Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

3 participants
7535 discussions

by ccernn

axonlib v0.1.0 back on track (and beyond), completely rewritten from scratch, plug-devel 'api' is solidifying, coagulating, lots of new features, lots of squashed bugs, basis is (hopefully) more stable than ever, and if we're lucky, more 'future proof', so, we bumped up the version number. "selling points": OVERVIEW: * opensource, c++ * binary format abstraction for vst plugins and excutables on linux and windows * common look, feel, functionality among platforms * few external dependencies * compile scripts with simplified command lines for the gnu gcc compiler * tiny and compact binaries with no big, external libraries needed * cpu efficient, code prepared for compiler analysis and optimization * options to disable code or functionality that is not needed * flexible axl license, (generally gpl w/ exception for proprietary use) CORE: * builtin fast memory allocator routines (and leak detection functionality) * builtin routines for low-level string and memory manipulation * heavily optimized mathematical functions and approximations * intuitive debugging functionality & helpers * static and runtime assertion GUI: * hierarchial gui, flexible, skinnable, auto-layout, sizeable, moveable etc * resizeable window/editor (in plugin hosts as well) * mouse cursor shapes, hovering hints, mouse capture, modal widgets * low level gfx (gdi/xlib) canvas, surface, bitmap etc * support decoding 32bit pngs from memory or from an external file * scalable, alpha blended bitmaps * partial support for antialiased, transparent lines & textured polygons DSP: * polyphonic voice manager and event scheduler * modular audio graph with connectable dsp modules * rbj filter bank * basic oversampling container * chamberlin state variable filter * rms approximation * envelope follower * basic waveform generators PLUGINS: * lots of included example vst plugins * simplified creation and use of parameters * easy host tempo/sync handling for audio and midi * can load external files directly from plugin folder on both linux and windows OTHER: * basic read/write access for external files * utility methods for bit manipulation and conversations * scripts, stack-based, 4th inspired, rudimentary compilation (bytecode) * builtin, random number generators * mersenne twister implementation, customized for small binary size impact * fft implementation * more... [..and this is probably already outdated..] http://dl.dropbox.com/u/249632/axonlib/0.1.0/screenshot010.png screenshot with: - jost (linux) + axDemo/fx_grains - reaper (win32, via wine) + axDemo/fx_grains - standalone axDemo test binaries: - http://dl.dropbox.com/u/249632/axonlib/0.1.0/bin/linux.tar.gz]linux.tar.gz (686k) - http://dl.dropbox.com/u/249632/axonlib/0.1.0/bin/win32.zip]win32.zip (973k) contains: - vst plugins: axDemo, fx_blur, fx_distortion, fx_freeze, fx_grains, fx_svf, fx_tepodelay, fx_wgtavg, midi_transpose, syn_poly, test_gain_gui, test_gain_gui_skin, test_gain_gui_nogui - executables: axDemo, fx_grains, fx_tempodelay, test_gain_gui, test_gain_gui_skin various levels of buggginess... these, and lots more will be bugfixed, tweaked and developed further as the library progresses. still available (for a limited time): some plugins made with an older version of axonlib (pre r151): - http://sites.google.com/site/ccernnaudio/vst-plugins we would appreciate: - bug reports - questions - ideas - comments - contributions ! - discussions - ... subversion (latest sources): http://axonlib.googlecode.com/ svn snapshot (always a little outdated): axonlib-v0.1.0.zip (r379, 07.jun.2010) - ccernn & neolit123 .....

15 years, 8 months

[LAD] Ardour/Jack issue on Ubuntu Jaunty netbook

by Aaron Krister Johnson

Hi, I have a young bright-eyed, bright-eared 12-year old composition student working on a ASUS EEE-PC netbook. On my advice he switched over to Ubuntu from Windows to do our work. I figured since I'm a veteran Linux user (since 1997!!!) I could help him if he had issues. He's trying to compose some "musique concrete" style things right now using Ardour, and of course, jack. Mostly it goes smoothly, but he does run into crippling issues more frequently. Two things I want to mention: 1) Ardour works fine as it should 90% of the time, except on a netbook, the windows fail to maximize correctly to available screen real estate (1024x600 I think), e.g., when you scroll down, you get the lowest ardour track leaving video trails, and basically it looks like a huge GTK bug of some sort---the interaction with the pointer of course becomes impossible. On my EEE-PC Arch Linux system I can confirm the same behavior. I know 600 pixels is not a lot to work with when you have all those tracks, but there shouldn't be video freezes and trails of graphical widgets. Seems to me to be a really obvious bug. 2) My student is reporting that at least on his Ubuntu machine, he's having a problem getting sound consistently out of Ardour: sometimes, he says, the mixer seems to randomly disconnect the tracks from the "Master Out" bus, and sometimes he reports that jack misbehaves and that he cannot reconnect to it. I'm going to try to get to the bottom of it, but I can report that I've experienced similar things on rare occasions (Ardour 2.8.7, and jack 0.118.0 on Arch, maybe Ubuntu has other destructive aspects?), although I'm much more able to hack around it, as a Linux beginner, and without my tech support, unfortunately he's had to open Windows and finish his assignments in Cakewalk. Which is of course MOST unfortunate for the cause of great Linux audio software advocacy! In talking to his Dad today, apparently, taking my advice and doing a 'sudo killall jackd' and restarting jackd and ardour may have worked--I have to confirm this w/Patrick. But this is not an ideal way of working...... Has anyone experienced similar things? Best, Aaron Krister Johnson http://www.akjmusic.com http://www.untwelve.org

15 years, 8 months

Re: [LAD] qjackctl & server name

by drew Roberts

On Monday 07 June 2010 11:03:06 you wrote: > Or it sees a jack running > when > > > it starts and asks if you want to connect to it or not. > > and for what purpose? qjackctl sole function is being attached to a > running jack server. why do you want to make it an option ? Because perhaps you want it to start up another jack server rather than connecting to this one? Can someone from the jack side speak to the future possibility of determining all running jack servers? if we settle on a naming scheme could this be done via a simple ps -ax? (Not ignoring the hairiness spoken of later in your post.) all the best, drew

15 years, 8 months

Re: [LAD] qjackctl & server name

by drew Roberts

On Monday 07 June 2010 08:59:09 you wrote: > On Mon, 7 Jun 2010 08:18:23 -0400, drew Roberts <zotz(a)100jamz.com> wrote: > >>>> JACK_DEFAULT_SERVER environment variable might be your (only) friend > >>>> here :) > > <snip> > > > Hold on a second. Let me try walking through this. > > > > We start qjackctl. Does it connect to a jack server at this point? If > > so, > > > always or only if jack is currently running. > > it connects only if jackd is currently running. That's what I thought. > > > In cases where it might connect on startup, must it? > > no. again it only connects automatically iif a (default) server is found > responsive to open qjackctl as one of its clients. Right, but does it *have* to do this? Is there no way to make this behavior a config option? Or a query on start option? Config file is set to not attach on startup and so it doesn't. Config file is set to attach on startup and so it does. Or it sees a jack running when it starts and asks if you want to connect to it or not. > > > Let's say no jack is running and we start qjackctl. > > Let's say it doesn't connect to jack at this point. > > i does not. > > > Could there not be a setup option to indicate what -n indicated now? > > qjackctl -n command line option is just convenient for you to start jackd > server with that precise server name and let qjackctl connect immediately > to it as client to that same server. > > > Let's say multiple jacks are running and we start qjackctl. > > Is it possible to discover that multiple jacks are running? > > nope. qjackctl will only "see" the default jack server or the one named by > JACK_DEFAULT_SERVER environment variable at the time qjackctl is launched. Sure. it will only see it as things stand now, but is it impossible for it to see the others? If so, where does this impossibility arise? > > > If so, would it be possible to allow a choice from within the gui as to > > which one to connect to? > > none atm. each qjackctl instance may only attach to one server at a time. I think we may be having language issues here. I am not asking here how things stand atm. Rather could qjackctl be modified to start, see multiple jacks running and list them and ask which you want to connect to and control? Thanks for your time and responses. > > > cheers all the best, drew

15 years, 8 months

[LAD] Job offer at BarcelonaMedia: Real-time audio software developers

by Pau Arumí

Dear All, The following position may be of interest to you. Please forward to anyone interested. Apologies in case of double posting. ==== About the Barcelona Media Audio Group ==== Fundacio Barcelona Media (BM) is a research centre created to foster the competitiveness of the Catalan and Spanish media and communication industry through innovative research activities and projects. BM promotes technology generation and development; research and creativity; transfer of research results to industry; promotion of the research results to society at large; training in all areas of communication; and social awareness of the communication industry in a culture of innovation. The Audio Group research embraces the whole chain of audiovisual productions, focusing specially on 3D surround sound technologies, from capturing, to post-production, to exhibition. Two main general goals are to automatize the capture and post-production workflow by automatic audio adaptation to given 3D scenes, and to make it easily adaptable to any final exhibition system (surround 5.1, 7.1, 22.2, binaural or 3D stereo, etc.). Audio group home page: http://www.barcelonamedia.org/linies/10/en ==== Profile ==== We are looking for one or more experienced software engineers with experience in real-time audio processing. The candidate should preferably have a degree on Computer Science, although other profiles might be taken into account. ==== Required skills ==== * Real-time programming techniques: lock-free multi-threading programming, synchronisation of events from different clocks, etc. * Digital audio processing techniques. * Software-engineering practices for developing large software systems: Design patterns, agile methodologies such as simple design, refactoring and automatic testing. In general, a keen sense for the aesthetics of code, documentation and interfaces. and thoroughness in all aspects of software development. * The ability and willingness to work with a team, communicate well, adhere to standards, share code ownership, use versioning systems, etc. * Programming languages: Proficiency in C++, but also C and Python * Operating systems: GNU/Linux and Mac OS X ==== Other valuable skills ==== * Experience with real-time audio environments (PureData, Max/MSP, Supercollider, Faust, CLAM, etc.) * Plugin development experience with architectures such as: LADSPA, LV2, VST, Audio Units and Protools RTAS. * Experience with communication protocols such as OSC and MIDI. * Experience with time-code protocols such as LTC, MTC and MMC * Knowledge in acoustics technologies * Knowledge in audio and video streaming codecs and architectures. * 3D modeling: Blender or Maya or 3D studio * Qt graphical toolkit * Scons build system * Knowledge in electronics ==== What we offer ==== We offer an opportunity to work in one exciting and creative project related to 3D audio for cinema. The job also includes developing systems for 3D audio exhibition, post-production and interaction, with applications ranging from 3D digital cinema, to sports broadcasting, and video-games. Salary will depend on experience and skills. Side opportunities: perform strategic research in a new promising domain, work in a small-medium young and multidisciplinary including physicists, software engineers and sound designers, collaborate with people from the industry and from other academic research groups, establish contacts with the international audio research community through the attendance to international conferences, etc. ==== How to apply ==== To apply, send email to jobs(a)barcelonamedia.org / cc: toni.mateos(a)barcelonamedia.org, pau.arumi(a)barcelonamedia.org with the subject "Audio job" * A brief presentation letter stating your interest in the offer. * A CV * Optionally, code samples (non open-source samples will be treated confidentially) ==== More background about Barcelona Media ==== BM grew from the Communication Station set up by Universitat Pompeu Fabra in 2001. It is a member of the Catalan and Spanish network of Technology Centres, and is the only one devoted to the Media sector. BM’s trustees are representatives of the Media industry, the Catalan Government, Barcelona City and four universities. BM has an extremely strong record in European collaborative R&D and Innovation projects, both as partner and coordinator. BM is currently involved in 14 EU funded research projects in information and communication technologies with over 5 million € EC funding. BM was coordinator of an FP6 IP and 2 STREPs, including IP-RACINE which researched and developed digital cinema technologies ‘from scene to screen’. It is now coordinating the FP7 ICT IP 2020 3D Media, developing 3D digital cinema and home entertainment. Other directly relevant projects are IP SALERO (‘intelligent content’ objects with context-aware behaviours), SEMEDIA (Search Environments for MEDIA) and FP5 SPEED-FX (very high resolution real-time graphic interaction for digital cinema).

15 years, 8 months

[LAD] [LAA] [ANN] guitarix-0.09.0 release 'Multi'

by hermann

guitarix is a simple Linux Rock Guitar amplifier and is designed to achieve nice thrash/metal/rock/blues guitar sounds. guitarix uses the Jack Audio Connection Kit as its audio backend and brings to the jack audio graph a mono amplifier input/output port, and a FX mono input with two (stereo) output ports. guitarix provide a jack midi input port to connect a midi controller (midi learn) and a (3 channel) jack midi output port, feed by a (scalable) mix of the tuner and a beat-detector. Release 0.09.0 comes with following changes : * make distortion a multiband distortion * fix bug in resample IR-files * resample IR-files now on-the-fly * add insert point ports to the effect chain * add --optimization flag for configure the build * switch to double precision instead of single precision * fix bug in reconfigure convolver when preset change * add patch from Peder Hedlund, build against gtk-2.12.9 and gcc-4.3.1 * add faust 0.9.24 to the "good_faust_versions" have fun ________________________________________________________________________ NOTE:// Andreas have found and solved a bug in the IR-file resampler, that means that all the files you have resampled with guitarix, are "NOT RESAMPLED", only the header was change. To get rid of this, please remove the resampled files and rename the files with the extension .orig (remove .orig) That sucks, sorry for that. hermann _________________________________________________________________________ guitarix is licensed under the GPL. Project page with screenshots: http://guitarix.sourceforge.net/ download: http://sourceforge.net/projects/guitarix/ please report bugs and suggestions in our forum here: http://sourceforge.net/apps/phpbb/guitarix/ here is a nice YouTube video about what you can do with guitarix: http://www.youtube.com/watch?v=KdlJZQT3zY4 (Thanks stevie :-)) ________________________________________________________________________ For capture, guitarix uses the great 'jack_capture' (version >= 0.9.30) written by Kjetil S. Matheussen. If you don't have it installed, you can look here: http://old.notam02.no/arkiv/src/?M=D For extra Impulse Responses, guitarix uses the zita-convolver library, and, for up/down sampling we use zita-resampler, both written by Fons Adriaensen. If you don't have it installed, get it here: http://www.kokkinizita.net/linuxaudio/index.html We use the marvellous faust compiler to build the amp and effects and will say thanks to : Julius Smith http://ccrma.stanford.edu/realsimple/faust/ : Albert Graef http://q-lang.sourceforge.net/examples.html#Faust : Yann Orlary http://faust.grame.fr/ ________________________________________________________________________ For faust users : All used Faust dsp files are included in /guitarix/src/faust, the resulting cc files are in /guitarix/src/faust-cc The tools we use to convert (post-processing and plot) the resulting faust cpp files to the needed include format, stay in the /guitarix/tools directory. ________________________________________________________________________ regards Hermann Meyer, James Warden, Andreas Degert

15 years, 8 months

[LAD] Help, someone interested in taking over a small orphaned jack-module

by Julien Claassen

Hello all! I just got back into asterisk, the big telephony pbx system. It has a JACK module, which is very helpful to me, as it allows me to use asterisk with its cLI to be my telephone. Since I found no other commandline telephone supporting JACK, this is the only way to get there. So asterisk has a small module (1025 lines, including comments), that allows to use JACK. But unfortunitely the programmer of that module has no more time to develop it. So he wrote on the list today. Now I was wondering if anyone of you might be willing to help out here. There are a few people how like to use this module and I'm not the only one, who ends up frustrated. :-( So here are a few links: I uploaded app_jack.c to my site: http://juliencoder.de/app_jack.c I also uploaded libresample - which I think it relies upon: http://juliencoder.de/libresample-310510.tar.bz2 The latest asterisk can be downloaded here: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.2.9-rc1.t… Basic Developer resources (coding guidelines, etc.) can be found here: http://www.asterisk.org/developer/resources The author of the JACK module has written an article, on how to write a module for asterisk, so one can understand the basic structure. It can be found through this link: http://www.venturevoip.com/news.php?rssid=2024 (If you're in a hurry, directly search for "part 1"). That's the best I could find for a starter. I think app_jack relies on libresample, which is somehow connected with asterisk. Beyond that I suppose it's pure JACK and asterisk basic structure. Is anyone here willing to help in some way? Please? 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 ======= AND MY PERSONAL PAGES AT: ======= http://www.juliencoder.de

15 years, 8 months

[LAD] Lost the nekosynth sources...

by Gordon JC Pearce

Unfortunately, due to a problem with the server the nekosynth subversion server was on, I suspect I may have lost all the source to nekobee, nekostring et al. I don't even have a recent copy checked out of svn. The problem has come about because the hosting company stopped invoicing me for the hosting, and bouncing any mail sent to them. It may also be that I need to email them in Finnish, but I don't speak the language. I don't know. The server went down some time ago, but came back up without SSH or svn access, so I wasn't able to pull off any backups. If there's anyone fluent in Finnish who fancies having a go at getting in touch with Seclan to discuss the problem, please contact me off-list. Gordon MM0YEQ

15 years, 8 months

[LAD] Unite file types & project settings

by Harry Van Haaren

Hey all, I've been pondering lately about loading sessions with Linux Audio, and realized that I would find it particularly useful if I could load an LMMS project file into Ardour, and then proceed to move that into QTractor, and do my last mastering in Audacity. As things stand, this is possible, but it requires exporting .wav files, importing them to Ardour, then positioning them, doing what you need to do, and proceed to export them again to move them to another program. I'm hoping to get opinions if dev's think it may be a good idea to try and standardize a common "file-type" or "project file" across Linux Audio Apps. Eg: Programs support a filetype which is known as a .lap (LinuxAudioProject..?), and would contain all the MIDI clips, .wav files, synth settings etc etc across the project. (even JACK output settings?) Open to opinions... -Harry

15 years, 8 months

[LAD] OSC Practices

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, 8 months

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Linux-audio-dev