Hi lists,
my little report on the lac-concerts and some other subjects of the lac
in Karlsruhe will be broadcasted tomorrow on SWR2 JetztMusik Magazin at 23h.
Michael
Well, mixed results tonight.
I was able to get some sound to go across the ADAT
cables from the PC to the AW4416. But not good sound.
On the bright side, I think I more or less understand
connecting things up with jack, ecasound, and so on.
On the bad side, so far it's not working too well.
I monitored things with "jackmeter" and this meter
registered peaks near 0dB for the stuff I was playing
with ecasound, and pretty high levels for the most part.
On the AW4416, the levels were registering between -30dB
and -48dB. I guess I don't understand how ADAT works.
I was under the impression the signal going across the
cables was digital -- and so to get a reduction in levels
like that, I would expect some digital numbers would have
to go from being big numbers to being small numbers, which
seems unlikely thing to happen to numbers encoded as pulses
going down a cable. So I conclude I don't know how ADAT
works, except it's not as I imagined it did.
Oh, and besides a drastic loss of signal level, the signal
was distorted strangely. Hard to describe. This may be
due to xruns... I haven't got things to work without xruns
yet, but that shouldn't cause a drop in levels, right? Just
kind of choppiness, dropouts, crappy sound, right?
Transfering from the AW4416 to the PC did not work at all.
on capture_1 and capture_2, I got very low level white noise
apparently. Are those the s/pdif ports? On the other
channels input was dead silence.
I tried both ADAT ports on the RME board, with similar results
on each. I tried swapping the two ADAT cables in case one of
the cables was bad... this did not seem to make a difference.
Maybe the RME just transmits harder than the Yamaha, so it's
signal makes it across (just barely, crossing the finish
line at -48dB) while the yamaha's signal dies.
I did change the RME's frequency to 44.1kHz in qjackctl's
setup window.
Maybe there are some clues in here:
[root@zuul R15]# cat /proc/asound/R15/rme9652
RME Digi9636 (Rev 1.5) (Card #2)
Buffers: capture f6a00000 playback f6400000
IRQ: 10 Registers bus: 0xea000000 VM: 0xf88a2000
Control register: 48029
Latency: 1024 samples (2 periods of 4096 bytes)
Hardware pointer (frames): 1024
Passthru: no
Clock mode: autosync
Pref. sync source: ADAT1
ADAT1 Input source: ADAT1 optical
IEC958 input: Internal
IEC958 output: Coaxial only
IEC958 quality: Consumer
IEC958 emphasis: off
IEC958 Dolby: off
IEC958 sample rate: error flag set
ADAT Sample rate: 44100Hz
ADAT1: No Lock
ADAT2: Sync
ADAT3: No Lock
Timecode signal: no
Punch Status:
1: off 2: off 3: off 4: off 5: off 6: off 7: off 8: off
9: off 10: off 11: off 12: off 13: off 14: off 15: off 16: off
17: off 18: off
__________________________________________________
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Hi!
What is the most acceptable way to announce an app update?
I use "[ANN] ..." message sending to LAD/LAU lists, but am not
sure it is the best way for LA community.
(the app under question is QLoud)
Andrew
QLoud is a tool to measure loudspeaker frequency and step responses and distortions.
Find it here:
http://gaydenko.com/qloud/
Changes:
- Step Response plotting is added, this is a direct screenshot link:
http://gaydenko.com/qloud/screenshots/shot04.png
These measurements are done for the same 2-way loudspeaker (see plots titles).
It is clear, a tweeter must be shifted lightly beyond from a listener.
Hi,
I was wondering if there are any tools out there to test audio
resampling quality. I am particularly interested in 44.1kHz to 48kHz
resampling due to the fact that most sound cards prefer 48kHz.
At least with up sampling (low rate to higher rate) one does not get
aliasing.
I really just want to find some algorithm that I can use to compare
44.1kHz audio signal with an 48kHz audio signal, and to see if there has
been any lose of quality during the up sample.
James
Hi!
Can anybody point me to theoretical and algorithmic fundamentals
of real-time (JACK-oriented) (pseudo)pink noise generation at
given frequency range?
Andrew
Are there sound editors with horizontal scaling support as
snd (http://ccrma.stanford.edu/software/snd/) has?
I mean *both* scaling degree (1e4 times) and scale itself printing.
Hi,
this is a small announcement for a minor update for a minor piece of software,
and at the same time a question :) So here it goes:
Kontroll is a small utility that generates midi cc messages from the mouse
position. It is inspired by the MouseX and MouseY UGens in Supercollider. It
simply creates an alsa sequencer port which you can then connect with your
favourite patchbay. The mouse position is independent of window focus and is
relative to the screen origin at the upper left.
- Another small update to kontroll. Now the controller and channel numbering
range from 1-128 and 1-16 as commonly seen in other midi applications and
hardware. previously it as 0-127 and 0-15 which was probably confusing to non
computer people.
- A minor update to this little program of mine called “Kontroll”. On shutdown
it saves the last used parameters to a file called ~/.kontroll and on startup
reads it again. This saves setting it up all over again on each start of the
program. You can also save special setups via the “File” menu.
Grab it here:
http://tapas.affenbande.org/?page_id=42
Or directly:
http://affenbande.org/~tapas/kontroll.tgz
And here's the question: A user suggested (and i'd like this idea very much)
that kontroll be able to make use of other input devices attached to the
computer (additional mice, joysticks, etc). Now i would like to avoid playing
with /dev/input directly, cause i imagine it to be a drag. So does anyone of
you guys know a small and easy to use input-library that makes accessing
these devices a breeze? If so, please let me know.
Regards,
Flo
P.S.: Ah, LASH support is still missing. Will add it right away (or at least
try) ;)
--
Palimm Palimm!
http://tapas.affenbande.org
Hi All. I'm using Snack Audio 2.2 and Tcl/Tk to develop some
application, I need to record a few seconds of voice but it's
impossible y record an empty file.
-- Snack Audio
-- vtcl(GUI for Tcl/tk)
-- Suse Linux 10.1
I use this Tcl code.
package require snack
snack :: sound s
proc ::Record {} {
global widget
snack::sound s
proc sstop {} {
s stop
set filename "tmpwave.wav"
s write $filename
s destroy
// Extract
}
after 5000 sstop
s record
best regards
Yosvany
Greetings:
Recently I tested Robert Reif's ASIO driver for WINE. It works okay for
some small test apps (asiosiggen and asiodump). I also tested it with
NI's FM7, the app opens fine but I got no sound from it. I even loaded
and played a MIDI file as a demo but still got no joy from the audio.
I'm curious to try other ASIO-driven apps but I need some
recommendations for light-to-middle weight programs for testing.
Free/shareware is best, but feel free to suggest commercial apps too. I
don't use Win/Mac music apps and I have no idea where to start.
Best,
dp