Greetings:
I know this is possible but I can't remember how. I want to send this
string to a MIDI device:
f0 7f f7 06 02 f7
Anyone know the command syntax ?
Best,
dp
Hello everybody,
I use mandrake and i want to switch to debian, i have mdk 10.0 and debian
sarge currently installed.
As a musician i use band in a box with wine alsa and timidity with the same
kernel + config (stock 2.6.8 no preemption) on both system.
However, the sound is great on mandrake whereas it is crappy (glitches,
scratches, not regular) on debian (even trying the LD_ASSUME_KERNEL=2.4.22
trick). I don't use jack.
How comes there is such a difference between two system that uses the same
kernel and config?
Where should i look since the kernel and modules are the same (are they?)?
Thanks
rico
Hi.
I tried to compile the latest MPlayer (1.0pre5) with JACK output enabled
(./configure did detect JACK after I installed the bio2jack, but not
before)
The compilation fails at linking giving me something about functions
JACK_GetVolume
JACK_SetVolume
not found. It must be something with the bio2jack install, since it has
those functions.
Has anyone succeeded in compiling the MPlayer with JACK output enabled?
Tommi Uimonen
Several of us are planning to get together at the Dog and Duck
this week. It looks like we'll be there around 6:00 PM on
Thursday. We haven't discussed recognition signals yet, so
stay tuned...
Bill
Hello All,
I have a bunch of old consumer sound cards. SBLive, Vibra128 and ensoniq 1370(es1370). I went
through quite a number of rounds trying out these cards and found that IMHO the es1370 gave me the
best sound amongst these cards. So i went ahead and try to start jack according to the
capabilities of this card.
With this card, i find that once i start jackd like this:
jackd -v -R -d alsa es1370 -r 44100
I get lots of xruns immediately.
Then i tried with the n=4 (ie 4 periods per hardware buffer), i.e "jackd -v -R -d alsa es1370 -n
4" and I am able to get rid of the xruns. But i see lots of "late driver wakeup: nframes to
process=2048" on the jackd output. What does this message mean?
For recording into ardour, n=4 and the default frames per period (1024) gives me quite significant
latency when recording. e.g when I pluck a note on the electric bass i can hear it on the line-in
monitor first and then from the capture slightly later.
As a compromise, i set the frames per period to either 64, 128 or 256. But I still get the "late
driver wakeup" message and occasional xrun when i quit ardour or sometimes even hydrogen. I notice
that when quitting jackd programs there will sometimes be a few xruns. Is this normal?
Seems like ideal to start n=2 but this card don't seem to allow me to do that. I could be wrong
but i think for low latency n=2 is ideal.
If i have saved an ardour project with Tim Goetze's plugin activated in some tracks, ardour may
report that it is too slow or (something like that) when i reload that project file. I suspect
this is to do with some ladspa plugins requiring low latency which my setup is not able to give.
I know this card is old, but it does give a great sound. I used SBLive previously and don't seem
to have these latency issues (able to start jack with n=2). For owners of this card, any sound
advice to get the most out of it?
Thank You very much,
Louis
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JACK RELEASE 0.98.16
JACK is a low-latency audio server, written primarily for the GNU/Linux
operating system. It can connect a number of different applications to
an audio device, as well as allowing them to share audio between
themselves. Its clients can run in their own processes (ie. as normal
applications), or can they can run within the JACK server (ie. as a
"plugin").
JACK is different from other audio server efforts in that it has been
designed from the ground up to be suitable for professional audio work.
This means that it focuses on two key areas: synchronous execution of
all clients, and low latency operation.
JACK is available at http://jackit.sf.net
--CHANGES--
Buffer resizing enabled by default.
Added jack_ringbuffer_peak() to API.
Added jack_last_frame_time() to API.
--verbose will print the maximum usecs used on jackd termination.
Better compatibility with NPTL.
--version output changed for easier parsing.
New --unlock/-u option so that large libraries (gtk, qt, fltk, wine)
aren't memlocked.
Jack's tmp files now have the uid appended to them, so if there is a
crash, and then another user tries to use jack, it will still work.
New jack_create_thread() cleans up threading for portability. Available
for use by jack clients too.
New CoreAudio driver from the Jackosx project included in jack tree.
Prettier configure output.
and of course, updated documentation, better error reporting, and misc
internal fixes and cleanups.
Greetings,
I no this is not strictly (or perhaps vaguely) linux
audio related, but I have seen mention of folks here
using MAC OS and Audacity, so I'll ask away...
I am able to record using the line input into a 12"
powerbook (1GHz / Aluminum / Panther OS)
Is there any way to get Audacity to use the onboard
sound card's output as a monitor channel?
My minimal portable config is often using just a cable
from the headphone out of a source and thene there is
no monitor available.
Thanks in advance for any hints.
aloha,
dave
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Hi,
 I want to use the JACK soundserver. I'm able to connect to the normal analog
output but not to spdif.
I use the snd-intel8x0 driver for my onboard nforce2 soundchip.
> ash-2.05b# cat  /proc/asound/devices
> Â 1: Â Â Â : sequencer
> Â 0: [0- 0]: ctl
> Â 18: [0- 2]: digital audio playback
> Â 25: [0- 1]: digital audio capture
> Â 16: [0- 0]: digital audio playback
> Â 24: [0- 0]: digital audio capture
> Â 33: Â Â Â : timer
> bash-2.05b#
Using the alsa output plugin in xmms I can choose hw:0,2 then the sound comes
out through spdif, but I don't know how to do this with JACK.
I tried qjackctl (grafical controller for JACK and ALSA) and qjackconnect
(grafical patchbay) and via console by typing 'jackd -d alsa -d hw:0,2' (same
with hw:1 and hw:2) but this doesn't work, neither as user nor as root.
 via console I get the following output:
 > ash-2.05b$ jackd -d alsa -d hw:1 -S
>
> Â jackd 0.98.1
> Â Copyright 2001-2003 Paul Davis and others.
> Â jackd comes with ABSOLUTELY NO WARRANTY
> Â This is free software, and you are welcome to redistribute it
> Â under certain conditions; see the file COPYING for details
>
> Â loading driver ..
> Â creating alsa driver ... hw:2|hw:2|1024|2|48000|0|0|nomon|swmeter|-|16bit
> Â jackd: pcm.c:690: snd_pcm_nonblock: Assertion `pcm' failed.
> Â Abgebrochen
> Â bash-2.05b$
qjackctl displays following errors:
> 13:16:14.915 /usr/bin/jackd -R -dalsa -dhw:0,2 -r48000 -p1024 -n2 -S -i2
> -o2 13:16:14.936 JACK was started with PID=12828 (0x321c).
> jackd 0.98.1
> Copyright 2001-2003 Paul Davis and others.
> jackd comes with ABSOLUTELY NO WARRANTY
> This is free software, and you are welcome to redistribute it
> under certain conditions; see the file COPYING for details
> loading driver ..
> apparent rate = 48000
> creating alsa driver ... hw:0,2|hw:0,2|1024|2|48000|2|2|nomon|swmeter|-|
16bit
> ALSA lib pcm_hw.c:1155:(snd_pcm_hw_open) open /dev/snd/pcmC0D2c failed: No
such file or directory
> jackd: pcm.c:690: snd_pcm_nonblock: Assertion `pcm' failed.
> 13:16:14.994 JACK was stopped successfully.
> 13:16:16.966 Could not connect to JACK server as client.
Is there some special configuration needed in '/etc/asound.conf'
Please help me, if it works I want to write a howto in the gentoo user forum
cause there is to little information about JACK and JACK is definitly a great
leap forward considering the sound quality.
Ruben
--
Wi....s is like a frozen turd in winter.
You can't excactly see what it is,
but when you take it home to check it up
its malodor blurts out it's true nature.
Dear all,
Roland UA-100 plugged into PC, usbview detects interface.
aplay -l doesn't report any knowledge of the device.
I believe I have loaded the appropriate modules, snd-usb-audio but the
system resolutely refuses to make music as it has before. I would
include the appropriate modules.conf files to show but this E-mailing
machine and the linux box are not connected and I feel cross reading
would be a little error strewn.
Can anyone outline a procedure for checking at which point a problem
might occur and, if possible, a list of command line utilities that can
reveal precisely what components are loaded. Is it possible to load
modules incompletly such that a component identifies it's presence but
isn't actual functioning. If this is so how does one check the dependancies?
I understand that amix and alsamixer won't recognise the presence of a
USB connected audio interface, is this the case?
Any help appreciated.
Chris Lyon