From petecrighton at gmail.com Sun Jun 1 01:45:50 2014 From: petecrighton at gmail.com (Peter Crighton) Date: Sun, 1 Jun 2014 03:45:50 +0200 Subject: [LAU] Recommended near-realistic strings section generator? In-Reply-To: References: Message-ID: An update: I had problems with loading VSTs through SAVIHost, which I had recommended earlier, but with FSTHost my professional VST instruments (Garritan Personal Orchestra, some from IK Multimedia, and Arturia Mini V) all load and play perfectly so far. So, should anyone have problems with loading VSTs (I had no luck with Festige) try FSTHost. -- Peter Crighton | Musician & Music Engraver based in Mainz, Germany http://www.petercrighton.de 2014-05-26 6:30 GMT+02:00 Danni Coy : > I have had some luck with Kontact and festige > > On Wed, May 21, 2014 at 5:54 AM, Peter Crighton > wrote: > > I have the great Garritan Personal Orchestra > > (http://www.garritan.com/products/personal-orchestra-4/) and have used > it > > successfully (on 64-bit Arch) ? I haven?t touched it for 1.5 years, and > it > > doesn?t load at the moment, but I?m also on a terribly out-of-date Arch > > version at the moment. It definitely worked beautifully back then. > > It?s not native to Linux, but with the help of SAVIHost > > (http://www.hermannseib.com/savihost.htm) through Wine it will load > > perfectly. Presets can be saved. I think it only allows a minimum of 256 > > frames per period, though. > > Miroslav Philharmonik from IK Multimedia > > (http://www.ikmultimedia.com/products/philharmonik/) works in the same > way, > > but I like GPO a lot more. > > > > > > -- > > Peter Crighton | Musician & Music Engraver based in Mainz, Germany > > http://www.petercrighton.de > > > > > > 2014-05-20 21:01 GMT+02:00 Jonathan E Brickman : > > > >> Thanks! > >> > >> Jonathan E. Brickman > >> Ponderworthy Music | jeb at ponderworthy.com | (785)233-9977 | > >> http://ponderworthy.com > >> > >> > >> > >> > >> > >> > >> ------ Original Message ------ > >> From: "James Stone" > >> To: "Linux Audio Users" > >> Sent: 5/20/2014 1:26:08 AM > >> Subject: Re: [LAU] Recommended near-realistic strings section generator? > >> > >> > >> Maybe loomer if you want synth strings? Otherwise a gig or SFZ -based > >> sample set to load in linuxsampler - free: > >> > >> Sonatina: http://sso.mattiaswestlund.net > >> > >> This thread discusses using cakewalk instruments: > >> > >> http://linuxmusicians.com/viewtopic.php?f=21&t=11323 > >> > >> James > >> > >> On 20 May 2014 03:28, "Jonathan E Brickman" > wrote: > >>> > >>> Right now I'm using Fluidsynth running a soundfont which I customized a > >>> good bit, but it's just not quite what I want; I want it recognizably a > >>> string section, 88-key range, rumbly power in the low, smooth but a > definite > >>> bit of fuzz in mids and highs. I'll take any technology, and will do > >>> payware, as long as it runs well on 64-bit Arch Linux. Anyone got a > >>> recommend? > >>> -- > >>> Jonathan E. Brickman > >>> Ponderworthy Music | jeb at ponderworthy.com | (785)233-9977 | > >>> http://ponderworthy.com > >>> > >>> _______________________________________________ > >>> Linux-audio-user mailing list > >>> Linux-audio-user at lists.linuxaudio.org > >>> http://lists.linuxaudio.org/listinfo/linux-audio-user > >>> > >> > >> _______________________________________________ > >> Linux-audio-user mailing list > >> Linux-audio-user at lists.linuxaudio.org > >> http://lists.linuxaudio.org/listinfo/linux-audio-user > >> > > > > > > _______________________________________________ > > Linux-audio-user mailing list > > Linux-audio-user at lists.linuxaudio.org > > http://lists.linuxaudio.org/listinfo/linux-audio-user > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From rosea.grammostola at gmail.com Sun Jun 1 17:21:44 2014 From: rosea.grammostola at gmail.com (rosea grammostola) Date: Sun, 1 Jun 2014 19:21:44 +0200 Subject: [LAU] octaver (plugin) for bass Message-ID: Hi, Quite a few e-bass players here, seems to use an octaver pedal. Is there some plugin for it on Linux? Regards, ~r -------------- next part -------------- An HTML attachment was scrubbed... URL: From brummer- at web.de Sun Jun 1 17:37:18 2014 From: brummer- at web.de (hermann meyer) Date: Sun, 01 Jun 2014 19:37:18 +0200 Subject: [LAU] octaver (plugin) for bass In-Reply-To: References: Message-ID: <538B64CE.7030605@web.de> Am 01.06.2014 19:21, schrieb rosea grammostola: > Hi, > > Quite a few e-bass players here, seems to use an octaver pedal. Is > there some plugin for it on Linux? > > Regards, > ~r > I've just added such a thing to guitarix (git) as gx and as LV2 plug. It's called GxDetune and could shift one octave up/down. Detune is possible for 1/4 semitone. Latency is ( in High Quality mode) 2048 samples minus jack-frame-size, but could as well set down to Lower rates down to realtime (true, quality get lost then) Latency could be compensate internal for mix with dry signal, as well latency is reported to host (in LV2 version) so that the host could compensate the latency, if supported. greets hermann From brummer- at web.de Sun Jun 1 18:09:00 2014 From: brummer- at web.de (hermann meyer) Date: Sun, 01 Jun 2014 20:09:00 +0200 Subject: [LAU] octaver (plugin) for bass In-Reply-To: <538B64CE.7030605@web.de> References: <538B64CE.7030605@web.de> Message-ID: <538B6C3C.6030608@web.de> Am 01.06.2014 19:37, schrieb hermann meyer: > Am 01.06.2014 19:21, schrieb rosea grammostola: >> Hi, >> >> Quite a few e-bass players here, seems to use an octaver pedal. Is >> there some plugin for it on Linux? >> >> Regards, >> ~r >> > > I've just added such a thing to guitarix (git) as gx and as LV2 plug. > It's called GxDetune and could shift one octave up/down. Detune is > possible for 1/4 semitone. Latency is ( in High Quality mode) 2048 > samples minus jack-frame-size, but could as well set down to Lower > rates down to realtime (true, quality get lost then) > Latency could be compensate internal for mix with dry signal, as well > latency is reported to host (in LV2 version) so that the host could > compensate the latency, if supported. > > greets > hermann Some more information here: http://www.linuxmusicians.com/viewtopic.php?f=48&t=12300&start=15#p52208 From rosea.grammostola at gmail.com Sun Jun 1 18:14:06 2014 From: rosea.grammostola at gmail.com (rosea grammostola) Date: Sun, 1 Jun 2014 20:14:06 +0200 Subject: [LAU] octaver (plugin) for bass In-Reply-To: <538B6C3C.6030608@web.de> References: <538B64CE.7030605@web.de> <538B6C3C.6030608@web.de> Message-ID: The fun of these pedals is that it *adds* an octave (higher or lower) to the tone being played http://www.jimdunlop.com/blog/new-bass-octave-deluxe-demo/ On Sun, Jun 1, 2014 at 8:09 PM, hermann meyer wrote: > Am 01.06.2014 19:37, schrieb hermann meyer: > > Am 01.06.2014 19:21, schrieb rosea grammostola: >> >>> Hi, >>> >>> Quite a few e-bass players here, seems to use an octaver pedal. Is there >>> some plugin for it on Linux? >>> >>> Regards, >>> ~r >>> >>> >> I've just added such a thing to guitarix (git) as gx and as LV2 plug. >> It's called GxDetune and could shift one octave up/down. Detune is possible >> for 1/4 semitone. Latency is ( in High Quality mode) 2048 samples minus >> jack-frame-size, but could as well set down to Lower rates down to realtime >> (true, quality get lost then) >> Latency could be compensate internal for mix with dry signal, as well >> latency is reported to host (in LV2 version) so that the host could >> compensate the latency, if supported. >> >> greets >> hermann >> > > Some more information here: > > http://www.linuxmusicians.com/viewtopic.php?f=48&t=12300&start=15#p52208 > -------------- next part -------------- An HTML attachment was scrubbed... URL: From brummer- at web.de Sun Jun 1 18:30:59 2014 From: brummer- at web.de (hermann meyer) Date: Sun, 01 Jun 2014 20:30:59 +0200 Subject: [LAU] octaver (plugin) for bass In-Reply-To: References: <538B64CE.7030605@web.de> <538B6C3C.6030608@web.de> Message-ID: <538B7163.6070102@web.de> Okay, so top posting is requested here? Yea, that's what it does actual. you can *adds* a octave up/down with GxDetune. You can mixed dry / wet. (seperate controllers for dry and wet signal from 0 - 100 % for each). You can delay the added octave for the latency, or you can compensate it internal, so that dry and wet signal comes in sync. screenshot: http://oi62.tinypic.com/34njmo3.jpg Am 01.06.2014 20:14, schrieb rosea grammostola: > The fun of these pedals is that it *adds* an octave (higher or lower) > to the tone being played > http://www.jimdunlop.com/blog/new-bass-octave-deluxe-demo/ > > > On Sun, Jun 1, 2014 at 8:09 PM, hermann meyer > wrote: > > Am 01.06.2014 19:37, schrieb hermann meyer: > > Am 01.06.2014 19:21, schrieb rosea grammostola: > > Hi, > > Quite a few e-bass players here, seems to use an octaver > pedal. Is there some plugin for it on Linux? > > Regards, > ~r > > > I've just added such a thing to guitarix (git) as gx and as > LV2 plug. It's called GxDetune and could shift one octave > up/down. Detune is possible for 1/4 semitone. Latency is ( in > High Quality mode) 2048 samples minus jack-frame-size, but > could as well set down to Lower rates down to realtime (true, > quality get lost then) > Latency could be compensate internal for mix with dry signal, > as well latency is reported to host (in LV2 version) so that > the host could compensate the latency, if supported. > > greets > hermann > > > Some more information here: > > http://www.linuxmusicians.com/viewtopic.php?f=48&t=12300&start=15#p52208 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bruviaro at scu.edu Mon Jun 2 00:35:06 2014 From: bruviaro at scu.edu (Bruno Ruviaro) Date: Sun, 1 Jun 2014 17:35:06 -0700 Subject: [LAU] Session Management In-Reply-To: References: <20140529180215.GM13263@silverninja.net> Message-ID: On Fri, May 30, 2014 at 11:47 PM, Harry van Haaren wrote: > I would love to see NSM be used as "the session management" system, and > I've offered > to assist other developers in implementing NSM: > http://permalink.gmane.org/gmane.linux.audio.devel/30699 > > I'm currently working with the Hydrogen project to make Hydrogen > NSM-capable: see http://openavproductions.com/news/ for a screenshot. > > Cheers, -Harry > > ?Just wanted to second Harry that I too would love to see NSM be used as THE widespread session management system. In my (limited) experience with session managers, I am very happy with it so far. Hydrogen supporting NSM will be awesome. Next one in my dream-list would be SooperLooper! ?Bruno? -------------- next part -------------- An HTML attachment was scrubbed... URL: From rosea.grammostola at gmail.com Mon Jun 2 14:04:44 2014 From: rosea.grammostola at gmail.com (rosea grammostola) Date: Mon, 2 Jun 2014 16:04:44 +0200 Subject: [LAU] octaver (plugin) for bass In-Reply-To: <538B7163.6070102@web.de> References: <538B64CE.7030605@web.de> <538B6C3C.6030608@web.de> <538B7163.6070102@web.de> Message-ID: Thanks. Seems to work ok (all though Carla doesn't find the plugin afaik). Numbers on the knobs would be nice. I've no idea how much I detune the tone now for example. -------------- next part -------------- An HTML attachment was scrubbed... URL: From blablack at gmail.com Tue Jun 3 16:27:46 2014 From: blablack at gmail.com (=?UTF-8?Q?Aur=C3=A9lien_Leblond?=) Date: Tue, 3 Jun 2014 17:27:46 +0100 Subject: [LAU] Session Management In-Reply-To: References: <20140529180215.GM13263@silverninja.net> Message-ID: I tried NSM quickly - can it deal at a simple level with applications that are not directly compatible with NSM? If i use gladish as an example, I can save any jack connections (which is really what I'm interrested here). On Mon, Jun 2, 2014 at 1:35 AM, Bruno Ruviaro wrote: > On Fri, May 30, 2014 at 11:47 PM, Harry van Haaren > wrote: >> >> I would love to see NSM be used as "the session management" system, and >> I've offered >> to assist other developers in implementing NSM: >> http://permalink.gmane.org/gmane.linux.audio.devel/30699 >> >> I'm currently working with the Hydrogen project to make Hydrogen >> NSM-capable: see http://openavproductions.com/news/ for a screenshot. >> >> Cheers, -Harry >> > > Just wanted to second Harry that I too would love to see NSM be used as THE > widespread session management system. In my (limited) experience with > session managers, I am very happy with it so far. > > Hydrogen supporting NSM will be awesome. Next one in my dream-list would be > SooperLooper! > > Bruno > > > > From harryhaaren at gmail.com Tue Jun 3 16:34:25 2014 From: harryhaaren at gmail.com (Harry van Haaren) Date: Tue, 3 Jun 2014 17:34:25 +0100 Subject: [LAU] Session Management In-Reply-To: References: <20140529180215.GM13263@silverninja.net> Message-ID: Yes: There's the "NSM Proxy" client, which can launch a program with command-line arguments. The jackpatch program included in NSM saves/restores JACK connections. HTH, -Harry On Tue, Jun 3, 2014 at 5:27 PM, Aur?lien Leblond wrote: > I tried NSM quickly - can it deal at a simple level with applications > that are not directly compatible with NSM? > > If i use gladish as an example, I can save any jack connections (which > is really what I'm interrested here). > > On Mon, Jun 2, 2014 at 1:35 AM, Bruno Ruviaro wrote: > > On Fri, May 30, 2014 at 11:47 PM, Harry van Haaren < > harryhaaren at gmail.com> > > wrote: > >> > >> I would love to see NSM be used as "the session management" system, and > >> I've offered > >> to assist other developers in implementing NSM: > >> http://permalink.gmane.org/gmane.linux.audio.devel/30699 > >> > >> I'm currently working with the Hydrogen project to make Hydrogen > >> NSM-capable: see http://openavproductions.com/news/ for a screenshot. > >> > >> Cheers, -Harry > >> > > > > Just wanted to second Harry that I too would love to see NSM be used as > THE > > widespread session management system. In my (limited) experience with > > session managers, I am very happy with it so far. > > > > Hydrogen supporting NSM will be awesome. Next one in my dream-list would > be > > SooperLooper! > > > > Bruno > > > > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bruviaro at scu.edu Tue Jun 3 19:34:55 2014 From: bruviaro at scu.edu (Bruno Ruviaro) Date: Tue, 3 Jun 2014 12:34:55 -0700 Subject: [LAU] Women in Computer Music - SuperCollider Workshop Scholarship Message-ID: Application deadline is June 10. Please spread the word if you know anyone who might be interested. https://ccrma.stanford.edu/women-in-computer-music-supercollider-workshop-scholarship This scholarship was created to encourage young women to engage with the field of computer music through SuperCollider. The workshop itself is open to all genders of course: https://ccrma.stanford.edu/workshops/supercollider-101 Best, Bruno -------------- next part -------------- An HTML attachment was scrubbed... URL: From rob at rektau.ukfsn.org Tue Jun 3 21:14:28 2014 From: rob at rektau.ukfsn.org (rob) Date: Tue, 03 Jun 2014 22:14:28 +0100 Subject: [LAU] octaver (plugin) for bass In-Reply-To: References: <538B64CE.7030605@web.de> <538B6C3C.6030608@web.de> <538B7163.6070102@web.de> Message-ID: <538E3AB4.9090403@rektau.ukfsn.org> On 02/06/14 15:04, rosea grammostola wrote: > Numbers on the knobs would be nice. I've no idea how much I detune the > tone now for example. > Options -> Show Values rob From rosea.grammostola at gmail.com Wed Jun 4 13:10:02 2014 From: rosea.grammostola at gmail.com (rosea grammostola) Date: Wed, 4 Jun 2014 15:10:02 +0200 Subject: [LAU] Session Management In-Reply-To: References: <20140529180215.GM13263@silverninja.net> Message-ID: yep, you need to add the clients: jackpatch (for (re)store connections) nsm-proxy (for command-line options) On Tue, Jun 3, 2014 at 6:34 PM, Harry van Haaren wrote: > Yes: There's the "NSM Proxy" client, which can launch a program with > command-line arguments. > > The jackpatch program included in NSM saves/restores JACK connections. > > HTH, -Harry > > > On Tue, Jun 3, 2014 at 5:27 PM, Aur?lien Leblond > wrote: > >> I tried NSM quickly - can it deal at a simple level with applications >> that are not directly compatible with NSM? >> >> If i use gladish as an example, I can save any jack connections (which >> is really what I'm interrested here). >> >> On Mon, Jun 2, 2014 at 1:35 AM, Bruno Ruviaro wrote: >> > On Fri, May 30, 2014 at 11:47 PM, Harry van Haaren < >> harryhaaren at gmail.com> >> > wrote: >> >> >> >> I would love to see NSM be used as "the session management" system, and >> >> I've offered >> >> to assist other developers in implementing NSM: >> >> http://permalink.gmane.org/gmane.linux.audio.devel/30699 >> >> >> >> I'm currently working with the Hydrogen project to make Hydrogen >> >> NSM-capable: see http://openavproductions.com/news/ for a screenshot. >> >> >> >> Cheers, -Harry >> >> >> > >> > Just wanted to second Harry that I too would love to see NSM be used as >> THE >> > widespread session management system. In my (limited) experience with >> > session managers, I am very happy with it so far. >> > >> > Hydrogen supporting NSM will be awesome. Next one in my dream-list >> would be >> > SooperLooper! >> > >> > Bruno >> > >> > >> > >> > >> > > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From paul at linuxaudiosystems.com Thu Jun 5 13:08:42 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Thu, 5 Jun 2014 09:08:42 -0400 Subject: [LAU] [LAD] Standalone resources browser - mockup, near of publishing, need discussion In-Reply-To: <1583613.k0EgnoIbnv@nick87720z> References: <1583613.k0EgnoIbnv@nick87720z> Message-ID: On Thu, Jun 5, 2014 at 7:25 AM, Zlobin Nikita wrote: > > As for plugins: currently each audio application has own plugin browser. In > some cases it is very handy (ingen, carla), sometimes awful (audacity, may > be > more). My hope is to make it external with filemanager-like workflow. > then you'd better be considering cases where the host has its own reasons to filter the list presented to the user. > Dragging to jack patchbay canvas may be used to load plugin in single host > (not sure, what about instrument, since there is usually separate sampler > engine, usually managing all instruments). > indeed. > Not implemented: > Just have to learn VAMP and somehow try VST - for some completeness. > VAMP is not a realtime plugin API. VAMP plugins are for feature analysis not audio processing. -------------- next part -------------- An HTML attachment was scrubbed... URL: From list at nilsgey.de Thu Jun 5 16:35:09 2014 From: list at nilsgey.de (Nils) Date: Thu, 05 Jun 2014 18:35:09 +0200 Subject: [LAU] Open Source Audio Meeting in Cologne, Germany (User Group) Message-ID: <53909C3D.30106@nilsgey.de> Dear musicians, programmers and normal people, I plan to create a user group, a regular meeting, in my home town Cologne, in Germany. The first meeting ever will be already on June 18th, 19:00. After that every two month or so. There are more dates on the website (see below). Here is a brief website with all the necessary information. http://cologne.linuxaudio.org/ (Any language is welcome but the chances are that most people will be from the area and therefore speak German. So the page is in German) If you intend to come you can put your name on this etherpad, but this is not required. Anybody can show up. http://yourpart.eu/p/linuxaudio-cologne Topics will be unorganized Q&A, showing off programs and music, sharing knowledge and tips and hopefully one day shorter or longer presentations, tutorials, workshops etc. I expect most people to use Linux but any OS is welcome, therefore I named it just "Open Source Audio" and not Linux Audio. So if you are in the area please join us! If you are not in the area but know people in the area, please tell them. Greetings, Nils http://cologne.linuxaudio.org/ http://www.nilsgey.de P.S. Despite the domain saying linuxaudio.org this is an independently and privately organized event. It is not intended to replace or get in conflict with the Linux Audio Conference. From gg3137 at vegri.net Thu Jun 5 21:44:54 2014 From: gg3137 at vegri.net (Giso Grimm) Date: Thu, 05 Jun 2014 23:44:54 +0200 Subject: [LAU] linux powered spatial audio concert for viol ensemble Message-ID: <5390E4D6.2010405@vegri.net> Dear Linux Audio enthusiasts! I would like to announce a concert with contemporary and early music, performed on five viola da gamba and life electronics: Harmony of the Spheres June 14th, 2014 7.30pm Malory/Santana/Stockhausen/Lawes/Cage/Tye 10pm Palestrina/Picforth/Stockhausen/Bach/?/Strogers 11.30pm Picforth/Santana/Stockhausen/Cage/Bach The concert is part of a contemporary music festival (Long night of music) in Oldenburg, Germany. Free admission. In the concert the acoustic instruments are spatially processed in real-time. Sounds of the street are analysed to control real-time composition in one piece. The tools involved are ambdec, ardour, jconvolver, tetraproc, and a bunch of tools developed specifically for this concert (partly presented at LAC2012): https://github.com/gisogrimm/hos-toolbox (completely undocumented) https://github.com/gisogrimm/tascar (very limited documentation) More infos on the programme can be found here: http://hos.orlandoviols.de/ -- Giso From el.doctor at laposte.net Fri Jun 6 03:53:09 2014 From: el.doctor at laposte.net (Manu Kebab) Date: Fri, 06 Jun 2014 05:53:09 +0200 Subject: [LAU] Yet another "io GNU/Linux" iso released Message-ID: <1782105.ohqO6NtJuQ@io> Hi, A new 64bit iso is up ;) io GNU/Linux is a Live DVD/USB based on Debian Sid and focused on multimedia. Kernel 3.14.4, Jack2 as default sound server, e18 as desktop environment and a big collection of installed software... Full persistence for USB install (with encryption) and more cool stuff... A great nomade studio :) For more infos: manual, packages list, screenshots, video etc... Check: -> http://manu.kebab.free.fr/iognulinux.html -> https://sourceforge.net/projects/io-gnu-linux/ Feedbacks welcome, enjoy :) MK -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 198 bytes Desc: This is a digitally signed message part. URL: From brummer- at web.de Fri Jun 6 10:43:48 2014 From: brummer- at web.de (hermann meyer) Date: Fri, 06 Jun 2014 12:43:48 +0200 Subject: [LAU] octaver (plugin) for bass In-Reply-To: References: <538B64CE.7030605@web.de> <538B6C3C.6030608@web.de> <538B7163.6070102@web.de> Message-ID: <53919B64.6040706@web.de> Am 02.06.2014 16:04, schrieb rosea grammostola: > Thanks. Seems to work ok (all though Carla doesn't find the plugin afaik). > I test all guitarix plugs with Ardour, Qtractor and of-course jalv. All guitarix plugs load and run well in those 3 hosts. As well do GxDetune. So I've no idea what the problem with carla could be, nor what I could do about that. > Numbers on the knobs would be nice. I've no idea how much I detune the > tone now for example. > Right click on a controller will open the num entry, you can set value by keyboard or mouse-wheel. You could even use the internal UI from ardour or qtractor which will show controllers with values. From rosea.grammostola at gmail.com Fri Jun 6 18:36:09 2014 From: rosea.grammostola at gmail.com (rosea grammostola) Date: Fri, 6 Jun 2014 20:36:09 +0200 Subject: [LAU] octaver (plugin) for bass In-Reply-To: <53919B64.6040706@web.de> References: <538B64CE.7030605@web.de> <538B6C3C.6030608@web.de> <538B7163.6070102@web.de> <53919B64.6040706@web.de> Message-ID: Must be said, I've a lot of fun using this plugin (in Carla git version), thanks! On Fri, Jun 6, 2014 at 12:43 PM, hermann meyer wrote: > Am 02.06.2014 16:04, schrieb rosea grammostola: > > Thanks. Seems to work ok (all though Carla doesn't find the plugin afaik). >> >> I test all guitarix plugs with Ardour, Qtractor and of-course jalv. All > guitarix plugs load and run well in those 3 hosts. As well do GxDetune. So > I've no idea what the problem with carla could be, nor what I could do > about that. > > > Numbers on the knobs would be nice. I've no idea how much I detune the >> tone now for example. >> >> Right click on a controller will open the num entry, you can set value > by keyboard or mouse-wheel. > You could even use the internal UI from ardour or qtractor which will show > controllers with values. > -------------- next part -------------- An HTML attachment was scrubbed... URL: From el.doctor at laposte.net Sat Jun 7 00:57:58 2014 From: el.doctor at laposte.net (Manu Kebab) Date: Sat, 07 Jun 2014 02:57:58 +0200 Subject: [LAU] Yet another "io GNU/Linux" iso released In-Reply-To: <1782105.ohqO6NtJuQ@io> References: <1782105.ohqO6NtJuQ@io> Message-ID: <2386408.q0oAuvFt4X@io> Le vendredi 6 juin 2014, 05:53:09 Manu Kebab a ?crit : > Hi, > > A new 64bit iso is up ;) > > io GNU/Linux is a Live DVD/USB based on Debian Sid and focused on > multimedia. > > Kernel 3.14.4, Jack2 as default sound server, e18 as desktop environment and > a big collection of installed software... Full persistence for USB install > (with encryption) and more cool stuff... A great nomade studio :) > > For more infos: manual, packages list, screenshots, video etc... Check: > > -> http://manu.kebab.free.fr/iognulinux.html > -> https://sourceforge.net/projects/io-gnu-linux/ > > > Feedbacks welcome, enjoy :) > > MK Update with kernel 3.14.5 Greetingz :) MK -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 198 bytes Desc: This is a digitally signed message part. URL: From brummer- at web.de Sat Jun 7 08:15:41 2014 From: brummer- at web.de (hermann meyer) Date: Sat, 07 Jun 2014 10:15:41 +0200 Subject: [LAU] octaver (plugin) for bass In-Reply-To: References: <538B64CE.7030605@web.de> <538B6C3C.6030608@web.de> <538B7163.6070102@web.de> <53919B64.6040706@web.de> Message-ID: <5392CA2D.5070104@web.de> Am 06.06.2014 20:36, schrieb rosea grammostola: > Must be said, I've a lot of fun using this plugin (in Carla git > version), thanks! Thanks have to go to Stephan M. Bernsee from dspdimension as well. GxDetune is based on his work here: http://www.dspdimension.com/admin/pitch-shifting-using-the-ft/ True, I've heavy modified the original source, but the underlying algorithm comes from Stephan. From csanchezgs at gmail.com Sat Jun 7 12:39:37 2014 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Sat, 7 Jun 2014 14:39:37 +0200 Subject: [LAU] RT kern for Raspberry Pi? In-Reply-To: <53770788.2010406@autostatic.com> References: <20140516043944.GA1361@tf101> <5375C0AD.7010806@autostatic.com> <20140516235105.GA3058@q400a.mobile.restivo.org> <53770788.2010406@autostatic.com> Message-ID: 2014-05-17 8:54 GMT+02:00 Jeremy Jongepier : > On 05/17/2014 01:51 AM, Ken Restivo wrote: >> On Fri, May 16, 2014 at 09:39:25AM +0200, Jeremy Jongepier wrote: >>> On 05/16/2014 06:39 AM, Ken Restivo wrote: >>>> Has anyone built one, i.e. off of latest Raspbian release, and made it available as a deb? >>>> >>>> For a while I thought about building one, but I kind of lost interest. Now feeling lazy and wondering if someone's already done it. >>> >>> Hello Ken, >>> >>> I have built RT kernels for Raspbian. No debs though :( I could upload a >>> tarball with the kernel and modules if you'd like. For what it's worth, >>> I have better results with the standard PREEMPT Raspbian kernel so I >>> actually never use RT kernels on the RPi besides doing some testing. >>> >> >> Thanks, yeah, it'd be cool to try them out. > > When I have something wrapped up I'll post a link. > >> >> You're running jackd and softsynths with PREEMPT? Really? I've never had that work right ever. Always needs RT to get the latency to a playable state with no Xruns, at least on Intel which is all I've tried so far. >> > > The Raspberry Pi is a different beast and I just get better results with > the default PREEMPT kernel and some tweaking > (http://wiki.linuxaudio.org/wiki/raspberrypi). I didn't really do any > serious testing though, it was just that the RT kernels generated more > xruns so I quickly went back to the default kernel. > > Jeremy > > > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > Ken, I also use the Raspbian distro and kernel with a few tweaks to the Raspberry config (Jeremy has great documentation on this) and that's it. It serves me well i.e. as a guitarr FX + looper station, controlled via MIDI footpedalboard and this mobile panel/control GUI interface that I'm developing so I don't need any screen attached to the RPi with. I also use an external USB sound card. So you can start right away and then tweak as you go. -- C. sanchiavedraZ: * NEW / NUEVO: www.sanchiavedraZ.com * Musix GNU+Linux: www.musix.es From fons at linuxaudio.org Sat Jun 7 16:08:24 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sat, 7 Jun 2014 16:08:24 +0000 Subject: [LAU] octaver (plugin) for bass In-Reply-To: <5392CA2D.5070104@web.de> References: <538B64CE.7030605@web.de> <538B6C3C.6030608@web.de> <538B7163.6070102@web.de> <53919B64.6040706@web.de> <5392CA2D.5070104@web.de> Message-ID: <20140607160824.GA27886@linuxaudio.org> On Sat, Jun 07, 2014 at 10:15:41AM +0200, hermann meyer wrote: > Thanks have to go to Stephan M. Bernsee from dspdimension as well. > GxDetune is based on his work here: > http://www.dspdimension.com/admin/pitch-shifting-using-the-ft/ This sort of works, but it's not what it claims to be. The whole part that finds the exact frequency by comparing phases is completely redundant. This information is never really used. It just looks as if it is used. For example, for one octave up, you could just as well take the magnitude and phase of bin k, multiply the phase by 2 and put the result in the input bin 2*k of the inverse FFT. The result would be just the same. No frequency calculation is ever made. The net result is also equivalent to: - overlap - windowing (as in your code) but then: - downsample by 2 - repeat the result so you get the original length - add to output Which doesn't even require an FFT. The way to really use the computed frequencies would be quite different. If you have a signal at some frequency F there will be significant energy in a number of bins close to F. The correct value of F can be found by comparing the phases as explained by Bernsee. Given this F you need some way to determine which contiguous group of bins is representative of that signal (one way would be to look for minima in magnitude left and right). Now for correct frequency scaling, you need to move that whole group up or down (as determined by the ratio, e.g. 2 for one octave up) *** but without scaling the group itself ***. In other words, if bin k moves to 2*k, then bin k-1 moves to 2*k-1 etc. This requires an *interpretation* of the signal: do bins that are close together 1. represent a single frequency signal, or 2. multiple signals that are close together. In case (1) the envelope of the signal is represented by the relative magitudes and phases of the adjacent bins. To preserve this envolope (i.e. to correctly reproduce transient signals), these bins need to remain adjacent. Another way to state this that any algorithm that does frequency scaling (or time stretching) needs some way to decide if certain features of the signal need to be interpreted as significant in the time domain or in the frequency domain. The correct decision depends on how a human listener would interpret that feature. It is not even possible to *define* a frequency scaling or time stretching algorithm without at least implicitly defining a way to decide on this. The implicit assumption in the current algorithm is that each bin is an separate feature in the frequency domain, and thus needs to be scaled independently of all others. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From csanchezgs at gmail.com Sat Jun 7 17:03:00 2014 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Sat, 7 Jun 2014 19:03:00 +0200 Subject: [LAU] Social Media In-Reply-To: <4E3E39E6.3060303@hawaii.rr.com> References: <4e37ef67.a701440a.7356.ffffd598SMTPIN_ADDED@mx.google.com> <4E3B072D.5080704@smoors.de> <4E3B1627.9070404@gmail.com> <4E3BD019.9090902@autostatic.com> <4E3D0CB7.8080008@autostatic.com> <4E3D9C23.6080300@hawaii.rr.com> <4E3E25BE.8020500@yahoo.fr> <4E3E39E6.3060303@hawaii.rr.com> Message-ID: 2011-08-07 9:08 GMT+02:00 david : > fred wrote: >> >> >> >> Le 06/08/2011 21:55, david a ?crit : >>> >>> Jeremy Jongepier wrote: >>>> >>>> On 08/06/2011 07:09 AM, Alexandre Prokoudine wrote: >>>>> >>>>> On Fri, Aug 5, 2011 at 3:12 PM, Jeremy Jongepier wrote: >>>>> >>>>>> My 2?: social media don't work for Linux Audio. I think the only thing >>>>>> that >>>>>> does work is good video tutorials, that's really big at the moment, >>>>>> good >>>>>> blogs and decent articles on authorative sites and in printed >>>>>> magazines. >>>>> >>>>> >>>>> I think you really want to expand your understanding of what social >>>>> media is. All of the above except printed magazines is part of it :) >>>>> >>>> >>>> Hello Alexandre, >>>> >>>> You're right, I'm confusing social media and social networking services. >>>> I meant the latter. >>>> >>>>>> the other things are imho not useful. Twitter, Facebook, Google+ won't >>>>>> work, >>>>>> simply too much dispersion, people don't collaborate on these >>>>>> platforms, >>>>>> they only click on buttons and leave pointless comments. >>>>> >>>>> >>>>> That's quite an exxageration. >>>> >>>> >>>> I know. I've already replied to Rosea where that originates from. Next >>>> time I'll keep the whole picture in mind :) >>>> >>>> I do some techsupport for Inkscape, GIMP >>>>> >>>>> and Scribus via Twitter. It's actually useful for helping people solve >>>>> simple issues. That's not a direct marketing, but it helps preserving >>>>> user base. Of course, one could go beyond that. >>>>> >>>>>> I call BS, with the band I convinced the others to ditch Cubase in >>>>>> favor of >>>>>> Qtractor because every rehearsal session we were totally lost again on >>>>>> how >>>>>> to record a simple track. >>>>> >>>>> >>>>> You mean you didn't know how to use Cubase? :) >>>> >>>> >>>> Ha ha, yeah, that's true. We even borrowed a 300 page Cubase book from a >>>> friend but still we didn't manage. >>> >>> >>> Well, for comparison, I once tried to make a simple stereo recording >>> using Ardour, and was completely unable to figure it out. But pro-level >>> software is complex and assumes that its user has a lot of domain-specific >>> knowledge (is an experienced audio engineer). I don't fault the UI for that! >>> >> Ardour is a really step above Cubase... > > > I'm not the one who tried to use Cubase, so I'll take your word for it. > > > -- > David > gnome at hawaii.rr.com > authenticity, honesty, community > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user Hello dear all. First, my apologies for rescuing this thread from the sands of time of LAU. But I bumped into this working on some of my projects I'm struggling to bring to the (web)light at last, and those words from Jeremy and your responses totally resonate with me, and the concept seems to be present nowadays even when this thread goes back to 2011. I just wanted to let go some words about the subject. I think Tools don't make the Artist. But it's clear that it's really helpful when a really well known artist or a great mixer/producer shows up in a video using tool X (multifx, plugin, daw...). Because if they are good at what they do it usually seems it's so easy to do Y (play, mix, record...) with that tool that the tool itself benefits from that perception, and people think that is the tool that's making them do what they do. And big enterprises, of course, take advantage of this, along with beautiful eye-candy graphical interfaces. So if there is well produced quality material that demonstrate how to get an amazing guitar FX, how to easily record yourself in you're room, how to mix you're band... then people might get the seminal idea of "hey, this works", and after that maybe they might be interested in going a little deeper and getting to know that those tools also take care of their rights and freedom and all that philosophy we are already aware of. That is critical to be aware of: Humans hate or are not comfortable with changes, and non tech-savvy people usually just care about if it "just works", and much better if it has a nice GUI; and moreover if there's some neighbor/friend that can make them a copy of the tool and teach them how to use it. Word of mouth is of course invaluable. Modestly, I still trying to do what I can from various perspectives, and hopefully I'll be taking another further steps to give at least just a little back of all that I receive from so many talented people that free their knowledge... and their software and projects as well. It seems that even all of us along with many people in the FLOSS community still trying to go against the current, the established, the easy way, there's way to go. But, again, it feels that nowadays, given so many changes that are happening, there's another chance to make a shift. Still trying to think global and act local, and still trying to put my modest and tiny two cents. Sorry for that lengthy spiel. Thanks you all. -- C. sanchiavedraZ: * NEW / NUEVO: www.sanchiavedraZ.com * Musix GNU+Linux: www.musix.es From brummer- at web.de Sat Jun 7 17:10:21 2014 From: brummer- at web.de (hermann meyer) Date: Sat, 07 Jun 2014 19:10:21 +0200 Subject: [LAU] octaver (plugin) for bass In-Reply-To: <20140607160824.GA27886@linuxaudio.org> References: <538B64CE.7030605@web.de> <538B6C3C.6030608@web.de> <538B7163.6070102@web.de> <53919B64.6040706@web.de> <5392CA2D.5070104@web.de> <20140607160824.GA27886@linuxaudio.org> Message-ID: <5393477D.9020801@web.de> Am 07.06.2014 18:08, schrieb Fons Adriaensen: > On Sat, Jun 07, 2014 at 10:15:41AM +0200, hermann meyer wrote: > >> Thanks have to go to Stephan M. Bernsee from dspdimension as well. >> GxDetune is based on his work here: >> http://www.dspdimension.com/admin/pitch-shifting-using-the-ft/ > This sort of works, but it's not what it claims to be. Do you talk about my reworked code here: http://sourceforge.net/p/guitarix/git/ci/master/tree/trunk/src/LV2/gx_detune.lv2/detune.cc > > The whole part that finds the exact frequency by comparing > phases is completely redundant. This information is never > really used. It just looks as if it is used. But without the accumulation on the phase the down shift simply sounds shitty. It work pretty well, with my rework it use a dsp load from 2-3% were it use previous in the original version around 50% I would say, it is the best sounding octave (pitch) shifter we have in open source, > For example, for one octave up, you could just as well take > the magnitude and phase of bin k, multiply the phase by 2 and > put the result in the input bin 2*k of the inverse FFT. The > result would be just the same. No frequency calculation is > ever made. > > The net result is also equivalent to: > > - overlap > - windowing > (as in your code) but then: > > - downsample by 2 > - repeat the result so you get the original length > - add to output > > Which doesn't even require an FFT. > > The way to really use the computed frequencies would > be quite different. > > If you have a signal at some frequency F there will > be significant energy in a number of bins close to F. > The correct value of F can be found by comparing the > phases as explained by Bernsee. Given this F you need > some way to determine which contiguous group of bins > is representative of that signal (one way would be to > look for minima in magnitude left and right). > Now for correct frequency scaling, you need to move > that whole group up or down (as determined by the ratio, > e.g. 2 for one octave up) *** but without scaling the > group itself ***. In other words, if bin k moves to 2*k, > then bin k-1 moves to 2*k-1 etc. > > This requires an *interpretation* of the signal: do bins > that are close together > > 1. represent a single frequency signal, or > 2. multiple signals that are close together. > > In case (1) the envelope of the signal is represented by > the relative magitudes and phases of the adjacent bins. > To preserve this envolope (i.e. to correctly reproduce > transient signals), these bins need to remain adjacent. > > Another way to state this that any algorithm that does > frequency scaling (or time stretching) needs some way > to decide if certain features of the signal need to be > interpreted as significant in the time domain or in the > frequency domain. The correct decision depends on how > a human listener would interpret that feature. > > It is not even possible to *define* a frequency scaling > or time stretching algorithm without at least implicitly > defining a way to decide on this. > > The implicit assumption in the current algorithm is that > each bin is an separate feature in the frequency domain, > and thus needs to be scaled independently of all others. > > > Ciao, > From brummer- at web.de Sat Jun 7 17:43:53 2014 From: brummer- at web.de (hermann meyer) Date: Sat, 07 Jun 2014 19:43:53 +0200 Subject: [LAU] octaver (plugin) for bass In-Reply-To: <5393477D.9020801@web.de> References: <538B64CE.7030605@web.de> <538B6C3C.6030608@web.de> <538B7163.6070102@web.de> <53919B64.6040706@web.de> <5392CA2D.5070104@web.de> <20140607160824.GA27886@linuxaudio.org> <5393477D.9020801@web.de> Message-ID: <53934F59.7040906@web.de> Am 07.06.2014 19:10, schrieb hermann meyer: > Am 07.06.2014 18:08, schrieb Fons Adriaensen: >> On Sat, Jun 07, 2014 at 10:15:41AM +0200, hermann meyer wrote: >> >>> Thanks have to go to Stephan M. Bernsee from dspdimension as well. >>> GxDetune is based on his work here: >>> http://www.dspdimension.com/admin/pitch-shifting-using-the-ft/ >> This sort of works, but it's not what it claims to be. > > Do you talk about my reworked code here: > http://sourceforge.net/p/guitarix/git/ci/master/tree/trunk/src/LV2/gx_detune.lv2/detune.cc > >> >> The whole part that finds the exact frequency by comparing >> phases is completely redundant. This information is never >> really used. It just looks as if it is used. > > But without the accumulation on the phase the down shift simply sounds > shitty. > It work pretty well, with my rework it use a dsp load from 2-3% were > it use previous in the original version around 50% > I would say, it is the best sounding octave (pitch) shifter we have in > open source, > >> For example, for one octave up, you could just as well take >> the magnitude and phase of bin k, multiply the phase by 2 and >> put the result in the input bin 2*k of the inverse FFT. The >> result would be just the same. No frequency calculation is >> ever made. >> >> The net result is also equivalent to: >> >> - overlap >> - windowing >> (as in your code) but then: >> >> - downsample by 2 >> - repeat the result so you get the original length >> - add to output >> >> Which doesn't even require an FFT. >> >> The way to really use the computed frequencies would >> be quite different. >> >> If you have a signal at some frequency F there will >> be significant energy in a number of bins close to F. >> The correct value of F can be found by comparing the >> phases as explained by Bernsee. Given this F you need >> some way to determine which contiguous group of bins >> is representative of that signal (one way would be to >> look for minima in magnitude left and right). >> Now for correct frequency scaling, you need to move >> that whole group up or down (as determined by the ratio, >> e.g. 2 for one octave up) *** but without scaling the >> group itself ***. In other words, if bin k moves to 2*k, >> then bin k-1 moves to 2*k-1 etc. >> >> This requires an *interpretation* of the signal: do bins >> that are close together >> >> 1. represent a single frequency signal, or >> 2. multiple signals that are close together. >> >> In case (1) the envelope of the signal is represented by >> the relative magitudes and phases of the adjacent bins. >> To preserve this envolope (i.e. to correctly reproduce >> transient signals), these bins need to remain adjacent. >> >> Another way to state this that any algorithm that does >> frequency scaling (or time stretching) needs some way >> to decide if certain features of the signal need to be >> interpreted as significant in the time domain or in the >> frequency domain. The correct decision depends on how >> a human listener would interpret that feature. >> That's a good point, maybe you find the time to listen to the results of this plug, that is what I do during the work on it, and, I'm very, very pleased with the result. >> It is not even possible to *define* a frequency scaling >> or time stretching algorithm without at least implicitly >> defining a way to decide on this. >> >> The implicit assumption in the current algorithm is that >> each bin is an separate feature in the frequency domain, >> and thus needs to be scaled independently of all others. >> >> >> Ciao, >> > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user From brummer- at web.de Sun Jun 8 04:38:50 2014 From: brummer- at web.de (hermann meyer) Date: Sun, 08 Jun 2014 06:38:50 +0200 Subject: [LAU] octaver (plugin) for bass In-Reply-To: <53934F59.7040906@web.de> References: <538B64CE.7030605@web.de> <538B6C3C.6030608@web.de> <538B7163.6070102@web.de> <53919B64.6040706@web.de> <5392CA2D.5070104@web.de> <20140607160824.GA27886@linuxaudio.org> <5393477D.9020801@web.de> <53934F59.7040906@web.de> Message-ID: <5393E8DA.9000603@web.de> Am 07.06.2014 19:43, schrieb hermann meyer: > Am 07.06.2014 19:10, schrieb hermann meyer: >> Am 07.06.2014 18:08, schrieb Fons Adriaensen: >>> On Sat, Jun 07, 2014 at 10:15:41AM +0200, hermann meyer wrote: >>> >>>> Thanks have to go to Stephan M. Bernsee from dspdimension as well. >>>> GxDetune is based on his work here: >>>> http://www.dspdimension.com/admin/pitch-shifting-using-the-ft/ >>> This sort of works, but it's not what it claims to be. >> >> Do you talk about my reworked code here: >> http://sourceforge.net/p/guitarix/git/ci/master/tree/trunk/src/LV2/gx_detune.lv2/detune.cc >> >>> >>> The whole part that finds the exact frequency by comparing >>> phases is completely redundant. This information is never >>> really used. It just looks as if it is used. >> That's true, we could leave that part out, that will spare us 0.2% dsp load. From fons at linuxaudio.org Sun Jun 8 09:49:31 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sun, 8 Jun 2014 09:49:31 +0000 Subject: [LAU] octaver (plugin) for bass In-Reply-To: <5393E8DA.9000603@web.de> References: <538B7163.6070102@web.de> <53919B64.6040706@web.de> <5392CA2D.5070104@web.de> <20140607160824.GA27886@linuxaudio.org> <5393477D.9020801@web.de> <53934F59.7040906@web.de> <5393E8DA.9000603@web.de> Message-ID: <20140608094931.GA6958@linuxaudio.org> On Sun, Jun 08, 2014 at 06:38:50AM +0200, hermann meyer wrote: > >>>The whole part that finds the exact frequency by comparing > >>>phases is completely redundant. This information is never > >>>really used. It just looks as if it is used. > > That's true, we could leave that part out, that will spare us 0.2% > dsp load. You're missing the point. Which is that the frequency info *should* be used to decide how to map bins to a new frequency. Also, it's easy to reduce CPU load by letting the whole thing run at a quarter of the system sample rate. Which means that for one octave down everything above 3 kHz is gone. But I guess CPU load was not the only reason for doing that. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From nicola.di.marzo at vodafone.it Sun Jun 8 10:22:21 2014 From: nicola.di.marzo at vodafone.it (Nicola) Date: Sun, 08 Jun 2014 11:22:21 +0100 Subject: [LAU] New EP - Everywhere Nowhere - indie rock Message-ID: <5394395D.3050606@vodafone.it> Hi all, The EP of my band is out now! It's been produced using Free and Open Source Software, Debian GNU/Linux + Tango Studio Repositories and Ardour 3. The license is CC BY-NC-SA Feel free to listen and download it freely from soundcloud https://soundcloud.com/bandage-indie-rock/sets/bandage-everywhere-nowhere or Jamendo: http://www.jamendo.com/en/list/a135179/everywhere-nowhere There's also a FLAC version available from Bandcamp http://bandage-indierock.bandcamp.com/album/everywhere-nowhere at 1 euro. Opinions or suggestions are welcome! Thanks Linuxaudio community! Regards, Nicola From brummer- at web.de Sun Jun 8 10:36:17 2014 From: brummer- at web.de (hermann meyer) Date: Sun, 08 Jun 2014 12:36:17 +0200 Subject: [LAU] octaver (plugin) for bass In-Reply-To: <20140608094931.GA6958@linuxaudio.org> References: <538B7163.6070102@web.de> <53919B64.6040706@web.de> <5392CA2D.5070104@web.de> <20140607160824.GA27886@linuxaudio.org> <5393477D.9020801@web.de> <53934F59.7040906@web.de> <5393E8DA.9000603@web.de> <20140608094931.GA6958@linuxaudio.org> Message-ID: <53943CA1.6010108@web.de> Am 08.06.2014 11:49, schrieb Fons Adriaensen: > On Sun, Jun 08, 2014 at 06:38:50AM +0200, hermann meyer wrote: > >>>>> The whole part that finds the exact frequency by comparing >>>>> phases is completely redundant. This information is never >>>>> really used. It just looks as if it is used. >> That's true, we could leave that part out, that will spare us 0.2% >> dsp load. > You're missing the point. Which is that the frequency info > *should* be used to decide how to map bins to a new frequency. > > Also, it's easy to reduce CPU load by letting the whole thing > run at a quarter of the system sample rate. Which means that > for one octave down everything above 3 kHz is gone. But I guess > CPU load was not the only reason for doing that. > > Ciao, > Without downsampling it use (well, 4xtimes more then now) 8% dsp load. Most costs in the original source comes from that used values are not pre-calculated. But indeed, the reason for downsampling is that the limited frequency range makes it sound good, and for guitar/bass 3kHz are far more then enough when you would add a octave up/down to the original sound. regards hermann From fons at linuxaudio.org Sun Jun 8 11:34:53 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sun, 8 Jun 2014 11:34:53 +0000 Subject: [LAU] octaver (plugin) for bass In-Reply-To: <53943CA1.6010108@web.de> References: <53919B64.6040706@web.de> <5392CA2D.5070104@web.de> <20140607160824.GA27886@linuxaudio.org> <5393477D.9020801@web.de> <53934F59.7040906@web.de> <5393E8DA.9000603@web.de> <20140608094931.GA6958@linuxaudio.org> <53943CA1.6010108@web.de> Message-ID: <20140608113453.GB6958@linuxaudio.org> On Sun, Jun 08, 2014 at 12:36:17PM +0200, hermann meyer wrote: > Without downsampling it use (well, 4xtimes more then now) 8% dsp > load. Most costs in the original source comes from that used values > are not pre-calculated. > But indeed, the reason for downsampling is that the limited > frequency range makes it sound good, because that removes most of the broadband junk that would be generated otherwise... > and for guitar/bass 3kHz are > far more then enough when you would add a octave up/down to the > original sound. True for bass and guitar. Still this algorithm is far from what it could be. I don't blame for you that, it's Bernsee who is missing the consequences of his own analysis (which is valid as far as it goes). Take alook at his table labeled 'pass #5'. The input signal is halfway between two bins. Assume we want one octave up. The expected output signal corresponds exactly to bin 225. For that signal, the output of the analysis FFT would be (similar to 'pass #1): bin amplitude ------------------ 223 0.000 224 0.500 225 1.000 226 0.500 227 0.000 And that is of course also what the correct input to the synthesis IFFT should be. Which is quite different from what the algorithm produces (by scaling each bin individually): bin amplitude ------------------- 222 0.170 223 0.000 224 0.849 225 0.000 226 0.849 227 0.000 228 0.170 The result of this after the IFFT is the correct frequency, but with two periods of the window applied (it will be zero at the center). The frequency values that are calculated provide exactly the information required to avoid this and to do the correct calculation. But it's just thrown away. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From massimo at fsfe.org Mon Jun 9 15:54:38 2014 From: massimo at fsfe.org (Massimo Barbieri) Date: Mon, 09 Jun 2014 17:54:38 +0200 Subject: [LAU] John Option: the source code of the music Message-ID: <5395D8BE.3010407@fsfe.org> Hi! A few months ago I started with a friend of mine a musical project that has particular similarities to the philosophy of free software. I would like to tell you about it in this post and I'd love to read your feedback. With our band John Option[1] we published our first single My monkey some week ago. Of course the song is published under the terms of the Creative Commons License (CC-BY-SA) and it's completely produced only with free software: Ardour, Hydrogen, Jack, Qsynth, CALF, and many other great free audio software that we used under a GNU/Linux system. Here you can listen the single: http://youtu.be/GdsyGlPkfEg But with the project of John Option we have done a little more in the direction of freedom. As for the free software the source code is accessible for the users, we decided to do the same thing for our music. So we published the single recording tracks of the song My monkey and the complete Ardour session. All this material is published in our official website[1] under the CC-BY-SA license so that anyone can use our tracks to produce a remix of our song or even a new song that have to be published under the same license. You can find all about our project here: http://johnoption.org I hope that you like our choice of freedom. If you feel like I'd love to read your feedback, because the encouragement of the people who listen to us and appreciate the philosophy of our project is a fuel for us to continue. Ciao, Max-B [1] Official site: http://johnoption.org -- XMPP: massimo at jabber.fsfe.org OpenPGP Key-Id: 0x5D168FC1 -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 274 bytes Desc: OpenPGP digital signature URL: From rosea.grammostola at gmail.com Tue Jun 10 15:27:13 2014 From: rosea.grammostola at gmail.com (rosea grammostola) Date: Tue, 10 Jun 2014 17:27:13 +0200 Subject: [LAU] octaver (plugin) for bass In-Reply-To: <20140608113453.GB6958@linuxaudio.org> References: <53919B64.6040706@web.de> <5392CA2D.5070104@web.de> <20140607160824.GA27886@linuxaudio.org> <5393477D.9020801@web.de> <53934F59.7040906@web.de> <5393E8DA.9000603@web.de> <20140608094931.GA6958@linuxaudio.org> <53943CA1.6010108@web.de> <20140608113453.GB6958@linuxaudio.org> Message-ID: I'm not sure if real hardware stompboxes of this type are better then this plugin, but this might be the kind of sound I might like to buy a hardware stompbox for. Maybe I can test one somewhere and compare it with Gxdetune. Thanks for the comments Fons, tips for improvement are always welcome I think. On Sun, Jun 8, 2014 at 1:34 PM, Fons Adriaensen wrote: > On Sun, Jun 08, 2014 at 12:36:17PM +0200, hermann meyer wrote: > > > Without downsampling it use (well, 4xtimes more then now) 8% dsp > > load. Most costs in the original source comes from that used values > > are not pre-calculated. > > But indeed, the reason for downsampling is that the limited > > frequency range makes it sound good, > > because that removes most of the broadband junk that would be generated > otherwise... > > > and for guitar/bass 3kHz are > > far more then enough when you would add a octave up/down to the > > original sound. > > True for bass and guitar. > > Still this algorithm is far from what it could be. I don't blame > for you that, it's Bernsee who is missing the consequences of his > own analysis (which is valid as far as it goes). > > Take alook at his table labeled 'pass #5'. The input signal is > halfway between two bins. Assume we want one octave up. The expected > output signal corresponds exactly to bin 225. For that signal, the > output of the analysis FFT would be (similar to 'pass #1): > > bin amplitude > ------------------ > 223 0.000 > 224 0.500 > 225 1.000 > 226 0.500 > 227 0.000 > > And that is of course also what the correct input to the synthesis > IFFT should be. Which is quite different from what the algorithm > produces (by scaling each bin individually): > > bin amplitude > ------------------- > 222 0.170 > 223 0.000 > 224 0.849 > 225 0.000 > 226 0.849 > 227 0.000 > 228 0.170 > > The result of this after the IFFT is the correct frequency, but > with two periods of the window applied (it will be zero at the > center). > > The frequency values that are calculated provide exactly the > information required to avoid this and to do the correct > calculation. But it's just thrown away. > > Ciao, > > -- > FA > > A world of exhaustive, reliable metadata would be an utopia. > It's also a pipe-dream, founded on self-delusion, nerd hubris > and hysterically inflated market opportunities. (Cory Doctorow) > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > -------------- next part -------------- An HTML attachment was scrubbed... URL: From gianfranco at portalmod.com.br Tue Jun 10 15:56:29 2014 From: gianfranco at portalmod.com.br (Gianfranco Ceccolini) Date: Tue, 10 Jun 2014 12:56:29 -0300 Subject: [LAU] octaver (plugin) for bass In-Reply-To: References: <53919B64.6040706@web.de> <5392CA2D.5070104@web.de> <20140607160824.GA27886@linuxaudio.org> <5393477D.9020801@web.de> <53934F59.7040906@web.de> <5393E8DA.9000603@web.de> <20140608094931.GA6958@linuxaudio.org> <53943CA1.6010108@web.de> <20140608113453.GB6958@linuxaudio.org> Message-ID: <5C37708C-DC68-4B56-8FA4-4A5889EAB2BE@portalmod.com.br> Sorry for the late posting, but I?d also recommend the mod-pitchshifter https://github.com/portalmod/mod-pitchshifter We have developed 4 plugins: Capo ( 1 to 7 semitones up), the SuperCapo (1 to 24 semitones up), Drop (1 to 12 semitones down) and SuperWhammy (continuous travel from -12 to 24 semitones) They are a bit CPU hungry but sound quality is quite good. Hope I?ve helped Gianfranco The MOD Team Em 10/06/2014, ?(s) 12:27, rosea grammostola escreveu: > I'm not sure if real hardware stompboxes of this type are better then this plugin, but this might be the kind of sound I might like to buy a hardware stompbox for. Maybe I can test one somewhere and compare it with Gxdetune. Thanks for the comments Fons, tips for improvement are always welcome I think. > > > On Sun, Jun 8, 2014 at 1:34 PM, Fons Adriaensen wrote: > On Sun, Jun 08, 2014 at 12:36:17PM +0200, hermann meyer wrote: > > > Without downsampling it use (well, 4xtimes more then now) 8% dsp > > load. Most costs in the original source comes from that used values > > are not pre-calculated. > > But indeed, the reason for downsampling is that the limited > > frequency range makes it sound good, > > because that removes most of the broadband junk that would be generated > otherwise... > > > and for guitar/bass 3kHz are > > far more then enough when you would add a octave up/down to the > > original sound. > > True for bass and guitar. > > Still this algorithm is far from what it could be. I don't blame > for you that, it's Bernsee who is missing the consequences of his > own analysis (which is valid as far as it goes). > > Take alook at his table labeled 'pass #5'. The input signal is > halfway between two bins. Assume we want one octave up. The expected > output signal corresponds exactly to bin 225. For that signal, the > output of the analysis FFT would be (similar to 'pass #1): > > bin amplitude > ------------------ > 223 0.000 > 224 0.500 > 225 1.000 > 226 0.500 > 227 0.000 > > And that is of course also what the correct input to the synthesis > IFFT should be. Which is quite different from what the algorithm > produces (by scaling each bin individually): > > bin amplitude > ------------------- > 222 0.170 > 223 0.000 > 224 0.849 > 225 0.000 > 226 0.849 > 227 0.000 > 228 0.170 > > The result of this after the IFFT is the correct frequency, but > with two periods of the window applied (it will be zero at the > center). > > The frequency values that are calculated provide exactly the > information required to avoid this and to do the correct > calculation. But it's just thrown away. > > Ciao, > > -- > FA > > A world of exhaustive, reliable metadata would be an utopia. > It's also a pipe-dream, founded on self-delusion, nerd hubris > and hysterically inflated market opportunities. (Cory Doctorow) > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user -------------- next part -------------- An HTML attachment was scrubbed... URL: From rosea.grammostola at gmail.com Tue Jun 10 16:15:50 2014 From: rosea.grammostola at gmail.com (rosea grammostola) Date: Tue, 10 Jun 2014 18:15:50 +0200 Subject: [LAU] octaver (plugin) for bass In-Reply-To: <5C37708C-DC68-4B56-8FA4-4A5889EAB2BE@portalmod.com.br> References: <53919B64.6040706@web.de> <5392CA2D.5070104@web.de> <20140607160824.GA27886@linuxaudio.org> <5393477D.9020801@web.de> <53934F59.7040906@web.de> <5393E8DA.9000603@web.de> <20140608094931.GA6958@linuxaudio.org> <53943CA1.6010108@web.de> <20140608113453.GB6958@linuxaudio.org> <5C37708C-DC68-4B56-8FA4-4A5889EAB2BE@portalmod.com.br> Message-ID: On Tue, Jun 10, 2014 at 5:56 PM, Gianfranco Ceccolini < gianfranco at portalmod.com.br> wrote: > Sorry for the late posting, but I?d also recommend the mod-pitchshifter > > https://github.com/portalmod/mod-pitchshifter > > We have developed 4 plugins: Capo ( 1 to 7 semitones up), the SuperCapo (1 > to 24 semitones up), Drop (1 to 12 semitones down) and SuperWhammy > (continuous travel from -12 to 24 semitones) > > They are a bit CPU hungry but sound quality is quite good. > Thanks, but the make files doesn't work / are not up-to-date -------------- next part -------------- An HTML attachment was scrubbed... URL: From emailgrant at gmail.com Tue Jun 10 17:05:00 2014 From: emailgrant at gmail.com (Grant) Date: Tue, 10 Jun 2014 10:05:00 -0700 Subject: [LAU] Gentoo mpd overlay? Message-ID: Can I contact someone in charge of the Gentoo mpd overlay here? If not, any idea where to do so? I'm running into this: https://github.com/musicpd/mpd-overlay/issues/3 - Grant From silvain at freeshell.de Tue Jun 10 17:28:11 2014 From: silvain at freeshell.de (F. Silvain) Date: Tue, 10 Jun 2014 19:28:11 +0200 (CEST) Subject: [LAU] John Option: the source code of the music In-Reply-To: <5395D8BE.3010407@fsfe.org> References: <5395D8BE.3010407@fsfe.org> Message-ID: <1406101924580.28238@freeshell.de> Hey hey Max, good track! It's amazing how good a production you can achieve with the samples involved. I'm thinking of the drums. I think I've heard and played those, when I first investigated Hydrogen. I never thought you could mix them that well. Chapeau! Not my current mood, but everything with this song sounds as it should do. Perhaps I will go for the Ardour session and take a look at your setup there. Ta-ta ---- Ffanci * Internet: http://freeshell.de/~silvain From khirai at ongaku.isa-geek.net Tue Jun 10 17:58:48 2014 From: khirai at ongaku.isa-geek.net (Kelly Hirai) Date: Tue, 10 Jun 2014 13:58:48 -0400 Subject: [LAU] Gentoo mpd overlay? In-Reply-To: References: Message-ID: <53974758.3060904@ongaku.isa-geek.net> mpd is in the main portage tree. * media-sound/mpd Available versions: 0.17.6 ~0.18.8 ~0.18.9 {adplug +alsa ao audiofile bzip2 cdio +curl debug faad +ffmpeg +fifo flac fluidsynth gme +id3tag inotify ipv6 jack lame lastfmradio libmpdclient libsamplerate +mad mikmod mms modplug mpg123 musepack +network ogg openal opus oss pipe pulseaudio recorder sid sndfile soundcloud soup sqlite systemd tcpd twolame unicode vorbis wavpack wildmidi zeroconf zip} Homepage: http://www.musicpd.org Description: The Music Player Daemon (mpd) this overlay is looking a little dusty, are you sure you still need this overlay? k. On 06/10/2014 01:05 PM, Grant wrote: > Can I contact someone in charge of the Gentoo mpd overlay here? If > not, any idea where to do so? > > I'm running into this: > > https://github.com/musicpd/mpd-overlay/issues/3 > > - Grant > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user From massimo at fsfe.org Tue Jun 10 19:51:14 2014 From: massimo at fsfe.org (Massimo Barbieri) Date: Tue, 10 Jun 2014 21:51:14 +0200 Subject: [LAU] John Option: the source code of the music In-Reply-To: <1406101924580.28238@freeshell.de> References: <5395D8BE.3010407@fsfe.org> <1406101924580.28238@freeshell.de> Message-ID: <539761B2.3020209@fsfe.org> Hi! Il 10/06/2014 19:28, F. Silvain ha scritto: > good track! It's amazing how good a production you can achieve with the > samples involved. I'm thinking of the drums. I think I've heard and > played those, when I first investigated Hydrogen. I never thought you > could mix them that well. Chapeau! Many thanks! Hydrogen is a geat free software and it comes with great drumkit! > do. Perhaps I will go for the Ardour session and take a look at your > setup there. Thanks for your interesting. If you investigate to my Ardour session and you would like to write your feedback here I will be glad to read it. For the guitar I used a 1991 Fender Telecaster American Standard with a Boss Blues Drive and a Fender Princeton Amp. I recorded the guitar from the amp cone with a Shure SM7 directlyconnected to my audio card Focusrite Scarlett 2i2. For the lead vocals I used the Shure SM57 with a GAP pre-73 DLX pre-amp. And for the backing vocal a Bluebird Condenser microphone with the same preamp. Ciao, Max-B -- XMPP: massimo at jabber.fsfe.org OpenPGP Key-Id: 0x5D168FC1 -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 274 bytes Desc: OpenPGP digital signature URL: From aiyumi.br at gmail.com Wed Jun 11 12:39:43 2014 From: aiyumi.br at gmail.com (Aiyumi Moriya) Date: Wed, 11 Jun 2014 09:39:43 -0300 Subject: [LAU] Recommended near-realistic strings section generator? In-Reply-To: References: Message-ID: 2014-05-26 1:30 GMT-03:00, Danni Coy : > I have had some luck with Kontact and festige Really? :D That got my hopes up. Does the free Kontakt 5 Player also work under Wine? There's this violin sample library that I'm interested for quite a while now: http://embertone.com/instruments/friedlanderviolin.php It is a solo violin library, but it also has an "ensemble" mode. I really liked the sound... But It runs on Kontakt Player. I googled about Kontakt 5 Player on Linux/Wine and didn't find anything useful. Also, from what I understood from some reading, it seems to use a software called Continuata for watermarking and downloading the library. I've never heard of it before - I'm a beginner in the area of (proprietary) sample libraries -, but from the forum posts I read, it seems to require the user to install a "Continuata utility program" that downloads and extracts the library. The posts I read were this one: http://www.kvraudio.com/forum/viewtopic.php?p=5448245 and the one directly below it. But I googled about it too, and found a post about someone downloading (not specifically this sample library) normally via the browser without using the Continuata thing. http://www.kvraudio.com/forum/viewtopic.php?t=376934 Does that mean it isn't required, and only the developers need to use it to watermark/upload or something? If it's really required, any idea if that works under Wine? The library is on sale right now, costing $99 USD until June 15th (before, it was $110), and I'm seriously considering buying it. I just would like some confirmation if it can work under Wine, even if JACK doesn't work and I have to use an external hardware recorder to capture the speaker outputs or something. :P Thank you to anyone that chimes in! :) -- ____________________ Blog: http://aiyumi.warpstar.net/ From emailgrant at gmail.com Wed Jun 11 14:37:36 2014 From: emailgrant at gmail.com (Grant) Date: Wed, 11 Jun 2014 07:37:36 -0700 Subject: [LAU] Gentoo mpd overlay? In-Reply-To: <53974758.3060904@ongaku.isa-geek.net> References: <53974758.3060904@ongaku.isa-geek.net> Message-ID: > mpd is in the main portage tree. > > * media-sound/mpd > Available versions: 0.17.6 ~0.18.8 ~0.18.9 {adplug +alsa ao > audiofile bzip2 cdio +curl debug faad +ffmpeg +fifo flac fluidsynth gme > +id3tag inotify ipv6 jack lame lastfmradio libmpdclient libsamplerate > +mad mikmod mms modplug mpg123 musepack +network ogg openal opus oss > pipe pulseaudio recorder sid sndfile soundcloud soup sqlite systemd tcpd > twolame unicode vorbis wavpack wildmidi zeroconf zip} > Homepage: http://www.musicpd.org > Description: The Music Player Daemon (mpd) > > this overlay is looking a little dusty, are you sure you still need this > overlay? It's nice to be able to pull the latest from git. - Grant >> Can I contact someone in charge of the Gentoo mpd overlay here? If >> not, any idea where to do so? >> >> I'm running into this: >> >> https://github.com/musicpd/mpd-overlay/issues/3 >> >> - Grant From gianfranco at portalmod.com.br Wed Jun 11 14:53:52 2014 From: gianfranco at portalmod.com.br (Gianfranco Ceccolini) Date: Wed, 11 Jun 2014 11:53:52 -0300 Subject: [LAU] octaver (plugin) for bass In-Reply-To: References: <53919B64.6040706@web.de> <5392CA2D.5070104@web.de> <20140607160824.GA27886@linuxaudio.org> <5393477D.9020801@web.de> <53934F59.7040906@web.de> <5393E8DA.9000603@web.de> <20140608094931.GA6958@linuxaudio.org> <53943CA1.6010108@web.de> <20140608113453.GB6958@linuxaudio.org> <5C37708C-DC68-4B56-8FA4-4A5889EAB2BE@portalmod.com.br> Message-ID: <5C455A98-934E-42E9-8336-F6655B25ADC3@portalmod.com.br> Hi Rosea There was o problem in the makefile that?s just been fixed. It shall work now. Kind regards Gianfranco The MOD Team Em 10/06/2014, ?(s) 13:15, rosea grammostola escreveu: > > > > On Tue, Jun 10, 2014 at 5:56 PM, Gianfranco Ceccolini wrote: > Sorry for the late posting, but I?d also recommend the mod-pitchshifter > > https://github.com/portalmod/mod-pitchshifter > > We have developed 4 plugins: Capo ( 1 to 7 semitones up), the SuperCapo (1 to 24 semitones up), Drop (1 to 12 semitones down) and SuperWhammy (continuous travel from -12 to 24 semitones) > > They are a bit CPU hungry but sound quality is quite good. > > Thanks, but the make files doesn't work / are not up-to-date -------------- next part -------------- An HTML attachment was scrubbed... URL: From rosea.grammostola at gmail.com Wed Jun 11 17:52:23 2014 From: rosea.grammostola at gmail.com (rosea grammostola) Date: Wed, 11 Jun 2014 19:52:23 +0200 Subject: [LAU] octaver (plugin) for bass In-Reply-To: <5C455A98-934E-42E9-8336-F6655B25ADC3@portalmod.com.br> References: <53919B64.6040706@web.de> <5392CA2D.5070104@web.de> <20140607160824.GA27886@linuxaudio.org> <5393477D.9020801@web.de> <53934F59.7040906@web.de> <5393E8DA.9000603@web.de> <20140608094931.GA6958@linuxaudio.org> <53943CA1.6010108@web.de> <20140608113453.GB6958@linuxaudio.org> <5C37708C-DC68-4B56-8FA4-4A5889EAB2BE@portalmod.com.br> <5C455A98-934E-42E9-8336-F6655B25ADC3@portalmod.com.br> Message-ID: /git/mod-pitchshifter/SuperCapo$ make g++ -I. -I../Shared_files -O3 -ffast-math -msse -msse3 -mfpmath=sse -Wall -Wextra -march=native -mtune=native -c -fPIC -DPIC -o ../Shared_files/Exp.o ../Shared_files/Exp.cpp virtual memory exhausted: Cannot allocate memory make: *** [../Shared_files/Exp.o] Error 1 Since when is 2Gb not enough on Linux? :/ :) -------------- next part -------------- An HTML attachment was scrubbed... URL: From brummer- at web.de Wed Jun 11 17:55:13 2014 From: brummer- at web.de (hermann meyer) Date: Wed, 11 Jun 2014 19:55:13 +0200 Subject: [LAU] octaver (plugin) for bass In-Reply-To: References: <53919B64.6040706@web.de> <5392CA2D.5070104@web.de> <20140607160824.GA27886@linuxaudio.org> <5393477D.9020801@web.de> <53934F59.7040906@web.de> <5393E8DA.9000603@web.de> <20140608094931.GA6958@linuxaudio.org> <53943CA1.6010108@web.de> <20140608113453.GB6958@linuxaudio.org> Message-ID: <53989801.50600@web.de> Am 10.06.2014 17:27, schrieb rosea grammostola: > I'm not sure if real hardware stompboxes of this type are better then > this plugin, but this might be the kind of sound I might like to buy a > hardware stompbox for. Maybe I can test one somewhere and compare it > with Gxdetune. Thanks for the comments Fons, tips for improvement are > always welcome I think. > > Indeed, I'm interested to learn more at any time. :-) And when I could learn from one of the Giants of the OpenSource DSP programming, I'm more then happy to listen. > The result of this after the IFFT is the correct frequency, but > with two periods of the window applied (it will be zero at the > center). > > The frequency values that are calculated provide exactly the > information required to avoid this and to do the correct > calculation. But it's just thrown away. > I'm really unsure about how I could use this to calculate it right, could you give me some more hints, or best at all, a patch will be very welcome, as most I've learned from reading source code. > > Ciao, > > -- > FA > > regards hermann -------------- next part -------------- An HTML attachment was scrubbed... URL: From fons at linuxaudio.org Thu Jun 12 08:41:26 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Thu, 12 Jun 2014 08:41:26 +0000 Subject: [LAU] octaver (plugin) for bass In-Reply-To: <53989801.50600@web.de> References: <5392CA2D.5070104@web.de> <20140607160824.GA27886@linuxaudio.org> <5393477D.9020801@web.de> <53934F59.7040906@web.de> <5393E8DA.9000603@web.de> <20140608094931.GA6958@linuxaudio.org> <53943CA1.6010108@web.de> <20140608113453.GB6958@linuxaudio.org> <53989801.50600@web.de> Message-ID: <20140612084126.GA23224@linuxaudio.org> On Wed, Jun 11, 2014 at 07:55:13PM +0200, hermann meyer wrote: > I'm really unsure about how I could use this to calculate it right, > could you give me some more hints, or best at all, a patch will be > very welcome, as most I've learned from reading source code. There's a paper by Dolson and Laroche (1999) which is a 'must read' for anyone dabbling with phase vocoders. See which has a free download link from columbia.edu. Their method does the right calculations even without using the computed frequencies (they wanted to avoid the atan2() calls), but it's possible to do the same thing using them. Hint: if you need the frequencies the Bernsee way, just compute them using bin number as the unit instead of Hz - it simplifies things quite a lot ans also shows much clearer what's going on. This is how it's done in some experimantal code I wrote years ago: for (i = 0; i <= hlen; i++) { x = _F [i][0]; y = _F [i][1]; p = atan2 (y, x) / twopi; // Phase in cycles. d = p - _phase1 [i] - i / div; // Minus previous and expected difference. d -= floorf (d + 0.5); // Reduce to [-0.5...+0.5] _phase1 [i] = p; // Store for next iteration. _freq [i] = i + d * div; // Frequency in bins. _magn [i] = hypotf (x, y); // Magnitude. } where 'div' is the overlap factor, e.g. 4 for 75%. Note this must be a float. I'll let you work out the inverse operation by yourself, you'll notice it's even simpler (as some terms cancel out). But it's not needed when using D&L's method. For code, maybe have a look at Rubberband which may contain interesting things (I don't know, never dared to look as it would probably have me hooked up for weeks experimenting with this sort of algorithms...) Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From letz at grame.fr Thu Jun 12 09:24:00 2014 From: letz at grame.fr (=?windows-1252?Q?St=E9phane_Letz?=) Date: Thu, 12 Jun 2014 11:24:00 +0200 Subject: [LAU] FaustLive is released ! Message-ID: GRAME is happy to announce the official release of FaustLive. FaustLive is an advanced self-contained prototyping environment for the Faust programming language with an ultra-short edit-compile-run cycle. Thanks to its fully embedded compilation chain, FaustLive is simple to install and doesn't require any external compiler, development toolchain or SDK to run. FaustLive is the ideal tool for fast prototyping. Faust programs can be compiled and run on the fly by simple drag and drop. They can even be edited and recompiled while running without sound interruption or Jack disconnection. 1) Dynamic Compilation : On FaustLive?s windows you can drop your Faust code as a file, a string or a url. The code will be dynamically compiled and executed. You can then choose to edit your code. It will be opened in the default editor for .dsp files (FOLLOW THE README TO CONFIGURE FILE ASSOCIATION). The application will be automatically recompiled, every time you save your document. A crossfade is calculated between two relaying applications in a window to avoid brutal sound interruptions. 2) Audio Drivers : Depending on your Operation System, you will have different drivers available: On OSX : Coreaudio, Jack and NetJack On Linux : Jack and NetJack On Windows : Portaudio You can then dynamically switch from one to another in FaustLive?s preferences. 3) Export Your DSP : Exporting your DSP as plugins is easy, thanks to FaustWeb, compilation service. In FaustLive?s export menu, you can find every platform and architecture that Faust can target. As you choose your target, your code is sent to FaustWeb and you receive the requested binary in exchange. 4) Save Snapshots : If you create a configuration you like, you can save it as a Snapshot. The state of FaustLive will be saved (running applications, Jack connections, interface parameters, ?). Later on, you will be able to whether : - recall the snapshot : closing any running application to restore the saved state - import the snapshot : adding the saved state to the current state 5) Remote Control Interfaces (only on Linux and OSX for now) : In the Windows Option toolBar, you can open a UDP port for OSC control or a TCP port for HTTP control. Moreover, the HTLM interface can be accessed through a QrCode that you can create from ?View QrCode ? in the menu ? Window ?. Download access: http://sourceforge.net/projects/faudiostream/files/ Tutorial video: http://www.youtube.com/watch?v=8ZUD2c5D-PU Sarah Denoux and St?phane Letz From cannam at all-day-breakfast.com Thu Jun 12 09:41:00 2014 From: cannam at all-day-breakfast.com (Chris Cannam) Date: Thu, 12 Jun 2014 10:41:00 +0100 Subject: [LAU] octaver (plugin) for bass In-Reply-To: <20140612084126.GA23224@linuxaudio.org> References: <5392CA2D.5070104@web.de> <20140607160824.GA27886@linuxaudio.org> <5393477D.9020801@web.de> <53934F59.7040906@web.de> <5393E8DA.9000603@web.de> <20140608094931.GA6958@linuxaudio.org> <53943CA1.6010108@web.de> <20140608113453.GB6958@linuxaudio.org> <53989801.50600@web.de> <20140612084126.GA23224@linuxaudio.org> Message-ID: <1402566060.31518.127964301.79F4305F@webmail.messagingengine.com> On Thu, Jun 12, 2014, at 09:41 AM, Fons Adriaensen wrote: > There's a paper by Dolson and Laroche (1999) which is a 'must read' > for anyone dabbling with phase vocoders. One warning, the method described in it is patented -- I had to do a hasty rewrite in some early Rubber Band code. I don't know whether anyone enforces the patent though. > For code, maybe have a look at Rubberband which may contain > interesting things (I don't know, never dared to look Probably wise -- I expect it would horrify you. And of course this application is an incredible time-sink simply because there's no right way to do it. It's a subject that can surely drive you mad. You gave a low-level example of the problem earlier (with neighbouring frequency bins). Looking at it at a high level, you're basically trying to synthesise a signal that corresponds to "what the same instruments would have sounded like if they were playing slower" (or higher, or whatever). You don't have anything like enough knowledge frame-by-frame to actually do that. You can get closer for many signals with a sinusoidal modelling decomposition (in which you track the frequencies that appear to be consistent frame-to-frame, adjust their phases, and treat the rest as noise whose phase you don't change) but those methods are still fairly expensive to do and of course even the best method can never actually be technically correct. Rubber Band does basically the same sums as you just gave. By default it fudges the time/frequency question for neighbouring bins by taking groups of bins that appear to be moving in the same direction (in frequency) and giving each one a phase advance somewhere between its single-bin predicted value and the value that would be expected if the group were all following the same path. That's mathematically... barely supportable at all, but in many cases it sounds OK. The other thing it does by default is reset the stretch factor and revert to the input phases when a sufficiently noisy transient is found, which is why you get quite poppy transients in e.g. drum loops that are either satisfying or unrealistic depending on your point of view. Chris From robin at gareus.org Thu Jun 12 21:00:57 2014 From: robin at gareus.org (Robin Gareus) Date: Thu, 12 Jun 2014 23:00:57 +0200 Subject: [LAU] LAC'14 video archive Message-ID: <539A1509.50906@gareus.org> Hi all, The video recordings of the LAC'14 presentations have just been uploaded to the conference website and are now directly linked from the archive: http://lac.linuxaudio.org/2014/program There are still a three videos missing and the workshop videos are also yet to come. Currently they are also only available as vp8/vorbis/webm (sorry IE and Safari users). But since it has been quite a while already, we decided to not hold back the release of these already finished videos any further. Once the collection is complete, we will provide a .torrent. Meanwhile, for those who prefer to download the videos incrementally, they are accessible via rsync://linuxaudio.org/ [1]. Many thanks for Frank and Moritz to get those done in really outstanding quality this year. Kudos to the complete stream-team. enjoy, robin - for the LAC'14 team [1] example to get the 720p versions: rsync -Pa --exclude "*360p.webm" \ rsync://linuxaudio.org/lac2014/ \ lac2014/ From fons at linuxaudio.org Thu Jun 12 22:16:01 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Thu, 12 Jun 2014 22:16:01 +0000 Subject: [LAU] octaver (plugin) for bass In-Reply-To: <1402566060.31518.127964301.79F4305F@webmail.messagingengine.com> References: <5393477D.9020801@web.de> <53934F59.7040906@web.de> <5393E8DA.9000603@web.de> <20140608094931.GA6958@linuxaudio.org> <53943CA1.6010108@web.de> <20140608113453.GB6958@linuxaudio.org> <53989801.50600@web.de> <20140612084126.GA23224@linuxaudio.org> <1402566060.31518.127964301.79F4305F@webmail.messagingengine.com> Message-ID: <20140612221601.GA3543@linuxaudio.org> On Thu, Jun 12, 2014 at 10:41:00AM +0100, Chris Cannam wrote: > One warning, the method described in it is patented -- I had to do a > hasty rewrite in some early Rubber Band code. I don't know whether > anyone enforces the patent though. Indeed... I'd be surprised if the patent is actually enforced... There's also a collection of related patents by Fraunhofer.. > > For code, maybe have a look at Rubberband which may contain > > interesting things (I don't know, never dared to look > > Probably wise -- I expect it would horrify you. And of course this > application is an incredible time-sink simply because there's no right > way to do it. It's a subject that can surely drive you mad. It sure can. Meanwhile, since you tickled me, I did have a look. I've seen worse :-) > You gave a low-level example of the problem earlier (with neighbouring > frequency bins). Looking at it at a high level, you're basically trying > to synthesise a signal that corresponds to "what the same instruments > would have sounded like if they were playing slower" (or higher, or > whatever). There's a more fundamental problem behind this. Suppose you have a sine wave at some frequency F, modulated (i.e. multiplied) by say a 8 Hz sine wave. Assume we want to transpose an octave up. Now is this signal a) just a single frequency (F) with some amplitude modulation on it, or b) two signals, at F-8 and F+8 Hz. Mathematically, and in the analysis spectrum, these are just the same thing. It's a matter of interpretation. In case (a) you'd want a sine at 2*F with the same 8 Hz modulation on it. In case (b) the wanted output is two signals at 2*(F-8) and 2*(F+8) Hz. You have the choice of interpreting the 'detail' in either the time or frequency domains. For high F, our hearing would probably favour (a). But at low frequencies things could be different. 64 Hz and 80 Hz would make a nice major third... Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From brummer- at web.de Fri Jun 13 07:04:59 2014 From: brummer- at web.de (hermann meyer) Date: Fri, 13 Jun 2014 09:04:59 +0200 Subject: [LAU] octaver (plugin) for bass In-Reply-To: <20140612221601.GA3543@linuxaudio.org> References: <5393477D.9020801@web.de> <53934F59.7040906@web.de> <5393E8DA.9000603@web.de> <20140608094931.GA6958@linuxaudio.org> <53943CA1.6010108@web.de> <20140608113453.GB6958@linuxaudio.org> <53989801.50600@web.de> <20140612084126.GA23224@linuxaudio.org> <1402566060.31518.127964301.79F4305F@webmail.messagingengine.com> <20140612221601.GA3543@linuxaudio.org> Message-ID: <539AA29B.40103@web.de> Am 13.06.2014 00:16, schrieb Fons Adriaensen: > On Thu, Jun 12, 2014 at 10:41:00AM +0100, Chris Cannam wrote: > >> One warning, the method described in it is patented -- I had to do a >> hasty rewrite in some early Rubber Band code. I don't know whether >> anyone enforces the patent though. > Indeed... I'd be surprised if the patent is actually enforced... > There's also a collection of related patents by Fraunhofer.. > >>> For code, maybe have a look at Rubberband which may contain >>> interesting things (I don't know, never dared to look >> Probably wise -- I expect it would horrify you. And of course this >> application is an incredible time-sink simply because there's no right >> way to do it. It's a subject that can surely drive you mad. > It sure can. Meanwhile, since you tickled me, I did have a look. > I've seen worse :-) > >> You gave a low-level example of the problem earlier (with neighbouring >> frequency bins). Looking at it at a high level, you're basically trying >> to synthesise a signal that corresponds to "what the same instruments >> would have sounded like if they were playing slower" (or higher, or >> whatever). > There's a more fundamental problem behind this. > > Suppose you have a sine wave at some frequency F, modulated (i.e. > multiplied) by say a 8 Hz sine wave. Assume we want to transpose > an octave up. Now is this signal > > a) just a single frequency (F) with some amplitude modulation on it, > or > b) two signals, at F-8 and F+8 Hz. > > Mathematically, and in the analysis spectrum, these are just the > same thing. It's a matter of interpretation. In case (a) you'd want > a sine at 2*F with the same 8 Hz modulation on it. In case (b) the > wanted output is two signals at 2*(F-8) and 2*(F+8) Hz. You have > the choice of interpreting the 'detail' in either the time or > frequency domains. > > For high F, our hearing would probably favour (a). But at low > frequencies things could be different. 64 Hz and 80 Hz would > make a nice major third... > > > Ciao, > Thanks for your hints. I've found a other interesting paper which seems to use the resampling technique you've talked about in a earlier post. http://qmplus.qmul.ac.uk/mod/resource/view.php?id=305410 From gianfranco at portalmod.com.br Fri Jun 13 12:55:51 2014 From: gianfranco at portalmod.com.br (Gianfranco Ceccolini) Date: Fri, 13 Jun 2014 09:55:51 -0300 Subject: [LAU] octaver (plugin) for bass In-Reply-To: References: <53919B64.6040706@web.de> <5392CA2D.5070104@web.de> <20140607160824.GA27886@linuxaudio.org> <5393477D.9020801@web.de> <53934F59.7040906@web.de> <5393E8DA.9000603@web.de> <20140608094931.GA6958@linuxaudio.org> <53943CA1.6010108@web.de> <20140608113453.GB6958@linuxaudio.org> <5C37708C-DC68-4B56-8FA4-4A5889EAB2BE@portalmod.com.br> <5C455A98-934E-42E9-8336-F6655B25ADC3@portalmod.com.br> Message-ID: <0A0D63AE-5DA4-4F56-A5D4-060E6110249B@portalmod.com.br> Hi Rosea. Sorry Again. We?ve been experimenting with some compile time details and a arctan table generation was still bugged. Andr? just fixed it. Cheers Gianfranco Em 11/06/2014, ?(s) 14:52, rosea grammostola escreveu: > /git/mod-pitchshifter/SuperCapo$ make > g++ -I. -I../Shared_files -O3 -ffast-math -msse -msse3 -mfpmath=sse -Wall -Wextra -march=native -mtune=native -c -fPIC -DPIC -o ../Shared_files/Exp.o ../Shared_files/Exp.cpp > virtual memory exhausted: Cannot allocate memory > make: *** [../Shared_files/Exp.o] Error 1 > > Since when is 2Gb not enough on Linux? :/ :) From aiyumi.br at gmail.com Sat Jun 14 01:37:50 2014 From: aiyumi.br at gmail.com (Aiyumi Moriya) Date: Fri, 13 Jun 2014 22:37:50 -0300 Subject: [LAU] Embertone Friedlander Violin on Linux - was Re: Recommended near-realistic strings section generator? Message-ID: 2014-06-11 9:39 GMT-03:00, Aiyumi Moriya : > 2014-05-26 1:30 GMT-03:00, Danni Coy : >> I have had some luck with Kontact and festige > > Really? :D That got my hopes up. > > Does the free Kontakt 5 Player also work under Wine? There's this > violin sample library that I'm interested for quite a while now: > > http://embertone.com/instruments/friedlanderviolin.php > > It is a solo violin library, but it also has an "ensemble" mode. I > really liked the sound... But It runs on Kontakt Player. I googled > about Kontakt 5 Player on Linux/Wine and didn't find anything useful. > Also, from what I understood from some reading, it seems to use a > software called Continuata for watermarking and downloading the > library. I've never heard of it before - I'm a beginner in the area of > (proprietary) sample libraries -, but from the forum posts I read, it > seems to require the user to install a "Continuata utility program" > that downloads and extracts the library. The posts I read were this > one: > > http://www.kvraudio.com/forum/viewtopic.php?p=5448245 > > and the one directly below it. > > But I googled about it too, and found a post about someone downloading > (not specifically this sample library) normally via the browser > without using the Continuata thing. > > http://www.kvraudio.com/forum/viewtopic.php?t=376934 > > Does that mean it isn't required, and only the developers need to use > it to watermark/upload or something? If it's really required, any idea > if that works under Wine? > > The library is on sale right now, costing $99 USD until June 15th > (before, it was $110), and I'm seriously considering buying it. I just > would like some confirmation if it can work under Wine, even if JACK > doesn't work and I have to use an external hardware recorder to > capture the speaker outputs or something. :P Yes, I'm still that desperate after this sample library. I'll trust the message that got my hopes up, hinting that Kontakt works on Wine. I decided to take the risk and bought the library. If all else fails, I can resort to my Windows partition, I guess (though I sincerely hope I won't need to do that. I really wanted to use this string library, and really wanted it to be on Linux :D ). Soon after the purchase, I got an email with the serial code, saying: > You can begin downloading right away. First, please install the latest > version of the Continuata Connect Download utility, our high-speed library > downloader, using the link below. > > http://continuata.net/download_app.php > > Just click the icon for your OS (PC or Mac - 64-bit OSX only). Once you've > downloaded and installed the Connect utility, run it and then copy-paste > your Download Code directly into the download code box, using the Paste > button. So, no download links. Using the app is really required... We have to install the app and paste the serial code into it, which it uses to identify and retrieve the file. I downloaded the Continuata Connect app and installed it through Wine. There was a "couldn't create directory" error at the end of the installation, but the program seems to run okay. I pasted the serial code into the program and it began downloading. The first file part went fine, but from the second part it decided to stop downloading every now and then, corrupting the file, then starting all over again. I don't know if that's supposed to be normal, but I don't believe it to be a Linux or Wine issue. Although I'm still stuck on the second file part, it seems it has chances of working, so I'll keep trying. If (no, "when") I get past the download issue, the next step is trying to install Kontakt Player. I'll post again when I have more news. PS: at the end of the email, they say: > If you can't install for whatever reason, get in touch and we'll get you up > and running as soon as humanly possible. I wonder if the "whatever reason" also includes "because I'm trying to run it on Linux", hahahaha :P. Jokes aside, if it indeed does work on Wine, letting them know might be a good idea too. -- ____________________ Blog: http://aiyumi.warpstar.net/ From brummer- at web.de Sun Jun 15 08:50:45 2014 From: brummer- at web.de (hermann meyer) Date: Sun, 15 Jun 2014 10:50:45 +0200 Subject: [LAU] Embertone Friedlander Violin on Linux - was Re: Recommended near-realistic strings section generator? In-Reply-To: References: Message-ID: <539D5E65.3040003@web.de> Am 14.06.2014 03:37, schrieb Aiyumi Moriya: > Does the free Kontakt 5 Player also work under Wine? There's this > >violin sample library that I'm interested for quite a while now: > > > >http://embertone.com/instruments/friedlanderviolin.php > > > >It is a solo violin library, but it also has an "ensemble" mode. I > >really liked the sound... Just for the case, / Glenn McArthur /Have a collection of .sfz library's online (mostly created by Jeff G) were you could find as well two violin sets. http://www.bandshed.net/sounds/sfz/ // -------------- next part -------------- An HTML attachment was scrubbed... URL: From aiyumi.br at gmail.com Tue Jun 17 00:53:45 2014 From: aiyumi.br at gmail.com (Aiyumi Moriya) Date: Mon, 16 Jun 2014 21:53:45 -0300 Subject: [LAU] Embertone Friedlander Violin on Linux - was Re: Recommended near-realistic strings section generator? In-Reply-To: <539D5E65.3040003@web.de> References: <539D5E65.3040003@web.de> Message-ID: 2014-06-15 5:50 GMT-03:00, hermann meyer : > Just for the case, / > Glenn McArthur /Have a collection of .sfz library's online (mostly > created by Jeff G) were you could find as well two violin sets. > http://www.bandshed.net/sounds/sfz/ I really appreciate their efforts on adding to the free sample libraries! I visit the samples section of the LinuxMusicians forums from time to time, to see if there's anything new. I think the solo violin from that set is too simple, only legato and with no way of controlling vibrato. There's also this: http://www.freesound.org/people/ldk1609/packs/ Not a library, but just the samples. There are vibrato and non-vibrato samples and a few articulations. If I remember right, it's the same one used in Sonatina (without the reverb), but I think they only used the vibrato samples (I might be wrong). With skill, patience and creativity, one probably should be able to get a nice SFZ out of these. And the good news is that... Jeff G seems to be doing just that! :D http://www.linuxmusicians.com/viewtopic.php?f=50&t=12530 (the download link seems broken though). ... But, I ended up buying the Embertone violin (held off doing that for almost one year :P), so now I'll have to find a way to use it. -- ____________________ Blog: http://aiyumi.warpstar.net/ From p8rpp at aol.com Tue Jun 17 18:06:57 2014 From: p8rpp at aol.com (Peter P.) Date: Tue, 17 Jun 2014 14:06:57 -0400 Subject: [LAU] Disabling IRQ #19 Message-ID: <20140617180655.GA19019@aol.com> Dear List, I have a strange and unreproducable error on my laptop, that halts the entire audio system randomly about once every three weeks. Jackd quits, with it all clients, and dmesg says: [21684.947293] irq 19: nobody cared (try booting with the "irqpoll"option) [21684.947303] Pid: 194, comm: irq/19-ehci_hcd Not tainted 3.2.0-4-rt-amd64 #1 Debian 3.2.54-2 [21684.947309] Call Trace: [21684.947327] [] ? __report_bad_irq+0x2c/0xb5 [21684.947336] [] ? note_interrupt+0x16f/0x1f2 [21684.947344] [] ? irq_thread_fn+0x32/0x32 [21684.947351] [] ? irq_thread_fn+0x32/0x32 [21684.947358] [] ? irq_thread+0x106/0x201 [21684.947367] [] ? irq_finalize_oneshot+0xb3/0xb3 [21684.947378] [] ? kthread+0x78/0x80 [21684.947385] [] ? get_parent_ip+0x9/0x1b [21684.947395] [] ? kernel_thread_helper+0x4/0x10 [21684.947405] [] ? rcu_read_unlock_sched_notrace+0x2a/0x2a [21684.947412] [] ? gs_change+0x13/0x13 [21684.947417] handlers: [21684.947429] [] irq_default_primary_handler threaded [] usb_hcd_irq [21684.947473] [] irq_default_primary_handler threaded [] ips_irq_handler [21684.947485] [] irq_default_primary_handler threaded [] snd_hdsp_interrupt [21684.947498] Disabling IRQ #19 The sound card is an RME Multiface via an ExpressCard in a Thinkpad X201s. The card itself had been working reliably since more than ten years by now (not with the ExpressCard though). Running on Debian RT Kernel. I have no way to set/change the IRQ assignments in this BIOS. Interrupt 19 is shared by modules ehci_hcd:usb2, ips, snd_hdsp, where the two others are USB and the thermal subsystem. I assume ehci_hcd:usb2 denotes the second USB bus. If this is correct, there are no devices connected to it at the moment, according to lsusb (except for USB hubs). Web research has shown that people suggest using the kernel option "irqpoll" or disable "threadirqs", use "noirqdebug" or "irqdebug" options (apparently quite expensive in computation), or the irqbalance package. https://bbs.archlinux.org/viewtopic.php?id=133327 gives a hint about deinstalling laptop-mode-tools, which I am running, and which made the problem go away for the poster. As the error occurs so infrequently, I can not yet verify if any of these help, and am hoping for some hints or experience from this list. Please, any help or ideas are desperately needed, as this is a production system that is used in live shows in front of large audiences, and I absolutely need to resolve that error. Thank you for all ideas, best, Peter From markus.seeber at spectralbird.de Tue Jun 17 18:49:50 2014 From: markus.seeber at spectralbird.de (Markus Seeber) Date: Tue, 17 Jun 2014 20:49:50 +0200 Subject: [LAU] Disabling IRQ #19 In-Reply-To: <20140617180655.GA19019@aol.com> References: <20140617180655.GA19019@aol.com> Message-ID: <53A08DCE.80601@spectralbird.de> On 06/17/2014 08:06 PM, Peter P. wrote: > Dear List, > > I have a strange and unreproducable error on my laptop, that halts the > entire audio system randomly about once every three weeks. > > Jackd quits, with it all clients, and dmesg says: > > [21684.947293] irq 19: nobody cared (try booting with the "irqpoll"option) > [21684.947303] Pid: 194, comm: irq/19-ehci_hcd Not tainted 3.2.0-4-rt-amd64 #1 Debian 3.2.54-2 > [21684.947309] Call Trace: > [21684.947327] [] ? __report_bad_irq+0x2c/0xb5 > [21684.947336] [] ? note_interrupt+0x16f/0x1f2 > [21684.947344] [] ? irq_thread_fn+0x32/0x32 > [21684.947351] [] ? irq_thread_fn+0x32/0x32 > [21684.947358] [] ? irq_thread+0x106/0x201 > [21684.947367] [] ? irq_finalize_oneshot+0xb3/0xb3 > [21684.947378] [] ? kthread+0x78/0x80 > [21684.947385] [] ? get_parent_ip+0x9/0x1b > [21684.947395] [] ? kernel_thread_helper+0x4/0x10 > [21684.947405] [] ? rcu_read_unlock_sched_notrace+0x2a/0x2a > [21684.947412] [] ? gs_change+0x13/0x13 > [21684.947417] handlers: > [21684.947429] [] irq_default_primary_handler threaded [] usb_hcd_irq > [21684.947473] [] irq_default_primary_handler threaded [] ips_irq_handler > [21684.947485] [] irq_default_primary_handler threaded [] snd_hdsp_interrupt > [21684.947498] Disabling IRQ #19 > > The sound card is an RME Multiface via an ExpressCard in a Thinkpad > X201s. The card itself had been working reliably since more than ten > years by now (not with the ExpressCard though). Running on Debian RT > Kernel. > > I have no way to set/change the IRQ assignments in this BIOS. > Interrupt 19 is shared by modules ehci_hcd:usb2, ips, snd_hdsp, > where the two others are USB and the thermal subsystem. > I assume ehci_hcd:usb2 denotes the second USB bus. If this is correct, > there are no devices connected to it at the moment, according to lsusb > (except for USB hubs). > > Web research has shown that people suggest using the kernel option > "irqpoll" or disable "threadirqs", use "noirqdebug" or "irqdebug" options > (apparently quite expensive in computation), or the irqbalance > package. > > https://bbs.archlinux.org/viewtopic.php?id=133327 gives a hint about > deinstalling laptop-mode-tools, which I am running, and which made the > problem go away for the poster. > > As the error occurs so infrequently, I can not yet verify if any of > these help, and am hoping for some hints or experience from this list. > > Please, any help or ideas are desperately needed, as this is a > production system that is used in live shows in front of large > audiences, and I absolutely need to resolve that error. > > Thank you for all ideas, > best, Peter > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > You are not alone with this issue, I tried to work around this on different hardware by turning off the devices that share the IRQ in BIOS. I suppose that's not a good idea or even impossible with the thermal subsystem, but it _seems_ to work on my system. In my case there was (probably) a USB device causing problems, but due to the dodgy nature of the problem, I could not reproduce it or verify a working solution. As a note, I was also using Kernel 3.2. maybe more recent Kernel versions don't have this issue? Can you trigger the Problem somehow? Maybe by connecting and disconnecting USB devices? If you try to reproduce this, please take notes on what you are doing and how. Greetings Markus From ralf.mardorf at rocketmail.com Tue Jun 17 19:42:44 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Tue, 17 Jun 2014 21:42:44 +0200 Subject: [LAU] Disabling IRQ #19 In-Reply-To: <53A08DCE.80601@spectralbird.de> References: <20140617180655.GA19019@aol.com> <53A08DCE.80601@spectralbird.de> Message-ID: <1403034164.15126.164.camel@archlinux> On Tue, 2014-06-17 at 20:49 +0200, Markus Seeber wrote: > On 06/17/2014 08:06 PM, Peter P. wrote: > > I have no way to set/change the IRQ assignments in this BIOS. > > Interrupt 19 is shared by modules ehci_hcd:usb2, ips, snd_hdsp, > > where the two others are USB and the thermal subsystem. > > I assume ehci_hcd:usb2 denotes the second USB bus. If this is correct, > > there are no devices connected to it at the moment, according to lsusb > > (except for USB hubs). > You are not alone with this issue, > I tried to work around this on different hardware by turning off the > devices that share the IRQ in BIOS. It's also possible to unbind devices by command line. On my machine the BIOS is useless too. Assumed the links mentioned in my script are still available, one of the scripts describes how to unbind a device. [rocketmouse at archlinux ~]$ cat /usr/local/sbin/tuning #!/bin/bash [snip] ### http://www.mythtv.org/wiki/PCI_Latency ### http://wiki.linuxmusicians.com/doku.php?id=system_configuration#pci_bus_latency [snip] ### Unbinding devices echo -n "0000:00:13.2" > /sys/bus/pci/drivers/ohci_hcd/unbind echo -n "0000:00:13.4" > /sys/bus/pci/drivers/ohci_hcd/unbind [snip] exit 0 From ralf.mardorf at rocketmail.com Tue Jun 17 19:53:39 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Tue, 17 Jun 2014 21:53:39 +0200 Subject: [LAU] Disabling IRQ #19 In-Reply-To: <1403034164.15126.164.camel@archlinux> References: <20140617180655.GA19019@aol.com> <53A08DCE.80601@spectralbird.de> <1403034164.15126.164.camel@archlinux> Message-ID: <1403034819.15126.166.camel@archlinux> On Tue, 2014-06-17 at 21:42 +0200, Ralf Mardorf wrote: > On Tue, 2014-06-17 at 20:49 +0200, Markus Seeber wrote: > > On 06/17/2014 08:06 PM, Peter P. wrote: > > > I have no way to set/change the IRQ assignments in this BIOS. > > > Interrupt 19 is shared by modules ehci_hcd:usb2, ips, snd_hdsp, > > > where the two others are USB and the thermal subsystem. > > > I assume ehci_hcd:usb2 denotes the second USB bus. If this is correct, > > > there are no devices connected to it at the moment, according to lsusb > > > (except for USB hubs). > > > You are not alone with this issue, > > I tried to work around this on different hardware by turning off the > > devices that share the IRQ in BIOS. > > It's also possible to unbind devices by command line. On my machine the > BIOS is useless too. > > Assumed the links mentioned in my script are still available, one of the > scripts describes how to unbind a device. ^^^^^^^ links ;) > ### http://wiki.linuxmusicians.com/doku.php?id=system_configuration#pci_bus_latency It's this link and it's still available. Take a look at the "Solve IRQ conflicts by unbinding devices" section of the link. From ralf.mardorf at rocketmail.com Tue Jun 17 20:12:45 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Tue, 17 Jun 2014 22:12:45 +0200 Subject: [LAU] Disabling IRQ #19 In-Reply-To: <1403034819.15126.166.camel@archlinux> References: <20140617180655.GA19019@aol.com> <53A08DCE.80601@spectralbird.de> <1403034164.15126.164.camel@archlinux> <1403034819.15126.166.camel@archlinux> Message-ID: <1403035965.15126.170.camel@archlinux> On Tue, 2014-06-17 at 21:53 +0200, Ralf Mardorf wrote: > On Tue, 2014-06-17 at 21:42 +0200, Ralf Mardorf wrote: > > On Tue, 2014-06-17 at 20:49 +0200, Markus Seeber wrote: > > > On 06/17/2014 08:06 PM, Peter P. wrote: > > > > I have no way to set/change the IRQ assignments in this BIOS. > > > > Interrupt 19 is shared by modules ehci_hcd:usb2, ips, snd_hdsp, > > > > where the two others are USB and the thermal subsystem. > > > > I assume ehci_hcd:usb2 denotes the second USB bus. If this is correct, > > > > there are no devices connected to it at the moment, according to lsusb > > > > (except for USB hubs). > > > > > You are not alone with this issue, > > > I tried to work around this on different hardware by turning off the > > > devices that share the IRQ in BIOS. > > > > It's also possible to unbind devices by command line. On my machine the > > BIOS is useless too. > > > > Assumed the links mentioned in my script are still available, one of the > > scripts describes how to unbind a device. > ^^^^^^^ links ;) > > > ### http://wiki.linuxmusicians.com/doku.php?id=system_configuration#pci_bus_latency > > It's this link and it's still available. Take a look at the "Solve IRQ > conflicts by unbinding devices" section of the link. Two replies to myself, to help others, are allowed? :D JFTR, if you unbind USB slots for a tower PC mobo you likely could get other USB slots by spending less money: http://www.reichelt.de/AK-666/3/index.html?&ACTION=3&LA=446&ARTICLE=79819&artnr=AK+666&SEARCH=usb+blende http://www.reichelt.de/AK-674-2/3/index.html?&ACTION=3&LA=446&ARTICLE=45871&artnr=AK+674%2F2&SEARCH=usb+blende From ivan_521521 at yahoo.com Wed Jun 18 01:17:09 2014 From: ivan_521521 at yahoo.com (Ivan K) Date: Tue, 17 Jun 2014 18:17:09 -0700 Subject: [LAU] Microphone Pre-amp with an S/PDIF output? Message-ID: <1403054229.51078.YahooMailNeo@web122606.mail.ne1.yahoo.com> The time has come for myself to acquire a microphone pre-amp. I do not require more than two channels and in fact, will probably do mostly mono recordings. My current audio card is a an M-Audio 2496, which has RCA and S/PDIF inputs. Due to the phasing out of PCI slots, it seems that USB interfaces are what most people are choosing to purchase these days. However, I was thinking about getting a microphone pre-amp with an s/pdif output.? My thinking is: ?(1) One will always be able to buy PCIe cards in the ?future with S/PDIF ?(2) I don't have to worry about drivers.? Just plug ?this pre-amp into the s/pdif input. However, it seems that the options for such a pre-amp are limited.? I found the following items: ?? (1) ART DPS-II Digital/Tube $265 ?? The price does not seem unreasonable for a quality pre-amp. ?? This thing does have a lot of knobs though, and therefore ?? probably has more functionality than I know what to do with. ?? (2) Samson C-Valve Tube Microphone Preamp $99 ?? This item is discontinued but still available on amazon. ?? This item did not review well on this list, but ... ?? it is inexpensive. Is anyone aware of other options? So what do people think of this plan?? Or do people recommend that I just go ahead and get a USB interface? Thank you for your help. From aiyumi.br at gmail.com Wed Jun 18 01:49:36 2014 From: aiyumi.br at gmail.com (Aiyumi Moriya) Date: Tue, 17 Jun 2014 22:49:36 -0300 Subject: [LAU] Embertone Friedlander Violin on Linux - was Re: Recommended near-realistic strings section generator? In-Reply-To: References: <539D5E65.3040003@web.de> Message-ID: Back to the Embertone library. Oops. Outdated information. I just realized I had the original price wrong. I said $110, but that was several months ago. Now it's $120 for the 16-bit version and $125 for 24-bit. Anyway, some progress. A good and a bad news: the good news is that I was able to download the files. The bad is that (as much as I didn't want to do it), I had to resort to Windows. I tried on Wine a few more times, but the download always hung and the file always got corrupted. So, I tried using the app on Windows to see if there was any difference, planning to contact support if there wasn't. And sadly, there was... On Windows, everything downloaded without complaint, no hangs and no file corruptions. Looks like it was a Wine issue, after all... That means I got past the download issue, but it would be nice to figure out why it wasn't working on Wine. I tried again to see if I discovered something. Strange results. Wine 1.7.12 on Slackware 64-bit with multilib: * With Wine set to "Windows XP" in Winecfg, the Connect app opens and runs, but it's the same problem I said before. The download hangs from time to time. It even continues when pressing the resume button, but when the download gets to 100%, it says there was an error and it restarts from 0%. * With Wine set to "Windows 7", the program doesn't even start. It says there was an abnormal error and crashes. The Continuata site says it's a Java program wrapped in a binary executable, so it should run on "any" computer (they mean with Windows or Mac of course :P). Some googling returned an FAQ from another sample library developer: http://store.agsoundtrax.com/download-and-installation/ My problem on Wine sounds suspiciously like this question: ""The Connect utility is unable to successfully download files and shows a "DL Error" or "Install Error" warning. If you're getting incomplete or corrupted downloads or the utility displays a "DL Error" or "Install Error" status message to the right of a file after more than 3 automatic retries, that may be a sign of router or ISP issues. Try rebooting your modem and router first and then check to see if your ISP has imposed a bandwidth cap on your connection. It is becoming more common for ISPs around the world to limit large file downloads. If you have a monthly bandwidth or data limit with your ISP and you think you might be close to reaching it, please check with your ISP before ordering or make sure order the DVD version of the library." I don't know if it's my ISP, but I never had any problem with other downloads before, plus, on Windows the app worked fine (same machine, same modem, same router, same configuration). Maybe it's indeed a Wine issue, or a Wine internet connection issue... I just found other posts hinting that Kontakt can run on Wine and am more confident now. There's still this download issue though. But as per the link above, as well as some forum posts by other people having problems with this app, it looks like the developers usually provide the manual download links if the customer requests them. It may be worth contacting Embertone to see if they provide the direct download links. If so, no need to resort to Windows. There is still hope for us Linux users :D . Anyway, I moved the downloaded files to my Linux partition. However, now I'm struggling with Kontakt Player. Any chance it can work without JACK+Wineasio? I'm on a multilib 64-bit system but still don't have 32-bit JACK nor Wineasio. Kontakt Player installed fine, but I'm stuck on the audio setup screen. There's "Wasapi (Shared mode)" as the only audio driver, status says "stopped", my onboard soundcard appears on the dropdown list, my MIDI ports get recognized, but no matter what is done, it complains that I need to set a "valid" audio interface... I don't know what's happening. Wine plays sounds from other apps just fine. Can't it work with the default driver, without JACK and Wineasio (not considering latency or anything yet), or am I missing something else? By what I got from reading their "Getting Started" guide, I can't add libraries if I don't get past this screen... -- ____________________ Blog: http://aiyumi.warpstar.net/ From len at ovenwerks.net Wed Jun 18 04:28:28 2014 From: len at ovenwerks.net (Len Ovens) Date: Tue, 17 Jun 2014 21:28:28 -0700 (PDT) Subject: [LAU] Microphone Pre-amp with an S/PDIF output? In-Reply-To: <1403054229.51078.YahooMailNeo@web122606.mail.ne1.yahoo.com> References: <1403054229.51078.YahooMailNeo@web122606.mail.ne1.yahoo.com> Message-ID: On Tue, 17 Jun 2014, Ivan K wrote: > My current audio card is a an M-Audio 2496, which has > RCA and S/PDIF inputs. > > Due to the phasing out of PCI slots, it seems > that USB interfaces are what most people are choosing > to purchase these days. > > However, I was thinking about getting a microphone pre-amp > with an s/pdif output.? My thinking is: That is what I do. I have an ART USBDualTube Pre which also has s/pdif out. The only thing I don't like about the unit is that there is no external sync on it. > ?(1) One will always be able to buy PCIe cards in the > ?future with S/PDIF I am not seeing this... in my price range :) > However, it seems that the options for such a pre-amp > are limited.? I found the following items: > > ?? (1) ART DPS-II Digital/Tube $265 > ?? The price does not seem unreasonable for a quality pre-amp. > ?? This thing does have a lot of knobs though, and therefore > ?? probably has more functionality than I know what to do with. I have been happy with my unit which is simpler but that one has clock in. I have heard that inout impedance is good for ribon mics too. I can't speak of the quality as I have not used one. There are really not so many knobs as you suppose: - input imedance - tube stage gain - output level - Phantom power - phase (not needed for mostly mono recording, but if you ever do vocals and guitar together...) - V3 is a lot about how distorted the signal gets ;) It allows The tube to act as a limiter. They call it valve voicing. It is one of those set it to where it "sounds nice" things. > ?? (2) Samson C-Valve Tube Microphone Preamp $99 > ?? This item is discontinued but still available on amazon. > ?? This item did not review well on this list, but ... > ?? it is inexpensive. Don't know. > So what do people think of this plan?? Or do people recommend > that I just go ahead and get a USB interface? USB support is up and down (I see a lot of alsa bug reports) minimum latency will be a bit higher than with a pci card. I would tend to choose firewire over USB, though the cost is a little higher. They seem to be more solid/stable. As I have done the same thing (I made sure my new MB had PCI slots), I am probably biased :) USB interupts and internal hubs make finding the right port to use a little more difficult, but I was able to get mine running with jack at 64/2 in a stable fashion... The ICE1712 PCI will do jack at 16/2 with no xruns wich is nice for using it for live effects/midi but probably not really needed for recording. -- Len Ovens www.ovenwerks.net From danni.coy at gmail.com Wed Jun 18 08:13:37 2014 From: danni.coy at gmail.com (Danni Coy) Date: Wed, 18 Jun 2014 18:13:37 +1000 Subject: [LAU] Recommended near-realistic strings section generator? In-Reply-To: References: Message-ID: yes kontakt player does work under wine I had problems with connecting midi doing it that way though. I am able to perform on my keyboard using festige and the kontakt vst plugin (part of the player) On Wed, Jun 11, 2014 at 10:39 PM, Aiyumi Moriya wrote: > 2014-05-26 1:30 GMT-03:00, Danni Coy : >> I have had some luck with Kontact and festige > > Really? :D That got my hopes up. > > Does the free Kontakt 5 Player also work under Wine? There's this > violin sample library that I'm interested for quite a while now: > > http://embertone.com/instruments/friedlanderviolin.php > > It is a solo violin library, but it also has an "ensemble" mode. I > really liked the sound... But It runs on Kontakt Player. I googled > about Kontakt 5 Player on Linux/Wine and didn't find anything useful. > Also, from what I understood from some reading, it seems to use a > software called Continuata for watermarking and downloading the > library. I've never heard of it before - I'm a beginner in the area of > (proprietary) sample libraries -, but from the forum posts I read, it > seems to require the user to install a "Continuata utility program" > that downloads and extracts the library. The posts I read were this > one: > > http://www.kvraudio.com/forum/viewtopic.php?p=5448245 > > and the one directly below it. > > But I googled about it too, and found a post about someone downloading > (not specifically this sample library) normally via the browser > without using the Continuata thing. > > http://www.kvraudio.com/forum/viewtopic.php?t=376934 > > Does that mean it isn't required, and only the developers need to use > it to watermark/upload or something? If it's really required, any idea > if that works under Wine? > > The library is on sale right now, costing $99 USD until June 15th > (before, it was $110), and I'm seriously considering buying it. I just > would like some confirmation if it can work under Wine, even if JACK > doesn't work and I have to use an external hardware recorder to > capture the speaker outputs or something. :P > > Thank you to anyone that chimes in! :) > > > -- > ____________________ > > Blog: http://aiyumi.warpstar.net/ > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user From danni.coy at gmail.com Wed Jun 18 08:15:25 2014 From: danni.coy at gmail.com (Danni Coy) Date: Wed, 18 Jun 2014 18:15:25 +1000 Subject: [LAU] Recommended near-realistic strings section generator? In-Reply-To: References: Message-ID: If memory serves me correctly I did have difficulty installing it from the Linux side. I did have to do the initial installation in windows. This might not be a limitation of the free player though On Wed, Jun 18, 2014 at 6:13 PM, Danni Coy wrote: > yes kontakt player does work under wine I had problems with connecting > midi doing it that way though. > I am able to perform on my keyboard using festige and the kontakt vst > plugin (part of the player) > > On Wed, Jun 11, 2014 at 10:39 PM, Aiyumi Moriya wrote: >> 2014-05-26 1:30 GMT-03:00, Danni Coy : >>> I have had some luck with Kontact and festige >> >> Really? :D That got my hopes up. >> >> Does the free Kontakt 5 Player also work under Wine? There's this >> violin sample library that I'm interested for quite a while now: >> >> http://embertone.com/instruments/friedlanderviolin.php >> >> It is a solo violin library, but it also has an "ensemble" mode. I >> really liked the sound... But It runs on Kontakt Player. I googled >> about Kontakt 5 Player on Linux/Wine and didn't find anything useful. >> Also, from what I understood from some reading, it seems to use a >> software called Continuata for watermarking and downloading the >> library. I've never heard of it before - I'm a beginner in the area of >> (proprietary) sample libraries -, but from the forum posts I read, it >> seems to require the user to install a "Continuata utility program" >> that downloads and extracts the library. The posts I read were this >> one: >> >> http://www.kvraudio.com/forum/viewtopic.php?p=5448245 >> >> and the one directly below it. >> >> But I googled about it too, and found a post about someone downloading >> (not specifically this sample library) normally via the browser >> without using the Continuata thing. >> >> http://www.kvraudio.com/forum/viewtopic.php?t=376934 >> >> Does that mean it isn't required, and only the developers need to use >> it to watermark/upload or something? If it's really required, any idea >> if that works under Wine? >> >> The library is on sale right now, costing $99 USD until June 15th >> (before, it was $110), and I'm seriously considering buying it. I just >> would like some confirmation if it can work under Wine, even if JACK >> doesn't work and I have to use an external hardware recorder to >> capture the speaker outputs or something. :P >> >> Thank you to anyone that chimes in! :) >> >> >> -- >> ____________________ >> >> Blog: http://aiyumi.warpstar.net/ >> _______________________________________________ >> Linux-audio-user mailing list >> Linux-audio-user at lists.linuxaudio.org >> http://lists.linuxaudio.org/listinfo/linux-audio-user From rtg at aapsc.com Wed Jun 18 11:45:04 2014 From: rtg at aapsc.com (Rick Green) Date: Wed, 18 Jun 2014 07:45:04 -0400 (EDT) Subject: [LAU] Microphone Pre-amp with an S/PDIF output? In-Reply-To: References: <1403054229.51078.YahooMailNeo@web122606.mail.ne1.yahoo.com> Message-ID: On Tue, 17 Jun 2014, Len Ovens wrote: > On Tue, 17 Jun 2014, Ivan K wrote: > >> However, I was thinking about getting a microphone pre-amp >> with an s/pdif output.? My thinking is: > > That is what I do. I have an ART USBDualTube Pre which also has s/pdif out. > The only thing I don't like about the unit is that there is no external sync > on it. I also have one of the USBDualTubePre units. I use its SPDIF output regularly, and find it fits the bill nicely for my needs, as well. And its less than $100USD. > >> So what do people think of this plan?? Or do people recommend >> that I just go ahead and get a USB interface? > The USBDualTubePre also is a class-compliant USB 1.1 stereo audio interface, which plays nice with my linux laptop... -- Rick Green We, the People of the United States of America, reject the U.S. Supreme Court's Citizens United ruling, and move to amend our Constitution to firmly establish that money is not speech, and that human beings, not corporations, are persons entitled to constitutional rights. http://www.MoveToAmend.org -------------- next part -------------- _______________________________________________ Linux-audio-user mailing list Linux-audio-user at lists.linuxaudio.org http://lists.linuxaudio.org/listinfo/linux-audio-user From hanswil at notam02.no Wed Jun 18 11:54:45 2014 From: hanswil at notam02.no (Hans Wilmers) Date: Wed, 18 Jun 2014 13:54:45 +0200 Subject: [LAU] Microphone Pre-amp with an S/PDIF output? In-Reply-To: <1403054229.51078.YahooMailNeo@web122606.mail.ne1.yahoo.com> References: <1403054229.51078.YahooMailNeo@web122606.mail.ne1.yahoo.com> Message-ID: <53A17E05.2030404@notam02.no> On 06/18/2014 03:17 AM, Ivan K wrote: > > The time has come for myself to acquire a microphone pre-amp. > I do not require more than two channels and in fact, will > probably do mostly mono recordings. > > My current audio card is a an M-Audio 2496, which has > RCA and S/PDIF inputs. > > Due to the phasing out of PCI slots, it seems > that USB interfaces are what most people are choosing > to purchase these days. > > However, I was thinking about getting a microphone pre-amp > with an s/pdif output. My thinking is: > > (1) One will always be able to buy PCIe cards in the > future with S/PDIF > > (2) I don't have to worry about drivers. Just plug > this pre-amp into the s/pdif input. > > However, it seems that the options for such a pre-amp > are limited. I found the following items: > > (1) ART DPS-II Digital/Tube $265 > The price does not seem unreasonable for a quality pre-amp. > This thing does have a lot of knobs though, and therefore > probably has more functionality than I know what to do with. > > (2) Samson C-Valve Tube Microphone Preamp $99 > This item is discontinued but still available on amazon. > This item did not review well on this list, but ... > it is inexpensive. > > Is anyone aware of other options? > Sounddevices USBPre2: http://www.sounddevices.com/products/usbpre2/key-features/ Superb quality, and doubles as a USB interface. But it has become quite expensive recently. / Hans --- Hans Wilmers NOTAM Sandakerveien 24 D, bygg F3 N-0473 Oslo Norway tlf.: +47 22358065 mob.: +47 92459361 http://www.notam02.no From aiyumi.br at gmail.com Wed Jun 18 12:09:06 2014 From: aiyumi.br at gmail.com (Aiyumi Moriya) Date: Wed, 18 Jun 2014 09:09:06 -0300 Subject: [LAU] Recommended near-realistic strings section generator? In-Reply-To: References: Message-ID: 2014-06-18 5:13 GMT-03:00, Danni Coy : > yes kontakt player does work under wine I had problems with connecting > midi doing it that way though. Thank you for the clarification :D . 2014-06-18 5:15 GMT-03:00, Danni Coy : > If memory serves me correctly I did have difficulty installing it from > the Linux side. I did have to do the initial installation in windows. I gave it a try. Here, installation went fine. But I'm stuck on the initial configuration (the "audio and MIDI settings" screen that appears when Kontakt is run for the first time). This is the problem I'm having (from the other thread): 2014-06-17 22:49 GMT-03:00, Aiyumi Moriya : > now I'm struggling with Kontakt Player. Any chance it can work without > JACK+Wineasio? I'm on a multilib 64-bit system but still don't have > 32-bit JACK nor Wineasio. Kontakt Player installed fine, but I'm stuck > on the audio setup screen. There's "Wasapi (Shared mode)" as the only > audio driver, status says "stopped", my onboard soundcard appears on > the dropdown list, my MIDI ports get recognized, but no matter what is > done, it complains that I need to set a "valid" audio interface... I > don't know what's happening. Wine plays sounds from other apps just > fine. Can't it work with the default driver, without JACK and Wineasio > (not considering latency or anything yet), or am I missing something > else? By what I got from reading their "Getting Started" guide, I > can't add libraries if I don't get past this screen... After that, I gave it some more thought. The "Getting Started" guide says it "skips" this audio configuration if used as VST (the audio is left up to the host). Would it be possible to add libraries through the VST interface? -- ____________________ Blog: http://aiyumi.warpstar.net/ From jh at brainiac.com Wed Jun 18 12:26:55 2014 From: jh at brainiac.com (Joe Hartley) Date: Wed, 18 Jun 2014 08:26:55 -0400 Subject: [LAU] Video card joy Message-ID: <20140618082655.1ddc62b9dd7ed1051bc45d60@brainiac.com> Because who else would even care besides you lot? After going for years with anemic video cards because the "good" ones would sound like a jet taking off, I splurged and got an Asus GeForce GE640 silent. No fans, just fins and heat pipes, and a GPU that isn't close to EOL already. Now, great performance on my two monitors and sweet, sweet silence. And having replaced video cards in Windows machines and having the OS freak the frak out about a system change, a renewed appreciation for Linux. I'd pre-installed the nouveau packages from Arch, and once the new card was in place, two simple config file changes and a mkinitcpio later I was rocking it. -- ====================================================================== Joe Hartley - UNIX/network Consultant - jh at brainiac.com Without deviation from the norm, "progress" is not possible. - FZappa From ivan_521521 at yahoo.com Wed Jun 18 17:36:27 2014 From: ivan_521521 at yahoo.com (Ivan K) Date: Wed, 18 Jun 2014 10:36:27 -0700 Subject: [LAU] Microphone Pre-amp with an S/PDIF output? In-Reply-To: <53A17E05.2030404@notam02.no> References: <1403054229.51078.YahooMailNeo@web122606.mail.ne1.yahoo.com> <53A17E05.2030404@notam02.no> Message-ID: <1403112987.31203.YahooMailNeo@web122601.mail.ne1.yahoo.com> Rick Green writes: > > On Tue, 17 Jun 2014, Len Ovens wrote: > > > That is what I do. I have an ART USBDualTube Pre which also > > has s/pdif out.? The only thing I don't like about the unit > > is that there is no external sync on it. > > I also have one of the USBDualTubePre units.? I use its SPDIF output > regularly, and find it fits the bill nicely for my needs, as well.? And > its less than $100USD. Thank you both for the recommendation.? I of course found this item: http://artproaudio.com/art_products/signal_processing/usb_audio_devices/product/usbdualtubepre/ But because it had the term "USB" in the name, I had just assumed that was the only way to connect it to your computer.? Yes, It is quite inexpensive though it looks like I better purchase it quick as this item has been discontinued. Thanks again!. From ivan_521521 at yahoo.com Wed Jun 18 17:42:48 2014 From: ivan_521521 at yahoo.com (Ivan K) Date: Wed, 18 Jun 2014 10:42:48 -0700 Subject: [LAU] Microphone Pre-amp with an S/PDIF output? In-Reply-To: References: <1403054229.51078.YahooMailNeo@web122606.mail.ne1.yahoo.com> Message-ID: <1403113368.77091.YahooMailNeo@web122603.mail.ne1.yahoo.com> Len Ovens writes: > > > > ?? (1) ART DPS-II Digital/Tube $265 > [...] There are really not so many knobs as you suppose: [...] Thank you for the explanation, as well as the USB review. From len at ovenwerks.net Wed Jun 18 20:33:32 2014 From: len at ovenwerks.net (Len Ovens) Date: Wed, 18 Jun 2014 13:33:32 -0700 (PDT) Subject: [LAU] Microphone Pre-amp with an S/PDIF output? In-Reply-To: References: <1403054229.51078.YahooMailNeo@web122606.mail.ne1.yahoo.com> Message-ID: On Wed, 18 Jun 2014, Rick Green wrote: > On Tue, 17 Jun 2014, Len Ovens wrote: > The USBDualTubePre also is a class-compliant USB 1.1 stereo audio interface, > which plays nice with my linux laptop... Which is to say that USB out is 16 bit and s/pdif is 24 bit. I found USB was nice and stable as well. -- Len Ovens www.ovenwerks.net From csanchezgs at gmail.com Thu Jun 19 17:14:05 2014 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Thu, 19 Jun 2014 19:14:05 +0200 Subject: [LAU] audio recording of my set at LAC 2014 (streaming, download) In-Reply-To: References: Message-ID: I'm also listening to it right now. Interesting and curious, it'd be nice to see a video of you live-coding to see how's the process. Thanks for sharing to us poor mortals that couldn't go to LAC. 2014-05-12 22:19 GMT+02:00 Matej Fr?be : > Great, I'm listening to the recording right now. I like your approach to > generating rhythms. > > Regards, > Matej > > > 2014-05-11 16:31 GMT+02:00 Renick Bell : >> >> Sorry, I messed up that mp3 link: >> >> http://renickbell.net/sound/renick-bell-live-lac-2014.mp3 >> >> On Sun, May 11, 2014 at 11:30 PM, Renick Bell wrote: >> > I'm really glad that I had a chance to perform at LAC this year. >> > Thanks to everyone in the audience! >> > >> > Here's a link to a streaming version on SoundCloud: >> > >> > >> > https://soundcloud.com/renick/live-at-the-linux-audio-conference-2014-karlsruhe-germany-may-5th-2014 >> > >> > If you prefer, here are direct download links in a variety of formats: >> > >> > ogg: http://renickbell.net/sound/renick-bell-live-lac-2014.ogg >> > >> > flac: http://renickbell.net/sound/renick-bell-live-lac-2014.flac >> > >> > mp3: http://renick/renickbell.net/sound/renick-bell-live-lac-2014.mp3 >> > >> > I'll put a video online after I've had a chance to edit it. >> > >> > Best to everyone, >> > >> > Renick >> > >> > -- >> > Renick Bell >> > - http://renickbell.net >> > - http://twitter.com/renick >> > - http://the3rd2nd.com >> >> >> >> -- >> Renick Bell >> - http://renickbell.net >> - http://twitter.com/renick >> - http://the3rd2nd.com >> _______________________________________________ >> Linux-audio-user mailing list >> Linux-audio-user at lists.linuxaudio.org >> http://lists.linuxaudio.org/listinfo/linux-audio-user > > > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > -- C. sanchiavedraZ: * NEW / NUEVO: www.sanchiavedraZ.com * Musix GNU+Linux: www.musix.es From l.gabrielli at univpm.it Thu Jun 19 17:31:36 2014 From: l.gabrielli at univpm.it (Leonardo Gabrielli) Date: Thu, 19 Jun 2014 19:31:36 +0200 Subject: [LAU] 4x4 usb audio interface Message-ID: <53A31E78.9000901@univpm.it> Dear all, have anyone had experience of a 4in 4out usb audio interface, working on Linux: - with all 4 channels (I can afford some tinkering if needed) - at least in USB1 compliant mode, i.e. stereo 16-bit 48kHz) I have seen the Alesis iO4 should work, as well as (maybe) the Focusrite 6i6, and the Presonus 44vsl, but I got no consistent report on neither of these. Leonardo From rncbc at rncbc.org Thu Jun 19 18:12:40 2014 From: rncbc at rncbc.org (Rui Nuno Capela) Date: Thu, 19 Jun 2014 19:12:40 +0100 Subject: [LAU] [ANN] QmidiNet 0.2.0 is out! Message-ID: <53A32818.4080801@rncbc.org> Headless finally! QmidiNet 0.2.0 is out! all that is to say that it may now run without the GUI, eg. qmidinet --no-gui QmidiNet [1] is a MIDI network gateway application that sends and receives MIDI data (ALSA-MIDI and JACK-MIDI) over the network, using UDP/IP multicast. Inspired by multimidicast [2] and designed to be compatible with ipMIDI [3] for Windows. See also: http://www.rncbc.org/drupal/node/790 Website: http://qmidinet.sourceforge.net Project pages: http://sourceforge.net/projects/qmidinet Downloads: - source tarballs: http://downloads.sourceforge.net/qmidinet/qmidinet-0.2.0.tar.gz - source package (openSUSE 13.1): http://downloads.sourceforge.net/qmidinet/qmidinet-0.2.0-3.rncbc.suse131.src.rpm - binary packages (openSUSE 13.1): http://downloads.sourceforge.net/qmidinet/qmidinet-0.2.0-3.rncbc.suse131.i586.rpm http://downloads.sourceforge.net/qmidinet/qmidinet-0.2.0-3.rncbc.suse131.x86_64.rpm Weblog (upstream support): http://www.rncbc.org License: QmidiNet is free, open-source software, distributed under the terms of the GNU General Public License (GPL) [4] version 2 or later. Change-log: - A man page has beed added (making up Alessio Treglia's work on debian, thanks). - First attempt to allow a headless application run mode, without GUI or system-tray icon accessibility, with all options given as command line arguments. - Allow the build system to include an user specified LDFLAGS. References: [1] QmidiNet - A MIDI Network Gateway via UDP/IP Multicast http://qmidinet.sourceforge.net [2] multimidicast - sends and receives MIDI from ALSA sequencers over network http://llg.cubic.org/tools/multimidicast [3] ipMIDI - MIDI over Ethernet ports - send MIDI over your LAN http://nerds.de [4] GPL - GNU General Public License http://www.gnu.org/copyleft/gpl.html Cheers && Enjoy -- rncbc aka Rui Nuno Capela rncbc at rncbc.org From brummer- at web.de Thu Jun 19 18:16:31 2014 From: brummer- at web.de (hermann meyer) Date: Thu, 19 Jun 2014 20:16:31 +0200 Subject: [LAU] New GxBluemann.lv2 Message-ID: <53A328FF.7060607@web.de> Hi I've made a new, simple basic guitar-amp lv2 plug. It's placed outside of the guitarix distribution, as a single plugin bundle. It comes without any fancy dependency (only make tools and gcc are needed). It comes without GUI. It is a tube screamer driving a powerful 2 stage 12ax7 tube amp with baxandall tone controls in the middle of the stages, followed by a cabinet. It needs here just ~5% DSP load. I'm very pleased with the sound of it, so I decide to share it with you, and keep it simple. I'm interested if this plug suite the needs of a bass player, as I didn't play bass, I cant really check this. But I think, with the baxandall controls it could be useful for bass as well. Simply run make to build it, run make install to install it (as user to ~./lv2 or as root (sudo) to /usr/lib/lv2, or run make deb to build a debian package and install that. http://sourceforge.net/projects/guitarix/files/lv2/gx_bluemann.lv2.tar.bz2/download From harryhaaren at gmail.com Thu Jun 19 20:46:51 2014 From: harryhaaren at gmail.com (Harry van Haaren) Date: Thu, 19 Jun 2014 21:46:51 +0100 Subject: [LAU] 4x4 usb audio interface In-Reply-To: <53A31E78.9000901@univpm.it> References: <53A31E78.9000901@univpm.it> Message-ID: On Thu, Jun 19, 2014 at 6:31 PM, Leonardo Gabrielli wrote: > - with all 4 channels (I can afford some tinkering if needed) > - at least in USB1 compliant mode, i.e. stereo 16-bit 48kHz) > I had an Esi Maya 44 USB: it only has (unbalanced) RCA in/out, but its class compliant USB 1.1: Always worked, no issues. If you don't need preams etc, I can recommend it, cost me 90 euros at the time IIRC. HTH, -Harry -------------- next part -------------- An HTML attachment was scrubbed... URL: From lol at sn0wcrash.net Thu Jun 19 22:18:09 2014 From: lol at sn0wcrash.net (Radovan Misovic) Date: Fri, 20 Jun 2014 00:18:09 +0200 (CEST) Subject: [LAU] EPH-O2 w/alsa - volume too low In-Reply-To: <1115804825.4072228.1403213411367.JavaMail.zimbra@sn0wcrash.net> Message-ID: <156556132.4073214.1403216289700.JavaMail.zimbra@sn0wcrash.net> Hi, I've recently purchased an Epiphany Acoustic O2 sound card, basically it's an USB DAC with a headphone amp. Problem is, the sound is barely audible. I've tested the amp independently (using the line in) and it works fine. So now I am trying to set up a preamp in alsa, but somehow I can't get it working. The config is below. I'd be grateful for any hints or ideas. Thanks, rad0 ------- asoundrc --------------------- # External USB DAC pcm.DAC { type hw card DAC } ctl.DAC { type hw card DAC } # Preamp on external DAC pcm.DAC { type plug slave.pcm "softvol" } pcm.softvol { type softvol slave { pcm "dmix" } control { name "preamp" card DAC } min_dB -5.0 max_dB 20.0 resolution 6 } ------- asoundrc --------------------- ++ Connection closed by remote ghost. From temcat at mail.ru Thu Jun 19 23:44:14 2014 From: temcat at mail.ru (Artem Vakhitov) Date: Fri, 20 Jun 2014 03:44:14 +0400 Subject: [LAU] Live bass guitar -> analog synth on a 2007 laptop: viable? Message-ID: <53A375CE.8040000@mail.ru> Hello fellow Linux audio users, I'm back to dabbling with Linux as an audio system. Among other things, I play bass guitar in a synth pop band and recently started to get interested in bass synthesizers. I could of course buy something like Markbass Super Synth, but then I thought - maybe I could cobble together something using my Samsung Q35 laptop and Linux? The laptop has a dual core processor and 2.5GB RAM. The sound card is a variable here: it could be the built-in Intel HDA (for this particular purpose, why not), or an Infrasonic DeuX (Firewire) that I have, or even some used Echo Indigo (PCMCIA). Is what I want viable at all with a reasonable latency? What software setup can I use for that? Does anybody here use something similar live? Regards, Artem Vakhitov From ralf.mardorf at rocketmail.com Fri Jun 20 00:10:37 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Fri, 20 Jun 2014 02:10:37 +0200 Subject: [LAU] Live bass guitar -> analog synth on a 2007 laptop: viable? In-Reply-To: <53A375CE.8040000@mail.ru> References: <53A375CE.8040000@mail.ru> Message-ID: <1403223037.1694.13.camel@archlinux> Rakarrak's audio to midi conversion does work for some kinds of guitar playing. I never tested it using a bass, it likely will become harder to use it the lower the string sound its, this already was an issue for old MIDI pickups. I don't know the quality of modern MIDI pickups, but I guess it's better you get such a pickup. Without a MIDI pickup the conversion only does work monophonic. As for the synth available for Linux, assumed there should be no xruns at usable latencies, the sound of an onboard audio device unlikely will be good enough for usage in a band, but the available synth might satisfy your needs assumed you're using a high quality sound card. From brummer- at web.de Fri Jun 20 05:09:26 2014 From: brummer- at web.de (hermann meyer) Date: Fri, 20 Jun 2014 07:09:26 +0200 Subject: [LAU] Live bass guitar -> analog synth on a 2007 laptop: viable? In-Reply-To: <53A375CE.8040000@mail.ru> References: <53A375CE.8040000@mail.ru> Message-ID: <53A3C206.8040606@web.de> Am 20.06.2014 01:44, schrieb Artem Vakhitov: > Hello fellow Linux audio users, > > I'm back to dabbling with Linux as an audio system. Among other > things, I play bass guitar in a synth pop band and recently started to > get interested in bass synthesizers. I could of course buy something > like Markbass Super Synth, but then I thought - maybe I could cobble > together something using my Samsung Q35 laptop and Linux? The laptop > has a dual core processor and 2.5GB RAM. The sound card is a variable > here: it could be the built-in Intel HDA (for this particular purpose, > why not), or an Infrasonic DeuX (Firewire) that I have, or even some > used Echo Indigo (PCMCIA). > > Is what I want viable at all with a reasonable latency? What software > setup can I use for that? Does anybody here use something similar live? > > Regards, > Artem Vakhitov > You may have a look at brian's amsynth site, he list and describe a couple of Linux soft synth's, give sound examples and patches for them. That could make it a bit easier for you to find what you are looking for. http://amsynth.com/ From csanchezgs at gmail.com Fri Jun 20 10:22:14 2014 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Fri, 20 Jun 2014 12:22:14 +0200 Subject: [LAU] New MOD Website In-Reply-To: References: Message-ID: 2014-05-07 0:00 GMT+02:00 Gianfranco Ceccolini : > Hi everybody > > Still recovering from the LAC2014 trip but already with news, here's the link to our new website (including english version) > > www.portalmod.com > > The video that was shown at the MOD presentation is right in the frontpage. > > Even better, the plugin library can now be navigated in the same way as in the MODs interface. On top of it, there's a dashboard section where visitors can try the interface without audio. > > Hope you all enjoy. > > Cheers > > Gianfranco Ceccolini > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user That promo is great, and the site. I've been following MOD for some time, really interesting and brave project. Some lines of my projects coincide with some of yours. I even sent you you quite some time ago an email with my congrats and a hint for possible collaborations, because I was developing in HTML5 something similar to that great interface you have achieved to connect pedals and effects. P.S: If I may say, switching my hat to webdev's hat, I would define more the fonts used for the site menu letters; they're too much blurred and it could be an accessibility issue. -- C. sanchiavedraZ: * NEW / NUEVO: www.sanchiavedraZ.com * Musix GNU+Linux: www.musix.es From csanchezgs at gmail.com Fri Jun 20 10:45:00 2014 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Fri, 20 Jun 2014 12:45:00 +0200 Subject: [LAU] Music made with fun thanx to linux In-Reply-To: References: Message-ID: 2014-05-16 18:10 GMT+02:00 Set Hallstr?m : > Hi! > > Recently i traded my earfull portastudio against memories of happy > unafortable OSystems some years ago, and then i made this: > https://soundcloud.com/sakrecoer/sets/music-made-with-linux > > I may hope you like it like i do, but i know life is full of inexpectations > :) > > With wishes of happy week endings, > > -- > Set Hallstr?m > AKA > reSet Sakrecoer > http://sakrecoer.com > > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > What a session, Almost 2 hours! Not my main kind of music but It sounds good, it develops and progress well. And the overall EQ and balance is also good. Thanks for sharing. -- C. sanchiavedraZ: * NEW / NUEVO: www.sanchiavedraZ.com * Musix GNU+Linux: www.musix.es From csanchezgs at gmail.com Fri Jun 20 10:53:00 2014 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Fri, 20 Jun 2014 12:53:00 +0200 Subject: [LAU] My debut EP, "Indigo" In-Reply-To: References: <1400107268.2675.117520949.01E4F900@webmail.messagingengine.com> Message-ID: 2014-05-29 20:21 GMT+02:00 Tim Goetze : > [William Light] >>I've posted up a number of my songs here over the years, and I'm very >>pleased to announce the availability of my first EP. I've posted a few >>of these songs here already, but these versions are updated and >>mastered, and, of course, there's plenty of new material as well. > > The remixed tracks are a bit too dense/busy for my taste but the > originals are great, especially "Swimming". > > Thanks! > Tim > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user I also like it. And as my collegues, I'd be also wating to howto videos or alike, or a list of programs used and effects. Kick sound punchy and bassy, as great some kickdrum with loose drumhead. Nice. -- C. sanchiavedraZ: * NEW / NUEVO: www.sanchiavedraZ.com * Musix GNU+Linux: www.musix.es From csanchezgs at gmail.com Fri Jun 20 11:12:02 2014 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Fri, 20 Jun 2014 13:12:02 +0200 Subject: [LAU] New EP - Everywhere Nowhere - indie rock In-Reply-To: <5394395D.3050606@vodafone.it> References: <5394395D.3050606@vodafone.it> Message-ID: 2014-06-08 12:22 GMT+02:00 Nicola : > Hi all, > > The EP of my band is out now! > It's been produced using Free and Open Source Software, Debian GNU/Linux + > Tango Studio Repositories and Ardour 3. > The license is CC BY-NC-SA > Feel free to listen and download it freely from soundcloud > https://soundcloud.com/bandage-indie-rock/sets/bandage-everywhere-nowhere > or Jamendo: > http://www.jamendo.com/en/list/a135179/everywhere-nowhere > > There's also a FLAC version available from Bandcamp > http://bandage-indierock.bandcamp.com/album/everywhere-nowhere at 1 euro. > > Opinions or suggestions are welcome! > > Thanks Linuxaudio community! > > Regards, > Nicola > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user Sounds wide and clean, and you can distinguish instruments. Good job. Any docu about the process? Thanks for sharing. -- C. sanchiavedraZ: * NEW / NUEVO: www.sanchiavedraZ.com * Musix GNU+Linux: www.musix.es From spamatica at gmail.com Fri Jun 20 12:29:12 2014 From: spamatica at gmail.com (Robert Jonsson) Date: Fri, 20 Jun 2014 14:29:12 +0200 Subject: [LAU] FaustLive is released ! In-Reply-To: References: Message-ID: Wow, this is fun! Now I really do need to learn Faust! So many fuzzes to create and so little time... Thanks /Robert 2014-06-12 11:24 GMT+02:00 St?phane Letz : > GRAME is happy to announce the official release of FaustLive. > > FaustLive is an advanced self-contained prototyping environment for the Faust programming language with an ultra-short edit-compile-run cycle. Thanks to its fully embedded compilation chain, FaustLive is simple to install and doesn't require any external compiler, development toolchain or SDK to run. > > FaustLive is the ideal tool for fast prototyping. Faust programs can be compiled and run on the fly by simple drag and drop. They can even be edited and recompiled while running without sound interruption or Jack disconnection. > > 1) Dynamic Compilation : > > On FaustLive?s windows you can drop your Faust code as a file, a string or a url. The code will be dynamically compiled and executed. > You can then choose to edit your code. It will be opened in the default editor for .dsp files (FOLLOW THE README TO CONFIGURE FILE ASSOCIATION). The application will be automatically recompiled, every time you save your document. > > A crossfade is calculated between two relaying applications in a window to avoid brutal sound interruptions. > > 2) Audio Drivers : > > Depending on your Operation System, you will have different drivers available: > > On OSX : Coreaudio, Jack and NetJack > On Linux : Jack and NetJack > On Windows : Portaudio > > You can then dynamically switch from one to another in FaustLive?s preferences. > > 3) Export Your DSP : > > Exporting your DSP as plugins is easy, thanks to FaustWeb, compilation service. In FaustLive?s export menu, you can find every platform and architecture that Faust can target. As you choose your target, your code is sent to FaustWeb and you receive the requested binary in exchange. > > 4) Save Snapshots : > > If you create a configuration you like, you can save it as a Snapshot. The state of FaustLive will be saved (running applications, Jack connections, interface parameters, ?). > > Later on, you will be able to whether : > - recall the snapshot : closing any running application to restore the saved state > - import the snapshot : adding the saved state to the current state > > 5) Remote Control Interfaces (only on Linux and OSX for now) : > > In the Windows Option toolBar, you can open a UDP port for OSC control or a TCP port for HTTP control. Moreover, the HTLM interface can be accessed through a QrCode that you can create from ?View QrCode ? in the menu ? Window ?. > > Download access: > > http://sourceforge.net/projects/faudiostream/files/ > > Tutorial video: > > http://www.youtube.com/watch?v=8ZUD2c5D-PU > > Sarah Denoux and St?phane Letz > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user From l.gabrielli at univpm.it Fri Jun 20 12:41:30 2014 From: l.gabrielli at univpm.it (Leonardo Gabrielli) Date: Fri, 20 Jun 2014 14:41:30 +0200 Subject: [LAU] Live bass guitar -> analog synth on a 2007 laptop:, viable? Message-ID: <53A42BFA.4030004@univpm.it> Also consider testing your sound cards prior to the software: if you can't get them to work, or you can't exploit very low period sizes (64-128) then all the software will come to no use. And don't forget about some tests with MIDI to audio conversion in Rakarrak at low pitches... These are the two most importante showstoppers IMO. If you want to go low latency, consider a clean distro, for some tests, I use Audiophile Linux (AP Linux), because the kernel comes patched and you have the bare minimum (also you can start with a handy fluxbox desktop manager with all the process info), or just go for a stable debian and maybe patch it RT PREEMPT. L. From spamatica at gmail.com Fri Jun 20 12:53:59 2014 From: spamatica at gmail.com (Robert Jonsson) Date: Fri, 20 Jun 2014 14:53:59 +0200 Subject: [LAU] Audio interface latency measurements Message-ID: Hello folks, Is there a site/repository somewhere with numbers on audio interface latency? I'm much interested in doing multichannel live audio processing and the device latency matters a lot in this case. I'm especially curious how recent USB audio interfaces provide in this regard. Anyone has any numbers? Lastly, here are some numbers for the firewire interfaces Edirol FA-66 and Presonus FIRESTUDIO Project I measured with jack_delay. FA-66 jackd -Z -t 9999 -d firewire -p 128 -r 44100 -X seq 618.760 frames 14.031 ms jackd -Z -t 9999 -d firewire -p 64 -r 44100 -X seq 420.924 frames 9.545 ms jackd -Z -t 9999 -d firewire -p 128 -r 48000 -X seq 618.780 frames 12.891 ms jackd -Z -t 9999 -d firewire -p 64 -r 48000 -X seq 440.792 frames 9.183 ms jackd -Z -t 9999 -d firewire -p 128 -r 96000 -X seq 724.485 frames 7.547 ms jackd -Z -t 9999 -d firewire -p 64 -r 96000 -X seq 560.492 frames 5.838 ms jackd -Z -t 9999 -d firewire -p 128 -r 192000 -X seq (unstable) 989.556 frames 5.154 ms FIRESTUDIO Project jackd -Z -t 9999 -d firewire -p 128 -r 44100 -X seq 543.411 frames 12.322 ms jackd -Z -t 9999 -d firewire -p 64 -r 44100 -X seq 351.412 frames 7.969 ms jackd -Z -t 9999 -d firewire -p 128 -r 48000 -X seq 543.438 frames 11.322 ms jackd -Z -t 9999 -d firewire -p 64 -r 48000 -X seq 351.438 frames 7.322 ms jackd -Z -t 9999 -d firewire -p 128 -r 96000 -X seq 566.642 frames 5.903 ms jackd -Z -t 9999 -d firewire -p 64 -r 96000 -X seq 374.642 frames 3.903 ms Regards, Robert From paul at linuxaudiosystems.com Fri Jun 20 13:07:37 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Fri, 20 Jun 2014 09:07:37 -0400 Subject: [LAU] Audio interface latency measurements In-Reply-To: References: Message-ID: On Fri, Jun 20, 2014 at 8:53 AM, Robert Jonsson wrote: > > Lastly, here are some numbers for the firewire interfaces Edirol FA-66 > and Presonus FIRESTUDIO Project I measured with jack_delay. > If you used jack_iodelay, it would do the math for you and show you how much extra latency is bing caused by the hardware (rather than the period size). -------------- next part -------------- An HTML attachment was scrubbed... URL: From fons at linuxaudio.org Fri Jun 20 13:19:10 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Fri, 20 Jun 2014 13:19:10 +0000 Subject: [LAU] Audio interface latency measurements In-Reply-To: References: Message-ID: <20140620131910.GC15318@linuxaudio.org> On Fri, Jun 20, 2014 at 09:07:37AM -0400, Paul Davis wrote: > If you used jack_iodelay, it would do the math for you and show you how > much extra latency is bing caused by the hardware (rather than the period > size). jack_delay will do this as well, use the -E option. jack_delay 0.4.0 (C) 2003-2013 Fons Adriaensen Measure round trip latency of a soundcard. Usage: jack_delay Options: -h Display this text -O Connect output to named port. -I Connect input to named port. -E Report excess latency, requires both -I and -O. The excess latency is the measured value minus the expected value for the given ports including any corrections set by Jack's -I and -O options. Ciao, -- FA From paul at linuxaudiosystems.com Fri Jun 20 13:23:58 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Fri, 20 Jun 2014 09:23:58 -0400 Subject: [LAU] Audio interface latency measurements In-Reply-To: <20140620131910.GC15318@linuxaudio.org> References: <20140620131910.GC15318@linuxaudio.org> Message-ID: On Fri, Jun 20, 2014 at 9:19 AM, Fons Adriaensen wrote: > On Fri, Jun 20, 2014 at 09:07:37AM -0400, Paul Davis wrote: > > > If you used jack_iodelay, it would do the math for you and show you how > > much extra latency is bing caused by the hardware (rather than the period > > size). > > jack_delay will do this as well, use the -E option. > and make sure your distribution has packaged and included the new version of jack_delay :) of course, since jack2 still isn't sharing the tools/ or headers in the JACK git repository, the presence of jack_iodelay there doesn't really help most Linux users at all since they don't have it either. -------------- next part -------------- An HTML attachment was scrubbed... URL: From fons at linuxaudio.org Fri Jun 20 14:10:45 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Fri, 20 Jun 2014 14:10:45 +0000 Subject: [LAU] Audio interface latency measurements In-Reply-To: References: <20140620131910.GC15318@linuxaudio.org> Message-ID: <20140620141044.GD15318@linuxaudio.org> On Fri, Jun 20, 2014 at 09:23:58AM -0400, Paul Davis wrote: > On Fri, Jun 20, 2014 at 9:19 AM, Fons Adriaensen > wrote: > > > On Fri, Jun 20, 2014 at 09:07:37AM -0400, Paul Davis wrote: > > > > > If you used jack_iodelay, it would do the math for you and show you how > > > much extra latency is bing caused by the hardware (rather than the period > > > size). > > > > jack_delay will do this as well, use the -E option. > > > > and make sure your distribution has packaged and included the new version > of jack_delay :) Released 18 April 2011, that is more than three years ago :-) Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From paul at linuxaudiosystems.com Fri Jun 20 16:05:51 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Fri, 20 Jun 2014 12:05:51 -0400 Subject: [LAU] Audio interface latency measurements In-Reply-To: <20140620141044.GD15318@linuxaudio.org> References: <20140620131910.GC15318@linuxaudio.org> <20140620141044.GD15318@linuxaudio.org> Message-ID: On Fri, Jun 20, 2014 at 10:10 AM, Fons Adriaensen wrote: > On Fri, Jun 20, 2014 at 09:23:58AM -0400, Paul Davis wrote: > > On Fri, Jun 20, 2014 at 9:19 AM, Fons Adriaensen > > wrote: > > > > > On Fri, Jun 20, 2014 at 09:07:37AM -0400, Paul Davis wrote: > > > > > > > If you used jack_iodelay, it would do the math for you and show you > how > > > > much extra latency is bing caused by the hardware (rather than the > period > > > > size). > > > > > > jack_delay will do this as well, use the -E option. > > > > > > > and make sure your distribution has packaged and included the new version > > of jack_delay :) > > Released 18 April 2011, that is more than three years ago :-) > It might even be in Debian Stable, then! :) -------------- next part -------------- An HTML attachment was scrubbed... URL: From ralf.mardorf at rocketmail.com Sat Jun 21 08:33:23 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sat, 21 Jun 2014 10:33:23 +0200 Subject: [LAU] Live bass guitar -> analog synth on a 2007 laptop:, viable? In-Reply-To: <53A42BFA.4030004@univpm.it> References: <53A42BFA.4030004@univpm.it> Message-ID: <1403339603.854.8.camel@archlinux> On Fri, 2014-06-20 at 14:41 +0200, Leonardo Gabrielli wrote: > Also consider testing your sound cards prior to the software: if you > can't get them to work, or you can't exploit very low period sizes > (64-128) then all the software will come to no use. Frames/Period 256, Periods/Buffer 2, IOW a latency of 10.7 ms IMO can be used at least for studio work. Using external synth could become an issue regarding to MIDI jitter, but this is not an issue for virtual synth. The OP should be aware that playing the instrument when using audio to MIDI conversion, can't be done, as it's done without the conversion. From spamatica at gmail.com Sat Jun 21 09:51:54 2014 From: spamatica at gmail.com (Robert Jonsson) Date: Sat, 21 Jun 2014 11:51:54 +0200 Subject: [LAU] Audio interface latency measurements In-Reply-To: References: <20140620131910.GC15318@linuxaudio.org> <20140620141044.GD15318@linuxaudio.org> Message-ID: 2014-06-20 18:05 GMT+02:00 Paul Davis : > > > > On Fri, Jun 20, 2014 at 10:10 AM, Fons Adriaensen > wrote: >> >> On Fri, Jun 20, 2014 at 09:23:58AM -0400, Paul Davis wrote: >> > On Fri, Jun 20, 2014 at 9:19 AM, Fons Adriaensen >> > wrote: >> > >> > > On Fri, Jun 20, 2014 at 09:07:37AM -0400, Paul Davis wrote: >> > > >> > > > If you used jack_iodelay, it would do the math for you and show you >> > > > how >> > > > much extra latency is bing caused by the hardware (rather than the >> > > > period >> > > > size). >> > > >> > > jack_delay will do this as well, use the -E option. >> > > >> > >> > and make sure your distribution has packaged and included the new >> > version >> > of jack_delay :) >> >> Released 18 April 2011, that is more than three years ago :-) > > > It might even be in Debian Stable, then! :) > Hehe Anyway, thanks guys for enlightening me about improving the test results! Follow up question: from a theoretical perspective, is it likely a usb 2.0 interface would have similar transport latency as firewire? Usb 1 I suppose would be worse due to lower clockspeed. Regards, Robert From nicola.di.marzo at vodafone.it Sat Jun 21 10:22:39 2014 From: nicola.di.marzo at vodafone.it (Nicola) Date: Sat, 21 Jun 2014 11:22:39 +0100 Subject: [LAU] New EP - Everywhere Nowhere - indie rock In-Reply-To: References: <5394395D.3050606@vodafone.it> Message-ID: <53A55CEF.1040005@vodafone.it> On 20/06/14 12:12, Carlos sanchiavedraz wrote: > 2014-06-08 12:22 GMT+02:00 Nicola : >> Hi all, >> >> The EP of my band is out now! >> It's been produced using Free and Open Source Software, Debian GNU/Linux + >> Tango Studio Repositories and Ardour 3. >> The license is CC BY-NC-SA >> Feel free to listen and download it freely from soundcloud >> https://soundcloud.com/bandage-indie-rock/sets/bandage-everywhere-nowhere >> or Jamendo: >> http://www.jamendo.com/en/list/a135179/everywhere-nowhere >> >> There's also a FLAC version available from Bandcamp >> http://bandage-indierock.bandcamp.com/album/everywhere-nowhere at 1 euro. >> >> Opinions or suggestions are welcome! >> >> Thanks Linuxaudio community! >> >> Regards, >> Nicola >> _______________________________________________ >> Linux-audio-user mailing list >> Linux-audio-user at lists.linuxaudio.org >> http://lists.linuxaudio.org/listinfo/linux-audio-user > Sounds wide and clean, and you can distinguish instruments. Good job. > Any docu about the process? > > Thanks for sharing. > Hi Carlos, Thanks for your comment. My intention is to publish a proper guide about "The making of" of the EP reporting all the steps done (recording, miking, mixing and etc) to achieve this release. The guide is still work in progress, Keep an eye here for a future publication: http://noeisnotunique.wordpress.com/ http://bandagerock.wordpress.com/diy-2/ I also would like to release the entire Ardour project for a song as "source". Anyway here's a brief description of the stuff I used: Digital Audio Workstation: Debian GNU/Linux 7.2 + Tango Studio Repositories Ardour 3 Mixing Plugins: EQ- LinuxDSP MKII-Graph(Commercial), EQ10Q. Compressor - Invada and Calf Compressor Gate - Steve Harris Gate Reverb - Roomy (Arty-FX - Open AV production), Calf Reverb, Steve Harris Plate Reverb. Delay - MDA Delay. Invada Tube Distortion and Steve Harris Valve Saturation. Mastering Plugins: Invada Meter and Calf Analyzer. MkII GraphEQ, Invada Compressor,Calf Stereo-tools and Limiter. There's been also a discussion for the mastering I did on Linuxmusicians: http://linuxmusicians.com/viewtopic.php?f=9&t=12505 Ciao Nicola From fons at linuxaudio.org Sat Jun 21 10:22:36 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sat, 21 Jun 2014 10:22:36 +0000 Subject: [LAU] Audio interface latency measurements In-Reply-To: References: <20140620131910.GC15318@linuxaudio.org> <20140620141044.GD15318@linuxaudio.org> Message-ID: <20140621102235.GA19081@linuxaudio.org> On Sat, Jun 21, 2014 at 11:51:54AM +0200, Robert Jonsson wrote: > Follow up question: from a theoretical perspective, is it likely a usb > 2.0 interface would have similar transport latency as firewire? Usb 1 > I suppose would be worse due to lower clockspeed. Clock frequency determines the amount of data per time that can be tranported, not latency. If a slower clock can still handle the data rate then there is no reason why it should make the data arrive 'later'. Every technology will have its particular features and limits. Nothing will beat a PCI(e) card if that card is designed well. USB probably won't give you the same advantage at 96 and 192 kHz Firewire offers. But generalisations can be misleading, and there's really no alternative to measuring each device. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From csanchezgs at gmail.com Sat Jun 21 12:15:23 2014 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Sat, 21 Jun 2014 14:15:23 +0200 Subject: [LAU] Yet another "io GNU/Linux" iso released In-Reply-To: <2386408.q0oAuvFt4X@io> References: <1782105.ohqO6NtJuQ@io> <2386408.q0oAuvFt4X@io> Message-ID: 2014-06-07 2:57 GMT+02:00 Manu Kebab : > Le vendredi 6 juin 2014, 05:53:09 Manu Kebab a ?crit : >> Hi, >> >> A new 64bit iso is up ;) >> >> io GNU/Linux is a Live DVD/USB based on Debian Sid and focused on >> multimedia. >> >> Kernel 3.14.4, Jack2 as default sound server, e18 as desktop environment and >> a big collection of installed software... Full persistence for USB install >> (with encryption) and more cool stuff... A great nomade studio :) >> >> For more infos: manual, packages list, screenshots, video etc... Check: >> >> -> http://manu.kebab.free.fr/iognulinux.html >> -> https://sourceforge.net/projects/io-gnu-linux/ >> >> >> Feedbacks welcome, enjoy :) >> >> MK > Update with kernel 3.14.5 > > Greetingz :) > > MK > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > Hello Manu. Congratulations for that work. We at Musix are always trying to improve our processes because we have very modest resources, talking about people and time. I would like to know if you have any documentation on how you remaster or build (i.e. live-helper or "the old" way) and the main aspects you tweak from SID that become your main differentiation, it would be really appreciated. Thanks for your work. -- C. sanchiavedraZ: * NEW / NUEVO: www.sanchiavedraZ.com * Musix GNU+Linux: www.musix.es From len at ovenwerks.net Sat Jun 21 17:18:49 2014 From: len at ovenwerks.net (Len Ovens) Date: Sat, 21 Jun 2014 10:18:49 -0700 (PDT) Subject: [LAU] Audio interface latency measurements In-Reply-To: References: <20140620131910.GC15318@linuxaudio.org> <20140620141044.GD15318@linuxaudio.org> Message-ID: On Sat, 21 Jun 2014, Robert Jonsson wrote: > Follow up question: from a theoretical perspective, is it likely a usb > 2.0 interface would have similar transport latency as firewire? Usb 1 > I suppose would be worse due to lower clockspeed. As was already stated, clock speed of the interface is not really relevant. It seems in fact that no one is really interested in USB3 because it does not have any improvement for audio, USB2 is enough. The limitation with USB1 is bit depth, bit rate and channel count. In general, throughput and latency are two different things. Larger packets mean better throughput, but smaller packets mean lower latency. I am not sure, but it seems to me the USB1.1 audio standard effectively means that the lowest latency for USB1 is jackd set to 64/2. This is the smallest buffer size supported. I do not know, but it seems that fire wire audio is about the same from what I have read (I don't have one of my own to confirm). The main trouble with USB is on the MB. Finding a USB port that is not shared with something else via an internal hub. I think adding a USB card would make things better, but trying different ports on a laptop gives good results too. With any audio interface, having it's own irq is important, I have moved PCI cards to different slots with a big difference. It shouldn't be, but it seems tunning a computer for audio is a must still for low latency. Audio is very definately _not_ plug and play for (semi)pro audio work. There is no silver bullet kernel or distro that just makes everything work. On my laptop, there is one USB port that gives good audio... so long as the port next to it is empty... and the wireless kernel module is unloaded and .... you get the picture :) -- Len Ovens www.ovenwerks.net From spamatica at gmail.com Sat Jun 21 18:19:49 2014 From: spamatica at gmail.com (Robert Jonsson) Date: Sat, 21 Jun 2014 20:19:49 +0200 Subject: [LAU] Audio interface latency measurements In-Reply-To: References: <20140620131910.GC15318@linuxaudio.org> <20140620141044.GD15318@linuxaudio.org> Message-ID: 2014-06-21 19:18 GMT+02:00 Len Ovens : > As was already stated, clock speed of the interface is not really relevant. > It seems in fact that no one is really interested in USB3 because it does > not have any improvement for audio, USB2 is enough. The limitation with USB1 > is bit depth, bit rate and channel count. You are probably right about USB3 but with USB1 and USB2 speeds, how can the clock speed not be relevant? As an example, wouldn't transferring 1024 frames of float on USB1 (12Mbit) take 2,7ms and only 68us on USB2 (480Mbit)? In raw throughput that is, there are probably other factors limiting the speed? Regards, Robert From jeremy at autostatic.com Sat Jun 21 18:43:23 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Sat, 21 Jun 2014 20:43:23 +0200 Subject: [LAU] Audio interface latency measurements In-Reply-To: References: <20140620131910.GC15318@linuxaudio.org> <20140620141044.GD15318@linuxaudio.org> Message-ID: <53A5D24B.2050306@autostatic.com> On 06/21/2014 07:18 PM, Len Ovens wrote: > On Sat, 21 Jun 2014, Robert Jonsson wrote: > >> Follow up question: from a theoretical perspective, is it likely a usb >> 2.0 interface would have similar transport latency as firewire? Usb 1 >> I suppose would be worse due to lower clockspeed. > > As was already stated, clock speed of the interface is not really > relevant. It seems in fact that no one is really interested in USB3 > because it does not have any improvement for audio, USB2 is enough. The > limitation with USB1 is bit depth, bit rate and channel count. > > In general, throughput and latency are two different things. Larger > packets mean better throughput, but smaller packets mean lower latency. > > I am not sure, but it seems to me the USB1.1 audio standard effectively > means that the lowest latency for USB1 is jackd set to 64/2. With my Edirol UA-25 lowest possible setting is 48/2 @ 48kHz. This is the > smallest buffer size supported. I do not know, but it seems that fire > wire audio is about the same from what I have read (I don't have one of > my own to confirm). With the FireWire interfaces I've owned I could go as low as 16/3 @ 48kHz. But settings like this are unusable, DSP load qickly rises as soon as you start doing something serious. Nice to brag about but that's about it ;) > > The main trouble with USB is on the MB. Finding a USB port that is not > shared with something else via an internal hub. I think adding a USB > card would make things better, but trying different ports on a laptop > gives good results too. With any audio interface, having it's own irq is > important, I have moved PCI cards to different slots with a big > difference. It shouldn't be, but it seems tunning a computer for audio > is a must still for low latency. Audio is very definately _not_ plug and > play for (semi)pro audio work. There is no silver bullet kernel or > distro that just makes everything work. On my laptop, there is one USB > port that gives good audio... so long as the port next to it is empty... > and the wireless kernel module is unloaded and .... you get the picture :) > It's indeed a matter of finding a free USB port. If you don't have any you have to resort to unloading kernel modules or even unbinding drivers. Jeremy > -- > Len Ovens > www.ovenwerks.net > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 836 bytes Desc: OpenPGP digital signature URL: From len at ovenwerks.net Sat Jun 21 19:23:49 2014 From: len at ovenwerks.net (Len Ovens) Date: Sat, 21 Jun 2014 12:23:49 -0700 (PDT) Subject: [LAU] Audio interface latency measurements In-Reply-To: <53A5D24B.2050306@autostatic.com> References: <20140620131910.GC15318@linuxaudio.org> <20140620141044.GD15318@linuxaudio.org> <53A5D24B.2050306@autostatic.com> Message-ID: On Sat, 21 Jun 2014, Jeremy Jongepier wrote: > On 06/21/2014 07:18 PM, Len Ovens wrote: >> I am not sure, but it seems to me the USB1.1 audio standard effectively >> means that the lowest latency for USB1 is jackd set to 64/2. > > With my Edirol UA-25 lowest possible setting is 48/2 @ 48kHz. I hadn't tried that... or to be honest 32/3. I have been lazy not thinking outside the 16/32/64/128 box :) > This is the >> smallest buffer size supported. I do not know, but it seems that fire >> wire audio is about the same from what I have read (I don't have one of >> my own to confirm). > > With the FireWire interfaces I've owned I could go as low as 16/3 @ > 48kHz. But settings like this are unusable, DSP load qickly rises as > soon as you start doing something serious. Nice to brag about but that's > about it ;) Is that the DSP load in the audio interface itself or the computer? I have found that 16/2 at 48k for use with guitarix (in other words as an effects rack) is very solid. (ice1712 pci interface) I am using a lowlatency kernel, but not RT. No hyperthreading (got the i5 for that reason) no CPU frequency changes. Not a super great tunning, but all the low hanging fruit for sure. Any modern computer should be able to deal with the load for at least some uses. Though I guess my atom should be considered "modern"... it does really well down to 64/2, I will have to try pushing a bit further, but I don't think it will go much lower and still have cpu cycles to run the SW. -- Len Ovens www.ovenwerks.net From jeremy at autostatic.com Sat Jun 21 19:48:38 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Sat, 21 Jun 2014 21:48:38 +0200 Subject: [LAU] Audio interface latency measurements In-Reply-To: References: <20140620131910.GC15318@linuxaudio.org> <20140620141044.GD15318@linuxaudio.org> <53A5D24B.2050306@autostatic.com> Message-ID: <53A5E196.20200@autostatic.com> On 06/21/2014 09:23 PM, Len Ovens wrote: >> >> With the FireWire interfaces I've owned I could go as low as 16/3 @ >> 48kHz. But settings like this are unusable, DSP load qickly rises as >> soon as you start doing something serious. Nice to brag about but that's >> about it ;) > > Is that the DSP load in the audio interface itself or the computer? The computer. I > have found that 16/2 at 48k for use with guitarix (in other words as an > effects rack) is very solid. (ice1712 pci interface) I was referring to USB and FireWire devices. PCI(e) apparently is still the interface to beat, even my USB2 interface can't go that low without becoming unstable. Jeremy -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 836 bytes Desc: OpenPGP digital signature URL: From len at ovenwerks.net Sat Jun 21 19:50:37 2014 From: len at ovenwerks.net (Len Ovens) Date: Sat, 21 Jun 2014 12:50:37 -0700 (PDT) Subject: [LAU] Audio interface latency measurements In-Reply-To: References: <20140620131910.GC15318@linuxaudio.org> <20140620141044.GD15318@linuxaudio.org> Message-ID: On Sat, 21 Jun 2014, Robert Jonsson wrote: > 2014-06-21 19:18 GMT+02:00 Len Ovens : >> As was already stated, clock speed of the interface is not really relevant. >> It seems in fact that no one is really interested in USB3 because it does >> not have any improvement for audio, USB2 is enough. The limitation with USB1 >> is bit depth, bit rate and channel count. > > You are probably right about USB3 but with USB1 and USB2 speeds, how > can the clock speed not be relevant? As an example, wouldn't > transferring 1024 frames of float on USB1 (12Mbit) take 2,7ms and only > 68us on USB2 (480Mbit)? > In raw throughput that is, there are probably other factors limiting the speed? Measure round trip. That is the only way to find out. As an aside, if you have to measure it, does it matter? In general, the manufacturers seem to have chosen to use the extra through put on USB2 to add extra bit depth (24bit instead of 16), extra bitrate (96k or more instead of 48k) and more channels. The latency would end up being simmilar. Being realistic... I can set my card to ~.6ms (one way) latency at 48k... maybe less at 96k but I haven't tried it. However, The dsp inside the audio interface itself already adds ~1ms of latency. That is 1.6ms one way. I can double the latency in jack to 1.2ms and the total is now 2.2ms not that much higher(this is not measured values BTW so the jackd added latency may be even less of the equation). There is a point where the latency of the interface itself is the main part of things... and how many inches away or towards the speaker do I have to move my head to have the same effect? When is a latency decrease really a gain? It seems to me from my own experience that a USB1 IF at 64/2 is good enough for live work as an effects rack or softsynth. For tighter needs the cost for the interface and the computer to run it goes up. Even audio imaging with multi speakers using delays for spacial placement is not overly latency dependant so long as all the channels are in good sync. -- Len Ovens www.ovenwerks.net From fons at linuxaudio.org Sat Jun 21 20:47:03 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sat, 21 Jun 2014 20:47:03 +0000 Subject: [LAU] Audio interface latency measurements In-Reply-To: References: <20140620131910.GC15318@linuxaudio.org> <20140620141044.GD15318@linuxaudio.org> Message-ID: <20140621204703.GA7353@linuxaudio.org> On Sat, Jun 21, 2014 at 08:19:49PM +0200, Robert Jonsson wrote: > You are probably right about USB3 but with USB1 and USB2 speeds, how > can the clock speed not be relevant? As an example, wouldn't > transferring 1024 frames of float on USB1 (12Mbit) take 2,7ms and only > 68us on USB2 (480Mbit)? For USB devices period size is just a property of the driver. Not of the interface, which will be using smaller packets. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From peter at peterlutek.com Sat Jun 21 20:57:34 2014 From: peter at peterlutek.com (Peter Lutek) Date: Sat, 21 Jun 2014 16:57:34 -0400 Subject: [LAU] Audio interface latency measurements In-Reply-To: <20140621204703.GA7353@linuxaudio.org> References: <20140620131910.GC15318@linuxaudio.org> <20140620141044.GD15318@linuxaudio.org> <20140621204703.GA7353@linuxaudio.org> Message-ID: <43e13a6d2d17d941c1392921c1e3cfe1@peterlutek.com> greetings, all! interesting that this USB latency discussion comes up now, as i'm re-examining my live performance setup. without having quantified rigorously, i CAN say that my latency performance before xruns is much better with a new PCIe USB adapter than with the internal USB ports on my Lenovo T520. ART USB Dual Tube Pre (USB 2.0) using a bunch of pd stuff, plus sooperlooper. cheers! .pltk. -- Peter Lutek improvising musician in Toronto, Canada http://peterlutek.com From len at ovenwerks.net Sat Jun 21 21:16:10 2014 From: len at ovenwerks.net (Len Ovens) Date: Sat, 21 Jun 2014 14:16:10 -0700 (PDT) Subject: [LAU] Audio interface latency measurements In-Reply-To: <43e13a6d2d17d941c1392921c1e3cfe1@peterlutek.com> References: <20140620131910.GC15318@linuxaudio.org> <20140620141044.GD15318@linuxaudio.org> <20140621204703.GA7353@linuxaudio.org> <43e13a6d2d17d941c1392921c1e3cfe1@peterlutek.com> Message-ID: On Sat, 21 Jun 2014, Peter Lutek wrote: > ART USB Dual Tube Pre (USB 2.0) Holding the box for one of those in my hand... and yes it does say USB 2.0 on the box. But, the spec looks a lot like USB 1.1 - two channels, 44.1k or 48k. SO I don't know what the gain is for that. It is interesting though that there seem to be separate a/d converters for USB and s/pdif so that one can be set to 44.1 while the other is at 48k. I am not sure how this would be useful though. The stated A/D latency is .4ms. The best thing from my POV was that I could take my laptop into the music store and plug it in and everything just worked. This was not the case with 2 or 3 other interfaces I tried the same day. (Native Instruments, and M-audio... can't remember the other now) The only thing missing is midi. -- Len Ovens www.ovenwerks.net From peter at peterlutek.com Sat Jun 21 21:27:59 2014 From: peter at peterlutek.com (Peter Lutek) Date: Sat, 21 Jun 2014 17:27:59 -0400 Subject: [LAU] Audio interface latency measurements In-Reply-To: References: <20140620131910.GC15318@linuxaudio.org> <20140620141044.GD15318@linuxaudio.org> <20140621204703.GA7353@linuxaudio.org> <43e13a6d2d17d941c1392921c1e3cfe1@peterlutek.com> Message-ID: <6051ac7f02f29ec2f8b442510bd9f34c@peterlutek.com> On 2014-06-21 17:16, Len Ovens wrote: > >> ART USB Dual Tube Pre (USB 2.0) > > ...The only > thing missing is midi. i add on an M-Audio midisport 2x2 for that... nice and small, and completely plug'n'play too! cheers! .pltk. From louigi.verona at gmail.com Sun Jun 22 10:01:19 2014 From: louigi.verona at gmail.com (Louigi Verona) Date: Sun, 22 Jun 2014 14:01:19 +0400 Subject: [LAU] Streaming radio and calls from listeners Message-ID: Hey everyone! For more than a year I am producing a skeptic-oriented podcast. So far it's been an offline venture. We have a nice audio mixer from Yamaha that we are recording as one track into Qtractor. However, I am thinking towards live streaming and accepting calls from listeners. Is this realistic with Linux? If yes - can anyone suggest how? -- Louigi Verona http://www.louigiverona.ru/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From fons at linuxaudio.org Sun Jun 22 10:21:06 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sun, 22 Jun 2014 10:21:06 +0000 Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: References: Message-ID: <20140622102106.GB27070@linuxaudio.org> On Sun, Jun 22, 2014 at 02:01:19PM +0400, Louigi Verona wrote: > For more than a year I am producing a skeptic-oriented podcast. So far it's > been an offline venture. We have a nice audio mixer from Yamaha that we are > recording as one track into Qtractor. > > However, I am thinking towards live streaming and accepting calls from > listeners. Is this realistic with Linux? If yes - can anyone suggest how? If you use the POTS you need an interface that provides transformer isolation, the correct impedance on the telephone line, and that removes as much as possible of the outgoing signal from the incoming one. Could be expensive. It is possible to hack together such a thing using relatively simple electronics (done it), but using such a device woul break the terms of use of your telco - they usually want a certified unit (for good reasons). If you use a cellular phone you need something like this: to connect the phone to a sound card. In both cases on the mixer you need to create an 'N-1' bus, which has all the inputs from your main mix except the signal from the phone. This is what you send back to the phone. You also need some way to talk to the caller before he/she goes on air, zita-mu1 could come in handy for this. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From ralf.mardorf at rocketmail.com Sun Jun 22 10:21:55 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 22 Jun 2014 12:21:55 +0200 Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: References: Message-ID: <1403432515.2245.13.camel@archlinux> Technically there likely is a way. I wanted to make Internet radio, but I never did it regarding to legal issues. JFTR I made citizen radio in Germany and it already caused serious legal issues, not regarding to collecting societies, but regarding to freedom of speech. For example, if there should be no doubt about the fact that a nation should kill people, you're not allowed to point this out in Germany, it's a crime, since it does touch the sovereignty of a state and we are not allowed to do it. Somebody now might claim, that such crimes are made public each day by official radio stations in Germany. Yes, but it's a long way and does cause a lot of trouble, if it should be against the policy, e.g. Germany makes lots of money by selling weapons and Germany needs the jobs. IOW it does matter what facts you point out on air. Good luck! Ralf From louigi.verona at gmail.com Sun Jun 22 10:30:34 2014 From: louigi.verona at gmail.com (Louigi Verona) Date: Sun, 22 Jun 2014 14:30:34 +0400 Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: <1403432515.2245.13.camel@archlinux> References: <1403432515.2245.13.camel@archlinux> Message-ID: I am not able to use the telephone line. I can rely only on Skype or similar online services for calls, unfortunately. I should have mentioned that, but I did not know someone is using an actual phone for it nowadays. On Sun, Jun 22, 2014 at 2:21 PM, Ralf Mardorf wrote: > Technically there likely is a way. I wanted to make Internet radio, but > I never did it regarding to legal issues. JFTR I made citizen radio in > Germany and it already caused serious legal issues, not regarding to > collecting societies, but regarding to freedom of speech. > > For example, if there should be no doubt about the fact that a nation > should kill people, you're not allowed to point this out in Germany, > it's a crime, since it does touch the sovereignty of a state and we are > not allowed to do it. Somebody now might claim, that such crimes are > made public each day by official radio stations in Germany. Yes, but > it's a long way and does cause a lot of trouble, if it should be against > the policy, e.g. Germany makes lots of money by selling weapons and > Germany needs the jobs. IOW it does matter what facts you point out on > air. > > Good luck! > Ralf > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > -- Louigi Verona http://www.louigiverona.ru/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From ralf.mardorf at rocketmail.com Sun Jun 22 10:36:37 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 22 Jun 2014 12:36:37 +0200 Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: <1403432515.2245.13.camel@archlinux> References: <1403432515.2245.13.camel@archlinux> Message-ID: <1403433397.2245.15.camel@archlinux> On Sun, 2014-06-22 at 12:21 +0200, Ralf Mardorf wrote: > Technically there likely is a way. I wanted to make Internet radio, but > I never did it regarding to legal issues. JFTR I made citizen radio in > Germany and it already caused serious legal issues, not regarding to > collecting societies, but regarding to freedom of speech. > > For example, if there should be no doubt about the fact that a nation > should kill people, you're not allowed to point this out in Germany, > it's a crime, since it does touch the sovereignty of a state and we are > not allowed to do it. Somebody now might claim, that such crimes are > made public each day by official radio stations in Germany. Yes, but > it's a long way and does cause a lot of trouble, if it should be against > the policy, e.g. Germany makes lots of money by selling weapons and > Germany needs the jobs. IOW it does matter what facts you point out on > air. > > Good luck! > Ralf PS: To make it absolutely clear! There never was an issue regarding the truth! The issue was, that the truth did offend the sovereignty of a state! At least in Germany you're not allowed to point out undisputed truth, if it does offend the sovereignty of a state, assumed it's a state that is important for Germany. You're free even to spread unproven rumours about ever nation that is unimportant for Germany. From ralf.mardorf at rocketmail.com Sun Jun 22 10:39:36 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 22 Jun 2014 12:39:36 +0200 Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: References: <1403432515.2245.13.camel@archlinux> Message-ID: <1403433576.2245.17.camel@archlinux> On Sun, 2014-06-22 at 14:30 +0400, Louigi Verona wrote: > I did not know someone is using an actual phone for it nowadays. AFAIK, but my experiences are a few years old, phone more often is used than Internet alternatives. Anyway, you still risk legal issues, assumed it's live. From list at nilsgey.de Sun Jun 22 13:10:03 2014 From: list at nilsgey.de (Nils) Date: Sun, 22 Jun 2014 15:10:03 +0200 Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: <1403433397.2245.15.camel@archlinux> References: <1403432515.2245.13.camel@archlinux> <1403433397.2245.15.camel@archlinux> Message-ID: <53A6D5AB.5010600@nilsgey.de> On 22.06.2014 12:36, Ralf Mardorf wrote: > > For example ...bla bla bla bla Oh, shut up Ralf! From fons at linuxaudio.org Sun Jun 22 11:23:15 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sun, 22 Jun 2014 11:23:15 +0000 Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: References: Message-ID: <20140622112315.GD27070@linuxaudio.org> On Sun, Jun 22, 2014 at 02:01:19PM +0400, Louigi Verona wrote: > For more than a year I am producing a skeptic-oriented podcast. So far it's > been an offline venture. We have a nice audio mixer from Yamaha that we are > recording as one track into Qtractor. > > However, I am thinking towards live streaming and accepting calls from > listeners. Is this realistic with Linux? If yes - can anyone suggest how? If you use Skype then the simplest solution is to run it on a separate PC using the built-in soundcard (this usually works). You then have to connect that PC's headphone output to your main soundcard input (this may require a DI-box to get a clean signal) and the main soundard line out to the mic input of the PC (which will require a passive attenuator and maybe also transforer isolation). It is in theory possible to run Skype on the main PC and connect it to jack using snd_aloop, zita-a2j and zita-j2a. It's not a stable setup (Skype being very fussy about sound cards), so I wouldn't recommend doing that. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From ralf.mardorf at rocketmail.com Sun Jun 22 11:41:48 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 22 Jun 2014 13:41:48 +0200 Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: <53A6D5AB.5010600@nilsgey.de> References: <1403432515.2245.13.camel@archlinux> <1403433397.2245.15.camel@archlinux> <53A6D5AB.5010600@nilsgey.de> Message-ID: <1403437308.2245.22.camel@archlinux> On Sun, 2014-06-22 at 15:10 +0200, Nils wrote: > On 22.06.2014 12:36, Ralf Mardorf wrote: > > > > For example ...bla bla bla bla > Oh, shut up Ralf! I seriously don't see a technical issue, but only a legal issue regarding to live streaming. YMMV! I'm not talking about theory, but about practice. Perhaps your mail is "...bla bla bla bla" and not mine, we never know, so ... or do we (at least I) already know? ... .. where I'm mistaken for the mails I wrote? TIA for your clarification ;)!!! Ralf PS: Let us know about your radio experiences, especially if you should be from Germany. What "Landesrundfunkanstalt" does chsk your radio transmissions? I'm fro NRW. Regards, Ralf From ralf.mardorf at rocketmail.com Sun Jun 22 11:44:54 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 22 Jun 2014 13:44:54 +0200 Subject: [LAU] Sorry, but I type faster than my Linux userspace keyboard thingy is able to follow, so pleas read this correction: Streaming radio and calls from listeners In-Reply-To: <1403437308.2245.22.camel@archlinux> References: <1403432515.2245.13.camel@archlinux> <1403433397.2245.15.camel@archlinux> <53A6D5AB.5010600@nilsgey.de> <1403437308.2245.22.camel@archlinux> Message-ID: <1403437494.2245.24.camel@archlinux> On Sun, 2014-06-22 at 13:41 +0200, Ralf Mardorf wrote: > On Sun, 2014-06-22 at 15:10 +0200, Nils wrote: > > On 22.06.2014 12:36, Ralf Mardorf wrote: > > > > > > For example ...bla bla bla bla > > Oh, shut up Ralf! > > I seriously don't see a technical issue, but only a legal issue > regarding to live streaming. YMMV! > > I'm not talking about theory, but about practice. Perhaps your mail is > "...bla bla bla bla" and not mine, we never know, so ... or do we (at > least I) already know? ... > > .. where I'm mistaken for the mails I wrote? > > TIA for your clarification ;)!!! > Ralf > > PS: Let us know about your radio experiences, especially if you should > be from Germany. What "Landesrundfunkanstalt" does chsk your radio check > transmissions? I'm fro NRW. from > > Regards, > Ralf From sam at vis.nu Sun Jun 22 11:45:59 2014 From: sam at vis.nu (Sam Mulvey) Date: Sun, 22 Jun 2014 04:45:59 -0700 Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: <20140622112315.GD27070@linuxaudio.org> References: <20140622112315.GD27070@linuxaudio.org> Message-ID: <53A6C1F7.8020205@vis.nu> On 06/22/2014 04:23 AM, Fons Adriaensen wrote: > It is in theory possible to run Skype on the main PC and connect it to > jack using snd_aloop, zita-a2j and zita-j2a. It's not a stable setup > (Skype being very fussy about sound cards), so I wouldn't recommend > doing that. I've used Skype in a loopback setup, but over the long term it wasn't that stable. I've been doing live radio on a linux setup since March, and we *are* using a loopback for an instance of Linphone, and that's been working like a champ. -Sam From csanchezgs at gmail.com Sun Jun 22 12:03:10 2014 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Sun, 22 Jun 2014 14:03:10 +0200 Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: <53A6C1F7.8020205@vis.nu> References: <20140622112315.GD27070@linuxaudio.org> <53A6C1F7.8020205@vis.nu> Message-ID: 2014-06-22 13:45 GMT+02:00 Sam Mulvey : > On 06/22/2014 04:23 AM, Fons Adriaensen wrote: >> It is in theory possible to run Skype on the main PC and connect it to >> jack using snd_aloop, zita-a2j and zita-j2a. It's not a stable setup >> (Skype being very fussy about sound cards), so I wouldn't recommend >> doing that. > > > I've used Skype in a loopback setup, but over the long term it wasn't > that stable. I've been doing live radio on a linux setup since March, > and we *are* using a loopback for an instance of Linphone, and that's > been working like a champ. > > -Sam > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user Given that Skype is available for many mobile platforms, and there are SIP clients as well, there's another possibility that should be simpler than an extra PC/loopback solution. You could install Skype/SIP client on your mobile or tablet and connect device's headphone output to the mixer. On the other hand, there are programs like Rivendel and distros for radio stations specific purposes that could be interesting to take into account. -- C. sanchiavedraZ: * NEW / NUEVO: www.sanchiavedraZ.com * Musix GNU+Linux: www.musix.es From ralf.mardorf at rocketmail.com Sun Jun 22 12:20:50 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 22 Jun 2014 14:20:50 +0200 Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: References: <20140622112315.GD27070@linuxaudio.org> <53A6C1F7.8020205@vis.nu> Message-ID: <1403439650.3789.1.camel@archlinux> On Sun, 2014-06-22 at 14:03 +0200, Carlos sanchiavedraz wrote: > 2014-06-22 13:45 GMT+02:00 Sam Mulvey : > > On 06/22/2014 04:23 AM, Fons Adriaensen wrote: > >> It is in theory possible to run Skype on the main PC and connect it to > >> jack using snd_aloop, zita-a2j and zita-j2a. It's not a stable setup > >> (Skype being very fussy about sound cards), so I wouldn't recommend > >> doing that. > > > > > > I've used Skype in a loopback setup, but over the long term it wasn't > > that stable. I've been doing live radio on a linux setup since March, > > and we *are* using a loopback for an instance of Linphone, and that's > > been working like a champ. > > > > -Sam > > _______________________________________________ > > Linux-audio-user mailing list > > Linux-audio-user at lists.linuxaudio.org > > http://lists.linuxaudio.org/listinfo/linux-audio-user > > Given that Skype is available for many mobile platforms, and there are > SIP clients as well, there's another possibility that should be > simpler than an extra PC/loopback solution. You could install > Skype/SIP client on your mobile or tablet and connect device's > headphone output to the mixer. > > On the other hand, there are programs like Rivendel and distros for > radio stations specific purposes that could be interesting to take > into account. My apologize, you're not talking about real radio and everything that is important, just abut some hobby channels and technical issues that aren't real issues for people who care about real radio issue. Wow, ho low can you go :D. From len at ovenwerks.net Sun Jun 22 16:13:27 2014 From: len at ovenwerks.net (Len Ovens) Date: Sun, 22 Jun 2014 09:13:27 -0700 (PDT) Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: References: Message-ID: On Sun, 22 Jun 2014, Louigi Verona wrote: > For more than a year I am producing a skeptic-oriented podcast. So far it's > been an offline venture. We have a nice audio mixer from Yamaha that we are > recording as one track into Qtractor. So, it sounds like you have been doing all your mixing outside the computer then. (just checking) > > However, I am thinking towards live streaming and accepting calls from > listeners. Is this realistic with Linux? If yes - can anyone suggest how? Try reading about http://idjc.sourceforge.net/ in particular http://idjc.sourceforge.net/tutorials_voip.html Though I have also seen setups with pulse->jack... I have not tried pulse->jack with my new computer but it required too much CPU on the old single core P4. (by the way the idjc here is _not_ Idaho Department of Juvenile Corrections) Remote content providing that I know has worked: -teamspeak (oss sound on a second computer mixed in an external mixer) -mumble -others as mentioned Skype likes to play with levels of the sound card... owned by MS so it likes to be in control... However, lots of people have a skype account, but only the free part so they can not call a number or a named voip account. They also can not accept a call from non-skype voip unless you have a paying account. With regard to content filtering of live calls, not just from a legal perspective, but also from what you want your listeners to hear. The standard method used here when I was in the business years ago was to delay the whole studio audio about 7 seconds and provide a cut button that shut off the audio after the delay. It gave some interesting echo effects when the call in had their radio turned up too high :) The idjc DSP button could be used for this if you were using that SW... most effects have a bypass button too. One thing I would like to try for remote content is to use netjack... Your master server has to be inet visible though. Which brings a question to mind... can a jackd server with a net backend also be a netjack master? I guess the real question is if a netjack master can be tied to a particular network interface. The idea being that the studio machine with the audio card is master on the local net. The webserver is a slave on the local net, but a master to the internet. The other way would be to put a soundcard in the webserver for timing only and make it master of all. Then use zita-aj for the real audio card. The problem with the second method is that the local net would have compressed audio and all machines using netjack would have the added overhead of the codec. -- Len Ovens www.ovenwerks.net From sam at vis.nu Sun Jun 22 20:21:47 2014 From: sam at vis.nu (Sam Mulvey) Date: Sun, 22 Jun 2014 13:21:47 -0700 Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: References: <20140622112315.GD27070@linuxaudio.org> <53A6C1F7.8020205@vis.nu> Message-ID: <53A73ADB.9060803@vis.nu> On 06/22/2014 05:03 AM, Carlos sanchiavedraz wrote: > Given that Skype is available for many mobile platforms, and there are > SIP clients as well, there's another possibility that should be > simpler than an extra PC/loopback solution. You could install > Skype/SIP client on your mobile or tablet and connect device's > headphone output to the mixer. > > On the other hand, there are programs like Rivendel and distros for > radio stations specific purposes that could be interesting to take > into account. If I'm doing prerecorded interviews with Skype, I generally use a smart phone. It's a stop gap solution, especially since the Skype caller isn't getting audio off the board, just audio from the smartphone. If your mixer has multiple sends, then you could go that route, but the mixers that most podcasters are going to have won't. For live radio, UI concerns are important[1], and having to take calls on a smartphone *and* manage the rest of the show turns it into a bit of a mess. It also makes call screening harder to do, which is something I really have to do. That might be well beyond what the OP was asking for though. -Sam [1]: To the point where I'm designing a hardware interface that acts like the older boards I'm used to, but is all digital underneath. Right now, I'm using touchscreens, and having to look at what I'm doing can actually be a problem sometimes. With pots and switches, I can feel around. From louigi.verona at gmail.com Sun Jun 22 20:59:30 2014 From: louigi.verona at gmail.com (Louigi Verona) Date: Mon, 23 Jun 2014 00:59:30 +0400 Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: <53A73ADB.9060803@vis.nu> References: <20140622112315.GD27070@linuxaudio.org> <53A6C1F7.8020205@vis.nu> <53A73ADB.9060803@vis.nu> Message-ID: Hey everyone! Thanks for all your advice! I do have a second laptop I can use and I was thinking of this scheme: channel this second laptop into the mixer or, indeed, channel the phone into the mixer and do it this way. Thanks to the mixer, I will be able to mute the call if someone says nasty things. I installed Internet DJ Console, will look into it. I wonder if streaming it live will work. A complicated setup, but manageable. I was also thinking of streaming it to YouTube, live. On Mon, Jun 23, 2014 at 12:21 AM, Sam Mulvey wrote: > On 06/22/2014 05:03 AM, Carlos sanchiavedraz wrote: > > Given that Skype is available for many mobile platforms, and there are > > SIP clients as well, there's another possibility that should be > > simpler than an extra PC/loopback solution. You could install > > Skype/SIP client on your mobile or tablet and connect device's > > headphone output to the mixer. > > > > On the other hand, there are programs like Rivendel and distros for > > radio stations specific purposes that could be interesting to take > > into account. > > > If I'm doing prerecorded interviews with Skype, I generally use a smart > phone. It's a stop gap solution, especially since the Skype caller > isn't getting audio off the board, just audio from the smartphone. If > your mixer has multiple sends, then you could go that route, but the > mixers that most podcasters are going to have won't. > > For live radio, UI concerns are important[1], and having to take calls > on a smartphone *and* manage the rest of the show turns it into a bit of > a mess. It also makes call screening harder to do, which is something I > really have to do. That might be well beyond what the OP was asking > for though. > > -Sam > > > [1]: To the point where I'm designing a hardware interface that acts > like the older boards I'm used to, but is all digital underneath. > Right now, I'm using touchscreens, and having to look at what I'm doing > can actually be a problem sometimes. With pots and switches, I can > feel around. > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > -- Louigi Verona http://www.louigiverona.ru/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From len at ovenwerks.net Sun Jun 22 21:00:46 2014 From: len at ovenwerks.net (Len Ovens) Date: Sun, 22 Jun 2014 14:00:46 -0700 (PDT) Subject: [LAU] Streaming radio and calls from listeners Message-ID: Louigi Verona wrote: > I am not able to use the telephone line. I can rely only on Skype or > similar online services for calls, unfortunately. I should have > mentioned that, but I did not know someone is using an actual phone for > it nowadays. Ok, I did the full setup minus the last step of encoding and streaming to the server. I used idjc as my console. I installed skype and created an account. My audio setup: ice1712 audio card. At the start of session jackdbus is started at 48k 2048/2, a2jmidid runs and pulse gets restarted so it sees jack. I already have all audio interfaces turned off in pulse so if jackdbus is not available there is only a "dummy" output. I use this: http://www.ovenwerks.net/software/index.html Which I have not yet packaged, and set to phone mode which is jack at 128/2 and pulse->jack bridged and the cpu governor set to performance. My studio mic (I happen to have a Peavy PVM 520NT dynamic set up just now) is into capture_1 through a mackie pre strip. Playback_1 and 2 go to studio monitors. I start idjc and move the pulse lines to the matching idjc voip lines. If I was doing this all the time... ie, this computers job was radio studio, I would have pulse start without connections and use jack.plumbing or some session manager to make sure this are connected right... I haven't looked into it but I think idjc can do this part too. I haven't made actual phone calls, but the "echo/sound test service" provides both incoming and outgoing audio, so I used that as it streamed audio to and from the skype server. The output was recorded rather than streamed. The recording of both incoming and outgoing audio was clear and dropout free. I had an mp3 being played through idjc as well at the same time and that was mixed in fine too. So if pulse is already running on your system, it is probably the easiest thing to use. It just works with pulse with no fussing needed. Using pulse means that the ports are always graphed and can remain connected all the time and any other desktop audio source can easily be used... for example a soundcloud track could be directly played from a browser or another voip client could be used for normal phone calls all without doing any extra setup. Pulse has it's bad points for sure, but with a suitable machine (I have a newer i5 machine at 3.2 Ghz with 8Gram... nothing special) it is the cat's meow for this application. Things to remember: Streaming audio means DSP work. Generally all your content is encoded and the idea of decoding to live play of at least three mp3/ogg files while still encoding the output for streaming is a reality. Then adding skype adds an encoding and a decoding step as well. This is assuming that the "on air talent" (dj) is playing music and talking to their next "on the phone" guest at the same time even if the phone audio is not being streamed at that time. This is where my old computer failed, it just ran out of CPU to do all that. This decoding includes sample rate changes too. I think skype likes 48k and then a lot of mp3 and oggs are 44.1k... though of coarse an effort could be made to make sure the file library is at the working sample rate which may reduce the cpu load some. I didn't have to do that in my test though. On this machine with an mp3 playing and a skype call running my cpu was still less than 10% so there is still room for some audio optimization of the final stream before output. (is there an LV2 plugin for on air optimization? In any case all the bits are there) I suspect there is also room for an on air monitor running separately. -- Len Ovens www.ovenwerks.net From ralf.mardorf at rocketmail.com Sun Jun 22 21:18:11 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 22 Jun 2014 23:18:11 +0200 Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: References: Message-ID: <1403471891.3789.5.camel@archlinux> On Sun, 2014-06-22 at 09:13 -0700, Len Ovens wrote: > With regard to content filtering of live calls, not just from a legal > perspective, but also from what you want your listeners to hear. The > standard method used here when I was in the business years ago was to > delay the whole studio audio about 7 seconds and provide a cut button > that shut off the audio after the delay. Sure, issues are not only that dramatically as I said. It simply could be swearing you don't want to send. Perhaps not a legal issue, but at least something that cause advertisers not to pay you anymore. Ok, the OP might not have advertisers, but seemingly listeners. It's all the same, advertisers drop a radio station, when the radio station doesn't have enough listeners. I know, everything I experienced in reality is just bla bla to the people subscribed to this list. Do yourself a favour, if you don't like me, then listen to Len. From ralf.mardorf at rocketmail.com Sun Jun 22 21:32:10 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 22 Jun 2014 23:32:10 +0200 Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: References: <20140622112315.GD27070@linuxaudio.org> <53A6C1F7.8020205@vis.nu> <53A73ADB.9060803@vis.nu> Message-ID: <1403472730.3789.9.camel@archlinux> On Mon, 2014-06-23 at 00:59 +0400, Louigi Verona wrote: > Thanks to the mixer, I will be able to mute the call if someone says > nasty things. As Len already pointed out, you need a delay. Btw. if you're talking to the people + taking care about issues, 7 seconds likely are not enough delay. A radio station has got somebody who is talking to the people and another person listening with the finger above the mute bottom and another finger to start an advertising or music, to avoid a gap. From fons at linuxaudio.org Sun Jun 22 21:48:01 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Sun, 22 Jun 2014 21:48:01 +0000 Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: References: Message-ID: <20140622214801.GC13032@linuxaudio.org> On Sun, Jun 22, 2014 at 09:13:27AM -0700, Len Ovens wrote: > With regard to content filtering of live calls, not just from a > legal perspective, but also from what you want your listeners to > hear. The standard method used here when I was in the business years > ago was to delay the whole studio audio about 7 seconds and provide > a cut button that shut off the audio after the delay. This seven seconds 'profanity delay' seems to be a US thing, and probably an FCC requirement. I've never seen such things used on this side of the Atlantic. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From ralf.mardorf at rocketmail.com Sun Jun 22 21:56:30 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Sun, 22 Jun 2014 23:56:30 +0200 Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: <20140622214801.GC13032@linuxaudio.org> References: <20140622214801.GC13032@linuxaudio.org> Message-ID: <1403474190.3789.11.camel@archlinux> On Sun, 2014-06-22 at 21:48 +0000, Fons Adriaensen wrote: > On Sun, Jun 22, 2014 at 09:13:27AM -0700, Len Ovens wrote: > > > With regard to content filtering of live calls, not just from a > > legal perspective, but also from what you want your listeners to > > hear. The standard method used here when I was in the business years > > ago was to delay the whole studio audio about 7 seconds and provide > > a cut button that shut off the audio after the delay. > > This seven seconds 'profanity delay' seems to be a US thing, > and probably an FCC requirement. I've never seen such things > used on this side of the Atlantic. Indeed we don't use a delay here, but there always is the person with the finger above the mute button and with the other finger on the play button. From len at ovenwerks.net Sun Jun 22 22:13:34 2014 From: len at ovenwerks.net (Len Ovens) Date: Sun, 22 Jun 2014 15:13:34 -0700 (PDT) Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: <20140622214801.GC13032@linuxaudio.org> References: <20140622214801.GC13032@linuxaudio.org> Message-ID: On Sun, 22 Jun 2014, Fons Adriaensen wrote: > This seven seconds 'profanity delay' seems to be a US thing, > and probably an FCC requirement. I've never seen such things > used on this side of the Atlantic. And Canadian too, but I hear lots of words from announcers that would have been out the door the same day words not that long ago. More likely to have trouble with "hate" words these days.... flavour of the week stuff. There is a big difference between internet radio and on air. On air requires a licence after which lots of money gets spent putting up transmitter and studio, then lots of money is spent paying employees and talent. The loss of a licence for a radio station is devastating. Internet radio? I would have no equipment outlay, no licence to worry about either. Probably it would start with as little as one person maybe not even paid. The biggest thing with music is making sure royalties are dealt with either by picking free to play or making sure payment was made. At worst someone may come and remove my computer... they may sue too, but you have to have something to sue for :) Most likely just stopping would quell complaints though. Ya, it depends where you are for sure. It changes all the time too. Generally when interviewing someone on the phone it is not a problem. Call in stuff can be more volitile... perhaps north americans are just less polite :) -- Len Ovens www.ovenwerks.net From ralf.mardorf at rocketmail.com Sun Jun 22 22:29:23 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Mon, 23 Jun 2014 00:29:23 +0200 Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: References: <20140622214801.GC13032@linuxaudio.org> Message-ID: <1403476163.3789.13.camel@archlinux> On Sun, 2014-06-22 at 15:13 -0700, Len Ovens wrote: > perhaps north americans are just less polite :) :D No, in Europe we are allowed to speak out thinks that American people don't want to hear, anyway, there are limits in Europa too. Btw. in Germany there are no black helicopters, there's no suing, just the editor gets fired and unlikely, but possible you receive a letter from the local Landesrundfunkanstalt. I don't know how long audio recordings have to be archived, but in Germany each radio station needs to record everything that is send and then it has to be archived for a while. From ats at offog.org Sun Jun 22 23:51:31 2014 From: ats at offog.org (Adam Sampson) Date: Mon, 23 Jun 2014 00:51:31 +0100 Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: <20140622214801.GC13032@linuxaudio.org> (Fons Adriaensen's message of "Sun, 22 Jun 2014 21:48:01 +0000") References: <20140622214801.GC13032@linuxaudio.org> Message-ID: Fons Adriaensen writes: > This seven seconds 'profanity delay' seems to be a US thing, and > probably an FCC requirement. I've never seen such things used on this > side of the Atlantic. Some UK broadcasters certainly use this kind of delay when dealing with live callers, with a "dump" button that the presenter can use to drop the last few seconds. This sometimes gets reported on by Ofcom (the UK communications regulator) when it goes wrong, e.g. when LBC were feeding their FM service from the wrong side of the delay: http://stakeholders.ofcom.org.uk/enforcement/broadcast-bulletins/obb68/ The Communications Act 2003 has a requirement to not include "offensive and harmful" material (which Ofcom interpret in a fairly liberal way, taking context into account), and a delay's a pretty good way of doing so when the presenter can't otherwise control what interviewees might say. -- Adam Sampson From ralf.mardorf at rocketmail.com Mon Jun 23 00:37:45 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Mon, 23 Jun 2014 02:37:45 +0200 Subject: [LAU] Off-topic: Streaming radio and calls from listeners In-Reply-To: References: <20140622214801.GC13032@linuxaudio.org> Message-ID: <1403483865.3789.19.camel@archlinux> On Mon, 2014-06-23 at 00:51 +0100, Adam Sampson wrote: > The Communications Act 2003 has a requirement to not include "offensive > and harmful" material (which Ofcom interpret in a fairly liberal way, > taking context into account), and a delay's a pretty good way of doing > so when the presenter can't otherwise control what interviewees might > say. What happens if a radio station does offend this commandment? In Germany you are free to speak out obscenities, OTOH for even correct political statements that are against the "vision" of Germany and allies, the editor gets fired, but nobody will sue the radio station, but there's one exception when black helicopters are nearly used, those guys definitively will be sued, while AFAIK American NAZIs are free to transmit agitation by radio, this is a no-go in Germany. From ralf.mardorf at rocketmail.com Mon Jun 23 00:51:48 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Mon, 23 Jun 2014 02:51:48 +0200 Subject: [LAU] PS: Off-topic: Streaming radio and calls from listeners In-Reply-To: <1403483865.3789.19.camel@archlinux> References: <20140622214801.GC13032@linuxaudio.org> <1403483865.3789.19.camel@archlinux> Message-ID: <1403484708.3789.22.camel@archlinux> On Mon, 2014-06-23 at 02:37 +0200, Ralf Mardorf wrote: > AFAIK American NAZIs are free to transmit agitation by radio I didn't spread FUD, I watched a movie and have got other sources and have got no reason to not believe it: IMO a rather good documentary from an African-German woman. http://www.die-arier.com/en/index.php From idragosani at gmail.com Mon Jun 23 01:00:43 2014 From: idragosani at gmail.com (Brett McCoy) Date: Sun, 22 Jun 2014 21:00:43 -0400 Subject: [LAU] PS: Off-topic: Streaming radio and calls from listeners In-Reply-To: <1403484708.3789.22.camel@archlinux> References: <20140622214801.GC13032@linuxaudio.org> <1403483865.3789.19.camel@archlinux> <1403484708.3789.22.camel@archlinux> Message-ID: On Sun, Jun 22, 2014 at 8:51 PM, Ralf Mardorf wrote: > On Mon, 2014-06-23 at 02:37 +0200, Ralf Mardorf wrote: >> AFAIK American NAZIs are free to transmit agitation by radio > > I didn't spread FUD, I watched a movie and have got other sources and > have got no reason to not believe it: It is true, in the US, the racist views of neo-Nazis, Ku Klux Klan, neo-Confederates, etc, are protected under Freedom of Speech. -- Brett W. McCoy -- http://www.brettwmccoy.com ------------------------------------------------------------------------ "In the rhythm of music a secret is hidden; If I were to divulge it, it would overturn the world." -- Jelaleddin Rumi From temcat at mail.ru Mon Jun 23 12:09:44 2014 From: temcat at mail.ru (Artem Vakhitov) Date: Mon, 23 Jun 2014 15:09:44 +0300 Subject: [LAU] Live bass guitar -> analog synth on a 2007 laptop: viable? In-Reply-To: <1403223037.1694.13.camel@archlinux> References: <53A375CE.8040000@mail.ru> <1403223037.1694.13.camel@archlinux> Message-ID: <53A81908.1000706@mail.ru> On 20.06.2014 3:10, Ralf Mardorf wrote: > Rakarrak's audio to midi conversion does work for some kinds of guitar > playing. I never tested it using a bass, it likely will become harder to > use it the lower the string sound its, this already was an issue for old > MIDI pickups. I don't know the quality of modern MIDI pickups, but I > guess it's better you get such a pickup. Without a MIDI pickup the > conversion only does work monophonic. > > As for the synth available for Linux, assumed there should be no xruns > at usable latencies, the sound of an onboard audio device unlikely will > be good enough for usage in a band, but the available synth might > satisfy your needs assumed you're using a high quality sound card. Will try Rakarrack. BTW, I don't need MIDI as such, I just need a way to control an analog synth using the audio signal from bass guitar. Regards, Artem Vakhitov From temcat at mail.ru Mon Jun 23 12:06:25 2014 From: temcat at mail.ru (Artem Vakhitov) Date: Mon, 23 Jun 2014 15:06:25 +0300 Subject: [LAU] Live bass guitar -> analog synth on a 2007 laptop: viable? In-Reply-To: <53A3C206.8040606@web.de> References: <53A375CE.8040000@mail.ru> <53A3C206.8040606@web.de> Message-ID: <53A81841.6060506@mail.ru> On 20.06.2014 8:09, hermann meyer wrote: > You may have a look at brian's amsynth site, he list and describe a > couple of Linux soft synth's, give sound examples and patches for them. > That could make it a bit easier for you to find what you are looking for. > http://amsynth.com/ Thank you Hermann, I will explore it. Regards, Artem Vakhitov From csanchezgs at gmail.com Mon Jun 23 11:28:08 2014 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Mon, 23 Jun 2014 13:28:08 +0200 Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: <53A73ADB.9060803@vis.nu> References: <20140622112315.GD27070@linuxaudio.org> <53A6C1F7.8020205@vis.nu> <53A73ADB.9060803@vis.nu> Message-ID: 2014-06-22 22:21 GMT+02:00 Sam Mulvey : > > -Sam > > > [1]: To the point where I'm designing a hardware interface that acts > like the older boards I'm used to, but is all digital underneath. > Right now, I'm using touchscreens, and having to look at what I'm doing > can actually be a problem sometimes. With pots and switches, I can > feel around. I would be interested in knowing more about that, if it's not a supersecret project. A common goal on my projects is to "feel" instead BTW, as I stated, my solution was to be some simple one but it's obvious that it's not aimed for a pro purppose. For that I would try something related with Jackd, I think you can get Skype "channel" through Jack in some way. Although SIP client it's still a viable solution even for a pro situation. -- C. sanchiavedraZ: * NEW / NUEVO: www.sanchiavedraZ.com * Musix GNU+Linux: www.musix.es From csanchezgs at gmail.com Mon Jun 23 11:30:09 2014 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Mon, 23 Jun 2014 13:30:09 +0200 Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: References: <20140622112315.GD27070@linuxaudio.org> <53A6C1F7.8020205@vis.nu> <53A73ADB.9060803@vis.nu> Message-ID: 2014-06-23 13:28 GMT+02:00 Carlos sanchiavedraz : > 2014-06-22 22:21 GMT+02:00 Sam Mulvey : >> >> -Sam >> >> >> [1]: To the point where I'm designing a hardware interface that acts >> like the older boards I'm used to, but is all digital underneath. >> Right now, I'm using touchscreens, and having to look at what I'm doing >> can actually be a problem sometimes. With pots and switches, I can >> feel around. > > > I would be interested in knowing more about that, if it's not a > supersecret project. A common goal on my projects is to "feel" instead > instead of "seeing" (screens, plugin UIs...). (Sorry, pushed send button accidentally) > BTW, as I stated, my solution was to be some simple one but it's > obvious that it's not aimed for a pro purppose. For that I would try > something related with Jackd, I think you can get Skype "channel" > through Jack in some way. Although SIP client it's still a viable > solution even for a pro situation. > > -- > > C. sanchiavedraZ: > * NEW / NUEVO: www.sanchiavedraZ.com > * Musix GNU+Linux: www.musix.es -- C. sanchiavedraZ: * NEW / NUEVO: www.sanchiavedraZ.com * Musix GNU+Linux: www.musix.es From ralf.mardorf at rocketmail.com Mon Jun 23 11:34:47 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Mon, 23 Jun 2014 13:34:47 +0200 Subject: [LAU] Live bass guitar -> analog synth on a 2007 laptop: viable? In-Reply-To: <53A81908.1000706@mail.ru> References: <53A375CE.8040000@mail.ru> <1403223037.1694.13.camel@archlinux> <53A81908.1000706@mail.ru> Message-ID: <1403523287.3789.28.camel@archlinux> On Mon, 2014-06-23 at 15:09 +0300, Artem Vakhitov wrote: > Will try Rakarrack. BTW, I don't need MIDI as such, I just need a way to > control an analog synth using the audio signal from bass guitar. I guess you wanted to point out that you will use a simulation of an analog synth ;). There are some virtual Linux synth that fit to my needs :), while I prefer to use MIDI to use real analog synth ;). Calf Monosynth, Phasex and last but not least Yoshimi (not really an anlog synth emulation ;) are some of the good once. Regarding to sound samplers that are able to play good, free available sounds we are screwed. Oops, I nearly forgot that some man really tries to emulate analog chips, http://bristol.sourceforge.net/ , I prefer my MIDI interface to use real synth, but it's worth to test his emulations, if you aren't addicted (as I'm) to CEM/Curtis microchips or the original C64 SID. [rocketmouse at archlinux ~]$ pacman -Qi linuxsampler | grep Licenses Licenses : GPL custom:exception For other distros than Arch Linux, there are usually are third party repositories providing linuxsampler + GUI's. From ralf.mardorf at rocketmail.com Mon Jun 23 11:41:06 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Mon, 23 Jun 2014 13:41:06 +0200 Subject: [LAU] Live bass guitar -> analog synth on a 2007 laptop: viable? In-Reply-To: <1403523287.3789.28.camel@archlinux> References: <53A375CE.8040000@mail.ru> <1403223037.1694.13.camel@archlinux> <53A81908.1000706@mail.ru> <1403523287.3789.28.camel@archlinux> Message-ID: <1403523666.3789.30.camel@archlinux> On Mon, 2014-06-23 at 13:34 +0200, Ralf Mardorf wrote: > http://bristol.sourceforge.net/ Stay away from this DX7 emulation. I own a real old DX7 in the metal case, if you like DX7 Bass sounds, better get an old DX7, but if you're willing to use an emulation, than better use Hexter instead of Bristol. From csanchezgs at gmail.com Mon Jun 23 12:45:35 2014 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Mon, 23 Jun 2014 14:45:35 +0200 Subject: [LAU] Live bass guitar -> analog synth on a 2007 laptop: viable? In-Reply-To: <53A81841.6060506@mail.ru> References: <53A375CE.8040000@mail.ru> <53A3C206.8040606@web.de> <53A81841.6060506@mail.ru> Message-ID: 2014-06-23 14:06 GMT+02:00 Artem Vakhitov : > On 20.06.2014 8:09, hermann meyer wrote: >> >> You may have a look at brian's amsynth site, he list and describe a >> couple of Linux soft synth's, give sound examples and patches for them. >> That could make it a bit easier for you to find what you are looking for. >> http://amsynth.com/ > > > Thank you Hermann, I will explore it. > > Regards, > Artem Vakhitov > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user Great resource to have an idea of the sound of each synth, good for Brian, and thanks for the resource. That will serve really well when I have to demonstrate the quality sounds that you can achieve. -- C. sanchiavedraZ: * NEW / NUEVO: www.sanchiavedraZ.com * Musix GNU+Linux: www.musix.es From louigi.verona at gmail.com Mon Jun 23 16:14:19 2014 From: louigi.verona at gmail.com (Louigi Verona) Date: Mon, 23 Jun 2014 20:14:19 +0400 Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: References: <20140622112315.GD27070@linuxaudio.org> <53A6C1F7.8020205@vis.nu> <53A73ADB.9060803@vis.nu> Message-ID: "Although SIP client it's still a viable solution even for a pro situation." Mmmm. Not really. I don't see people creating an account in some SIP client just to call a show. This would drastically decrease the amount of callers to almost zero. -- Louigi Verona http://www.louigiverona.ru/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From edogawa at aon.at Mon Jun 23 16:43:23 2014 From: edogawa at aon.at (Edgar Aichinger) Date: Mon, 23 Jun 2014 18:43:23 +0200 Subject: [LAU] Audio interface latency measurements In-Reply-To: <6051ac7f02f29ec2f8b442510bd9f34c@peterlutek.com> References: <6051ac7f02f29ec2f8b442510bd9f34c@peterlutek.com> Message-ID: <3193695.zc0zkloQYO@edhp> Am Samstag, 21. Juni 2014, 17:27:59 schrieb Peter Lutek: > On 2014-06-21 17:16, Len Ovens wrote: > > > >> ART USB Dual Tube Pre (USB 2.0) > > > > ...The only > > thing missing is midi. > > i add on an M-Audio midisport 2x2 for that... nice and small, and > completely plug'n'play too! Since when is that? Ii own a midisport 2x2 since many years and it needs the midisport firmware, I even had to alter the udev rule (patch went upstream I seem to remember) to make it handle my model revision properly... I just recently wrote a wiki page about it, for the linuxaudio hardware database, at http://wiki.linuxaudio.org/hw/m-audio_midisport_2x2 . If there's misleading, not sufficient or incorrect information I'd like to know to be able to correct/extend that page. Thanks, Edgar > > cheers! > .pltk. > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user From martin.peach at sympatico.ca Mon Jun 23 16:59:28 2014 From: martin.peach at sympatico.ca (Martin Peach) Date: Mon, 23 Jun 2014 12:59:28 -0400 Subject: [LAU] Audio interface latency measurements In-Reply-To: <3193695.zc0zkloQYO@edhp> References: <6051ac7f02f29ec2f8b442510bd9f34c@peterlutek.com> <3193695.zc0zkloQYO@edhp> Message-ID: On 2014-06-23 12:43, Edgar Aichinger wrote: > Am Samstag, 21. Juni 2014, 17:27:59 schrieb Peter Lutek: >> On 2014-06-21 17:16, Len Ovens wrote: >> i add on an M-Audio midisport 2x2 for that... nice and small, and >> completely plug'n'play too! > > Since when is that? Ii own a midisport 2x2 since many years and it needs the midisport firmware, I even had to alter the udev rule (patch went upstream I seem to remember) to make it handle my model revision properly... > I just recently wrote a wiki page about it, for the linuxaudio hardware database, at http://wiki.linuxaudio.org/hw/m-audio_midisport_2x2 . If there's misleading, not sufficient or incorrect information I'd like to know to be able to correct/extend that page. The new ones since a few years are class-compliant and don't need drivers. I have a 2X2 "Anniversary Edition" from 2012 running on a debian system with no issues. Martin From nescivi at gmail.com Mon Jun 23 17:52:55 2014 From: nescivi at gmail.com (nescivi) Date: Mon, 23 Jun 2014 19:52:55 +0200 Subject: [LAU] Creative Music Coding lab #12 in Amsterdam Message-ID: <53A86977.3090804@gmail.com> Calling all ChucK?ers, SuperColliders, Max and PureData patchers, CSounders, Fluxites, Overtoners, and all other tongues of creative coders. We welcome you to attend the fifth edition of the Creative Music Coding lab at STEIM. The CMC lab is an autonomous zone to try out sonic experiments as a group. And an opportunity to leverage the expertise of the group in realizing new artistic tools and processes through the medium of code. Many of the founding members of the group are indeed experts in their favorite languages, but we come from all technical levels of proficiency and enjoy helping one-another out. http://steim.org/event/creative-music-coding-lab-12/ DETAILS DATE: Tuesday, 24 June 2014 TIME: 19:30 ENTRY: FREE LOCATION: STEIM Concert Space, Utrechtsedwarsstraat 134 Amsterdam What ARE the goals for the lab? * a place for creative music coders to show work in progress, regardless of programming language or platform * a place to discuss and question techniques with fellow computer musicians * an informal stage for playing with others, livecoding sessions, jamming, and other fun experiments * an opportunity to meet like-minded artists, share talents, and start new collaborations * an opportunity to be exposed to new languages and improve cross-language fluency * a way to discover lesser known and emerging creative programming paradigms * a place to to discuss interconnections between programming environments * an environment for discussions on cultural contexts surrounding coding in the arts We encourage members of the group to use this as a platform for exploring ways in which we might be able to create and play together. Proposals for creative group investigations of hardware, software, and coding as process are welcome. As always, entry is free, tea and coffee will be provided. From csanchezgs at gmail.com Mon Jun 23 18:18:47 2014 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Mon, 23 Jun 2014 20:18:47 +0200 Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: References: <20140622112315.GD27070@linuxaudio.org> <53A6C1F7.8020205@vis.nu> <53A73ADB.9060803@vis.nu> Message-ID: 2014-06-23 18:14 GMT+02:00 Louigi Verona : > "Although SIP client it's still a viable > solution even for a pro situation." > > Mmmm. Not really. I don't see people creating an account in some SIP client > just to call a show. This would drastically decrease the amount of callers > to almost zero. > > > > -- > Louigi Verona > http://www.louigiverona.ru/ Hi Louigi. You're right, but I know at least some shows that use Skype, and you have to have an account then. Anyway, I would use SIP depending on the kind of PRO we're talking. I would not use for critical/sensible/important situations like TV or Radio; I'm referring more in amateur almost-pro situations. Anyway It was just to add some possibility to the conversation without knowing the exact context. P.S: One of my in progress projects maybe could one day be used for this kind purpose, and it would not use SIP nor anything alike, but rather browser to browser technology. -- C. sanchiavedraZ: * NEW / NUEVO: www.sanchiavedraZ.com * Musix GNU+Linux: www.musix.es From studiochanning at yahoo.com Mon Jun 23 18:32:07 2014 From: studiochanning at yahoo.com (Studio Channing) Date: Mon, 23 Jun 2014 12:32:07 -0600 Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: References: <20140622112315.GD27070@linuxaudio.org> <53A6C1F7.8020205@vis.nu> <53A73ADB.9060803@vis.nu> Message-ID: <53A872A7.1010901@yahoo.com> there are several "Internet Telephony Service Providers" that will sell you a "real" telephone number to use with a SIP address your listeners call a normal phone number and it rings your SIP phone it's not a free solution but you are just paying for the number, you need to buy a number anyway? On 06/23/2014 10:14 AM, Louigi Verona wrote: > "Although SIP client it's still a viable > solution even for a pro situation." > > Mmmm. Not really. I don't see people creating an account in some SIP > client just to call a show. This would drastically decrease the amount > of callers to almost zero. > > > > -- > Louigi Verona > http://www.louigiverona.ru/ > > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user -------------- next part -------------- An HTML attachment was scrubbed... URL: From len at ovenwerks.net Mon Jun 23 19:58:02 2014 From: len at ovenwerks.net (Len Ovens) Date: Mon, 23 Jun 2014 12:58:02 -0700 (PDT) Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: References: <20140622112315.GD27070@linuxaudio.org> <53A6C1F7.8020205@vis.nu> <53A73ADB.9060803@vis.nu> Message-ID: On Mon, 23 Jun 2014, Carlos sanchiavedraz wrote: > 2014-06-22 22:21 GMT+02:00 Sam Mulvey : >> >> [1]: To the point where I'm designing a hardware interface that acts >> like the older boards I'm used to, but is all digital underneath. >> Right now, I'm using touchscreens, and having to look at what I'm doing >> can actually be a problem sometimes. With pots and switches, I can >> feel around. > > > I would be interested in knowing more about that, if it's not a > supersecret project. A common goal on my projects is to "feel" instead One of the cheapest, easiest to hack pieces of hardware out there is a keyboard. Either the old serial ones (mini din) or the USB type already have 100 plus switches worked out. It is recognized by linux and X and so by any app, most of which have key mappings already. So long as you are using less than the full number of switches, it is pretty easy to select switch points where there can be multiple key presses at once. A pot is a bit harder (and there is midi pots out there) but could be done with two switches where moving the pot one way selects one switch and moving it the other way selects the other. Keyboards (if you just can't find them free... a coffee spill is not a problem) can be had for $3 each at thrift stores or even the dollar store brand new. -- Len Ovens www.ovenwerks.net From marc at hacklava.net Mon Jun 23 21:05:54 2014 From: marc at hacklava.net (Marc =?UTF-8?B?TGF2YWxsw6ll?=) Date: Mon, 23 Jun 2014 17:05:54 -0400 Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: <53A872A7.1010901@yahoo.com> References: <20140622112315.GD27070@linuxaudio.org> <53A6C1F7.8020205@vis.nu> <53A73ADB.9060803@vis.nu> <53A872A7.1010901@yahoo.com> Message-ID: <20140623170554.65f3f6b9@telecino> Mon, 23 Jun 2014 12:32:07 -0600, Studio Channing wrote : > there are several "Internet Telephony Service Providers" that will > sell you a "real" telephone number to use with a SIP address > > your listeners call a normal phone number and it rings your SIP phone > > it's not a free solution but you are just paying for the number, you > need to buy a number anyway? With my SIP VOIP provider, it's $1 per month for a local telephone number (with $10 setup fee), and it can also provide international numbers at a reasonable monthly fee. It uses the G711 codec, so the final quality is good enough for a radio show. Since it obviously work for people with SIP phones, maybe better codecs can be used. So I also believe that VOIP is a viable option for a pro situation, and I would prefer to use the Linphone software because of its command line client (which useful for automation and integration), and there's probably other free options. - Marc > On 06/23/2014 10:14 AM, Louigi Verona wrote: > > "Although SIP client it's still a viable > > solution even for a pro situation." > > > > Mmmm. Not really. I don't see people creating an account in some > > SIP client just to call a show. This would drastically decrease the > > amount of callers to almost zero. > > > > > > > > -- > > Louigi Verona > > http://www.louigiverona.ru/ > > > > > > _______________________________________________ > > Linux-audio-user mailing list > > Linux-audio-user at lists.linuxaudio.org > > http://lists.linuxaudio.org/listinfo/linux-audio-user > From len at ovenwerks.net Mon Jun 23 22:05:05 2014 From: len at ovenwerks.net (Len Ovens) Date: Mon, 23 Jun 2014 15:05:05 -0700 (PDT) Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: <20140623170554.65f3f6b9@telecino> References: <20140622112315.GD27070@linuxaudio.org> <53A6C1F7.8020205@vis.nu> <53A73ADB.9060803@vis.nu> <53A872A7.1010901@yahoo.com> <20140623170554.65f3f6b9@telecino> Message-ID: On Mon, 23 Jun 2014, Marc Lavall?e wrote: > for people with SIP phones, maybe better codecs can be used. So I > also believe that VOIP is a viable option for a pro situation, and I > would prefer to use the Linphone software because of its command line > client (which useful for automation and integration), and there's > probably other free options. In a profesional situation, asterisk might be better (perhaps on it's own machine) as it would provide for a switch board as well. On the same number you could have more than one person call in as the number is only used in sip to initiate the call. Plus, it does have a jack interface. I have not played with asterisk myself, so I don't know how easy it would be to set up. The main thing with any of these solutions... SIP or a local phone number limits you to local callers or people with sip clients. Skype, like it or not, has infected more computers than most virus have... so using skype gives you the world. For an Inet radio station... this is best. Another consideration with Inet radio stations is that often the stream server does not belong to the studio operator and the studio operator does not have an IP that is visible. It is possible to do ip to ip voip/sip calls so long as one end is visible no server is needed, but for a lot of people that is not the case... and even if it was, not that many people would know how to use it. So it would be fine for remote news gathering with trained people, but not casual call in. -- Len Ovens www.ovenwerks.net From ken at restivo.org Tue Jun 24 03:58:22 2014 From: ken at restivo.org (Ken Restivo) Date: Mon, 23 Jun 2014 20:58:22 -0700 Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: <20140622214801.GC13032@linuxaudio.org> References: <20140622214801.GC13032@linuxaudio.org> Message-ID: <20140624035822.GA28868@q400a.mobile.restivo.org> On Sun, Jun 22, 2014 at 09:48:01PM +0000, Fons Adriaensen wrote: > On Sun, Jun 22, 2014 at 09:13:27AM -0700, Len Ovens wrote: > > > With regard to content filtering of live calls, not just from a > > legal perspective, but also from what you want your listeners to > > hear. The standard method used here when I was in the business years > > ago was to delay the whole studio audio about 7 seconds and provide > > a cut button that shut off the audio after the delay. > > This seven seconds 'profanity delay' seems to be a US thing, > and probably an FCC requirement. I've never seen such things > used on this side of the Atlantic. > It's definitely an FCC thing, and a US-ism. Background: http://en.wikipedia.org/wiki/Federal_Communications_Commission_v._Pacifica_Foundation Source material, (warning NSFFCC): http://www.youtube.com/watch?v=kyBH5oNQOS0 -ken From len at ovenwerks.net Tue Jun 24 04:56:53 2014 From: len at ovenwerks.net (Len Ovens) Date: Mon, 23 Jun 2014 21:56:53 -0700 (PDT) Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: References: Message-ID: On Sun, 22 Jun 2014, Len Ovens wrote: > Ok, I did the full setup minus the last step of encoding and streaming to the > server. Tried it with streaming. Steaming to server adds no more than 5% CPU use. -- Len Ovens www.ovenwerks.net From ken at restivo.org Tue Jun 24 05:39:45 2014 From: ken at restivo.org (Ken Restivo) Date: Mon, 23 Jun 2014 22:39:45 -0700 Subject: [LAU] Live bass guitar -> analog synth on a 2007 laptop: viable? In-Reply-To: <53A375CE.8040000@mail.ru> References: <53A375CE.8040000@mail.ru> Message-ID: <20140624053945.GB32510@q400a.mobile.restivo.org> On Fri, Jun 20, 2014 at 03:44:14AM +0400, Artem Vakhitov wrote: > Hello fellow Linux audio users, > > I'm back to dabbling with Linux as an audio system. Among other > things, I play bass guitar in a synth pop band and recently started > to get interested in bass synthesizers. I could of course buy > something like Markbass Super Synth, but then I thought - maybe I > could cobble together something using my Samsung Q35 laptop and > Linux? The laptop has a dual core processor and 2.5GB RAM. The sound > card is a variable here: it could be the built-in Intel HDA (for > this particular purpose, why not), or an Infrasonic DeuX (Firewire) > that I have, or even some used Echo Indigo (PCMCIA). > > Is what I want viable at all with a reasonable latency? What > software setup can I use for that? Does anybody here use something > similar live? > In my experience, you've got more than enough hardware there to do many things. I made tons of music with softsynths on top of softsynths, plugins, etc, on a 2007-era Asus Core2Duo at 2Ghz with 2GB RAM. Much of that music was posted here. I also mixed a band CD using that same hardware. And I used it as a live synth for a few years too. The key is to get a low-latency kernel. At the time (2007) I had to homegrow that stuff, but nowadays it seems AVLinux is the easiest to set up with that. The latency I got with my low-end FastTrack Pro USB interface was 3 periods at 128 each, 44.1khz. More than goood enough for recording/mixing/softsynths, and totally reliable, never glitched on me once after I got everything dialed in. -ken From csanchezgs at gmail.com Tue Jun 24 08:14:51 2014 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Tue, 24 Jun 2014 10:14:51 +0200 Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: References: <20140622112315.GD27070@linuxaudio.org> <53A6C1F7.8020205@vis.nu> <53A73ADB.9060803@vis.nu> Message-ID: 2014-06-23 21:58 GMT+02:00 Len Ovens : > On Mon, 23 Jun 2014, Carlos sanchiavedraz wrote: > >> 2014-06-22 22:21 GMT+02:00 Sam Mulvey : >>> >>> >>> [1]: To the point where I'm designing a hardware interface that acts >>> like the older boards I'm used to, but is all digital underneath. >>> Right now, I'm using touchscreens, and having to look at what I'm doing >>> can actually be a problem sometimes. With pots and switches, I can >>> feel around. >> >> >> >> I would be interested in knowing more about that, if it's not a >> supersecret project. A common goal on my projects is to "feel" instead > > > One of the cheapest, easiest to hack pieces of hardware out there is a > keyboard. Either the old serial ones (mini din) or the USB type already have > 100 plus switches worked out. It is recognized by linux and X and so by any > app, most of which have key mappings already. So long as you are using less > than the full number of switches, it is pretty easy to select switch points > where there can be multiple key presses at once. A pot is a bit harder (and > there is midi pots out there) but could be done with two switches where > moving the pot one way selects one switch and moving it the other way > selects the other. > > Keyboards (if you just can't find them free... a coffee spill is not a > problem) can be had for $3 each at thrift stores or even the dollar store > brand new. > > -- > Len Ovens > www.ovenwerks.net > Good idea, very convenient and plastic, and quite cheap so it is less risky to start something for a long run. Keep us informed, please. Thanks -- C. sanchiavedraZ: * NEW / NUEVO: www.sanchiavedraZ.com * Musix GNU+Linux: www.musix.es From jeremy at autostatic.com Tue Jun 24 08:35:19 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Tue, 24 Jun 2014 10:35:19 +0200 Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: References: <20140622112315.GD27070@linuxaudio.org> <53A6C1F7.8020205@vis.nu> <53A73ADB.9060803@vis.nu> Message-ID: <53A93847.5050503@autostatic.com> On 06/23/2014 06:14 PM, Louigi Verona wrote: > Mmmm. Not really. I don't see people creating an account in some SIP client > just to call a show. This would drastically decrease the amount of callers > to almost zero. Hi Louigi, Couple the SIP account to a DID (http://en.wikipedia.org /wiki/Direct_inward_dial) and one can call from anywhere. Or does calling in have to be free? Jeremy -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 836 bytes Desc: OpenPGP digital signature URL: From jeremy at autostatic.com Tue Jun 24 08:37:30 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Tue, 24 Jun 2014 10:37:30 +0200 Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: References: <20140622112315.GD27070@linuxaudio.org> <53A6C1F7.8020205@vis.nu> <53A73ADB.9060803@vis.nu> Message-ID: <53A938CA.2020208@autostatic.com> On 06/23/2014 08:18 PM, Carlos sanchiavedraz wrote: > P.S: One of my in progress projects maybe could one day be used for > this kind purpose, and it would not use SIP nor anything alike, but > rather browser to browser technology. That already exists, it's called WebRTC: http://en.wikipedia.org/wiki/WebRTC Jeremy -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 836 bytes Desc: OpenPGP digital signature URL: From fons at linuxaudio.org Tue Jun 24 08:44:03 2014 From: fons at linuxaudio.org (Fons Adriaensen) Date: Tue, 24 Jun 2014 08:44:03 +0000 Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: <20140624035822.GA28868@q400a.mobile.restivo.org> References: <20140622214801.GC13032@linuxaudio.org> <20140624035822.GA28868@q400a.mobile.restivo.org> Message-ID: <20140624084403.GB24657@linuxaudio.org> On Mon, Jun 23, 2014 at 08:58:22PM -0700, Ken Restivo wrote: > It's definitely an FCC thing, and a US-ism. > > Background: http://en.wikipedia.org/wiki/Federal_Communications_Commission_v._Pacifica_Foundation > > Source material, (warning NSFFCC): http://www.youtube.com/watch?v=kyBH5oNQOS0 See also: http://en.wikipedia.org/wiki/Broadcast_delay#Early_use Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) From atte at youmail.dk Tue Jun 24 08:47:04 2014 From: atte at youmail.dk (Atte) Date: Tue, 24 Jun 2014 10:47:04 +0200 Subject: [LAU] howto get pcmcia firewire card working Message-ID: <53A93B08.8090905@youmail.dk> Hi I got a new laptop without firewire. So I bought a delock pcmcia firewire card to be able to use my firewire soundcard. Since I couldn't see any hardware specs for the cards I looked at before buying, I simply went for the cheapest... I've been working with firewire a lot in the past, but only with build in controllers, so I'm not sure if I should do something special to get this card going. I obviously tried plugging it in and starting jack with the firewire driver for all available interfaces, but I get these errors: firewire ERR: FFADO: Error creating virtual device Cannot attach audio driver JackServer::Open failed with -1 no message buffer overruns Failed to open server 10:35:53.707 JACK was stopped with exit status=255. 10:35:54.785 Could not connect to JACK server as client. - Overall operation failed. - Unable to connect to server. Please check the messages window for more info. Cannot connect to server socket err = No such file or directory Cannot connect to server request channel jack server is not running or cannot be started Those look familiar, they are exactly the same as on my desktop pc (that works with wirewire) when I don't have the soundcard plugged in. Of course it might be that the pcmcia card is simply not working under linux (I have no other os running, so I have to assume the card is in fact working under, say, windows). But what actions could I take to either get it working or convince myself that it's simply not supported? NB: I run crunchbang linux (=debian wheezy), which were installed prior to getting the card. The laptop is a Lenovo x220. NB2: I couldn't see anything in dmesg after inserting the card and lspci is the same with the card attached and detached... Any pointers appreciated! -- Atte http://atte.dk http://modlys.dk From clemens at ladisch.de Tue Jun 24 09:03:21 2014 From: clemens at ladisch.de (Clemens Ladisch) Date: Tue, 24 Jun 2014 11:03:21 +0200 Subject: [LAU] howto get pcmcia firewire card working In-Reply-To: <53A93B08.8090905@youmail.dk> References: <53A93B08.8090905@youmail.dk> Message-ID: <53A93ED9.3050902@ladisch.de> Atte wrote: > I got a new laptop without firewire. So I bought a delock pcmcia > firewire card > > The laptop is a Lenovo x220. I'd guess the card is not PCMCIA but ExpressCard. > NB2: I couldn't see anything in dmesg after inserting the card and lspci > is the same with the card attached and detached... There's your problem. Try loading the acpiphp module. Regards, Clemens From willgodfrey at musically.me.uk Tue Jun 24 09:17:03 2014 From: willgodfrey at musically.me.uk (Will Godfrey) Date: Tue, 24 Jun 2014 10:17:03 +0100 Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: <20140624084403.GB24657@linuxaudio.org> References: <20140622214801.GC13032@linuxaudio.org> <20140624035822.GA28868@q400a.mobile.restivo.org> <20140624084403.GB24657@linuxaudio.org> Message-ID: <20140624101703.6e15f86d@debian> On Tue, 24 Jun 2014 08:44:03 +0000 Fons Adriaensen wrote: > On Mon, Jun 23, 2014 at 08:58:22PM -0700, Ken Restivo wrote: > > > It's definitely an FCC thing, and a US-ism. > > > > Background: http://en.wikipedia.org/wiki/Federal_Communications_Commission_v._Pacifica_Foundation > > > > Source material, (warning NSFFCC): http://www.youtube.com/watch?v=kyBH5oNQOS0 > > See also: http://en.wikipedia.org/wiki/Broadcast_delay#Early_use > > Ciao, > Can't be certain, but I have a feeling the BBC has, or had something like this for live radio. If so, knowing the BBC they probably developed their own kit. After all, at their height, they designed and built their own mixing desks. -- Will J Godfrey http://www.musically.me.uk Say you have a poem and I have a tune. Exchange them and we can both have a poem, a tune, and a song. From willgodfrey at musically.me.uk Tue Jun 24 09:20:41 2014 From: willgodfrey at musically.me.uk (Will Godfrey) Date: Tue, 24 Jun 2014 10:20:41 +0100 Subject: [LAU] audio recording of my set at LAC 2014 (streaming, download) In-Reply-To: References: Message-ID: <20140624102041.7aaeb402@debian> On Thu, 19 Jun 2014 19:14:05 +0200 Carlos sanchiavedraz wrote: > I'm also listening to it right now. > > Interesting and curious, it'd be nice to see a video of you > live-coding to see how's the process. > > Thanks for sharing to us poor mortals that couldn't go to LAC. > > > 2014-05-12 22:19 GMT+02:00 Matej Fr?be : > > Great, I'm listening to the recording right now. I like your approach to > > generating rhythms. > > > > Regards, > > Matej > > > > > > 2014-05-11 16:31 GMT+02:00 Renick Bell : > >> > >> Sorry, I messed up that mp3 link: > >> > >> http://renickbell.net/sound/renick-bell-live-lac-2014.mp3 > >> > >> On Sun, May 11, 2014 at 11:30 PM, Renick Bell wrote: > >> > I'm really glad that I had a chance to perform at LAC this year. > >> > Thanks to everyone in the audience! > >> > > >> > Here's a link to a streaming version on SoundCloud: > >> > > >> > > >> > https://soundcloud.com/renick/live-at-the-linux-audio-conference-2014-karlsruhe-germany-may-5th-2014 > >> > > >> > If you prefer, here are direct download links in a variety of formats: > >> > > >> > ogg: http://renickbell.net/sound/renick-bell-live-lac-2014.ogg > >> > > >> > flac: http://renickbell.net/sound/renick-bell-live-lac-2014.flac > >> > > >> > mp3: http://renick/renickbell.net/sound/renick-bell-live-lac-2014.mp3 > >> > > >> > I'll put a video online after I've had a chance to edit it. > >> > > >> > Best to everyone, > >> > > >> > Renick > >> > > >> > -- > >> > Renick Bell > >> > - http://renickbell.net > >> > - http://twitter.com/renick > >> > - http://the3rd2nd.com > >> > >> > >> > >> -- > >> Renick Bell > >> - http://renickbell.net > >> - http://twitter.com/renick > >> - http://the3rd2nd.com Remember this well. Very impressive work. -- Will J Godfrey http://www.musically.me.uk Say you have a poem and I have a tune. Exchange them and we can both have a poem, a tune, and a song. From willgodfrey at musically.me.uk Tue Jun 24 09:29:59 2014 From: willgodfrey at musically.me.uk (Will Godfrey) Date: Tue, 24 Jun 2014 10:29:59 +0100 Subject: [LAU] Music made with fun thanx to linux In-Reply-To: References: Message-ID: <20140624102959.086c5429@debian> On Fri, 20 Jun 2014 12:45:00 +0200 Carlos sanchiavedraz wrote: > 2014-05-16 18:10 GMT+02:00 Set Hallstr?m : > > Hi! > > > > Recently i traded my earfull portastudio against memories of happy > > unafortable OSystems some years ago, and then i made this: > > https://soundcloud.com/sakrecoer/sets/music-made-with-linux > > > > I may hope you like it like i do, but i know life is full of inexpectations > > :) > > > > With wishes of happy week endings, > > > > -- > > Set Hallstr?m > > AKA > > reSet Sakrecoer > > http://sakrecoer.com > > > > > > _______________________________________________ > > Linux-audio-user mailing list > > Linux-audio-user at lists.linuxaudio.org > > http://lists.linuxaudio.org/listinfo/linux-audio-user > > > > What a session, Almost 2 hours! Not my main kind of music but It > sounds good, it develops and progress well. And the overall EQ and > balance is also good. > > Thanks for sharing. > Not the sort of thing I would listen too very often, but very well done and interesting, some for the vocal sections especially. -- Will J Godfrey http://www.musically.me.uk Say you have a poem and I have a tune. Exchange them and we can both have a poem, a tune, and a song. From atte at youmail.dk Tue Jun 24 10:30:46 2014 From: atte at youmail.dk (Atte) Date: Tue, 24 Jun 2014 12:30:46 +0200 Subject: [LAU] howto get pcmcia firewire card working In-Reply-To: <53A93ED9.3050902@ladisch.de> References: <53A93B08.8090905@youmail.dk> <53A93ED9.3050902@ladisch.de> Message-ID: <53A95356.7060703@youmail.dk> On 06/24/2014 11:03 AM, Clemens Ladisch wrote: > Atte wrote: >> I got a new laptop without firewire. So I bought a delock pcmcia >> firewire card >> >> The laptop is a Lenovo x220. > > I'd guess the card is not PCMCIA but ExpressCard. Might be, never used either before, actually I thought they were two names for the same thing... >> NB2: I couldn't see anything in dmesg after inserting the card and lspci >> is the same with the card attached and detached... > > There's your problem. > > Try loading the acpiphp module. Hmmm atte at vestbjerg:~$ lsmod | grep acpip atte at vestbjerg:~$ sudo modprobe acpiphp [sudo] password for atte: atte at vestbjerg:~$ lsmod | grep acpip atte at vestbjerg:~$ sudo modprobe acpiphp_ibm ERROR: could not insert 'acpiphp_ibm': No such device atte at vestbjerg:~$ Strange thing (but what do I know), "sudo modprobe acpiphp" completes to "sudo modprobe acpiphp_ibm", suggesting that there's no acpiphp available on the system, still I get no error when loading it *and* it's not showing in lsmod after attempted load. Here's what dmesg have to say: atte at vestbjerg:~$ dmesg | tail -n 2 [ 45.626648] iwlwifi 0000:03:00.0: Tx aggregation enabled on ra = a0:21:b7:d7:fa:15 tid = 0 [ 90.387533] acpiphp_ibm: ibm_acpiphp_init: acpi_walk_namespace failed Reading http://www.thinkwiki.org/wiki/ExpressCard_slot + your reply, makes me thing I made the mistake of buying a pcmcia, whereas I in fact should have bought an expresscard/54. -- Atte http://atte.dk http://modlys.dk From atte at youmail.dk Tue Jun 24 10:42:01 2014 From: atte at youmail.dk (Atte) Date: Tue, 24 Jun 2014 12:42:01 +0200 Subject: [LAU] howto get pcmcia firewire card working In-Reply-To: <53A95356.7060703@youmail.dk> References: <53A93B08.8090905@youmail.dk> <53A93ED9.3050902@ladisch.de> <53A95356.7060703@youmail.dk> Message-ID: <53A955F9.1060002@youmail.dk> On 06/24/2014 12:30 PM, Atte wrote: > whereas I in fact > should have bought an expresscard/54. Or expresscard/34, since http://www.thinkwiki.org/wiki/ExpressCard_slot suggests the 34 or 54 refer to the width (in mm) of the physical card, that my x220 has an expresscard/54 *and* that "ExpressCard/54 slots can accept ExpressCard/34 cards, but not the other way around." Correct? The reason I'm asking this, is because I can't find any /54 cards but a lot of /34 cards... -- Atte http://atte.dk http://modlys.dk From grib at billgribble.com Tue Jun 24 10:59:55 2014 From: grib at billgribble.com (Bill Gribble) Date: Tue, 24 Jun 2014 06:59:55 -0400 Subject: [LAU] howto get pcmcia firewire card working In-Reply-To: <53A95356.7060703@youmail.dk> References: <53A93B08.8090905@youmail.dk> <53A93ED9.3050902@ladisch.de> <53A95356.7060703@youmail.dk> Message-ID: <395936B8-18EF-486C-BB85-EAEA60B72E40@billgribble.com> There were problems with ExpressCard hotplug last I checked. Getting the kernel to recognize a card when it's plugged in is not possible all the time. Try booting with the card already plugged in and see if it's recognized then. Thanks, Bill Gribble > On Jun 24, 2014, at 6:30, Atte wrote: > >> On 06/24/2014 11:03 AM, Clemens Ladisch wrote: >> Atte wrote: >>> I got a new laptop without firewire. So I bought a delock pcmcia >>> firewire card >>> >>> The laptop is a Lenovo x220. >> >> I'd guess the card is not PCMCIA but ExpressCard. > > Might be, never used either before, actually I thought they were two > names for the same thing... > >>> NB2: I couldn't see anything in dmesg after inserting the card and lspci >>> is the same with the card attached and detached... >> >> There's your problem. >> >> Try loading the acpiphp module. > > Hmmm > > atte at vestbjerg:~$ lsmod | grep acpip > atte at vestbjerg:~$ sudo modprobe acpiphp > [sudo] password for atte: > atte at vestbjerg:~$ lsmod | grep acpip > atte at vestbjerg:~$ sudo modprobe acpiphp_ibm > ERROR: could not insert 'acpiphp_ibm': No such device > atte at vestbjerg:~$ > > Strange thing (but what do I know), "sudo modprobe acpiphp" > completes to "sudo modprobe acpiphp_ibm", suggesting that there's no > acpiphp available on the system, still I get no error when loading it > *and* it's not showing in lsmod after attempted load. > > Here's what dmesg have to say: > > atte at vestbjerg:~$ dmesg | tail -n 2 > [ 45.626648] iwlwifi 0000:03:00.0: Tx aggregation enabled on ra = > a0:21:b7:d7:fa:15 tid = 0 > [ 90.387533] acpiphp_ibm: ibm_acpiphp_init: acpi_walk_namespace failed > > Reading http://www.thinkwiki.org/wiki/ExpressCard_slot + your reply, > makes me thing I made the mistake of buying a pcmcia, whereas I in fact > should have bought an expresscard/54. > > -- > Atte > > http://atte.dk http://modlys.dk > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user From atte at youmail.dk Tue Jun 24 11:18:13 2014 From: atte at youmail.dk (Atte) Date: Tue, 24 Jun 2014 13:18:13 +0200 Subject: [LAU] howto get pcmcia firewire card working In-Reply-To: <395936B8-18EF-486C-BB85-EAEA60B72E40@billgribble.com> References: <53A93B08.8090905@youmail.dk> <53A93ED9.3050902@ladisch.de> <53A95356.7060703@youmail.dk> <395936B8-18EF-486C-BB85-EAEA60B72E40@billgribble.com> Message-ID: <53A95E75.2080606@youmail.dk> On 06/24/2014 12:59 PM, Bill Gribble wrote: > There were problems with ExpressCard hotplug last I checked. Getting > the kernel to recognize a card when it's plugged in is not possible > all the time. Try booting with the card already plugged in and see if > it's recognized then. Thanks for the suggestion, but unfortunately that doesn't change anything :-( -- Atte http://atte.dk http://modlys.dk From jeremy at autostatic.com Tue Jun 24 11:23:55 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Tue, 24 Jun 2014 13:23:55 +0200 Subject: [LAU] howto get pcmcia firewire card working In-Reply-To: <53A93B08.8090905@youmail.dk> References: <53A93B08.8090905@youmail.dk> Message-ID: <53A95FCB.3010602@autostatic.com> On 06/24/2014 10:47 AM, Atte wrote: > I got a new laptop without firewire. So I bought a delock pcmcia > firewire card to be able to use my firewire soundcard. Since I couldn't > see any hardware specs for the cards I looked at before buying, I simply > went for the cheapest... Hi Atte, Do you have any exact specifications of the delock card? Does it have a model number? Other designations that might help figuring out what kind of card this is? And if you're looking for a cheap option, try to get a hold of a Dynex DX-ECFW ExpressCard which has a LSI/Agere FW643 chipset. Jeremy -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 836 bytes Desc: OpenPGP digital signature URL: From atte at youmail.dk Tue Jun 24 11:35:29 2014 From: atte at youmail.dk (Atte) Date: Tue, 24 Jun 2014 13:35:29 +0200 Subject: [LAU] howto get pcmcia firewire card working In-Reply-To: <53A95FCB.3010602@autostatic.com> References: <53A93B08.8090905@youmail.dk> <53A95FCB.3010602@autostatic.com> Message-ID: <53A96281.8080003@youmail.dk> On 06/24/2014 01:23 PM, Jeremy Jongepier wrote: > Do you have any exact specifications of the delock card? Does it have a > model number? Other designations that might help figuring out what kind > of card this is? All I can find is online, for instance: http://www.delock.de/produkte/G_61114/dokumente.html -- Atte http://atte.dk http://modlys.dk From clemens at ladisch.de Tue Jun 24 12:16:57 2014 From: clemens at ladisch.de (Clemens Ladisch) Date: Tue, 24 Jun 2014 14:16:57 +0200 Subject: [LAU] howto get pcmcia firewire card working In-Reply-To: <53A96281.8080003@youmail.dk> References: <53A93B08.8090905@youmail.dk> <53A95FCB.3010602@autostatic.com> <53A96281.8080003@youmail.dk> Message-ID: <53A96C39.7060807@ladisch.de> Atte wrote: > http://www.delock.de/produkte/G_61114/dokumente.html This is CardBus, not ExpressCard. See . You need a different card. Regards, Clemens From csanchezgs at gmail.com Tue Jun 24 14:45:59 2014 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Tue, 24 Jun 2014 16:45:59 +0200 Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: <53A938CA.2020208@autostatic.com> References: <20140622112315.GD27070@linuxaudio.org> <53A6C1F7.8020205@vis.nu> <53A73ADB.9060803@vis.nu> <53A938CA.2020208@autostatic.com> Message-ID: 2014-06-24 10:37 GMT+02:00 Jeremy Jongepier : > On 06/23/2014 08:18 PM, Carlos sanchiavedraz wrote: >> P.S: One of my in progress projects maybe could one day be used for >> this kind purpose, and it would not use SIP nor anything alike, but >> rather browser to browser technology. > > That already exists, it's called WebRTC: http://en.wikipedia.org/wiki/WebRTC > > Jeremy > > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > Yes, you got it (I didn't mean I was inventing from scratch :) ), I use that technology along with others that HTML5 and WebAPIs provides. -- C. sanchiavedraZ: * NEW / NUEVO: www.sanchiavedraZ.com * Musix GNU+Linux: www.musix.es From csanchezgs at gmail.com Tue Jun 24 14:50:29 2014 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Tue, 24 Jun 2014 16:50:29 +0200 Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: References: <20140622112315.GD27070@linuxaudio.org> <53A6C1F7.8020205@vis.nu> <53A73ADB.9060803@vis.nu> <53A938CA.2020208@autostatic.com> Message-ID: 2014-06-24 16:45 GMT+02:00 Carlos sanchiavedraz : > 2014-06-24 10:37 GMT+02:00 Jeremy Jongepier : >> On 06/23/2014 08:18 PM, Carlos sanchiavedraz wrote: >>> P.S: One of my in progress projects maybe could one day be used for >>> this kind purpose, and it would not use SIP nor anything alike, but >>> rather browser to browser technology. >> >> That already exists, it's called WebRTC: http://en.wikipedia.org/wiki/WebRTC >> >> Jeremy >> >> >> _______________________________________________ >> Linux-audio-user mailing list >> Linux-audio-user at lists.linuxaudio.org >> http://lists.linuxaudio.org/listinfo/linux-audio-user >> > > Yes, you got it (I didn't mean I was inventing from scratch :) ), I > use that technology along with others that HTML5 and WebAPIs provides. > > > -- > > C. sanchiavedraZ: > * NEW / NUEVO: www.sanchiavedraZ.com > * Musix GNU+Linux: www.musix.es BTW, to the OP, maybe you can get the output of a web browser from Jack using this alternative if you ever have no other choice: https://talky.io/ -- C. sanchiavedraZ: * NEW / NUEVO: www.sanchiavedraZ.com * Musix GNU+Linux: www.musix.es From ken at restivo.org Tue Jun 24 15:12:23 2014 From: ken at restivo.org (Ken Restivo) Date: Tue, 24 Jun 2014 08:12:23 -0700 Subject: [LAU] Streaming radio and calls from listeners In-Reply-To: <20140624084403.GB24657@linuxaudio.org> References: <20140622214801.GC13032@linuxaudio.org> <20140624035822.GA28868@q400a.mobile.restivo.org> <20140624084403.GB24657@linuxaudio.org> Message-ID: <920238a2-1079-4ca8-934f-60ad620f5e2f@email.android.com> I remember those! I interned in a radio station in high school. These delays were implemented with a "cart", an 8-track-style cartridge. The carts were usually used for commercials and occasionally music. They were cut to a minute or several minutes of 1/4" tape, specially lubed for repeated use. The delay unit was a specially wired cart machine. The tape delay cart was just a cart with a very short loop, like 3 or 7 seconds. The cart machine had read and write heads and erase heads. The layout was, read, erase, write. So the delay machine would write the material from the write head, the tape would loop around, the material would be read off of the read head, then erased. If you turned the erase head off you'd get Frippertronics-like effects. The carts would wear out fast, and in the course of a live show you'd have to replace them often, like every hour or so, usually during commercial. The boards were wired up with sends so you could send the whole program thru the delay. I remember the boards, usually an RCA BC-7A. The pots were huge and made of bakelite, and they had gold contacts in a radial step pattern. They were already ancient by the time I touched one. If you potted things up slowly you could hear the stepping. Dunno why I just remembered all that from 30 years ago, but thanks for jogging my memory. Fons Adriaensen wrote: >On Mon, Jun 23, 2014 at 08:58:22PM -0700, Ken Restivo wrote: > >> It's definitely an FCC thing, and a US-ism. >> >> Background: >http://en.wikipedia.org/wiki/Federal_Communications_Commission_v._Pacifica_Foundation >> >> Source material, (warning NSFFCC): >http://www.youtube.com/watch?v=kyBH5oNQOS0 > >See also: http://en.wikipedia.org/wiki/Broadcast_delay#Early_use > >Ciao, > >-- >FA > >A world of exhaustive, reliable metadata would be an utopia. >It's also a pipe-dream, founded on self-delusion, nerd hubris >and hysterically inflated market opportunities. (Cory Doctorow) From peter at peterlutek.com Tue Jun 24 18:37:56 2014 From: peter at peterlutek.com (Peter Lutek) Date: Tue, 24 Jun 2014 14:37:56 -0400 Subject: [LAU] Audio interface latency measurements In-Reply-To: References: <6051ac7f02f29ec2f8b442510bd9f34c@peterlutek.com> <3193695.zc0zkloQYO@edhp> Message-ID: <5a7fae2230a5baebe1b57a379042c592@peterlutek.com> On 2014-06-23 12:59, Martin Peach wrote: > On 2014-06-23 12:43, Edgar Aichinger wrote: >> Am Samstag, 21. Juni 2014, 17:27:59 schrieb Peter Lutek: >>> On 2014-06-21 17:16, Len Ovens wrote: >>> i add on an M-Audio midisport 2x2 for that... nice and small, and >>> completely plug'n'play too! >> >> Since when is that? Ii own a midisport 2x2 since many years and it >> needs the midisport firmware, I even had to alter the udev rule (patch >> went upstream I seem to remember) to make it handle my model revision >> properly... >> I just recently wrote a wiki page about it, for the linuxaudio >> hardware database, at >> http://wiki.linuxaudio.org/hw/m-audio_midisport_2x2 . If there's >> misleading, not sufficient or incorrect information I'd like to know >> to be able to correct/extend that page. > > The new ones since a few years are class-compliant and don't need > drivers. I have a 2X2 "Anniversary Edition" from 2012 running on a > debian system with no issues. yes, mine is also the "Anniversary Edition", purchased in 2013. cheers! .pltk. From unaudio at gmail.com Wed Jun 25 09:49:01 2014 From: unaudio at gmail.com (Vytautas Jancauskas) Date: Wed, 25 Jun 2014 12:49:01 +0300 Subject: [LAU] Music made with fun thanx to linux In-Reply-To: References: Message-ID: On Fri, May 16, 2014 at 7:10 PM, Set Hallstr?m wrote: > Hi! > > Recently i traded my earfull portastudio against memories of happy > unafortable OSystems some years ago, and then i made this: > https://soundcloud.com/sakrecoer/sets/music-made-with-linux > > I may hope you like it like i do, but i know life is full of > inexpectations :) > > With wishes of happy week endings, > > I like your version of the English language. -------------- next part -------------- An HTML attachment was scrubbed... URL: From atte at youmail.dk Wed Jun 25 12:09:39 2014 From: atte at youmail.dk (Atte) Date: Wed, 25 Jun 2014 14:09:39 +0200 Subject: [LAU] howto get pcmcia firewire card working In-Reply-To: <53A96C39.7060807@ladisch.de> References: <53A93B08.8090905@youmail.dk> <53A95FCB.3010602@autostatic.com> <53A96281.8080003@youmail.dk> <53A96C39.7060807@ladisch.de> Message-ID: <53AABC03.7040404@youmail.dk> On 06/24/2014 02:16 PM, Clemens Ladisch wrote: > You need a different card. I suspected that, at least it was cheap :-) Thanks! -- Atte http://atte.dk http://modlys.dk From sakrecoer at gmail.com Wed Jun 25 16:17:31 2014 From: sakrecoer at gmail.com (=?UTF-8?Q?Set_Hallstr=C3=B6m?=) Date: Wed, 25 Jun 2014 18:17:31 +0200 Subject: [LAU] Music made with fun thanx to linux In-Reply-To: References: Message-ID: > > > Thanks!! :) <3 your response make me very happy! I do have strange ways with tongues and languages, i speak fluent french, swedish, german spannish and english (but i spell them all like a fool). Hence i like to play arround with synonyms, onomatopeias and so forth.... For good and worse: it has taken me as far as it has gotten me in deep trouble for being missunderstood. haha. After all words become the frames of our thoughts? I mainly use hardware synths and sequencers, which makes the EQ'ing and arranging a bit easier, because i tune the sound to my taste before recording and arrange on the go, or while layering takes. Hence, my computer process is mainly deletion of errors and fine-mixing when i figure its really needed. I record all the audio with ardour thru a teratec phase88 firewire rack: a pure delight to work with! Again, thank you for getting back to me! Yours, -- Set Hallstr?m AKA reSet Sakrecoer http://sakrecoer.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From waynedpj at in-giro.org Wed Jun 25 23:37:47 2014 From: waynedpj at in-giro.org (Wayne DePrince Jr.) Date: Wed, 25 Jun 2014 19:37:47 -0400 Subject: [LAU] howto get pcmcia firewire card working In-Reply-To: <53A93B08.8090905@youmail.dk> References: <53A93B08.8090905@youmail.dk> Message-ID: <1403739467.16698.11.camel@localhost.localdomain> On mar, 2014-06-24 at 10:47 +0200, Atte wrote: > Hi > > I got a new laptop without firewire. So I bought a delock pcmcia > firewire card to be able to use my firewire soundcard. Since I couldn't > see any hardware specs for the cards I looked at before buying, I simply > went for the cheapest... > > I've been working with firewire a lot in the past, but only with build > in controllers, so I'm not sure if I should do something special to get > this card going. > > I obviously tried plugging it in and starting jack with the firewire > driver for all available interfaces, but I get these errors: > > firewire ERR: FFADO: Error creating virtual device > Cannot attach audio driver > JackServer::Open failed with -1 > no message buffer overruns > Failed to open server > 10:35:53.707 JACK was stopped with exit status=255. > 10:35:54.785 Could not connect to JACK server as client. - Overall > operation failed. - Unable to connect to server. Please check the > messages window for more info. > Cannot connect to server socket err = No such file or directory > Cannot connect to server request channel > jack server is not running or cannot be started > > Those look familiar, they are exactly the same as on my desktop pc (that > works with wirewire) when I don't have the soundcard plugged in. > > Of course it might be that the pcmcia card is simply not working under > linux (I have no other os running, so I have to assume the card is in > fact working under, say, windows). > > But what actions could I take to either get it working or convince > myself that it's simply not supported? > > NB: I run crunchbang linux (=debian wheezy), which were installed prior > to getting the card. The laptop is a Lenovo x220. > > NB2: I couldn't see anything in dmesg after inserting the card and lspci > is the same with the card attached and detached... > > Any pointers appreciated! > i can "atte"st ;) to this card's performance with GNU/Linux, JACK, Ardour, etc.: http://www.startech.com/Cards-Adapters/FireWire/2-Port-ExpressCard-1394a-FireWire-Laptop-Adapter-Card~EC13942A2 only problem is the previously mentioned problem with hot-plugging (i.e. where the card must be in the laptop at boot up). otherwise works great. peace, w -------------- next part -------------- An HTML attachment was scrubbed... URL: From brouits at free.fr Thu Jun 26 00:52:41 2014 From: brouits at free.fr (=?ISO-8859-1?Q?Beno=EEt_Rouits?=) Date: Thu, 26 Jun 2014 02:52:41 +0200 Subject: [LAU] Music made with fun thanx to linux In-Reply-To: References: Message-ID: <53AB6ED9.1040602@free.fr> Le 20/06/2014 12:45, Carlos sanchiavedraz a ?crit : > 2014-05-16 18:10 GMT+02:00 Set Hallstr?m : >> Hi! >> >> Recently i traded my earfull portastudio against memories of happy >> unafortable OSystems some years ago, and then i made this: >> https://soundcloud.com/sakrecoer/sets/music-made-with-linux >> >> I may hope you like it like i do, but i know life is full of inexpectations >> :) >> >> With wishes of happy week endings, >> >> -- >> Set Hallstr?m >> AKA >> reSet Sakrecoer >> http://sakrecoer.com > > What a session, Almost 2 hours! Not my main kind of music but It > sounds good, it develops and progress well. And the overall EQ and > balance is also good. > > Thanks for sharing. > Yes, thanks for sharing, it's a very nice progression. right now, it is almost 3AM in Paris (FR), and i enjoy your set so that i cannot go and sleep now, until the end ! - Ben From jeremy at autostatic.com Thu Jun 26 07:11:19 2014 From: jeremy at autostatic.com (Jeremy Jongepier) Date: Thu, 26 Jun 2014 09:11:19 +0200 Subject: [LAU] howto get pcmcia firewire card working In-Reply-To: <1403739467.16698.11.camel@localhost.localdomain> References: <53A93B08.8090905@youmail.dk> <1403739467.16698.11.camel@localhost.localdomain> Message-ID: <53ABC797.60803@autostatic.com> On 06/26/2014 01:37 AM, Wayne DePrince Jr. wrote: > http://www.startech.com/Cards-Adapters/FireWire/2-Port-ExpressCard-1394a-FireWire-Laptop-Adapter-Card~EC13942A2 > > only problem is the previously mentioned problem with hot-plugging (i.e. > where the card must be in the laptop at boot up). otherwise works > great. > > peace, w http://subversion.ffado.org/wiki/HostControllers#VIA So does it also work at 88.2kHz and higher for you? Jeremy -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 836 bytes Desc: OpenPGP digital signature URL: From sakrecoer at gmail.com Thu Jun 26 14:02:39 2014 From: sakrecoer at gmail.com (=?UTF-8?Q?Set_Hallstr=C3=B6m?=) Date: Thu, 26 Jun 2014 16:02:39 +0200 Subject: [LAU] Music made with fun thanx to linux In-Reply-To: <53AB6ED9.1040602@free.fr> References: <53AB6ED9.1040602@free.fr> Message-ID: On Thu, Jun 26, 2014 at 2:52 AM, Beno?t Rouits wrote: > > > Yes, thanks for sharing, it's a very nice progression. right now, it is > almost 3AM in Paris (FR), and i enjoy your set so that i cannot go and > sleep now, until the end ! > - Ben > > Merci for this delicious feedback! I hope it was worth staying up and that it didn't end up being too painfull in the morning!!!! :) You see, i belong to those who claim that "the good sound" is the sound that triggers the best memories. :D I have some more and wget:able here if anyone is interested http://sakrecoer.com/assets/mu/files/ Yours, happy and thankfull, -- Set Hallstr?m AKA reSet Sakrecoer http://sakrecoer.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From carlo.ratm at gmail.com Thu Jun 26 14:08:11 2014 From: carlo.ratm at gmail.com (Carlo Ascani) Date: Thu, 26 Jun 2014 16:08:11 +0200 Subject: [LAU] Music made with fun thanx to linux In-Reply-To: <53AB6ED9.1040602@free.fr> References: <53AB6ED9.1040602@free.fr> Message-ID: 2014-06-26 2:52 GMT+02:00 Beno?t Rouits : > Le 20/06/2014 12:45, Carlos sanchiavedraz a ?crit : >> 2014-05-16 18:10 GMT+02:00 Set Hallstr?m : >>> Hi! >>> >>> Recently i traded my earfull portastudio against memories of happy >>> unafortable OSystems some years ago, and then i made this: >>> https://soundcloud.com/sakrecoer/sets/music-made-with-linux >>> >>> I may hope you like it like i do, but i know life is full of inexpectations >>> :) >>> >>> With wishes of happy week endings, >>> >>> -- >>> Set Hallstr?m >>> AKA >>> reSet Sakrecoer >>> http://sakrecoer.com >> >> What a session, Almost 2 hours! Not my main kind of music but It >> sounds good, it develops and progress well. And the overall EQ and >> balance is also good. >> >> Thanks for sharing. >> > I quote everything! May I ask what piece of software did you use to make the tracks ? A tracker, maybe ? Thank you for sharing -- Carlo Ascani | carlorat.me skype: carloratm irc: carloratm at freenode From sakrecoer at gmail.com Thu Jun 26 15:18:04 2014 From: sakrecoer at gmail.com (=?UTF-8?Q?Set_Hallstr=C3=B6m?=) Date: Thu, 26 Jun 2014 17:18:04 +0200 Subject: [LAU] Music made with fun thanx to linux In-Reply-To: References: <53AB6ED9.1040602@free.fr> Message-ID: Hi Carlos! On Thu, Jun 26, 2014 at 4:08 PM, Carlo Ascani wrote: > > I quote everything! > > May I ask what piece of software did you use to make the tracks ? > A tracker, maybe ? > > Thank you for sharing > > > Hey! thanks for getting back! Maybe my messages didn't go thru? I wrote this in an earlier response: > I mainly use hardware synths and sequencers, which makes the EQ'ing and arranging a bit easier, > because i tune the sound to my taste before recording and arrange on the go, or while layering > takes. Hence, my computer process is mainly deletion of errors and fine-mixing when i figure its > really needed. I record all the audio with ardour thru a teratec phase88 firewire rack: a pure delight > to work with! So, no no software for sequencing (well... akai software of the mpc500 and some octatrack) hence the title of my thread: *Music made _with fun_ thanks to GNU/Linux* ;) I use ardour as a mixer and its sequencer to delete eventual errors that occured during the recording session. Pretty oldschool to be frank, but my workflow was build on physical recorders. I am a big fan of Lee scratch Perry style. Maybe some of you notcied that i really love the calf plugins suit. Especialy the compressors and spaciotemporal FXs (delay reverb). I usualy go one take for the beat, one take with "instrumental" adlibs, one take for vocals, and some takes for backing vocals. In that order. But tracks like "Ner I Malm?" or "Par Hazard" are pure oneshots vocals included, that i ran thru a compressor before uploading. Some tracks i ran thru JAMin also. But i've got a lot of work to do on my mastering skills. I still tend to lose focus on what needs what when i fiddle with finetuning. Please feel comfortable with asking anything about what how when and whatnots :) Yours, -- Set Hallstr?m AKA reSet Sakrecoer http://sakrecoer.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From carlo.ratm at gmail.com Thu Jun 26 16:18:43 2014 From: carlo.ratm at gmail.com (Carlo Ascani) Date: Thu, 26 Jun 2014 18:18:43 +0200 Subject: [LAU] Music made with fun thanx to linux In-Reply-To: References: <53AB6ED9.1040602@free.fr> Message-ID: 2014-06-26 17:18 GMT+02:00 Set Hallstr?m : > Hi Carlos! > Carlo, please. :) > > Hey! thanks for getting back! > > Maybe my messages didn't go thru? > > I wrote this in an earlier response: > >> I mainly use hardware synths and sequencers, which makes the EQ'ing and >> arranging a bit easier, >> because i tune the sound to my taste before recording and arrange on the >> go, or while layering >> takes. Hence, my computer process is mainly deletion of errors and >> fine-mixing when i figure its >> really needed. I record all the audio with ardour thru a teratec phase88 >> firewire rack: a pure delight >> to work with! > > So, no no software for sequencing (well... akai software of the mpc500 and > some octatrack) hence the title of my thread: > *Music made _with fun_ thanks to GNU/Linux* > ;) > > I use ardour as a mixer and its sequencer to delete eventual errors that > occured during the recording session. Pretty oldschool to be frank, but my > workflow was build on physical recorders. I am a big fan of Lee scratch > Perry style. Oh it was on spam here, sorry. Very nice, I do love hardware things! > > Maybe some of you notcied that i really love the calf plugins suit. > Especialy the compressors and spaciotemporal FXs (delay reverb). I usualy go > one take for the beat, one take with "instrumental" adlibs, one take for > vocals, and some takes for backing vocals. In that order. But tracks like > "Ner I Malm?" or "Par Hazard" are pure oneshots vocals included, that i ran > thru a compressor before uploading. Some tracks i ran thru JAMin also. But > i've got a lot of work to do on my mastering skills. I still tend to lose > focus on what needs what when i fiddle with finetuning. > > Please feel comfortable with asking anything about what how when and > whatnots :) > > Yours, > > -- > Set Hallstr?m > AKA > reSet Sakrecoer > http://sakrecoer.com > > > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > -- Carlo Ascani | carlorat.me skype: carloratm irc: carloratm at freenode From sakrecoer at gmail.com Thu Jun 26 16:46:04 2014 From: sakrecoer at gmail.com (=?UTF-8?Q?Set_Hallstr=C3=B6m?=) Date: Thu, 26 Jun 2014 18:46:04 +0200 Subject: [LAU] Music made with fun thanx to linux In-Reply-To: References: <53AB6ED9.1040602@free.fr> Message-ID: On Thu, Jun 26, 2014 at 6:18 PM, Carlo Ascani wrote: > > > Carlo, please. :) > > Of course!! sorry Carlo :) Thanks again for getting thru!!! -------------- next part -------------- An HTML attachment was scrubbed... URL: From csanchezgs at gmail.com Thu Jun 26 17:16:19 2014 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Thu, 26 Jun 2014 19:16:19 +0200 Subject: [LAU] LAC'14 video archive In-Reply-To: <539A1509.50906@gareus.org> References: <539A1509.50906@gareus.org> Message-ID: 2014-06-12 23:00 GMT+02:00 Robin Gareus : > Hi all, > > The video recordings of the LAC'14 presentations have just been uploaded > to the conference website and are now directly linked from the archive: > http://lac.linuxaudio.org/2014/program > > > There are still a three videos missing and the workshop videos are also > yet to come. Currently they are also only available as vp8/vorbis/webm > (sorry IE and Safari users). But since it has been quite a while > already, we decided to not hold back the release of these already > finished videos any further. > > Once the collection is complete, we will provide a .torrent. Meanwhile, > for those who prefer to download the videos incrementally, they are > accessible via rsync://linuxaudio.org/ [1]. > > Many thanks for Frank and Moritz to get those done in really outstanding > quality this year. Kudos to the complete stream-team. > > enjoy, > robin - for the LAC'14 team > > > [1] example to get the 720p versions: > rsync -Pa --exclude "*360p.webm" \ > rsync://linuxaudio.org/lac2014/ \ > lac2014/ > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user I've just watched the Conference Welcome and the Keynote. My kudos to the organization and everyone involved, it's very nice to be able to watch conferences for us that couldn't afford to get there and enjoy them live. @J?rn, you've spotted many questions on your presentation (here in the list previuos to this LAC as well) that I've wonder as well over the years. Great intro for a LAC which encourages a healthy self-criticism to get better. Not long ago I was at a conference related to Multimedia/3D/Animation FLOSS (not only under Linux) hanging out with speakers and presenters having some coffee. At a certain moment they were talking about how good, ethic, philosophical and bla bla FLOSS was (sarcasm wink), and I modestly dare to say something in the lines of: Yes... of course it is but... we do already know and we're convinced; so maybe what we should also be doing (in addition to enjoy some coffee/beer making believe to believers) is to tell others out there and try to put this stuff close to them. You mentioned one important point that's to contact and attract enterprise and companies. This surely could help to create some momentum and get public knowledge. Related to this point, given that for the moment that high end aspect escapes me (but it's on my radar), I'm trying to effectively do something with several in progress projects aimed mainly to musicians and artist not very tech-savvy and not aware of other other ways of thinking that care and preserve their freedom and autonomy (and their wallets in some way), and are completely useful and functional in addition (maybe just a little less eye-candy sometimes). P.S: Just wanted to mention that I had to change to 360p because 720p stream cuts off. Just FWIW. Surely it's a temporary peak. -- C. sanchiavedraZ: * NEW / NUEVO: www.sanchiavedraZ.com * Musix GNU+Linux: www.musix.es From elmastero74 at gmail.com Thu Jun 26 21:43:00 2014 From: elmastero74 at gmail.com (Aaron L.) Date: Thu, 26 Jun 2014 14:43:00 -0700 Subject: [LAU] ardour file name plus "2000"??? Message-ID: Whenever I open an older ardour project, it keeps getting renamed with "2000" appended to it. This still happens with the newest release too. What do I need to do to stop this? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: From idragosani at gmail.com Thu Jun 26 21:50:50 2014 From: idragosani at gmail.com (Brett McCoy) Date: Thu, 26 Jun 2014 17:50:50 -0400 Subject: [LAU] ardour file name plus "2000"??? In-Reply-To: References: Message-ID: On Thu, Jun 26, 2014 at 5:43 PM, Aaron L. wrote: > Whenever I open an older ardour project, it keeps getting renamed with > "2000" appended to it. > > This still happens with the newest release too. > > What do I need to do to stop this? Ardour3 renames projects made in older versions (Ardour2 & MixBus) with 2000 appended so you can still open that project in the older versions. -- Brett W. McCoy -- http://www.brettwmccoy.com ------------------------------------------------------------------------ "In the rhythm of music a secret is hidden; If I were to divulge it, it would overturn the world." -- Jelaleddin Rumi From elmastero74 at gmail.com Thu Jun 26 21:56:20 2014 From: elmastero74 at gmail.com (Aaron L.) Date: Thu, 26 Jun 2014 14:56:20 -0700 Subject: [LAU] ardour file name plus "2000"??? In-Reply-To: References: Message-ID: Thanks, Brett. But it seems to me that it's appending a "2000" every time I open the project. On Thu, Jun 26, 2014 at 2:50 PM, Brett McCoy wrote: > On Thu, Jun 26, 2014 at 5:43 PM, Aaron L. wrote: > > Whenever I open an older ardour project, it keeps getting renamed with > > "2000" appended to it. > > > > This still happens with the newest release too. > > > > What do I need to do to stop this? > > Ardour3 renames projects made in older versions (Ardour2 & MixBus) > with 2000 appended so you can still open that project in the older > versions. > > > -- > Brett W. McCoy -- http://www.brettwmccoy.com > ------------------------------------------------------------------------ > "In the rhythm of music a secret is hidden; If I were to divulge it, > it would overturn the world." > -- Jelaleddin Rumi > -------------- next part -------------- An HTML attachment was scrubbed... URL: From cecchilorenzo at gmail.com Thu Jun 26 22:05:51 2014 From: cecchilorenzo at gmail.com (Lorenzo Cecchi) Date: Fri, 27 Jun 2014 00:05:51 +0200 Subject: [LAU] ardour file name plus "2000"??? In-Reply-To: References: Message-ID: That's because you keep opening the "-2000" file with A3. The ".ardour" file is now for the A3 session, the new "-2000" file is to open with the previous Ardour versions. 2014-06-26 23:56 GMT+02:00 Aaron L. : > Thanks, Brett. > > But it seems to me that it's appending a "2000" every time I open the > project. > > > On Thu, Jun 26, 2014 at 2:50 PM, Brett McCoy wrote: > >> On Thu, Jun 26, 2014 at 5:43 PM, Aaron L. wrote: >> > Whenever I open an older ardour project, it keeps getting renamed with >> > "2000" appended to it. >> > >> > This still happens with the newest release too. >> > >> > What do I need to do to stop this? >> >> Ardour3 renames projects made in older versions (Ardour2 & MixBus) >> with 2000 appended so you can still open that project in the older >> versions. >> >> >> -- >> Brett W. McCoy -- http://www.brettwmccoy.com >> ------------------------------------------------------------------------ >> "In the rhythm of music a secret is hidden; If I were to divulge it, >> it would overturn the world." >> -- Jelaleddin Rumi >> > > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > > -- Lorenzo Cecchi cecchilorenzo at gmail.com +44 07796341045 30 Gilbert Rd, bs5 9dp, Bristol -------------- next part -------------- An HTML attachment was scrubbed... URL: From idragosani at gmail.com Thu Jun 26 23:51:48 2014 From: idragosani at gmail.com (Brett McCoy) Date: Thu, 26 Jun 2014 19:51:48 -0400 Subject: [LAU] ardour file name plus "2000"??? In-Reply-To: References: Message-ID: On Thu, Jun 26, 2014 at 5:56 PM, Aaron L. wrote: > Thanks, Brett. > > But it seems to me that it's appending a "2000" every time I open the > project. You need to open the version the new version (the one without 2000), which can only be opened in A3 now. -- Brett W. McCoy -- http://www.brettwmccoy.com ------------------------------------------------------------------------ "In the rhythm of music a secret is hidden; If I were to divulge it, it would overturn the world." -- Jelaleddin Rumi From louigi.verona at gmail.com Fri Jun 27 11:08:52 2014 From: louigi.verona at gmail.com (Louigi Verona) Date: Fri, 27 Jun 2014 15:08:52 +0400 Subject: [LAU] Writing jingles Message-ID: I recently started writing jingles, because I need them for my podcast and videos that our skeptic group produces. I did not want to use somebody else's when I have the ability to write my own. I found jingles to be a very complex and interesting material to work on. So far I've written only one jingle that I am completely happy with and I am presenting it to you here. It is a jingle that we use for our videos, when the title of the video is shown and then the video proceeds to the lecturer. www.louigiverona.ru/files/TV_jingle_1.flac Has anyone else had this experience? -- Louigi Verona http://www.louigiverona.ru/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From sakrecoer at gmail.com Fri Jun 27 14:30:12 2014 From: sakrecoer at gmail.com (=?UTF-8?Q?Set_Hallstr=C3=B6m?=) Date: Fri, 27 Jun 2014 16:30:12 +0200 Subject: [LAU] LAC'14 video archive In-Reply-To: References: <539A1509.50906@gareus.org> Message-ID: On Thu, Jun 26, 2014 at 7:16 PM, Carlos sanchiavedraz wrote: > 2014-06-12 23:00 GMT+02:00 Robin Gareus : > > Hi all, > > > > The video recordings of the LAC'14 presentations have just been uploaded > > to the conference website and are now directly linked from the archive: > > http://lac.linuxaudio.org/2014/program > > I've just watched the Conference Welcome and the Keynote. My kudos to > the organization and everyone involved, it's very nice to be able to > watch conferences for us that couldn't afford to get there and enjoy > them live. > Thanks for bumping this thread Carlos! The keynote by J?rg made me think quite a lot. It's a risky thing to open conferences by questioning the conference itself. But it was well put and constructive. I'm a still pretty much a n00b here, although i have lurked the web-archive of LAU since 2009. But i think i would like to participate in the 2015 session after watching some videos. I would like to feed back my impressions after watching J?rg speak: I work quite hard on a peer-2-peer AFK level to promote GNU/linux and opensource, but also the philosophy arround and within it. Mainly by making my music with it, and teaching people to use blender, gimp and inkscape. I have noticed that it is better to let the people come to me, then to put my reality in their face. For example when they complain about having to buy expensive upgrades for their adobe products, i show them my results and only then i speak about my tools. But last year i tried something on a higher and more political level. I first try finding out who in what department is responsible for buying software in the swedish government (where i am now) Once i found who to talk with, i ask this simple question: "What is the total anual cost of all apple and windows products bought by the state?" At first i got pingponged all over the place. But after a few month i got an answer: Only microsoft cost aprox 500 Million SEK (aprox. 50 million euros). But i could tell by the email which contained may answer and the past log of many questions and comments that the employess had asked each other, that i layed a little seed in their minds. Unfortunately, i don't posses the human ressources or experience to push it further than that simple question. But unless you already have asked this question to your country's governement, i suggest you do it. The more people asking, the better probably. Now for a note about the ecnomical questions raised by J?rg arround LAC: Perhaps some of you guys are familiar with the frenchspeaking meetings called RMLL ( http://rmll.info ) I worked with them on the 2012 edition. And whilst it was held in switzerland where funding for culture and knowledge is easier to arrange than most place, i would like to suggest you ask them how they finance their meetings. Because i think pretty much all active participants (conferenciers, workshopers etc) got their ticket and a hot meal per day, with free to very cheap accomodation. They can be found on IRC freenode. But if anyone would like a direct link to the administration, please get in touch with me and i will arrange it as far as i can. I hope i'm not repeating anything that has been said 20'000 times before with this email. I would however like to end it with this little anecdote: i like to talk to random people. Hence, when ever i'm in a bar, i tigh alot of temporary to permanent connections. Lately, a few of them have advized me on their own behalf to try out linux "because its really easy now, and opensource/transparency is important if society has to become computerized". I think this is perhaps an insignifcant detail of luck and circumstances, but also a great sign! Before 2012, i don't think it ever happened to me. :) Happy Audio Creation! -- Set Hallstr?m AKA reSet Sakrecoer http://sakrecoer.com On Thu, Jun 26, 2014 at 7:16 PM, Carlos sanchiavedraz wrote: > 2014-06-12 23:00 GMT+02:00 Robin Gareus : > > Hi all, > > > > The video recordings of the LAC'14 presentations have just been uploaded > > to the conference website and are now directly linked from the archive: > > http://lac.linuxaudio.org/2014/program > > > > > > There are still a three videos missing and the workshop videos are also > > yet to come. Currently they are also only available as vp8/vorbis/webm > > (sorry IE and Safari users). But since it has been quite a while > > already, we decided to not hold back the release of these already > > finished videos any further. > > > > Once the collection is complete, we will provide a .torrent. Meanwhile, > > for those who prefer to download the videos incrementally, they are > > accessible via rsync://linuxaudio.org/ [1]. > > > > Many thanks for Frank and Moritz to get those done in really outstanding > > quality this year. Kudos to the complete stream-team. > > > > enjoy, > > robin - for the LAC'14 team > > > > > > [1] example to get the 720p versions: > > rsync -Pa --exclude "*360p.webm" \ > > rsync://linuxaudio.org/lac2014/ \ > > lac2014/ > > _______________________________________________ > > Linux-audio-user mailing list > > Linux-audio-user at lists.linuxaudio.org > > http://lists.linuxaudio.org/listinfo/linux-audio-user > > I've just watched the Conference Welcome and the Keynote. My kudos to > the organization and everyone involved, it's very nice to be able to > watch conferences for us that couldn't afford to get there and enjoy > them live. > > @J?rn, you've spotted many questions on your presentation (here in the > list previuos to this LAC as well) that I've wonder as well over the > years. Great intro for a LAC which encourages a healthy self-criticism > to get better. > > Not long ago I was at a conference related to Multimedia/3D/Animation > FLOSS (not only under Linux) hanging out with speakers and presenters > having some coffee. At a certain moment they were talking about how > good, ethic, philosophical and bla bla FLOSS was (sarcasm wink), and I > modestly dare to say something in the lines of: Yes... of course it is > but... we do already know and we're convinced; so maybe what we should > also be doing (in addition to enjoy some coffee/beer making believe to > believers) is to tell others out there and try to put this stuff close > to them. > You mentioned one important point that's to contact and attract > enterprise and companies. This surely could help to create some > momentum and get public knowledge. Related to this point, given that > for the moment that high end aspect escapes me (but it's on my radar), > I'm trying to effectively do something with several in progress > projects aimed mainly to musicians and artist not very tech-savvy and > not aware of other other ways of thinking that care and preserve their > freedom and autonomy (and their wallets in some way), and are > completely useful and functional in addition (maybe just a little less > eye-candy sometimes). > > P.S: Just wanted to mention that I had to change to 360p because 720p > stream cuts off. Just FWIW. Surely it's a temporary peak. > > -- > > C. sanchiavedraZ: > * NEW / NUEVO: www.sanchiavedraZ.com > * Musix GNU+Linux: www.musix.es > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > -- Set Hallstr?m AKA reSet Sakrecoer http://sakrecoer.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From waynedpj at in-giro.org Fri Jun 27 15:12:13 2014 From: waynedpj at in-giro.org (Wayne DePrince Jr.) Date: Fri, 27 Jun 2014 11:12:13 -0400 Subject: [LAU] howto get pcmcia firewire card working In-Reply-To: <53ABC797.60803@autostatic.com> References: <53A93B08.8090905@youmail.dk> <1403739467.16698.11.camel@localhost.localdomain> <53ABC797.60803@autostatic.com> Message-ID: <1403881933.16698.159.camel@localhost.localdomain> On gio, 2014-06-26 at 09:11 +0200, Jeremy Jongepier wrote: > On 06/26/2014 01:37 AM, Wayne DePrince Jr. wrote: > > http://www.startech.com/Cards-Adapters/FireWire/2-Port-ExpressCard-1394a-FireWire-Laptop-Adapter-Card~EC13942A2 > > > > only problem is the previously mentioned problem with hot-plugging (i.e. > > where the card must be in the laptop at boot up). otherwise works > > great. > > > > peace, w > > http://subversion.ffado.org/wiki/HostControllers#VIA > > So does it also work at 88.2kHz and higher for you? > > Jeremy > though i mainly work at 48 kHz/24 bit, it works fine at 96/24 as well with my Editorl FA-66. however, it appears my controller is not VIA but TI: 04:00.0 PCI bridge: Texas Instruments XIO2000(A)/XIO2200A PCI Express-to-PCI Bridge (rev 03) 05:00.0 FireWire (IEEE 1394): Texas Instruments XIO2200A IEEE-1394a-2000 Controller (PHY/Link) (rev 01) while we are on the topic, does anyone know of a non-Apple laptop (current or old) that has a powered Firewire 400 6 conductor port? though the StarTech ExpressCard solution works well and provides 2x 6 conductor ports, the physical ExpressCard connection is sensitive to vibration/movement and is easily dislodged. also, since it does not provide power via the FireWire ports, i need to also carry around the FA-66's power brick. peace, w -------------- next part -------------- An HTML attachment was scrubbed... URL: From list at nilsgey.de Fri Jun 27 15:20:05 2014 From: list at nilsgey.de (Nils) Date: Fri, 27 Jun 2014 17:20:05 +0200 Subject: [LAU] Writing jingles In-Reply-To: References: Message-ID: <53AD8BA5.9010607@nilsgey.de> On 27.06.2014 13:08, Louigi Verona wrote: > It is a jingle that we use for our videos. Can you show the video? From elmastero74 at gmail.com Fri Jun 27 16:44:20 2014 From: elmastero74 at gmail.com (Aaron L.) Date: Fri, 27 Jun 2014 09:44:20 -0700 Subject: [LAU] ardour mixer info missing in reference? Message-ID: http://manual.ardour.org/ardours-interface/introducing-the-mixer-window/ There's nothing there. Am I missing something? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: From paul at linuxaudiosystems.com Fri Jun 27 16:51:03 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Fri, 27 Jun 2014 12:51:03 -0400 Subject: [LAU] ardour mixer info missing in reference? In-Reply-To: References: Message-ID: On Fri, Jun 27, 2014 at 12:44 PM, Aaron L. wrote: > http://manual.ardour.org/ardours-interface/introducing-the-mixer-window/ > > There's nothing there. > > Am I missing something? > You could be missing a few weekends as you decide to write the missing pages. The Ardour manual is (and probably always will be) incomplete. It is user-editable: git clone git://git.ardour.org/ardour/manual.git Send me patches! -------------- next part -------------- An HTML attachment was scrubbed... URL: From gnome at hawaii.rr.com Fri Jun 27 17:11:04 2014 From: gnome at hawaii.rr.com (david) Date: Fri, 27 Jun 2014 07:11:04 -1000 Subject: [LAU] howto get pcmcia firewire card working In-Reply-To: <1403881933.16698.159.camel@localhost.localdomain> References: <53A93B08.8090905@youmail.dk> <1403739467.16698.11.camel@localhost.localdomain> <53ABC797.60803@autostatic.com> <1403881933.16698.159.camel@localhost.localdomain> Message-ID: <53ADA5A8.7070902@hawaii.rr.com> On 06/27/2014 05:12 AM, Wayne DePrince Jr. wrote: > On gio, 2014-06-26 at 09:11 +0200, Jeremy Jongepier wrote: >> On 06/26/2014 01:37 AM, Wayne DePrince Jr. wrote: >> >http://www.startech.com/Cards-Adapters/FireWire/2-Port-ExpressCard-1394a-FireWire-Laptop-Adapter-Card~EC13942A2 >> > >> > only problem is the previously mentioned problem with hot-plugging (i.e. >> > where the card must be in the laptop at boot up). otherwise works >> > great. >> > >> > peace, w >> >> http://subversion.ffado.org/wiki/HostControllers#VIA >> >> So does it also work at 88.2kHz and higher for you? >> >> Jeremy >> > though i mainly work at 48 kHz/24 bit, it works fine at 96/24 as > well with my Editorl FA-66. however, it appears my controller is not > VIA but TI: > > 04:00.0 PCI bridge: Texas Instruments XIO2000(A)/XIO2200A PCI Express-to-PCI Bridge (rev 03) > 05:00.0 FireWire (IEEE 1394): Texas Instruments XIO2200A IEEE-1394a-2000 Controller (PHY/Link) (rev 01) Hmmm, my understanding is that the TI firewire chips were THE best chips to use with Linux? > while we are on the topic, does anyone know of a non-Apple laptop > (current or old) that has a /powered/ Firewire 400 6 conductor port? > though the StarTech ExpressCard solution works well and provides 2x 6 > conductor ports, the physical ExpressCard connection is sensitive to > vibration/movement and is easily dislodged. also, since it does not > provide power via the FireWire ports, i need to also carry around the > FA-66's power brick. Don't know about that. My ~$1000 Linux laptop from System76 doesn't even have an ExpressCard port: it's all USB2 and 3 only. -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From elmastero74 at gmail.com Fri Jun 27 17:11:41 2014 From: elmastero74 at gmail.com (Aaron L.) Date: Fri, 27 Jun 2014 10:11:41 -0700 Subject: [LAU] ardour mixer info missing in reference? In-Reply-To: References: Message-ID: I was afraid you'd say that! ;-) Thanks for the response! On Fri, Jun 27, 2014 at 9:51 AM, Paul Davis wrote: > > > > On Fri, Jun 27, 2014 at 12:44 PM, Aaron L. wrote: > >> http://manual.ardour.org/ardours-interface/introducing-the-mixer-window/ >> >> There's nothing there. >> >> Am I missing something? >> > > You could be missing a few weekends as you decide to write the missing > pages. > > The Ardour manual is (and probably always will be) incomplete. It is > user-editable: > > git clone git://git.ardour.org/ardour/manual.git > > Send me patches! > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From gnome at hawaii.rr.com Fri Jun 27 17:23:24 2014 From: gnome at hawaii.rr.com (david) Date: Fri, 27 Jun 2014 07:23:24 -1000 Subject: [LAU] ardour mixer info missing in reference? In-Reply-To: References: Message-ID: <53ADA88C.2040909@hawaii.rr.com> That's a common problem with F/OSS; shortage of documentation. To my existing plans for retirement in 10 years or so, I might turn my old tech writer hand that direction sometime. On 06/27/2014 07:11 AM, Aaron L. wrote: > I was afraid you'd say that! > > ;-) > > Thanks for the response! > > > On Fri, Jun 27, 2014 at 9:51 AM, Paul Davis wrote: > > > On Fri, Jun 27, 2014 at 12:44 PM, Aaron L. wrote: > > http://manual.ardour.org/ardours-interface/introducing-the-mixer-window/ > > There's nothing there. > > Am I missing something? > > > You could be missing a few weekends as you decide to write the > missing pages. > > The Ardour manual is (and probably always will be) incomplete. It is > user-editable: > > git clone git://git.ardour.org/ardour/manual.git > > Send me patches! -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From waynedpj at in-giro.org Fri Jun 27 20:28:34 2014 From: waynedpj at in-giro.org (Wayne DePrince Jr.) Date: Fri, 27 Jun 2014 16:28:34 -0400 Subject: [LAU] howto get pcmcia firewire card working In-Reply-To: <53ADA5A8.7070902@hawaii.rr.com> References: <53A93B08.8090905@youmail.dk> <1403739467.16698.11.camel@localhost.localdomain> <53ABC797.60803@autostatic.com> <1403881933.16698.159.camel@localhost.localdomain> <53ADA5A8.7070902@hawaii.rr.com> Message-ID: <1403900914.16698.165.camel@localhost.localdomain> On ven, 2014-06-27 at 07:11 -1000, david wrote: > On 06/27/2014 05:12 AM, Wayne DePrince Jr. wrote: > > On gio, 2014-06-26 at 09:11 +0200, Jeremy Jongepier wrote: > >> On 06/26/2014 01:37 AM, Wayne DePrince Jr. wrote: > >> >http://www.startech.com/Cards-Adapters/FireWire/2-Port-ExpressCard-1394a-FireWire-Laptop-Adapter-Card~EC13942A2 > >> > > >> > only problem is the previously mentioned problem with hot-plugging (i.e. > >> > where the card must be in the laptop at boot up). otherwise works > >> > great. > >> > > >> > peace, w > >> > >> http://subversion.ffado.org/wiki/HostControllers#VIA > >> > >> So does it also work at 88.2kHz and higher for you? > >> > >> Jeremy > >> > > though i mainly work at 48 kHz/24 bit, it works fine at 96/24 as > > well with my Editorl FA-66. however, it appears my controller is not > > VIA but TI: > > > > 04:00.0 PCI bridge: Texas Instruments XIO2000(A)/XIO2200A PCI Express-to-PCI Bridge (rev 03) > > 05:00.0 FireWire (IEEE 1394): Texas Instruments XIO2200A IEEE-1394a-2000 Controller (PHY/Link) (rev 01) > > Hmmm, my understanding is that the TI firewire chips were THE best chips > to use with Linux? i am not sure about it being the best FW chipset, but i can vouch that it works well with no problems (aside from the hotplug issue already mentioned). > > > while we are on the topic, does anyone know of a non-Apple laptop > > (current or old) that has a /powered/ Firewire 400 6 conductor port? > > though the StarTech ExpressCard solution works well and provides 2x 6 > > conductor ports, the physical ExpressCard connection is sensitive to > > vibration/movement and is easily dislodged. also, since it does not > > provide power via the FireWire ports, i need to also carry around the > > FA-66's power brick. > > Don't know about that. My ~$1000 Linux laptop from System76 doesn't even > have an ExpressCard port: it's all USB2 and 3 only. > -------------- next part -------------- An HTML attachment was scrubbed... URL: From waynedpj at in-giro.org Fri Jun 27 20:30:10 2014 From: waynedpj at in-giro.org (Wayne DePrince Jr.) Date: Fri, 27 Jun 2014 16:30:10 -0400 Subject: [LAU] howto get pcmcia firewire card working In-Reply-To: <53ADA5A8.7070902@hawaii.rr.com> References: <53A93B08.8090905@youmail.dk> <1403739467.16698.11.camel@localhost.localdomain> <53ABC797.60803@autostatic.com> <1403881933.16698.159.camel@localhost.localdomain> <53ADA5A8.7070902@hawaii.rr.com> Message-ID: <1403901010.16698.166.camel@localhost.localdomain> On ven, 2014-06-27 at 07:11 -1000, david wrote: > On 06/27/2014 05:12 AM, Wayne DePrince Jr. wrote: > > On gio, 2014-06-26 at 09:11 +0200, Jeremy Jongepier wrote: > >> On 06/26/2014 01:37 AM, Wayne DePrince Jr. wrote: > >> >http://www.startech.com/Cards-Adapters/FireWire/2-Port-ExpressCard-1394a-FireWire-Laptop-Adapter-Card~EC13942A2 > >> > > >> > only problem is the previously mentioned problem with hot-plugging (i.e. > >> > where the card must be in the laptop at boot up). otherwise works > >> > great. > >> > > >> > peace, w > >> > >> http://subversion.ffado.org/wiki/HostControllers#VIA > >> > >> So does it also work at 88.2kHz and higher for you? > >> > >> Jeremy > >> > > though i mainly work at 48 kHz/24 bit, it works fine at 96/24 as > > well with my Editorl FA-66. however, it appears my controller is not > > VIA but TI: > > > > 04:00.0 PCI bridge: Texas Instruments XIO2000(A)/XIO2200A PCI Express-to-PCI Bridge (rev 03) > > 05:00.0 FireWire (IEEE 1394): Texas Instruments XIO2200A IEEE-1394a-2000 Controller (PHY/Link) (rev 01) > > Hmmm, my understanding is that the TI firewire chips were THE best chips > to use with Linux? > > > while we are on the topic, does anyone know of a non-Apple laptop > > (current or old) that has a /powered/ Firewire 400 6 conductor port? > > though the StarTech ExpressCard solution works well and provides 2x 6 > > conductor ports, the physical ExpressCard connection is sensitive to > > vibration/movement and is easily dislodged. also, since it does not > > provide power via the FireWire ports, i need to also carry around the > > FA-66's power brick. > > Don't know about that. My ~$1000 Linux laptop from System76 doesn't even > have an ExpressCard port: it's all USB2 and 3 only. > thanks for the info. i guess everything is going USB these days. but can USB 3 perform as well as FW? -------------- next part -------------- An HTML attachment was scrubbed... URL: From len at ovenwerks.net Fri Jun 27 23:36:55 2014 From: len at ovenwerks.net (Len Ovens) Date: Fri, 27 Jun 2014 16:36:55 -0700 (PDT) Subject: [LAU] howto get pcmcia firewire card working In-Reply-To: <1403901010.16698.166.camel@localhost.localdomain> References: <53A93B08.8090905@youmail.dk> <1403739467.16698.11.camel@localhost.localdomain> <53ABC797.60803@autostatic.com> <1403881933.16698.159.camel@localhost.localdomain> <53ADA5A8.7070902@hawaii.rr.com> <1403901010.16698.166.camel@localhost.localdomain> Message-ID: On Fri, 27 Jun 2014, Wayne DePrince Jr. wrote: > ??? thanks for the info.? i guess everything is going USB these days.? but > can USB 3 perform as well as FW? Unless you know different, The last I heard there are no USB3 Audio interfaces. Many people feel there is nothing to be gained over USB2 for audio. _If_ you can find a usb port that has it's own irq and that is not used (via an internal usb hub) for any other bits (webcam, wifi card, touchpad, etc.) then very good performance can be had. Up to about 18 channels plus midi at 48k/24bit... fewer at 96 (because normally 8 of the ports are adat). There are some better FW units (64 channels?) with the right FW IF. For a reality check on this. The only computers where USB is the only options are laptops and smaller computers and 18 i/os already require a BOB bigger than the computer. For something that requires more channels than 8 it is not that much more trouble to cart around a desktop box... or rackmount unit. Then other options exist anyway. A desktop MB _can_ be cooled better than a laptop and quieter too. (though many desktops are actually louder) -- Len Ovens www.ovenwerks.net From gnome at hawaii.rr.com Sat Jun 28 05:07:36 2014 From: gnome at hawaii.rr.com (david) Date: Fri, 27 Jun 2014 19:07:36 -1000 Subject: [LAU] howto get pcmcia firewire card working In-Reply-To: References: <53A93B08.8090905@youmail.dk> <1403739467.16698.11.camel@localhost.localdomain> <53ABC797.60803@autostatic.com> <1403881933.16698.159.camel@localhost.localdomain> <53ADA5A8.7070902@hawaii.rr.com> <1403901010.16698.166.camel@localhost.localdomain> Message-ID: <53AE4D98.5000801@hawaii.rr.com> On 06/27/2014 01:36 PM, Len Ovens wrote: > On Fri, 27 Jun 2014, Wayne DePrince Jr. wrote: > >> thanks for the info. i guess everything is going USB these days. >> but >> can USB 3 perform as well as FW? > > Unless you know different, The last I heard there are no USB3 Audio > interfaces. Many people feel there is nothing to be gained over USB2 for > audio. > > _If_ you can find a usb port that has it's own irq and that is not used > (via an internal usb hub) for any other bits (webcam, wifi card, > touchpad, etc.) then very good performance can be had. Up to about 18 > channels plus midi at 48k/24bit... fewer at 96 (because normally 8 of > the ports are adat). There are some better FW units (64 channels?) with > the right FW IF. > > For a reality check on this. The only computers where USB is the only > options are laptops and smaller computers and 18 i/os already require a > BOB bigger than the computer. For something that requires more channels > than 8 it is not that much more trouble to cart around a desktop box... > or rackmount unit. Then other options exist anyway. A desktop MB _can_ > be cooled better than a laptop and quieter too. (though many desktops > are actually louder) Just built a replacement for my dying desktop system. Used a Zen fanless PS. Even with 4 fans running inside it, the desktop is quieter than the laptop cooler my laptop sits on. My problem on desktops is that they're all just PCIe only. So now cannot use the 2 Audiophile 24/96s I have. :( I suppose I should pick up a PCIe Firewire card and talk the church into loaning me their unused Presonus ... -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From gnome at hawaii.rr.com Sat Jun 28 05:22:21 2014 From: gnome at hawaii.rr.com (david) Date: Fri, 27 Jun 2014 19:22:21 -1000 Subject: [LAU] howto get pcmcia firewire card working In-Reply-To: <1403900914.16698.165.camel@localhost.localdomain> References: <53A93B08.8090905@youmail.dk> <1403739467.16698.11.camel@localhost.localdomain> <53ABC797.60803@autostatic.com> <1403881933.16698.159.camel@localhost.localdomain> <53ADA5A8.7070902@hawaii.rr.com> <1403900914.16698.165.camel@localhost.localdomain> Message-ID: <53AE510D.10703@hawaii.rr.com> On 06/27/2014 10:28 AM, Wayne DePrince Jr. wrote: > On ven, 2014-06-27 at 07:11 -1000, david wrote: >> On 06/27/2014 05:12 AM, Wayne DePrince Jr. wrote: >> > On gio, 2014-06-26 at 09:11 +0200, Jeremy Jongepier wrote: >> >> On 06/26/2014 01:37 AM, Wayne DePrince Jr. wrote: >> >> >http://www.startech.com/Cards-Adapters/FireWire/2-Port-ExpressCard-1394a-FireWire-Laptop-Adapter-Card~EC13942A2 >> >> > >> >> > only problem is the previously mentioned problem with hot-plugging (i.e. >> >> > where the card must be in the laptop at boot up). otherwise works >> >> > great. >> >> > >> >> > peace, w >> >> >> >>http://subversion.ffado.org/wiki/HostControllers#VIA >> >> >> >> So does it also work at 88.2kHz and higher for you? >> >> >> >> Jeremy >> >> >> > though i mainly work at 48 kHz/24 bit, it works fine at 96/24 as >> > well with my Editorl FA-66. however, it appears my controller is not >> > VIA but TI: >> > >> > 04:00.0 PCI bridge: Texas Instruments XIO2000(A)/XIO2200A PCI Express-to-PCI Bridge (rev 03) >> > 05:00.0 FireWire (IEEE 1394): Texas Instruments XIO2200A IEEE-1394a-2000 Controller (PHY/Link) (rev 01) >> >> Hmmm, my understanding is that the TI firewire chips were THE best chips >> to use with Linux? > > i am not sure about it being the best FW chipset, but i can vouch > that it works well with no problems (aside from the hotplug issue > already mentioned). I meant "best" in the sense of they worked reliably and were well supported by Linux. My old Toshiba laptop had a Firewire (FW400) port on it, with a TI chipset, but I was never able to get to work with the FW800 device I had to test with. I think problem may have been the adaptor I had to use. But that laptop finally died and vanished into recycling heaven. Sometimes I think a compact case with a microATX mobo and a PCIe FW card wouldn't be much bigger than some laptops. Bring a small LCD display and keyboard along with it, I suppose. -- David W. Jones gnome at hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com From waynedpj at in-giro.org Sat Jun 28 11:09:29 2014 From: waynedpj at in-giro.org (Wayne DePrince Jr.) Date: Sat, 28 Jun 2014 07:09:29 -0400 Subject: [LAU] howto get pcmcia firewire card working In-Reply-To: <53AE510D.10703@hawaii.rr.com> References: <53A93B08.8090905@youmail.dk> <1403739467.16698.11.camel@localhost.localdomain> <53ABC797.60803@autostatic.com> <1403881933.16698.159.camel@localhost.localdomain> <53ADA5A8.7070902@hawaii.rr.com> <1403900914.16698.165.camel@localhost.localdomain> <53AE510D.10703@hawaii.rr.com> Message-ID: <1403953769.16698.182.camel@localhost.localdomain> On ven, 2014-06-27 at 19:22 -1000, david wrote: > On 06/27/2014 10:28 AM, Wayne DePrince Jr. wrote: > > On ven, 2014-06-27 at 07:11 -1000, david wrote: > >> On 06/27/2014 05:12 AM, Wayne DePrince Jr. wrote: > >> > On gio, 2014-06-26 at 09:11 +0200, Jeremy Jongepier wrote: > >> >> On 06/26/2014 01:37 AM, Wayne DePrince Jr. wrote: > >> >> >http://www.startech.com/Cards-Adapters/FireWire/2-Port-ExpressCard-1394a-FireWire-Laptop-Adapter-Card~EC13942A2 > >> >> > > >> >> > only problem is the previously mentioned problem with hot-plugging (i.e. > >> >> > where the card must be in the laptop at boot up). otherwise works > >> >> > great. > >> >> > > >> >> > peace, w > >> >> > >> >>http://subversion.ffado.org/wiki/HostControllers#VIA > >> >> > >> >> So does it also work at 88.2kHz and higher for you? > >> >> > >> >> Jeremy > >> >> > >> > though i mainly work at 48 kHz/24 bit, it works fine at 96/24 as > >> > well with my Editorl FA-66. however, it appears my controller is not > >> > VIA but TI: > >> > > >> > 04:00.0 PCI bridge: Texas Instruments XIO2000(A)/XIO2200A PCI Express-to-PCI Bridge (rev 03) > >> > 05:00.0 FireWire (IEEE 1394): Texas Instruments XIO2200A IEEE-1394a-2000 Controller (PHY/Link) (rev 01) > >> > >> Hmmm, my understanding is that the TI firewire chips were THE best chips > >> to use with Linux? > > > > i am not sure about it being the best FW chipset, but i can vouch > > that it works well with no problems (aside from the hotplug issue > > already mentioned). > > I meant "best" in the sense of they worked reliably and were well > supported by Linux. > > My old Toshiba laptop had a Firewire (FW400) port on it, with a TI > chipset, but I was never able to get to work with the FW800 device I had > to test with. I think problem may have been the adaptor I had to use. > But that laptop finally died and vanished into recycling heaven. > > Sometimes I think a compact case with a microATX mobo and a PCIe FW card > wouldn't be much bigger than some laptops. Bring a small LCD display and > keyboard along with it, I suppose. > agreed, and probably cooler thus quieter than a laptop. i have an older 2009 12" Darter from System76 and while the thing is still running great, the fans woosh up for every little thing, which makes recording more challenging. that being said, simply carrying a small laptop/netbook and FW interface is much nicer. or i guess for lots of recording only situations a portable recorder like the Zoom H4n/H6 would be the simplest. thanks for all the info. -------------- next part -------------- An HTML attachment was scrubbed... URL: From waynedpj at in-giro.org Sat Jun 28 11:11:20 2014 From: waynedpj at in-giro.org (Wayne DePrince Jr.) Date: Sat, 28 Jun 2014 07:11:20 -0400 Subject: [LAU] howto get pcmcia firewire card working In-Reply-To: References: <53A93B08.8090905@youmail.dk> <1403739467.16698.11.camel@localhost.localdomain> <53ABC797.60803@autostatic.com> <1403881933.16698.159.camel@localhost.localdomain> <53ADA5A8.7070902@hawaii.rr.com> <1403901010.16698.166.camel@localhost.localdomain> Message-ID: <1403953880.16698.184.camel@localhost.localdomain> On ven, 2014-06-27 at 16:36 -0700, Len Ovens wrote: > On Fri, 27 Jun 2014, Wayne DePrince Jr. wrote: > > > thanks for the info. i guess everything is going USB these days. but > > can USB 3 perform as well as FW? > > Unless you know different, The last I heard there are no USB3 Audio > interfaces. Many people feel there is nothing to be gained over USB2 for > audio. > > _If_ you can find a usb port that has it's own irq and that is not used > (via an internal usb hub) for any other bits (webcam, wifi card, touchpad, > etc.) then very good performance can be had. Up to about 18 channels plus > midi at 48k/24bit... fewer at 96 (because normally 8 of the ports are > adat). There are some better FW units (64 channels?) with the right FW IF. > > For a reality check on this. The only computers where USB is the only > options are laptops and smaller computers and 18 i/os already require a > BOB bigger than the computer. For something that requires more channels > than 8 it is not that much more trouble to cart around a desktop box... or > rackmount unit. Then other options exist anyway. A desktop MB _can_ be > cooled better than a laptop and quieter too. (though many desktops are > actually louder) > thanks. good to know that USB 2 could replace my FW ExpressCard+FW interface, as that seems the only way to get a bus-powered interface on a non-Apple laptop these days. -------------- next part -------------- An HTML attachment was scrubbed... URL: From louigi.verona at gmail.com Sat Jun 28 13:12:59 2014 From: louigi.verona at gmail.com (Louigi Verona) Date: Sat, 28 Jun 2014 17:12:59 +0400 Subject: [LAU] Writing jingles In-Reply-To: <53AD8BA5.9010607@nilsgey.de> References: <53AD8BA5.9010607@nilsgey.de> Message-ID: Sure, but it will be out in a week. I can post here. Also - it will be in Russian. Right now also doing the graphics with OpenShot. I did not know it could do quite a lot of stuff. It has various plugins that allow you to do rotation and changing parameters of the clip, like blur something and make it go from blur to clear, moving things and such. Not bad, although crashes a lot if you have lots of things loaded into it, so you have to save your work often. -- Louigi Verona http://www.louigiverona.ru/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From list at nilsgey.de Sat Jun 28 13:19:36 2014 From: list at nilsgey.de (Nils) Date: Sat, 28 Jun 2014 15:19:36 +0200 Subject: [LAU] Writing jingles In-Reply-To: References: <53AD8BA5.9010607@nilsgey.de> Message-ID: <53AEC0E8.7070103@nilsgey.de> On 28.06.2014 15:12, Louigi Verona wrote: > Sure, but it will be out in a week. I can post here. Also - it will be > in Russian. That would be nice in any case. But I meant just the part where the jingle can be heard. Nils From louigi.verona at gmail.com Sat Jun 28 13:29:33 2014 From: louigi.verona at gmail.com (Louigi Verona) Date: Sat, 28 Jun 2014 17:29:33 +0400 Subject: [LAU] Writing jingles In-Reply-To: <53AEC0E8.7070103@nilsgey.de> References: <53AD8BA5.9010607@nilsgey.de> <53AEC0E8.7070103@nilsgey.de> Message-ID: Actually, I just finished making the intro sequence, so you can take a look here: http://youtu.be/OFE4fe_d4bw On Sat, Jun 28, 2014 at 5:19 PM, Nils wrote: > > > > On 28.06.2014 15:12, Louigi Verona wrote: > >> Sure, but it will be out in a week. I can post here. Also - it will be in >> Russian. >> > That would be nice in any case. But I meant just the part where the jingle > can be heard. > > Nils > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > -- Louigi Verona http://www.louigiverona.ru/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From len at ovenwerks.net Sat Jun 28 15:15:24 2014 From: len at ovenwerks.net (Len Ovens) Date: Sat, 28 Jun 2014 08:15:24 -0700 (PDT) Subject: [LAU] howto get pcmcia firewire card working In-Reply-To: <53AE4D98.5000801@hawaii.rr.com> References: <53A93B08.8090905@youmail.dk> <1403739467.16698.11.camel@localhost.localdomain> <53ABC797.60803@autostatic.com> <1403881933.16698.159.camel@localhost.localdomain> <53ADA5A8.7070902@hawaii.rr.com> <1403901010.16698.166.camel@localhost.localdomain> <53AE4D98.5000801@hawaii.rr.com> Message-ID: On Fri, 27 Jun 2014, david wrote: > Just built a replacement for my dying desktop system. Used a Zen fanless PS. > Even with 4 fans running inside it, the desktop is quieter than the laptop > cooler my laptop sits on. My experience was not the same... but my laptop is atom powered. I used as much of what I already had in my upgrade as I could. What I was spending already felt extravagant at the time. But I could still change out the PS without too much hurt and I may. At least there is a door between the computer and any mic. > My problem on desktops is that they're all just PCIe only. So now cannot use > the 2 Audiophile 24/96s I have. :( That is not true at all. Please feel free to send your cards here :) (well one of them) One of the things I made sure on my new MB was to have PCI slots. I have three PCI slots and use two, one for a d66 and I have an old pciaudio that I use for the MIDI port. There were actually quite a selection of MB with 3 pci slots and a lot more with 1 and 2. If I wanted to set up a xeon powered board. Then pci slots are harder to find. However the cost of board, cpu(s), case, ps and memory would make the cost of a new Audio Science PCIe card seem ok. In other words they are a different playing field. In my range the i5 CPUs are the best for audio. Only buy an i7 if you also do a lot of video or games, but for low latency audio you would want to turn off it's i7-ness and use it like an i5 anyway. > I suppose I should pick up a PCIe Firewire card and talk the church into > loaning me their unused Presonus ... You are in luck then. Any unused stuff at our church is unused for good reason... and some of the stuff we do use makes me shudder. I am just now on the hunt for a new mixing board (mixing desk is beyond my means) to replace the http://www.long-mcquade.com/64/Pro_Audio_Recording/Mixers/Yorkville_Sound/Micromix_800-Watt_Stereo_10_Channel_Powered_Mixer.htm rental that doesn't work worth anything. I am looking at the mackie, soundcraft and allen & heath stuff (16 to 24 ch) but we will probably be using the power amp from the above for now. -- Len Ovens www.ovenwerks.net From csanchezgs at gmail.com Sun Jun 29 14:37:05 2014 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Sun, 29 Jun 2014 16:37:05 +0200 Subject: [LAU] Writing jingles In-Reply-To: References: <53AD8BA5.9010607@nilsgey.de> <53AEC0E8.7070103@nilsgey.de> Message-ID: 2014-06-28 15:29 GMT+02:00 Louigi Verona : > Actually, I just finished making the intro sequence, so you can take a look > here: > http://youtu.be/OFE4fe_d4bw > > I really like it, Louigi, and the synth bass sounds deep and full. Very appropriate for an intro. And the video it seems interesting. What is the vidcast/show is about? > On Sat, Jun 28, 2014 at 5:19 PM, Nils wrote: >> >> >> >> >> On 28.06.2014 15:12, Louigi Verona wrote: >>> >>> Sure, but it will be out in a week. I can post here. Also - it will be in >>> Russian. >> >> That would be nice in any case. But I meant just the part where the jingle >> can be heard. >> >> Nils >> >> _______________________________________________ >> Linux-audio-user mailing list >> Linux-audio-user at lists.linuxaudio.org >> http://lists.linuxaudio.org/listinfo/linux-audio-user > > > > > -- > Louigi Verona > http://www.louigiverona.ru/ > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > -- C. sanchiavedraZ: * NEW / NUEVO: www.sanchiavedraZ.com * Musix GNU+Linux: www.musix.es From louigi.verona at gmail.com Sun Jun 29 22:07:32 2014 From: louigi.verona at gmail.com (Louigi Verona) Date: Mon, 30 Jun 2014 02:07:32 +0400 Subject: [LAU] Writing jingles In-Reply-To: References: <53AD8BA5.9010607@nilsgey.de> <53AEC0E8.7070103@nilsgey.de> Message-ID: Hey Carlos! The video is about critical thinking and science. The video is ready, you can take a look at how the jingle looks with the rest of the video. Although OpenShot was killing me with its constant crashes. It is such a pity Linux does not have stable video editors (yeah, I tried kdenlive too, but at some point it stopped exporting audio for some reason). http://www.youtube.com/watch?v=ZMcPcxzN70M L.V. -- Louigi Verona http://www.louigiverona.ru/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From len at ovenwerks.net Sun Jun 29 23:43:13 2014 From: len at ovenwerks.net (Len Ovens) Date: Sun, 29 Jun 2014 16:43:13 -0700 (PDT) Subject: [LAU] Live mixers/recording Message-ID: Some of you may remember a while ago (year and a half or so) I was asking about mixers for live use. I finally have some money to spend, so I went back over the thread. I also looked at some of the lower end digital boards like the A&H QU and the Soundcraft Si series. I found it interesting that the mackie has firewire but the other two have chosen USB2.0 even for a channel count as high as 32 i/o. The manual says "standard compliant" and will work with the Apple no extra drivers (nothing works with windows without drivers). Has anyone tried one of these with alsa? Considering the cost of a high count audio interface, the cost of this as an audio IF plus controler starts to look not too bad. It is obvious that the main cost is hardware, pots, switches and connectors. The digital boards don't try to put in as much hardware, yet seem to be able to give more features for the same price range. It is easy to include a whole effects rack on almost a per channel basis only by changing cpu/dsp power. I guess in many ways that is what we are doing with a daw... time for the open console design? Actually we have much of it already, it is just the hardware parts we don't seem to have. I/O ports and control surfaces is all thats missing. -- Len Ovens www.ovenwerks.net From paul at linuxaudiosystems.com Sun Jun 29 23:54:55 2014 From: paul at linuxaudiosystems.com (Paul Davis) Date: Sun, 29 Jun 2014 19:54:55 -0400 Subject: [LAU] Live mixers/recording In-Reply-To: References: Message-ID: On Sun, Jun 29, 2014 at 7:43 PM, Len Ovens wrote: > Some of you may remember a while ago (year and a half or so) I was asking > about mixers for live use. I finally have some money to spend, so I went > back over the thread. > > I also looked at some of the lower end digital boards like the A&H QU and > the Soundcraft Si series. I found it interesting that the mackie has > firewire but the other two have chosen USB2.0 even for a channel count as > high as 32 i/o. The manual says "standard compliant" and will work with the > Apple no extra drivers (nothing works with windows without drivers). Has > anyone tried one of these with alsa? Considering the cost of a high count > audio interface, the cost of this as an audio IF plus controler starts to > look not too bad. > > It is obvious that the main cost is hardware, pots, switches and > connectors. The digital boards don't try to put in as much hardware, yet > seem to be able to give more features for the same price range. It is easy > to include a whole effects rack on almost a per channel basis only by > changing cpu/dsp power. > > I guess in many ways that is what we are doing with a daw... time for the > open console design? Actually we have much of it already, it is just the > hardware parts we don't seem to have. I/O ports and control surfaces is all > thats missing. > why not just use midibox or similar open source MIDI hardware to build a suitable controller? audio interfaces are well taken care of already ... -------------- next part -------------- An HTML attachment was scrubbed... URL: From gurusonic at gmail.com Mon Jun 30 01:20:05 2014 From: gurusonic at gmail.com (Roger) Date: Mon, 30 Jun 2014 11:20:05 +1000 Subject: [LAU] Live mixers/recording In-Reply-To: References: Message-ID: <53B0BB45.2090708@gmail.com> On 30/06/14 09:43, Len Ovens wrote: > Some of you may remember a while ago (year and a half or so) I was > asking about mixers for live use. I finally have some money to spend, > so I went back over the thread. > > I also looked at some of the lower end digital boards like the A&H QU > and the Soundcraft Si series. I found it interesting that the mackie > has firewire but the other two have chosen USB2.0 even for a channel > count as high as 32 i/o. The manual says "standard compliant" and will > work with the Apple no extra drivers (nothing works with windows > without drivers). Has anyone tried one of these with alsa? Considering > the cost of a high count audio interface, the cost of this as an audio > IF plus controler starts to look not too bad. > > It is obvious that the main cost is hardware, pots, switches and > connectors. The digital boards don't try to put in as much hardware, > yet seem to be able to give more features for the same price range. It > is easy to include a whole effects rack on almost a per channel basis > only by changing cpu/dsp power. > > I guess in many ways that is what we are doing with a daw... time for > the open console design? Actually we have much of it already, it is > just the hardware parts we don't seem to have. I/O ports and control > surfaces is all thats missing. > > -- > Len Ovens > www.ovenwerks.net > > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > Midas Pro1 (which runs on Linux) or Digico SD11 (which doesn't) would be my top choices for a small digital desk. Probably not in your budget though given the desks you mentioned. We run Digico SD10s and SD9 where I work. I have always had good experiences with Soundcraft, but haven't used their digital desks. Cheers Roger From len at ovenwerks.net Mon Jun 30 04:26:05 2014 From: len at ovenwerks.net (Len Ovens) Date: Sun, 29 Jun 2014 21:26:05 -0700 (PDT) Subject: [LAU] Live mixers/recording In-Reply-To: References: Message-ID: On Sun, 29 Jun 2014, Paul Davis wrote: > I guess in many ways that is what we are doing with a daw... > time for the open console design? Actually we have much of it > already, it is just the hardware parts we don't seem to have. > I/O ports and control surfaces is all thats missing. > > > why not just use midibox or similar open source MIDI hardware to build a > suitable controller? audio interfaces are well taken care of already ... Cost mostly. I guess one has to decide what open source is worth. digital mixer is 3k msr but 2k street it seems. 17 motor faders, 16inputs with mic pres, 16 outputs, remote controlable from ipad, or laptop. Built in port for digital snake. At least one touch screen. No fans for cooling. On board recording. So assuming I am buying the computer as well as the interfaces... It seems I would spend 1k2 just on the computer with fanless cooling and 1k on the 16 i/o interface. So the controler would be all extra cost. There are some interesting projects out there in the diy world but I still can't see getting much under $200 if not more... depending on scrounging. If I stick to having a fader control for every channel, I don't need motorizing. For my home studio (workstation is more accurate) a diy set of controls is actually a great idea. I already have everything else. 6/4 i/o is more than I need. The most inputs I have used at once is 2, but normally 1. For a live setup from scratch where I am not the primary operator, I will probably end up with an analog setup. Unless I can talk my local supplier into a good discount :) -- Len Ovens www.ovenwerks.net From len at ovenwerks.net Mon Jun 30 04:43:01 2014 From: len at ovenwerks.net (Len Ovens) Date: Sun, 29 Jun 2014 21:43:01 -0700 (PDT) Subject: [LAU] Live mixers/recording In-Reply-To: <53B0BB45.2090708@gmail.com> References: <53B0BB45.2090708@gmail.com> Message-ID: On Mon, 30 Jun 2014, Roger wrote: > Midas Pro1 (which runs on Linux) or Digico SD11 (which doesn't) would be my > top choices for a small digital desk. Probably not in your budget though > given the desks you mentioned. We run Digico SD10s and SD9 where I work. > I have always had good experiences with Soundcraft, but haven't used their > digital desks. Ya, a bit over. Not horible though. I expect either of the boards I am looking at will give my sound guy room to grow his talent (Not to indicate he is bad, I am no better)... as well as an instant improvement in sound. (it will replace a yorkville m810 powered mixer bought from rental stock) The reality is I will be able to add some new mics if I go analog (we seems to have lost another since last week - no output). Someone else is slowly replacing mic stands. -- Len Ovens www.ovenwerks.net From ralf.mardorf at rocketmail.com Mon Jun 30 05:23:02 2014 From: ralf.mardorf at rocketmail.com (Ralf Mardorf) Date: Mon, 30 Jun 2014 07:23:02 +0200 Subject: [LAU] Live mixers/recording In-Reply-To: <53B0BB45.2090708@gmail.com> References: <53B0BB45.2090708@gmail.com> Message-ID: <1404105782.22829.3.camel@archlinux> On Mon, 2014-06-30 at 11:20 +1000, Roger wrote: > I have always had good experiences with Soundcraft, but haven't used > their digital desks. I at least know one analog discrete circuit mixer that is very good. Unlikely that a digital desk will max out the advantages of good designed discrete circuits. IOW their digital mixers might be good or not, the known good quality of the analog discrete circuits don't say anything about the quality of a Soundcraft's digital desk. From csanchezgs at gmail.com Mon Jun 30 13:16:26 2014 From: csanchezgs at gmail.com (Carlos sanchiavedraz) Date: Mon, 30 Jun 2014 15:16:26 +0200 Subject: [LAU] Writing jingles In-Reply-To: References: <53AD8BA5.9010607@nilsgey.de> <53AEC0E8.7070103@nilsgey.de> Message-ID: 2014-06-30 0:07 GMT+02:00 Louigi Verona : > Hey Carlos! > > The video is about critical thinking and science. > > The video is ready, you can take a look at how the jingle looks with the > rest of the video. Although OpenShot was killing me with its constant > crashes. It is such a pity Linux does not have stable video editors (yeah, I > tried kdenlive too, but at some point it stopped exporting audio for some > reason). > > http://www.youtube.com/watch?v=ZMcPcxzN70M > > > L.V. > > > -- > Louigi Verona > http://www.louigiverona.ru/ Great, I'll watch it. BTW, I'm helping a friend and talented multimedia artist that's made the change to open tools and Linux coming from closed and captive technologies. He's making really good progresses with Kdenlive and Blender, for the moment he's quite happy and surprised by the quality and versatility that now is available to him. Maybe I can ask any specific question if you want. -- C. sanchiavedraZ: * NEW / NUEVO: www.sanchiavedraZ.com * Musix GNU+Linux: www.musix.es From alexandre.prokoudine at gmail.com Mon Jun 30 13:29:16 2014 From: alexandre.prokoudine at gmail.com (Alexandre Prokoudine) Date: Mon, 30 Jun 2014 17:29:16 +0400 Subject: [LAU] Writing jingles In-Reply-To: References: <53AD8BA5.9010607@nilsgey.de> Message-ID: On Sat, Jun 28, 2014 at 5:12 PM, Louigi Verona wrote: > Sure, but it will be out in a week. I can post here. Also - it will be in > Russian. > > Right now also doing the graphics with OpenShot. I did not know it could do > quite a lot of stuff. It has various plugins that allow you to do rotation > and changing parameters of the clip, like blur something and make it go from > blur to clear, moving things and such. That sounds a bit like you are trying to apply every LADSPA/LV2 plugin out there to a single track :) Alex From rncbc at rncbc.org Mon Jun 30 16:45:07 2014 From: rncbc at rncbc.org (Rui Nuno Capela) Date: Mon, 30 Jun 2014 17:45:07 +0100 Subject: [LAU] [ANN] Vee One Suite 0.5.0 - Officially beta now! Message-ID: <53B19413.3090306@rncbc.org> Greetings, The Vee One Suite of old-school soft(ware)-instruments, primarily slated for Linux, featuring synthv1 [1], a polyphonic synthesizer, samplv1 [2], a polyphonic sampler and drumkv1 [3], a drum-kit sampler, are officially in beta phase now. Change-log for the new release: - LV2 UI Idle and Show interfaces support added. - First attempt to allow a headless stand-alone JACK client application run mode, without a GUI, with option given as command line argument (--no-gui). - A man page has beed added (re. stand-alone JACK client). - Reverse sample option and parameter knob added. (samplv1 and drumkv1 only). - Allow the build system to include an user specified LDFLAGS. As before, all made available in dual form: - a pure stand-alone JACK [4] client with JACK-session, NSM [5] (Non Session management) and both JACK MIDI and ALSA [6] MIDI input support; - a LV2 [7] instrument plug-in. The Vee One Suite are free and open-source Linux Audio software, distributed under the terms of the GNU General Public License (GPL) [8] version 2 or later. The fine print follows ;) * synthv1 - an old-school polyphonic synthesizer [1] * synthv1 0.5.0 released! synthv1 is an old-school all-digital 4-oscillator subtractive polyphonic synthesizer with stereo fx. LV2 URI: http://synthv1.sourceforge.net/lv2 website: http://synthv1.sourceforge.net downloads: http://sourceforge.net/projects/synthv1/files - source tarball: http://download.sourceforge.net/synthv1/synthv1-0.5.0.tar.gz - source package: http://download.sourceforge.net/synthv1synthv1-0.5.0-17.rncbc.suse131.src.rpm - binary packages: http://download.sourceforge.net/synthv1synthv1-0.5.0-17.rncbc.suse131.i586.rpm http://download.sourceforge.net/synthv1synthv1-0.5.0-17.rncbc.suse131.x86_84.rpm * samplv1 - an old-school polyphonic sampler [2] * samplv1 0.5.0 released! samplv1 is an old-school polyphonic sampler synthesizer with stereo fx. LV2 URI: http://samplv1.sourceforge.net/lv2 website: http://samplv1.sourceforge.net downloads: http://sourceforge.net/projects/samplv1/files - source tarball: http://download.sourceforge.net/samplv1/samplv1-0.5.0.tar.gz - source package: http://download.sourceforge.net/samplv1/samplv1-0.5.0-17.rncbc.suse131.src.rpm - binary packages: http://download.sourceforge.net/samplv1/samplv1-0.5.0-17.rncbc.suse131.i586.rpm http://download.sourceforge.net/samplv1/samplv1-0.5.0-17.rncbc.suse131.x86_84.rpm * drumkv1 - an old-school drum-kit sampler [3] * drumkv1 0.5.0 released! drumkv1 an old-school drum-kit sampler synthesizer with stereo fx. LV2 URI: http://drumkv1.sourceforge.net/lv2 website: http://drumkv1.sourceforge.net downloads: http://sourceforge.net/projects/drumkv1/files - source tarball: http://download.sourceforge.net/drumkv1/drumkv1-0.5.0.tar.gz - source package: http://download.sourceforge.net/drumkv1/drumkv1-0.5.0-13.rncbc.suse131.src.rpm - binary packages: http://download.sourceforge.net/drumkv1/drumkv1-0.5.0-13.rncbc.suse131.i586.rpm http://download.sourceforge.net/drumkv1/drumkv1-0.5.0-13.rncbc.suse131.x86_84.rpm References: [1] synthv1 - an old-school polyphonic synthesizer http://synthv1.sourceforge.net/ [2] samplv1 - an old-school polyphonic sampler http://samplv1.sourceforge.net/ [3] drumkv1 - an old-school drum-kit sampler http://drumkv1.sourceforge.net/ [4] JACK Audio Connection Kit http://jackaudio.org/ [5] NSM, Non Session Management http://non.tuxfamily.org/nsm/ [6] ALSA, Advanced Linux Sound Architecture http://www.alsa-project.org/ [7] LV2, Audio Plugin Standard, the extensible successor of LADSPA http://lv2plug.in/ [8] GNU General Public License http://www.gnu.org/copyleft/gpl.html See also: http://www.rncbc.org/drupal/node/793 Enjoy && ever again, have fun. -- rncbc aka Rui Nuno Capela rncbc at rncbc.org From alexandre.prokoudine at gmail.com Mon Jun 30 16:49:45 2014 From: alexandre.prokoudine at gmail.com (Alexandre Prokoudine) Date: Mon, 30 Jun 2014 20:49:45 +0400 Subject: [LAU] [ANN] Vee One Suite 0.5.0 - Officially beta now! In-Reply-To: <53B19413.3090306@rncbc.org> References: <53B19413.3090306@rncbc.org> Message-ID: 30 ???? 2014 ?. 20:45 ???????????? "Rui Nuno Capela" ???????: > Change-log for the new release: > - LV2 UI Idle and Show interfaces support added. Which in layman terms means...? :) Also, glad to see this project alive and kicking! Alex -------------- next part -------------- An HTML attachment was scrubbed... URL: From rncbc at rncbc.org Mon Jun 30 17:09:52 2014 From: rncbc at rncbc.org (Rui Nuno Capela) Date: Mon, 30 Jun 2014 18:09:52 +0100 Subject: [LAU] [ANN] Vee One Suite 0.5.0 - Officially beta now! In-Reply-To: References: <53B19413.3090306@rncbc.org> Message-ID: <53B199E0.8080705@rncbc.org> On 06/30/2014 05:49 PM, Alexandre Prokoudine wrote: > > 30 ???? 2014 ?. 20:45 ???????????? "Rui Nuno Capela" > ???????: > > > Change-log for the new release: > > - LV2 UI Idle and Show interfaces support added. > > Which in layman terms means...? :) > the layman shouldn't bother with this atm. sorry :) but if you bother, LV2 UI Idle+Show interfaces are part of the conspiracy to deprecate lv2_external_ui. oops. i said it :) > Also, glad to see this project alive and kicking! > thanks. btw. may i ask why if ever you thought about the project going awol.? cheers -- rncbc aka Rui Nuno Capela rncbc at rncbc.org From louigi.verona at gmail.com Mon Jun 30 20:57:22 2014 From: louigi.verona at gmail.com (Louigi Verona) Date: Tue, 1 Jul 2014 00:57:22 +0400 Subject: [LAU] Writing jingles In-Reply-To: References: <53AD8BA5.9010607@nilsgey.de> Message-ID: No, not really. I had that problem even with very simple projects. And this is very unfortunate, since Open Shot's possibilities are quite enough for what I want and it is only it's instability which tires me and gives me thoughts to stop this torment and do video with Windows. On Mon, Jun 30, 2014 at 5:29 PM, Alexandre Prokoudine < alexandre.prokoudine at gmail.com> wrote: > On Sat, Jun 28, 2014 at 5:12 PM, Louigi Verona wrote: > > Sure, but it will be out in a week. I can post here. Also - it will be in > > Russian. > > > > Right now also doing the graphics with OpenShot. I did not know it could > do > > quite a lot of stuff. It has various plugins that allow you to do > rotation > > and changing parameters of the clip, like blur something and make it go > from > > blur to clear, moving things and such. > > That sounds a bit like you are trying to apply every LADSPA/LV2 plugin > out there to a single track :) > > Alex > _______________________________________________ > Linux-audio-user mailing list > Linux-audio-user at lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-user > -- Louigi Verona http://www.louigiverona.ru/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From murks at tuxfamily.org Mon Jun 30 21:23:33 2014 From: murks at tuxfamily.org (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=) Date: Mon, 30 Jun 2014 23:23:33 +0200 Subject: [LAU] Writing jingles In-Reply-To: References: <53AD8BA5.9010607@nilsgey.de> Message-ID: <20140630232333.3402d15f@eeyore.mozart.uni-klu.ac.at> On Tue, 1 Jul 2014 00:57:22 +0400 Louigi Verona wrote: > No, not really. I had that problem even with very simple projects. > And this is very unfortunate, since Open Shot's possibilities are > quite enough for what I want and it is only it's instability which > tires me and gives me thoughts to stop this torment and do video with > Windows. I did this admittedly crappy little clip for a small game project using openshot using a single track and extensive effects. I didn't count the number of crashes, but there were many. https://freeshell.de/~murks/public/Tallest.ogg Regards, Philipp