> From: Fons Adriaensen <fons(a)linuxaudio.org>
> On Fri, Feb 18, 2011 at 07:36:44AM +1300, Jeff McClintock wrote:
>
> > With a RMS VU Meter you measure a 1KHz tone as a reference.
>
> A contradiction... A VU does not measure RMS, whatever does measure
> RMS is not a VU.
Isn't a VU Meter a standard root-mean-square function followed by a 300ms
integration to give it some 'weight'? ...calibrated against a 1kHz tone?
Hi Experts.
I wanna normalize my sound stream by loudness (energy / pressure /
intensity) , not by peaks.
How i do it ?
Is available Jack plugin for so what ?
What is (we hear as) "loudness" ?
RMS or +(average) or something else ?
Is somewhere available examples how to calculate RMS ?
Is it done simlpe by :
 int i, n;  double sums, rms;
 sums=0.0;  n=10;  rms=0;
 for(i=0; i<n;i++)
 {   sums = sums + ( (double)i * (double)i );  }
 rms = sqrt(sums / n);
 printf("         
rms = %12.12fnn", rms );
Is so sipmple algo good enough for frequencies > 10 kHz ?
How to calculate RMS with hi-precision for frequencies > 10 kHz ?
With inerpolation or so ?
What is reference ( 0dB ) RMS for example for 16bit PCM signal 1024
samples ?
  sqrt( 1024*(0x7FFF ^ 2) / 1024 )  ==  sqrt( (0x7FFF
^ 2)  ) == 0x7FFF 
Is it simple  0x7FFF (32000 dec) ?
How to calc RMS for stereo signal ?
So , or somehow else ?
for(i=0; i<n;i++)
 {  ....
   sums = sampleL/2 + sampleR/2;
 }
In case operating with float point, what should be a bit
faster,  / 2  or * 0.5 ?
How to gotta RMS value in dB ?
   20 x log10(Measured/Reference0_dB) 
or 10 x log10(Measured/Reference0_dB)  ??
Im just physics student and im new in DSP,
so pleaz no angree about simple or stuppid questions.
Any examples and hints welcomed.
Many Tnx in advance.
Alf
----
Paul Davis:
>
> On Thu, Feb 17, 2011 at 4:53 PM, Robin Gareus <robin(a)gareus.org> wrote:
>
>>> Aside of that, what about locks? I've many times been told that one mustn't do
>>> anything that could block in a realtime thread. What are the consequences of
>>> that? Could a malicious app freeze the system by blocking in a realtime thread?
>
> it poses no risks to anything except itself if it does that. blocking
> in an RT thread matters to the thread, not to anything else.
>
> to demote RT threads that are doing too much you'd need a user-space
> watchdog like das_watchdog
>
Actually, das_watchdog is not very useful anymore after the kernel
developers implemented a scheme to avoid a process to take over
the machine. This built-in scheme is also a watchdog, but much
more fine graided than das_watchdog. And it is also (more often
than not) useless, so one has to press the reboot button anyway.
The sad thing is that this built-in watchdog in newer kernels fools
das_watchdog into thinking that the system is operational.
(@#$@#%%!#$$!!!)
I should look into it though, it might not be impossible to
tune das_watchdog to work again.
Hi folks,
inspired by a plan of a german onlinemag called amazona.de
I came up with the idea that a virtual analogue opensource softsynth
nativly running on Linux
would be really nice. (a nice filterbank too, but thats another thing)
Amazona planned a complete synth based on userpolls (only in german, sorry):
http://www.amazona.de/index.php?page=26&file=2&article_id=3191
which is now realized as vst: (only german, too)
http://www.amazona.de/index.php?page=26&file=2&article_id=3202
I know that Zynaddsubfx/yoshimi has a really strong soundengine and I
asked myself,
if it would be possible to take this engine or the DSSI-API and build
a polyphonic softsynth
with a nice UI like the new calf plugins or guitarix, a bit like the
loomer aspect, with some discoDSP,
a bit from the Tyrell or the Roland Gaia SH-01 with midilearn, ......
The problem I have are my programming skills, that are not good enough
to code this kind of software
by myself.
Are there some LAD's willing to join/take/realise this idea??
If there is interest I could translate the ideas of amazona.de and we
all could share our visions for a
new kind of controllable virtual analogue softsynth.
kind regards, saschas
Hello all,
As posted before, kokkinizita.net went offline last week as the
result of some stupid but apparently uncorrectable issues with
my hosting service, hosteurope.de. They are flatly refusing to
put the site back online or allow me to create a new one.
The kokkinizita.net domain will appear again (I still own the
domain name), but this could take some time. Meanwhile the site
has been recreated at <http://kokkinizita.linuxaudio.org>.
You won't be able to access it until linuxaudio.org's DNS will be
updated to include this url, which will happen some time friday.
In the meantime, if you need something urgently, just add the line
kokkinizita.linuxaudio.org 198.82.152.114
to your /etc/hosts, but don't forget to remove it later.
Many thanks to Robin Gareus, Ico Bukvic and Virginia Tech for
providing this solution, and also to all people who suggested
alternative hosting.
For email, please use the from: address of this message, not
the gmail one I posted earlier (I'll still check it, but not
as frequently as I do now).
Ciao,
--
FA
Hi,
I've forked the Specimen sampler to create Petri-Foo[1]. The main goal
of Petri-Foo was to make the ADSRs and LFOs independent from the items
they modulate (ie amplitude/pitch/etc). This has been achieved and is
quite nice to play with switching modulation sources on the go as a
sequences plays and the output is recorded.
I'm currently overhauling the sample-'editor'. So far I've ported the
deprecated GDK drawing code to Cairo, and prettified the display of
play start/stop and loop start/stop points and areas.
What I want to do next is explained here Sample editor zoom [2]
Your ideas would be welcome in how it should work. I'll try and
implement it. If you're interested subscribe to the Petri-Foo-Devel
list[3].
Please note though, it's not ready for general usage by inexperienced
users who don't know how to download and compile software from git
repo's nor for users who aren't expecting things to be broken.
Thanks,
James.
[1] http://petri-foo.sourceforge.net/
[2] http://sourceforge.net/mailarchive/message.php?msg_id=27069524
[3] https://lists.sourceforge.net/lists/listinfo/petri-foo-devel
The other day I tried somthing unusual. I use 96 frame buffers and so I
allocate two of those for each of in and out, and then read/write
(blocking) 96 frames at a time - makes sense, yes?
Well no, it turns out that - without doing any processing - looping over
read/writing only 16 frames at a time is more robust.
How come?
The same effect can be observed with 64 frames and 16 appears to be the
optimal value in the loop.
/j
Hello all,
Last week my mail and web host (Hosteurope in Germany) blocked my accound due
to failed credit card payments. I got no notice of this until it was
too late, and the
result is that I can't re-activate my account or even start a new one.
For anything urgent you can reach me via the gmail address used to
send this messsage.
I looking out for a new host, and hope to have the website online
before the end of the
week.
Ciao,
--
Fons Adriaensen