Linux-audio-dev

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4 participants
7441 discussions

[linux-audio-dev] Telephone audio and linux options

by Steven Clift

Has anyone developed any tools to capture voice conversations via a voice capable modem on linux? I am interested in using a speakerphone and phone line to webcast a meeting where I don't have a decent Internet connection. Steven Clift Steven Clift - http://publicus.net - Reply to: clift(a)publicus.net Join DoWire: http://dowire.org E-Democracy: http://e-democracy.org

19 years, 8 months

[linux-audio-dev] [ANN] liblo 0.16

by Steve Harris

Liblo, the Lite OSC library, is an implementation of the Open Sound Control [1] protocol for POSIX systems. It is written in ANSI C and released under the GNU General Public Licence. It is designed to make developing OSC applictions as easy as possible. http://plugin.org.uk/liblo/ Changes: Patch from Dave Robillard that fixes bundles of more than 4 messages. Some const char * for C++ compatibility. Added a source field to messages that represents the source from which a message was received. Useful in method handlers to determine which client sent the message. Added patch from Walco van Loon and Pix that fixes a bug in the hostname detection fallback code - Steve [1] http://www.cnmat.berkeley.edu/OpenSoundControl/

19 years, 8 months

[linux-audio-dev] seeking minimal C++ library for ear-training

by Jeffrey Brown

Hi everyone. It's an honor to be read by you. I'm not a professional programmer, but I wrote some text-based C++ programs that quiz me on musical stuff, and was so pleased that I want to do something similar with musical tones. All I'll ever need is a library that will let me playback a sample at a certain set of frequencies -- like 6 at a time (bass tone, high tone, and a four-note chord) -- until keyboard input signals it to stop. In fact, even that's more than I need -- if playing a sample is hard, I'd be perfectly happy listening to square waves. I've looked at some audio libraries (OpenAL, Penguinsound, Sound Object Library) but they're all very intense, designed to do far more than I could even understand, let alone want. Is there a non-threatening, perhaps even easy-to-install, C++ library that will do what I have in mind? (In case it's relevant, I run Mandrake Linux 10.1 Official, KDE, ALSA, and an old Sound Blaster Live. I often go to the San Fernando Valley LUG, in Los Angeles, California.) Thanks in advance, Jeff __________________________________ Do you Yahoo!? All your favorites on one personal page � Try My Yahoo! http://my.yahoo.com

19 years, 8 months

[linux-audio-dev] korg oasys

by Jens M Andreasen

Hi! There is an article on slashdot today about KORGs Linux based music- workstation christened OASYS. The java-script on KORGSs page kills my Mozilla (well, actually the whole Xsession goes bunkers ...) Is this machine somehow related to the prototype HiEnd keyboard Benno introduced a while ago? Price and features seems to be about the same ... -- ( ) c[] // Jens M Andreasen

19 years, 8 months

[linux-audio-dev] [ANN] LACommunity goes gentoo...

by Arnold Krille

Hi all, if you are not a gentoo user stop here, otherwise read on. All the english linux-audio-lists please accept my apologies for crossposting to german gentoo list. [DE: Gentoo-user-de, bitte vergebt mir das Crossposting an die Englischen Listen.] Today I decided to make my little but constant gentoo-portage overlay available for the public. It contains only some apps not in already in portage. Currently available are aeolus-0.3.1 with aeolus-stops-0.1.1, fmit-0.9.[89], museseq-0.7.0, tuneroid-0.9.4 and (not an linux-audio-app) ktechlab-0.1.2. You can access the repository via svn, the address is: http://roederberg.dyndns.org/svn/apps/portage-arnold/ If there is interest, I could also create a tarball, just ask... Feel free to use it and feel free to send patches or to apply for write access. Currently I am trying to follow the anounces on this list, filter away the apps, where gentoo.org is fast enough and add the remaining into my repo. The chances are better if I personally use this app. :-) Thanks for your patience, Arnold PS: [EN: This will be the last time I do such a crossposting. Promised!] [DE: Das ist das letzte mal, das ich so ein Crossposting mache. Versprochen!] -- There is a theory which states that if ever anyone discovers exactly what the Universe is for and why it is here, it will instantly disappear and be replaced by something even more bizarre and inexplicable. There is another theory which states that this has already happened. -- Douglas Adams, The Restaurant at the End of the Universe

19 years, 8 months

[linux-audio-dev] [ANN] QjackCtl 0.2.14 released!

by Rui Nuno Capela

Hi, QjackCtl 0.2.14 has been released. No big features, only a bunch of optimizations and cleanups. Check it out: http://qjackctl.sourceforge.net From the change log: - Put a limit on XRUN callback messages and statistics report rate, preventing the potential hosing of the GUI due to a XRUN cascade storm. The maximum reasonable report rate has been fixed to be one XRUN callback occurrence per second. - Set to ignore the SIGPIPE ("Broken pipe") signal, where available, as the default handler is usually fatal when a JACK client is zombified abruptly. - All conection view items are now sorted in natural case insensitive order, not just as audio port names as was before. - Got rid of those nonsense paranoid and rather stupid pseudo-mutex guards on the connections management framework and event notifications (nuff said :). - Optional confirmation warning on audio server shutdown, if there's some audio clients still active and connected (as suggested by Sampo Savolainen). - Check for <jack/statistics.h> on configure time (as of JACK 0.99.42+ CVS). - "Unlock memory" server setup option was added, allowing the release of memory used by common toolkit libraries (GTK+, Qt, FLTK, Wine) that were being superfluously locked on every GUI JACK client; number of periods has now the minimum allowed value of 2; server start delay widget converted to spinbox; setup dialog layout slighly changed. - Removed stand-alone usx2y driver support. Since JACK 0.99.41+ CVS, the special "rawusb" support on the Tascam US-122/224/428 USB Audio/MIDI interface controllers have been merged and properly integrated into the regular alsa backend driver. Being still experimental, this special mode of operation is now triggered only when "hw:N,2" is specified as the alsa device name (N = soundcard index of snd-usb-usx2y module). - Messages window limit is now enforced only when the line count exceeds in one third the user configured line count maximum; if Qt 3.2.0+ is in use, the QTextView widget is otherwise set to the optimized Qt::LogText format. - XRUN status items are kept double-dashed if none has been detected. Enjoy. -- rncbc aka Rui Nuno Capela rncbc(a)rncbc.org

19 years, 8 months

[linux-audio-dev] linuxsampler cvs from this weeks compile problem

by Julien Claassen

Hi! I just downloaded linuxsampler cvs at the beginning of this week. After copying some standard scripts (ltmain, ltconfig, config.sub...) into the linuxsampler directory I could execute: aclocal, autoconf, autoheader and automake without any problems. When I typed make though I got the following problem: cd . && /bin/sh /home/src/tarballs/new/linuxsampler/missing --run autoheader autoheader: `config.h.in' is unchanged touch ./config.h.in cd . && /bin/sh ./config.status config.h config.status: creating config.h config.status: config.h is unchanged gmake all-recursive gmake[1]: Entering directory `/home/src/tarballs/new/linuxsampler' Making all in src gmake[2]: Entering directory `/home/src/tarballs/new/linuxsampler/src' Making all in network gmake[3]: Entering directory `/home/src/tarballs/new/linuxsampler/src/network' gmake[3]: Nothing to be done for `all'. gmake[3]: Leaving directory `/home/src/tarballs/new/linuxsampler/src/network' Making all in engines gmake[3]: Entering directory `/home/src/tarballs/new/linuxsampler/src/engines' Making all in gig gmake[4]: Entering directory `/home/src/tarballs/new/linuxsampler/src/engines/gig' source='Synthesizer.cpp' object='Synthesizer.lo' libtool=yes \ depfile='.deps/Synthesizer.Plo' tmpdepfile='.deps/Synthesizer.TPlo' \ depmode=gcc3 /bin/sh ../../../depcomp \ /bin/sh ../../../libtool --mode=compile g++ -DHAVE_CONFIG_H -I. -I. -I../../.. -ffast-math -fpermissive -g -O2 -c -o Synthesizer.lo `test -f 'Synthesizer.cpp' || echo './'`Synthesizer.cpp rm -f .libs/Synthesizer.lo g++ -DHAVE_CONFIG_H -I. -I. -I../../.. -ffast-math -fpermissive -g -O2 -c Synthesizer.cpp -MT Synthesizer.lo -MD -MP -MF .deps/Synthesizer.TPlo -fPIC -DPIC ../common/Resampler.h: In static member function `static void LinuxSampler::gig::Synthesizer<IMPLEMENTATION, CHANNELS, USEFILTER, INTERPOLATE, DOLOOP, CONSTPITCH>::Synthesize(sample_t*, void*, float&, float*, float*, uint&, float*, float*, float*, LinuxSampler::gig::Filter&, LinuxSampler::gig::Filter&, LinuxSampler::biquad_param_t&, LinuxSampler::biquad_param_t&) [with implementation_t IMPLEMENTATION = ASM_X86_MMX_SSE, LinuxSampler::gig::channels_t CHANNELS = MONO, bool USEFILTER = false, bool INTERPOLATE = false, bool DOLOOP = false, bool CONSTPITCH = true]': Synthesizer.h:136: instantiated from `static void LinuxSampler::gig::Synthesizer<IMPLEMENTATION, CHANNELS, USEFILTER, INTERPOLATE, DOLOOP, CONSTPITCH>::Synthesize(VOICE_T&, void*, sample_t*, uint&) [with VOICE_T = LinuxSampler::gig::Voice, implementation_t IMPLEMENTATION = ASM_X86_MMX_SSE, LinuxSampler::gig::channels_t CHANNELS = MONO, bool USEFILTER = false, bool INTERPOLATE = false, bool DOLOOP = false, bool CONSTPITCH = true]' Synthesizer.h:110: instantiated from `static void LinuxSampler::gig::Synthesizer<IMPLEMENTATION, CHANNELS, USEFILTER, INTERPOLATE, DOLOOP, CONSTPITCH>::SynthesizeFragment(VOICE_T&, unsigned int, sample_t*, uint&, uint&, unsigned int, unsigned int, unsigned int, uint&, void*, float&, float&) [with VOICE_T = LinuxSampler::gig::Voice, implementation_t IMPLEMENTATION = ASM_X86_MMX_SSE, LinuxSampler::gig::channels_t CHANNELS = MONO, bool USEFILTER = false, bool INTERPOLATE = false, bool DOLOOP = false, bool CONSTPITCH = true]' Synthesizer.h:74: instantiated from `static void LinuxSampler::gig::Synthesizer<IMPLEMENTATION, CHANNELS, USEFILTER, INTERPOLATE, DOLOOP, CONSTPITCH>::SynthesizeFragment(VOICE_T&, unsigned int, sample_t*, unsigned int) [with VOICE_T = LinuxSampler::gig::Voice, implementation_t IMPLEMENTATION = ASM_X86_MMX_SSE, LinuxSampler::gig::channels_t CHANNELS = MONO, bool USEFILTER = false, bool INTERPOLATE = false, bool DOLOOP = false, bool CONSTPITCH = true]' Synthesizer.cpp:152: instantiated from here ../common/Resampler.h:70: can't find a register in class `GENERAL_REGS' while reloading `asm' gmake[4]: *** [Synthesizer.lo] Error 1 gmake[4]: Leaving directory `/home/src/tarballs/new/linuxsampler/src/engines/gig' gmake[3]: *** [all-recursive] Error 1 gmake[3]: Leaving directory `/home/src/tarballs/new/linuxsampler/src/engines' gmake[2]: *** [all-recursive] Error 1 gmake[2]: Leaving directory `/home/src/tarballs/new/linuxsampler/src' gmake[1]: *** [all-recursive] Error 1 gmake[1]: Leaving directory `/home/src/tarballs/new/linuxsampler' gmake: *** [all] Error 2 Here are some facts about my machine: uname -a Linux bach 2.6.10 #2 Wed Jan 12 02:06:11 CET 2005 i686 unknown /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 6 model : 8 model name : Celeron (Coppermine) stepping : 6 cpu MHz : 699.961 cache size : 128 KB fdiv_bug : no hlt_bug : no f00f_bug : no coma_bug : no fpu : yes fpu_exception : yes cpuid level : 2 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 sep mtrr pge mca cmov pat pse36 mmx fxsr sse bogomips : 1376.25 gcc -v Reading specs from /usr/lib/gcc-lib/i486-suse-linux/3.2/specs Configured with: ../configure --enable-threads=posix --prefix=/usr --with-local-prefix=/usr/local --infodir=/usr/share/info --mandir=/usr/share/man --libdir=/usr/lib --enable-languages=c,c++,f77,objc,java,ada --enable-libgcj --with-gxx-include-dir=/usr/include/g++ --with-slibdir=/lib --with-system-zlib --enable-shared --enable-__cxa_atexit i486-suse-linux Thread model: posix gcc version 3.2 Any idea about that? How could I fix it (or is it already fixed?). Kindest regards Julien -------- Music was my first love and it will be my last (John Miles) ======== FIND MY WEB-PROJECT AT: ======== http://ltsb.sourceforge.net - the Linux TextBased Studio guide

19 years, 8 months

[linux-audio-dev] Realtime audio and IO schedulers

by Florian Schmidt

Hi, i wonder if there's a way to grant a process priority over all other processes when it comes to disk i/o.. Somewhat similar like SCHED_FIFO scheduling for cpu priorities. This would be very useful especially for high reliability hard disk recording. I mean it would be simply great to record 24 tracks in ardour and not having to worry about disk underrun [or overrun] when a cronjob kicks in that scans the whole filesystem.. I heard and read a bit about the new io schedulers in the 2.6.x kernels, but i wasn't able to find any infos specifically on this subject.. So anyone knows more? Please let us know! Maybe current io-schedulers actually take the cpu priorities [nice level or relatime priorities] into account when assigning io bandwidth to processes, but if so i wouldn't know where to look to find out.. Thanks, Flo -- Palimm Palimm! http://affenbande.org/~tapas/

19 years, 8 months

Re: [linux-audio-dev] Realtime audio and IO schedulers

by Florian Schmidt

On Sun, 23 Jan 2005 10:44:48 +1100 Cal <cal(a)graggrag.com> wrote: > Indeed it does, so let's not stir it up again unnecessarily! Sorry, i did not see a reply in that thread. Flo -- Palimm Palimm! http://affenbande.org/~tapas/

19 years, 8 months

Re: [linux-audio-dev] Re: Fixing Freeverb and Denormals

by Neil Klepeis

Hi, I compiled the CMT plugins with the g++ options -msse2 and -march=i686 and the freeverb denormals problem (seen with ardour on a dual Xeon machine) seems to be _almost_ entirely fixed. Also, I don't see any weird problems with the ardour host (yet :). But, I still see the DSP load increasing from around 5% to about 15%, although now it is _much_ farther along in the tails of the reverb than before and I get _zero_ xruns. [before, the DSP shot up to 60-90% with tons of xruns and usually the zombification of ardour]. Much cooler. --Neil PS: I also am using the new undenormalize inline function described in an earlier message. On Sat, Jan 22, 2005 at 03:17:28PM +0100, Christian Schoenebeck wrote: >> I think for such an important piece like freeverb it might be worth to add >> architecture dependant assembly optimizations. E.g. with SSE(1) you simply >> have to change the SSE control register to get rid of the denormal problem. >> With that no denormal fixes for individual sample points are necessary >> anymore which definitely pays off in performance. Agreed, but I'm not sure how doing that will affect the host - ie does -mfpumath=SSE (or whatever the option is) always leave the FPU stack in a sane state when control goes back to the host? I dont think it would normally return the FPU to 387 mode, which might do bad things to the host occasionally. OTOH it might be fine, I dont understand the interaction with the 387 and SSE. - Steve

19 years, 8 months

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Linux-audio-dev