Linux-audio-dev

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7516 discussions

by Juhana Sadeharju

Hello. The old OASYS had a hardware card with five DSPs. Korg provided the patch editor for those who asked it separately. The same editor was used to make the preset patches by Korg people, if I remember correctly. It looks like I don't have the module documentation, I have only Patches and Effects doc. Perhaps it came with the patch editor. Anyone? Hey. Nobody wrote details here. Was the given url empty or what? Is it standalone keyboard with Linux inside? Is it an audio card like the old AOSYS but they now provide the patch editor running in Linux? Does it have the patch editor included? Juhana -- http://music.columbia.edu/mailman/listinfo/linux-graphics-dev for developers of open source graphics software

20 years, 9 months

[linux-audio-dev] Lexicon USB reverb

by Steve Harris

http://namm.harmony-central.com/WNAMM05/Content/Lexicon/PR/MX200.html Seems relevant to the linux reverb plugin questions. I would imagine that the plugin sends out a packet of samples over USB, and receives a packet back, maybe with a one packet slip to reduce the plugin processing latency, at the cost of overall latency. I wonder if they'd be willing to release the USB protocol they use. Its possible they compute some of the reverb inside the plugins (eg. the ERs) as a cheap hack to hide USB latency. More likly they dont bother though. If the protocol is simple enough it could be sniffed from a windows box. No info on pricing yet. - Steve

20 years, 9 months

[linux-audio-dev] emu10k1 multichannel support

by Lee Revell

I posted this to alsa-devel but since my previous post on this list generated a lot of interest, I am just reposting it here. As promised, here's an updated patch to add real multichannel playback support (and improved multichannel capture) to the emu10k1 driver. http://www.alsa-project.org/~rlrevell/emu10k1-multichannel-v001.patch Please test it and report any problems. I am especially interested in any regressions that impact regular PCM playback (the hw:0,0 device). QuickStart: $ jackd -R -v -d alsa -P hw:0,3 -C hw:0,2 -S I tested this and it works well with 16in/16out at 128, 256, 512 frames. 32 and 64 should work too but I can't test as I'm running a stock 2.6.10 kernel for now ;-). You can check that the routing is correct by connecting a JACK client to the playback ports corresponding to the FX buses described in Documentation/Audigy-mixer.txt and Documentation/SB-Live-mixer.txt, and verifying that the output appears on that channel (the FX buses are numbered from 0 but JACK numbers clients from 1). For example (from SB-Live-mixer.txt): name='Music Playback Volume',index=0 This control is used to attenuate samples for left and right MIDI FX-bus accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples. The result samples are forwarded to the front DAC PCM slots of the AC97 codec. So "alsaplayer -o jack -d alsa_pcm:playback_5,alsa_pcm:playback_6" should output to FX buses 4 and 5, which you can test by lowering the 'Music' control in alsamixer. With an SBLive, use ports 1 and 2 for the front channels, 3 and 4 for the rear channels. The Audigy uses different channels, see the above docs for more info. In addition to multichannel recording applications, this should also be useful for OpenAL implementations, which are currently restricted to using 21 sources due to the use of an extra voice per stereo PCM. This should allow up to 63 sources. This also adds some new register info including a per channel half loop interrupt that I have discovered by reverse engineering the Windows drivers. Improvements over previous versions: - Routes the 16 channels to the 16 FX buses by default. - Enables the first 16 FX capture outputs by default, required for full duplex operation at latencies lower than 512 frames. - Rewrote the voice allocator to use a more efficient round robin algorithm, eliminating the need to reserve the first 16 voices for the multichannel device. The next free voice is maintained in the card record and the search starts from there. - Use an extra voice for playback timing rather than the EFX capture interrupt. I was only ever able to get that to work at 64 frames. Also there are definite advantages to being able to use the capture and playback devices independently. - Use the newly discovered per-channel half loop interrupt source for the extra voice rather than the channel loop interrupts. For unknown reasons, this works better for multichannel playback, and does not seem to affect regular PCM playback at all. TODO: - Fix the send routing and volume controls for the multichannel device. The current (copy and paste) solution assumes either one or two voices per PCM. So the default settings work fine but changing them with the mixer is likely to have unpredictable effects. - EFX capture should capture output channels 16-32 (mostly unused now) by default, so that we only capture the sources the user has connected to the multichannel recording inputs in the DSP manager. Typically FX buses 0-15 would be connected directly to FX outputs 16-32 so the capture channels would correspond directly to the playback channels. In order for this to work the default DSP configuration has to be changed slightly. Lee

20 years, 9 months

[linux-audio-dev] Still on usx2y

by Antonio "Willy" Malara

So.. exams gone OK this semester, so i'm returning to linux-audio world :) unfortunately my tascam us122 still dont work, and the new jack backend doesn't even compile to me.. dont know what to do.. but, since someone is using this thing right now on linux, i think there's hope even for me :) willy@Zeryn:~$ uname -a Linux Zeryn 2.6.10-mm3 #1 Wed Jan 12 21:01:17 CET 2005 ppc GNU/Linux willy@Zeryn:~$ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.8rc2 (Wed Jan 05 06:44:40 2005 UTC). willy@Zeryn:~$ jackd -d alsa -d hw:0 jackd 0.99.0 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 JACK compiled with System V SHM support loading driver .. creating alsa driver ... hw:0|hw:0|1024|2|48000|0|0|nomon|swmeter|-|32bit control device hw:0 configuring for 48000Hz, period = 1024 frames, buffer = 2 periods Couldn't open hw:0 for 32bit samples trying 24bit instead Couldn't open hw:0 for 24bit samples trying 16bit instead Sorry. The audio interface "hw:0" doesn't support any of the hardware sample formats that JACK's alsa-driver can use. ALSA: cannot configure capture channel cannot load driver module alsa willy@Zeryn:~$ cd Samples/HistoricalBeats willy@Zeryn:~/Samples/HistoricalBeats$ aplay -v -D hw:0 amen.wav Playing WAVE 'amen.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Hardware PCM card 0 'TASCAM US-X2Y' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 16384 period_size : 4096 period_time : 92879 tick_time : 1000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 4096 xfer_align : 4096 start_threshold : 16384 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 willy@Zeryn:~/Samples/HistoricalBeats$ willy@Zeryn:~/Pacchi/jack-0.99.10.0usx2y$ ./configure [...] willy@Zeryn:~/Pacchi/jack-0.99.10.0usx2y$ make [...] make[3]: Entering directory `/home/willy/Pacchi/jack-0.99.10.0usx2y/drivers/usx2y' if /bin/sh ../../libtool --mode=compile --tag=CC gcc -DHAVE_CONFIG_H -I. -I. -I../.. -I../../config -I../.. -I../.. -D_REENTRANT -D_POSIX_PTHREAD_SEMANTICS -Wall -g -O2 -I../../config -I../.. -I../.. -D_REENTRANT -D_POSIX_PTHREAD_SEMANTICS -Wall -g -O2 -MT usx2y_driver.lo -MD -MP -MF ".deps/usx2y_driver.Tpo" -c -o usx2y_driver.lo usx2y_driver.c; \ then mv -f ".deps/usx2y_driver.Tpo" ".deps/usx2y_driver.Plo"; else rm -f ".deps/usx2y_driver.Tpo"; exit 1; fi gcc -DHAVE_CONFIG_H -I. -I. -I../.. -I../../config -I../.. -I../.. -D_REENTRANT -D_POSIX_PTHREAD_SEMANTICS -Wall -g -O2 -I../../config -I../.. -I../.. -D_REENTRANT -D_POSIX_PTHREAD_SEMANTICS -Wall -g -O2 -MT usx2y_driver.lo -MD -MP -MF .deps/usx2y_driver.Tpo -c usx2y_driver.c -fPIC -DPIC -o .libs/usx2y_driver.o usx2y_driver.c: In function `alsa_driver_setup_io_function_pointers': usx2y_driver.c:292: error: `sample_moveswap_dS_s16' undeclared (first use in this function) usx2y_driver.c:292: error: (Each undeclared identifier is reported only once usx2y_driver.c:292: error: for each function it appears in.) usx2y_driver.c:300: error: `sample_moveswap_dS_s24' undeclared (first use in this function) usx2y_driver.c: In function `alsa_driver_configure_stream': usx2y_driver.c:334: error: `SND_PCM_FORMAT_S16BE' undeclared (first use in this function) usx2y_driver.c:334: error: initializer element is not constant usx2y_driver.c:334: error: (near initialization for `formats[3].format') usx2y_driver.c:334: error: initializer element is not constant usx2y_driver.c:334: error: (near initialization for `formats[3]') usx2y_driver.c:335: error: `SND_PCM_FORMAT_S16LE' undeclared (first use in this function) usx2y_driver.c:335: error: initializer element is not constant usx2y_driver.c:335: error: (near initialization for `formats[4].format') usx2y_driver.c:335: error: initializer element is not constant usx2y_driver.c:335: error: (near initialization for `formats[4]') make[3]: *** [usx2y_driver.lo] Error 1 make[3]: Leaving directory `/home/willy/Pacchi/jack-0.99.10.0usx2y/drivers/usx2y' [...] willy@Zeryn:~/Pacchi/jack-0.99.10.0usx2y$ if someone has some clue... :) thanx for the time spent reading this wil --- Il jazz e' lasciare che la luce brilli, lasciarla essere. -- Keith Jarrett

20 years, 9 months

[linux-audio-dev] linuxsampler crashing anyway

by Julien Claassen

Hi! I did try the following things to get current cvs to work. I used the appropriate configuration method and then edited the Synthesizer.cpp and exchanged "ASM_X86_MMX_SSE" with "CPP" as in the other synthesizer_mode functions. The sampler compiled cleanly. But when running it (with OR WITHOUT the --no-tune option) it crashes after a few seconds. It says: Zombified - calling shutdown handler. Then it keeps idling there. I have to use killall -9 linuxsampler to terminate it. CTRL-C doesn't work. Any idea. I use jack 0.99.47 (CVS from sunday. Is there anything I can do to help find this "bug?"? Some debug information, etc... 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

20 years, 9 months

[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

20 years, 9 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/

20 years, 9 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

20 years, 9 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

20 years, 9 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

20 years, 9 months

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Linux-audio-dev