Linux-audio-dev

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7516 discussions

by Orlarey Yann

================ | FAUST 0.9.24 | ================ GRAME - Centre National de Creation Musicale - is happy to announce the release of FAUST 0.9.24. This version fixes several bugs, and introduces some new possibilities in the language. ------------- About FAUST : ------------- FAUST (Functional Audio Stream) is a functional programming language specifically designed for real-time signal processing and synthesis. A distinctive characteristic of FAUST is to be fully compiled. The FAUST compiler translates DSP specifications into very efficient C++ code that works at sample level. It targets high-performance signal processing applications, libraries and audio plug-ins for a variety of audio platforms and standards. A same FAUST specification can be used to easily generate native JACK or ALSA applications, as well as CSOUND, LADSPA, MAX/MSP, PD, Q, SC and VST plugins. The Faust distribution can be downloaded at: http://sourceforge.net/projects/faudiostream Two mailing lists are available: https://lists.sourceforge.net/lists/listinfo/faudiostream-devel https://lists.sourceforge.net/lists/listinfo/faudiostream-users In order to test FAUST without installing it, please refer to the Online Faust Compiler: http://faust.grame.fr ------------ What's new : ------------ - Explicit substitutions. The language has been extended with new expressions type : exp[x1=def1; x2=def2; ...] allowing explicit substitutions in the lexical environment of an expression. This extension allows for instance, to customize an existing component by replacing some of its internal definitions without having to modify its source code. This extension is particularly useful to promote better code reuse. - Improved mathematical description (--mathdoc option) and support for two new languages : German (-mdlang de) and Italian (-mdlang it) - Support for floating point numbers in scientific notation and better precision for floating point constants. The precision used to print a floating point constant in the generated C++ code is no more limited to 6 digits. It is now dynamically adjusted to find the minimal number of digits that will produce the same internal representation when read back. This approach guarantees accuracy without sacrificing for readability. - All expressions are now systematically represented in polynomial forms. For example x*x will be replaced by x^2. If x is a complex expression the later form has several advantages, in particular to limit CSE. - Lazy semantics to select2 and select3 : the code generated for select2 and select3 is now based on conditional expressions ((cond)?exp1:exp0 ) instead of tables. The resulting code is more efficient as the stateless parts of the branches are not computed every time but only when really needed. - new --task-graph option. It produces a graphical representation of the internal DAG of task in dot format (Graphviz http://www.graphviz.org/). This DAG is useful for example to understand the potential parallelism of a program as analyzed by the Faust compiler - Two new tools : faust2graph and faust2graphviewer. These tools make use of the --task-graph option in order to produce the graphical representation, as a PDF file, of the internal DAG of tasks of a Faust program (require Graphviz). - new reduce.lib library. It provides various operations on block of samples based on a high order 'reduce(op, n)' fold-like function. Moreover the music.lib library has been extended with break-point functions and multiple decorrelated random and noise generators. New flanger and stereowidth control have been added to the effect.lib library. - new iPhone architecture. It consists in a iphone-cocoa.cpp architecture file and an Xcode template project to be used to produce the applications. Use "make iphone" in the example folder to build the examples for the iPhone. - improved cross plateform compatibility and brand new visual studio 2008 project to build Faust on windows machines. ---------- Bug Fixes: ---------- - Report error when non-integer table size is detected during compilation - Corrected partial application of power operator. Now ^(n) is equivalent to \(x).(x^n) and not anymore to \(x).(n^x) - Added missing faustpower definition when power function is used only in table content. - Fixed lock-free implementations of PopHead and PopTail functions on work stealing queues in --scheduler mode - Corrected missing dependencies in the internal DAG of tasks - Added missing cache code to slow shared expressions used delayed - Added missing cache code to foreign functions ---------------- Acknowledgments: ---------------- Many persons have been contributing to the FAUST project by providing code for the compiler, architecture files, libraries, examples, documentation, scripts, bug reports, ideas, etc. I would like to thank them and especially: Fons Adriaensen, Tiziano Bole, Baktery Chanka, Thomas Charbonnel, Damien Cramet, Etienne Gaudrin, Albert Graef, Stefan Kersten, Victor Lazzarini, Matthieu Leberre, Mathieu Leroi, Kjetil Matheussen, Remy Muller, Sampo Savolainen, Nicolas Scaringella, Stephen Sinclair, Travis Skare, Julius Smith, as well as my colleagues at GRAME, in particular : Dominique Fober, Stephane Letz and Karim Barkati, and from the ASTREE project : Jerome Barthelemy (IRCAM), Alain Bonardi (IRCAM), Raffaele Ciavarella (IRCAM), Pierre Jouvelot (Ecole des Mines/ParisTech), Laurent Pottier (U. Saint-Etienne) Yann Orlarey GRAME

15 years, 5 months

LAC "Tools" round-table wrapup

by Robin Gareus

Hi LADs, Following up on the LAC Tools round-table, I've started a wiki page: http://wiki.linuxaudio.org/wiki/tools_comparison Since I've not taken part in the discussion, I'm missing a few footnotes and explanation for keys in the context (eg. batch, sync). Could you please enlighten me, or just fill in the missing content there. IMHO it'd also make sense to do a 2nd table with orthogonal practical information, for instance: audio: JACK,ALSA,.. midi: JACK,ALSA,ALSA-seq audio-file-formats: pcm,mp3,gig,mid,.. control: OSC,TCP,pipe,MMC,.. interop/sync: jack-transport,MTC plugins (if applicable): LV2,LADSPA,VST,AU,.. Come to think of it, those could be integrated in the apps-database as tags for each tool, similar to: http://wiki.linuxaudio.org/apps/categories/jack_transport but let's get the categories/tags straightened out before starting on an implementation and posting it to LAU. What do you think? Cheers! robin

15 years, 5 months

Need to unsub, but server is down.

by Gene Heskett

Greetings; Can someone go look at lists.linuxaudio.org? The bottom line of the message is timing out. Thanks. -- Cheers, Gene "There are four boxes to be used in defense of liberty: soap, ballot, jury, and ammo. Please use in that order." -Ed Howdershelt (Author) "Live or die, I'll make a million." -- Reebus Kneebus, before his jump to the center of the earth, Firesign Theater

15 years, 5 months

[OT] lectureship in composition, music dept. NUIM Ireland

by Victor Lazzarini

Hi everyone, We have a new post in our department, which might interest some of you. http://www.publicjobs.ie/publicjobs/en/star/goToJobDetails.do?id=419 Unfortunately, given my involvement in the selection process, I can't answer any queries related to this post, but my colleague Gordon Delap will be most happy to do so. Maynooth is an interesting place to be at the moment, so I hope some of you will find this worth a go. All the best Victor

15 years, 5 months

guitarix-0.08.0 release 'Rockers'

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 in one input and two output ports to the jack audio graph. 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 0.08.0 comes with a bunch of major changes : * new portmap window * new BiQuad Filter * new Moog Filter * new Flanger * new post processing tube3 * new two 10 band parametric EQ's * new oversample mode selector(1x -8x) * fix some bugs in midi out * reworked tuner interface (show frequency(hz)) * new configure options (see ./waf --help for more info) have fun ________________________________________________________________________ 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/ ________________________________________________________________________ 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, 5 months

LAC 2010 performances

by Marc Groenewegen

Patrick Shirkey wrote: >... > Wanda da Bizzarre Bunny Rocks!!!! Very heavy dynamics they are > achieving. How were the ears after that set? Or maybe a better question > is how many heads exploded during that set? None. Stefan Janssen was at the mixer, which meant that we could walk around safely without earplugs. Great set indeed ! Cheers, Marc

15 years, 5 months

file and communications for RT audio

by Stephen Sinclair

I have an idea in mind for an application that would involve a core audio callback responsible for playing several sounds at the same time, each being streamed in by some as-yet-undetermined means. Before I get too far into it, I have a few questions about the best method for ensuring that the audio callback is not interrupted for lengthy disk access, etc. Obviously I am not planning on doing the main disk I/O in the callback, but I am thinking about the best means for the callback to communicate with the rest of the application. Possibly I might like to support having some of these streams come from external processes, opened through popen() for example. So, the idea for an RT audio callback is that it should not wait on data, (whether it comes from a file or process), but continue processing the other streams if audio data is not immediately available. There are a few ways to do this in Linux: 1) Have a secondary thread responsible for passing data to the audio callback through a wait-free ring buffer. 2) Read from a pipe, FIFO, or socket from another process (e.g. popen), using select() or poll() to check when there is actually data to read. 3) Read from a file, using select()? 4) The async I/O API. 5) Interprocess shared memory, presumably using a semaphore of some kind. I guess this is similar to (1) but for inter-process communication. The question is, which one of these methods is the most "real-time friendly"? Under what conditions, if any, can I be sure a read() will not block? Is there any advantage to threads vs. processes? Using async I/O I suppose I could avoid either one. Are there any general guidelines somewhere for dealing with I/O in audio applications? thanks in advance, Steve

15 years, 6 months

Is it possible to use two RME cards in sync?

by Pau Arumí

Hi, i have another question to RME experts (hi Florian!). the system i need to build should have 8 AES inputs and 24 AES outputs. For that, i see two basic approaches 1) using two RME AES-32 (*) cards in world-clock synch and 2) using a RME hammerfall with 24 ADAT input/outputs and then several 8ch ADAT<->AES converters such as the Aphex 144 (+), which costs about 400€ each 8ch unit. Option 1 seems to me simpler and cheaper but i ignore if is feasible and reliable: Have anyone had good experiences using two clock-synced RME cards with jack? Are there other options apart from 1 and 2? Thanks a lot! P * http://www.rme-audio.de/en_products_hdsp_aes32.php + http://www.aphex.com/144.htm

15 years, 6 months

LAC 2010 stream recordings...

by Jörn Nettingsmeier

hi *! the lac2010 presentation recordings are now available at http://www.linuxproaudio.org/lac2010/ - kudos to faberman for very-close-to-realtime post-production! let me take the opportunity to thank all stream team people (many of them members of the linux video community, who put in many hours of volunteer work to cover our beloved little annual meeting)! check out their works and websites, join their projects, send them beer! christian thäter, germany (http://lumiera.org/) - cam operator, vision mixing florian faber, germany - post production, archive, software support, club mate frank neumann, germany - cam operator, emcee herman robak, norway (http://developer.skolelinux.no/~herman/) - director of photography, cams, vision mixing, hardware marc-olivier barre, france (http://marcochapeau.org/) - relay operator yours truly, germany (http://stackingdwarves.net) - vision mixing, technical supervisor, relay operator raffaella traniello, italy (http://www.g-raffa.eu/, http://vimeo.com/raffatraniello) - cam operator robin gareus, france (http://gareus.org/) - hardware, setup, technical support thijs koerselman, the netherlands (http://www.vauxlab.com) - cam operator wouter verwijlen, the netherlands (http://www.wouterverwijlen.nl) - cam operator and of course major kudos to marc groenewegen and everyone at hku for a great conference! best, jörn

15 years, 6 months

Rendering Softsynth Instruments To Sample Based Instruments - Automation

by Guru Prasad

Hello everyone, This is my first post on this list, so please excuse me if I'm not following list etiquette. I don't think I can be called a developer, although I can get things done with Python scripts. I'm much more of an end user, a musician using linux live - LinuxSampler being my 'bread and butter' tool. I have a specific application in mind, and I dont think there's anything out there that does it: I want to 'sample' a softsynth (e.g. Yoshimi) / modelled software instrument. In other words, I want something that automates the following process: 1. Sequencially play 88 X N notes, where N is the number of velocity layers 2. Play them through a particular preset on the softsynth/modelled instrument (basically just send MIDI output to the softsynth/modelled piano, through ALSA/JACK) 3. Record each sound (corresponding to each note/velocity) as a .wav file (at a specified sample rate, etc) The sample files can then be converted into a .gig instrument sample using Gigedit. The advantage of this process would be that one can now use (more or less) the same sounds with MUCH less CPU usage. the disadvantage, of course, is the loss of tweakability, but I can live with that, especially in live contexts. I searched for an application that does this, and the closest I got to was this feature in Renoise: http://tutorials.renoise.com/wiki/Render_or_Freeze_Plugin_Instruments_to_Sa… But I don't use Renoise live. Which brings me to my question(s): 1. Am I right about there not being any application that does this? (I hope I'm not!) 2. If yes, can someone point me in the direction of how I might get about writing a script that does this? 2a. What Python modules would be useful? I just came across PyAudio and PySndObj, and checking them out. I'm sure there are *quite* a few people who would find such a script very useful, and I intend to make it available ASAP. My programming skills are limited, though, so any help in this regard would be appreciated. Thanks for your time and consideration! Cheers, Guru

15 years, 6 months

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Linux-audio-dev