Ok guys so here's one (told ya it wouldn't be long before I was yakkin again -
I tried to send this earlier but it might have gone bouncy due to html in
message) - this is for all you MIDI and Ardour experts :)
How about syncing Ardour with MIDI - let's see there's a few different things
I need to do -
1) sync ardour to outside tape source via ADAT sync...I set it to ADAT in
options window, and expect it to wait and play when I play tape, but no cigar
2) sync ardour to outside source via MTC sync - same deal - although if I
play, and then hit stop, the transport marker sort of sits there waitiing and
jiggling - does this mean something?
3) sync Ardour to sequencer inside computer - say rosegarden - via MTC I guess
- whatever works...
anxiously awaiting replies :) thanks, have fun and yay! :)
--
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Aaron Trumm
NQuit
www.nquit.com
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Good Morning (?, European :) ),
Well, do you know sth about MIDI-keyboards uses other ... defaults ( than MIDI or others ),
so that Linux/alsa can't handle them correctly ??
I have a Evolution MK-149 .... In an alsa-tutorial Dr. Nagorni sais that some Keyboards
seem to use other NOTE-OFF evens, e.g. send just anther NOTE-ON event ... Is my keyboard
such a keyboard ??
greez, Sascha Retzki
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feel free to shoot me ;)
MTC sync WAS awesome - then it stopped being awesome, don't know why...ardour
terminal window saying
received new MTC status
MTC stopped ...
received new MTC status
MTC stopped ...
it seems like other people have probably tried to deal with this and had this
happen is why I mention...
--
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Aaron Trumm
NQuit
www.nquit.com
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whoa I'm fast ;)
syncing ardour to outside MTC seems to be solved (repatched midi patch bay,
boom it seems to slave like a - well, like a slave ;) )
--
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Aaron Trumm
NQuit
www.nquit.com
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Hi,
dealing with linux GUI audio applications
I have found that the following
feature would be usefull:
I want to stick or glue some windows of
*different* applications together so that
they be organized in "solid body"
i.e. so that I could minimize/restore,
move them together.
(you can find an example of similar behavior
in how xmms organizes its subwindows)
I know that some window managers offer
"border gravity" and some others offer
virtual desktops,
but sometimes it's just not enough.
As fas as I can tell the linux audio
goes the unix way in the sence that there are
many small interacting programs each doing
its own thing. So one needs to have *many*
small GUI apps open at the same time.
And the feature Im talking about would
be realy usefull.
So my question is:
Which window manager support such "sticky"
feature?
horsh
-------- directBOX Reply ---------------
From: fbar(a)footils.org
To : linux-audio-user(a)music.columbia.edu
Date: 04.10.2003 17:16:43
Hallo,
hexe_2003(a)directbox.com hat gesagt: // hexe_2003(a)directbox.com wrote:
> Well, I thought exactly the same. Maybe I'll try another way ... Do you know
> possibilities to connect a MIDI-keyboard to applications via the OSS-layer ?
For routing OSS or more specific rawmidi devices you need to use
aconnect. and a snd-virmidi virtual midi card.
> Or can I make something wrong with alsa, too ?
> I used 'aconnect 64 65' , where 64 is my External MIDI port and 65 is
> the internal FM synthesizer... .
Basically this is correct. You didn't give full portnames, though.
First you need to find out the correct input ports with "aconnect -i"
$ aconnect -i
client 0: 'System' [type=kernel]
0 'Timer '
1 'Announce '
client 64: 'Rawmidi 0 - M Audio Audiophile 24/96 MPU-401' [type=kernel]
0 'M Audio Audiophile 24/96 MPU-401'
....
and then the output ports with "aconnect -o". Your FM synth should
show up here. On my box it looks like this:
$ aconnect -o
client 64: 'Rawmidi 0 - M Audio Audiophile 24/96 MPU-401' [type=kernel]
0 'M Audio Audiophile 24/96 MPU-401'
....
client 73: 'Emu10k1 WaveTable' [type=kernel]
0 'Emu10k1 Port 0 '
1 'Emu10k1 Port 1 '
2 'Emu10k1 Port 2 '
3 'Emu10k1 Port 3 '
....
Now I can route input coming from 64:0 to output 73:0 to use a
keyboard with the SBLive hardware synth with "aconnect 64:0 73:0".
If you use an ALSA sequencer enabled software like Muse or RG4 you
shouldn't need to fuss with aconnect, BTW. Just configure it inside
Muse or RG4.
ciao
--
Frank Barknecht _ ______footils.org__
---------
Well, Muse doesn't work because my jack server seems to be the wrong
version (libjack), and I did not recompile it yet ( I use Debian unstable );
Could you give me a link to the mentioned "RG4" ( never heard of it
before ;) )
Retzki
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WAIT ! RoseGarden4 ..... my mistake
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Hi folks,
I've been trying for months now to get arecord to work with the built-in
microphone and/or mic jack of my laptop but it just won't work. Since
the alsa developers seem totally uninterested in fixing or even
understanding this problem (yes, I've posted on alsa-devel many many
times) and the built-in A/D on the laptop is unlikely to be any good
anyhow I've decided to investigate an external A/D system. It's for a
laptop so it would have to be either PC card, USB, or firewire, not
PCI. Compatability and reliability are a big deal to me -- I'm fed up
with sending mail to alsa-devel, which seems to route to /dev/null as
far as I can tell. Can anybody recommend a solution that's really
simple to set up and reliable to use under linux and doesn't cost an
arm and a leg (e.g. less than $400)? I'm not too picky about number of
ins and outs or fancy features, more picky about quality,
compatability, ease-of-use, and reliability.
Thanks,
-n8
--
>>>-- Nathaniel Gray -- Caltech Computer Science ------>
>>>-- Mojave Project -- http://mojave.cs.caltech.edu -->
GuyCLO~ wrote:
>I agree. I found that larger latencies (100 < latency < 200) are usable for
>having fun. I mean I have used softsynths on a computer without tuning
>latency and without beeing root.
Hmm. How are you measuring latency? I'm not sure how to do it (sufficiently
accurately). I'm running SuSE 8.2, which some have suggested includes the
low-latency patch, but I don't think so. I've been too busy (lazy?) to check
or implement. I find even when playing some .ogg file to jam along with, that
hiccups are "disturbing" or distracting. Maybe I've got problems (as a
musician wannabe) with my timing? I've got my .ogg files on a server, but I
believe xmms pre-buffers (I recall setting it to 1/2 second at one point?)
its compressed audio stream, so I think I'm only hearing jitter from variable
interrupt response (and temporarily blocked interrupts?)?
Your other comment (in another post) about "real life lost packets" (UDP
comparison) is interesting. You would have to transmit a time code in each
packet, so the player can continually "re-sync" the audio. I don't think the
current audio streams do that? I think they "assume" (Benny Hill?) that the
audio stream is continuous, and therefore you can derive the timing?
--
Juhan Leemet
Logicognosis, Inc.
Hello again - I hope this is the right list for this topic (so far I can only
get subscribed to two and I keep bouncing back and forth between them...)
first off, may I say thank you to the list members for bearing me, and extra
thank you still to those who take the time to respond - amazing gentlemen and
ladies you all are; I realize I have written what amounts to several novels
worth of emails in the past couple weeks! :)
well I started a couple threads earlier about strangeness (or what I think was
strangeness) with compiling and installing hdspmixer and such...the questions
I had I didn't figure out, but somehow hdspmixer was able to run - whether
it's ACTUALLY working, I don't know for sure sure...
THAT having been said,
I only know of a couple people who are using the HDSP 9652, along with a
couple who've developed for it - Mark K, Thomas, Kevin, et al - does any of
this sound familiar? :
--
a) I've been doing a lot of back and forth and reading but I'm a bit confused
as to what hdspmixer is DOING (which makes it harder to tell if it's working
:) )
b) my goal, as I may have said on other threads, is to be running ardour with
my HDSP 9652, sending each of 24 individual adat optical channels out to 1
channel input in my behringer ddx3216 (outfitted with adat i/o). An earlier
thread I started was solved by me adding -d hw:0 to my jackd command line
(duh), where I wasn't seeing 24 (actually 26) possible outputs when I clicked
on "output" in the mixer in ardour
now I'm able to see the outputs, and assign them as they should be, meanwhile
in hdspmixer i've picked the preset which assigns ins, out and "playback"
(what would be the difference between "out" and "playback" I wonder) to adat
outputs, etc:
--
first I used hdspconf to change the HDSP to 44.1 for purposes of these tests
next I ran hdspmixer
I kept hdspmixer running, then started jack from a terminal using:
su
jackd -R -d alsa -d hw:0 -r 44100 -p 2048
then started ardour from a terminal:
su
ardour -n <--- that's so no splash screen will appear
but what was showing up at the board wasn't what I was expecting *laugh* if I
routed a track to channel 1, then it showd up a 3 db too quiet at channel 1
on the mixer, and at the right volume on channel 2 (I was using a -12 1khz
test tone that I generated with the board and was able to record to ardour -
ardour indeed is playing it at -12.1 - I think the point 1 is explanable but
I'll skip that 'cause it doesn't matter just now) if I routed the track to
channel 2, it didn't show up anywhere. odd numbered tracks after that seemed
to be showing up on 1-2 just like 1 did (although I can't confirm the
consistancy of this)
I experimented with opening up qjackconnect, which was the only one of the
many patch bay programs from planet that showed 24 capture and playback
channels and connecting captures to playbacks. this didn't seem to do
anything, which made me wonder just what qjackconnect was for.
then I closed all, closed the hdspmixer program, and started just jack and
ardour. the results from that are:
routed to channel 1, it shows up at channel 1, 3 db quiet, channels 3, 7, 9,
11, 17, 19 and 23 at 30 db quiet and channels 5, 13 and 21 at 12 dbs quiet -
wha?
routed to channel 2, it shows up only at channel 2! 3 dbs under - actually,
this 3 dbs may be a non-problem issue in the board, so possibly it can be
ignored
routed to channel 3, it shows up on 1, 2, 3 and 4 and some other channels at
half volume - wha wha?
channel 4, nothing
channel 5 --> 4, 5, 6 and others
ok you get the picture. randomness. a weird mess.
I don't know if this is a driver problem, a HDSP 9652/alsa driver patch
problem, an hdspmixer problem (I did, as I mentioned, have oddities on
compiling that program), a jack issue, an alsa issue, a bug in ardour, a bug
in alsa drivers, etc. etc. etc.
I only know of a couple people who are using the HDSP 9652 - Mark K, does any
of this sound familiar?
clearly there are thousands of details - anyone who wants to talk about this
and wants other info, as usual, ask and ye shall receive :) (although in
some cases, a request for details may yield first the question "how do I find
that?" :) )
thanks and I hope this is interesting and stimulating, maybe even
educational!!! :)
--
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Aaron Trumm
NQuit
www.nquit.com
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