Linux-audio-dev

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7513 discussions

Re: [LAD] [LAU] New modular synthesis library w/ ruby bindings

by Harry Van Haaren

On Mon, Jul 19, 2010 at 8:06 PM, Bernardo Barros <bernardobarros2(a)gmail.com>wrote: > At least for me it would be easier to think as c = 0 > > c = 0 > c' = 12 > c, = -12 > c'' = 24 > c,, = -24 > > because most of the notes in a regular score is mostly like to happen > in the middle and things looks simpler that way. UNLESS you're already > familiar with all kind of notes with MIDI number notation. > Hey, Yes I was referring to the MIDI standard there.. Should have mentioned that.. There's no "rule" telling you you should do it differently, I'm expressing my opinion that for me it would seem more logical to have middle C at 60. -Harry

15 years, 2 months

Re: [LAD] [LAU] Free Software Support Re: Reverb to use with LinuxSampler/piano?

by drew Roberts

On Monday 19 July 2010 15:13:54 Bernardo Barros wrote: > Yes, but a lot of stuff is not presently appealing to industry. > Like... ,music and multimedia stuff: LilyPond, SoundCard drivers, > InkScape, GIMP, Ardour, Qtractor and so on. But yes, even though I'm a > socialist I agree with you that big corporations would benefit with > Free Software a lot. I wouldn't be too sure about that. You might just need to think of which industry. And A lot of the groups I mentioned are normally small businesses around here. There are a decent number of one man operations wrt loclsmithing. Barber shops go from one barber to several but they are not big corporations by any means. The associations may be bigger, but if the members wanted the software written for them to be Free, they could use their association to coordinate their efforts. > > 2010/7/19 drew Roberts <zotz(a)100jamz.com>: > > On Monday 19 July 2010 12:12:54 you wrote: > >> Maybe they always overlap somehow, software needs support. Really good > >> and complex software need more support than one person can do just > >> with his free time. I am not sure from where this support could come > >> from. > > > > I keep thinking that industry groups would be wise to support the > > development of Free software for their industries. > > > > For instance: > > > > Locksmiths > > http://www.aloa.org/ > > http://www.uklocksmithsassociation.co.uk/ > > http://www.masterlocksmiths.com.au/home.aspx > > > > Barbers, Beauty Salons, Furniture Stores, Ocean Engineers, Corner Grocery > > Stores, what have you. > > > > all the best, > > > > drew > > > >> From universities, donations, companies, governments? The Linux > >> kernel, for example, is very complex and needs a lot of work all the > >> time since new hardware and software need it , and it is well > >> supported by really big companies like google, novell and red har. > >> Other areas this is not so easy to get support from companies, so I > >> now just can think of the other options: donations from people that > >> make money or work seriouly with the software, universities and > >> government programmes. The last one seems unlikely to happen but here > >> in Brazil the present government is officially supporting Linux and > >> some software projects. Donations from professionals and Universities > >> can help Free Software a lot too. > >> > >> > >> 2010/7/19 drew Roberts <zotz(a)100jamz.com> > >> > >> > On Sunday 18 July 2010 11:36:42 you wrote: > >> > > On Sun, Jul 18, 2010 at 10:04 AM, drew Roberts <zotz(a)100jamz.com> wrote: > >> > > > Commercial and Free are not opposites. > >> > > > >> > > no, just siblings who fight with each other like crazy for 18 years > >> > > before developing a deep and abiding love for each other :) > >> > > >> > Free and Commercial can intersect / overlap. Venn wise. > > > > _______________________________________________ > > Linux-audio-user mailing list > > Linux-audio-user(a)lists.linuxaudio.org > > http://lists.linuxaudio.org/listinfo/linux-audio-user

15 years, 2 months

Re: [LAD] May I asked something OT?

by Sakari Bergen

On July 18, 2010 03:57:06 pm Ralf Mardorf wrote: > A lot of kids wish to have a kill switch for their guitars. > A kill switch is a short circuit, to 'stop' the audio signal. > > I'm not fine with this solution, but the kids argue, that e.g. an > interruption does cause unwanted noise, especially for over drive > sounds. IMO even using opto-electronics won't solve the issue, because > the noise of the transistor overdrive effect still would be hearable, > while for a short circuit there is silence. > > Has anybody an idea to solve this without a short circuit? > > I'm really not a fan of short circuits. Note, it's not possible to do an > interrupt all the times behind the latest noise generator and even an > interrupt could cause noise itself, while a short circuit indeed is a > good way to cancel sound. If you don't like short circuiting the mics, just switch the output jack hot lead between ground and the volume pot(s). Connecting the output to ground is the same as turning the volume down. There is no need to short the mics... On 19.07.2010 08:30, Tim E. Real wrote: > If you play a Gibson you can set the neck pickup volume to zero and > the bridge pickup volume to full and then toggle the pickup switch, > rapidly if desired, like Eddie van Halen on You Really Got Me. > Tim. Tom Morello gets real creative with his guirar, and uses this technique alot. He also unplugs his guitar to make the pedals and amps make noise (oscillate), and controls the noise with his wah pedal and by touching the tip of the plug to the guitar bridge (which is grounded). Good demonstration of how just cutting the signal lead can cause lots of noise, while a short will be more or less silent :) -Sakari-

15 years, 2 months

New modular synthesis library w/ ruby bindings

by Evan Buswell

Hi all, I've just released a beta of a modular synthesis library using Jack2, with a Ruby module. See http://ebuswell.github.com/Cshellsynth/ There's no mailing list for the project yet, but feel free to email me directly if you have questions/comments/concerns. Enjoy! Evan

15 years, 2 months

[Somewhat OT] Strange failure mode of a PC

by fons＠kokkinizita.net

Hello all, Early this week one of the three 'rendering' PCs of the WFS system in the Sala Bianca failed. It just appeared completely dead and didn't even try to boot when the power button was pressed, but the standby power (for the network interface) was available. I suspected the power supply, so ordered a new one which arrived two days ago. Installed it and things worked again. But now comes the interesting part. While installing the new PS, I also disconnected the wires to the power button, and to test I just used a screwdriver to short the two pins that normally connect to it. But when I reconnected the power button the PC switched off after a few seconds. So it seemed as if the power button was permanently being pressed. I again installed the old PS, and things worked as long as the power button was not connected. Measuring the power button switch with a multimeter showed an unstable resistance value of between 1 and 3k while it was not pressed. So I removed the thing, which turned out to be a cheap miniature switch, a little cube of around 8 mm size. I opened and disassembled it, and noticed that the contacts had some black dirt on them. Cleaned with aceton and reassembled, and things worked perfectly again. What I don't understand is how the contacts got so dirty. If a resistance of a few kOhm is enough to make it look as a closed contact then it can't be handling large currents, so there should not be any arcing. And the construction of the thing is such that it is virtually closed, no dust or whatever could ever creep in. Still it's quite sobering that this cheap 0.30 Euro thing was capable of bringing down a 1600 Euro workstation... Who would suspect a switch to fail in this way ? Ciao, -- Je veux que la mort me trouve plantant mes choux, mais nonchalant d’elle, et encore plus de mon jardin imparfait. (Michel de Montaigne)

15 years, 2 months

6 months job offer at Grame

by Orlarey Yann

The following job offer for a C++ developer may be of interest to you or some one you know. ==== About GRAME ==== Based in Lyon (France), GRAME, Centre National de Création Musicale (national center for musical creation) is a leading music institution dedicated to contemporary music and digital arts. Its activities range from artistic to scientific. GRAME's research department is involved in various French and European research projects as well as in several open source projects (Faust, Jackdmp, MidiShare, libMusicXML, GuidoLib, etc.) Grame's home page: http://www.grame.fr ==== Job Description ==== GRAME is looking for two C++ developers willing to collaborate together with its research team in the development of the future version of FAUST. FAUST (Functional Audio Stream) is a functional programming language specifically designed for high level real-time signal processing and synthesis. FAUST targets high-performance signal processing applications and audio plug-ins for a variety of platforms and standards. Its compiler translates Faust programs into equivalent C++ programs taking care of generating the most efficient code. Thanks to advanced compilation techniques, the resulting code can generally compete with, and sometimes even outperform, C++ code written by seasoned programmers. The future version of FAUST will introduce several new and exciting features, especially the support for multirate and multidimension processing. This will open a whole range of real time applications, from spectral to video processing. The recruited developers will be hired on a full-time limited-term contract of 6 months, from October 1st, 2010 to March 31, 2011. FAUST's home page: http://faust.grame.fr ==== Profile ==== * Excellent C++ programming skills : - methodical work, - clear programming and documentation style, - high productivity - good at understanding and modifying existing code * Strong Computer Science background : - compiler technologies - Lambda-calculus and functional programming - type systems and type inference, - code synthesis * Good Knowledge of audio/video processing techniques and associated Maths. * Familiarity with Linux and GNU Tools ==== How to apply ==== To apply, please send an email to orlarey at grame.fr mentioning "job offer at Grame" in the subject and adding : * A brief presentation letter stating your interest in the offer. * Your resume * Optionally, code samples and pointers to open source projects you have contributed to Yann Orlarey Scientific Director GRAME 9 rue du Garet BP 1185 69202 Lyon Cedex 01 France

15 years, 2 months

ALSA MIDI thru for live playing is ok (at least at the moment)

by Ralf Mardorf

Without running JACK and with running JACK -p2048 44.1Khz and 96KHz, without HR timer enabled and with HR timer enabled ALSA MIDI used to make a thru connection was always ok when playing live, not only for the PCI MIDI, but even for USB MIDI. When playing the recordings from my last test by Qtractor, there wasn't the issue anymore that FluidSynth and the internal Linux recorded FluidSynth sometimes were out of sync. The MIDI and the audio track were always in sync. I only tested it with system timer. Anyway, there definitive is jitter for the recorded FluidSynth. It's inaudible jitter, resp. it is audible when loop playing, because some beats are played to early, because of jitter with negative delay. So it's not an issue of bad timing, but because of notes that shouldn't be played, when loop playing. It's still a recording from the last test, not from today (note, I'm not writing about the hw MIDI at the moment). There must have been something bad, that is ok today. I wondered myself, because I never had this issues before. I didn't test MIDI jitter for hw MIDI today. Anyway, ALSA MIDI for MIDI thru is ok and also Qtractor always plays the MIDI and audio track in sync. For the important jitter tests I'll switch back to 64 Studio 3.0-beta3, there might be something broken for 64 Studio 3.3 alpha or it was just bad luck, anyway at least for FluidSynth with 3.0-beta3 there never were issues. Over the weekend I had no time to test MIDI as it was planned. It likely will be the same for the next days. Again there's hope that PCI MIDI jitter will be inaudible. Because even the USB MIDI was ok for live playing, I suspect that the jitter for the recordings is not regarding to ALSA MIDI. I wish I would have the time to do some recordings right now.

15 years, 2 months

Linux Plumbers Conference: Call For Papers

by rosea grammostola

Hi, AFAIK Lennart and some other Red Hat devs, where talking about the Linux Plumbers Conference at LAC2010, telling why it would be good if people from the linuxaudio world would join that conference and why it could improve the state of linuxaudio in the Linux environment. Here is a call for papers: http://www.linuxplumbersconf.org/2010/06/03/linux-plumbers-conference-call-… \r

15 years, 2 months

Midi tempo sync'd live drum machine

by Nathanael Anderson

Here is the setup i'm looking to do: midi master -> special program ddr pad -> special program special program -> hydrogen I've got a hardware midi device controlling the master tempo. aseqdump shows the tempo messages x times a second. I want to either find or make a program so I can use a ddr pad (usb gamepad) and assign different timed midi patterns to the buttons, so for example: if the program is configured for 4/4 time, when I hold down or toggle a button if note 36 is a bass drum and 41 is a snare then you'd have a basic drum and snare beat when both buttons are held down or toggled, of course i'd like to be able to have multiple patterns bound per button as well button one is set to play a quarter note midi note 36, then rest for the remaining 3 beats 1 - 36,quarter|rest,quarter|36,quarter|rest,quarter 2 - 41,eighth|rest,eighth|41,eighth|rest,eighth|41,eighth|rest,eighth|41,eighth|rest,eighth Each pattern would describe one measure, and the flow of a live solo performance could easily be changed on the fly, leaving hands free to play guitar. Anything out there like this, and I just don't know about it? If I don't find anything I want to hack something together, so a primer on midi tempo sync, and how to program with it would be appreciated. Thanks, Nathanael

15 years, 2 months

Re: [LAD] [PlanetCCRMA] FIxed alsa-tools' envy24control missing peak level meters and "Reset Peaks"

by Niels Mayer

On Fri, Jul 16, 2010 at 5:27 AM, Geoff King <gsking1(a)gmail.com> wrote: > Niels, Thank you for taking the time to work on improving Envy24Ctl. > I tried your patch for the peaks and it seemed to work fine for me. I > can't comment on the code (as I'm much more musician than programmer), > but had no problem patching and installing. Looking forward to trying > this new patch and whatever else you come up with. Geoff Geoff -- thank you for trying out the patch! (And, OT, thanks for working w/ Rui to help fix http://sourceforge.net/tracker/?func=detail&atid=733076&aid=3021645&group_i… :: "help fix qtractor crash on bus changing/configuration (3021645)" -- Envy24 and multichannel qtractor users, yes, it's time for a "svn up" ). FYI, to try out the new levelmeters "easy", if you're running Fedora x86_64, use the "envy24control" binary directly, or drop the levelmeters.c file into your build: http://nielsmayer.com/npm/Efficient-Meters-Envy24Control.tgz ( for details http://nielsmayer.com/npm/Efficient-Meters-Envy24Control.README ) ........ On a completely different "note" the envy 24 manual ( http://alsa.cybermirror.org/manuals/icensemble/envy24.pdf ) has interesting info on the envy24 digital mixer that I've snapshotted: Diagram: http://nielsmayer.com/npm/envy24mixer-architecture.png Note the way it truncates in the mixer: the more inputs you "mix" at once, the fewer bits each input source gets, and it's not clear what kind of dither, or it's a straight truncate of 24... ultimately the envy24 seems oriented towards producing a 16 bit, and not 24 bit master (which makes sense given that the chip is well over a decade old and HD audio production for "prosumer" was rare): .......... 4.5.5 Multi-Track Digital Monitoring The Envy24 integrates a 36-bit resolution digital hardware mixer. The width of the data path is strictly to ensure that during processing of all the channels, under any condition, no resolution is lost. The dynamic range of the end user system will be limited by the range of the physical output devices used. In order to maintain identical gain to the input stream (i.e. 0dB), the resulting 24-bit is not msb-aligned to the 36-bit. The overflow bits correspond to the analog distortion due to saturation. The user would need to reduce the overall attenuation of the inputs to avoid clipping. Insertion of the digital mixer adds only a single sample cycle delay with respect to the original data. This extremely low latency all digital mixer provides monitoring functionality and can replace a traditional external analog input mixer. There are 20 independent audio data streams to mix and control the volume. ... .............. --Niels http://nielsmayer.com

15 years, 2 months

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Linux-audio-dev