Linux-audio-dev

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7534 discussions

[linux-audio-dev] Hi, is it possible to record multi channel speech audio and down sample it 2 stereo channels BUT with horizontal audio positioning of each audio channel ie. like dolby surround effect. I'm not looking to make multi channel podcast, just a plain stere

by Steve Harris

On 13 Mar 2007, at 09:15, Valent Turkovic wrote: > Hi, > is it possible to record multi channel speech audio and down sample it > 2 stereo channels BUT with horizontal audio positioning of each audio > channel ie. like dolby surround effect. Sure, but you might be disappointed by the results, Pro Logic is not very sophisticated (eg. there's no separate rear left and right channel, and the bandwidth is limited). Also the only free software implementation that I know of is not state of the art, partly due to patent problems, and partly due to lack of interest. There's a LADSPA plugin: http://plugin.org.uk/ladspa-swh/docs/ladspa- swh.html#id1401 which does Pro Logic matrix encoding, you will need some other software to mix your independent streams into LCR channels (you can do it by ear with a 3+ variable send bus mixer), but once you have that you can feed it to the plugin and get a pro logic compatible stream out. PS ignore the documentation, it has been tested, and it does work, just not brilliantly. You will get better results if you EQ the mics a bit to cut the level of the highs and lows a bit, especially in the rear channel. You would get much better results with an ambisonic stream, but I don't /think/ you can make stereo-compatible ambisonic streams, and not many people have ambisonic decoders. Other people on this list know a lot more about ambisonics though. - Steve

18 years, 10 months

[linux-audio-dev] [ANN] FreeBoB bugfix release - libfreebob-1.0.3

by Pieter Palmers

Dear FireWire enabled Linux audio users, libfreebob 1.0.3 is available as from today. It is downloadable at our SourceForge page: http://downloads.sourceforge.net/freebob/libfreebob-1.0.3.tar.gz This is a maintenance release for the freebob 1.0 branch, and contains no new features. It fixes two bugs: - a buffer reset bug that prevented jackd freewheeling from working. - a bug that caused MIDI output to fail on all but the last channel of a device. Greets, Pieter

18 years, 10 months

[linux-audio-dev] Loss Livecode festival, Sheffield 20-22 July 2007

by alex

Festival/conference about live coding in Sheffield, UK this Summer... ====================================================================== _ ___ ____ ____ _ _ ____ _ | | / _ \/ ___/ ___| | | (_)_ _____ / ___|___ __| | ___ | | | | | \___ \___ \ | | | \ \ / / _ \ | / _ \ / _` |/ _ \ | |__| |_| |___) |__) | | |___| |\ V / __/ |__| (_) | (_| | __/ |_____\___/|____/____/ |_____|_| \_/ \___|\____\___/ \__,_|\___| ---------------------> LOSS Livecode Festival <----------------------- Sheffield, UK -- 20-22 July 2007 http://livecode.access-space.org/ In association with Access Space, TOPLAP and lurk When we improvise music, we are creating music while it is being performed. "Live Coding" is the creation of software while it is being executed; the software in turn generating music or video. Thanks to dynamic programming languages, the live coder is able to modify and extend their program without restarts, their music and/or visual growing with the code that describes it. This way of working allows instant results for every sourcecode edit. Programming becomes a fast, creative process - expressive enough that a whole audio/visual performance may be created as software. Live Coding began during the 1980s, primarily with FORTH and Lisp. In recent years new live coding environments and languages such as Chuck, Fluxus, Impromptu and SuperCollider 3 have appeared, with enthusiastic communities growing around them. Live Coding performances have also used Smalltalk, PureData, Scheme, Perl, Haskell, Ruby, Python... In early 2004 the "Temporary Organisation for the Promotion of Live Algorithm Programming" (TOPLAP) was formed to support open dialog between all live coders. Since its early beginnings in a smoky bar in Hamburg, TOPLAP has reached 178 members worldwide, gaining coverage in mass media and collaboratively organising several international meetings. In 2005 Access Space initiated the L.O.S.S. project (http://loss.access-space.org) to support free music creativity and distribution. It featured a series of commissions leading to a Creative Commons licensed audio CD and repository website produced entirely with open source tools. Continuing their series of LOSS commissions and events, Access Space have teamed up with TOPLAP and lurk to create a three day international festival, bringing live coding musicians and video artists together to explore and showcase new approaches in live performing and participatory arts. ---> CALL FOR PARTICIPATION <----------------------------------------- Your performance and/or presentation proposal is called for. For the latest version of this call, please refer to http://livecode.access-space.org/ Commissions are available to help realise ambitious projects and performances. Presenters and performers will gain free entry throughout the festival, and those without institutional support may apply for a small bursary. ---> IMPORTANT DATES <--- * 14th March - Call for participation * 14th April - Deadline for proposals * 1st May - Notification of acceptance * 16th June - Copy deadline for proceedings (to be confirmed) * 20th-22nd July - Conference - schedule TBA ---> PRESENTATIONS <--- Short (up to 20 minute) presentations during a day long symposium. The remit is broad, but possible subjects may include * A demo of a novel live coding language/environment * Historical context of live coding * Live coding without computers * Critique of live coding practice * Live patching * Reflections on live coding experiences * Adapting general purpose languages to live coding * Analysis of live coding performances * Live algorithms that live code * Life coding * Portable live coding devices * Reflective/self-modifying code * Live visualisation of sourcecode * Collaborative networked live coding * ... Proposals do not have to be long - however much or little you need to explain your ideas is fine. If you are unsure if you can make it, submit your idea anyway - we may be able to accommodate a small number of remotely streamed presentations for those unable to attend in person. There will also be time for a brief (around three hours) introductory workshop. Please indicate if you would like to be involved. ---> PERFORMANCES <--- There will be at least two evenings of performances, ranging from 10 to 40 minutes. Please outline what you would like to perform, including technical requirements. We plan to have at least three data projectors, many pairs of small speakers for participatory improvisations, enough headphone amps for 100 pairs of headphones, and a big stereo sound-system for 'traditional' performances. Please state your preference, and feel free to be creative (see commissions below). We are also thinking about a pre-event in London, UK some days before the festival, let us know if you would like to take part. ---> PROCEEDINGS <--- If your proposal is accepted you will be encouraged to submit short texts and images for publication in the proceedings. All speakers and performers will receive a free copy at the beginning of the conference. ---> COMMISSIONS <--- If you would like time or resources to develop a new way of performing, some new language or software feature, or something else interesting then please include a short estimated budget in your brief, which may include an artist fee. Note that due to funding constraints the project should have a strong audio component. The maximum commission will be of £1000 (about 1470 euro). ---> BURSARY <--- A small bursary is available to contribute towards travel and accommodation. Please include an estimated budget for your attendance and we will apportion this money based on need. Money is however very short, if you are a member of an academic institution we are keen to help you apply for local funding. ---> PROPOSAL SUBMISSION <--- Preferably in plaintext, but all common formats are accepted. Supporting material including web links to previous work, audio and video files are welcome but not mandatory. Proposals should arrive before midnight, 14th April 2007. Proposals are accepted by email (preferred): livecode(a)access-space.org Or by post: LOSS Livecode c/o Alex McLean Access Space 1 Sidney Street Sheffield S1 4RG Royaume Uni If sending via email please do not include large attachments - either include URLs or contact us in advance. If sending via post include an email address so that we may confirm receipt. ---> MAILING LIST <--- As members of the "keep avant garde internet tidy" campaign, we keep our cross posts to a minimum. To continue receiving news of the conference, please sign up to our mailing list: http://lists.lurk.org/mailman/listinfo/lc/ ---> ABOUT ACCESS SPACE <--- Based in Sheffield, Access Space is the UK's first "Free Media Lab" - a community space equipped with locally recycled computers running free, open source software. It provides a framework, resources and support for self-directed learning, arts and creativity. Taking part is totally free, and anyone can walk in and contribute: http://access-space.org ---> FOR MORE INFORMATION <--- Don't hesitate to email with questions to the submission address above. The conference website is not yet ready, but more information about live coding may be found at the official TOPLAP wiki: http://toplap.org/ Hope to see you in July! LOSS Livecode is funded by Arts Council England, Yorkshire and The PRS Foundation. ====================================================================== -- Alex McLean http://yaxu.org/ http://slub.org/ http://lurk.org/ http://doc.gold.ac.uk/~ma503am/

18 years, 11 months

Re: [linux-audio-dev] Getting out of the software game

by Taybin Rutkin

By continually breaking the drivers, it forces someone to look over them when updating and maybe fix other problems. Taybin -----Original Message----- >From: Christian Schoenebeck <cuse(a)users.sourceforge.net> >Sent: Mar 14, 2007 10:17 AM >To: The Linux Audio Developers' Mailing List <linux-audio-dev(a)music.columbia.edu> >Subject: Re: [linux-audio-dev] Getting out of the software game > >Am Mittwoch, 14. März 2007 14:16 schrieb Paul Davis: >> in theory, you certainly can. but the kernel development team, and linus >> in particular, are not interested in an engineering effort/long term >> approach that makes this feasible. if you define a stable driver binary >> interface, you can change the kernel out around it and drivers keep >> working. linus has made it clear that he sees no reason to do this, and >> is perhaps even opposed to it for some possibly sound engineering >> arguments (though that is open to debate). > >And what are these arguments? > >CU >Christian

18 years, 11 months

[linux-audio-dev] Some ALSA mixer API questions

by Ross Vandegrift

Hey everyone, I'm working my way through a simple mixer application using ALSA's mixer API. However, the mixer section of the documentation is blank, so I've taken to reading through amixer's source code to try and figure out how it does it's things. It's a bit hard to piece together how some of the stuff goes together. I've got the basics of getting/setting volume values and getting volume ranges from mixer elements. I'm a bit confused by how to get element types. It seems like amixer opens the ctl device to get a mixer element's type? Am I right about this? It seems like a very confusing API to require the ctl device to get info on an element from the mixer device. If it does require a handle to the ctl device, could someone give me the quick overview of how this works together in the big picture? If not, I'd appreciate a quick explanation of the right way to determine the type of a mixer element. -- Ross Vandegrift ross(a)kallisti.us "The good Christian should beware of mathematicians, and all those who make empty prophecies. The danger already exists that the mathematicians have made a covenant with the devil to darken the spirit and to confine man in the bonds of Hell." --St. Augustine, De Genesi ad Litteram, Book II, xviii, 37

18 years, 11 months

Re: [linux-audio-dev] Hi, is it possible to record multi channel speech audio and down sample it 2 stereo channels BUT with horizontal audio positioning of each audio channel ie. like dolby surround effect. I'm not looking to make multi channel podca

by Valent Turkovic

> To start with have a look at Wikipedia. They have a pretty good section > about audio engineering which covers a lot of topics including > sterophony. (At least the German edition does) > > > > Yours sincerely, > Dennis Schulmeister Thank you for your suggestions.

18 years, 11 months

by Valent Turkovic

Hi, is it possible to record multi channel speech audio and down sample it 2 stereo channels BUT with horizontal audio positioning of each audio channel ie. like dolby surround effect. I'm not looking to make multi channel podcast, just a plain stereo podcast. Ok, not just plain, maybe better term is enhanced podcast with "dolby surround" or "binaural" effect so that different voices appear to be coming from different positions. The idea is to make a podcast with 5 (or more/less people) and record each of them individually and to make an stereo file for distribution but such that each person has a distinct "location" in your ear. So you can hear that person A is in front of you, person B is on the far left, person C is to the right but closer to center... and so on. I would maybe have 3 channels sometimes and sometimes maybe 7 channels, it differs by the amount of guests there are in each podcast show. Is there some linux audio tool which I could feed 3-7 wav tracks and define where in horizontal space I want each channel to be located - and it makes final stereo track with this "binaural" or "Dolby surround" enhanced podcast? Thank you in advance. -- http://kernelreloaded.blog385.com/ linux, blog, anime, spirituality, windsurf, wireless registered as user #367004 with the Linux Counter, http://counter.li.org. ICQ: 2125241 Skype: valent.turkovic

18 years, 11 months

[linux-audio-dev] ANN: munger1~ minor update

by Ivica Ico Bukvic

http://ico.bukvic.net/Max/munger1~_1.0.1.tar.gz (includes changelog and install how-to documentation fixes) Best wishes, Ivica Ico Bukvic, D.M.A. Composition, Music Technology, CCTAD, and CHCI Virginia Tech Dept. of Music - 0240 Blacksburg, VA 24061 (540) 231-1137 (540) 231-5034 (fax) ico(a)vt.edu http://www.music.vt.edu/people/faculty/bukvic/

18 years, 11 months

by Valent Turkovic

On 3/13/07, Georg Holzmann <georg.holzmann(a)student.kug.ac.at> wrote: > Hallo! > > > I'm not looking to make multi channel podcast, just a plain stereo > > podcast. Ok, not just plain, maybe better term is enhanced podcast > > with "dolby surround" or "binaural" effect so that different voices > > appear to be coming from different positions. > [...] > > Is there some linux audio tool which I could feed 3-7 wav tracks and > > define where in horizontal space I want each channel to be located - > > and it makes final stereo track with this "binaural" or "Dolby > > surround" enhanced podcast? > > Yes, there are various implementations of binaural spatialization in > Pure Data (and I guess also in Supercollider and CSound ...). > > I am just making some ready-made tutorial patches regarding this topic > for the linux audio conference - so if you wait some days you can have > them ! > > LG > Georg > Great! I would love to gave it! Thank you in advance. Valent.

18 years, 11 months

[linux-audio-dev] ANN: munger1~ port (from the Percolate library) with new stuff

by Ivica Ico Bukvic

Greetings all, For the impatient, download at: http://ico.bukvic.net/Max/munger1~_1.0.0.tar.gz (270KB, includes source, Linux-Pd-i386, Mac-Max-i386, and Win32-Max-i386 binaries, and 3 cases of beer) OVERVIEW ======== munger1~ (March 12, 2007 1.0.0 release) a realtime multichannel granulator a.k.a. the swiss-army-knife of realtime granular synthesis a flext (cross-platform PD & Max/MSP) port of the munger~ object from the PeRColate library (0.9 beta5) http://www.music.columbia.edu/PeRColate/ Original PeRColate library by: Dan Trueman http://www.music.princeton.edu/~dan/ R. Luke DuBois's http://www.lukedubois.com/ Flext port and additions by: Ivica Ico Bukvic http://ico.bukvic.net Ji-Sun Kim hideaway(a)vt.edu http://www.music.vt.edu http://www.cctad.vt.edu Released under GPL license (whichever is the latest version--as of this release, version 2) For more info on the GPL license please visit: http://www.gnu.org/copyleft/gpl.html ACKNOWLEDGEMENTS ================ Many thanks to Dan Trueman for open-sourcing this great object! SOURCE INSTALL ============== If you simply intend to use prebuilt binaries, please skip to the INSTALL section. Otherwise take a big breath and read on... 1) You need stk library which can be downloaded from: http://ccrma.stanford.edu/software/stk/ 2) You need to also install latest flext library (this is a library that allows for creation of externals for both Max/MSP and PD using the same source). Version 0.4.x can obtained from the following link: http://grrrr.org/ext/flext/ Latest CVS version (0.5.1) is found in the Pure-Data CVS (this one is recommended): http://sourceforge.net/cvs/?group_id=55736 3) If you are using latest CVS version (0.5.1) Before compiling the source you will need to add the following to the top of the flext/source/flstk.h file right below the #define __FLSTK_H: #ifdef PI #undef PI #endif This step will probably become quickly obsolete once Thomas updates CVS. Until then, this is needed to be able to compile flext against stk. 4) To compile flext, read flext instructions (it boils down to running build.sh with appropriate parameters and then editing two simple config files, i.e. "build pd gcc build" or "build max gcc" or "build max msvc" etc.) Your will need to edit buildsys/config-<platform-compiler-pdormax>.txt to adjust paths to various folders. Then you will need to edit config.txt file. You do not need to include SndObj for this external but you do need stk option to be properly set. On Windows+MSVC, STK flag at the time of this release does not work, so you will have to use included testmunger1 MSVC project file and adjust path settings to compile munger1~. 5) Once stk and flext are compiled, go into munger1~ folder and type: <path to flext folder>/build.sh <platform> <compiler> <build/clean/install> NB: on Mac <build/clean/install> is not needed. On Windows, please use MSVC and open the testmunger1 project file in the root of the folder. 6) Once compiled, your binary will be created in a <maxorpd-platform> subfolder (i.e. pd-linux, or max-darwin), followed by another subfolder which reflects whether a threaded or singlethread flext was used. Inside you will find your external. INSTALL ======= You can either use the prebuilt externals (found in the bin/ folder) or ones built using the "SOURCE INSTALL" instructions above. Binaries are provided for Intel-based Macs, Win32, and Intel-based Linux OS. The included prebuilt binaries DO NOT REQUIRE you to install flext or stk as these are statically linked. 1) Copy the external in your externals folder (i.e. /usr/lib/pd/extra or C:\Program Files\Cycling '74\MaxMSP 4.6\Cycling '74\externals\, or "Applications/MaxMSP 4.6/Cycling '74/externals) 2) Copy appropriate help file (found in the help/ folder) into the help folder (i.e. /usr/lib/pd/doc/5.reference or C:\Program Files\Cycling '74\MaxMSP 4.6\max-help, or "Applications/MaxMSP 4.6/max-help) NB: Pd help file has a ".pd" extension, while Max/MSP help file has a ".help" extension. 3) Start your app (PD or Max) and create object called munger1~. Right-click (ctrl-click on Macs) and select "help" and this should open the help file with additional documentation. Questions? See OVERVIEW for contact and Q&A info. Enjoy! FAQ === The following is Ico's FAQ, so it may or may not reflect other project participants' opinions, including original author(s) of munger~, flext, etc. Q: Why porting to flext? A: Flext library (by Thomas Grill) is a layer which allows creation of externals for both Max/MSP and PD without any alterations to the code (obviously once it is adapted to use flext). While there have been a number of Max/MSP <-> PD external ports in the past, many of them have become outdated because such attempts required either maintaining one code full of ugly #ifdefs, or worse--maintaining two sources. Either way, what usually turned out to be the case is that original authors did not have the time, interest, or simply the software/hardware to deal with the newly generated overhead and/or test the code, while volunteers who made the original porting efforts eventually moved on to other projects. The result was/is outdated and/or broken externals. Flext circumvents this problem by allowing one clean code to compile on both platforms while also supplying in many cases cleaner (more legible) API and (as a whipped cream on top) object-oriented environment (C++). Q: Why bother with PD <-> Max/MSP cross-platform compatibility... ...when I use only <insert-your-favorite-application-here>? ...<insert-your-favorite-application-here> is better? A: Choice is what makes us human (this is also what makes Arts so vibrant and exciting). And while everyone's welcome to express their own preferences, we also have to realize that in this case these same preferences are also the main cause of a virtual divide which manifests itself at everyone's detriment. Wouldn't it be nicer if we could share externals transparently, or even better, open PD patches in Max and vice-versa? This would help in both the cross-pollination of ideas as well as creative efforts. This project has also taught me that creating flext-ready externals is as easy if not easier (due to the aforesaid API's legibility) than native objects (whether that be PD or Max/MSP). Finally, if all else fails, such externals are bound to reach wider audience, and are much easier to maintain if cross-platform compatibility is to be pursued. Q: If flext is so cool, why don't we see more porting efforts? A: Good question. The fact is that flext is much more widely known among PD users than it is among the Max/MSP community, so this seemingly one-way road may have contributed to the current situation. One could only hope that projects like this may help reverse this unfortunate trend. Q: So, is all really that peachy in the flext-land? A: Well, our lives teach us that nothing is truly free in this world. Flext is no exception. Its "fees," however are not tied to our checkbooks. Rather, they manifest themselves in a slightly greater CPU overhead in signal flow due to message translation. Thus, one could consider flext a "middle-person" between the <app-of-your-choice> and the external. This, however, in today's world is so negligible that during the testing phase I was unable to measure any noticeable CPU-overhead difference. Another consideration is that flext might not be complete (see KNOWN ISSUES for an example). That being said, in its current state it did the trick for a relatively complex external such as munger~ or even FFTEASE collection which had been ported several years ago. All this leads me to believe that it is more than ready for the day-to-day use. Q: I already have Dan and Luke's awesome PeRColate lib. Why should I download this one? A: This is a cross-platform port of the latest version with several new features. Thus, it allows for those platforms which have not had the beta6 available (Linux, Windows) to finally dig into all the goodies it brings. Plus you also get the cool stuff such as verbose modes, discrete panning, more thorough documentation, up to 500 grains per sample (instead of 50), up to 24-channel output (instead of 2 or 16, depending which one you used), etc. KNOWN ISSUES ============ munger1~ has been tested extensively on Linux+PD, OSX+Max/MSP and Win32+Max/MSP setups, suggesting that it should work on other setups as well. Your mileage may vary, though. Currently there is only one known issue in the wild which requires changes to flext in order to be fixed. Namely, if you use munger1~ object in conjunction with an external buffer in PD (known as an array) and if you dubiously decide to delete that particular buffer in the middle of your performance while munger1~ is still associated with it, this will [unsurprisingly] crash PD. Max/MSP currently has a check implemented against that via flext layer so Max/MSP will simply stop outputting anything until buffer is reset. The flext author is aware of this and PD fix should appear in the flext CVS hopefully soon. That being said, the lingering question is why would you want to do this in the first place... FYI, even though munger1~ allows up to 500 simultaneous grains per sample and has been compiled with all available optimizations (SSE, Altivec is supposedly available via flext but has not been tested), on MBP (Core Duo 1.83GHz) I was unable to get more than 160 simultaneous grains per sample (or ~32,000 grains/second) without dropouts, even though CPUs were not getting maxed out, so something else might be the cause of this limitation (flext?). Win32 machine (3-year old AMD64 3000+) fared marginally better at around 165 simultaneous grains per sample (or ~33,000 grains/second) before its CPU was maxed out. Linux on the same AMD64 3000+ hardware fared the best. It topped off at 47,999 grains per second at 48KHz sampling rate which for some reason the sampling rate appears to be the upper limit (i.e. if you run PD or Max/MSP at lower sampling rates, your upper limit will be restricted to the sampling rate), even though the code allows for multiple initiations of grains per cycle. This, however, is also the way original munger~ works. An interesting bit is that while on Linux/PD combo 48K grains are already reached when we get 64 simultaneous grains, on Win32/Mac even 160 simultaneous grains yield only ~32-33K grains. Could this be a flext bug? Best wishes, Ivica Ico Bukvic, D.M.A. Composition, Music Technology, CCTAD, CHCI Virginia Tech Dept. of Music - 0240 Blacksburg, VA 24061 (540) 231-1137 (540) 231-5034 (fax) ico(a)vt.edu http://www.music.vt.edu/people/faculty/bukvic/

18 years, 11 months

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Linux-audio-dev