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Hi there,
I'm currently thinking about using lash for my project.
But their website is down, when I used it a year ago it was worked like
crap with crashing etc.
Therefore I'm wondering how the actual state of this project is(is it
stable?) and if you recommend using it for developing.
Christian
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hi everyone!
the videos from lac 2009 are being made available at
http://lad.linuxaudio.org/events/2009_cdm/videos/ .
most of the footage is there - the missing stuff will follow in a few
days as florian (aka faberman) gets home from a production in italy and
finds time to encode the rest.
the video quality this year is quite stunning - thanks to the theora
team for huge improvements in efficiency and quality. if you are a video
person and haven't upgraded to the thusnelda branch encoder, do it NOW
(before even going for a coffee). then hit #theora on freenode.net and
sign hymns of praise until they kick you out.
you will note that the videos are raw - they could use trimming some
talks consist of several fragments that need concatenating, but due to
some stream problems they caused oggCat to barf, even though every
single fragment plays ok. your help is appreciated - get in touch with
me off-list or via l-a-d if you have worked on the stuff and would like
to upload your improved version.
best regards,
jörn
(Sorry for crossposting, but according to the statistics there are many people who are in only one list and I want to hear the opinion of both devs and users)
Hi people,
If would like you to have a look at my initial thoughts about live-midi with notation, or the developing of a notation-based midi sequencer (for Denemo). It would be interesting to hear your thoughts, additions or comments (besides "You made a spelling/grammatical error").
http://denemo.org/index.php/Join_the_team!#Live_MIDI_out
Of course if you are interested in this or other things your help would be welcome, we are always searching for help @ Denemo. Denemo is a Open Source Music Notation Editor, based on lilypond and written in C. (And I know no better one :)
greetings,
Nils Gey
Cologne
http://www.denemo.org
Hello,
Just joined the list.
I was toying with the following idea: take jamin's graphical multiband EQ and reshape it into an LV2 plugin. Is this a redundant idea in case someone else is already working on something similar ?
Just let me know before I start looking into this.
Cheers,
J.
Heya!
Doing (audio) infrastructure work on Linux? If so, please make to sure
to join us at the Linux Plumbers Conference 2009 in Portland/Oregon in
September. I will be doing an Audio track there and would like to make
sure that everyone who does audio infrastructure work submits a
paper, or at least attends, or maybe just knows about this conference!
It's a conference about the kernel-userspace interfacing, about the
lower levels in our software stack.
Last year we had a very successfuly track about Audio and this year we
will hopefully have a successfuly track again. Last year among the
speakers we had Takashi Iwai (ALSA), Jonathan Woithe (FFADO) and me
(PulseAudio). This year, we'll hopefully have YOU as well!
To get an idea of the conference, here's a writeup of last year's
track available from LWN:
http://lwn.net/Articles/299211/
More details about the conference you can find here:
http://linuxplumbersconf.org/
The full CFP is available here:
http://linuxplumbersconf.org/2009/2009/04/lpc-2009-call-for-proposals/
Spread the word,
Lennart
--
Lennart Poettering Red Hat, Inc.
lennart [at] poettering [dot] net ICQ# 11060553
http://0pointer.net/lennart/ GnuPG 0x1A015CC4
Hi,
Just a post to say I got the LADSPA working with SDL for my game. I've
scrapped the synthesis idea at the moment, but might go back to it.
The 1st stage can be downloaded from:
http://www.jwm-art.net/XorGramana/XorGramana-0.0.9_19-04-2009.tar.bz2
If you have spare time on your hands, could you give the code the
once-over and offer any suggestions/criticism/advice?
Cheers,
James.
relevant files:
smixer*
audio*
sfx*
ladspa*
further info:
The mixer code started off based upon the mixer from David Olofson's
DT-42 but has been converted to mix floating point data instead. It uses
libsndfile to load audio (flac) and libsamplerate for converting audio
during playback for different speed/pitch (but this has not been
incorporated into the api, the ratio is set randomly).
LADSPA plugins used are from SWH and C* (CAPS) sets. The ladspa_loader
code is based upon that used in my Wav Composer Not Toilet.
Still a lot more to do yet. Migrate effects processing from audio_gen.c
to smixer.c next job. The 1bar sequence/loop is hard-coded. Needs a
sequencer, probably will base it loosely upon DT-42 to begin with. Plan
is to have player position and immediate surroundings influence the
effects settings and mute voices/add extra hits/switch loops etc.
Free-up some allocations upon exit etc.
Info about the game itself can be found at:
http://www.jwm-art.net/XorGramana
-
Hi all,
I am using mplayer, sox and tee to capture streaming internet radio
and send it both to an audio recognition program and to a file
(recording).
This is how I do it:
mplayer -playlist {url} -nocache -af volnorm -msglevel all=1 -nolirc
-vc dummy -vo null -ao pcm:file={$fifo1} &
sox -S {$fifo1} -c 1 -r 8000 -t wav - resample | tee {$recording} |
tee {$fifo2} & ......
$fifo1 and $fifo2 are named pipes, $fifo2 is processed by an audio
recognition program. The idea is that the audio is converted to low
quality wav (8000Hz mono) after which it is fed to the program and
simultaneously recorded.
This works great most of the time but sometimes the recording is too
long/slow or too short/fast. Sometimes the speed is slighly wrong and
sometimes it is terribly wrong. When the speed is too slow, I notice
that the file is also much larger and if I open the file in Audacity,
the program tells me that the duration of the recording is for example
two hours in stead of 1 hour (when the speed is twice as slow as it
should be). Below is a list of three recordings that are the product
of this setup. The middle one is larger and when I play it back it is
three times too slow.
55M 2009-04-09 07:00 /data/station_recordings/48_20090409-0600.wav
152M 2009-04-09 06:00 /data/station_recordings/48_20090409-0500.wav
55M 2009-04-09 05:00 /data/station_recordings/48_20090409-0400.wav
My guess is that it has something to do with the sample rate. If
anyone has any idea where this goes wrong or what I could do to try to
fix it, please let me know.
Arnold.
Greetings;
I'm building a rhumba box not exactly like anything I've been able to google
up on the net. But I saw one a week ago that I thought worked very well as an
acoustic standup bass substitute when it was close miked.
My boxes panels are 5.2mm birch plywood, with double rabbeted cherry glue
strips, and except for the sound port holes in the front panel, will be air
sealed for best resonance. Total capacity then is 3.535 cu feet.
My question is, given that the port holes will be a larger one in the dead
center of the front panel, and two smaller ones flanking it, splitting the
space so the smaller ones will be centered on 1/2 the distance from the center
point of the center hole, sort of a | o 0 o | where the vertical lines are the
ends of the box by its longest measure.
Is there a formula for lowest frequency vs cubic feet that is commonly used to
determine the sizes of these 'sound' portholes?
Thanks all.
--
Cheers, Gene
"There are four boxes to be used in defense of liberty:
soap, ballot, jury, and ammo. Please use in that order."
-Ed Howdershelt (Author)
Yes, but every time I try to see things your way, I get a headache.