is there a firewire or usb soundcard wich would have the same quality
and low latency as the rme-(cardbus/expresscard)-multiface together with
a rme-quadmic?
i think i dont need the big number of channels...
any suggestions?
thanks
olaf
I have Debian installed and was wondering if it was possible to get a
real-time kernel in Debian? I do not want to use 64 studio, but was
hoping for a site like Planet CCRMA for Fedora, just for Debian. Is
there anything like this? Many Thanks, Preston
Hi we are on our way to create a special distro dedicated to MIDIBox
development tools
Please have a look at this thread :
http://www.midibox.org/forum/index.php/topic,11913.0/topicseen.html
if you want to give your opinion/input and don t want to sign up at
MIDIBox i can forward your messages, copying and pasting this thread
If you don t know what MIDIBox is have a look at
www.ucapps.de
Simone
--
.wmv , .wma , .pps along with all proprietary Windows formats won t be
accepted and/or viewed....
Greetings,
Yet another round of upgrades here. I installed another gig of RAM in
each machine. The JAD box (32-bit) reads correctly at 3.1 G during the
boot, but the 64 Studio box reads only 3.67 G. It's supposed to be 4
gigs, what happened to the rest of it ?
Mobos are identical Gigabyte boards, CPUs are both AMD64, so the amount
should be supported, yes ? Video boards are also identical, with 512 MB
on-board memory.
The BIOS and dmesg yield the same report. Did I miss something ? DIMMs
not arranged properly, or ... ?
Best,
dp
Download from:
http://old.notam02.no/arkiv/src/?M=D
Realtime priority patch for the linux kernel
============================================
To make sure the Linux kernel is able to grant realtime
priority, and full nice and mlock capabailites can still
be a little bit inconvenient and/or frustrating.
However, this super-tiny patch:
http://old.notam02.no/arkiv/src/realtime.diff
against the linux source shortcuts all other
methods. No more pam, realtime-lsm, rtlimits etc.
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 0.9.19 -> 0.9.23:
*Minor spellings
*Check for out of memory
*Clean up source a bit
*Stop connection thread before closing jack client.
*Made --help a tiny bit cleaner
*Removed shut down code from the SIGINT signal handler.
*Fixed segfault in case jack shuts down. Thanks to Julien Claassen
for reporting the bug.
(Note that there is also a 0.9.24 release. 0.9.24 has
changed internal data representation from lockless ringbuffer to
lockless lifo and fifo stacks. (Unmodified lifo/fifo code taken
from midishare. (Copyright Grame 1999-2005)) 0.9.24 probably works
fine, but it shouldn't be used for important recordings since
it hasn't been much tested yet.)
Rollendurchmesserzeitsammler v0.0.5
------------------------------------
The Audio Rollendurchmesserzeitsammler is a conservative garbage
collector especially made for running inside an audio DSP thread.
New about this release is that I have finally replaced TLSF
(http://rtportal.upv.es/rtmalloc/) with a pool-based dynamic
memory allocator, which makes allocation using the
rollendurchmesserzeitsammler approximately as fast as
using custom memory pools.
Using the rollendurchmesserzeitsammler
should be a lot more convenient than memory pools though, and
since memory is not freed manually, but instead
is automatically freed in a separate thread, there is a
slight chance that using rollendurchmesserzeitsammler
instead of custom memory pools could make some DSP code run
faster.
In non-synthetic benchmarks, I have not been able to see any
significant improvement in CPU use because of this compared to
using the TLSF allocator. But for programs doing
millions of allocations per second, the new memory allocator
will probably perform significantly better than TLSF, if it
would ever make sense doing so many and frequent allocations
of course...
Changes 0.0.4 -> 0.0.5
* Implemented a custom pool-based dynamic memory allocater. This new
memory allocator is now set as default. To use TLSF instead, set
"USE_TLSF=-DUSE_TLSF" in the Makefile before compiling. The following
changes are caused by this switch:
* Allocating memory is now approx 10 times faster (13 vs. 168
instructions for the allocation itself, but there are some GC
overhead too)
* The allocator copies used memory only (not just the whole heap).
But not always! This was a lot more complicated to to with TLSF
so I didn't do that. Note that for the garbage collector to still
be hard realtime safe, the programs must ensure that full copies
are taken now and then. (There's a change in the API for doing that)
* Doing a garbage collection is much faster since the heaps are usually
much smaller (because only used memory is copied) and that freeing
is 10-20 times faster.
* Further improvements for reducing memory overhead and make
searching for used mem to be O(log n) instead of O(n) is
much simpler now. (this is TODO though)
* However 1: In case the code using the garbage collector will
continue forever to allocate memory of different sizes, the
new dynamic memory allocator could eventually run out of memory
even if the program itself doesn't use very much memory.
I don't think this is very likely to happen for DSP routines though,
and there might even be solutions to fix this problem if it should
ever come up. For now, just switching to TLSF fixes the problem.
* However 2: in non-synthetic benchmarks, I have not been able to see any
practical improvement in CPU use, apart from the slight improvement
in CPU available for use in non-realtime threads because taking
snapshots usually takes a lot less time now. But for programs doing
millions of allocations per second, the new memory allocator will
probably perform significantly better than TLSF, if it would ever
make sense doing so many allocations of course.
Hi !
I am wondering how to use JACK with more than one soundcard. For
instance, what if I want to get one of the inputs of my soundcard #1,
process it with a JACK program, and finally send it to one of the
outputs of my soundcard #2 ?
Should I launch several JACK deamons ? If so, would I be able to
interconnect them ?
Any idea ?
Thanks !
Adrien
NB : I cannot test anything, because I only possess one soundcard for
the moment...
NB2 : I have reposted this message because I think that my first post
was in HTML, sorry !..