Howdy,
(origonally posted to gmane.linux.agnula.general,
it was suggested to try this list as well)
I need to monitor and record a number of
radio and telephone inputs.
When the conversation is finished, I need to
compress the audio and log the time and source
of the recording.
I installed Agnula DeMuDi on a spare machine,
it seems to know all about the M-Audio Delta 1010lt
card which is in there.
That's about all I know.
Questions;
- am I asking these questions on the correct list?
if not, please give directions.
- does a project exist which seeks to do multi-channel
voice actuated recording?
- if not, is this a reasonable task to attempt for a
low/mid level Python programmer?
Thanks,
Kent
Hi friends,
I prepare an installation with 10 xmms playing soundfiles through jackd &
ardour. I use the last xmms-jack 0.14 plugin but it seems to have a problem with
the biojack, each time I launch a second xmms, biojack want to re-number as the
first one and not give it a new biojack number... So I can't use different xmms
at the same time. Did some of you used the plugin recently?
I am trying to patch the bio2jack source to allow xmms to play multiple session
at the same time and give a new biojack number... but I can't contact the
programmer "Chris Morgan" (his email is protected from spam abuse, and it
rejected my email...).
So if someone has an idea, thanks for your help!!
cheers
julien
APODIO : http://www.apodio.org
Playback and capture streams can never be exactly synchronized -- even
if both use the same sample rate, packet sizes are not guaranteed to
be the same. This means that it's possible to use smaller buffers
with Jack by using only one direction (-P or -C instead of -d), if
only one direction is needed.
Due to synchronization issues with the host controller, stopping and
restarting a stream will take several milliseconds. This means that
it is often preferrable to use Jack's --softmode option to reduce the
length of the gap when an underrun happens.
The snd-usb-audio driver in CVS, or in ALSA 1.0.10rc1 (to be released
Real Soon Now(TM)) has several improvements:
- Maximum additional latency of captured data reduced to 1 ms (this
was a bug in the earlier version).
- Consistent handling of USB packets that go over the buffer boundary.
Sample rates that aren't a multiple of 1000 now work just as well as
sample rates that are. This affects programs (like Jack) that don't
use a buffer size that is a multiple of rate/1000, too.
- The nr_packs parameter can now be changed after the module has been
loaded, e.g.:
echo 1 > /sys/module/snd_usb_audio/parameters/nr_packs
This change will take effect when a stream is started the next time.
Regards,
Clemens
Hello
And very many thanks for all the c-port2000 answers, it's been some time that
I've encountered this friendly mailing list :-).
I'm looking for some simple software to accomplish the following:
- do realtime monitoring(VU) and mixing
- output mixed data for headphones-monitoring (using same card, I have the
infamous st-audio rig, on which I didn't manage to get _any_ of the MIDI
ports to work)
- optionally route some of the input channels trough LADSPA plugins
A Gtk+2 GUI would be much preferred to anything else.
Has anyone seen such a beast? If not, how does this sound:
- a jack client that will get the input data
- will do the mixing and drive the stuff trough ladspa
- output result using jack (this should be possible, only I skimmed trough
some jack-related docs, difficult to get handle on it)
- display VU/implement GUI for levels using GTK+ (2.x).
Does anyone see any architectural problems with this approach? I'd guess that
implementing mixing using a jackd plugin would be more efficient, but is it
worth it and how to connect the controls to the GUI?
ak.
Hello folks,
I have two midi controller connected to my UM-2 usb midi interface
And I have a lot of trouble to figure out how to configure ecasound to use
them.
here is the aseqdump outputs :
$ aseqdump -l
Port Client name Port name
0:0 System Timer
0:1 System Announce
72:0 PHASE 26 USB(16/48) PHASE 26 USB(16/48) MIDI 1
80:0 UM-2 UM-2 MIDI 1
80:1 UM-2 UM-2 MIDI 2
if I want to monitor the first device (connected on the in 1 of the UM-2)
this is what I mostly get :
$ aseqdump -p 80:0
Cannot connect from port 80:0 - Resource temporarily unavailable
sometimes it works, but it is absolutely not reproductible
on the other hand, the second device (connected on the in 2 of the UM-2)
is always working :
$ aseqdump -p 80:1
Waiting for data. Press Ctrl+C to end.
Source_ Event_________________ Ch _Data__
80:1 Control change 0 1 69
80:1 Control change 0 1 70
but this information is useless to set up ecasound so I have tested with
gmidimon. The approach is not the same as it uses /dev/midix or
/dev/snd/midiCXDX
the only device I can monitor is /dev/midi2 and /dev/snd/midiC2D0 (same
device) and it gives me this information :
Status Chan Data1 Data2 time
Control(176) 0 1 54 1123881037
I suspect I can use this information in ecasound so when I run ecasound
like that :
# ecasound \
> -G:jack,ecasound,notransport \
> -Md:rawmidi,/dev/snd/midiC2D0 \
> -a:6 -i:jack \
> -ea:1000 -km:1,0,1000,176,0 -o:jack
I hear no sound even if I tweek the channel 0 of the suitable device.
If I remove the midi control :
# ecasound \
> -G:jack,ecasound,notransport \
> -Md:rawmidi,/dev/snd/midiC2D0 \
> -a:6 -i:jack \
> -ea:1000 -o:jack
I hear the sound
I'm sure that the midi devices are all working because when I connect them
to AMS or om-synth with qjackctl I can control everything with both
devices.
What am I doing wrong ? And why the midi monitoring is so erratic ?
Thanks in advance for your help.
Philippe
Hi,
I'm discovering Demudi live, I have setup a box with two differents
sound cards (pci), both are working.
I'm not familiar with jack, I believed that it could handle several
sound output but can't see where to add the second card, It seems that I
can switch between the two cards but not to have twice in jack patch bay.
Is someone could light my mind ? :)
Thanks,
Jody
>
>
> All:
> I would like to announce the Mondrian Project. Mondrian is an attempt
> to create a text-based setting for writing and performing music.
actually, the way we do pushing/popping note state with [] is exactly the
same :)
more up to date docs are here (long url alert):
http://savannah.nongnu.org/cgi-bin/viewcvs/livenoisetools/livenoisetools/no…
cheers,
dave
Hi friends,
for some days I was looking for some open & free hardware and specially
soundcards.
I find some informations about openhardware... but not really on the
question of soundcard (some dac to build yourself). I would like to know
if some of you guys using open & free soundcard and if not at least a
way to do it yourself... or even if you know people who had a project to
release some prototypes or have similar projects.
I could be really great to use a gnu/linux plateform with so much & good
free audio software and to use an open & free soundcard to record & play
this sounds!!
thanks for the help
Julien