Ii's only taken me about 40 years to finish this little project. But my
CD is finished and ready for the masses. Preview tracks and place an
order at
http://mellowood.ca/music/cedars/index.html
Backgrounds with MMA, mastering audacity, plus other tools on linux.
Best,
--
Bob van der Poel ** Wynndel, British Columbia, CANADA **
EMAIL: bob(a)mellowood.ca
WWW: http://www.mellowood.ca
> Message: 17
> Date: Mon, 14 Apr 2008 10:48:50 -0400
> From: nescivi <nescivi(a)gmail.com>
> Subject: [LAU] looking for synchronisable and OSC controllable video
> player
> To: LAU <linux-audio-user(a)lists.linuxaudio.org>
> Message-ID: <200804141048.50806.nescivi(a)gmail.com>
> Content-Type: text/plain; charset=us-ascii
>
> Hi all,
>
> I am looking for a video player that could be started and stopped
> through OSC
> and maybe synchronised as well.
>
> We have a LADSPA plugin lying around which could maybe do the starting
> and
> stopping (we wrote it for Ardour), so if it can do LADSPA plugins that
> may be
> enough.
>
> MIDI control could be an alternative maybe...
>
> Does anyone have any good suggestions?
>
> sincerely,
> Marije
Veejay is what you're looking for.
http://www.veejayhq.net/
Gasten
The first (beta) release of Jmeters is available at
<http://www.kokkinizita.net/linuxaudio/downloads>
Jmeters is a Jack multichannel audio level meter app.
It looks very similar to meterbridge since it uses the
same pixmaps.
This first release offers VU, PPM and stereo versions of the
same. The stereo versions have two indicator needles on the
same scale. I've never seen a twin VU, but stereo (or M/S)
PPMs *do* exist. Later releases will add bargraph meters,
digital peak indicators for the analog ones, and a stereo
correlation meter.
The main difference to meterbridge is that Jmeters has the
correct ballistics for both the VU and the PPM.
The VU meter measures the average of the absolute value of
the signal, 'average' meaning a second order filter that
reaches 99% in 300ms and overshoots between 1.0 and 1.5%.
It is calibrated to indicate 0dB for a sine wave at -10dB
w.r.t. digital full scale (which is +/-1 peak in this case).
The particular VU scale used is not entirely linear and
starts at -20dB (it's one designed for a passive VU meter
with a diode bridge), so the meter will not move at all
for inputs below that level. Later versions may use a
VU scale designed for an active meter which doesn't have
this threshold.
The PPM is a pseudo-peak meter. It will indicate 80% of
the steady-state value for a 10ms burst, and fall by 24db
in 2.8s. Each scale division (1..7) represents 4dB. It is
calibrated to indicate '7' (+12 dB on the EBU scale) for
0dB FS.
For speech and music with distinct short peaks the PPM
will usually indicate higher. For music with continuous
long notes, and for heavily compressed signals the VU
indicates higher. Both meters require some 'getting used
to' in order to read them correctly.
You can modify the calibration by using the -g(ain) option,
but it should normally not be necessary.
Enjoy !
--
FA
Laboratorio di Acustica ed Elettroacustica
Parma, Italia
Lascia la spina, cogli la rosa.
Hi all,
I recieved a message on the muse mailinglist and thought maybe you like to hear the news...
I wanted to let you all know of a site I put up not long ago called
LinuxMusicians.Com. It's just a forum where us musicians can get
together and discuss creating music on Linux, what apps we
like/dislike, share our recordings and scores, and generally just have
a good time! I'd love it if you all would stop by and check it
out...we don't have very many members or posts right now, but I think
this could be a really exciting community
> > > > Find it at http://linuxmusicians.com
> > > >
> > > > Take care,
> > > >
> > > > Nathan
I think it could be something linux audio is needed. A central community forum to share thoughts wisdoms etc. Maybe it is possible to link it with a better documentation/ howto's etc. in a topic with howto's/ documentation and or links to usefull documentations and tutorials...
Would be great if many people join the forum :)
REgards,
~dirk
Isn't latency an issue with Pulse? If you want the best performance with a single application, isn't the solution to disable it?
-----Original Message-----
>From: Steve Lindsay <stephen.a.lindsay(a)gmail.com>
>Sent: Apr 14, 2008 7:27 AM
>To: linux-audio-user(a)lists.linuxaudio.org
>Subject: Re: [LAU] PulseAudio
>
>On Mon, Apr 14, 2008 at 6:04 PM, Arnold Krille <arnold(a)arnoldarts.de> wrote:
>>
>> Where does PulseAudio come into that picture? - When the gnome-guys realized
>> that esd is out of date and they want a new api/lib. Unfortunately they
>> decided to a) write their own and not adopt what is there and b) to go
>> audio-only which means no chance of KDE adopting it (apart from the fact that
>> kde already has Phonon). So PulseAudio is by design not _the_ solution for
>> sound on the desktop. It is just another middle-layer for sound. And why
>> should a desktop-app-dev adopt PulseAudio when he would have to use another
>> api/lib for video? Isn't it better to use one api/lib that has both and even
>> does them in sync?
>
>Isn't comparing Phonon and PulseAudio apples and oranges though? If I
>understand the situation correctly Phonon is just an abstraction layer
>that interfaces with various multimedia frameworks, whereas PulseAudio
>is an actual sound server. Even if the gnome guys had adopted Phonon
>that wouldn't have fixed the esd situation (old, unmaintained sound
>server). PulseAudio is a drop in replacement for esd and if you want
>to use Phonon you could (I assume at least in theory) use PulseAudio
>as a backend for it (or PulseAudio via gstreamer or xinelib or
>whatever).
>
>> And PulseAudio claims to unify both desktop-needs and pro-audio-needs. Another
>> place it will fail big time. Because it will never be good enough to have
>> ardour use PulseAudio. (Hint: Jack was designed for ardour...)
>>
>
>Are there fundamental design decisions in PulseAudio that would make
>this impossible, as opposed to just difficult or a lot of work?
>(sincere question, I have no idea)
>As cool as Jack is surely it would be nice to have a sound
>architecture on linux that seamlessly supported pro-needs as well as
>typical desktop needs?
>
>Cheers....Steve
>_______________________________________________
>Linux-audio-user mailing list
>Linux-audio-user(a)lists.linuxaudio.org
>http://lists.linuxaudio.org/mailman/listinfo/linux-audio-user
Hi,
There is a Free Software event in south of France this summer.
We would like to make a place for libre audio and libre culture, which
includes conferences and workshop around the tools (softwares), the
diffusion (medias, communities, ...), the licences, the users (examples of
audio studio using free technologies, examples of libre artits, ... )
So don't hesitate to send your propositions
Here is the official call for conference :
>From 1st to 5th July 2008, the Ninth Libre Software Meeting(9th LSM) will
take place in Mont-de-Marsan, in south of France. Hundreds of conferences,
workshops and showings will be programmed.
To maintain the interest of the public, it is important for LSM to renew the
content of different themes. So, if you participate in a new or unknown
project, if you wish to share your passion and your libre software
experience, we will gladly welcome you during LSM 2008.
We start a large call for communications on the following axes :
- thematic conferences - all subjects are interesting, particularly
those we have not thought about ...
- projects or libre softwares presentations
- workshop animations (development, initiation, ... )
To propose to the public a rich and various content, please send your
propositons, suggestions and ideas before February 8th to appel2008 at
rmll.info
*Contacts :* (program directors)
Nicolas Ducoulombier : nicolas at ldd.fr <nicolas%20at%20ldd.fr>
Christophe Merlet : redfox at redfoxcenter.org<redfox%20at%20redfoxcenter.org>
The definite programme, theme list, conferences and contributors will be
highly influenced by the feedback we will get from this call for
contributions.
Do not hesitate to participate and give worthy projects wide visibility.
--
Benjamin Coudrin
benjamin.coudrin(a)gmail.com
(+33)6.09.11.00.83
Hi peole,
I'm looking for a really cheap hardwaresynth or hardwaresampler with
midi-interface and some audio-out.
I want to use my masterkeyboard even when my pc is powered off. Just for
some playing and practising, nothing important.
So the sound doesn't have to be that great. And its only for home-use, too.
Maybe you know something, maybe some indie-stuff. The Location is
Germany, but it matters not.
Greetings and thanks!
Nils
On Sun, 13 Apr 2008, Justin Smith wrote:
> On Sun, Apr 13, 2008 at 9:24 AM, Kjetil S. Matheussen <
> k.s.matheussen(a)notam02.no> wrote:
>
> >
> > "Lee Revell":
> > > On Fri, Apr 11, 2008 at 11:43 PM, naysayer <gateswideopen(a)gmail.com>
> > wrote:
> > > > hi crew...
> > > >
> > > > i have just been looking at the benchmarking stats for reiser4 and it
> > looks
> > > > pretty cool. i was wondering if there is there is any reason to
> > implement
> > > > this kind of file system in multimedia environments. i current have my
> > > > root/system partition as reiserFS and my home partition as ext3. i
> > find this
> > > > to be quite efficient but perhaps if there would be an improvement to
> > > > recording stability and speed, then perhaps it could work well with
> > apps
> > > > like ardour. although, reiser seems to be happiest with small files,
> > reiser4
> > > > claims to be more efficient than ext3.... or that could be just spin.
> > > >
> > >
> > > These days the choice of filesystem should make no significant
> > > difference. Certainly it won't matter if an -rt kernel is used.
> > >
> >
> > Well, doesn't reiserfs use quite a lot more cpu than ext3? I think
> > that may make a significant difference...
> > _______________________________________________
> > Linux-audio-user mailing list
> > Linux-audio-user(a)lists.linuxaudio.org
> > http://lists.linuxaudio.org/mailman/listinfo/linux-audio-user
> >
>
> Reiserfs is optimized for smaller files (10 times as fast for files under 1k
> size, compared to ext3), and sluggish on huge files, so it would perform
> worse if you are mainly using large data (multitracking, wavetable
> synthesis), and slightly better if you are mainly working with small data,
> or data created in ram at runtime (fm, am, algorithmic synthesis, real time
> sampling). For a multitract machine reiser would definitely not help, for a
> machine emulating a moog, or doing experimental realtime synthesis, it may
> be a small improvement.
>
That's already been said. What I pointed out was that reiserfs uses more
cpu time than ext3, and for audio use, cpu usage may make a significant
difference, while throughput, which you are talking about, probably
don't.
Hello,
I'm experiencing some tonality problems with qsynth and jack.
I use them on a Debian SID, with non-realtime kernel.
I use both rosegarden and nted, and both package give me the same
results.
The problem is very simple : all the "melodic" instrument are played
half a tone upper than they should (a A sound like a A#).
I thought the problem was coming from my non-realtime kernel, but a
friend of mine has the same problem on an Ubuntu Studio distribution
with a realtime kernel and realtime enabled in jack.
The problem doesn't come from the soundfonts we use (there are many,
Crisi General 1.8, Super Evandro Drummer, and others...), as another
friend of mine, running also with Ubuntu Studio and the same settings
than the previous one (buffers sizes, etc.) doesn't have the problem.
I don't know what from it could come.
Any idea?
Thanks a lot.
--
==============================
ORL /// AMMD Booking (www.ammd.net)
° Sebkha-Chott (www.sebkhachott.net - Ohreland [FR]) - next touring period: 04-06/2008
° Unexpect (www.unexpect.com - Montréal [CA]) - Europe tour w/ Sebkha-Chott period: 09-12/2008
° La Muette (www.myspace.com/muette - Paris [FR]) - next touring period: 05/2008 (Bretagne)
° Mel-P (www.mel-p.net - Le Mans [FR]) - next touring period: 09/2008
Phone: +33 (0)95 234 72 48