Sorry folks!
But I'mtoo stupid. I have a struct, which contains a pointer to itself as an
element. How to write this. I once knew, but now I fail. It's a shame! :-(
Kindest regards
Julien
--------
Music was my first love and it will be my last (John Miles)
======== FIND MY WEB-PROJECT AT: ========
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
======= AND MY PERSONAL PAGES AT: =======
http://www.juliencoder.de
Hello,
I have been trying to use the hdspm driver with a recently purchased RME
Madiface device. The functionality that I require is to play various
files through various channels under program (script) control. I am
happy to report that it *almost* works - in that I can use my simple
userspace programs to play files. However, there are subtle problems -
in that the external MADI A-to-D (an Euphonix 703) does not seem to lock
into the sampling rate (something that works under windows). Its very
close to achieving full functionality (for my needs) - but just that
tiny bit away from being fully useable.
Any ideas?
Havent had much luck contacting RME and Winfried Ritsch (who
wrote/maintains the code).
Information on the the hardware and/or windows code would help to figure
out how to modify the driver (which was originally written for the rme
9652/aes32 devices).
Regards,
D. Sen
Version 1.3 of the Vamp plugin SDK is now available.
http://www.vamp-plugins.org/
Vamp is a plugin API for audio analysis and feature extraction plugins written
in C or C++. Its SDK features an easy-to-use set of C++ classes for plugin
and host developers, a reference host implementation, example plugins, and
documentation. It is supported across Linux, OS/X and Windows.
Version 1.3 is a maintenance release, with several bugfixes (almost all of
which only affect hosts, not plugins) and no new features.
Changes since 1.2:
* PluginBufferingAdapter has several important fixes to bugs that could
cause incorrect timings or output descriptors to be returned
* Conversion between real-time and frames has been improved to avoid
rounding error in round-trip calculations
* Plugin lookup no longer relies on non-portable DT_REG
* The SDK now compiles with gcc 4.3 (knowing my luck it probably
won't, but it's supposed to)
Plugins and hosts remain binary compatible with those built using the 1.0
version of the SDK.
Chris
Rubber Band is an audio time-stretching and pitch-shifting library and
utility designed for musical applications.
http://www.breakfastquay.com/rubberband/
It includes a library that supports a sample-accurate multithreaded
offline mode and a real-time lock-free streaming mode; a command-line
utility program; and a LADSPA pitch-shifter plugin. Version 1.2 is
faster in most situations, better sounding in many, and less
potentially subject to patent claims than version 1.0.1 was.
Rubber Band is Free Software under the GNU GPL.
Chris
MarcOChapeau:
> > In the pd-world, we had a similar discussion to get rid of double- and
> > triple-announcements some time ago, and solved it so that mails to the
> > announce list get forwarded to the main list automatically. So people
> > who want to announce something, only send mails to pd-announce. If you
> > subscribe pd-list, you automatically also receive pd-announce, while
> > if you subscribe pd-announce only, you only get pd-announce mails.
>
> Interesting suggestion. I guess this could work. This means that there
> should be some web page somewhere stating the rules that I could use in a
> template for denied emails.
It would be great if you did that.
>
> Other ideas anyone ?
How about automatically add follow-up to linux-audio-dev and/or
linux-audio-user on mails posted to linux-audio-announce?
I wonder if we could change the post-to-all-three-lists
policy? I don't like to spam three mailing lists
for everytime I announce a program, and it doesn't
seem like everyone knows that we are supposed
to post to all lists either.
Download from:
http://old.notam02.no/arkiv/src/?M=D
jack_capture
============
jack_capture is a program for recording soundfiles with jack. Its default
operation is to capture whatever sound is going out to your speakers into
a file. (But it can do a number of other operations as well...)
Changes 0.9.17 -> 0.9.19:
*Do not accept filename starting with "-" when filename is last argument.
*Added the "--filename"/"-fn" option to set filename not
as the last argument. Can also contain "-" as first character.
*Fixed colors a bit and removed a superfluous linebreak.
*Fixed segfault for missing values in commandline.
*Manually clear allocated memory instead of using calloc.
*Removed buffer-info line when recording to stdout.
*Removed printing of "continue recording" when writing the last overruns.
*Niceify the disk writing thread to -20 when more than half the buffer is used.
*Continously show buffer usage, total number of overruns and whether
the disk thread has high priority, in the console. Turn off by using
the -hdu option.
*Added option -dc to disable console update. (same as "-dm -hdu")
*Fancier colors.
*Removed "hepp".
*Made dB meter the default and replaced the -dB option with the -lm option.
*Made the console meter wider to fit the info line at the bottom.
Snd-ls
======
Snd-ls is a distribution of Bill Schottstaedt's sound editor SND.
(http://ccrma.stanford.edu/software/snd/)
Its target is people that don't know scheme very well, and don't want
to spend too much time configuring Snd. It can also serve
as a quick introduction to Snd and how it can be set up.
Snd-ls also serves as base code for the San-Dysth softsynth
(http://www.notam02.no/~kjetism/sandysth/) and the Snd-rt music
programming language (http://www.notam02.no/arkiv/doc/snd-rt)
Changes 0.9.8.16 -> 0.9.8.17:
*(read-enable 'positions) has been wrongfully
turned off for some time. Bactrace should work
again now. (This only affects people using
snd-ls for programming.)
Rollendurchmesserzeitsammler
============================
The Audio Rollendurchmesserzeitsammler is a conservative garbage collector
providing general ways to efficiently allocate and use memory
inside a realtime audio thread without having to manually free
it later.
Changes 0.0.1 -> 0.0.4
* Added support for iterior pointers. (Pointers which points inside
middle of allocated memory). Turned out Stalin depended on this,
plus that c compiler are free to store pointers in registers
and stack however they want, so it's better to be safe.
* Fixed header.
* Made it packagable. Write make and make install to install library file
and header files.
* Added the function tar_get_dynamic_roots_for(char *pointer,char **start,char **end).
This function requires the source for HBGC:
http://www.hpl.hp.com/personal/Hans_Boehm/gc/
* Note that there are ways to increase the execution speed
significantly, I just haven't had the time yet, so this update doesn't
contain all the changes I wished for.
Hello!
Please Christian or someone else, can you help me. I just reread the
LinuxSampler features page and seems to be some way of marking points as
implemented, not done or partly implemented. But I can really find those
markings. Can someone give me a list of clearly marked features (clearly as in
text not colours or other visually markings).
In particular I'm interested in the following:
1. Does LS support gigaStudio3?
2. Articluations (.art-files)
3. Convolution reverbs
And another not completely related question: Is there a telnet equivalnet
that supports a bit more line-editing and probably globbing. If I say telnet I
mean the program "telnet". It's just annoying when misstyping not being able
to correct the typos and when entering long filenames not being able to
complete them.
Kindest regards
Julien
--------
Music was my first love and it will be my last (John Miles)
======== FIND MY WEB-PROJECT AT: ========
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
======= AND MY PERSONAL PAGES AT: =======
http://www.juliencoder.de
For a while I've been using the sirlabs.de vocoder plugin:
http://www.sirlab.de/linux/descr_vocoder.html
But it has some problems, notably expecting two inputs but only offering
one output. There was a patch on linuxrockstar which gave it two
outputs, but something seemed a bit lacking, still.
An evening of hacking and here we have it:
http://www.gjcp.net/~gordonjcp/vocoder.tar.bz2
It's hard-set to eight bands, all with the same level. There's a
bandspread control to alter the frequency spacing of the partials, and a
width control to vary the stereo effect where odd bands go to the left
output and even bands go to the right output.
Give it a go and let me know how you get on.
Gordon
jack_capture
============
jack_capture is a program for recording soundfiles with jack. It's default
operation is to capture whatever sound is going out to your speakers into
a file, but it can do a number of other operations as well.
Normal URL:
http://www.notam02.no/arkiv/src/?M=D
Note that the above link doesn't seem to currently work
because of something which seems to be www upgrade
at Notam. But this link does seem to work for now:
http://old.notam02.no/arkiv/src/?M=D
0.9.10 -> 0.9.17:
-----------------
*Made sure the process thread won't continue sending
data when jack_capture is told to quit. This
led to a race conditions when recording too many channels
at once (which was 256 channels on my machine).
*Added the jack_capture_gui script, based on code
by Svend-Erik Kjær Madsen's.
*Replaced sh by bash to make it work in ubuntu.
*Fixed exact port name match and gen_setformat. Thanks to
Gabriel J.L. Beckers for reporting the bugs.
*Increased the time the console meter is displayed in red from
50ms to 2000ms. (when the recorded jack value is equal to or
higher than 1.0.)
*Added the -dBr argument to specify reference level when using
the console dB meter.
*Added the -mr argument to specify reference level when using
the meterbridge meter.
*Added peak indicators to the console meter. Code to do so taken
from meterbridge by Steve Harris.
*Updated --help and README with the new options.
*Added option "-mt" to change meterbridge type. Current valid
options are vu (default), ppm, dpm, jf or sco. It's not necesarry
to specify "-mb" if using "-mt".
*Added option "-dB" to get a dB meter for the console meter.
*Added examples how to record ogg and mp3 files using the -ws option.
*Decreased default buffersize from 60 to 20 seconds.
(I even had trouble provocing underruns using a minimal
0.05 seconds long buffer, so 60 seconds was obviously overkill)
*Fixed message and error printing to stderr when
vu meter is running.