Mike Cookson:
>
> Hello all. I have compiled latest stable snd, version 11.6. I looked for
> LADSPA support and found it at core, but ladspa-related scheme files
> (extensions) are missing. I extracted them from Debian package since
> could not find initial place, where they appeared. When I am loading
> ladspa.scm throught listener, I have message:
> ;use-modules: unbound variable, ladspa.scm[56]
>
> Approriate string is (use-modules (ice-9 threads)).
> I wrote (load "full_path_to_ice-9/threads.scm") and then got other message:
> ;define-module: unbound variable, /usr/share/guile/1.6/ice-9/threads.scm[64]
>
> LADSPA menu is not appeared neither at menu bar, nor plugins menu.
>
Lots of stuff was recently removed from snd, so
the ladspa menu probably doesn't work anymore.
However, snd-ls still supports ladspa and all the realtime stuff,
and it is going to do so in the future too.
However(2), you are using Guile 1.6, which is really really really
old, so snd-ls probably won't work either unfortunately. :-(
even though I found a workaround and won't need this approach, I just
wanted to share something I found out, for "hot rodding outputs" of
consumer equipment, aka "doing it like M-audio's balanced outs."
Executive summary: Use this configuration:
http://www.dself.dsl.pipex.com/ampins/balanced/balfig4b.gif
from http://www.dself.dsl.pipex.com/ampins/balanced/balanced.htm
Details: see part about ''"hot-rod" the output stage'' from
http://www.jensen-transformers.com/an/an003.pdf
.........................
2 - UNBALANCED to BALANCED INTERFACES
The interfaces on the following page do NOT provide the 12 dB gain
necessary to raise the nominal -10 dBV (316 mV RMS) "consumer"
reference level to the nominal +4 dBu (1.23 V RMS) "pro" reference
level. If the pro equipment doesn't have enough gain "reach", an
active interface may be necessary. A step-up transformer, even an
ideal lossless one, is not a viable source of gain in this application.
Reflected impedances cause excess level losses and compromise
both low frequency response and distortion.
Adding a Balanced Output
A simple modification to equipment with unbalanced outputs can
convert it to have true balanced outputs. Get (or trace the circuit to
make) a schematic of the equipment's output circuitry. Depending
on available panel space, the new 3-conductor output connector can
be added or used to replace the existing connector. This
modification uses the existing unbalanced output as the + output
and adds an impedance matched passive network to ground for the
- output. In most cases, it is as simple as shown above.
The output impedance of the existing output is defined by the
network between the op-amp output (whose closed loop output
impedance is negligible) and the output connector. An identical
network to ground is then added as shown.
This is also a good opportunity to "hot-rod" the output stage, by
lowering and tightly matching its output impedances. Lowering RS
to 100 S, ±1% and increasing CC to 220 μF, ±20%, works well with
any popular op-amp known to the author, except for the TL06x,
TL07x, or TL08x series (their high open loop output impedance
makes them unstable with capacitive loads such as cables). For op-
amps operating from symmetrical supplies up to ±18 volts, we
recommend Panasonic 16 volt bi-polar electrolytics, part number
ECE-A1CN221S, available from Digi-Key or other Panasonic
distributors. These parts have the lowest distortion characteristic of
any we've tested. The modified output will have balance as good or
better than most current pro gear and, with the exception of the
possible "gain reach" problem mentioned earlier, will produce
excellent results in a professional environment. If the unbalanced
output is retained, do not use (or connect cables to) both outputs at
the same time.
..............
http://search.digikey.com/scripts/DkSearch/dksus.dll?vendor=0&keywords=ECE-…
[[ comment: not available any more as they contain Lead and not ROHS
compliant, any suggested replacements?? How about those "10,000hr
Japanese solid capacitor" types like they advertise for premium
motherboards?? also is a 220 μF cap overdoing it, or is that what it
takes do de-anemic-the-bass on a consumer card? ]]
For reference: http://en.wikipedia.org/wiki/Balanced_audio#Differential_signalling
..............
Signals are often transmitted over balanced connections using the
differential mode, meaning the wires carry signals of opposite
polarity to each other (for instance, in an XLR connector, pin 2
carries the signal with normal polarity, and pin 3 carries an inverted
version of the same signal).
Despite popular belief, this is not necessary for noise rejection. As
long as the impedances are balanced, noise will couple equally into
the two wires (and be rejected by a differential amplifier),
regardless of the signal that is present on them.[1][2] A simple
method of driving a balanced line is to inject the signal into the
"hot" wire through a known source impedance, and connect the "cold"
wire to ground through an identical impedance. Due to common
misconceptions about differential signalling, this is often referred
to as a quasi-balanced or impedance-balanced output, though it is, in
fact, fully balanced and will reject common-mode interference.
..............
Niels
http://nielsmayer.com
Hello all.
I have compiled latest stable snd, version 11.6. I looked for LADSPA support and found it at core, but ladspa-related scheme files (extensions) are missing. I extracted them from Debian package since could not find initial place, where they appeared.
When I am loading ladspa.scm throught listener, I have message:
;use-modules: unbound variable, ladspa.scm[56]
Approriate string is (use-modules (ice-9 threads)).
I wrote (load "full_path_to_ice-9/threads.scm") and then got other message:
;define-module: unbound variable, /usr/share/guile/1.6/ice-9/threads.scm[64]
LADSPA menu is not appeared neither at menu bar, nor plugins menu.
I was working on some new material and I seemingly hit on a limitation
in Hydrogen. Is it not possible to make songs of more than 400 bars
length (yeah, it's getting long...)? Any insight here is most welcome!
Regards,
Arve Barsnes
Hi everyone,
I'm looking for an app that assists in organizing and previewing
samples/wave files. I vaguely remember that there exists at least one
app for that, but Google fails me. Any idea?
Also, are there sample editing apps on Linux which support "elastic
time" editing, such as Pro Tools or Cubase do? I know Ardour has
timestretching and chopping up regions, but I'm looking for something
more comfortable...
Thanks,
Leonard
Hello everyone!
So here's my first real Bach piece. The one I learnt back in 98, because we
worked on it at school. I found it so lovely, that I wanted to learn it: The
2-part invention in D minor.
http://juliencoder.de/nama/j_s_bach-d_minor_invention.ogg
Or the dinosaur version :-)
http://juliencoder.de/nama/j_s_bach-d_minor_invention.mp3
Or of course the website:
http://juliencoder.de/nama/music.html
I played it using the Tubed Keys Rhodes 73 gigasampler library from
Sampletekk and the jreverb (standford reverb), from the CAPS LADSPA collection
(unique ID: 1778).
I hope you enjoy it. It's played a bit differently and I'm afraid a little
sloppy. But that's art. :-)
Feedback as always very welcome!
Kindest regards
Julien
--------
Music was my first love and it will be my last (John Miles)
======== FIND MY WEB-PROJECT AT: ========
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
======= AND MY PERSONAL PAGES AT: =======
http://www.juliencoder.de
I have, for the moment, my RME HDSP Multiface at home along with a nice
pair of Sennheiser headphones. The headphones show up the shortcomings
of the integrated audio I/F on the motherboard so I'd like to be able to
route "normal" audio to the RME, e.g. use it for things like listening
to music with rhtythmbox, which seems to mean persuading PulseAudio to
output to it.
So far the PulseAudio sound preferences does not see the HDSP even after
restarting PulseAudio. The kernel module is loaded, the firmware loaded
and the card works fine with jack and ardour.
What do I have to do to persuade PulseAudio to see it?
The distribution is Ubuntu 10.04. There is no /etc/asound.conf
or .asoundrc file. I did try creating a .soundrc to enable ALSA
applications to output to Jack in the hope this would cause PulseAudio
to offer jack as an output option but this didn't seem to make any
difference.
Interestingly aplay -l shows the card and aplay -L does not:
$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: DSP [Hammerfall DSP], device 0: RME Hammerfall DSP + Multiface
[RME Hammerfall DSP + Multiface]
Subdevices: 0/1
Subdevice #0: subdevice #0
card 1: NVidia [HDA NVidia], device 0: VT1708B Analog [VT1708B Analog]
Subdevices: 1/2
Subdevice #0: subdevice #0
Subdevice #1: subdevice #1
card 1: NVidia [HDA NVidia], device 1: VT1708B Digital [VT1708B Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0
$ aplay -L
pulse
Playback/recording through the PulseAudio sound server
front:CARD=NVidia,DEV=0
HDA NVidia, VT1708B Analog
Front speakers
surround40:CARD=NVidia,DEV=0
HDA NVidia, VT1708B Analog
4.0 Surround output to Front and Rear speakers
surround41:CARD=NVidia,DEV=0
HDA NVidia, VT1708B Analog
4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=NVidia,DEV=0
HDA NVidia, VT1708B Analog
5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=NVidia,DEV=0
HDA NVidia, VT1708B Analog
5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=NVidia,DEV=0
HDA NVidia, VT1708B Analog
7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
iec958:CARD=NVidia,DEV=0
HDA NVidia, VT1708B Digital
IEC958 (S/PDIF) Digital Audio Output
Any ideas, please?
Andrew C:
>
> Hey Kjetil,
>
> Just one thing, jack_capture_gui2 appears to overwrite the current
> ja_ca_setrc when it's opened/closed, so that means if I set the option -jt
> and set an alternative location for the files to be saved, they get
> overwritten with whatever is in the GUI.
>
I think Hermann, who wrote jack_capture_gui2, needs to answer that
question.
> Also, would anyone know what I need to set in Rosegarden's Midi Sync
> settings to get Rosegarden to send the proper messages to jack transport
> (I'd have no problem with using CLI jack_capture if it weren't for this.)?
>
I don't understand the problem, why can't you use jack_capture
instead of jack_capture_gui2?
Ooops...looks like my copy and pasting skills are bit clumsy today, so
apologies if the end of that email is a little confusing.
Please note we're also asking for proposals to perform for between 25
and 45 minutes
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http://loss.access-space.org/
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--
Cheers,
Jake
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