I want to run a sound recorder like audacity to record
what is coming thru my soundcard when playing
streaming audio thru xmms.
However audacity can't open the audio port /dev/dsp
when streaming audio is playing. This is on knoppix
3.4 using the OSS driver.
Would using ALSA make this possible?
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hi everyone!
in case you were wondering how to get ices-jack to stream your jack
graphs out on the net, here's a quick howto:
browse svn.xiph.org, get the following modules from /trunk:
ao, vorbis, ogg, ogg2, theora, speex, vorbis-tools, ogg-tools
(do this even if you have ogg packages from your distro installed,
it won't do no harm and makes sure you've got the latest'n'greatest)
there's nothing interesting to configure afaik, so you can compile
them (in that order) without interaction:
for i in ao vorbis ogg ogg2 theora speex vorbis-tools ogg-tools; do
svn co http://svn.xiph.org/trunk/$i; cd $i; ./configure && make
install ; cd .. ; done
from icecast/branches/kh, check out
libshout, icecast, ices
again, not really anything to configure, so the for-loop can do the
grunt work...
now fire up icecast, fire up ices, connect it to your jack graph,
and the fun starts.
the default config files are extensively commented, but here's my
config, in case you need some more inspiration:
http://spunk.dnsalias.org/download/ices.xmlhttp://spunk.dnsalias.org/download/icecast.xml
(the source and server run on different hosts, and icecast runs
chrooted and as user icecast)
btw, a graph with an ogg edge between ices-jack and xmms-jack
vertices makes a nice delay effect :) if you use feedback, there's
interesting sound deterioration due to repeated
ogg-encoding/decoding and noise buildup. here's me toying around
with my bass and such a setup:
http://spunk.dnsalias.org/download/netjam.ogg
have fun
jörn
--
"Debugging is twice as hard as writing the code in the first place.
Therefore, if you write the code as cleverly as possible, you are,
by definition, not smart enough to debug it."
- Brian W. Kernighan
Jörn Nettingsmeier
Lortzingstr. 11, 45128 Essen, Germany
http://spunk.dnsalias.org (my server)
http://www.linuxaudiodev.org (Linux Audio Developers)
Chris Cannam posted:
> Because sfArk and sfpack compress soundfonts much better than
> zip/gzip/bzip2 can.
This is absolutely correct. The reason is that text files repeat character
sequences exactly whenever words and other character combinations are
used over and over again. The "normal" compression methods build tables
and utilize this precisely repetitive nature effectively. Audio files
repeat character sequences only approximately, so are not recognized as
being almost the same even though they sound the same, hence are regarded
as new sequences. True binary files (e.g. stripped executables) look pretty
much random, also, to one of these compression algorithms. The random
appearance of audio files is one reason why MP3, sfark, etc. were developed.
To distinguish between these latter techniques: As most people know by now,
MP3 is a lossy technique which means that information is lost never to be
seen again. sfark is not. These soundfont compression techniques are a
compromise between loss of information and effective compression. Some
out there may be using MP3's in place of such compression schemes as
sfark's, believing that MP3's are as good or better due to the good
compression ratios obtainable. This is at a fairly heavy cost. I would
advise against it generally, which is the reason I'm spending time posting
this. MP3 is fine for distribution over limited-resource channels, but
not so fine for soundfonts/samples.
Hope this helps someone out there. Now I really am going to try to get
caught up....
Regards to everyone,
Dave.
Hi:
I 've try to use the alsa driver for my nforce2... when i try load the
module snd-intel8x0, the system say me:
/lib/modules/2.4.26-1-386/alsa/snd-intel8x0.o: init_module: No such
device
Hint: insmod errors can be caused by incorrect module parameters,
including invalid IO or IRQ parameters.
You may find more information in syslog or the output from dmesg
/lib/modules/2.4.26-1-386/alsa/snd-intel8x0.o: insmod
/lib/modules/2.4.26-1-386/alsa/snd-intel8x0.o failed
/lib/modules/2.4.26-1-386/alsa/snd-intel8x0.o: insmod snd-intel8x0
failed
My /proc/pci is:
Multimedia audio controller: nVidia Corporation nForce2 AC97 Audio
Controler (MCP) (rev 161).
IRQ 5.
Master Capable. No bursts. Min Gnt=2.Max Lat=5.
I/O at 0xd800 [0xd8ff].
I/O at 0xdc00 [0xdc7f].
Non-prefetchable 32 bit memory at 0xec002000 [0xec002fff].
Bus 0, device 8, function 0:
I use Debian sarge, and instal this packages:
alsa-base, alsa-modules, alsa-utils. alsa-lib
¿Any idea to fix this problem?
--
Leito Monk
--------------------
Miembro de CaFeLUG
www.cafelug.org.ar
-------------------
On Wednesday 22 September 2004, Dave Phillips wrote:
> I'm trying to send a MMC message to Ardour to set it up to receive MIDI
> time code. Apparently Ardour will follow MTC but it has to receive an
> MMC Start message first. So far I'm still unsuccessful, even with your
> suggestions, but I'll keep at it.
You can use "amidi", from "alsa-utils":
$ amidi -S 'F0 43 10 4C 00 00 7E 00 F7'
sends an XG Reset to the default port; `man amidi` for details.
To can use "echo", too:
$ echo -ne '\xf0\x7f\x01\x06\x02\xf7' > /dev/midi01
BTW, you can try a MIDI realtime message "start" like this:
$ echo -ne '\xfa' > /dev/midi
The message "f0 7f f7 06 02 f7" is not a valid MMC command, because the
device-id 0xf7 is also the EOX status byte. It should be a number between 00
and 0x7f.
Regards,
Pedro
How do I get my TB SantaCruz (cs46xx) to make sound when it gets midi
events? I assume this would involve a wavetable, but I'd be happy with
FM synth or any terrible sounding piano sound as well.
My cpu is not really up to the task of sequencing and softsynthing at
the same time.
--
De gustibus non disputandum est.
On Tue, 21 Sep 2004 08:52:17 -0700, Russell Hanaghan
<hanaghan(a)starband.net> wrote:
> On Tuesday 21 September 2004 08:34 am, Hans Fugal wrote:
> > The card works great for PCM.
> >
> > Which software I use to patch the midi depends on my mood, but usually
> > aconnectgui or aconnect.
> >
> > Haven't been fiddling with jack or Qjackctl yet.
> >
> > I'm trying to get my midi controller keyboard to make noise out the
> > soundcard, is all.
>
> Hmmm. Ok...so maybe the wrong midi device? How many midi devices show
> in /dev/snd ? If more than one, try patching around.
$ ls /dev/snd
controlC0 midiC0D0 pcmC0D0p pcmC0D2p pcmC1D0c pcmC1D1c seq
controlC1 pcmC0D0c pcmC0D1p pcmC0D3p pcmC1D0p pcmC1D1p timer
The TB is card 0.
> TB also has several outs for pcm (Surround stuff) but sound like your already
> using the card for normal stereo audio.
I will fiddle with the mixer more this evening. Perhaps the fm synth
is on master1 or some other pcm...
fugalh@falcon:~$ aconnect -oil
client 0: 'System' [type=kernel]
0 'Timer '
1 'Announce '
Connecting To: 63:0
client 64: 'CS46XX - Rawmidi 0' [type=kernel]
0 'CS46XX '
(the keyboard is turned off and I'm not at home to turn it on)
--
De gustibus non disputandum est.
Hi. I'm tring to use a GNU/Linux machine as a real-time effects box (live -
reverb, eq). I wonder if I could use real-time LADSPA plug-ins but with no X
server (to improve latency). I mean something like JACK Rack but from a shell
terminal.
Thanks
MartÃn.
So now that my machine is %99 percent there I have made a page that
represents my work so that others might get some help without asking all
the questions I did:
http://www.aproximation.org/application/AMD64laptop.html
Its filled with misspellings and is not really complete. Only use will
determine what actually works and what doesnt, but its a beginning at
any rate.
Im going to post this on a few other lists as well, so I appologize if you
get this message more than once.
Thanks for all your help!
-thewade