Linux-audio-dev

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2 participants
7534 discussions

[linux-audio-dev] LASH and LASH_Terminal client flag problem

by Florian Schmidt

Hi, in the course of LASH'ifying jack_convolve i stumbled across the LASH_Terminal client flag which specifies to LASH that the client wants to be run in its own terminal when the session is restored. The code for this is in liblash/loader.c:121 The problem i see is: when the client in the term exits (either by means of LASH telling it to, or by sending i.e. a SIGINT), it just drops to a bash prompt instead of exiting the terminal. For reference i have included the code in question below. I also wonder why it is necessary to start another bash anyways? I tried to remove the extra bash call and use xterm -e command_buffer directly, but then the program doesn't even start correctly. Any other thought on this? static void loader_exec_program_in_xterm(int argc, char **argv) { size_t command_size; char *command_buffer; char *xterm_argv[6]; command_size = get_command_size(argc, argv); command_buffer = lash_malloc(command_size); create_command(command_buffer, argc, argv); xterm_argv[0] = "xterm"; xterm_argv[1] = "-e"; xterm_argv[2] = "bash"; xterm_argv[3] = "-c"; xterm_argv[4] = command_buffer; xterm_argv[5] = NULL; /* execute it */ execvp("xterm", xterm_argv); ... Regards, Flo -- Palimm Palimm! http://tapas.affenbande.org

20 years, 2 months

[linux-audio-dev] Linux soundapps pages updated

by Dave Phillips

Greetings: Once again, the inevitable update: http://linuxsound.atnet.at (Europe) http://linuxsound.jp (Japan) http://linux-sound.org (US) No Musings column this time. :( Lots of nice New Additions. :) Best, dp

20 years, 2 months

[linux-audio-dev] Jack and NPTL (again?)

by Jonathan Woithe

Hi everyone I've recently moved a system over to Slackware 10.2 which utilises NPTL and I'm aware of the issues NPTL has raised in the past. Based on a comment on the Jack website though I sort of assumed that things were now in hand and that Jack had a workaround in place for the issue. Despite this I have found that jackd itself (when run using set_rtlimits) gives an error (-11, EAGAIN) when "creating realtime thread". I have already confirmed that set_rtlimits does operate correctly in the NPTL environment; it appears that for some reason pthread_create flatly refuses to create the RT thread even though the appropriate priority limits are set for the process. The LD_ASSUME_KERNEL hack does make jackd work; however, it would be nice to know what's going on (or whether in fact the problems are still being worked on). Are there any ideas? Interestingly enough, jack applications (ardour for example) do NOT need the LD_ASSUME_KERNEL hack - they start and operate fine without out. Only jack itself appears to have issues. Glibc is 2.3.5, getconf reports "NPTL 2.3.5" and kernel is 2.6.14 plus 2.6.14-rt13. 2.6.14-rc3-rt4 has the same issues. Oddly enough, it appears that under 2.6.13 things work fine even for jackd, but I need to confirm that tonight. Jack command line (from memory): jackd -R -d alsa -d hw:0 -n 4 -p 512 Jackd compiled against 2.6.14 headers with gcc 3.3.6. Kernel compiled with 3.3.6 also. Regards jonathan

20 years, 2 months

[linux-audio-dev] /. article and paper

by Lee Revell

Linked from this article: http://linux.slashdot.org/article.pl?sid=05/11/20/2255217 I found this paper from MontaVista. In addition to being a good high level overview of how modern Linux kernels achieve low latency they give special credit to the Linux audio developer community and Paul Davis's open letter to the kernel developers for driving the evolution of Linux 2.6 into a viable soft RT platform. http://linuxdevices.com/articles/AT9697872711.html Lee

20 years, 2 months

[linux-audio-dev] SMDITools and ESI4000

by Dan Richert

hello. I've been trying to get SMDITools working between linux and an E-Mu ESI4000. My sampler is being detected although on the wrong ID (the sampler is set to 4 and it's being detected at 0 ). Whenver I try to receive from or send to the sampler, I'm getting this: *** Error : Out of range *** as the README suggests, i made a symbolic link to /dev/sga from /dev/scsi/host0/bus0/target4/lun4/generic (i've also got sg0 - sg8 in /dev ... could it be looking at those first? ) the only scsi devices i have on the system is the esi4000 and an iomega zip drive between the computer and sampler. iomega drive is set to id 5 (scsi interface is at 7). also, does anybody know of any linux programs that work with the esi4000 file system? it would be great to be able to generate programs/banks. thanks. -d

20 years, 2 months

[linux-audio-dev] Channels and best practice

by Hans Fugal

I'm writing a library in ruby for dealing with audio data, and I'm faced with a design decision. For several reasons, the best thing to use in ruby for numerical data is NArray[1] which is implemented in C for efficiency. So my code is basically a wrapper around NArray which gives some more specific functionality. I want to support multichannel data, and this is where the design decision comes. Most of the time I've seen code that handles multichannel information in an interleaved fashion (each frame is consecutive samples in the array), but I have once or twice seen channels placed end-to-end or in different arrays altogether. It will of course be possible to extract and/or merge channels to deal with libraries (e.g. libsndfile, which I will also be wrapping) or existing code that works one way or the other, but I wonder which should be the internal format to use. What are your thoughts? What is best practice on multichannel audio, or is it always application-specific? For a fluctuating peek (think CVS, although I use darcs) into what I'm doing, check out http://hans.fugal.net/src/ruby-audio 1. http://www.ir.isas.ac.jp/~masa/ruby/index-e.html -- Hans Fugal ; http://hans.fugal.net There's nothing remarkable about it. All one has to do is hit the right keys at the right time and the instrument plays itself. -- Johann Sebastian Bach

20 years, 2 months

[linux-audio-dev] [ANN] QjackCtl 0.2.19 released!

by Rui Nuno Capela

Hi, QjackCtl 0.2.19 has been released. The Qt(cutie)-based GUI control panel for the JACK Audio Connection Kit service has come of age. You can grab it right away from the "official" home site: http://qjackctl.sourceforge.net Release notes as taken straight from the change-log: - Connections widget views are now properly refreshed after renaming client/ports (aliases). - Disabled system tray and ALSA sequencer support on configure time, whenever building for MacOSX as default. - Fixed the major issues with selecting an audio interface on Mac OSX; the button the right of the interface combo is now much better looking than it was before; input/output channel counts are also updated automatically now (thanks to Jesse Chappell for the patch). - Prevent the setting of the coreaudio device id on the jackd command line (-n) whenever the default interface is being selected. - The connections and patchbay windows are now allowed to have a wider connection lines frame panel; splitter width sizes are now persistent across application sessions (thanks to Filipe Tomás for the hint). - Activation toggling feedback on the patchbay widget has been fixed; additionally and as found convenient, the most recently used patchbay definitions can now be loaded immediately by selecting from a drop-down list widget, which replaces the old static patchbay name status text, and adds a lil'icon too :) - All widget captions changed to include proper application title prefix. - Attempt to bring those aging autoconf templates to date; sample SPEC file for RPM build is now being included and generated at configure time. - The current selected device is now shown with a checkmark on the device selection menu(s), while on the settings dialog. - Set to use QApplication::setMainWidget() instead of registering the traditional lastWindowClosed() signal to quit() slot, just to let the -geometry command line argument have some effect on X11. Enjoy, -- rncbc aka Rui Nuno Capela rncbc(a)rncbc.org

20 years, 2 months

[linux-audio-dev] LADSPA, GUIs, OSC and LADSPA_HINT_SAMPLE_RATE

by Luis Garrido

Hi! I am working on a GUI designing tool for LADSPA/DSSI around Qt designer. The idea is to extend the DSSI gui concept to LADSPA (OSC control included) and provide a skinnable tool to manually design GUIs. I plan also to include some user interaction enhancements. You can see some fancy concept diagram here http://flam.sourceforge.net/ There is some code behind it, too, but at the moment its structure is fluctuating and no effort of autotoolization/sconification has been done. There is a working skinnable slider and flamwizard is half finished. The skinnability will by provided by means of QStyle plugins. You can have very minimalistic depictions of the UI gadgets or hardware panel simulations. The GUI viewer, called flammer, will allow for loading/saving presets and load alternate GUI layouts for a given plugin at runtime. Now I have a doubt about LADSPA_HINT_SAMPLE_RATE. To effectively incorporate this hinting in the GUI, the host must have some means of letting know the GUI what the current sample rate is. Is there any standard procedure for achieving this? Comments, ideas, please? Cheers, Luis

20 years, 2 months

[linux-audio-dev] alsa help

by Aaron

Hi all, Denemo is really lacking alsa support. Unfortunatly Denemo has only one coder who is working a full time job. Is there a kind soul out there who could help implement ALSA support for Denemo. And if you live nearby I would gladly buy you a beer of your choice :) thats if you drink beer... Thanks Aaron

20 years, 2 months

[linux-audio-dev] the Linux-Powered Korg OASYS

by Jan Weil

<http://digitalmedia.oreilly.com/2005/11/09/inside-the-korg-oasys.html> Cheers Jan

20 years, 2 months

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Linux-audio-dev