Linux-audio-dev

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7441 discussions

[linux-audio-dev] Dumb question regarding CMT's ambisonic plugins

by Ivica Ico Bukvic

Hi all, I've posted this one before on the LAU list but unfortunately I've not received any response. Please accept my apologies for cross-posting, but I am hoping that someone on LAD might be able to help me learn more on this particular topic. I would greatly appreciate help in this matter! At any rate, here's the question: I have been messing with the ambisonic plugins provided in the CMT ladspa collection. I am namely looking into a simple implementation of B-format 1ch->4ch encoder. The problem is that I am failing to figure out how to use this on a simple mono (non-encoded) sound input in order to control its diffusion over 4 channels. When running it via Ardour and sending it to 4 separate busses via this plugin, even though the x, y, and z coordinates do affect the output of the 4 channels, they do not conform to the expected 3,4,2,1 output (I presume these correspond to 4 speaker outputs, but I am also having doubts about this as well, so your help in understanding this better would be most appreciated). By now you can see that although I know a bit about ambisonics (namely theory), that I am quite a newb in this area, but for what it's worth I am eager to learn :-). The pd set of plugins pretty much work as expected but they are also designed differently, likely with discrete 4-8 output that can be then patched directly to main outs. In the case of ladspa plugins I am not even sure whether what I am getting is encoded version of a 4-channel stream that then needs to be decoded. Yet, when I tried having: mono sound-> b-format 1ch to 4ch encoder (LADSPA)->4ch to 4ch decoder (LADSPA) -> out That gave me bunch of garbage with two out of 4 channels consistently clipping, so this means that this is probably not the case either... So, to conclude I must be not using the CMT plugins properly and would therefore greatly appreciate it if someone would enlighten me. I am already aware of the theory behind the ambisonics, it's just that I am a bit puzzled how the CMT plugins (if at all) could be used with mono and/or stereo non-encoded streams in order to spatialize them in real-time. I would most appreciate a simple practical example as to how to do this (if such proves to be possible). Thank you very much! Best wishes, Ico

19 years, 4 months

[linux-audio-dev] ALSA Sequencer API - How to Receive Events?

by Sean Don

Hi all. I'd like to receive Midi note events from another Midi Player. But before I attempt to dive into a wad of documentation, I'd like to know if it's even possible under the following situation... 1) I don't have a hardware sequencer; so, I use timidity as an ALSA sequencer, then pmidi a player... timidity -iA -B8,8 -Os & pmidi -p 128:0 <midi file> 2) That plays fine, but how to tap into the events pmidi is sending to 128:0? I've tried modifying sample code such as dump-alsa.c, but no output. Also tried tips at "ALSA related stuffs", but was unable to get "aseqview" to compile. Any tips? Thank you in advance. - Sean http://www.home.unix-ag.org/simon/files/dump-alsa.c http://mitglied.lycos.de/iwai/alsa.html __________________________________ Do you Yahoo!? Yahoo! Small Business - Try our new Resources site http://smallbusiness.yahoo.com/resources/

19 years, 4 months

[linux-audio-dev] ANN: Oscilloscope plugin 0.2.0

by Lars Luthman

Here is an oscilloscope DSSI plugin that I've been hacking on the last few weeks: http://www.student.nada.kth.se/~d00-llu/music_dssi.php?lang=en It has two audio input ports and will display the two input signals as two waves in the display. The trigger level and direction is controllable, as well as the amplification and offset for each channel and the time resolution. Note that this plugin will NOT work with jack-dssi-host from version 0.9 of the DSSI package since that does not support audio input ports. It will work with jack-dssi-host or Om from current CVS (I haven't tested it with Rosegarden). Om 0.1.1 and earlier will not work because of an optimisation that breaks this type of plugins. The plugin includes a DSSIUIClient class that might be of interest to plugin UI writers using libsigc++ and Glibmm - it handles all the required communication with the plugin host and exposes public member functions and signals for the UI -> host and host -> UI OSC commands specified in the DSSI RFC. It can also handle the nasty business of setting up an initial shared memory segment for the plugin and the UI (the actual implementation of the shared memory handling is in a separate C file, so it could be useful for people who don't use libsigc ++ and Glibmm or even C++). Source code documentation is here http://www.student.nada.kth.se/~d00-llu/plugins/ll-scope/dox/html/classDSSI… and here http://www.student.nada.kth.se/~d00-llu/plugins/ll-scope/dox/html/dssi__shm… For more details about the plugin, read the README file in the source package. It's all GPL. -- Lars Luthman PGP key: http://www.d.kth.se/~d00-llu/pgp_key.php Fingerprint: FCA7 C790 19B9 322D EB7A E1B3 4371 4650 04C7 7E2E

19 years, 4 months

[linux-audio-dev] Re: Best-performing Linux-friendly MIDI interfaces?

by Kjetil Svalastog Matheussen

Paul Davis: > > >what is the 'easiest' MIDI interface to get working under the average > >Linux kernel? what sort of experiences do folks have with getting > >MIDI working (on a programming level) using API's such as (but not > >limited to) ALSA, and MidiShare? > > > >for me so far, MidiShare seems to offer the most direct and usable > >approach .. while ALSA is wraught with complexity and dependencies > >which often seem out of control. i'd like to know what your > >experience, as developers, has been with getting a working MIDI > >subsystem under Linux ... > > i can speak up as someone who has avoided any interactions with the > ALSA sequencer for years because of a similar perception. i recently > converted Tim Thompson's KeyKit from its raw MIDI port-based > implementation under Linux to one that uses the ALSA sequencer (and > adding multiport support along the way). it was suprisingly easy and > obvious, given a simple example (in this case, the "ALSA sequencer > MIDI Port object" that someone contributed to Ardour). even more > gratifying was comparing the code to the keykit implementations for > win32 and CoreMidi. the ALSA sequencer one was smaller, more logical > and less bogged down in details. > Yepp, same experience as you. Before using the alsa seq api myself, it seemed to be horrible complicated. But it really wasn't. Tip: the source for jack-rack by bob ham has some clean alsa-seq code to look at. --

19 years, 4 months

[linux-audio-dev] sblive hints

by Aaron

Hi all, I have a soundblaster live. I also am using debian/demudi and have a dual PIII 500 mhz. I am looking for hints on how to optimize my card with alsa and jack etc. I once had found on the alsa site example of .asoundrc tailored for the sblive but now on searching I don't see such an animal. I want to use skype or a voice over ip in additon to audio processing. I find that with my setup I get clicks where I can't hear the other person. I am doing some mixes where I need as much umph from my system as possible and especially my sound card. I see that when I check my irq's my sound card is: theone:/home/aamehl# cat /proc/interrupts CPU0 CPU1 0: 190504469 36529323 IO-APIC-edge timer 0/33792 2: 0 0 XT-PIC cascade 0/0 7: 0 0 IO-APIC-edge parport0 0/0 14: 852744 235050 IO-APIC-edge ide0 0/87773 15: 14 0 IO-APIC-edge ide1 1/12 145: 15477006 1925905 IO-APIC-level aic7xxx, aic7xxx, nvidia 0/2911 153: 15069558 1527074 IO-APIC-level EMU10K1 0/96632 161: 3021752 703622 IO-APIC-level uhci_hcd, eth0 0/25374 NMI: 0 0 LOC: 227042766 227042753 ERR: 0 MIS: 0 153 and most of the other devices are not addon cards but part of the mother board. I can't move my video card and the ethernet is onboard. Is all lost or can I give more priority to the sblive? Any other suggestions are welcome as far as squezzing performance out of my 128megs rams. Thanks Aaron

19 years, 4 months

[linux-audio-dev] ANN: "0th Sound" contemporary multimedia art concert

by Ivica Ico Bukvic

Greetings all, Apologies for cross-posting. Please allow me to use this opportunity to bring to your attention an upcoming contemporary multimedia art concert titled "0th Sound." The evening-long event will take place in the Cincinnati area on May 29th (Sunday) 2005 at 8pm, and will feature a portfolio of my latest works, including a number of compositions created exclusively using GNU/Linux tools. For more info, sound clips and other goodies, please visit: http://meowing.ccm.uc.edu/~ico For promotional and press-release materials please visit: http://meowing.ccm.uc.edu/cgi-bin/ico/yabb/YaBB.cgi?board=News_id;action=dis play;num=1116212106;start=0 Many thanks! Best wishes, Ivica Ico Bukvic, composer & multimedia sculptor http://meowing.ccm.uc.edu/~ico/

19 years, 4 months

[linux-audio-dev] references

by Kevin Sookocheff

Hello everyone, I'm looking to lend a hand at an audio project. I'm not a new programmer but I'm new to audio programming. Is there any good references out there to help me get up to speed? How about any essential libraries, system code that I should learn? Thanks a bunch, Kevin

19 years, 4 months

[linux-audio-dev] The HDSP DJ Interface (RPM)

by Neil Klepeis

Has anyone had success using the latest RME HDSP interface for DJ's ("RPM")? http://www.rme-audio.com/english/hdsp/hdsprpm.htm --Neil

19 years, 4 months

[linux-audio-dev] Beginner question about ring buffer and block processing

by David Cournapeau

Hi folks, I try to get my head around a basic problem, and I don't manage to get around it: how to adapt to variable sized blocks of samples given by a ladspa or jack host/server to an existing, fixed length block processing algorithm, for example a N points fft. AFAIK, the ring buffer is the standart tool to solve this problem, but I don't understand how this helps ? I tried to read the source of jamin, which implements this kind of scheme in io.c, but I couldn't manage to get the whole thing. I would be glad to hear any hint about the right way to use ringbuffer for this task, cheers, David

19 years, 4 months

[linux-audio-dev] Re: Minimum reasonable latency

by Juhana Sadeharju

>From: fons adriaensen <fons.adriaensen(a)skynet.be> > >On Thu, May 19, 2005 at 08:57:24PM +0300, Juhana Sadeharju wrote: > >> Yes. The priority should depend on the buffer size. >> Longer buffers would run with non-soft-RT schdulers. > >That problem is really the same as what you get with partitioned >convolution. That was the problem which did lead me to develop the system a few year ago. Any hardware can be used in the system for lowest latencies. (hw -- RT Linux or soft-RT -- user scheduling) Juhana -- http://music.columbia.edu/mailman/listinfo/linux-graphics-dev for developers of open source graphics software

19 years, 4 months

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