Hi again, Steve:
> Reverb is largly a matter of taste I think.
Here is an example of a reverb that sounds hilarious to me: In a large
concert hall type of room, a vocalist may be singing or saying something
at a very low volume, but very close by. There shouldn't be much reverb
if there is any absorption in the room by clothing, carpeting, etc.
However, some models, because the amplitude is high, will go ahead and
produce a resounding reverb because the model is not actually physically
based. In order to fix this, one has to go in and tweak all the parameters
and create yet another preset for "concert hall." Even though you're
in one hall, you may need half a dozen different custom presets to make
everything sound OK. Even then, you have to just guess at the parameters.
Now the echo may be wrong, so you have to go fix that. It becomes a cycle
until you finally just decide it's good enough and quit.
I think that what has happened is that we have all become accustomed to
hearing badly modelled rooms and we are just used to it. I have found
that listening to good models (OK, better models!) is a lot less tiring.
Not only do things sound better, but they are easier to listen to.
Best regards,
Dave.
oliver oli <smoerk(a)gmail.com> writes:
> Is it possible to remote control the player (via OSC or
> Telnet, XML-RPC or some other interface)?
Midi control is an absolute requirement for me
> btw, why is the first and most important feature of most audio players
> skinability? ;-)
Both a non-interactive shell interface and a interactive shell or
ncurses interface is also a requirement for me, but a gui should also
be available when others use the computer.
mpd, http://musicpd.sf.net is just up my alley, but it lacks jack
support for the time being. It has gapless playback, though.
I also need to live mix and mpd doesn't fill that requirement yet
either.
--
Esben Stien is b0ef(a)esben-stien.name
http://www.esben-stien.name
irc://irc.esben-stien.name/%23contact
[sip|iax]:b0ef@esben-stien.name
Hi
newbie
Just grasping the ins and outs of jack
basic sound blaster card
tone present on line in
qjackctl shows capture_1, .. _2.
I can remove the route of capture_1 etc to things like ardour by
disconnecting it in qjackctl as expected.
however I cant stop the line input from being "mirrored" on the cards
output.
gnomes volume control ..line slider allows me to adjust the volume.
(like a pre fader) but no matter what i disconnect in jack the tone on
line input also comes out of the cards output.
Is this typical of sound cards. or do i still have something not quite
right?
ta
On Wed, 2 Feb 2005 12:06:23 -0500, linux-audio-user-request wrote
> On Wednesday 02 February 2005 15:01, Peter Lutek wrote:
> > would someone please recommend a softsynth which includes fm capability
> > and is efficient enough to run with MINIMUM 16-note polyphony (preferably
> > more), for use as a keyboard sound source (esp. as a rhodes clone). alsa
> > modular synth seems to make jack's cpu usage runaway when started with
> > decent polyphony and anything more than a trivial patch.
Try hexter: http://dssi.sourceforge.net/hexter.html
Great DX7 FM emulation, flexible control, and it doesn't work up a steam with
16 polyphony.
Plus you can draw on thousands of existing DX7 patches.
Good luck,
Mercury
These are not free but if you have them under windows, they work just fine in
WINE:
Tabledit, tabledit viewer (tablature editor and player, abc import, etc. -- A
good application.)
Abcmus (abc editor/player)
Harbal (Equalization to improve audio, compare to stuff you like--I was quite
surprized when this one played! Misidentifies the sound card but plays! The
only audio app so far I found usable and in real time yet.)
XGEdit (If you have an XG sound card, this one is handy. Caveat--the rightmost
column of button showing the voices does not appear because the author is
playing a few tricks, I suppose. Still usable to an extent, however.)
These are free:
AnalogX VPiano (computer keyboard plays MIDI. Could probably be of use using
jack if this works correctly in WINE).
DBPowerAmp music converter (Convert wav <--> MP3 -- works on right click or
standalone under Windows, standalone using WINE. Very useful!)
greetings!
would someone please recommend a softsynth which includes fm capability
and is efficient enough to run with MINIMUM 16-note polyphony (preferably
more), for use as a keyboard sound source (esp. as a rhodes clone). alsa
modular synth seems to make jack's cpu usage runaway when started with
decent polyphony and anything more than a trivial patch.
i'm on an acer laptop, pentium-m(a)1.6GHz, 1Gb RAM, with an RME Multiface.
thanks!
-p
Dear list,
I am trying to write a very simple script that launches jack and then
alsaplayer. I have tried a variety of techniques, but the only thing I
can get to work is a file containing:
/usr/bin/jackd -T -R -P60 -dalsa -dhw:0 -r44100 -p128 -n2 &
sleep 5
/usr/bin/alsaplayer -r -o jack
I am sure I am missing some core linux/unix knowledge here, because the
above strikes me as a bit of a nasty hack. Does anyone know of a more
elegant solution?
Jamie
About a month ago I was introduced to Agnula and Linux in general and to make
things short I'm making an attempt to switch to Linux for audio work. So I've
sold all my commercial VSTs, MAX/MSP, and will move to Csound, PD and Audacity.
The only things left to do are to buy a combo DVDrw/cdrw drive, switch my cpu,
ram and HD's over to a Shuttle lunchbox sized PC (for a less noisy system) and
find a replacement for my MOTU 828 MK2. I am gearing towards an RME Hammerfall
card and a behringer A8000 adat interface. Or would I be better off getting a
PCMCIA adapter and going for the cardbus interface and Multiface? This is incase
I get a laptop in the future.
Too bad MOTU products aren't supported.
Thanks
Josh
-------------------------------------------------
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The "correct" or perhaps "most correct" impulse is a Dirac delta so-called
"function." Practically speaking, we cannot generate one.
Steve Harris said that the impulse does not matter too much as long as it's
wide band. Loosely speaking, and theoretically speaking, this is quite
correct. But because we cannot actually physically generate Dirac delta
"functions," it turns out to matter. You can see (hear) this for yourself by
creating square waves, ramps, truncated sin waves, etc. and generating the
response to these. One could argue that these don't qualify as impulses,
but that's what we actually generate in any empirical situation --- a
nonideal impulse.
This is why I use calculated IR's rather then recorded ones (see below).
The IR's calculated use as close to an ideal Dirac delta "function" as is
possible on a particular machine.
How much the impulse matters also depends on things further downstream,
which are usually ignored in theoretical discussions.
----------------------------
Some comments were posted about specific IR's at some web locations, and I
might as well address that here: I also really don't like any of the IR's
that I've tried to use from various web sites to compare with those that
I've calculated. The ones I've tried so far don't really sound any better
than the DSP-oriented ones in my hardware synths. Some of them don't even
sound as good.
Wolfgang has posted that he uses convolution for everything except reverb.
I use convolution with the IR's I've calculated for reverb, echo, stereo
separation, and equalization. With calculated IR's the stereo separation
is so good that it mimics binaural recordings. I'm not at all convinced
that using stereo mics in a cathedral to record an impulse would do as well
because of the spatial extent of the mics as well as that of the source,
but perhaps it can be done. This spatial extent limitation plus other
practical problems with attempting to physically record an IR may limit
the empirical approach.