hello friends,
i have successfully built timidity (2.12.0), then jack (0.34.0), and
finally muse --with-jack (0.6.0pre5).
timidity works by itself, i can play midi files and it sounds. my intention
is to use it as a software synthesizer for muse, so i launch it with the
option -iA.
but muse also requires jack to be running, so first i invoke jack like
this:
jackd -R -d alsa -d hw:0
and then i can run muse. the problem is that as soon as i have jack running
timidity doesn't sound.
all right, the real problem is that i never got to fully understand jack,
so i have no idea how it works or how i can make things work together. as
soon as i begin more confused than before so i give up.
any charitable soul could give me a quick hint on how i should invoke
timidity and/or jack so that i can use the combination of the two as a
software midi device for muse? (i used to use it like this in the past, but
that was without jack coming into play)
thanks for any help.
best,
lj
hi all,
i know that music theory is off-topic for this list.
generally, how tolerated are off-topics posts on this list? i've been
experimenting with some new theory i've learned and wanted to know why
something sounds terrible.
if this list wants to be totally on-topic (which is understandable), can
someone recommend a flame-free mailing list for people who want to ask
questions about jazz theory?
my teacher doesn't have an email address, and i'd like to get an answer
before my next lesson. :)
sorry for asking, but i'm very new to this list and don't know the group
dynamic yet.
pete
--
Fingerprint: B9F1 6CF3 47C4 7CD8 D33E 70A9 A3B9 1945 67EA 951D
Hi,
If anyone is using this card and actually has tried to get MIDI
working, could you please drop me a note and let me know if your results
were positive or negative?
I've been messing with this for a few days and it's just not
happening for me.
Thanks,
Mark
Hello all,
here we go, another release of the hackish jack module player.
What's it about?
JMP is a MOD-player outputting each channel individually to jack. This is achieved using xmp module-player's channels soloing feature, and ecasound's multitracking abilities.
New features in v0.1b
-channel-naming: now you can see channels in jack with the names you assign to them in commandline
-sticking channels together for better performance: maybe you don't need EVERY channel showing separately in jack
-better error recovery: should leave no dead processes hanging around
(NOTE: kills all xmp processes on exit, be careful! ecasound is left in peace..)
--------------------
# ./jmp.py --help
Jack Module Player v0.1b
usage: jmp.py [options] module-file(s)
jmp.py options:
-b, --buffer set buffersize
-r, --rate set sample rate
-c, --channels override module's number of channels
-N, --names name individual channels (or channels sticked)
with comma separated list of names
e.g. bassdrum,hihat,snare
-S, --stick stick individual channels together
with colon separated list of channel-numbers
e.g. 0-3,5:6,7:8 (channel-numbering starts at 0)
-l, --looping loop module
-n, --noconnect don't connect jack-ports to alsa-playback
-q, --quiet
-v, --verbose
NOTE! external programs required:
Extended Module Player: http://xmp.sourceforge.net/
Ecasound: http://eca.cx/
# ./jmp.py /home/janne/biisit/wik.xm --names bassdrum,hihat,hihat,snare -c 4
...
# jack_lsp
alsa_pcm:capture_1
alsa_pcm:capture_2
alsa_pcm:playback_1
alsa_pcm:playback_2
ecasound:bassdrum_1
ecasound:hihat_2
ecasound:hihat_3
ecasound:snare_4
-------------------
Get it from
http://www.pp.htv.fi/jhalttun/jmp.py
..still coding when should be making music :)
janne
hi,
i want to connect my keyboard to my pc (sb Live!) in oder to use it
as a softwaresynthy.
i've read aconnect 64:0 65:0 should do this ... but:
stef: ~ $ aconnect -lio
client 0: 'System' [type=kernel]
0 'Timer '
1 'Announce '
Connecting To: 63:0
client 65: 'Emu10k1 WaveTable' [type=kernel]
0 'Emu10k1 Port 0 '
1 'Emu10k1 Port 1 '
2 'Emu10k1 Port 2 '
3 'Emu10k1 Port 3 '
i can't find this port 64:0.
here is my the alsa passage of my modules.conf:
# ALSA portion
alias char-major-116 snd
alias snd-card-0 snd-emu10k1
# module options should go here
# OSS/Free portion
alias char-major-14 soundcore
alias sound-slot-0 snd-card-0
# card #1
alias sound-service-0-0 snd-mixer-oss
alias sound-service-0-1 snd-seq-oss
alias sound-service-0-3 snd-pcm-oss
alias sound-service-0-8 snd-seq-oss
alias sound-service-0-12 snd-pcm-oss
#my own settings
alias midi snd-synth-emu10k1
alias snd-synth-midi snd-seq-emu10k1
alias snd-seq-midi snd-seq-emu10k1
(i tried it also without the "alias midi snd-synth-emu10k1" line)
i compiled alsa with
./configure --with-cards=emu10k1 --with-sequencer=yes;make; make install
i use debian sid. what am i doing wrong?
with muse i can use the /dev/snd/midiC0D0 for recording
thank you
stefan
>> The Midiman DiO 2448 seems to be the cheapest sound card with digital
>> I/O and I use it with the OSS drivers, paid for, from 4Front
>> technologies at http://www.opensound.com. The card has analogue output
>> but no analogue inputs at all.
>> The Midiman 2496 is more expensive but it may be easier to get drivers
>> working on it.
>
>Looking at the ALSA soundcard matrix and associated doco, the 2496
>doesn't have SPDIF support yet. However the 2448 (CMI8738 chip) seems
>to be well supported by the current drivers (0.9.0rc6).
I thought that I remembered that the Midiman Audiophile 2496 had been working with spdif for a while for me at least, though I have the dat on the Delta 1010 in the music room now.
I know that there was a lot of ups and downs with the drivers and spdif at first.
I did have to cook up my own .asoundrc for it, but that's kind of expected with a mutichannel chip being used on a simple card.
Tracey
Hi all,
This might seem like a silly question, but I'm having an
unexpectedly hard time putting together something I consider
to be quite simple in essence.
The intent: play music from PC (eg .ogg/.mp3 etc) via S/PDIF to
an external DAC, and thence to sound system.
The problem: finding a soundcard supported by Linux and which
offers the option of unadulterated digital output. Ie. I want the
soundcard to simply stream the bits at 44.1Khz (or 48) and not
shag around with it in any way.
My current setup is a compromise. I have onboard sound with
a VIA8233 (Avance AC'97) 6ch chipset, and am using Windows
XP with this, since the latest ALSA driver doesn't produce any
sound on the S/PDIF output (although the optical port lights up).
Trouble is, I don't trust the Windows driver, since the control
panel for it allows me to select DSP environments and use a
graphic equaliser. With the environment set to "none" and the
equaliser off is it providing a "pure" bitstream un-molested in any
way? Who knows.
So, the question to you is simply: do you know of any soundcards
which would fit the bill - ie. with a fully working Linux driver, and
known to be configured to produce "pure" digital out.
Cheers,
Paul.
--
Lubarsky's Law of Cybernetic Entomology:
There's always one more bug.
Good morning,
I read (somewhere) that it is possible to make a directly useable Hard
Disk 'Image File' from an Audio-CD.
I am familiar with mkisofs for sets of data files and mounting the
iso-9660 images as loop(ed) devices to access the files within the
resulting image file.
Also, I know that cdrdao builds a .bin temp file when you make a copy
...
Are these files interchangeable? How would one mount an Audio 'image
file' to play it, since you usually do not mount Audio CD's - players
like xmms just use the raw device (right?)
Any hints or pointers (urls) would be appreciated.
aloha,
dave
Hi,
ams-1.5.5 is available from http://www.suse.de/~mana/kalsatools.html.
It fixes a serious bug in synth.cpp which causes the machine to freeze
when ams is started as root.
Some example patches for the bode frequencer LADSPA plugin included in
the new 0.3.3 version of Steve's plugin set have been added as well.
Have fun !
Matthias
--
Dr. Matthias Nagorni
SuSE GmbH
Deutschherrnstr. 15-19 phone: +49 911 74053375
D - 90429 Nuernberg fax : +49 911 74053483
Hello list!
(I have searched for an answer to this question in
google and some other lists but didn't find something
helpful... sorry if this has been asked before)
i want to have realtime-encoding of my soundcard
line-in with sox and lame (or something similar - i
need command line tools). Now i use
rec -t wav -r 44100 - | lame -m m -a -b 64 -
(i need 44.1 khz/64 kbit). But all 5 or 7 hours sox
crashes; there's an older thread on the sox mailing
list:
http://sourceforge.net/mailarchive/forum.php?thread_id=1083489&forum_id=3958
which says that this is due to a counter in the wav
header which overflows and that you should use .raw
instead of .wav. But whatever i use doesn't sound
right:
sox -t ossdsp -w -s /dev/dsp -t raw -r 44100 -c 2 - |
lame -r -m j -b 64 -x -s 44.1
--resample 44.1 -
It sounds dump:
http://www.kochenmachtspass.de/test3-ms.mp3
I have tried many other options like -c 1 (leads to
noise) but i cannot find a solution... Maybe someone
of you knows a solution?
Thanks
Chris
=====
http://www.crupp.de
__________________________________________________
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