Hello,
Here is just a simple question about linux audio. I'm looking for some
software which doesn't appear to exists. But it should be quite few
projects which have not release any code for the moment. So I'm asking
the question :)
Is there any windows manager that supports jack and lash, in order to
make the ''Desktop'' become a modular studio ?
Is it possible at least ?
If it is, is there anyone here with the time and knowledges to start a
project of this kind ?
Regards,
Elthariel.
Rob wrote:
> On Friday 10 November 2006 11:11, Rui Nuno Capela wrote:
>> and will do pass the message. But, ...be warned. At the
>> slightest hint that ALSA and KDE is getting behind the
>> community idea, I will step ahead (i.e flee :)
>
> While SuSE may have contributed to KDE, Novell actually owns
> Ximian, which for all intents and purposes created GNOME. I've
> always been wary of the movement to incorporate Mono into GNOME,
> and now I'm much, much more so.
>
you should have read between the lines that I'm not very found of gnome
and all that. my reasoning is simple: miguel started it just because
someone told KDE was evil because Qt wasn't pure GPL at the time; then a
whole wasteful effort begun to make gnome as the official gnu/fsf
desktop; then come ximian; then miguel found that not.net was some kind
of bethelem star. go figure why people didn't stuck with (imho) higher
brained C++ and KDE just for the doctrine, go figure. :))
(what a waste of great minds) ... and what we have now?
Yes, we do still have JACK and Ardour, ALSA, KDE and Linux as true
high-tech, open-source, top-of-the-notch projects in the whole known
universe. Go catch it, corporate-BS-savvy-boys :)))
> I don't see any particular threat to Linux audio users that isn't
> also present for Linux users in general, though.
>
Nor do I. After all, we the guys/gals are few and numbered, but do it
for the fun and love, not for the money. Bah, did I say "love" ?
Ah, for you who that don't know me, I was a hardcore windows developer
once before. Ah, read "once" with emphasis :) thing of the
first-half-nineties, blerk!
Bottom line is, and that's not an opinion, the F/OSS (r)evolution is
already unstoppable. Evidence comes to that IBM knows it. Ultimately, MS
knows it too. And you just got to face it now?
If you stay with me, and I believe MS strategy cannot be far behind, the
OS business is already a dead-end. People are in charge. If you speak
BS-like, you can read: the customer is in charge. And don't reply that
was RH who come original with that slogan. I do remember something of
the sort from Hammer's "reengineering the corporation". Yes early
nineties gone old :)
Just for the topic, funny thing is, a copy of gamma's/GOF is still
around here somewhere and being a mid-nineties foundation truth must be
told,... I'm just an old donkey who doesn't learn new languages anymore :D
Cheers.
--
rncbc aka Rui Nuno Capela
rncbc(a)rncbc.org
Just wondering how the framing is specified in that protocol, does it use some lsb's just like the VST System Link protocol does? It would be also nice if you documented the protocol and made it PD so the support could be added also to ASIO4ALL drivers etc. (http://www.asio4all.com/).
Btw. In theory you could support 8 x 24-bit (less the framing bit) / 48kHz audio transfer over the SPDIF if you used the 192 kHz sampling rate! Not bad, where do we need the ADAT protocol actually?
- Mikko
...................................................................
Luukku Plus paketilla pääset eroon tila- ja turvallisuusongelmista.
Hanki Luukku Plus ja helpotat elämääsi. http://www.mtv3.fi/luukku
I'm happy to announce the release of kbdz 0.2.0 alpha, a set of tool
which allows you to transform classical pc keyboards and mouses plugged
via USB into midi keyboards and controller.
Support for joysticks is planned for soon.
It uses alsa sequencer and the 'new' event devices stack (evdev module).
You can find it here:
http://sourceforge.net/projects/nahlwe/
Please give me feedback if you test it.
Regards,
Elthariel.
lemmel:
>
> Hi everyone.
>
> for a project, we need to be able to play sound (at first look wav file), and
> we made several tests ; with a created stereo sound, we try to use alsa but
> the results doesn't fullfill our needs :
>
> sound played at the time, T, we want, and finished at the date, D=T+sound
> duration.
> (thi is a software with strict time constraints)
>
> The sound was always troncated (even with finished software such as xmms,
> amarok), and even randomly truncated, (sound created with audacity, and
> exported as WAV 16/32 bit etc).
>
> When we use OSS, all seems to be perfect.
>
> But, it seems that OSS is nowadays "deprecated", and consequently we shouldn't
> use OSS. What we can do ? Are our alsa results due to misconfigurations ?
>
Only the oss modules in the linux kernel are deprecated. Programs using
the OSS api will still continue to work, currently most importantly
because of the oss emulation module in alsa.
If you should choose between alsa and oss, and the oss version works
just as well, or better than the alsa version, choose oss, because its a
more portable API than alsa.
However, if I were you, I would use sndlib, portaudio, jack, or some other
higher level audio input/output library instead of oss or alsa.
Snd-ls v0.9.7.7
===============
Snd-ls is a distribution of Bill Schottstaedt's sound editor SND.
Its target is people that don't know scheme very well, and don't want
to spend too much time configuring Snd. It can also serve
as a quick introduction to Snd and how it can be set up.
Changes 0.9.7.7 -> 0.9.7.12:
----------------------------
-Fixed listener.
-Removed various debug printing.
-added --without-builtin-gtkrc configuration option.
-Downgraded Snd from 8.4/26.9 back to 8.4/12.9 again. That upgrade was a
mindless mistake.
-Copied all files from my private snd three into snd-ls. Hopefully, this
should make everything work again.
-Added fix to make jackdmp work with standard installation of guile.
-Don't quit snd-ls in case file can't be opened during startup. Bug
reported by Dragan Noveski.
-Disable FAM for now, because it fails for no reason during startup.
Problem reported by Dragan Noveski.
Download from http://www.notam02.no/arkiv/src/snd/
jack_capture v0.3.9
===================
jack_capture is a program for recording soundfiles with jack. Its default
operation is to capture whatever sound is going out to your speakers into
a file. This is the program I always wanted to have for jack, but no
one made. So here it is.
Changes 0.3.8 -> 0.3.9:
-----------------------
*Changed the -rt option name to -d, to be compatible with jackrec.
*Do not stop recording in case of disk errors.
*Replaced deprecated libsndfile functions.
*Added the --format/-f option. ("jack_capture -f flac", nice :-) )
(adding "-f w64" solves the 4GB limitation of wav files)
Hi all,
While hacking around with aliasing effects in digital compressors (Yes
it is real, yes you can hear it!), I happened to run a 10Khz sine wave
into jamin with an instance of Jaaa hooked up to the output.
The results were 'interesting' as it appears that jamin introduces
easily measurable harmonic distortion even with all compressors and eq
bypassed! Switching the master bypass in jamin however does make the
effect go away.
This was at a level of -26dbFS with no boost and with the limiter
bypassed.
I say 'harmonic distortion', but it really is not quite that as it
appears as a series of narrow spikes every few hundred Hz.
Further checks show that the same effect appears with the test tone
turned down to 2.5Khz and that eq bypass has no effect.
Now with a single tone we are looking at per 'harmonic' energy almost a
hundred db down on the test tone, but there are a lot of these spikes
and they increase in number as additional inputs are added, so there
might be quite a lot of energy here all told.
Anyone have any ideas? This thing should be linear under these
conditions!
Regards, Dan.
Send instant messages to your online friends http://uk.messenger.yahoo.com
Hello all,
Core Sound <http://www.core-sound.com/default.php> willsoon
be offering a tetrahedral (Ambisonic) microphone at a very
reasonable price. They are also working on a combined preamp
+ AD converter unit for this mic. This will be able to multiplex
the 4 channels over a single SPDIF link, by using it a the
double sample frequency.
I'm currently working on a software controller unit for this
microphone. It will perform A-format to B-format conversion,
and allow measured impulse responses to be used for calibrating
the four mics. The result should be a very high quality portable
surround recording system at a reasonable price (compared to other
solutions which cost easily five times as much).
The remaining problem is the demultiplexing of the two double
speed SPDIF channels to four channels. It could either be done
within the ALSA layer, or in JACK's ALSA backend. Doing this
in a JACK client will not work unless it would be the only
client - all others would get the wrong idea of the sample
frequency and buffer size.
So here's my question to both the ALSA and JACK teams: what
would be your idea of a solution for this ?
--
FA
Lascia la spina, cogli la rosa.