Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

2 participants
7501 discussions

[linux-audio-dev] jackd fails to start: cannot create shm segment /jack-shm-registry (No such file or directory)

by Ivica Bukvic

Hi all, any ideas on this one? I have 2 soundcards configured: sblive rme 9652 my modules.conf looks like this: #install rme9652 Hammerfall buffer first before memory gets fragmented pre-install snd-hammerfall-mem alias eth0 3c59x probeall usb-interface usb-uhci alias /dev/nvidia* nvidia # ALSA native device support alias char-major-116 snd options snd major=116 cards_limit=3 alias snd-card-0 snd-emu10k1 alias snd-card-1 snd-rme9652 alias snd-card-2 snd-usb-audio # OSS/Free compatibility # SBLIVE (Card 0) alias char-major-14 soundcore alias sound-slot-0 snd-0 alias sound-service-0-0 snd-mixer-oss alias sound-service-0-1 snd-seq-oss alias sound-service-0-3 snd-pcm-oss alias sound-service-0-8 snd-seq-oss alias sound-service-0-12 snd-pcm-oss #We'll use Hammerfall (Card 1) only as an ALSA device # USB-MIDI (Card 2) alias sound-slot-2 snd-2 alias sound-service-2-1 snd-seq-oss alias sound-service-2-8 snd-seq-oss lsmod reflects that everything is loaded ok: Module Size Used by Tainted: P isofs 28244 1 (autoclean) zlib_inflate 21156 0 (autoclean) [isofs] nls_iso8859-1 3516 1 (autoclean) snd-pcm-oss 42468 0 (autoclean) snd-mixer-oss 14296 0 (autoclean) [snd-pcm-oss] snd-seq-oss 30016 0 (autoclean) (unused) snd-seq-midi 4576 0 (autoclean) (unused) udf 92128 0 (autoclean) sr_mod 16952 2 (autoclean) floppy 56028 0 (autoclean) snd-emu10k1-synth 5116 0 (unused) snd-emux-synth 32092 0 [snd-emu10k1-synth] snd-seq-midi-emul 6652 0 [snd-emux-synth] snd-seq-virmidi 4556 0 [snd-emux-synth] snd-seq-midi-event 5800 0 [snd-seq-oss snd-seq-midi snd-seq-virmidi] snd-seq 44560 2 [snd-seq-oss snd-seq-midi snd-emux-synth snd-seq-midi-emul snd-seq-virmidi snd-seq-midi-event] snd-usb-audio 47744 0 (unused) snd-rme9652 21164 0 snd-hammerfall-mem 2848 0 [snd-rme9652] snd-emu10k1 71348 0 [snd-emu10k1-synth] snd-pcm 79680 0 [snd-pcm-oss snd-usb-audio snd-rme9652 snd-emu10k1] snd-timer 19272 0 [snd-seq snd-pcm] snd-rawmidi 18304 0 [snd-seq-midi snd-seq-virmidi snd-usb-audio snd-emu10k1] snd-page-alloc 7960 0 [snd-emu10k1 snd-pcm] snd-util-mem 2984 0 [snd-emux-synth snd-emu10k1] snd-ac97-codec 40416 0 [snd-emu10k1] snd-seq-device 5740 0 [snd-seq-oss snd-seq-midi snd-emu10k1-synth snd-emux-synth snd-seq snd-emu10k1 snd-rawmidi] snd-hwdep 6272 0 [snd-emu10k1] snd 41476 0 [snd-pcm-oss snd-mixer-oss snd-seq-oss snd-seq-midi snd-emux-synth snd-seq-virmidi snd-seq-midi-event snd-seq snd-usb-audio snd-rme9652 snd-emu10k1 snd-pcm snd-timer snd-rawmidi snd-util-mem snd-ac97-codec snd-seq-device snd-hwdep] parport_pc 25320 1 (autoclean) lp 8224 0 (autoclean) parport 35232 1 (autoclean) [parport_pc lp] agpgart 41248 3 (autoclean) nvidia 1593216 10 (autoclean) nfsd 74992 8 (autoclean) af_packet 16872 1 (autoclean) 3c59x 29936 1 (autoclean) supermount 74908 2 (autoclean) ide-scsi 11440 1 scsi_mod 105556 2 [sr_mod ide-scsi] ide-cd 33952 0 cdrom 31712 0 [sr_mod ide-cd] usb-midi 19256 0 (unused) soundcore 6500 0 [snd usb-midi] usbmouse 2936 0 (unused) keybdev 2720 0 (unused) mousedev 5332 1 hid 21028 0 (unused) input 5760 0 [usbmouse keybdev mousedev hid] usb-uhci 25932 0 (unused) usbcore 75616 1 [snd-usb-audio usb-midi usbmouse hid usb-uhci] rtc 8892 0 (autoclean) ext3 62252 3 jbd 41500 3 [ext3] my /etc/asound.conf looks like this: pcm.sblive { type hw card 0 } ctl.sblive { type hw card 0 } pcm.rme9652 { type hw card 1 } ctl.rme9652 { type hw card 1 } When I do: aplay -D sblive <filename> it works just fine. Yet, when I do jackd -d alsa -d hw:0 or jackd -d alsa -d sblive or simply jackd -d alsa I get the following _every time_: jackd 0.68.0 Copyright 2001-2003 Paul Davis and others. jackd comes with ABSOLUTELY NO WARRANTY This is free software, and you are welcome to redistribute it under certain conditions; see the file COPYING for details key for /tmp/jack/shm/jack-shm-registry is 0x6a06252b cannot create shm segment /jack-shm-registry (No such file or directory) cannot create engine jack main caught signal 15 I am using MDK 9.1 with low-lat kernel (otherwise vanilla), and the CVS jack (4/17/03). The rpm for MDK (0.66) also has the same issue. Any help on this one is greatly appreciated! Thank you! Sincerely, Ivica Ico Bukvic, composer & multimedia sculptor http://meowing.ccm.uc.edu/~ico

22 years, 1 month

[linux-audio-dev] SoundFontCombi v0.005

by holborn

Hi! Sorry for the english .. :-) SoundFontCombi takes the advantages of ALSA sequencer and the wavetable devices for emulate a synthetizer. I'm not programmer, this program dont have project page, CVS, or mailing list, is in proof of concept rigth now. And only the binaries are available. Is my first program in C++, my first graphic program for *nix, and belive me the source code needs a "making up" rigth now. Maybe next month when i learned how to do a Makefile :-) SoundFontCombi was tested in Debian sid and Gentoo. Requires FLTK v1.x.x (tested on v1.1.1) and ALSA with sequencer. The binaries are available in : http://personal.telefonica.terra.es/web/soudfontcombi/sfc-v0.005.tar.gz Any commets will be apreciated. Josep

22 years, 1 month

[linux-audio-dev] How can you make Jack always available?

by Nick Tsocanos

Hello, How can I make Jackd always run? Right now I have to set it up in a console window. I'd like to learn how to run it always at startup and not have to start/stop it when I need it. Also how do you use Jack in real-time mode? In order to run it I have to be root, but if I run it as root, nobody else can see Jack unless they are also root. I wish somebody would write a Jack-HOWTO so dummies like me can figure out how to run this thing. Thanks

22 years, 1 month

[linux-audio-dev] Found out what was wrong with my LADSPA manager programmer

by Nick Tsocanos

More like a golden buddha child but hey... I got them all working, it was problemos in my plugin manager code. The plugins were fine. Are you itching to know what the problem was? I'll tell you. There is the mysterious number called NAN. I found out where NAN was coming from. I was calculating the log(number) to put in the control value The culprit number=>0. Put 0 into log, and it makes NAN. Put NAN into plugin dry/wet mix control, and plugin makes zillions of NANs. And all those NANs will make your computer run slower than an Atari 800. What's even worse is trying to convert NAN to an integer and play it on a sound card.. I guess between the massive buffer underruns and trying to synthesize all NANs was why I was hearing the most horrible noises I ever heard in my life. I have no idea what NAN becomes when you cast it to an int. Some demon posessed number I suppose. It doesn't sound like white noise either. If you want to hear the most hideous noise try it. Put log(0) into a control volume and watch what happens. Try it with about 15 plugins at once. It is super loud too. I am sure it will destroy your speakers too. Don't try it with expensive speakers. Now which controls do it? A good many of them. It's not really a design defect of the plugin programmer. I am sure, they don't expect anyone but an idiot to put NAN into the control value. I didn't do it on purpose! It was a mistake! A simple error on my part. I hope somebody else has done this by accident too.. it'll make me feel better...

22 years, 1 month

[linux-audio-dev] LADSPA Hosting

by Nick Tsocanos

I got my app to host plugins, but I am having problems figuring out how to filter plugins out that won't work in my app. I only need F/X processors. If a plugin does not process an audio stream, or is used only for synthesis, or has peculiar controls that are not defined in the hints, than I want to ignore them. Is it possible? How can I find out how a plugin is used? I am reading the API header but it's not entirely clear what properties determine how a plugin is used or how I can tell if a plugin won't work for my app. Thanks

22 years, 1 month

[linux-audio-dev] Re: a little C++ guidance needed

by Andrew Burgess

>>On Tue, Apr 15, 2003 at 10:22:40AM -0400, Dave Phillips wrote: >>> Nick's advice to upgrade my compiler is timely (I'm using GCC 2.96 from >>> RH 7.2) but unfortunately I can't make the switch right now. If there's >>> an obvious (or non-obvious) solution to my dilemma I'd be happy to hear >>> of it. The redhat rpms work fine side by side. gcc3 is the default (/usr/bin/gcc). To use 2.96, you have to change your Makefiles to CC=gcc296, or set the env var and then just make. I've had no problem with executables from either compiler finding the correct runtime libs. The rpms are: compat-gcc-7.3-2.96.110 gcc-3.2-7 They are from the redhat 8.0 distribution but my system is a mismash, 7.0 with many upgrades. HTH

22 years, 1 month

[linux-audio-dev] aseq -> smf or audio

by Hans Fugal

Hi all, I am doing a modest partial MWPP implementation for a networking class and I want to demo it for the professor and class, but am not keen on carting my midi keyboard around campus if I can help it. Realtime would of course be nice, but a pre-recorded sound file would be ok too. I thought timidity++ would be a reasonable way to do this, but unfortunately timidity sounds _awful_ when used as an ALSA sequencer client, even when I'm feeding it with pmidi (i.e. I've ruled out network jitter). It's very jittery. pmidi or my own program going to the hardware midi port has no such jitter. Is there a setting to make timidity behave better? I've tried a few things (like -B 0) but to no avail. What's more, I can't get timidity to output to a file with the -iA switch even if I do get the jitter down. Alternatively, and probably more cool, I could create an SMF and play it with timidity or playmidi or make a sound file with csound or something. I like the csound option. :) Is there something out there that will take midi from the ALSA sequencer and create an SMF? Thanks, Hans -- Hans Fugal | De gustibus non disputandum est. http://hans.fugal.net/ | Debian, vim, mutt, ruby, text, gpg http://gdmxml.fugal.net/ | WindowMaker, gaim, UTF-8, RISC, JS Bach --------------------------------------------------------------------- GnuPG Fingerprint: 6940 87C5 6610 567F 1E95 CB5E FC98 E8CD E0AA D460

22 years, 1 month

Re: [linux-audio-dev] aseq -> smf or audio

by John Lazzaro

> Hans Fugal <hans(a)fugal.net> > > I am doing a modest partial MWPP implementation for a networking class > and I want to demo it for the professor and class, but am not keen on > carting my midi keyboard around campus if I can help it. [...] Sfront has a pre-IETF-era version of MWPP, and a bunch of demos that ship to show it off (including a 2-person interactive session demo). These are all under sfront/examples/rtime ... mirror, nmp_audio, nmp_null, and nmp_stream. The demos use -cin ascii for real-time input via the ASCII keyboard, and nmp_stream streams an SMF out. The trick here will be that, as written, the MWPP network drivers look to a SIP server at Berkeley to do the session setup, which has random non-standard hacks in it to do NAT breaking and MD5-based authentication -- the code is a few years old, and pre-dates the IETF standards to do those things. So, you'll either need to hack sfront networking to match your implementation, or hack your implementation to work with the Berkeley SIP server (the former is probably much easier ...). Let me know if you want to go this route and I can offer advice ... --jl ------------------------------------------------------------------------- John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley lazzaro [at] cs [dot] berkeley [dot] edu www.cs.berkeley.edu/~lazzaro -------------------------------------------------------------------------

22 years, 1 month

Re: [linux-audio-dev] Gibson magic protocol

by John Lazzaro

On Tue, Apr 15, 2003 at 06:40:23 +0100, Martijn Sipkema wrote: > It will _not_ coexist with standard Ethernet and I doubt a NIC can be > used because the timing is very critical. You'd probably have to buy a > dedicated MaGIC interface. See: http://www.ietf.org/mail-archive/working-groups/avt/current/msg00575.html That whole thread, and an related one Dominique Fober started: http://www1.ietf.org/mail-archive/working-groups/avt/current/msg01605.html Are both good reads, some highlights: http://www1.ietf.org/mail-archive/working-groups/avt/current/msg01619.html http://www.ietf.org/mail-archive/working-groups/avt/current/msg00578.html ------------------------------------------------------------------------- John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley lazzaro [at] cs [dot] berkeley [dot] edu www.cs.berkeley.edu/~lazzaro -------------------------------------------------------------------------

22 years, 2 months

[linux-audio-dev] Gibson magic protocol

by Ben Loftis

Has anybody investigated the Gibson MaGIC protocol for use on Linux? It looks like a perfect match: lots of audio channels and control data over CAT5. Just buy a separate network card, port their protocol to Linux, and then you could use their 16-in 16-out box, or get input directly from one of their new guitars with the RJ45 port on it. The The only scary bit is that the license is free "for the next10 years" only. -Ben

22 years, 2 months

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

Linux-audio-dev