Linux-audio-dev

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7516 discussions

by Olivier Guilyardi

Hi everyone, a Jackbeat user is currently trying to use TouchOSC, an iPhone app, as a controller, and there is a need to translate in/out OSC messages. I have looked at TouchOSC messages, and the translation is nothing like straightforward. There is a so-called BeatMachine[1] controller in TouchOSC, which looks like a drum machine, but in the messages that it sends: 1. the track order is reversed, the first track being at the bootom 2. the indexes are one-based in TouchOSC, and zero-based in Jackbeat 3. the track/beat position is transmitted as a path element instead of a value The fact that some translation is needed between OSC units is quite normal to me, but I would like to try and follow some wide-spread practices, if any, maybe refactoring the Jackbeat OSC dialect, to make such translation easier. I originally looked at some Monome messages docs[2], and that's the reason for 2. and 3. above. The point 1. seems absolutely odd to me however. Now, there's another thing: currently Jackbeat features quite a few OSC methods/commands but little OSC events[3]. This forbids using search an external controller as some kind of UI replacement. For instance, volume sliders can be controlled, but no event is sent when they change. One of the reason for this lack is that I was unsure about the right protocol to choose.. So, in TouchOSC, input and output messages seem to always be symmetrical, which looks like a clever and intuitive design to me. For instance, a slider both sends and receives messages at /1/fader1 (example). And you just can't setup a different prefix for in and out. Once again, this isn't the way Monome is doing it: it sends /40h/press, and receives at /40h/led. So I'm a bit confused. What would be ideal is to follow some kind of practices and/or standard, so that, in many cases, translation is only a matter of changing OSC prefixes. And if that's not realistic, then I would like to avoid some of the inconsistencies listed above with tools as TouchOSC and such. Is there any such standard or common/recommended practices? Cheers, [1] http://hexler.net/pub/touchosc/touchosc-manual-v1-1.pdf [2] http://docs.monome.org/doku.php?id=tech:protocol:osc [3] http://jackbeat.samalyse.org/wiki/BasicUsage#OSCInterface -- Olivier

15 years, 5 months

SConstruct for building minicomputer with fltk versions 1 + 2 in simultaneous existence

by James Morris

Hi, The MiniComputer softsynth has problems building on many distros when fltk-1.x is installed alongside fltk2, and/or the #include directories don't match where MiniComputer expects them to be and so the build fails at the configuration stage. In an effort to fix this quite long standing problem, I've adapted the SConstruct to use fltk-config and pkg-config to solve these problems. If you have problems building MiniComputer replace the SConstruct that comes with MiniComputer with the following file (obviously rename it to SConstruct first): http://jwm-art.net/art/text/MiniComputer_SConstruct It works for me on Gentoo (using accept keyword ~amd64). I posted here because: 1) I saw similar questions without answers on the list in the past. 2) I've spent several hours getting this to work so people might benefit from it. 3) I'm a python/scons newbie and hope that those of you with better knowledge can check it for problems (I've only got this to work by using Google, trial, and error). Cheers, James.

15 years, 5 months

[Release] Autotalent LV2 Port

by Jeremy

Hi all, I just finished up my port of Autotalent (A real-time pitch manipulation program by Tom Baran) to an LV2 plugin (Okay, It's not finished, but all the important features are there.) It should work basically the same as Autotalent except: - It is an LV2 plugin instead of an LADSPA plugin - It provides MIDI output of the pitch - It accepts MIDI input - It separates the pull to semitone and snap to scale functionality - It uses FFTW for the DFT routines, greatly improving performance. - Minor performance tweaks (substituting memcpy for loops, etc) - It is greatly refactored (broken into methods and structures, variables renamed) - Does not include smooth pitch jumping or LFO functionality in the first release (will add later) - The formant corrector causes artifacts not present in the original (I'm not sure how to fix, as it's the only part of the original I don't understand) I think especially significant is the midi input and output. This would allow you to do the following: - Record an audio track in your favorite DAW while running it through the plugin, and outputting the MIDI to another track - Correct the midi however you want (keep it monophonic) - Feed the corrected MIDI track back into the plugin with the recorded audio, and listen as it corrects the pitch to match the MIDI you gave! There are also other options, such as controlling an MIDI synth with your voice, or getting a vocoder-like effect by controlling your voice with a MIDI keyboard. You can check out the project page for more information here<http://code.google.com/p/autotalentlv2/> or if you want a direct source download, see here<http://autotalentlv2.googlecode.com/files/autotalent_source.tar.gz> . Any feedback on the source code or plugin would be welcome. Jeremy Salwen

15 years, 5 months

plugins page updated at linux-sound.org

by Dave Phillips

Greetings, Every now and then I decide to update a page on the original Linux soundapps site. Recently I cleaned up this page for audio/MIDI plugins : http://linux-sound.org/plugins.html Please advise if I've left out anything or if any information there is wrong. I've covered plugins in LADSPA, DSSI, MESS, LV2, and VST formats, hopefully I got all the major projects, and I'd like to know if I've missed anything. TIA! Best, dp

15 years, 5 months

A little quiz about audio measurements...

by fons＠kokkinizita.net

Hello all, This week I had to perform measurements on a audio interface, and this resulted in some quite interesting results. Before revealing what happened, I'll let you have a look at some of the data and come up with your own conclusions, see <http://www.kokkinizita.net/linuxaudio/quiz.html> (Free beer at the next LAC for the best analysis) Ciao, -- FA O tu, che porte, correndo si ? E guerra e morte !

15 years, 5 months

Re: [LAD] plugins page updated at linux-sound.org

by Dave Phillips

Johannes Kroll wrote: > In the native Linux VST section, you might want > to add Wolpertinger, a little subtractive synth in development, open > source: http://tumbetoene.tuxfamily.org/index.php?category=1 > Done. :) > Another excellent Linux native VST host is Renoise: www.renoise.com. > Non-free though, demo available, the full version costs 58 euros. > > I've added it to the list of compliant hosts. Thanks ! dp

15 years, 5 months

Quiz -- The winners

by fons＠kokkinizita.net

Hello all, This has stirred up more interest (also off-list) than I ever imagined it would ! OK, beers will go to (in no particular order) Niels: his comments on ADC/DAC reference voltages provide the key to understanding what happened. Veronica: for the most to-the-point one-liner. Florian: (off list) who got it right in the end. I guess that at least initially all of you put the blame of card X - either it's crap or it's defective. But the real culprit is card A, even if it did show quite well in the loopback test. And what happens is 100 Hz (rectified power grid 50 Hz) ripple on the DAC and ADC reference voltages, both derived from the same source. Most audio ADCs and DACs have two reference voltages (or one, and then the second is derived internally). In many cases these are the voltages that correspond to the min and max PCM values, and even if that is not the case the conversion factor of the ADC or DAC depends linearly on these voltages. If they are not well filtered, any fluctuations do generate AM, and the effects of the same eror on a ADC and DAC will cancel out. Cancellation also occurs if the sample clock (if shared by ADC and DAC) is modulated, but here the effect is delay modulation, which for a single tone translates into phase modulation proportional to the frequency of the signal. Since in this case the sideband levels do not depend on the measurement frequency, the sidebands are not the result of clock jitter. So Veronica is quite right: you can't trust a loopback test. And if, as in this case, the variations are the result of mains hum, you can't even trust a setup of two cards using the same mains power (although exact cancellation as in this case would be rare). It is also true that you have to be very careful when using a low-fi card to test a much better one. In this case the only purpose of the USB card was to provide a calibrated analog level, which is required for e.g. EIN measurement. The card is portable so I can easily take it to the lab and calibrate it against a high precision certified RMS meter which I can't take out without first filling out a hundred forms and then waiting two weeks. Meanwhile I did measure the signal quality of card X using a *decent* generator - it is extremely good. Ciao, -- FA O tu, che porte, correndo si ? E guerra e morte !

15 years, 5 months

Simple LV2 jack host

by Jeremy

Hi, I'm trying to find a simple host for my LV2 plugin that allows both midi in and out, and also allows you to control plugin parameters. I've found lv2_jack_host, which allows my port configuration, but doesn't seem to allow you to change the parameters, and I've found lv2rack and zynjacku, which allow you to change plugin controls, but don't seem to allow nonstandard input/output port configurations. Any suggestions would be nice. Thank you Jeremy

15 years, 5 months

like "qjackctl", but trimmed of all fat

by Aaron Krister Johnson

Hi linux-audio peoples, I want to announce my new and fun and useful python script -- I call it 'jackctl' -- it basically is a frontend to the jack tools 'jack_lsp' and 'jack_connect'. It's like 'qjackctl' , but since it's command line, no 'q' !!! get it here: http://www.akjmusic.com/software/jackctl20100526.py Why would I embark on writing such a script? Who would want to use this? You'll want to try this if one or any of the the following apply: 1) You don't want to have to install the entire QT toolkit for a single program. 2) If you like the command line, use a console, but still use jack often enough 3) You've used jack and 'jack_connect' through the command line before, but hated having to type the full name of the jack ports. 4) You've noticed that 'qjackctl' introduces CPU overhead and xruns you don't have when you use command-line jack, i.e. you are obsessed with the lowest possible latency 5) You want a fast, simple interface to connect jack ports that is even faster than a GUI---no need to point the mouse, just type two numbers and go! It's very simple. All you need is Python (and who doesn't have that on their machine). You put the script in your PATH, or link it/rename it, making sure it's exectuable. When you run it, you'll see a list of current numbered jack ports, and you can connect them by typing two numbers separated by a space. You can disconnect them by typing 'd' then the two numbers separated by a space. No hassles, and a nice feature is that it will protect you from making ear-blasting feedback connections. It's even quicker than qjackctl, b/c it takes more time to point your mouse at the ports in the GUI and then click 'connect' than it does to type two single-digit numbers and then hit return, yes? Let me know how you like it...I'm interested in reasonable feature requests. One potential TODO would be to make this script have a user-friendly way to start the jack daemon, but for now, I do that manually.... Enjoy, comments welcome! -- Aaron Krister Johnson http://www.akjmusic.com http://www.untwelve.org

15 years, 5 months

Re: [LAD] A little quiz about audio measurements...

by Ralf Mardorf

Olivier Guilyardi wrote: > On 05/28/2010 07:36 PM, Ralf Mardorf wrote: > >> Folderol wrote: >> >>> On Fri, 28 May 2010 19:20:54 +0200 >>> Ralf Mardorf <ralf.mardorf(a)alice-dsl.net> wrote: >>> >>> >>> >>>> Veronica Merryfield wrote: >>>> >>>> >>>>> You can't trust a loop back test. >>>>> Any instability or dither on the reference clock of card A (fifo >>>>> clocking say) is not going to show in a loop back test. >>>>> Vrnc >>>>> >>>>> >>>> Is Veronica Merryfield the winner? >>>> >>>> >>> I'm highly suspicious of the USB link, but can't quite put my finger >>> on why. >>> >>> >> Card A is the USB card. For USB there could be several issues, but I >> don't have knowledge about buffering etc., but I guess it's card A and >> that there's a "instability" = jitter. I don't know what dither for CLK >> is. I guess the winner is Veronica Merryfield. >> > > I mentioned the clock problem first ;-) However, I thought it the other way > around: I said that clocks being asynchronous that would generate artefacts, but > Veronica seems to say that these are hidden when using a single clock. > > That's pretty much the same thing to me :p > > -- > Olivier > Did you also say for what card? A or X? If so, is Oliver the winner? Btw.: Ralf Mardorf wrote: > Gabriel Beddingfield wrote: >> On Fri, May 28, 2010 at 12:39 PM, Gabriel M. Beddingfield >> <gabrbedd(a)gmail.com> wrote: >> >>> The 100 Hz (being 2x 50Hz, the power freq. in Italy) >>> suggests that it is probably related to some manner of >>> power supply. However, I have no theory why we're >>> getting 2x 50Hz (and I think I need one :-)). >>> >> >> Doh! When the AC wave is rectified, it results in a signal that is 2x >> the freq. because the negative part gets inverted. That's why we see >> 100 Hz instead of 50 Hz. >> >> -gabriel >> > > On card A or X? > Why AM and not additive signals? Is the jitter caused because of residual ripple? Summarized: Residual ripple for the DC could cause clock jitter and this for card A.

15 years, 5 months

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Linux-audio-dev