and website links get redirected to the correct
place).
I'll wait for Paul's reply how to proceed (redirect linuxdj.com etc).
cheers,
Benno
Paul Davis wrote:
On Mon, 2006-01-30 at 17:44 -0500, Paul Davis wrote:
On Tue, 2006-01-31 at 01:10 +0100, Esben Stien wrote:
Benno Senoner [EMAIL
Hi LADers,
During the last months the LAD website (
http://www.linuxdj.com/audio/lad ) was hosted on the lionstracs.com server.
Domenico from Lionstracs told me that he does not want do host the LAD
site anymore since it consumes so much bandwidth,
200 GB in January, see here.
Davis wrote:
On Mon, 2006-01-30 at 11:29 +0100, Benno Senoner wrote:
Hi LADers,
During the last months the LAD website (
http://www.linuxdj.com/audio/lad ) was hosted on the lionstracs.com server.
Domenico from Lionstracs told me that he does not want do host the LAD
site anymore since
Hi all,
I would like to route a microphone through a sound card and back to
powerful amplified speakers.
As we know in analog PA gear you have the microphone feedback problem
(usually it comes in form
of high pitched whistle sounds).
But if I route a mic from into the soundcard and out to
My suggestion is to handle buffering in the convolution plugin and
accept any buffer size from the host.
I'd do it without threading to ensure the lowest possible latency.
For example:
assume we run convolution at 512 samples.
use a ringbuffer structure (eg like RingBuffer.h in
Florian Schmidt wrote:
On Wed, 29 Jun 2005 13:20:31 +0200
Benno Senoner [EMAIL PROTECTED] wrote:
assume we run convolution at 512 samples.
process(float *input,float *output, int numframes) {
if(numframes == 512) {
convolve(input, output, 512);
return;
}
This has
fons adriaensen wrote:
On Wed, Jun 29, 2005 at 06:02:24PM +0200, Florian Schmidt wrote:
What other goodies does it have?
Nothing special. I'd like to add a mode using multiple block
sizes for minimal delay. This is not trivial and requires
multiple threads as well.
Another
Stéphane Letz wrote:
Good question
I would say that one the may problem is that jack MIDI does not has
the concept of time-stamps in the future . The Jack MIDI events
are received in real-time (similar to what MidiShare does with the
Receive callback concepts) but have also to be
Lee Revell wrote:
FWIW, Linspire (formerly Lindows) have made the same conclusion, and
their next release will use JACK for all desktop audio.
Interesting. Do you know how they intend to use jack for desktop apps ?
using artsd/gstreamer/esd piped into jack or other methods ?
Benno
I agree with Rui,
arts piped into jack is probably the best solution currently.
And when doing paranoid low latency audio work just kill artsd as Rui said.
I'm not a big user of consumer audio apps (eg mailer that emits
BOING.WAV) but
I guess due to certain apps being KDE centric and some GNOME
The SkinDial class is written in Qt [1] provides flexible pixmap based
dial/knob widget
compatible with QDial [2].
The idea was to make it using Qt's QDial API so that apps (mainly audio
apps) using this class can make
use of eyecandy knobs without changing the code. (except for the
Arnold Krille wrote:
On Saturday 11 June 2005 20:44, Benno Senoner wrote:
The SkinDial class is written in Qt [1] provides flexible pixmap based
dial/knob widget
compatible with QDial [2].
The idea was to make it using Qt's QDial API so that apps (mainly audio
apps) using this class can
Hi,
cool news about the echo canceller.
some questions:
I've found this algorithm on the web (but I did not test it)
http://home.arcor.de/andreadrian/echo_cancel/draft-aec-03.txt
How do you think your's and the one above compare quality wise ?
Do linux telephony apps like linphone,
Chris Camisa wrote:
Hello!
Sorry if I'm starting in the wrong place, but after several months
of thinking and two weeks of working, I have a couple of questions.
1.) can FreeBSD 5.4-RELEASE operate without a sound card such that the
Network Audio Server allows applications running on it
Since we cannot increase the speed at which the sound travels and even
DACs add some latency (1msec or so)
I see any effort to reduce latency below 2-3msec quite useless.
We know that halving the number of frames (soundcard period size) we run
the audio software at is doubling the IRQ rate
and
Florian's pratical experience confirms what I stated in my other message.
My take on the matter is to get a good soundcard with fast DAC/ADC (
2msec round trip AFAIK)
and use something like 64frames latency. 64 frames gives you still
good,reliable RT performance and does not
push the kernel to
Hi Lee,
Interesting.
But what's the exact behaviour in 2.6 kernels then ?
Does nanosleep() provide non-busywait usec precision using hires timers,
or does it simply do the same as usleep() ?
Eg providing only 1/HZ precison.
cheers,
Benno
http://www.linuxsampler.org
Lee Revell wrote:
Right,
Paul Davis wrote:
usleep (2) is the canonical high resolution sleep function on POSIX
(although nanosleep(2) is a close cousin). the resolution is limited
by the system timer interrupt frequency, which is typically either
100Hz (2.4 and older kernels) or 1kHz (2.6 kernels)
Right,
nanosleep()
John Lazzaro wrote:
gives up and throws away a packet. But, its very
rare -- 0.1% or less, if I had to put a number on it.
But if that 0.1% was a NoteOff sent to an Hammond
organ patch, you care :-). Thus, the recovery journal
technology in RTP MIDI.
While I agree that a missed note-off is bad,
not big secrets here.
try to look up the docs for the WAV format.
or just search for big/little endian encoding
24bit packed words etc
in substance there are only a few mainstream raw pcm data encodings.
8bit , in some case signed, in others unsigned (amiga IIRC).
16bit signed (2 complement)
Steve Harris wrote:
No, the roundtrip latency is *at least* 100usecs (or whatever), the hardware
will keep re-transmitting until the packets get through.
Even if it is 100usec it's still a negligible amount of time.
Keep in mind serial MIDI is relatibely slow, press a 7 key chord and the
7th
Steve Harris wrote:
Any good suggestion how to best implement it ?
You can't just duplicate or drop samples, it will sound terrible. You need
to do some resampling. For clusters this shouldn't be an issue, just
make one device be the i/o machine and sync everything else off that.
Of
Lee Revell wrote:
anyway.
Plus what's very important is that every kernel developer and driver
developer (even thirdparty, especially those
that do closed source stuff like Nvidia etc) takes into account the
latency problems that code paths that run for
too long time (or disable IRQs for too
While locking a RT process to a CPU to achieve even lower latencies
might be useful to some
the general userbase wants good latencies on simple UP, non HT-enabled
hardware too.
(AMD is gaining marketshare and we cannot simply expect that good
multimedia performance (aka low latency)
can be
Takashi Iwai wrote:
Is it possible that I am simply pushing my hardware past its limits?
Keep in mind this is a 600Mhz C3 processor.
I think yes. 32 frames / 44.1kHz = 0.725 ms.
I don't think so, I think it's because the Linux scheduler (and kernel
in general) since it's not a RTOS
is
Depending from your particular needs, the fastes real time safe/ lock
free allocator is using an allocator that
manages a list of equal sized elements (of any type, tanks to C++
templates).
We use such an allocator in LinuxSampler and it's really fast, all
inlined and an alloc() usually takes
Martijn Sipkema wrote:
It is often heard in the Linux audio community that mutexes are not realtime
safe and a lock-free ringbuffer should be used instead. Using such a lock-free
ringbuffer requires non-standard atomic integer operations and does not
guarantee memory synchronization (and should
Lee Revell wrote:
On Mon, 2004-07-12 at 19:31, Andrew Morton wrote:
OK, thanks. The problem areas there are the timer-based route cache
flushing and reiserfs.
We can probably fix the route caceh thing by rescheduling the timer after
having handled 1000 routes or whatever, although I do wonder
Hi,
Ron Kuper wrote:
But ASIO isn't the only way around KMIXER. With the advent of Win32
Driver Model (WDM) Kernel Streaming (KS), the Windows O/S is indeed
capable of very low latency. WDM KS has a standardized device I/O
control set that's part of the Windows audio stack. KS makes it
possible
Jens M Andreasen wrote:
Why not just use modf?
double fullindex, increment, integer, fraction;
// int i;
fullindex += increment;
fraction = modf(fullindex, integer);
// i = integer;
C99 have float and long double versions as well.
The problem of modf is that it is slow (it generates call
Ruben van Royen wrote:
First of all, I was not yet talking about vectorizing your code which is often
hard, especially for a compiler. but SSE can be used on scalars as well (as
you probably know).
The fact is that the intel pentium 4 optimization guide says that SSE code is
generally as fast
Erik de Castro Lopo wrote:
On Tue, 29 Jun 2004 10:19:32 +0200
Benno Senoner [EMAIL PROTECTED] wrote:
In LinuxSampler we do
double increment;
double fullindex;
int integer;
double fractional;
for (;;)
{
/* Bunch of other code. */
fullindex += increment;
integer = lrintf(fullindex
Erik de Castro Lopo wrote:
The fix in this case was this:
for (;;)
{
/* Bunch of other code. */
fractional += increment ;
rem = fmod (fractional, 1.0); /* floating point modulus */
integer += lrint (round (fractional - rem));
fractional = rem;
[EMAIL PROTECTED] wrote:
Wasn't there also a proposal for MIDI over ethernet? It would be
quite a neat idea for a totally software based studio - have the
sequencer on one machine, while softsynths, samplers and whatever
else could be spread across more powerfull machines. If such an
interface
Audio traffic has a constant data rate.
eg 44.1khz 16bit stereo is 176400 bytes/sec
since the audio cards use audio fragments (or periods) of N frames
it is natural to send audio using packets over the network of that size
(or multiples of it).
UDP is the natural choice because of the low
Michael Ost wrote:
Perhaps sysex calls for a second midi-only packet type. Sysex could be
encoded as a start packet and some number of continuation packets. A
midi only packet could also let a driver send more than the 344 midi
messages at a time you spec'd out, if it needs to.
this could be
Juhana Sadeharju wrote:
From: Tim Hockin [EMAIL PROTECTED]
Quick question: disk thread may suspend if there are no disk use.
How the disk thread is woken up to read the lock-free buffer?
Semaphore. Every time you put something into the buffer, up() the
How this all is done in
Paul Davis wrote:
in a lock-free way. This ensures zero-copy operation.
until you want to start processing the data but keep the original
around. i was always attached to the zero-copy model, but it just
doesn't seem to pan out in real life.
I don't know how ardour works internally, so in
Joshua Haberman wrote:
At the moment I am favoring an approach that uses multiple discrete
buffers and passing pointers to them, rather than pushing data on a
ring buffer. One major reason is that it allows you to send the data
to multiple places. Say you want to send a captured buffer to
Paul Davis wrote:
A question regarding jack use in a live setting. As far as I can tell
the soft mode only works with non realtime jack. What should I do if I
it should work with realtime mode too. this was specifically added
several (many?) months ago, for precisely the reason you gave.
Paul Davis wrote:
One thing I am still looking to learn more about is how to adjust
thread priorities and such to make sure that your threads are run often
enough (especially the disk thread), and how to decide how big your
disk buffers need to be.
4 years ago, Benno and I measured this
Very nice knobs Thorsten !
We could use them in LinuxSampler :)
Rui Nuno Capela has started work on the GUI
http://www.linuxsampler.org/screenshots.html
For example the channel strips could use a knob for volume instead of a
fader
which would save some space.
Thorsten in what format do you plan
Dave Robillard wrote:
You misunderstand. I'm not saying you're stupid for suggesting using
the win32 codecs (if that's what you have to do to play the damn
things, that's what you have to do, it's hardly your fault). I'm
saying Lionstracs is stupid for requiring you to do so. Especially
Hi,
just back from NAMM, just a few infos for the interested people:
Lionstracs had a booth demoing a preliminary version of the
Mediastation X-76 ( http://www.lionstracs.com )
I've read a thread here about people complaining about the
price of the keyboard. ( $12,000).
I fully agree, the problem
Jens M Andreasen wrote:
If the GUI is not running in realtime, then things like changing the
patch number from the midi stream won't be reflected instantaniously on
screen. To the contrary: You can almost imagine SuperMario running up
and down the interface with his little brushes, slowly
Paul Davis wrote:
now, if the audio thread is burning so much CPU time that the GUI
doesn't get to run, its certainly a problem. but step back - is it a
problem you want to fix by raising the priority of the GUI thread so
that it steals time from the audio thread? or even from a disk
butler
Every CD player I tested seem only to skips sectors when doing fast
forward.
You hear short pieces of the song at original pitch. Perhaps they do
some crossfading between
audio chunks to lower pops and clicks, I don't know.
I think the only exceptions could be DJ-style players that perhaps allow
Disclaimer: I have not read the entire GUI thread so please don't flame
me (but correct me)
if I say nonsense.
Having written a few engines (midi player, audio player) and the
corresponding GUIs of
the upcoming Lionstracs Mediastation keyboard, we have faced the same
problem old problem:
we
Thomas Webb wrote:
I love people when people makes such allegations.
Where did you get the proof that it is crippled ?
It's a full fledged PC with
[...]
I love when people quote me out of context. The next
thing i said was withold the cripple, but double the
cost It is the full-flegedness
Hi,
a few clarifications about the Mediastation X-76
I think it was inappropriate at this time to post this on slashdot but
you cannot hide a website from the
technology hungy geeks.
So far so good:
I'm one of the guys involved in that project. (as external software
developer)
The Mediastation
Thomas Webb wrote:
It's just a frikkan computer! I'm sorry, but I'm not a
big fan of these synth workstations. They are just
crippled computers and in this case,
I love people when people makes such allegations.
Where did you get the proof that it is crippled ?
It's a full fledged PC with
Ivica Bukvic wrote:
But the sample rate *was* specified to 44.1 kHz in this case, wasn't
it...?
Well if you wanna get *technical* about it, the hdsp tools (which was in
the screenshot) on Windows reflects the same latency values regardless
of what sampling rate you use (they do not change
Florian Schirmer wrote:
Hi,
Anyone got more infos about this ?
(I'd interested if Wine is performant enough etc )
Threading and file access functions are kind of sluggish but overall
behaviour is very good. With some tweaks like direct ALSA glibc interface
the (worst case) latency
Hi,
I regularly read the german Keyboards magazine and in some
occasions they tested some Windows softsynths/HDR apps and measured
voicecount/number of parallel instances of plugins.
They usually talk about 3 msec latency case but in one occasion
I've seen they talked about 1.5msec latency too.
I
basically my assumption seems to be correct (I hope so :-) ).
Perhaps these latency journalists are exactly those that say that
Linux audio is not a reality
and will have no future anyway :-)
cheers,
Benno.
Benno Senoner wrote:
Hi,
I regularly read the german Keyboards magazine and in some
occasions
This is a multi-part message in MIME format.
Frank Barknecht wrote:
Hallo,
Benno Senoner hat gesagt: // Benno Senoner wrote:
Perhaps these latency journalists are exactly those that say that
Linux audio is not a reality and will have no future anyway :-)
Write them about the latency figures. I'm constantly confused by the
numbers
From here:
http://www.native-instruments.com/index.php?fsfeatures_us
In TRAKTOR FS the Final Scratch System has a high-performance DJ
software that has been specifically optimized for the needs of Final
Scratch users.
System Requirements:
*Runs on standard PCs, operating on Linux OS (all
Hi,
I was wondering what's the correct way to handle the sustain pedal when
implementing a MIDI sound generating module.
from the MIDI specs:
---
Hold Pedal, controller number: 64:
When on, this holds (ie, sustains) notes that are playing, even if the
musician releases the notes. (ie, The
Alfons Adriaensen wrote:
On Mon, Oct 27, 2003 at 12:44:45PM +0100, Benno Senoner wrote:
Assume no sustain pedal for now.
When I press C2 I hear the note. When I release it the sound does not
vanish immediately but takes a small amout of time to decay due to the
release envelope. If after
Robert Jonsson wrote:
Since we are in a world where 64bit native is only a recompile away (tm)...
Anyone have any insights on the possible performance improvements with that
perspective?
I've heard the Athlon64 has lots of registers, lots of cache, more pipelines
and, well, it's 64bit, it
Taybin Rutkin wrote:
From my understanding, they are completely different chips with different instruction sets. I seriously doubt the intel compiler will produce x86-64 instructions. Not to mention the business perspective...
Yep Taybin is right, AMD64 and IA64 (Itanium) instruction sets are
Hi,
yesterday was the opening of the Southtyrol game,
one of the worlds largest hand carved pinball style game machines.
(length: 11 metres, weight: 2.6 tons, 16 audio speakers)
It is an interesting combination of art, electronics and audio powered
by Linux. The game is located in Southtyrol -
Paul Davis wrote:
I hope this is not true:
Embedded systems often need to poll hardware or do other tasks on a
fixed schedule. POSIX timers make it easy to arrange any task to get
scheduled periodically. The clock that the timer uses can be set to
tick at a rate a fine as one kilohertz, so
the current LAD site (http://www.linuxdj.com/audio/lad )
(or http://www.linuxaudiodev.org )
runs on a machine with PHP4 and full MySQL support.
Joern N. (the LAD web-meister) can install any package he likes
(wi-ki etc) he should just contact me if he needs a MySQL DB
so I can setup it within
Greg Reddin wrote:
Is there a web resource or book or something where one can learn
about these programming caveats for audio developers? I knew nothing
of this problem, but it makes sense when you mention it.
I don't think that there is an all-in-one howto around, most infos
are scattered
Andrew Morton wrote:
At http://www.zip.com.au/~akpm/linux/2.4.20-low-latency.patch.gz
Very much in sustaining mode. It includes a fix for a livelock
problem in fsync() from Stephen Tweedie.
Hi,
I have not currently the possibility to test this patch for the next 2-3
weeks but I'd be interested
I heard of USB speakers, could this be a valid solution ?
Are they supported under linux ? If yes which kind of models ?
they're basically usb audio devices with amp and speaker in one
box - so pretty much all of them work out of box (even the harman
kardon ones that don't work with windows)
Hi,
a friend of mine is building a system that plays sounds on different
speakers (up to 8-10 mono channels ) based on certain input events such
as light sensors and mechanical triggers.
The audio quality does not need to be high.
Latency can be as high as 100 msec.
My question is what is the
Interesting article about achieving sub-msec response times by
dedicating one of the CPUs of a SMP box to high priority tasks using
a CPU shielding method.
http://www.linuxdevices.com/articles/AT8610061752.html
Unfortunately most of us have only single CPU boxes which means that
we rely on the
Hi,
does anyone know if it is possible to make the Free Intel C Compiler
work on Red Hat 8.0 ?
It used to work on RH7.3 but Jussi L. reported failure on RH8.0 too.
I would just be courious about Intel compiler's efficiency since I am
currently performing some resampling / mixing benchmarks using
Don't worry David, I think we are wasting our precious time with these
talks (but it is interesting anyway :-) )
Regarding source and binary code:
Being or app opensource, we have the advantage that the source code
does not constitute a runnable application and this provides several
legal
Gigasampler / Gigastudio vs Halion comparison:
http://www.nemesysmusic.com/news/NewsLet/newslet_Mar2002.html
no comment :-)
Benno
--
http://linuxsampler.sourceforge.net
Building a professional grade software sampler for Linux.
Please help us designing and developing it.
Paul what you described is very unfortunate. Although I have not had
time to perform any tests lately, I always had the bad feeling that the
out-of-process model would cost us some performance because the kernel
would screw us in some way.
As said, two years ago I was able to achieve 2.1msec
Paul Davis wrote:
it shouldn't be *too* difficult ...
Someone suggested LADSPA but I do not see an easy way to do it since LADSPA
does not support MIDI and this seems a VST2 instrument.
Time to intruduce an instrument API or extend LADSPA ?
Of course one can write a standalone JACK client
Hi,
I meant of sending midi events from the master keyboard via USB to the PC
which runs an internal soft-synth/sampler.
In that case USB should improve timing because of the bigger bandwidth,
(but will add 1msec of latency due to USB1 polling mechanism as far as I can
understand).
Correct me
AFAIK the Oxygen8 does not behave as a standard usb midi device, it uses a
Midiman protocol instead, so the standard drivers don't know what to do
with it. If you connect it and do an lsusb -v you don't see much...
Ok if the Edirol keyboards support standard MIDI then there shouldn't be
Perhaps old news but I saw these interesting USB MIDI master keyboards
http://www.harmony-central.com/Newp/2002/PCR-30-PCR-50.html
I'm considering getting one so my questions are:
- What's the status of USB MIDI on Linux ? Are such kind of keyboards fully
supported ? (eg transmit all the
As promised, I'd like to revive the linux audio sampler I was working on
about 2 years ago.
I was forced to take a long pause (almost 2 years) from LAD stuff
because I had to finish my CS degree before the retirement age.
But speaking speaking with various developers on LAD there seems big
Hello Joern,
Are you using ALSA right ?
Perhaps an OSS emulation problem of ALSA ?
I recall that I got grabled sound (even lockups) on the SBLive
when using 128byte fragments (seemed like a driver or hardware problem).
What kind of audio card do you have ?
Anyway latencytest is completely outdated
If I remember correctly, the guy that wrote the VQF (an mp3-like codec)
plugin for xmms
(home page here:http://www.csn.ul.ie/~mel/projects/linux/vqfplugin/ ) used
wine to run the windows version of the audio codec under linux. Reading form
his page he has now switched to a native version of
I don't know if the 3 points I've made have already been done, I'm new
to LAD, but they are important. Especially 3 as it gives a migration
path to existing windoze users. I know, I know, DirectX plugins are
written against Microsoft libraries, but how come mplayer on Linux can
use windows
Hi,
Udo Jocher (a friend of mine) has built RPM packages of the 2.4.18 kernel +
lowlatency + preemption patches. (both UP and SMP)
The kernel should deliver the performance described in the latest redhat
lowlatency tests.
He compiled the kernels on a Redhat 7.2 box.
the kernels are here:
On Saturday 23 March 2002 04:46 am, you wrote:
there is no threads mode unless you mean IPC mode. that works fine
at a hardware interrupt time of 64 frames 48kHz with at least 3
clients, at least on some user's machines (other people seem to have
problems that are hard to identify). in
Hi all, got a some spare cycles to read LAD again :-)
I saw the linuxdevices.com article too, indeed quite nice results especially
given that the testing period was 15 hours (1.5msec max latency with the
combined preemt+lowlat patch).
BTW: what does the performance of jack in plugin mode vs
Sounds good, certainly more marketable than LAAGA
(which reminds me lago which in italian means lake)
:-)
cheers,
Benno.
On Thursday 26 July 2001 17:38, you wrote:
would it be too dreadfully obnoxious and steinberg sniping to rename
LAAGA as FreeWire ?
On Monday 23 July 2001 19:13, Richard C. Burnett wrote:
I found this article on www.prorec.com on Windows 2000, thought you all
would find it interesting:
in the average PC Windows 2000 is able to meet worst case high priority
task latencies close to 1 ms, while Windows 9X
does not even
Hello,
first thank you guys for the ALSA tips:
it turned out to be a bug in ALSA 0.9b5.
Now I got ALSA-CVS + Muse 0.4.3 and EVO running through alsa seq.
My app crashes quite a lot but this is because the code is really a hack.
I hope to clean it up a bit and make it more stable soon. At that
On Friday 13 July 2001 14:37, Dave Phillips wrote:
Yes, you do. I do so by utilizing the sfxload utility:
sfxload /home/dlphilp/sfonts/8mbgmsfx.sf2
I think it comes from Takashi's site (?). The SF2 package came from the
SBLive CD (I think).
thanks , I will download th sfxload stuff.
On Thursday 12 July 2001 08:04, Abramo Bagnara wrote:
still no luck.
You have not installed alsa.conf. There was a bug I fixed some time ago
pointed to me by Paul. I don't remember if 0.9b5 contains the fix or
not.
Please use current CVS.
Ok I will do so, but will I be able to run
On Wednesday 11 July 2001 15:22, Greg Berchin wrote:
Even without SIMD, GP processors are getting so fast that DSPs are no
longer the only game in town.
Yes, I do agree: my samples does not use SIMD and it does pretty well already.
The only problem with PCs is that you have to be careful
On Wednesday 11 July 2001 19:11, Paul Davis wrote:
So what do you suggest Paul ?
Is my problem normal ?
(I can't even run alsactl store, while apps in OSS emulation mode work
perfectly)
if you have one soundcard, then ALSA should work out-of-the-box, so to
speak.
you only need a
Hi,
don't worry no overrun occurred: there is only a single peak at the beginning.
It happens on some cards (my old Tropez+ shows the same behaviour).
It seems that it is due to the card needing some time to startup when you
write the first fragment.
Your latency diagrams perfectly fine and
The results are horrendous for that kind of machine.
Did you tune all your disks ? (see the README of latencytest)
did you turn APM off ? (it caused some spikes on some of my boxes too)
(you can boot the kernel with apm=off on the lilo commandline).
are you using 2.4.5+andrew lowlat patches
David Olofson has done some testing on that field some tima ago
(the driver works only with ES1370 (AudioPCI) )
http://www.linuxdj.com/maia/audiality/download.html
But the key question is: do you really need 0.5msec latency for your app ?
What kind of apps are you working on ?
Josh Green has
Hi, I'm trying to setup ALSA on my box (last version I compiled was 0.5.x
months ago), in order to perform some stresstests using
a sequencer (Muse) that sends midi events to the disk sampler usin the alsa
seq API.
I sent a posting to alsa-user but I'm not on the list and did not get back
On Mon, 02 Jul 2001, you wrote:
HI!
Worked on a shirt logo for the LAD crew on the LinuxTag over the weekend.
Check it out at http://rangoon.50g.com
Comments wellcome!
cool !
Will the shirts be ready for LinuxTag ?
Would it be possible to print the URL www.linuxaudiodev.org below the
On Wed, 27 Jun 2001, you wrote:
- Original Message -
From: Paul Davis [EMAIL PROTECTED]
Phil, thanks for the pointer to PA. I'm glad to see that you've also
adopted a callback model. PA looks quite pleasant to use and not too
far from part of what LAAGA is aiming at. However,
On Wed, 27 Jun 2001, Steve Harris wrote:
Hi,
Does anyone know of a program that is eaasy to build and will let me look
at the events in a MIDI file? The file only containts CC's. I've tried
jazz++ (didn't seem to work) and midimountain (wouldn't build).
Have you tried Muse ?
On Tue, 26 Jun 2001, you wrote:
Hello,
I would like LAAGA to be a fast, lean, easy to use, higher
level (than the sound driver itself) API that _hides_ the
hardware interface completely from all clients that use it. It
I want slightly more than that actually. I want it to hide (like
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