Re: [LAD] Patching linux 2.6.26.3
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Rui Nuno Capela wrote: Tim Goetze wrote: [Tim Goetze] [victor] I was told to revert to 2.6.24.17 (not possible in my specific case, but there you go), in this list. Or to join the tuner's list. http://www.kernel.org/pub/linux/kernel/projects/rt/patch-2.6.26.3-rt2.bz2 worked for me. Applied cleanly and compiled well after turning off some RCU-related preemption options that caused compilation errors, but I was in no mood to find out the exact how and why. So far, it has collected a few hours of solid uptime too, but I haven't done any latency measuring. Update: on my laptop, the patched 2.6.26.3 kernel boots into an endless list of tracebacks on the console. On the main box, USB MIDI input is only read as soon as a key is pressed on the USB keyboard (the one with the letters, not the MIDI one ...). USB MIDI out is broken, too. So it's back to the old version. ah, it seems i'm not alone. hehe , nice try. We won't let you go that easily ;) i'm currently recovering myself from schock after returning from vacation and while trying 2.6.26.x-rt for the rentr�e it all seemed to work fine except omg... midi timing is a wreck, specially wrt.alsa sequencer. event delivery is completely fubar. i mean, completely. true showstopper, whatever :( cacophony seems to be the right word to express what it is. however, didn't had the time to check whether NOHZ is at stake. i'm certainly going back to 2.6.25.x-rt where things are still sane and pleasant for a while. btw, having NOHZ=y (aka tickless kernel) has been the norm here, since its inception CONFIG_NO_HZ=y - same norm here; with good results iff it works. I'll forward this email to LAT. MIDI with 2.6.26 MIDI has not been mentioned there. Since there's more of you LAD's compiling your own kernel that does not work, what about posting .config, version/patchset and compiler info before we all run into the same walls. - We're planning something like this with http://wiki.linuxaudio.org/wiki/kernel/ Does anyone know of similar projects? How in your opinion can LAO best supplement [rt] kernel bug tracking? There's 2.6.26-rtX testing packages from various distributions eg: http://linux.ilmainen.net/musix/temp/ or at http://guests.goto10.org/~krgn/files/ more at LAT - none of which is stable at the moment. robin -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkiu6E4ACgkQeVUk8U+VK0ItUwCguWgCtuPlnZoQQHVIDpVJRtvt suYAoKkv8JvmehvEOtKq81ameZnxw7yn =DXcK -END PGP SIGNATURE- ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev
Re: [LAD] LV2 in Ardour 3?
Darren Landrum wrote: I seem to recall some discussion involving the implementation of LV2 as a part of Ardour 3, along with the MIDI functionality. I was just hoping to confirm whether this is true or not. for MIDI see http://ardour.org/node/1162 . LV2 is even supported in ardour-2.5 (starting 2.3?). If it is, what LV2 extensions are going to be supported? I'm really hoping for MIDI, GUIs, and probably the port grouping as well. The ardour-users or ardour-dev mailing lists at http://lists.ardour.org/ may have an answer to that. sorry I can't be of more help there. ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev
Re: [LAD] best setup for RT audio with kernel 2.6.25
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 victor wrote: Hi everyone, what would be your suggestions as to the best setup (parameters/patches) for customising 2.6.25 regarding scheduling and realtime audio? i think there's no straight answer. run 2.6.24[.7-rt17] or join http://lists.linuxaudio.org/listinfo/linux-audio-tuning robin -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkicgqEACgkQeVUk8U+VK0Ln+ACfazfX2JwVG/DzKsxMYOS4qIvt AQ4AnRpvfJlY/og3+s+R3MpDu2HmZszt =fYsJ -END PGP SIGNATURE- ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev
Re: [LAD] Regarding the linux-audio lists announcing policy
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 MarcOChapeau wrote: On Tue, 08 Jul 2008 16:52:16 +0200 (CEST), Kjetil S. Matheussen [EMAIL PROTECTED] wrote: 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. Hi Kjetil, Robin Gareus has made last year an interesting study on this matter. It seems that the proportion of users that are subscribed to all three lists is quite small. So cross posting announces on the 3 lists isn't such a bad thing. Now regarding the fact that not everyone knows they should post on the 3 lists, I don't really think that's the case. most announces I see are posted at least on 2 lists. LAA is nearly always one of them. Some people choose to add LAD and LAU, some choose to add also their project list. I just ran the script again: as of July 8 2008 there are 1760 unique subscribers @linuxaudio.org's 4 lists (laa,lau,lad,consortium). The majority of users is subscribed to ONE list only. # [EMAIL PROTECTED] = head-count [1] = 1255 [2] = 315 [3] = 160 [4] = 30 so long, robin -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) iD8DBQFIc42BeVUk8U+VK0IRAvq4AJ0WbEynpWrl71XGVi6SyW0q9VuZOwCfRiry ATg2dheRwpweXSucCn+7v5E= =Lzvy -END PGP SIGNATURE- ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev
[LAD] test message - please ignore
hello world, again. ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev
[LAD] maintenance @ linuxaudio.org
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hooray, To provide better performance and reliability linuxaudio.org is replacing old hardware and upgrading the network bandwidth ! Scheduled server maintenance - Fri Oct 12 2007. The servers will be down for a short time around 18.00 CEST, 12pm EST or 9am PST. The HTTP services will resume almost immediately (reboot). but the mailing lists might be off-line for up to 3 hours. During this time the list server will not accept incoming messages and will not distribute messages. Any messages sent to the lists should be queued by the sending mail servers and will be accepted once the new list server is brought into service. #robin, for the linuxaudio.org team -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) iD8DBQFHDjxieVUk8U+VK0IRAobPAJ9OgEje2zELBE5Bug1Y3f6gTJgD1QCfQx+I Q/5QoxJoq63BEcg4rP64uzY= =5saU -END PGP SIGNATURE- ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo.cgi/linux-audio-dev
Re: [LAD] How to get correct midi timings from ALSA using the library only
Carlo Florendo wrote: How could I get the app to u|nanosleep() in the most accurate way in userspace clock_nanosleep() - see also http://linuxaudio.org/pipermail/linux-audio-dev/2007-March/018691.html robin ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo.cgi/linux-audio-dev
Re: [LAD] simulating analog audio devices part II
hello again. Sorry for the late reply, but I could not spare any time to work on it the previous week(s).. porl sheean wrote: I have not progressed to simulating tube-amps or synths yet.. lack of time, netlists and tube-models; it's low priority ATM. i don't mind making the circuits (in oregano or whatever), but i don't have decent models of tubes or transformers, I'm sorry; can't help you on this one. Should you find some, I'm interested, too. Try searching the internet for device-name and add some spice model syntax (eg. Is= Rs= ... #more `locate .model`) - that helped me to find a specific Zener-Diode model, but tubes are more tricky; I suppose most models of them are copyrighted somehow. and am not sure how to easily work with them in oregano (it's been a while, be kind)... I barely scratched the surface of oregano. Modellibrary preferences are not accessible in the 0.60.0 version - I got lucky placing a .model file named after the device in /usr/share/oregano/models/ before I switched to editing netslists with a text-editor after an initially generating them with oregano. I've only just installed qucs and like it. Working on schematic entry it's much faster! (oreagno's wiring still sucks) i do however have access to a lot of classic amp schematics (fender, marshall, vox etc) and don't mind spending time capturing them into oregano etc if it will help. It'd be great to run a full simulation of a setup. more later, robin ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo.cgi/linux-audio-dev
Re: [LAD] simulating analog audio devices part II
Giuseppe Zompatori wrote: Wow, your ltSpice simulation sounds really real! I hoped to get ngSpice to do the same. Now days I am messing around with GNUCap and QUCS (http://qucs.sf.net) to which I just ported my first tube model whose plate voltage/grid current curves look like that: great! looking forward to listening to those ;) Hopefully a way to inject wav files in both GNUCap and QUCS will come true as they both seem better than spice derived programs. the sndfileresample wrappers that I patched into ngSpice just work with GNUCap and QUCS too. GNUCap is still quite unusable (segfaults on large data sets) but I've elaborated on the QUCS hack - It's much slower than ngSpice, but indeed more accurate (at least the timing); I'm still to run some tests.. http://mir.dnsalias.com/oss/spicesound/qucs - (highly experimental) - I did not get to clean it up last week as I intended to and can't spare any time this week either.. - but I plan to revisit the issue properly over the summer. keep in touch, robin ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo.cgi/linux-audio-dev
Re: [LAD] simulating analog audio devices part II
Stefano D'Angelo wrote: Hi, I read on spicy sound website: However todays computing power allows to do so almost in real-time!... do you think you'll get it real-time? no, not any more. Without oversampling, small circuits run /almost/ in real-time. However the posted 3 seconds of guitar took almost 4 mins to be processed with ngSpice simulating at 1s/(64*48k) timesteps. robin ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo.cgi/linux-audio-dev
Re: [LAD] simulating analog audio devices part II
Stefano D'Angelo wrote: Il giorno gio, 07/06/2007 alle 09.20 +0200, Robin Gareus ha scritto: Stefano D'Angelo wrote: Hi, I read on spicy sound website: However todays computing power allows to do so almost in real-time!... do you think you'll get it real-time? no, not any more. Without oversampling, small circuits run /almost/ in real-time. However the posted 3 seconds of guitar took almost 4 mins to be processed with ngSpice simulating at 1s/(64*48k) timesteps. 4 mins? OMG! :-\ indeed - at least it scales linearly with decreasing timesteps ;) - for guitar a *8 or *16 oversampling is sufficient. ngSpice is clearly is not suitable to be used as audio-effect; yet it's handy to testdebug audio-circuits, or even create convolution samples. Does anyone here have access to the full version of http://www.acusticaudio.com/modules.php?name=Productsfile=nebula2free VST plugin? - It would be interesting to know if ngSpice-snd samples can be used to program it. nudged by Erik at http://linuxaudio.org/pipermail/linux-audio-user/2007-May/045899.html I've Taylor-expanded a convolution integral, and started to look for some papers. The maths is not too wicked after all; but coding a volterra transfer function best with CPU optimization is no laughing matter.. - I've chatted with a physicist friend who has access to some pro-maths software to crunch integrals and product series and experience doing so - I'm gonna spend a few weeks with Florian during the summer and 'll get back to you guys. ATM my ToDo list is way too overloaded and just glimpsing at bruteFIR or aliki: this is no small project either! robin ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo.cgi/linux-audio-dev
[LAD] simulating analog audio devices part II
Hello again, I've added libsamplerate for resampling/oversampling which - as expected - dramatically improves the quality of the ngspice processed sound. Here's some example 3sec guitar sound from current testing: http://mir.dnsalias.com/_media/oss/spicesound/git-fuzz64.mp3 (left channel: resampled input-sound, right channel: fuzz-effect out) http://mir.dnsalias.com/oss/spicesound/examples#audio_example The fuzz effect still sounds a little odd, but I believe that a DI-recording would sound just like that ;) - I have not progressed to simulating tube-amps or synths yet.. lack of time, netlists and tube-models; it's low priority ATM. robin ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo.cgi/linux-audio-dev
Re: [LAD] simulating analog audio devices
Pieter Palmers wrote: Pieter Palmers wrote: Erik de Castro Lopo wrote: Robin Gareus wrote: hey LADs. For those of you who have not followed the 'fuzztone' thread on LAU. I'd like to announce a /cool hack/ to ngSpice that provides soundfile I/O capabilities. - it's more a LAD than a LAU issue anyway. This is a CoolAsAllFsck (tm) hack as far as I am concerned. Congrats. Simple tests sound rather promising; but my fuzztone experiment is not really satisfying yet. well, maybe it's just meant to sound *that* weird ;) I used Spice as part of my engineering degree and also in jobs later, but my memory of this is definitely hazy. However, my memories of how spice works is my discretizing the differential equations that describe the components. I believe the specification of the time step determines how fine grained (in time) a grid will be used for operating on the diff equations. One thing you might have missed is matching up the samplerate of the input file with the time grid of the spice calculations. You alos want to be *really* careful about how you interpolate the file data to spice sample rate. Please, please, please, do not use linear interpolation. Secret Rabbit Code is probably the way to go here. hehe, right. ATM i don't interpolate at all. The input just stays for the duration of the sample. - there might even be some rounding issues here as ngSpice stores the time in (double) seconds. Can I feed the rabbit at irregular intervals: ie. specify time and sample? All spices I use (although this excludes ngspice) have an option to force them to calculate the response at specified timesteps (along with the ones they need for accuracy). In this case it seems obvious to set this timestep to 1/Fsample. So far I've only found the command to set the *maximum* time-step, not an option to enforce it. - setting it to 1/Fsample is ok. setting it to 0.25/Fsample and using only about every 4th value gives much better results.. using 0.01/Fsample makes me fetch a few more coffees; robin ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo.cgi/linux-audio-dev
Re: [LAD] simulating analog audio devices
Erik de Castro Lopo wrote: Robin Gareus wrote: When Spice loads the libsndfile component, can the component find out the inter-sample period used by Spice? no. Spice does not use a constant interval; else it would be straight forward to use libsamplerate. If the above is not possible, I suggest that you up sample the audio by say 128 times and then linearly interpolate to get whatever inter sample values you need. sounds like a great idea. I should do that with the output too, but it's a little more complex there: In the beginning I've tried to take the average for all the spice-samples matching the corresponding audio-sample. I've only tried that up to 1/(4*Fsample) and it gave much worse results than just taking the closest sample... - combining interpolation and resampling could be the solution. now that I've got the framework, I'll dig into the details.. robin ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo.cgi/linux-audio-dev
Re: [LAD] Yamaha XS - more audio products using Linux
Ben Loftis wrote: Here's my list of audio products that are based on Linux. Does anybody have more to add? Is there somewhere online we can post and maintain a list? Harrison consoles Yamaha Motif XS Korg OASYS Waves DPA Roland Edirol RG-100 Lemur multi-touch controller Muse Receptor Plugzilla HHB Portadrive Midas XL8 Lawo consoles Hartman Neuron Lionstracs MediaStation keyboard LCS Systems great - thanks for the list! - I could not find URLs or a Linux-reference for all of them.. but I've started to add these to the apps-wiki tagged as Linux_Based_Products. http://apps.linuxaudio.org/apps/categories/linux_based_products ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo.cgi/linux-audio-dev
Re: Lil' hack-together2
Maciej Podkomorzy wrote: It does look a lot like Insanity, and if that's the general direction you're headed in, Sham and I would be happy to have you help out on insanity. :) Well, the main difference is that Insanity is GTK/Python, and Blastwave GTK/C so I don't think I could be of any assistance:) gx2ctl is also GTK/C; (actually glade-2/libgnome) there are a few bugs, but it works and is low-pri on my ToDo list. http://mir.dnsalias.com/oss/xmms2/gx2ctl git clone git://robstar.dyndns.org/gx2ctl.git might come in handy for you. @xmms2 team: do you guys want to provide a mirror or master-repo for it on git.xmms.se ? - you could just 'pull' and I'll drop by on IRC some time for PM for a push-account. cheers, robin ___ Xmms2-devel mailing list Xmms2-devel@lists.xmms.se http://lists.xmms.se/cgi-bin/mailman/listinfo/xmms2-devel
[linux-audio-dev] april fool!
Re: [linux-audio-dev] [ANN] List migration on the 1st of april
Marc-Olivier Barre wrote: On 3/30/07, Frank Barknecht [EMAIL PROTECTED] wrote: Hallo, Bengt G�rd�n hat gesagt: // Bengt G�rd�n wrote: List-id: lad.lists.linuxaudio.org List-id: lau.lists.linuxaudio.org List-id: laa.lists.linuxaudio.org I would love if the current List-id's could be kept, except the list-addresses. I already filter by List-Id, but only up to the mail address: The List-Id's will be the same. no problem on that. are you sure? - IIRC we've changed the domainname ;) NTL, Frank's procmailrc (/^List-Id: A list for linux audio users/) will continue to work after the migration. Here are a few example headers of a Test-messages to the *new* LAD: X-BeenThere: [EMAIL PROTECTED] Reply-To: The Linux Audio Developers' Mailing List [EMAIL PROTECTED] List-Id: The Linux Audio Developers' Mailing List linux-audio-dev.lists.linuxaudio.org Sender: [EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] We are canonicalizing the Email-addresses (eg. rewriting [EMAIL PROTECTED] to [EMAIL PROTECTED]) - for all *new* email. - BUT we will not touch/modify any archived emails (@music.columbia.edu)! ciao, robin
Re: [linux-audio-dev] [ANN] List migration on the 1st of april
Marc-Olivier Barre wrote: Dear members, I am proud to announce that we will proceed with the list migration during the night between the 31st March and the 1st April (at 0h00 GMT). From this point on, emails should be sent to these addresses : - [EMAIL PROTECTED] for the developper mailing list - [EMAIL PROTECTED] for the user mailing list - [EMAIL PROTECTED] for the announce mailing list The following aliases have been defined, to make it look nicer. Consider them as shorter equivalents : - [EMAIL PROTECTED] for the developer mailing list - [EMAIL PROTECTED] for the user mailing list [EMAIL PROTECTED] [EMAIL PROTECTED] work too! - there's no shortcut for the announce list! No alias for the announce list since it is an open list (non subscribers are allowed to post) and we fear a increase in spam with a too short name. Also, to respond to a request to a few of you working with command line mail clients, the list tag at the beginning of the subject line will be shortened : [LAD] for linux-audio-dev [LAU] for linux-audio-user [LAA] for linux-audio-announce You may need to update your email filters accordingly. i suggest not to filter on Subject but on Email-Header: X-BeenThere: [EMAIL PROTECTED] X-BeenThere: [EMAIL PROTECTED] X-BeenThere: [EMAIL PROTECTED] cheers, robin
[linux-audio-dev] OT: dev helpers - WAS: Realtime problems with midi/osc sequencer
Stephen Sinclair wrote: You could also try sigaction and setitimer. I've had good timing results with this approach in the past. (I haven't tried it for audio tasks though.) does anyone know a URL to a cute POSIX C fn-reference postscript? judging by http://www.oreilly.com/catalog/posix4/toc.html it seems possible to get all of them on a max. dbl-sided A4 sheet?! the man-pages are great if one knows about the function in the first place - is there a fn-list on a hidden man-page? I'd even be happy with some offline-RFC that I can GrepSedAwk(TM) but prefer sth. printable ;-) - what do you guys use? robin
[linux-audio-dev] LAC aftermath - non-GUI announce hattrick
From the debug OSC during lecture talks dept.
* dump UDP data from a port to stdout
* forward/relay UDP data to one or more UDP ports.
- C source attached - no compile flags needed
The hacked in Berlin's U-bahn dept. pre-releases
jplay2 (formerly jadio) - jack-plays-it-all
http://mir.dnsalias.com/oss/jplay2/start
- current Feature cloud:
Open-Sound-Controlled resample JACK-transport vari-speed
scrub-audio ffmpeg libquicktime libsndfile audio-cache
The no more xruns on app shutdown dept. has added a few
jack_deactivate() calls to various xj* gj* SVN and git repos.
and finally the berlin sync-it syndicate announces it's involvement in
http://ltcsmpte.sf.net
#basically all of it is C-code in progress that I did not consider to be
#releasable software before the LAC2007 ;-) - anyway they're helpers for
#an average day of A/V RD..
#robin
/*
udp_repeater -
Copyright (C) 1999,2006 Robin Gareus [EMAIL PROTECTED]
This program is free software; you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation; either version 2 of the License, or
(at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
$Id: repeater_udp.c,v 1.1.1.1 2006/07/16 11:24:50 rgareus Exp $
*/
#include sys/types.h
#include sys/socket.h
#include arpa/inet.h
#include netinet/in.h
#include netdb.h
#include stdio.h
#include string.h
#include stdlib.h
#include stdarg.h
#include unistd.h
typedef struct {
int sock;
struct sockaddr_in server, from;
char *hostname; // debug info
int port; // debug info
} remoteconnection;
void usage (char *name) {
fprintf(stderr, usage: %s listen-port -d \n,name);
fprintf(stderr, UDP dumper - listen on a local UDP port and dump\nraw data to STDOUT\n);
fprintf(stderr, usage: %s listen-port [host]:port [ [host]:port ]* \n,name);
fprintf(stderr, UDP repeater - this program listens on a local UDP port and \nforwards messages to one or more UDP ports.\nNot specifiying a hostname is equivalent to 'localhost'\nexample: %s :3334 :3335 192.168.6.66:666 receiver.jails.org:\n, name);
exit(0);
}
int splithp (char *arg, char **host, int *port) {
char *tmp = strchr(arg,':');
if (!tmp) return (1);
*tmp=0;
if (port) *port= atoi(tmp+1);
if (host) { if (arg == tmp) *host=strdup(localhost); else *host= strdup(arg); }
*tmp=':';
return (0);
}
void error(const char *format,...) {
va_list arglist;
char text[BUFSIZ];
va_start(arglist, format);
vsnprintf(text, BUFSIZ, format, arglist);
va_end(arglist);
text[BUFSIZ -1] =0;
perror(text);
exit(0);
}
int open_rc(remoteconnection *rc, char *host, int port) {
struct hostent *hp;
rc-sock= socket(AF_INET, SOCK_DGRAM, 0);
if (rc-sock 0) error(socket);
rc-server.sin_family = AF_INET;
hp = gethostbyname(host);
if (hp==0) error(Unknown host: '%s',host);
memmove((char *)rc-server.sin_addr,(char *)hp-h_addr, hp-h_length);
rc-server.sin_port = htons(port);
return(0);
}
int send_rc (remoteconnection *rc, char *buffer, size_t len) {
int n;
socklen_t length = sizeof(struct sockaddr_in);
n=sendto(rc-sock,buffer, len,0,(struct sockaddr *)(rc-server),length);
if (n 0) error(Sendto (%s:%i),rc-hostname,rc-port);
return (0);
}
int main(int argc, char *argv[])
{
int want_dump = 0;
int sock, n;
size_t length, fromlen;
struct sockaddr_in server;
struct sockaddr_in from;
char buf[BUFSIZ];
int lport = ;
int i = 2;
remoteconnection *rc = NULL;
if (argc 3) {
usage(argv[0]);
}
lport = atoi(argv[1]);
// printf( read from ANY LOCAL INTERFACE p:%i\n,lport);
rc=calloc((argc-2),sizeof(remoteconnection));
if (argc == 3 !strncmp(argv[2],-d,2)) {
want_dump = 1;
printf(raw data dump mode!\n);
argc=2;
}
for (i=2; i argc; i++) {
char *hostname=NULL;
int port=0;
if(!splithp(argv[i],hostname,port)) {
open_rc((rc[(i-2)]),hostname,port);
rc[(i-2)].port=port;
rc[(i-2)].hostname=hostname;
//printf( dup to h:%s p:%i\n,hostname,port);
//free(hostname);
}
}
sock=socket(AF_INET, SOCK_DGRAM, 0);
if (sock 0) error(Opening socket);
length = sizeof(server);
bzero(server,length);
server.sin_family=AF_INET;
server.sin_addr.s_addr=INADDR_ANY;
server.sin_port=htons(lport);
if (bind(sock,(struct sockaddr *)server,length)0) error(binding to port %i failed.,lport);
fromlen = sizeof(struct sockaddr_in);
while (1) {
n = recvfrom(sock,buf,BUFSIZ,0,(struct sockaddr *)from,fromlen
Re: [linux-audio-dev] promoting LAC 2007
Ivica Ico Bukvic wrote: We are, as we have been always, quite on our own in this little pocket universe of ours. I think this issue is amplified by the fact that the conference also targets by and large the same crowd. what's the difference between Linux-Audio-Developer meeting and Linux-Audio-Conference? historically Karlsruhe Thu/Fri vs. Sat/Sun - this year was different: Berlin weekdays vs. Berlin weekend - with a smaller audience on the last two days ;-) what is the location in Cologne like? - will there be Students lingering during week but not the weekend next year? I suggest a single (no parallel) devel-track next year - maybe some workshops can overlap or be in parallel with the last devel-talk of the day, which can be a demo.. all other tracks should be for general audience: musicians, home-users or professionals. Leaving the devs to attend, chat, hack, live-code or redo their presentation in a different style.. as for promoting Linux-Audio, what about setting up a Music Made with Linux webradio station? - what happened to all the audio-transmittals of LAC? is there an archive? robin
Re: [linux-audio-dev] Realtime problems with midi/osc sequencer
Christian wrote:
Robin Gareus schrieb:
usleep( iTick-( passedTime-startTime ) );
AFAIR usleep is not exact! - did you
echo 1024 /proc/sys/dev/rtc/max-user-freq ?
try sth like:
void select_sleep (int usec) {
fd_set fd;
int max_fd=0;
struct timeval tv = { 0, 0 };
tv.tv_sec = 0; tv.tv_usec = usec;
FD_ZERO(fd);
if (remote_en) {
max_fd=remote_fd_set(fd);
}
select(max_fd, fd, NULL, NULL, tv);
}
Interesting timing approach.
But I can't find remote_en and remote_fd_set in the man pages.
What does these arguments stand for?
sorry, cut the 3 if(remote_en) lines - I was too quick with pasting
sending the mail - remote_en is some global var. that allows to
interrupt the sleep, if some other-event occurs... - actually you'd only
needed select (0,fd,0,0,tv);
anyway clock_nanosleep seems better; at least it takes less code
to set it up. I did not know about it, and it's even POSIX, how cool!
#robin
[linux-audio-dev] note: apps.linuxaudio.org
There have been a few edits on our deveopment-wiki apps-devel.linuxaudio.org ! sorry for the confusion: http://apps.linuxaudio.org is the real site. http://apps-devel.linuxaudio.org/ is our *test server* that we use to experiment with PHP, plugins, designs, software updates, etc - it's actually a 1:1 mirror, but not intended for the public. - we ought to set up a jailed webserver behind openvpn for testing, but we're just not there yet :( robin
Re: [linux-audio-dev] Realtime problems with midi/osc sequencer
usleep( iTick-( passedTime-startTime ) );
AFAIR usleep is not exact! - did you
echo 1024 /proc/sys/dev/rtc/max-user-freq ?
try sth like:
void select_sleep (int usec) {
fd_set fd;
int max_fd=0;
struct timeval tv = { 0, 0 };
tv.tv_sec = 0; tv.tv_usec = usec;
FD_ZERO(fd);
if (remote_en) {
max_fd=remote_fd_set(fd);
}
select(max_fd, fd, NULL, NULL, tv);
}
Re: [linux-audio-dev] List migration to linuxaudio.org : remaining issues.
Marc-Olivier Barre wrote: Hi everyone, As I said yesterday when I said the migration to linuxaudio.org was postponed till 1st April, there are still a few issues asking for comments. It has been suggested by a few people(Ivica Ico Bukvic, Jan Weil) that the email addresses [EMAIL PROTECTED] was kind of redundant. The following suggestions were made : � just switch the names to [EMAIL PROTECTED], [EMAIL PROTECTED], and [EMAIL PROTECTED] � Make aliases : [EMAIL PROTECTED] - [EMAIL PROTECTED] To this Leonard Ritter replied that this could lead to more spam. he might be right. a random bot is more likely to spam [EMAIL PROTECTED] than [EMAIL PROTECTED] a compromise woud be la-user, la-dev , la-ann - or [EMAIL PROTECTED] (mailing-list), etc. (but that's not catchy enough) I'd suggest to alias the other way round: if [EMAIL PROTECTED] is Spammed, you can announce and drop this feature again while everyone with [EMAIL PROTECTED] in their address book won't ever notice. in short term [EMAIL PROTECTED] will have less SPAM than linux-audio-dev, and in the long-run SpamAssassin will mark dup.-postings in blacklists.. OTOH; with [EMAIL PROTECTED] - we'd benefit from a shorter subject! [la-dev] compared to [linux-audio-dev] - there are usually 50 chars for subject in a 80x24 terminal :) - for email shortcuts there is an addressbook and personally I use LAD as nick for it. robin
Re: [linux-audio-dev] List migration to linuxaudio.org : remaining issues.
Marc-Olivier Barre wrote: On 3/1/07, Robin Gareus [EMAIL PROTECTED] wrote: he might be right. a random bot is more likely to spam [EMAIL PROTECTED] than [EMAIL PROTECTED] Ok, right... a compromise woud be la-user, la-dev , la-ann - or [EMAIL PROTECTED] (mailing-list), etc. (but that's not catchy enough) Agreed I'd suggest to alias the other way round: if [EMAIL PROTECTED] is Spammed, you can announce and drop this feature again while everyone with [EMAIL PROTECTED] in their address book won't ever notice. If I understand you well, it means making [EMAIL PROTECTED] an alias (that can be dropped if causing too much issues) right ? yes. -lad , lau laa are even better aliases! retaining the orginal mailbox names, should also make it easier for (external-)email-archives to migrate to the new archive... in short term [EMAIL PROTECTED] will have less SPAM than linux-audio-dev, and in the long-run SpamAssassin will mark dup.-postings in blacklists.. Well, to be honest ATM there seems to be no spam on linux-audio-dev to be moderated and none received in my mailbox (from lad). It has always surprised me. lucky you :) - maybe those spammers are friendly when they sense a lost case or is linux-audio scaring them off? .grin. in this case you can even provide more aliases, not that it would make much sense... OTOH; with [EMAIL PROTECTED] - we'd benefit from a shorter subject! [la-dev] compared to [linux-audio-dev] - there are usually 50 chars for subject in a 80x24 terminal :) - for email shortcuts there is an addressbook and personally I use LAD as nick for it. Prefix for the subject line can be changed. It's the subject_prefix parameter. we could make it [LAD], [LAU] and [LAA] (that's the shortest I can think of and it would make terminal users happy). it might cause loads of readers to change their mail-filter(s). but moving to linuxaudio.org will alter the X-Been-There header anyway. just do it in one step. april fools day! gonna be fun, robin
[linux-audio-dev] Wiki for Linux Sound MIDI Software
With ongoing preparation to Integrate End User and Developer Resources at linuxaudio.org (workshop @ LAC2007) we are proud to announce a first version of Dave's linux-sound application index as public-wiki: http://apps.linuxaudio.org/ this is by far no cute end-user resource portal yet, but a first step in merging data and setting up a prototype back-end! We are looking for interested people to contribute at various levels and suggest to use the mailing list for general discussion while keeping dokuwiki and style related requests at http://apps.linuxaudio.org/wiki:open_discussion Further [major] back-end development on http://apps.linuxaudio.org/ will probably not happen until LAC2007, but we are interested to tweak the look and feel, fix bugs, and collect feedback. We're also looking for qualified LAU's and LAD's to verify and maintain the current content with editor privileges, responsibilities and deeds. next up: docs.linuxaudio.org: - static documentation (docbook, LateX, [x]html) - slides and presentations (PDF, ps, png jpg,..??) - music (collaborate with or outsource to freesound et al.) - include (external?) video presentations, movies, etc - source code (coding examples: wiki-page; svn-mirror/server ??) for each hosted documentation there will be an option for user annotation(s). eg. wiki-discussion page, blog-comments, email-notify[-mailing-list]? - some of it will be merged with apps.linuxaudio.org which will provide automatic indexing; details remain to be discussed. some of the intended supported formats imply moderation / maintained upload or installation of static documentation. - the ring of maintainers could evolve with key-signatures, and parts of the system may become completely open. - suggestions: http://apps-devel.linuxaudio.org/wiki:suggest_doc ico robin
Re: [linux-audio-dev] What does it mean for jack to be rolling (newby)
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Jonathan Ryshpan wrote: On Mon, 2007-02-19 at 13:18 -0800, vreuzon wrote: Jonathan Ryshpan a �crit : The above recording session was done while jack was Stopped. Would jack work better if it were Rolling? This play button refers to jack transport functions : The JACK Audio Connection Kit provides simple transport interfaces for starting, stopping and repositioning a set of clients. This document describes the overall design of these interfaces, their detailed specifications are in jack/transport.h from : http://jackit.sourceforge.net/docs/reference/html/transport-design.html Thanks for your quick reply. However... I have read this, and also part of the documentation of the transport.h File Reference to which it refers. Rolling is not defined anywhere; it's just used. Rolling (like Starting and Stopped) is a state of the jack-transport (SMPTE timecode) mechanism. (the diagram on the page) This has nothing to do with JACK audio-process callbacks which is/are always running! Stopping the jack-transport is just like turning off the motor on an old tape recorder while the amp (and patchbay) keeps working. every JACK application can *optionally* synchronize it's play position to jack-transport! AFAIR audacity does not support this (it has it's own motor ) #robin -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) iD8DBQFF2iCLeVUk8U+VK0IRAi5yAJ4i0G8y+v4ji/gBbUT6ePfzTRkO7gCgqJA0 z4Vk82zhQGzOj8bHTXlpFbw= =XyWe -END PGP SIGNATURE-
Re: [linux-audio-dev] donations or sponsoring?
Leonard Ritter wrote: How is a beginning open source project funded, usually? great goodness of heart and an ex-employer who lets you keep the keys - minimizing unnecessary accessories (shaving-kit, vacuum-cleaner, etc) helps a lot ;-) open-source coding is an Art not an Industry. - unless you want to sell support or merchandise, I recommend to get similar funding as Artists. Alas, most open-source code is [considered] craftsmanship not Art and it sells as such. - now compare it to music-business: it's a pitiful career unless (and even) if you're at the TOP; - ..usual exceptions.. a donation button to sell improvement-on-request seems the best option. If you get annoyed/broke: show it: donation-progress-splash-screen, change the default-app background to a picture of your fridge if a user has not made a donation for a month, etc... There is no dependency between you and your users, meaning that the choices you make might not necessarily be choices embraced by the community. Again, there is no contract, just a requirement of trust. that's a feature not e bug :) - I did not yet read your BLOG - but maybe you're looking for a shareware-license instead of GPL. It would be a pity though. open-source 2007: erst kommt das Fressen, dann kommt die Moral - You can certainly sell coding services to a community or individuals and make a binding contract under GPL. Is this an issue that the linux-audio-consortion could address? - set up a foundation to pay developers on project basis. (much like sourceforge donations - but first we need a wiki, blog, forum and project-MS :) ) robin PS. facts-from-a-parallel-universe: Vincent cut off his ear after 3days of debugging his rendering code.
Re: [linux-audio-dev] Old hat - comparison against windows
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Steve Harris wrote: On 31 Jan 2007, at 11:27, Bob Ham wrote: [...] I don't think that's necessarily the case, just because Linux had better RT performance in 2000 doesn't mean it still does today, with Vista and general improvements. I think it's reasonable for management to question if it's still the best choice. pick two out of three: cheap, quick, good. - Who's in the management? :) IMO most modern PC/PPC hardware and OS are equally good enough for non-hi-end audio engineering. - tweak your personal flavor! [hardware-]end-users are concerned about low-latency, scalability and reliability. (audio-processing-quality, easy-use, etc. are irrelevant here) Pro's for gnu/Linux that come to my mind: - flexibility (all kinds of... GNU) - reliability (you have the option to see what's the OS/HW is doing) - existing free Live-cds - active and open developer community ;-) - ... Cons; If windows wants it can perform better than a fully fledged rt-unix-kernel. - but that remains to be proven for Vista! It would be nice to have a software-suite to compare linux-audio-realtime performance .. - For linux/linux tests there seem to be various code-snippets out there... does anyone care to share pointers? robin -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) iD8DBQFFwJGUeVUk8U+VK0IRAlBcAJ92OtWRExkXGwzf1OFxdCTlXW2KLQCgpX+9 AbZjWHeYCQ47Zy5OnRebsnY= =incw -END PGP SIGNATURE-
Re: [linux-audio-dev] Old hat - comparison against windows
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Lars Luthman wrote: On Wed, 2007-01-31 at 13:54 +0100, Robin Gareus wrote: Cons; If windows wants it can perform better than a fully fledged rt-unix-kernel. - but that remains to be proven for Vista! Are you saying that this is true for XP? Are there any references for that? maybe not for audio-apps. I once read some books about subverting the windows kernel - one can do marvelous stuff and get great performance out of it!! - only half kiddingly. - luckily I have not booted into XP for years and this Con might just be crap... however some instinct tells me that a UNIX - no matter how pre-emptive or tweaked - will always have more overhead than some DirectX-app. - but IMO this is irrelevant for normal audio apps on modern machines.. - memory management and -locking are more crucial.. robin -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) iD8DBQFFwKJJeVUk8U+VK0IRAguvAKC5WkK6enqRl5YXpuqoY6hpdcjiCwCeKxSG Mqz4u/A+Eg8g/LD21Jvrlbc= =+Sl0 -END PGP SIGNATURE-
Bug#403651: libacl1: symbol setxattr ATTR_1.0 not defined libattr.so.1 (ATTR_1.1)
Package: libacl1 Version: 2.2.42-1 Severity: normal libattr1 2.4.35-1 + libacl1 2.2.42-1 cause major trouble on my system. eg: $ls -l ls: relocation error: /lib/libacl.so.1: symbol getxattr, version ATTR_1.0 not defined in file libattr.so.1 with link time reference I did not dare to reboot and downgraded to the prev version of both libs. It could be a false alarm: eg. libattr was still in use, but that should have been fixed by running ldconfig; and strings+grep found ATTR_1.1 in libattr but not libacl! #robin -- System Information: Debian Release: 4.0 APT prefers unstable APT policy: (500, 'unstable') Architecture: i386 (i686) Shell: /bin/sh linked to /bin/bash Kernel: Linux 2.6.17.13-rg Locale: LANG=en_US, LC_CTYPE=en_US (charmap=UTF-8) Versions of packages libacl1 depends on: ii libattr1 2.4.32-1Extended attribute shared library ii libc62.3.6.ds1-9 GNU C Library: Shared libraries libacl1 recommends no packages. -- debconf-show failed -- To UNSUBSCRIBE, email to [EMAIL PROTECTED] with a subject of unsubscribe. Trouble? Contact [EMAIL PROTECTED]
Re: [linux-audio-dev] [ANN] yet another linux sound wiki.
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Dave Phillips wrote: Robin Gareus wrote: I've started to convert the information from linux-sound.org into a distributable doku-wiki. http://linux-sound.sonologic.nl/ This is a prototype installation and experimental suggestion! Please review and comment before further development proceeds. Would I be too egotistical to request that there be some notice of the fact that the pages were originally mine ? Forked projects often nod towards their original makers. But it's just a courtesy, not a requirement. :) I'm sorry - I really am. even if you don't require it: I'd like to go GPL, CC or somewhat in that direction. Come to think of it i don't even remember putting my name in there somewhere yet. I'll get started on a About page.. robin -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFFfUHteVUk8U+VK0IRAtEhAJ46IBAUIopQqGVno7TCL795/bXMRQCgoq9o Pm2kYS+/uL5oxXMBxC1W0m8= =BjCn -END PGP SIGNATURE-
Bug#369469: libasound2: invalid datadir path - alsa.conf not found
Package: libasound2
Version: 1.0.11-6
Severity: important
/usr/lib/libasound.so.2.0.0 is compiled with
datadir=/home/jordi/svn/pkg-alsa/trunk/build-area/alsa-lib-1.0.11/${prefix}/share/alsa
libasound2 searches for alsa.conf in the build-dir rather than /usr/share/alsa/
- thus alsa apps fail to work until I create /usr/share/alsa as /home/jordi/...
or set the ALSA_CONFIG_PATH enviroment variable.
adding --datadir=/usr/share to the configure lines in debian/rules and
rebuilding libasound2 helped me out.
-robin
alsa-lib-1.0.11/configure.in:65 says:
eval dir=$datadir
case $dir in
/*) ;;
*) dir=$PWD/$dir
esac
\${prefix} is not replaced (maybe due to 40_relibtoolise.dpatch ??).
datadir starts with '$' instead of '/' - pwd is prepended and
'..${prefix}..' ends up in include/config.h and bibuild/include/config.h
$ strace alsamixer
...
stat64(/home/jordi/svn/pkg-alsa/trunk/build-area/alsa-lib-1.0.11//share/alsa/alsa.conf,
0xaf951b1c) = -1 ENOENT (No such file or directory)
write(2, ALSA lib control.c:816:(snd_ctl_..., 47ALSA lib
control.c:816:(snd_ctl_open_noupdate) ) = 47
write(2, Invalid CTL default, 19Invalid CTL default) = 19
write(2, \n, 1
) = 1
fstat64(1, {st_mode=S_IFCHR|0620, st_rdev=makedev(136, 9), ...}) = 0
mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) =
0xa7f3d000
write(1, \n, 1
) = 1
write(2, alsamixer: function snd_ctl_open..., 79alsamixer: function
snd_ctl_open failed for default: No such file or directory
) = 79
munmap(0xa7f3d000, 4096)= 0
exit_group(1) = ?
...
$ strings /usr/lib/libasound.so.2.0.0 | grep share
...
/home/jordi/svn/pkg-alsa/trunk/build-area/alsa-lib-1.0.11/${prefix}/share/alsa/%s
/home/jordi/svn/pkg-alsa/trunk/build-area/alsa-lib-1.0.11/${prefix}/share/alsa/alsa.conf
/home/jordi/svn/pkg-alsa/trunk/build-area/alsa-lib-1.0.11/${prefix}/share/alsa/smixer.conf
/home/jordi/svn/pkg-alsa/trunk/build-area/alsa-lib-1.0.11/${prefix}/share
...
-- System Information:
Debian Release: testing/unstable
APT prefers unstable
APT policy: (500, 'unstable')
Architecture: i386 (i686)
Shell: /bin/sh linked to /bin/bash
Kernel: Linux 2.6.16
Locale: LANG=C, LC_CTYPE=C (charmap=ANSI_X3.4-1968)
Versions of packages libasound2 depends on:
ii libc6 2.3.6-9GNU C Library: Shared libraries
libasound2 recommends no packages.
-- no debconf information
--
To UNSUBSCRIBE, email to [EMAIL PROTECTED]
with a subject of unsubscribe. Trouble? Contact [EMAIL PROTECTED]
Re: pam_coda module alpha2
... and I suggest you mark it as unsafe for use in any exposed network environment. The unsafeness of relying on Coda tokens for authentication to a service other than Coda (i.e., the local machine) has been discussed previously on the mailing list .. I completely agree with you, and will add a mark on the web-page. nevertheless i will not give up that module ;) because i consider the pam_module as usefull, since I use it (not for authentication but for creating tokens) on a local network, here. robin - Robin Gareus Universitaetsrechenzentrum Im Neuenheimer Feld 293 69120 Heidelberg Email : [EMAIL PROTECTED] Phone : 06221/54 4599 PGP : Public Key at http://www.gareus.de/public.asc Fingerprint : 12 0F DE 6B 58 E4 4B 23 D2 92 6A 23 EA 0B 14 9F -
Re: rdsinit death
my rdsinit is dying when I try to initialize an rvm data segment of 315M. When I tried it again at 22M, it worked fine (trying again w/ 200M now). Any hints? As far as I know Coda keeps a in memory-copy of the complete RVM partition. Correct me if I'm wrong: After rdsinit zeroed the data partition, the rds_zap_heap is used to create a segment/heap stucture on the new RVM partition and thereby also needs a lot of memory. So may it be, that you have too less RAM/Swap for a 315M Partition? Watch a "top" or /proc/meminfo while initialising the RVM and see what happens. And here my question to that : Can i get rid of codasrv wasting that lot of Memory? robin ----- Robin Gareus Universitaetsrechenzentrum Im Neuenheimer Feld 293 69120 Heidelberg Email : [EMAIL PROTECTED] Phone : 06221/54-4599
Coda Login
Hi ! I'm currently working on a Coda/Shadow Login (no PAM) for Linux. (the Workstations/Terminals it will be installed are all SuSE 6.1 - therefore no PAM). For more Information : http://www.rzuser.uni-heidelberg.de/~x42/coda/index.html comments or Information about other Coda-Logins, are welcome... robin - Robin Gareus Universitaetsrechenzentrum Im Neuenheimer Feld 293 69120 Heidelberg Email : [EMAIL PROTECTED] Phone : 06221/54-4599
Re: HOWTO-add a second NIC?
Paul Tader wrote: First NIC installed correctly. I'm adding a second nic (non-PnP, same vendor - SMC). What steps do I follow to configure it. You have to tell your kernel that he (it) should not stop looking for a NIC when already found one. Enable multiple ethernet devices on your machine by adding this to your /etc/lilo.conf, and re-run lilo: append = "ether=0,0,eth1" If you have three interfaces on your bridge, use this line instead: append = "ether=0,0,eth1 ether=0,0,eth2" More interfaces can be found by adding more ether statements. By default a stock Linux kernel probes for a single ethercard, and once one is found the probe ceases. The above append statement tells the kernel to keep probing for more ethernet devices after the first one is found. Alternatively, the boot parameter can be used instead: linux ether=0,0,eth1 the configuration is tha same as with your first NIC. this was taken from the Bridging HOWTO. I think you'll find more information about this in you /usr/doc/howto Folder robin --------- Robin Gareus Universitaetsrechenzentrum Im Neuenheimer Feld 293 69120 Heidelberg Email : [EMAIL PROTECTED] Phone : 06221/54-4599
