So I am not at my machine at the moment but I think you need to put the
headers in to the vst directory, then run the fixheaders script then do
a make.
However I am also just a simple user so of course you may have already
done this and it still doesn't work. But I remember getting your error
on Saturday night and fixing it somehow.
m.
> -----Original Message-----
> From: linux-audio-user-bounces(a)music.columbia.edu [mailto:linux-audio-
> user-bounces(a)music.columbia.edu] On Behalf Of Mark Knecht
> Sent: Friday, June 04, 2004 3:03 PM
> To: A list for linux audio users
> Subject: Re: [linux-audio-user] jack_fst build question
>
> Matthew Allen wrote:
> > I sent a message about this on Monday. Did you run the fixheaders
> > script? After I did that I was still getting errors with jfst.c so I
> > gave up.
> >
> > m.
> >
> I did run the fixheaders thing. I think this is part of the problem.
> From the jack_fst README:
>
> <SNIP>
> You need to have AEffect.h and aeffectx.h in your include path.
> they can be downloaded from
>
>
http://ygrabit.steinberg.de/users/ygrabit/public_html/vstsdk/Download/VS
T%
> 20Plug
> -Ins%20SDK%202.3/vstsdk2.3.zip
>
> and are located in source/common. unfortunately aeffectx.h
> contains some constructors making the C compiler grok. Remove them we
> dont use t
> hem.
>
> copy them to your include path.
> <SNIP>
>
> This might as well be Greek to me. What's a constructor? What exactly
to
> I remove. What include path are they talking about.
>
> Me no understando. Stupid user.... ;-)
>
> - Mark
>
>
> _________________________________________________
> Scanned on 04 Jun 2004 22:05:45
> Scanning by http://erado.com
_________________________________________________
Scanned on 04 Jun 2004 22:14:41
Scanning by http://erado.com
>The real good news is that the feature freeze and push
>for 1.0 will cause these interfaces to be worked on.
>I'm getting pretty excited about it because I think
>it's going to happen soon. :)
>
>ron
Oh Please Please Please Please Please etc.etc.etc ...... drool, slobber, dribble, tremble, dream, pray etc.etc.etc..... ahhhhhh; will I wake up and find it was all a dream ? The minute my Linux boxes can communicate with the real world reliably via MTC / SMPTE I'm in full time!!! Just wish I could get the whole CVS thing,love to try beta 15 but alas just not geeky enough.
till later,
Geoff.
Hello all,
I asked this on the MusE and the Rosegarden list, but no answers so far.
Until now, I used MusE 0.6.3 for sequencing, as a slave being controlled
by my external BR-8 HD-recorder.
Switching to MusE 0.7.0pre3 breaks things: When I start the BR-8, the
MusE cursor does not move, but after I pressed stop it direclty jumps to
the correct place (without having played any sound).
So I decided to give Rosegarden a try: The MIDI events show up in the
Transport Window, but Rosegarden does neither start nor stop.
I think I have configured both programs correctly to run as a slave.
Any hint?
Thank you,
t.
--
Thorsten Mika mailto: tmika(a)t-online.de
Hamm / Germany TM5173-RIPE
Public Key ID: 41338C37 -- http://www.keyserver.net
Greetings:
I've made some minor updates and URL corrections for the Linux
soundapps site, but I've also discovered a problem with the European
mirror. The site at www.linuxsound.at now presents an advertisement for
ATNET, and the advert includes a link to http://linuxsound.atnet.at.
Alas, that link doesn't work correctly either. I've written to ATNET
twice already and have heard nothing from them. If the problem persists
for another week I'll remove that link. The Japanese mirror is also
experiencing a problem: apparently it isn't updating the top and TOC
pages, which is uncool because I've added material to both those parts.
Hopefully the Japanese mirror will update completely by this weekend.
For the time, the only completely current site for the Linux soundapps
pages is now:
http://linux-sound.org
And you all know the rest...
Best regards,
dp
Hi, I've just migrated my studio over to Linux (whoohoo!) from
Windows/Finale/Cakewalk and need some more details on synchronization
between applications, specifically Ardour and Rosegarden4. The Ardour
manual is a little sparse in this area.
Specifically, I am using Rosegarden to control a MIDI keyboard and a
MIDI drum machine which route their audio outputs into an external mixer
and then back into the DAW hardware (Audiophile 2496). Works great, of
course, with much less fuss than Finale or Cakewalk. My drums are
broken up into multiple tracks (one for kick drum, one for snare, etc),
so I can mix them separately as needed (the way you would with a real
drum kit). So I need to capture the audio coming into Ardour (will have
to record each track individually), which is easy enough to do, but I
need to make sure each track stays in synch with the other drum tracks
or I will have a bloody mess. The version of Cakewalk I used didn't
have SMPTE support, but I assume I will have to use something along
those lines.
Any online documentation to look at? I couldn't find any... if there is
also some kind of auto-sync so when the MIDI playback starts in
Rosegarden, Ardour will automatically start recording the incoming audio
would be useful also.
I was recently converting some compact cassettes to ogg/flac when I
noticed the stereo image was reversed.
After poking about, I came to the conclusion that:
Left Channel:
White or black phono (RCA) plug
Tip of stereo jack
Right Channel:
Red phono (RCA) plug
Ring of stereo jack
but I couldn't find a definitive answer on the web. I have one
commercially made hookup cable that goes red phono to the tip of the
jack on the other end, and yes, this was the one pugged into the sound
card...
Is this correct?
Regards,
Pete.
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Ok, DAW/sequencer sync has been discussed.
But how about sequencer/sequencer?
Case in point:
I'd like to use Rosegarden or Muse as a sequencer for the solo parts, or
for anything that has long sequences that do not repeat too much.
But i'd like to use Seq24 for the small repetitive loops, because it's
such an ideal tool for that.
How do i ensure that the sequencers stay in sync? I mean, both in terms
of beats / minute and in terms of synchronized operation
(start/stop/record).
--
Florin Andrei
http://florin.myip.org/
> ------------------------------
>
> Message: 4
> Date: 01 Jun 2004 13:44:13 -0500
> From: "Jack O'Quin" <joq(a)io.com>
> Subject: Re: [linux-audio-user] JACK, Realtime-lsm, 2.6, Clients can't
> connect
> To: A list for linux audio users <linux-audio-user(a)music.columbia.edu>
> Message-ID: <87d64j9kzm.fsf(a)sulphur.joq.us>
> Content-Type: text/plain; charset=us-ascii
> Greg Moss <gmoss(a)TheWorld.com> writes:
>
> > I'm running a 2.6 kernel with the realtime module added. I'm able to get
> > jack running with realtime capabilites using
> >
> > jackstart -r -d alsa
>
> I assume you're actually using -R. What happens if you run jackd
> instead of jackstart?
Yes sorry about that, jackstart -R -d alsa
I think with Jackd I don't get realtime:
registered builtin port type 32 bit float mono audio
required capabilities not available
capabilities: =
loading driver ..
new client: alsa_pcm, id = 1 type 1 @ 0x8055830 fd = -1
creating alsa driver ... hw:0|hw:0|1024|2|48000|0|0|nomon|swmeter|-|32bit
control device hw:0
configuring for 48000Hz, period = 1024 frames, buffer = 2 periods
Couldn't open hw:0 for 32bit samples trying 24bit instead
Couldn't open hw:0 for 24bit samples trying 16bit instead
Couldn't open hw:0 for 32bit samples trying 24bit instead
Couldn't open hw:0 for 24bit samples trying 16bit instead
new buffer size 1024
JACK: unable to mlock() port buffers: Operation not permitted
registered port alsa_pcm:capture_1, offset = 4096
registered port alsa_pcm:capture_2, offset = 8192
registered port alsa_pcm:playback_1, offset = 0
registered port alsa_pcm:playback_2, offset = 0
++ jack_rechain_graph():
client alsa_pcm: internal client, execution_order=0.
-- jack_rechain_graph()
cannot set thread to real-time priority (FIFO/20) (1: Operation not
permitted)
no realtime watchdog thread
cannot use real-time scheduling (FIFO/10) (1: Operation not permitted)
2401 waiting for signals
load = 0.0516 max usecs: 22.000, spare = 21311.000
But I can with JAckstart
#jackstart -v -R -d alsa
registered builtin port type 32 bit float mono audio
capabilities: = cap_setpcap,cap_ipc_lock,cap_sys_nice,cap_sys_resource+eip
loading driver ..
new client: alsa_pcm, id = 1 type 1 @ 0x8055830 fd = -1
creating alsa driver ... hw:0|hw:0|1024|2|48000|0|0|nomon|swmeter|-|32bit
control device hw:0
configuring for 48000Hz, period = 1024 frames, buffer = 2 periods
Couldn't open hw:0 for 32bit samples trying 24bit instead
Couldn't open hw:0 for 24bit samples trying 16bit instead
Couldn't open hw:0 for 32bit samples trying 24bit instead
Couldn't open hw:0 for 24bit samples trying 16bit instead
new buffer size 1024
registered port alsa_pcm:capture_1, offset = 4096
registered port alsa_pcm:capture_2, offset = 8192
registered port alsa_pcm:playback_1, offset = 0
registered port alsa_pcm:playback_2, offset = 0
++ jack_rechain_graph():
client alsa_pcm: internal client, execution_order=0.
-- jack_rechain_graph()
2490 waiting for signals
load = 0.0820 max usecs: 35.000, spare = 21298.000
> > from the command line. But if I run qjackctl, jack starts up, and I get
> > "Could not connect to JACK sever as client." error.
> >
> > Alsaplayer seems to be ok, but ardour is a no go.....
I should add that alsaplayer is going through jack, from best I can tell -
I am getting audio from it, and the "alsaplayer -o jack" command didn't
produce a "Failed to load output plugin "jack". Trying defaults." reply
> > I've tried mobprobing as any=1 and with the gid option too.
>
> Are you using allcaps=1?
yes, and I'm having the same problems
> This could be caused by other problems, such as the JACK --tmpdir
> directory.
This seems possible, perhaps alsaplayer doesn't try to make use of the
tmpdir?
> What version of JACK? Of realtime-lsm?
jackstart -V
back from read, ret = 1 errno == Success
jackd version 0.98.1
default tmp directory: /tmp
Realtime is realtime-lsm-0.1.1
> What do these commands print?
>
> $ lsmod | grep realtime
realtime 10128 0
> $ grep Realtime /var/log/messages
Jun 1 19:07:05 localhost kernel: Realtime Capability LSM exiting
Jun 1 19:14:37 localhost kernel: Realtime LSM enabling all capabilities
Jun 1 19:14:37 localhost kernel: Realtime LSM initialized (group 101,
> --
Is there any other info I can provide that might help?
Thanks,
-G.
Hello y'all - this is WAY off topic but I am not connected to any web
developement lists - does anyone know of some, where people are great like
here and can help with web devel questions?
thanks in advance!
ps:
what my question is is this: I want to have a script that will rotate
between different content pages, so say user 1 hits the site, they see "blah
blah1" and user 2 hits the site they see "blah blah2" and so on...