago).
This is irrelevant. Xrm has nothing to do with the X protocal, and
Also then the Tascam (and maybe some other) USB boxes won't work, as
they also need a firmware plus loader. But then, I think, both
packages to my knowledge (apt-cache search ...) aren't available in
mainstream Debian as well. Maybe some kind of wishlist bug could be
debian is about to exclude
The modern AMD chips support SSE, and its not really the AMDs fault, its
just that someone put some dodgy optimation code in the Makefile or auto*.
^^^
paul
--p
On Saturday 15 May 2004 16:53, Jack O'Quin wrote:
Can you (or anyone) name all the people in this group picture?
http://www.linuxdj.com/audio/lad/contrib/zkm_meeting_2004/photos/frank_neum
ann-misc/LAConf2004/DSCN2917.JPG
I can name, erm, four I think. That's a bit poor.
OK, in the interest
Welcome to the 0.9beta12 release of Ardour. We are now moving ever
closer to a 1.0 release. This release includes an incredible number of
changes and improvements since beta11. We expect some instabilities
compared to beta11 to remain - please test, collect debug traces,
etc. etc.
On Wed, May 19, 2004 at 11:24:36AM +0100, Steve Harris wrote:
On Tue, May 18, 2004 at 10:36:03 +0200, Alfons Adriaensen wrote:
We don't have a scripting language problem, we have a metadata problem.
Square peg, round hole.
Scripting will handle metadata without any problem.
This is not right. A scripting language (well, nay suitably expressive
syntax for that matter) can express metadata structures, but thats only a
partial solution.
What's missing ?
fons, do you know what steve actually gets paid for? :))
No, and whatever it is does not IMHO imply that
I went to ambisonic and read the FAQ. I do not agree with them when they
say that 2-channel stereo is only good for imaging between the speakers.
It is possible by using phase differences (and the assumption that
people are not living in sound-dead laboratories) to project sounds
outside of the
Hi,
forgive me if this question has been asked before: Is there something like a
Linux Live CD optimized for and containing audio (editing) applications?
Something like this
http://www.frozentech.com/content/livecd.php
where I can simply insert the CD, boot from it, and start working on my
What specific benefits have folks seen by turning on the kernel
preemption patch in a 2.4.19 kernel?
From the benchmarks that have been done, the preemption patch seems to
have only very small benefits when added to the low latency patch. By
itself, it adds on the order of 80-90% of the low
I am currently trying to write a player for the DAISY 2 and 3 Digital Talking
Book formats for UNIX machines. One of the big great features of the
hardware DAISY players available is to set ones own prefers playback
speed while retaining the original pitch of the voice. This only works
within a
On Tuesday 08 June 2004 20:22, Chris Cannam wrote:
On Tuesday 08 Jun 2004 7:46 pm, [EMAIL PROTECTED] wrote:
Right click on any slider in JAMin and it immediately goes to
the default position, whether center or zero.
Ah, now I looked for that feature but didn't find it. In Rosegarden
On Tue, Jun 08, 2004 at 05:06:16 -0500, Jan Depner wrote:
This is exactly the point I was trying to get across. Do something
different. I've been toying lately with the idea of zoomable sliders
when you have too little real estate or pop up sliders that are larger
than their normal
Ardour: default value ctrl-button2 click
context menu = button3 click
chris - heh, i am as bad as you! i got ardour's wrong :) default value
is shift-button1. ctrl-button2 initiate MIDI binding.
There is chasm both broad and deep between
plugin, show your GUI now
and an actual implementation of such functionality.
Yeah osc.udp://localhost:2134/ui/show ;) I'm very much in favour of
simple, UNDERcomplicated solutions. If you make its easy to do almost
everything and possible
The discussion about linear or radial mouse movement for
knobs finaly got me to mockup an idea i had in my mind
for sometime already.
For now I call it fan sliders:
http://wrstud.urz.uni-wuppertal.de/~ka0394/forum/04-06-10_fan_slider_01.png
[ ... ]
I think that's a really good idea..
On Thu, Jun 10, 2004 at 12:40:33PM -0400, Dave Robillard wrote:
Not that I wasn't being truthful though - I really don't want
proprietary software anywhere around here (freedom's pretty much all we
have), but that's just one man's opinion, take it or leave it.
I'm happy to have the freedom to
Hallo,
Steve Harris hat gesagt: // Steve Harris wrote:
On Thu, Jun 10, 2004 at 09:26:54 +0200, Frank Barknecht wrote:
Better use Mutt, it sucks less.
In all seriousness, all the non elm/mutt apps I've used dont let me go
though my mail quickly enough. The ability to switch inboxes, delete
Hallo,
Thorsten Wilms hat gesagt: // Thorsten Wilms wrote:
The discussion about linear or radial mouse movement for
knobs finaly got me to mockup an idea i had in my mind
for sometime already.
For now I call it fan sliders:
the major weakness in this setup for me is the neccesity to render the
midi before mixing. At least that means all elements of the arrangement
agreed. this is part of the reason why i was interested in supporting
vst/i's on linux. whether you run them as a plugin, or as a standalone
sample-synced
moreover, iirc, the design of jackd makes no consideration for
'live' routing changes. At least on my system, changing the graph
results in an xrun.
last time i plug an analog signal into an analog patchbay, there was a
click.
you cannot modify the graph in JACK while the graph is being used to
that the only people who work on open source projects are kids in
college and weekend warriors? The things that I don't understand (like
DSP :) are covered by other bedroom hackers like Paul Davis, Steve
Harris, Jack O'Quin, Taybin Rutkin, Jesse Chappell, Andrew Morton, Linus
Torvalds
On Fri, Jun 11, 2004 at 12:06:44PM -0400, Paul Davis wrote:
i have a very hard rule against hacking in the bedroom. i have broken
it once and will never do it again. garage hackers, living room
hackers and right now on the bar in the kitchen hackers .. sure,
but never, ever a bedroom hacker
Does anybody have any opinion on which threading system is superior?
I've been using glib for a lot of things, but for whatever reason I'm
hesitant about using it for threading if the only benefit it will
provide is consistency (I'm guessing it's just a wrapper for pthread
anyway).
yes, its just
The MAudio driver setup card was downright dishonest. It said latency:
128 samples, when actually latency was 256... that should have been
labeled buffer size to be more honest.
michael - windows and linux use the term buffer in different
ways. on windows, it generally means the chunk of memory
Paul falls into the first category. But i have a strong feeling that he
thinks he falls into both categories.
He makes better UIs than anything I've ever seen that came from you!
Why don't you get involved with ardour and help with whatever problems
you see in the UI? Well?
Posting useless
From: Alfons Adriaensen [EMAIL PROTECTED]
On Fri, Jun 11, 2004 at 08:37:09AM -0400, Paul Davis wrote:
you cannot modify the graph in JACK while the graph is being used to
process audio. you do not know how long the graph modification will
take if you try to do it (for example) right after
A question regarding jack use in a live setting. As far as I can tell
the soft mode only works with non realtime jack. What should I do if I
it should work with realtime mode too. this was specifically added
several (many?) months ago, for precisely the reason you gave.
but yes, write better
One thing I am still looking to learn more about is how to adjust
thread priorities and such to make sure that your threads are run often
enough (especially the disk thread), and how to decide how big your
disk buffers need to be.
4 years ago, Benno and I measured this and concluded that under
Paul, does ardour allow to specify the size of the per-track-buffers
you use for disk streaming, if yes perhaps you should add this as an
option since it's handy for the user having the possibility to
increase the default values to achieve optimal track count.
yes, you can control both the
pipe()s here too. last time i benchmarked on an early 2.4 kernel, pipe
and socketpair gave about the same timing figures, quicker than
msgsnd/rcv. i don't remember the exact numbers but i remember being
positively surprised.
from the whitepapers that IBM did, the linux FIFO appears to be just
in a lock-free way. This ensures zero-copy operation.
until you want to start processing the data but keep the original
around. i was always attached to the zero-copy model, but it just
doesn't seem to pan out in real life.
Plus we added a
wrap space so that a section of the beginning of the
I wonder how to find out which frame_time corresponds to the first frame in th
e buffer passed to the process callback..
Is it possible at all? does jack use an internal frame counter which correspon
ds to jack_frame_time?
i am not sure your question is well-formed. do you mean
: free
How this all is done in Ardour? I browsed the source but there are
a lot of stuff there. How about LinuxSampler?
We use a very simple approach in LinuxSampler: in the disk thread if all
Ardour also uses a simple mechanism, but the overall result is a bit
more complex. Basically, the
forget all that ranting i made about jack_frames_since_cycle_start().
jack_frame_time() is an interpolated function that can be safely
called from any thread (its not RT-safe, but it is fast), and will
return a monotonically increasing frame counter. i wrote it.
--p
well, basically the idea is to write a simple drum softsynth.. a playing groun
d for me to try out different synthesis approaches. for the start it will be a
simple substractive synthesis..
many softsynths just schedule the audio events for the start of the next perio
d which introduces jitter
Pete Bessman wrote:
[ ... ]
Fons is an intelligent human being, and even if he is being somewhat
more elitist than some may consider necessary, there is nothing
gained by berating his viewpoint with sarcasm.
That's a straw man. The original point was something to the effect of
a volume
something went wrong with make dist and the file that was uploaded
was abnormally large. it will be corrected around lunchtime (EST)
today. sorry for the confusion.
Less SNAFU's, Less bytes, More Features
This release is more or less identical to yesterdays 0.9beta17, but
its only about 3MB instead of 35MB+, and looping while running in sync
with JACK is sort-of working. Packagers please take note.
Next release expected in less than 8 days but more than
We work together on a project, and i would like to be able to record
and play back the audio mix from the vs-1680 to my computer together
with the midi clock data (he takes his machine home to work on the
audio part). That way i can work on the sequences on my mpc1000, and
play back the audio
fwiw, i'm achieving quite satisfying results driving MIDI out from a
1024 Hz RTC thread, with external hardware locking steadily onto the
output MIDI clock stream, even at tempi up to 240 bpm.
OK, I guess i think about this too theoretically. I suppose that with
reasonable limits on the
Hi everyone,
How should I perform resampling at runtime ? Like : I load samples with
different bitrates, then JACK calls my process() callback function using
its own bitrate... If the JACK bitrate and the sample one match, there's
nothing to do, but if they differ, there's a need for
I am using the latest 2.6.7 kernel (tried also 2.6.5) but with hdsp I cannot s
elect anything lower than 1024x2 buffer settings in jackd without having massi
ve xruns.
Give up on 2.6 for now. If you can't give up on it, then at least run
all (audio) apps with the environment variable
It's been a while, although this time there's not much. Just minor fixes,
nothing very outstanding. However here it is, a new public release for
QjackCtl, the little Qt (cutie:) application to control the JACK sound
server daemon, specific for the Linux Audio Desktop infrastructure.
any idea when
Hi Paul,
It's been a while, although this time there's not much. Just minor
fixes, nothing very outstanding. However here it is, a new public
release for QjackCtl, the little Qt (cutie:) application to control
the JACK sound server daemon, specific for the Linux Audio Desktop
infrastructure.
It can be avoided in deed, if just you set its value to the jackd default
one (500 msecs). In general, this is what happens with all other server
parameters.
the problem is that there is supposed to be *no* jackd default for
this :) i'll have to fix this in JACK itself. qjackctl should include
a
I have a program with some (p)threads, and a jack ringbuffer. One
thread is writing to the ringbuffer, and another is reading from it.
The question is: is it (thread-)safe to have a _third_ thread that
looks at the ringbuffer via jack_ringbuffer_read_space() at random
times to determine how much
resierfs: yes, it's a problem. I fixed it multiple times in 2.4, but the
fixes ended up breaking the fs in subtle ways and I eventually gave up.
andrew, this is really helpful. should we conclude that until some
announcement from reiser that they have addressed this, the reiserfs
should be
OK, thanks. The problem areas there are the timer-based route cache
flushing and reiserfs.
We can probably fix the route caceh thing by rescheduling the timer after
having handled 1000 routes or whatever, although I do wonder if this is a
thing we really need to bother about - what else was that
Other low latency while high network I/O uses are video conferencing
applications, diskless settop boxes that stream
videos over IP etc.
none of these are low latency applications.
Thus, the fact that Linux does not support protocols to prevent priority
inversion (please correct me if I am wrong) kind of suggests that supporting
realtime applications is not considered very important.
we went through this (you and i in particular) right here on LAD a
year or so ago. while i
On Tue, Jul 13, 2004 at 10:50:38AM -0400, Paul Davis wrote:
Other low latency while high network I/O uses are video conferencing
applications, diskless settop boxes that stream
videos over IP etc.
none of these are low latency applications.
Fileservering for audio playback/recording
Hmm, I've just recently learned about the Priority Ceiling Protocol,
an extension to Priority Inversion Protocol, which explicitly prevents
deadlocks. And I've learned about both in a RTOS course, so I'm a little
surprised by your statement about them not being useful for RT purposes :-)
solving
Yes! i was just about to write the list this very same message!
is there a link available yet?
Choice awards. Congratulations to Paul and everyone on the team !
aw crap, i'm gonna to have mess with the website again :)
--p
We made some tests to get a more clear picture, and in fact we got
rather confused instead. I will just give you the results, without
Windows-Options-Sync-Align captured material with [ Capture Time ]
Recording from soft-synths or other sample-synced sources requires a
different post-capture
Hallo,
Dave Phillips hat gesagt: // Dave Phillips wrote:
Btw, where can I get Softwerk now ? Is there an announcement on the way ?
Huh, I thought, Softwark was on hold? Is there a new version?
ok, the word is out :)
dave asked me innocently about softwerk on the train from
paris-bordeaux. i
As mentioned briefly, Dave Phillips innocently asked me about SoftWerk
while we were riding on the train from Paris to Bordeaux last week. I
thought how hard could it be to get it working again? and thereby
hangs a tale (and a tail).
To cut a short story shorter:
correction to a minor build issue:
http://linuxaudiosystems.com/softwerk/softwerk-2.0.1.tar.bz2
correction to a minor build issue:
http://linuxaudiosystems.com/softwerk/softwerk-2.0.1.tar.bz2
Just out of curiousity, what gui library did you use for softwerk, and
if you were to do it over again, would you still use the same one? I'm
i used gtk--, a C++ thin wrapper of GTK+. i would
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