I don't think I understand this jackplug concept.
I'm trying to get an alsa client, such as aplay to play through jackd.
I've setup jackplug in /etc/asound.conf
pcm.jackplug
{
type plug
slave
{
pcm "jack"
}
}
pcm.jack
{
type jack
playback_ports
{
0 alsa_pcm:playback_1
1 alsa_pcm:playback_2
}
capture_ports
{
0 alsa_pcm:capture_1
1 alsa_pcm:capture_2
}
}
I then tell aplay to use:
#aplay -d jackplug foobar.wav
But aplay plays the file even if jackd is not started; why?. Isn't
jackplug supposed to show up in my jack connections when aplay is
playing?
I seem to be missing something vital.
Is there some way to make a "fake audio device", like hw:3,0 that alsa
applications can connect to so that I can get the audio out and then
send it through jack, process it and then send it to the real audio
device?
--
Esben Stien is b0ef(a)esben-stien.name
http://www.esben-stien.name
irc://irc.esben-stien.name/%23contact
[sip|iax]:b0ef@esben-stien.name
Greetings:
I've added another recording to my "music made with Ardour" page, a
guitar duet this time. It's a performance of an old Jimmy Dorsey tune
called Maria Elena, you can check it out here:
http://linux-sound.org/ardour-songs.html
Best,
dp
Gidday everyone. This is not a Linux topic but I'm sure there must be
some saxophonists amongst you.
I need to get some more intermediate level Saxophone music for teaching
purposes. I've got some good James Rae books and have been getting the
best stuff out of Fake books but good music is hard to come by. Very few
sax players want to learn classical and lots of the jazz stuff gets hard
pretty quick.
Any advice for books? Heck, even throw me the name of a good theme tune
so that I can write it out. I always like using the Indianna Jones theme.
Cheers.
Hi,
I've just finished writing up a HOWTO on setting up realtime audio on
SuSE Linux Pro 9.2.
You can get it here: http://danharper.org/linuxdesktopblog/
After previously using Planet CCRMA under Redhat, all I can say is
Fernando, you are a huge blessing to us Linux Audio users, but SuSE is
too good to pass up!
Dan
--
Dan Harper
http://danharper.org
--- Enhancing the Linux desktop for desktop users ---
--- http://danharper.org/linuxdesktopblog/ ---
Greetings, Earthlings:
As a long-time fan of FM synthesis I've been wondering whether I
should try getting NI's FM7 running under libfst or vstserver. I own two
TX802s and am very fond of their sounds. NI's advertising makes some
extraordinary claims about FM7, so I thought I'd ask here to see if
anyone on this list has used it. I'm especially interested in whether
anyone has actually compared it to its hardware antecedents.
Anyone ?
Best,
dp
Folks,
I've got a early Holiday present in the form of a new violin. While my
eventual aim to to put a pickup on it and start playing around with MIDI, i
have to get it tuned first. I don't have a pitch pipe or piano handy.
I have a very accurate (musically) system set up, of which my computer is a
major part.
I'm looking for wav files or a tuning program that would help me tune this
violin. If i had a wav file or a tone generator, I could just put it on
repeat while i was tuning. I've been searching the web for an hour now and
have found nothing, but i may just be looking in the wrong places.
The starting point for a violin is A (440) then D, then G then E. All are a
perfect fifth apart.
For the record, i use KDE mostly, but have the gnome libs installed in case
anyone has a good app suggestion?
Anyone out there able to help?
Much appreciated,
Bearcat M. Sandor
Hello everybody,
when linux-audio-announce was created, it was agreed that announcements
should be crossposted to all three lists. The reasoning was that this way
people wouldn't have to subcsribe to LAA if they were already on LAD+LAU.
Now when you look at LAA archives, some people post only to LAA, some
(like me) to all three lists, and some to either LAA+LAD or LAA+LAU.
My suggestion is that we drop this policy altogether: announcements should
be sent to LAA and optionally to LAD and/or LAU. At least I've always had
the nasty feeling that I'm spamming LAD+LAU with my Ecasound release
announcements. Of course, when announcing conferences, major new versions
(JACK-1.0.0 maybe? :)), etc, I see no harm in cross-posting to all the
three lists.
So in other words, if you really want to see _all_ the announcements,
you should subcsribe to LAA.
Any comments? If no objections, at least I will from now on send non-major
release announcements only to LAA.
--
http://www.eca.cx
Audio software for Linux!
I tried JACK today with vanilla 2.6.10 and had excellent reaults - it
works with 32 and 64 frames, which previously required Ingo's patches.
Many of the latency fixes have been going upstream, and it looks like we
are finally showing some results. I think we may finally have a kernel
that's usable out of the box for low latency audio.
Can someone else try to verify these results?
Lee
Hi gang,
I'm staggering my way through the BruteFIR documentation in hopes of
building a simple soundfile convolver. Not being a mathematician, I'm
not too good at writing FIR filter coefficients ;-) Besides that, using
coeficients is not what I am looking for. If anyone here is using
BruteFIR to convolve one sound file or sound stream with a soundfile
impulse, could you post an example config for that?
Otherwise, are there simpler ways of doing this? I've tried the impulse
convolution in Rezound, for example, and gotten almost nothing but pure
DC out of it. What I miss in Linux is a simple impulse convolver like
the one in SoundHack for Mac OS9, Sonic Mirror or [especially] HOG for
windoze [doesn't run under WINE, I tried!]:
http://music.calarts.edu/hog/
thanks much!
d.
--
derek holzer ::: http://www.umatic.nl
---Oblique Strategy # 36:
"Consult other sources
-promising
-unpromising"
Hi there,
i recently bought a Creative Labs SB Live 24bit 7.1 soundcard (SB0410),
to replace my old SB live player 1024.
i suddenly noted that this card is _NOT_ based on emu10k1 chip, but
on an CA0106-DAT.
The latest alsa CVS has a driver for this chip, but with mplayer,
using the "mplayer -ao alsa:device=hw=0 -channels 6 -alang de *.VOB" command
i only get wired sound out of it - but when using oss, i get good
sound, but only
stereo.
i want to know if anyone had success getting this card to work.
i also want to know if this card supports mixing in hardware. ( using
snd_pcm_oss
with multiple oss apps.)
thanks, tommy