Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

3 participants
7520 discussions

by Aaron

Hi all, I am trying to create a mix from a non jack application sbagen with a bunch of mp3's. My son who is ADD has some songs he really loves, which I have mp3s of. I created a script file for sbagen to produce binaural beats but now I would like to create a mix. songs in one track the binaural beats in another. I tried sox and ecasound but sox didn't like being piped from standard out. I couldn't figure out ecasound but I would love to use ardour or rezound if possible. how do I go about feeding the output of sbagen to a recording application (what application should I use for this) I don't need a fancy mix because it is for my special education child and he won't care about quality. I do need a way to bring this about. Thanks Aaron

20 years, 10 months

Re: [linux-audio-dev] realtime-lsm in Andrew Morton's latest kernel patchset

by Luke Yelavich

On Sun, Feb 13, 2005 at 02:00:38PM EST, Cal wrote: > Luke Yelavich wrote: > >Is the patch that is in the mm tree available anywhere? I would like to > >build some test kernels for Slackware but don't want to build mm > >kernels, so the patch would be great in this instance. > > http://www.kernel.org/pub/linux/kernel/people/akpm/patches/2.6/2.6.11-rc3/2… > > Seems to fit well against 2.6.11-rc3-bk9 too. Thanks. -- Luke Yelavich http://www.audioslack.com luke(a)audioslack.com

20 years, 10 months

[linux-audio-dev] realtime-lsm in Andrew Morton's latest kernel patchset

by Jack O'Quin

Those of you who enjoy working with bleeding-edge kernels, please try Andrew's latest 2.6.11-rc3-mm2 patchset. I am hopeful that this may be included in the base kernel, soon. It would help for some of you to build this kernel, run it and report on any problems (or successes) with the realtime LSM. This version no longer supports the `allcaps' option (too much risk), so you need to run `jackd' rather than `jackstart' with it, unless you have a very recent version of JACK. (Even so, using `jackd' is preferable.) You'll need the 2.6.10 sources from <http://kernel.org/>, then the 2.6.11-rc3 patchset, followed by 2.6.11-rc3-mm2 (in that order). Configure the realtime-lsm under `security'. It can either be built-in or loaded as a module (y or m). When the module is active, its parameters can now be changed dynamically via sysfs like this... # echo 1 > /sys/module/realtime/parameters/any # echo 29 > /sys/module/realtime/parameters/gid # echo 1 > /sys/module/realtime/parameters/mlock If you try it, please let me know how well it works. -- joq

20 years, 10 months

[linux-audio-dev] Change midi device order?

by airplays55＠yahoo.com

Is there a way for me to re-order the OSS (not alsa) midi devices that show up from sndstat? The reason is I am using the SB-live! soundcard and am unable to record input in rosegarden21 because it records midi from the first device (#0) and I am having somehow some kind of conflict resulting from 2 midi devices. Sndstat says: MIDI devices: 0: Error: No such device or address /dev/midi00 : Device or resource busy 1: EMU10K1 MIDI I don't know why one soundcard results in 2 drivers but I'd like to disable the bad device or reorder them. This is for knoppix (ALSA isn't an option) and and I have confirmed that midi output works (since playmidi -e works) and that midi input works (by running a short program that echo midi input to screen, using /dev/midi) __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

20 years, 10 months

[linux-audio-dev] Aureal souncard

by Jens M Andreasen

Hi! I Noticed that the Aureal drivers are now in Alsa. I used to be an early adopter (with bad experience ...) Anybody around here care to share some modern succes stories? -- ( ) c[] // Jens M Andreasen

20 years, 10 months

[linux-audio-dev] [ANN] QjackCtl 0.2.15a (fix) released!

by Rui Nuno Capela

Hi, QjackCtl 0.2.15a has been released! This is just a fix release, update highly recommended. As from the change log: - Regression from 0.2.13, of the not so stupid pseudo-mutex guards on the connections management framework, after fixing some crash reports from Fernando Pablo Lopez-Lezcano and Dave Phillips (thanks!); it pays to be such a paranoid after all :). Check it out from the usual place: http://qjackctl.sourceforge.net Enjoy. -- rncbc aka Rui Nuno Capela rncbc(a)rncbc.org

20 years, 10 months

[linux-audio-dev] [OT] How to attend LAC 2005?

by Bengt Gördén

This is sort of really of topic. I'm not a professional developer of music software or something (I do have some ideas though) but I would very much like to attend the conference. The problem is that private I can't afford the trip and stay. Now, I have a nice boss that says that if I could find something in the neighborhood that is close to my normal day job (network engineer) he is willing to pay for the trip as long as he gets something out of the trip. Is there anyone here that knows anything close to networking at LAC 2005 or is there something in Karlsruhe that I could have as a reason for traveling there. regards, /bengan

20 years, 10 months

[linux-audio-dev] about LAC 2005 paper template

by Juhana Sadeharju

I compared the LaTeX paper templates of LAC 2005 and DAFX 2004, and the output screenshots are available at: ftp://ftp.funet.fi/pub/sci/audio/devel/dsp/papertemplates/ It looks to me that DAFX template is better, both in the text quality and the example text. Should we switch? They say DAFX templates are the same what are used in IEEE audio conferences. The text is smaller but I'm sure it is tuned for readability. http://dafx04.na.infn.it/Tmpl_LaTeX.zip http://dafx04.na.infn.it/Tmpl_Word.zip http://www.zkm.de/lac/templates-lac2005.tgz Juhana -- http://music.columbia.edu/mailman/listinfo/linux-graphics-dev for developers of open source graphics software

20 years, 10 months

[linux-audio-dev] Re: [OT] RFA

by Juhana Sadeharju

>> technical operations website: http://techweb.rfa.org/. techweb also [ ... ] >> I see an opportunity here for future use of numerous >> linux audio applications in a working digital audio production facility >> that puts out 36+ hours of programming each day. I would be honored and >> excited if my new boss were to come from the linux audio community. OK. Here is a project suggestion: Remember the old days when Spectrum and Commodore computer software was broadcasted through radio to computer users? This system could be brought to this day using latest tech in coding. This would make it possible to make forbidden webpages available to countries who restrict the access. People would need a simple and cheap radio receiver, and plug it to the audio card. The signal coding would differ from communication coding systems in that sample rate is limited to 44100 Hz and that audio has more signal level resolution. I invented this "Radio Internet" several years ago because Internet costed (and still costs) too much and because the transfer direction is mostly from Internet to home for personal users. The idea of using this tech to broadcast Internet to China became a few months ago after BBC World's Click Online show reported about the Internet retrictions in China. With non-cheap radio tech the transfer rate would not be limited to the audio card properties, but simple and cheap might be better when it comes to the freedom spreading. Juhana -- http://music.columbia.edu/mailman/listinfo/linux-graphics-dev for developers of open source graphics software

20 years, 10 months

[linux-audio-dev] the bandwidth of the each harmonic

by Paul

...a simple, but powerfull paradigm that helps to make simple algorithms that produces very beautiful sounds. Hi. I want to open a topic about what I call "bandwidth of each harmonic", to say some ideas that helped me to make beautifull sounds. I am Paul, the author of ZynAddSubFX software synthsizer ( http://zynaddsubfx.sourceforge.net ) and I wish to share some ideas ;) Referring to musical sounds,usually the harmonics are considered simple sine function. Of course, the reality shows different thing, and so, the harmonics was started to be considered as sine functions modulated by a lower frequencies (I saw some time ago a link that describes that the choir sound is beautifull because of the "micromodulations",etc...). A very good thing is to "have a look" in a frequency domains. Let's take for example a choir that sing same note (like A=440Hz). Because all are human, even they are very well trained singer, they will not sing exactely the same note; for example one will sing at 435 Hz or other, at 443 Hz, and so on. Now, the first harmonic (the fundamental note) is not longer a sine of 440 Hz, but a narrow band signal that has a certain bandwith. For example, let's take a very simple case: if all sing at same loudness, and they sing from 435 Hz and the 445 Hz. In this case, the bandwidth will be 10 Hz. Of course, in real choir the frequency distribution of the harmonic will be not flat, but usually will be a curve that looks like a normal (gaussian) curve(found this after I did some research= very fine frequency analysis with very long FFTs). Now, let's go to the second harmonic: if you multiply 435 and 445 with 2, the difference will be 20 Hz. So, the bandwidth of the second harmonic is 20 Hz. So here a important rule of real instruments (esp. ensembles): the bandwith of each harmonic is proportional to the it's frequency; so in this example the first harmonic has bandwidth 10Hz, the second 20Hz, the third 30Hz, and so on. Here I made a very fine frequency analisys of a synthsized sound(a real orchestra sound will give similar results): [img]http://zynaddsubfx.sourceforge.net/doc/paul1.png[/img] You see, that the harmonic's bandwith increase according to their frequency. If you don't increase the bandwidth of higher harmonics, the resulting sound will be unpleasant especially on higher bandwiths of the first harmonic. So, not all quasi-periodic sounds are good, usually only if you increase the bandwidth of the harmonics. Now, what's happens if there are a lot of harmonics or the pitch of the sound is enough low? Let's see: [img]http://zynaddsubfx.sourceforge.net/doc/paul2.png[/img] The upper harmonics will merge to a single frequency band that will sound like a hiss, that is pleasent to the ear(eg. choir). I found that bandwidth of each harmonics can go even greater than 50 cents (quater of a tone) and still sound musical - I can give you some example wavs. Please note that this a different situation than 2 notes detuned by 50 cents (that sound very dissonand). Also, the best is that the phases are random. In real life this is ensured by the reverberation of the hall, for example if a instrument plays with a vibratto (a flute). Perhaps you noticed that always if more instruments plays the same note(as a enseble), the sound is very pleasant. I consider that this is the cause: the bandwidth of each harmonic. Unfortunately, there is very little on the internet about this stuff, because, I think, that are used more complicated paradigms (like some statistics on how sine harmonics modulates, or so). Now I want to give examples on how you can synthesize sounds, using this ideea. You can use zynaddsubfx or another synthesizer in most cases. Of course: I don't claim that I invented the bandwidth of each harmonic (or frequency distribution of each harmonic), because the ensembles, the choirs, the reverberations, exists since thousands of years (or more :) ). I just consider this a very important fact on how sounds are beautifull and it helped me to synthsize good sounds even since I was on highschool :D. You can make bandwidth of each harmonics by: 1) make more oscillators and detune them a bit (and a slight vibratto helps alot). Most synth use this and allows this, and perhaps it is known by you. This is one of the simplest method. This is implemented in ZynAddSubFX as "ADDsynth" module 2) generate white noise and filter each harmonics with a bandpass filter and mix the results. Be carefull to make higher harmonics to have higher bandwidth. This is implememnted in zynaddsubfx as "SUBsynth" module. 3) You see that above graphs. You can represent them as numbers in the frequency domain that represent the amplitudes of the frequencies, add random phases and do a SINGLE IFFT and voila! A very beautifull sound will "born". This new ideea is implemented in ZynAddSubFX as "PADsynth" 4) you can do other things, like a vibratto on a periodic oscillator and do a FFT to all sound, put random phases, and do a IFFT. For more information, I made few years ago a page that describes above things at: http://zynaddsubfx.sourceforge.net/doc_0.html Good luck. Paul __________________________________ Do you Yahoo!? The all-new My Yahoo! - What will yours do? http://my.yahoo.com

20 years, 10 months

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Linux-audio-dev