Hi
After way too long, a new release of the BLOP LADSPA Plugin set.
Orginally named Bandlimited LADSPA Oscillator Plugins, but there's more
than oscillators... they're more useful in a modular host, such as
Spiral Synth Modular, gAlan, Alsa Modular Synth etc.
Plugins in the set:
New since last release:
* Quantiser (quantise to arbitrary values in a range)
* Signal Tracker (envelope follower, sample+hold, track+hold, ...)
* DAHDSR envelope generator
* Difference and Ratio utility plugins
Existing ones, with less bugs I hope:
* Bandlimited DCOs (wavetable based):
- Sawtooth
- Square
- Pulse with variable width
- Triangle with variable slope (Saw->Triangle)
* Analogue-style step sequencer
* Fast'n'Dirty Moog 4 Pole implementation
* Two ADSR envelope generators
* Random wave generator
* A mono amplifier
* Non-bandlimited square and pulse clock oscillators
* 1V / Octave CV to Frequency converter
* Control rate to Audio rate interpolator
* Sum and Product utility plugins
Some plugins come in control and audio rate variations.
RDF metadata is also included if your host supports it.
All plugins pass the demolition test:
http://www.ecs.soton.ac.uk/~njl98r/code/ladspa/demolition.html
Website:
http://blop.sourceforge.net
-
Myk
>From: Joshua Haberman <joshua(a)reverberate.org>
>
>Also, I have a strategy in mind for gracefully degrading when the GUI
>thread can't pull data from the FIFO fast enough (again, for VU meters
One could use plain variables for the VU values.
Engine updates them. And GUI reads them. GUI may not
get all values, but the most important temporary max level
changes so slowly that GUI does not skip on that.
Another choise is to resample the VU signals to low enough rate
in the engine. If 30 frames per second is the display refresh
rate, then using rate 20 Hz for VU signals allows GUI to catch
up if VU freezes.
>that the loads and stores are observed in the required order, you have
>to use memory barriers (again, in theory). See:
>
>http://en.wikipedia.org/wiki/Memory_barrier
What it means practically? Any code? (Cannot check wikipedia now.)
One could write some ID numbers to the buffer next to the buffer
pointers. If the ID number is not what it should, then the problem
has been recognized. What to do in this situation? If it is
a non-realtime thread (disk thread), the disk thread could idle
a bit or do something else. But if it is the realtime audio thread,
then what? Any this kind of trickery could be slower than any
memory barriers, guess.
Quick question: disk thread may suspend if there are no disk use.
How the disk thread is woken up to read the lock-free buffer?
Are the following yes or no?
(1) audio thread sends SIGCONT signal to disk thread,
(2) audio thread sends sys/msg.h message to disk thread,
(3) audio thread sends one byte via a pipe to disk thread,
(4) disk thread uses timer to wake up itself.
Something else?
Audio thread sends the wake up relatively rarely. So, if pipe has
a small buffer, then pipe could be practically lock-free. How it is?
Juhana
Oh well, just read through the more than 30 mails ...
Thanks to Arnold, Marek, Tim Hockin, Benno, Uwe, Tim Orford
for the nice words.
Thanks to everyone for the interest.
About a standard toolkit:
The general desktop toolkits do not adress some special needs
of audio apps, that's clear.
While an integrated look/feel with other, non-audio apps, would
be nice, it can hardly be achieved (think alone of theming).
Therefor a somewhat alien look could be of advantage (better
clearly different than somewhat alike).
Unification advantages: write once, get used to once, allowing
usage to become somewhat subconcious (switching between different
looks/feels doesn't allow that).
It would be great to have all scalable vector guis (especialy
interesting options for modular systems, overview vs tuning).
Even without scaling, I think I would prefer to deliver plain
SVG.
Beside implementations in the major toolkits there's also
Y at the horizone, with a base collection of widgets that
is planned to be extendable with modules.
About generating plugin guis from parameter descriptions:
don't forget that layout is important. There would have to be
some hints for that at least.
I'm working on something very general regarding user interfaces,
and widgets will be one topic. This should give me a nice
foundation for an audio toolkit design from the look and feel
side. So how about building a standard for exactly that: look
and feel? Who's with me?
About the implementation: wouldn't it be possible to wrap things
up in enough ways to every capable developers likes?
For people like me it would be more interesting to have gui
editors / high level descriptions for concentrating on design.
(hand writing xml or similar is acceptible for me, fiddling
around with c code rather not)
To Rtaylor:
I'm all for breaking away from analog limitations. But I still
think that a little 3d look helps with making clear what is
manipulatable and what is not. And I hope it's possible to make
both purists and eye-candy freaks happy enough (I'm myself more
on the purist side, being german industrial design student, think
Bauhaus and all :)
And would you say what I presented is not clear/easy to
read/decipher?
About advanced knob functionality:
For resetting knob with center (pan) I would rather replace the
simple line mark with an arrow, that can be clicked for reset.
Mousewhell should work, of course! Backwards decrease, forwards
increase, to match with scrollbars/sliders. When Ctrl is used
for finer adjustment, Shift or Alt could boost.
Plus/minus buttons could also work.
Right-click must be preserved for context menus (knobs could
of course have such).
For numerical input making use of a text field close to the knob
might be better (but in any case only when numerical exactness
makes sense).
Tim Hockin: please elaborate about velocity sensitve knobs like
used by OhmForce.
A special fine adjustment accessible by a non-left click or with
modifier key is a little problematic, because it breaks the
connection between scale/indicator and mouse movement (not that
I'm against it, only wanted to point this out).
Clicks/modifier keys should not be user options. No need for
extra coding, overloaded preferences / layout problems and
users fiddling around instead of using the software.
Radial?
For those who think radial is bad: in that case knobs shouldn't
be used at all! (Tim Hockin: it makes no sense to disagree to a
half sentence/meaning!) Popping up a slider doesn't help much,
the nature of the control should be clear right away.
However, I never had problems dealing with radial knobs, but
always found linear ones do not feel right.
There's no need to move the pointer accurately in circular
fashion, a soft bow works perfectly fine.
I do not say knobs are perfect / better than sliders
or other solutions in all cases, and I'm aware of the problems.
Oh, and about linear needing 2 directions, up/down for gain
and right left for pan (Uwe): Sure. And the widget has
to be clear about that in advance.
About releasing the knobs
Everybody can have the existing svg, png export or Blender 3d
files (just ask). I doubt I will continue with Blender, so future
work will be vector or directly bitmap. Like I said, it's not
finalized at all.
ATM I do not know how to best generate series of bitmaps
for rotations. Would be nice if someone could help with a
little imagemagick or gimp scripting to do the rotation and
combine it with fixed parts.
Benno:
I think Dave Robillard is right. Only when you're very sure the
saved space can be put to better use, knobs ar an option there.
And the design would have to be adapted to fit in.
---
Thorsten Wilms
>From: Paul Davis <paul(a)linuxaudiosystems.com>
>>>
>>>> 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 you're done with
>>>> one process cycle. the only sure way to do this is to use lock free
>>>> parallel programming techniques.
>>
>>Would anyone please explane details of these techniques?
>
>No. Google for "herlihy lock free". Dozens of references. Papers range
>from about 12 years ago to this year.
I did mean the details on the graph modification. Up so far I have
not understood what is the problem in modifying the graphs in the engine.
It sounded like you solved the problem using lock-free par prog
techniques. By reading the solution I could catch up the problem. ;-)
Want explane what kind of modifications you mean? Why they take
so much time?
Juhana
Greetings, all. I have not been subscribed to this list for several
months now, but someone pointed me to a link to Paul's message that I
quote below. I realized that now is a good time to check back in, see
what everyone is up to, and let you know what I'm up to.
A few days ago Paul posted:
> the only sure way to do [live graph modification] is to use lock free
> parallel programming techniques. this is still a highly experimental
> area of computer science, and i have yet to see more than a glimmer of
> a technique that would actually work in a complex real time system
> like JACK.
Right now I am very interested in lock-free data structures, and I am
working with Ross Bencina (of PortAudio and AudioMulch) to implement
them in an open source library. See [0] and [1]. It turns out there
are several practical algorithms being published in this field, most
notably by Maged Michael from IBM research. Ross and I have been in
contact with him for advice and clarification about his papers, which
are available on his web site [2]. We have some working code, but
since all this stuff is highly architecture-specific we need to do more
research. A significant part of the library will probably end up being
in assembly.
For me, this is part of a bigger effort: rewriting Audacity's back-end,
and making it a GUI-independent library. We're tentatively calling
this library "Mezzo" -- more about Mezzo's goals and basic API is at
[3]. Mezzo's data representation scheme is just a refactoring of what
Audacity uses now, but the real-time system will be completely
redesigned to be similar to JACK's model: several nodes with input
ports and output ports that implement callbacks. The whole graph will
be run from the audio api's callback.
Getting back to lock-free data structures, my intention is not to use
locks anywhere in this real-time system. Since I haven't written the
whole thing yet I'm not sure if this is completely possible, but here
is my general strategy:
1. signal graph is constructed in main thread
2. once audio starts, the audio thread runs the graph once for every
callback
3. if the main thread wants to change the graph while it's running, it
does any heavy lifting (memory allocation, etc), then sends a message
to the audio thread with a lock-free queue (ex: "add this node and
connect it to this other node").
4. buffers are passed between the disk thread and the audio thread
using lock-free queues
5. in the real-time thread, buffers are allocated from and returned to
a lock-free freelist. For example, if you have a node that generates a
sine wave, it needs to allocate a buffer when it runs to put its data
in. Instead of calling malloc, it asks for a free buffer from the
freelist.
The real-time code is still in its beginning stages, but I have gotten
it to play data from disk by sending it from a disk thread to an audio
thread. You can see the code in our SVN repository. [4] I'm also
keeping a log of my activity and my thought process on my web site [5].
I am interested in any reactions people may have to this work. I drew
a lot of the inspiration for Mezzo's real-time system from JACK and
other things I have read on this list.
Regards,
Josh Haberman
[0] http://www.audiomulch.com/~rossb/code/lockfree/
[1] http://www.audiomulch.com/~rossb/code/lockfree/spec.txt
[2] http://www.research.ibm.com/people/m/michael/pubs.htm
[3] http://www.audacityteam.org/wiki/index.pl?Mezzo
[4] http://mezzo.homelinux.net/svn/
[5] http://www.reverberate.org/log/
Hi.
I am currently trying to figure out how to restructure the code
of my little yatm tool to be flexible and extensible in the future.
Basically, I'd like to be able to become a JACK client at
some point in the future. Since JACK uses a poll model, I have
to majorly restructure my code. I think what I need is the following:
Three threads, one for the UI, one for the MPEG/Vorbis/Speex decoder,
and one for audio output (to either libao or JACK).
I'll also need a buffer between the decoder and the audio thread, and some
way to signal the decoder thread that more audio data is needed.
Additionally, I'd like to have the ability to make the buffer
in between remember the last n seconds of audio data and make
it possible to jump back in time <=n seconds during playback.
I've looked at the lock-free ringbuffer of JACK, but I am not
sure if this is really what I need here. I am very new to
linux audio programming, and especially to pthreads (wrote
my first thread using program two days ago).
I'd be very happy if some of you experts here could give me
a hand here and point me into the right direction.
Maybe there is already code out there which does roughly that
which I can look at?
Or maybe you can suggest which mechanisms of pthreads I should
be using for this kind of scenario.
--
CYa,
Mario | Debian Developer <URL:http://debian.org/>
| Get my public key via finger mlang(a)db.debian.org
| 1024D/7FC1A0854909BCCDBE6C102DDFFC022A6B113E44
Greetings:
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Please update your bookmarks.
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Best regards,
dp
Hi all,
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
want realtime performance, but a really forgiving server, so dropouts
never cause the server to timeout. It's more important to me to have
continuous rather than glitch free audio (I never ever want everything
to shut down)
If the answer is ultimately to write better jack clients - thats Ok ;)
cheers,
dave
OK.
I have permission to get one of these programs moving with some
constraints (appended). WHich do you guys think would be most useful
(publishable :-) )?
1. The OpenGL Spectrogram implemented into Steve Harris's meterbridge
2. A 31-BAND (or more?) graphic EQ
These would both be implemented into JACK
My advisor requested these specs:
1- This should be run on any platform including PC platform.
<Yeah, right. Perhaps I"ll just implement the code, and wrap it with JACK
and WIN-API with preprocessor commands....Grrr...>
2- The interface would be build with a Tcl/Tk script which is compatible
with SGI/ Aplle and PC platform.
<I'm just not sure he knows what he's talking about...Anyway, I think he
just wants an interface that will work on linux and windows...again, see
#1>
3- The FFT algorithm should be run in parallel and optimize at least for
two platforms.
<I guess once we're splitting up the spectrum, this shouldn't be a
problem...BTW,the advisor teaches an MPI course. This idea probably came
from this...I'm not sure what he means by "2 platforms.">
4- We can use a Wavelet Transformation inside of FFT and to parallize it.
<*shrug* I'm gonna hafta do some research. wikipedia says it's like a FFT
running in O(N) instead of O(N log N):
http://en.wikipedia.org/wiki/Wavelet_transform
>
I can recomend against some of these. I just think he wants me to do some
extra work... Granted, without these recomendations, I'm not looking at
the suggested 500 Hours of work...
This project is starting to look big and clunky:
1. Implementing both windows and linux
2. Parrellizing the code? I don't mind writing multithreaded or MPI... it
just looks like overkill....
3. Can JACK compile on IRIX? That's his OS of choice (Being his only *nix
experience...) and, well, SGI's are fun! Perhaps I can convince him to put
linux on his old O2.
Anyway, bottom line, it'll be fun, and I don't mind spending my time on
this, if ANY part of it will be of use....
...I'de just hate to kill Steve's code, though.
Anyway, suggestions?
Thanks y'all!
-Mike
