Hi,
I am writing a program, where I have to receive osc messages with have their
last argument as a blob type.
Does anyone have an example of how to decode the blob using liblo?
sincerely,
Marije
The authors are proud to announce the release of Aqualung 0.9beta8.
Aqualung is an advanced music player originally targeted at GNU/Linux,
today also running on other operating systems such as FreeBSD and
MS Windows. We are striving to create one of the finest music players
available, with respect to sound quality, stability, features and
ease of use.
This release is the latest in a series of beta releases on our way to
the future stable release of Aqualung 1.0, which is anticipated to be
released at the end of this year. The current release adds support
for internet radio and tabbed playlists, also containing several
smaller improvements and important bugfixes.
The ChangeLog for this release is listed below.
Homepage: http://aqualung.sf.net
Enjoy,
Tom
2007-07-07 Tom Szilagyi <tszilagyi at users dot sourceforge dot net>
* Aqualung 0.9beta8
http://aqualung.sf.net
This is a major release bringing significant new functionality and
many important fixes. All users are encouraged to upgrade.
Major additions:
* Support for internet radio streams using Ogg Vorbis and MP3
audio encoding.
* Tabbed playlist support, very similar in concept to the tabbed
browsing feature of Firefox.
Smaller fixes, rewrites and additions have been also done,
particularly to the following areas:
* Cut/copy/paste functionality implemented in playlist. Works with
the usual Ctrl-X/C/V key combinations.
* MPEG decoder: more robust in case of corrupt UBR files.
* Command-line local and remote file loading.
* M3U and PLS parsers.
* HTTP proxy handling.
* RVA handling: now supports setting a default value for
unmeasured tracks.
* Icons and Documentation.
* Added Italian translation.
Hello all,
Some updates and new things on my webpages.
First official release of jconv.
Jconv is a command line jack client performing FFT-based
convolution using a mix of up to five partition sizes,
small ones at the start of the IR, and building up to
the optimum size further on. It allows zero-delay
convolution with moderate CPU load. Jconv uses the
multi-threaded convolution engine developed for use
in Aella, a convolution engine optimised for reverb
processing that is nearing completion.
Main features of jconv:
- Any convolution matrix up 64 by 64, as long as your
CPU(s) can take it.
- Allows to trade off processing delay to CPU load, and
remains efficient even when configured for zero delay.
- Reads the same config files as Jace which it will
eventually replace.
The beta version released two months ago to some
volunteers contained a bug discovered and patched
by Martin Rumori. It did not affect configurations
using independent 1-to-1 convolutions, only matrix
operations.
Update of Ambdec
Some small changes and bugfixes. The configuration
format now also permits the use of mixed-order de-
coders (2nd order horizontal, 1st order vertical).
Manual updated.
TetraProc / TetraCal.
Both are ready and have been used for real Ambisonic
recordings, but release of TetraCal awaits a required
update of Aliki, and completion of the manual that
describes the calibration procedure.
If you want to use Core Sound's TetraMic with
TetraProc this is possible today. I signed an NDA
with Core Sound giving me access to the impulse
response measurements performed by Core Sound on
each mic. Given the serial number, I can process
these using TetraCal and provide a matching config
file for TetraProc.
You can see some examples of this on the updated
screenshots page for TetraProc.
As usual: <www.kokkinizita.net/linuxaudio>
Ciao,
--
FA
Follie! Follie! Delirio vano è questo !
Hi everyone,
Qsynth 0.3.0 is now out for you to try and guess what? This marks the
point of no return to the aging Qt3 framework. Yes, Qt4 migration was
complete.
Hints from the change-log might be shallow, but nevertheless:
- Qt4 migration has comenced and is now complete. Care must be taken
with this new configuration file and location: this release starts a new
one from scratch and won't reuse any of the previous existing ones,
although cut and paste might help if you know what you'll be doing :)
- Application icon is now installed to ${prefix}/share/pixmaps;
application desktop entry file is now included in installation; spec
file (RPM) is now a bit more openSUSE compliant; initial debianization.
- Default font option names were adjusted to "Sans Serif" and
"Monospace", wherever available.
- The "keep child windows always on top" option is not set as default
anymore, because window focus behavior gets tricky on some desktop
environments (eg. Mac OS X, Gnome).
- Autoconf (configure) scripting gets an update.
Good grief. All this is rteadily available from the usual place:
http://qsynth.sourceforge.nethttp://sourceforge.net/projects/qsynth
Cheers && Enjoy &
--
rncbc aka Rui Nuno Capela
rncbc(a)rncbc.org
Hi everyone,
Not much to say, but an apologise: QjackCtl 0.2.23 has been released and
is the one first ever introducing explicit JACK MIDI support (JACK >=
0.107.0).
The version minor number did not bump as promised. Fact is I'm making
that reservation for the coming and ever nearer Qt4 migration. So near
that it will crash land around now sooner than after ;)
The change-log says all the lesser bits too:
- JACK MIDI support is now being introduced. Connections window now has
a brand new MIDI tab, the older being renamed to ALSA, as for the
ALSA/MIDI sequencer conveniency. The server settings now include the
MIDI driver setup option (ALSA backend only).
- Application icon is now installed to ${prefix}/share/pixmaps;
application desktop entry file is now included in installation; spec
file (RPM) is now a bit more openSUSE compliant; initial debianization.
- Invalidation of the JACK client handle is now forced right on
jack_shutdown notification, preventing a most probable fatal crash due
to jack_deactivate and/or jack_client_close being called after the
jack_watchdog kicks in.
- Default font option names were adjusted to "Sans Serif" and
"Monospace", wherever available.
- The "keep child windows always on top" option is not set as default
anymore, because window focus behavior gets tricky on some desktop
environments (eg. Mac OS X, Gnome).
- Autoconf (configure) scripting gets an update.
Of course, official source tarball is made available from the project site:
http://qjackctl.sourceforge.nethttp://sourceforge.net/projects/qjackctl
Enjoy, as always :)
--
rncbc aka Rui Nuno Capela
rncbc(a)rncbc.org
Sorry for X-posting.
PhD Scholarships
Multimedia Signal Processing
TU Berlin, Germany
The Communication Systems Group led by Prof. Dr.-Ing. Thomas Sikora
offers several PhD scholarships in the following research fields:
* Single- and Multi-view Video Coding
* 2D/3D/stereoscopic Image Processing
* Audio Analysis
* Description of Humans in Video Sequences
Please submit your full application until 2007-07-31.
More information:
<http://www.nue.tu-berlin.de/news/phdscholarships.html>
<http://www.nue.tu-berlin.de>
Hi all,
Core Sound <http://www.core-sound.com/default.php> will
start shipping their A-format TetraMic in a few days.
This is one of the Ambisonic microphones that can be
used with TetraProc, see
<http://www.kokkinizita.net/papers/tetraproc.pdf>.
I have signed an NDA with Core Sound, giving me access to
the original measurement files for each microphone. So if
you purchase a TetraMic you now have two options, either
perform the IR measurements required for calibration
yourself, or just send me the serial number which will
permit me to generate a config file for TetraProc based
on the IR measurements done by Core sound. Free service !
--
FA
Follie! Follie! Delirio vano è questo !
Hi,
Thanks for mail.
>Your terminology is not very clear. What exactly do you mean by
>'average' RMS and 'max' RMS ? The 'M' in RMS stands for 'mean',
>so it is already an average over all samples considered, and
I'm using 25ms of window for calculating Max RMS,meaning is in
every 25 ms I calculate RMS and compare it with previous value
if current 25 ms RMS value is bigger than previous value I retain
this value, using this I find MAX RMS value for 25ms in file and call
it MAX RMS value as it is in adobe audition.
>That is, by definition of RMS = square Root of the Mean of the
>Squares, the RMS value of the N samples, expressed in the same
>unit as the samples themselves.
> >AvgRMS = 20.0 * log10 ( rms /2^N-1)
>You may be confusing two values of N here, the first being the
>number of samples, as in equation (1), and the second being
>the number of bits.
Sorry for confusion in dB conversion algorithm, it is surely
depth of sample in number of bits.
> normalised_rms = rms / (2^(B-1))
>and then convert to dB:
> normalised_rms_in_dB = 20 * log10 (normalised_rms).
your are right here as i'm also doing the same. My requirement is
like Adobe audition my application has to calculate the Average RMS power
for a wave audio file using time window system.
I'm trying to do the same to calculate the Average RMS but confusion is my
calculated Avrg RMS value matches with Total RMS value of the adobe edition.
How to calculate the Average RMS , if above algorithm calculates Total RMS.
BR
chandrashekhar
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