Hi!
I Noticed that the Aureal drivers are now in Alsa. I used to be an early
adopter (with bad experience ...)
Anybody around here care to share some modern succes stories?
--
(
)
c[] // Jens M Andreasen
Hi,
QjackCtl 0.2.15a has been released!
This is just a fix release, update highly recommended.
As from the change log:
- Regression from 0.2.13, of the not so stupid pseudo-mutex guards on the
connections management framework, after fixing some crash reports from
Fernando Pablo Lopez-Lezcano and Dave Phillips (thanks!); it pays to be
such a paranoid after all :).
Check it out from the usual place:
http://qjackctl.sourceforge.net
Enjoy.
--
rncbc aka Rui Nuno Capela
rncbc(a)rncbc.org
This is sort of really of topic. I'm not a professional developer of
music software or something (I do have some ideas though) but I would
very much like to attend the conference. The problem is that private I
can't afford the trip and stay. Now, I have a nice boss that says that
if I could find something in the neighborhood that is close to my
normal day job (network engineer) he is willing to pay for the trip
as long as he gets something out of the trip. Is there anyone here that
knows anything close to networking at LAC 2005 or is there something
in Karlsruhe that I could have as a reason for traveling there.
regards,
/bengan
>> technical operations website: http://techweb.rfa.org/. techweb also
[ ... ]
>> I see an opportunity here for future use of numerous
>> linux audio applications in a working digital audio production facility
>> that puts out 36+ hours of programming each day. I would be honored and
>> excited if my new boss were to come from the linux audio community.
OK. Here is a project suggestion:
Remember the old days when Spectrum and Commodore computer software
was broadcasted through radio to computer users? This system could
be brought to this day using latest tech in coding. This would
make it possible to make forbidden webpages available to countries
who restrict the access.
People would need a simple and cheap radio receiver, and plug it to
the audio card. The signal coding would differ from communication
coding systems in that sample rate is limited to 44100 Hz and that
audio has more signal level resolution.
I invented this "Radio Internet" several years ago because Internet
costed (and still costs) too much and because the transfer direction
is mostly from Internet to home for personal users. The idea of using
this tech to broadcast Internet to China became a few months ago
after BBC World's Click Online show reported about the Internet
retrictions in China.
With non-cheap radio tech the transfer rate would not be limited to
the audio card properties, but simple and cheap might be better
when it comes to the freedom spreading.
Juhana
--
http://music.columbia.edu/mailman/listinfo/linux-graphics-dev
for developers of open source graphics software
...a simple, but powerfull paradigm that helps to make
simple algorithms that produces very beautiful sounds.
Hi.
I want to open a topic about what I call "bandwidth of
each harmonic", to say some ideas that helped me to
make beautifull sounds. I am Paul, the author of
ZynAddSubFX software synthsizer (
http://zynaddsubfx.sourceforge.net ) and I wish to
share some ideas ;)
Referring to musical sounds,usually the harmonics are
considered simple sine function. Of course, the
reality shows different thing, and so, the harmonics
was started to be considered as sine functions
modulated by a lower frequencies (I saw some time ago
a link that describes that the choir sound is
beautifull because of the "micromodulations",etc...).
A very good thing is to "have a look" in a frequency
domains. Let's take for example a choir that sing same
note (like A=440Hz). Because all are human, even they
are very well trained singer, they will not sing
exactely the same note; for example one will sing at
435 Hz or other, at 443 Hz, and so on. Now, the first
harmonic (the fundamental note) is not longer a sine
of 440 Hz, but a narrow band signal that has a certain
bandwith. For example, let's take a very simple case:
if all sing at same loudness, and they sing from 435
Hz and the 445 Hz. In this case, the bandwidth will be
10 Hz. Of course, in real choir the frequency
distribution of the harmonic will be not flat, but
usually will be a curve that looks like a normal
(gaussian) curve(found this after I did some research=
very fine frequency analysis with very long FFTs).
Now, let's go to the second harmonic: if you multiply
435 and 445 with 2, the difference will be 20 Hz. So,
the bandwidth of the second harmonic is 20 Hz. So here
a important rule of real instruments (esp. ensembles):
the bandwith of each harmonic is proportional to the
it's frequency; so in this example the first harmonic
has bandwidth 10Hz, the second 20Hz, the third 30Hz,
and so on.
Here I made a very fine frequency analisys of a
synthsized sound(a real orchestra sound will give
similar results):
[img]http://zynaddsubfx.sourceforge.net/doc/paul1.png[/img]
You see, that the harmonic's bandwith increase
according to their frequency. If you don't increase
the bandwidth of higher harmonics, the resulting sound
will be unpleasant especially on higher bandwiths of
the first harmonic. So, not all quasi-periodic sounds
are good, usually only if you increase the bandwidth
of the harmonics.
Now, what's happens if there are a lot of harmonics or
the pitch of the sound is enough low?
Let's see:
[img]http://zynaddsubfx.sourceforge.net/doc/paul2.png[/img]
The upper harmonics will merge to a single frequency
band that will sound like a hiss, that is pleasent to
the ear(eg. choir).
I found that bandwidth of each harmonics can go even
greater than 50 cents (quater of a tone) and still
sound musical - I can give you some example wavs.
Please note that this a different situation than 2
notes detuned by 50 cents (that sound very dissonand).
Also, the best is that the phases are random. In real
life this is ensured by the reverberation of the hall,
for example if a instrument plays with a vibratto (a
flute).
Perhaps you noticed that always if more instruments
plays the same note(as a enseble), the sound is very
pleasant. I consider that this is the cause: the
bandwidth of each harmonic.
Unfortunately, there is very little on the internet
about this stuff, because, I think, that are used more
complicated paradigms (like some statistics on how
sine harmonics modulates, or so).
Now I want to give examples on how you can synthesize
sounds, using this ideea.
You can use zynaddsubfx or another synthesizer in most
cases. Of course: I don't claim that I invented the
bandwidth of each harmonic (or frequency distribution
of each harmonic), because the ensembles, the choirs,
the reverberations, exists since thousands of years
(or more :) ). I just consider this a very important
fact on how sounds are beautifull and it helped me to
synthsize good sounds even since I was on highschool
:D.
You can make bandwidth of each harmonics by:
1) make more oscillators and detune them a bit (and a
slight vibratto helps alot). Most synth use this and
allows this, and perhaps it is known by you. This is
one of the simplest method. This is implemented in
ZynAddSubFX as "ADDsynth" module
2) generate white noise and filter each harmonics with
a bandpass filter and mix the results. Be carefull to
make higher harmonics to have higher bandwidth. This
is implememnted in zynaddsubfx as "SUBsynth" module.
3) You see that above graphs. You can represent them
as numbers in the frequency domain that represent the
amplitudes of the frequencies, add random phases and
do a SINGLE IFFT and voila! A very beautifull sound
will "born". This new ideea is implemented in
ZynAddSubFX as "PADsynth"
4) you can do other things, like a vibratto on a
periodic oscillator and do a FFT to all sound, put
random phases, and do a IFFT.
For more information, I made few years ago a page that
describes above things at:
http://zynaddsubfx.sourceforge.net/doc_0.html
Good luck.
Paul
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Hi,
QjackCtl 0.2.15 has been released.
As a major new feature you are now allowed to rename (alias) the JACK/ALSA
connections client/port names to something intelligible. Another nice one
is about actual ALSA hardware device names which are now presented for
selection as a pull-down menu on the setup dialog.
Grab it from:
http://qjackctl.sourceforge.net
As taken from the change log:
- JACK/ALSA client and port name aliasing (renaming) is now an optional
feature for the connections window; all client/port aliases are saved on a
per preset basis (as proposed for Lionstracs' Mediastation).
- Server state now shown (back gain) on the system tray icon tooltip;
speaking of which, tooltips are now also featured on connections, status
and patchbay windows.
- New actual hardware device selection menu featured on setup dialog;
these new button menus are only available for the ALSA driver settings.
- Server path factory default to jackd instead of jackstart; preset setup
button icons are back.
- Fixed rare connection port item removal/disconnection dangling pointer bug.
Have fun.
--
rncbc aka Rui Nuno Capela
rncbc(a)rncbc.org
I've been trying for months to get my midi input
(using OSS/free) on a knoppix liveCD so that I can
record midi in the included rosegarden2.1, and here is
where I am now:
I found a very short program that all it does is
simply print out the bytes recieved at the midi input
- its at
http://ccrma.stanford.edu/~craig/articles/linuxmidi/input/section1.html
My /dev/sequencer is working fine for output (verified
with playmidi -e) however this program does not
display any midi input.
Actually what happens is BEFORE running the program, I
verify that my Roland sound module (connected to the
midi output of the midi cable plugged into the
soundcard) DOES recieve input from the midi keyboard
plugged into the midi input, BUT then when I run the
program, the midi input is no longer passed through to
the midi output.
I've verified that the test program displays no midi
input using /dev/sequencer, or /dev/midi, /dev/midi00
(I changed the device in the code)
Here are the only steps I followed which succeeded in
getting the ESS1868 midi output working.
modprobe sb io=0x220 irq=5 dma=1 dma16=5
mpu_io=0x330 # OR dma16=-1
modprobe -v mpu401 io=0x330 irq=7
modprobe -v opl3 io=0x388
I guess that if I just get this sample program to
display midi input, then I will be able to start using
rosegarden2.1 to record midi? But why would the test
program use /dev/sequencer for input (I thought
/dev/midi was the input?)
Why would my midi output (/dev/sequencer) but my midi
input wouldn't work?
here is the relevant output of sndstat:
Synth devices:
0: Yamaha OPL3
MIDI devices:
0: MPU-401 0.0 Midi interface #1
Any advice would be VERY appreciated!
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![img]http://zynaddsubfx.sourceforge.net/doc/paul1.png[/img](http://img]http://zynaddsubfx.sourceforge.net/doc/paul1.png[/img){kind=link}
![img]http://zynaddsubfx.sourceforge.net/doc/paul2.png[/img](http://img]http://zynaddsubfx.sourceforge.net/doc/paul2.png[/img){kind=link}