Have you tried starting Qsynth as root rather than as "user"? The failure to set the scheduling params indicates you don't have sufficient permissions and logging in as, or 'su'ing root would at least overcome that issue. Not sure why the program exits though as I did not write it - perhaps it will not start the midi process unless it has rescheduled? Pure conjecture, I carry on regardless although it could be argued that.....
And yes, linux software is rather terse. Its not really written for users. I have submitted a couple of times on how it could be made easier however most linux users seem to equite 'easier' to 'dumbing down', which I disagree with.
Nick.> Date: Wed, 22 Aug 2007 21:24:24 +0100> From: simon(a)systemparadox.co.uk> To: linux-audio-user(a)lists.linuxaudio.org> Subject: Re: [LAU] Synths (software vs hardware, speed)> > Matthias Schönborn wrote:> > On Wednesday 22 August 2007 14:22:41 Simon Williams wrote:> >> I'm now at the point where I can play midi files, and I have a midi> >> keyboard connected and can play sounds to and from that.> >> Timidity is rubbish, frankly, so I'm using fluidsynth, which works fine.> >> However, for some reason I can't get qsynth to work.> > > > What soundcard do you use?> nforce (on an asus mobo).> > > Do you use jack?> Qsynth won't start without it.> > > If so, what settings do you use?> SERVER_PARAMS="-s -d alsa"> DRIVER_PARAMS="-d hw:0 -p 1024"> > That's all I could find- and even those are the defaults. I'm running it > with qjackctl.> > With the default setting (alsa_seq), qsynth gives this:> > 21:14:04.045 Qsynth1: Creating synthesizer engine...> 21:14:05.078 Qsynth1: Creating audio driver (jack)...> 21:14:05.144 Qsynth1: Creating MIDI router (alsa_seq)...> 21:14:05.150 Qsynth1: Creating MIDI driver (alsa_seq)...> 21:14:05.153 Qsynth1: Failed to create the MIDI driver (alsa_seq). No > MIDI input will be available.> JACK tmpdir identified as [/tmp]> fluidsynth: warning: Couldn't set high priority scheduling for the MIDI > input> fluidsynth: warning: Couldn't set high priority scheduling for the MIDI > input> fluidsynth: panic: Couldn't create the midi thread.> > Changing the setting to alsa_raw or oss makes little difference to these > errors. My usual rant when dealing with linux sound issues is about the > total lack of any meaningful errors when something doesn't work.> > I get the impression that qsynth has a major bug- I've posts on other > mailing lists from people with similar problems- fluidsynth works fine, > but qsynth is broken. If anyone can shed some light on this that would > be great.> > Thanks> Simon> _______________________________________________> Linux-audio-user mailing list> Linux-audio-user(a)lists.linuxaudio.org> http://lists.linuxaudio.org/mailman/listinfo.cgi/linux-audio-user
_________________________________________________________________
Make every IM count. Download Windows Live Messenger and join the i’m Initiative now. It’s free.
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Splank and crunck makes four is completely perfect.
I love the sounds in aSidBreaks, but it's a bit too broken to be as
enjoyable as Splank and crunck. Not that it's bad, just not easy
listening.
Column is very neat. I love the lead/bass thing.
Skunk is beautiful. It's along the lines of Windowlicker - broken and
wrong, but very very beautiful.
Track 8 on cd001 is also very very nice.
9, too.
And 10 is a nice jolly mix of stuff that works very well. The
distorted vocals are great.
Some very very nice stuff!
James
----- Forwarded message from Chris McCormick <chris(a)mccormick.cx> -----
From: Chris McCormick <chris(a)mccormick.cx>
Subject: Re: [LAU] Chris McCormick
To: james(a)dis-dot-dat.net
Cc: A list for linux audio users <linux-audio-user(a)lists.linuxaudio.org>
Date: Tue, 21 Aug 2007 23:34:26 -0400
User-Agent: Mutt/1.5.13 (2006-08-11)
Message-ID: <20070822033426.GA25804(a)podsix.com.au>
On Tue, Aug 21, 2007 at 08:01:45PM +0100, james(a)dis-dot-dat.net wrote:
> I love your beats.
Oooh, way to make me blush. :)
> Just listened to Meteors versus Dinosaurs.
>
> Have you got more? Overall, the track wasn't quite right for me - not
> easy enough to listen to - but I'm guessing that i'd love your more
> restrained stuff if you have any.
I've been sitting on this one for a very (very) long
time, but today is as good a day as any to release it:
<http://sciencegirlrecords.com/chr15m/music/CD005/Chris%20McCormick%20-%20Es…>
You might also like the more beats oriented stuff on my
first couple of disc-fulls of music, which you can find here:
<http://sciencegirlrecords.com/chr15m/page/discography/CD001>
<http://sciencegirlrecords.com/chr15m/page/discography/CD002>
I'm currently in the process of naming and updating the id3 tags on CD001
so the files may shift around and disappear over the next couple of days,
but if you refresh the page it should all be in order.
Thanks for enjoying, and most especially saying something! As much as I
hate to admit being driven by ego, it really is such a huge motivator.
Best,
Chris.
-------------------
http://mccormick.cx
----- End forwarded message -----
I've just found that I have a game port midi adapter, so while I wait
for my USB-midi device to arrive I thought I'd give it a try. However, I
cannot for the life of me find any information on how I go about
enabling midi on my nforce mobo.
A post on the alsa mailing list said that adding the option
mpu_port=330x0 to the snd-intel8x0 module worked, but that wasn't
specifically for an nforce, and my version of snd-intel8x0 doesn't
actually have any mpu options!
I've loaded every module I could find called snd*midi or snd*seq, and
still the only output I get from aconnect -io is:
client 0: 'System' [type=kernel]
0 'Timer '
1 'Announce '
client 128: 'TiMidity' [type=user]
0 'TiMidity port 0 '
1 'TiMidity port 1 '
2 'TiMidity port 2 '
3 'TiMidity port 3 '
Any ideas?
Thanks
Simon
First, my instructions in /etc/profile aren't being loaded. I have
OPCODEDIR for Csound exported there, and it never does. Anyone know what
might cause this?
I'm also confused about setting multiple values for a variable. I try $:
export LADSPA_PATH="/usr/lib/ladspa:/home/chuckk/ladspa", and this causes
Rosegarden to crash on start. All of the Debian binaries for LADSPA plugins
install to /usr/lib/ladspa, so I'd rather not just make
LADSPA_PATH=/home/chuckk/ladspa. But Csound's csLADSPA turns .csd files
into plugins, and I have to be able to edit them, preferably not always as
root. Is my syntax correct?
-Chuckk
--
http://www.badmuthahubbard.com
I posted my thoughts on this over a year ago and was a bit disappointed
at the lack of interest. Maybe you were all deeply engrossed in other
projects so I've posted it again!
Well I don't know if this term actually exists or if I've just invented
it!
This is an idea I've thought about for quite some time, years in fact,
but don't have the programming ability to try to put it into practice.
I'd be very interested in other people's thoughts on it.
Preamble over :)
All the quantisation systems I've seen so far only work if the music
has reasonably constant timing, and then produces much too rigid a
structure for my tastes.
I usually record live work without a metronome as these always inhibit
me. However I find that in a very long piece, I sometimes gradually
speed up or slow down. This is often only noticable if you go back to
the start of a piece and replay it immediately it has finished. If
'standard' quantisation is applied to this then the results can be
quite grotesque as notes progressivley fall outside the quantisation
capture range and get placed into the wrong positions.
What I would like to see is quantisation algorythm the detects trends
rather than absolute values, then progressively applies small
corrections to keep overall timing correct. (it would of course have to
operate over all tracks simultaneously).
For example, the musician could put markers on notes in, say, an
accompaniment section, that aught to fall on the first beat of each
bar. The quantisation would then stretch or shrink the time positions
so most of these fit. I say 'most' as it is the trend we are
controlling not specific notes. Intervening notes of ALL tracks are
then adjusted a proportionate amount. Later bars can then be
interpolated and occasional bars that don't actually have a note on the
first beat will still be adjusted based on averaging. Deliberate note
delays, syncopation etc. would then be perfectly preserved and the
music would retain its liveliness.
Having the musician place these markers rather than some automatic
system, means that not only are the correct notes used as a reference,
but the music can be brought into line even if it initially has
absolutely no relationship with the bar lines in the sequencer (this
happens to me a lot when I try to record live). The quantisation system
would match marked notes against 'real' bar lines. Overall timing can
then of course be adjusted by altering the beat rate.
This whole idea could then be turned on it's head. I find it VERY hard
to get several tracks to slow down at the end of a piece and stay
'together'. This quantisation system could do just this by having
'target' time/beat rates at the start and end of the section that is to
be slowed (or speeded up). Once again, natural variations would be
preserved and only the overal trend would be adjusted.
--
Will J Godfrey
http://www.musically.me.uk
Le 21 août 07 à 23:18, linux-audio-user-request(a)lists.linuxaudio.org
a écrit :
>
> ------------------------------
>
> Message: 2
> Date: Tue, 21 Aug 2007 21:20:35 +0200
> From: Fons Adriaensen <fons(a)kokkinizita.net>
> Subject: Re: [LAU] Re: Re: Jackmp - libjackdmp.so (Fons Adriaensen)
> To: linux-audio-user(a)lists.linuxaudio.org
> Message-ID: <20070821192035.GA6212(a)linux-2.site>
> Content-Type: text/plain; charset=iso-8859-1
>
> On Tue, Aug 21, 2007 at 07:02:27PM +0200, Stéphane Letz wrote:
>
>>> Because it doesn't read the sample rate, but counts on jackd
>>> calling the sample rate callback. Apparently jackdmp doesn't
>>> do this. The main code thinks that the jack interface failed
>>> when it sees a sample rate of zero. Then it opens OSS (!!!)
>>
>>
>> Just stupid. sample rate callback has been deprecated i think....
>>
>> i guess reading the sample rate with jack_get_sample_rate should work
>
> Yes, I added that, and now Alsaplayer works with jackdmp.
Can you send a patch to AlsaPlayer developers?
>
> So the only thing holding me from switching to jackdmp now is
> the system: ports issue.
Now it works on SVN.
>
> Why switch ? Not because there's any problem with jackd, I just
> want to use jackdmp for a while for all daily work to put it to
> the test, but the precondition for that is that everything just
> works.
>
Stephane
If you enjoy my music I'd like to suggest a peek at my updates page:
http://www.musically.me.uk/updates.html
There are a few new pieces as well as some re-polished ones.
If you haven't heard my music I would also like to suggest a visit.
And if you don't like my style, have a look to see if there's something
new there :)
Oh, and if any of you use ZynAddSubFX, my improved collection of
instrument patches is at:
http://www.musically.me.uk/Collection.zip
--
Will J Godfrey
http://www.musically.me.uk
>
> Message: 5
> Date: Tue, 21 Aug 2007 12:18:49 +0200
> From: Fons Adriaensen <fons(a)kokkinizita.net>
> Subject: Re: [LAU] Re: Jackmp - libjackdmp.so
> To: linux-audio-user(a)lists.linuxaudio.org
> Message-ID: <20070821101849.GC6214(a)linux-2.site>
> Content-Type: text/plain; charset=iso-8859-1
>
> On Tue, Aug 21, 2007 at 11:39:12AM +0200, Fons Adriaensen wrote:
>
>> - Why does Alsaplayer wrongly think it can't connect to jackdmp ?
>
> Because it doesn't read the sample rate, but counts on jackd
> calling the sample rate callback. Apparently jackdmp doesn't
> do this. The main code thinks that the jack interface failed
> when it sees a sample rate of zero. Then it opens OSS (!!!)
Just stupid. sample rate callback has been deprecated i think....
i guess reading the sample rate with jack_get_sample_rate should work
>
>> - Why, if that happens, it falls back on ALSA while the user has
>> clearly requested to use JACK ?
>
> IMHO it should _never_ do this. Starting to play on a interface
> that the user has not specified - and even specifically excluded
> by requesting another one - is as wrong as writing to a random
> existing file when the one specified can't be opened.
>
Hum also...
Stephane
hi list,
finally i got to the point to convince my girlfriend to switch from win
to linux, so i am looking now in the internet for a good-price-laptop
for her.
since i am on ibm here and very happy with this, we are looking now for
'lenovo 3000 n100' here:
http://www.notebooksbilliger.de/product_info.php/lenovo_3000_n100_megaaktio…
(sorry the link is in german).
so the question i have is first about the processor:
Intel Core™ Duo für Mobile
<javascript:popupWindow('http://indigo.intel.com/Syndication/DistributeModule.aspx?a=165&m=133&l=6&p…','infocontent','top=200,left=100,resizable=1,scrollbars=0,width=960,height=655');>
T2450 2x 2,0 GHz
does anyone can recommend this processor or is there someone who could
say "don't get that!"
i have a celeronM here and that works pretty good for me, but i do not
have any experience in setting up linux for dual-technologies.
another question is about the hd:
there is not specified which hd is build in, but it only makes 5400rpm.
should we expect any problems on a hd with that speed?
so in generally i would go for some hardware, where we can get ubuntu or
sth similar installed, without too much problems. in the end the machine
should be used for multimedia and some networking.
i am really looking forward for some answers on this!
cheers,
doc
--- Matthias =?iso-8859-1?q?Sch=F6nborn?= <mbs1303(a)gmx.de wrote:
Hi list,
>
> Lately I've got some issues with ardour repeatedly giving an error
message
> saying that ardour wasn't able to continue playback because the
disk was too
> slow and ardour couldn't fetch the data. How can this be
fixed?
>
> I'm using Kubuntu Feisty with a low latency kernel, and in my
setup I run
> Jack, QSynth, Hydrogen, Rosegarden, and Ardour - is the load
just too heavy
> (I can hardly imagine)? Ah, and I'm using an external M-Audio
Interface if
> that's of any importance.
>
> Regards,
> Matthias
>
_______________________________________________
> Linux-audio-user mailing
list
> Linux-audio-user(a)lists.linuxaudio.org
> http://lists.linuxaudio.org/mailman/listinfo.cgi/linux-audio-user
>
A message like this really does mean that your hard disk can't keep
up with the demand for data. The cause of this however can come from several
sources. Since you are running several programs at once, most likely synced
together through the Jack transport, this means that they are probably all
trying to access data from the drive(s) at the same time, and as you know,
audio can be rather dense when uncompressed. So the solution? It is often
recommended to have a separate disk for recording audio and whatnot to, that
way, when linux needs to read info for running a program it wont interfere
with the audio data access on another disk. Of course faster sata/serial disk
drives will help as well.
