Dear linux-audio list!
(sorry for cross-posting)
PDradio is a self-regulating internet radio station for PD (Pure Data) related art (and is of course build with PD ;).
Upload your PD-related music, vote for songs, listen to the automatic radio moderator ...
The link: http://pdradio.iem.at
Please feel free to upload your music and participate!
We hope you have fun with it,
LG
Winfried+Georg
------------------------------------------------
PDradio:
PD Webradio is a self-regulating internet radio station for the PD-community.
Registered users can upload PD-related music and share them with the community.
Because we didn't want to compile the radio program all the time we build an automatic radio DJ.
All user can vote for a song they would like to hear and the DJ will generate a playlist according to all votes.
Additionally there is an automatic radio moderator, who speaks a little bit about the songs (metadata which can be uploaded together with the music) and maybe is joking somtimes ...
PD-webradio is running on a Linux Server with Zope/Plone.
The audio streaming, speech synthesis, radio-moderator, DJ, ... is implemented in PD and Python for the communication to plone.
If you want to listen to the PDradio, vote for songs or you have music related to PD and want to share it - follow the link:
http://pdradio.iem.at
------------------------------------------------
Hi
Just found this item on Ebay being sold:
**********************
"
The best thing about this is that when you are running all 26 channels
of inputs and outputs, it adds no load to your CPU.
I had this card running flawlessly with Cubase with 6mS latency (It
would work down to 1mS but if I pushed the PC sometimes it would crackle
so I settled for 6mS)
I'm selling this because I have moved to Pro-tools and it won't work
with the RME card.
I'm now running with 26ms latency and hardware that definatley does make
demands on my PC. I guess that's progress! "
********************************
Thats how thing 'improve' in the proprietary world
PS I get this latency with Ardour on my Hammerfall lite!
Cheers
Bob
www.hearmymusic.co.uk
Hi
I've written a bunch of scripts to start synths, and connect them. But
sometimes it get's screwed mostly because of timing issues. When
counting on client numbers for connecting stuff which app comes up first
is important... So:
1) Can I avoid the client numbers alltogether? This seems to require a
way to rename ams, since all clients are calles AlsaModularSynth. Is
that (starting ams with a specific clientname?) possible?
2) Alternately: is there a better way to handle timing? So app2 will not
start before app1 is up and running?
Here's an example of the scripts:
#!/bin/bash
options='--sched -+rtaudio=jack -d -odac -M'
csound_path='.patches'
ams_path='.patches'
zyn_path='.patches'
key1=`usb_keyboards | head -n 1`
key2=`usb_keyboards | head -n 2 | tail -n 1`
key3=`usb_keyboards | head -n 3 | tail -n 1`
sine_balls=$csound_path/bouncing_sine_balls.csd
noisepad=$csound_path/filtered_noise_pad.csd
stringish_arpeggio=$csound_path/stringish_arpeggio.csd
mellotron=$csound_path/mellotron_strings.csd
warm_muted=$ams_path/muted_lead_big_mistake.ams
brass=$ams_path/res_sweep_big_mistake.ams
midisplit 60 &
midisplit1=$!
midisplit 45 60 &
midisplit2=$!
midisplit 48 60 &
midisplit3=$!
ams -j --preset $warm_muted &
sleep 1
ams -j --preset $brass &
sleep 1
csound -+jack_client=csound5_1 $options 4 $sine_balls &
csound -+jack_client=csound5_2 $options 3 $noisepad &
csound -+jack_client=csound5_4 $options 5 $stringish_arpeggio &
csound -+jack_client=csound5_3 $options 8 $mellotron &
sleep 2
aconnect $key1 129:0
aconnect 130:1 133:0
aconnect 130:2 132:0
#aconnect 129:5 133:0
aconnect $key2 130:0
csound_auto_connect_audio
ams_auto_connect_audio
echo press key to shut down...
read
aconnect -x
killall csound
killall ams.real
kill $midisplit1
kill $midisplit2
kill $midisplit3
killall python
--
peace, love & harmony
Atte
http://www.atte.dk
I need to be able to pipe stereo audio through a notch filter so that
I can adjust it while listening and analyzing the effects it is
having. Unfortunately I cannot find a way to do this. Here's why:
- The only LADSPA notch filter I can find is "Mag's notch filter"
(notch_iir_1894.so)
-Jack Rack works great with this plugin, except when I change the
number of stages in the interface, the change is only applied to one
channel. (the other controls are applied to both channels)
-AlsaModularSynth immediately segfaults when trying to load this
particular plugin
-Ardour will not work with this either. The controls don't do what
they are supposed and al I can accomplish with it is some sort of
wacky low pass filter. (how this is accomplished with a notch filter
plugin is beyond me...)
So... What else is there? these are the only programs I know of that I
can pipe audio through very easily. (please don't suggest Pd, I've
never gotten it to compile correctly) Anybody else have similar
problems with this plugin?
-Reuben
Hi everyone
I'm going to be spending 4 days in Paris towards the end of the month.
I'm looking for a decent but not expensive hotel / guest house. I'd be
grateful for any recommendations.
The only thing I really have to do is visit Vigier guitars in Grigny. So
I'm going to have some time on my hands, and I'd be really interested to
hear about any venues where world-music-ish types bands might be
playing.
thanks
John
Lee,
I built a new kernel 2.6.11 for this system (using the debian
kernel-package tool). I was under the impression that 2.6 kernel source
included the ALSA by default.
When I try to ./configure the latest ALSA it says that my kernel as the
ALSA built-in.
Can you clarify this for me? If my kernel has ALSA built-in, how can I
be sure that I am building it with the latest ALSA code as well?
aloha,
dave
On 9/1/05, Lee Revell <rlrevell(a)joe-job.com> wrote:
On Thu, 2005-09-01 at 10:05 -0600, Dave Price wrote:
> Any ideas what I should adjust to get a mixer/volume control
> to do
> anything meaningful?
Get a newer ALSA (at least 1.0.9b).
Lee
Hello,
I have a problem using linphonec with ALSA on my MIPS AU1100. The linphone
console client seems to work quite fine but there's no sound output. I
made a test together with sipomatic and traced the behaviour of linphone.
(see attachment) Whatever I try there will be no sound. I even sent the
process in sleep state (ctrl+z) and checked the mixer settings but they
were okay.
I can play sounds with aplay normally and linphone states having found
alsa, too. So I assume it should work. May there be buffer problems or
something like that?
Any help would be quite appreciative.
- Marcel Karras
--
------------------------------------------------------------------------
Contact: toka(a)freebits.de karma(a)informatik.tu-chemnitz.de
http://www.freebits.dehttp://www.tu-chemnitz.de
Unix, Linux && OpenSource Student of Chemnitz University of Technology
------------------------------------------------------------------------
Hi all,
I'd like to pass on an article that may be of interest to your comrades
itching to get into Linux audio - but have read all the horrible rumors
about configuration hell. Well.. I think Fervent has a hit. O'Reilly
Digital Audio feature story this week is a review of Fervent Software's
Studio To Go!.
<shameless self promotion>
Yeah.. it's written by me.... but I do believe Fervent has created a
great way for windows users to get into Linux Audio. It's been mentioned
on this ml before. But: Great Job Fervent!
Please check it out at!
http://digitalmedia.oreilly.com/2005/08/31/studiotogo.html
</shameless self promotion>
thanks!
brad
Me again.
More music: http://blog.dis-dot-dat.net/2005/08/stress-and-music.html
Also includes sample breakdown.
Why not take the same samples and do something different? Would be
interesting...
James
--
"I'd crawl over an acre of 'Visual This++' and 'Integrated Development
That' to get to gcc, Emacs, and gdb. Thank you."
(By Vance Petree, Virginia Power)