Linux-audio-dev April 2004

linux-audio-dev@lists.linuxaudio.org

99 participants
90 discussions

[linux-audio-dev] jack_fst: a JACK client to run VST's

by Paul Davis

Torben Hohn and I are pleased to announce the initial release of jack_fst, a small JACK client designed to run VST FX and VST/i's with connections to the rest of the JACK world, and, for VST/i's the ALSA sequencer. Tarball is available at: http://linuxaudiosystems.com/fst/jack_fst-1.2.tar.gz You will need the recently announced FST, a recent version of Wine, GTK+2 and JACK (from CVS at this time). Here is a list of VST plugins (actually) that are known to run successfully with jack_fst: Ambience.dll AnechoicRoomSimulator.dll BarsStripes.dll Big Tick Hexaline.dll BLOCKFISH.dll Boss DS-1.dll Boss SD-1.dll Classic EQ.dll Crystal.dll Cyanide2.dll DebaserDemo.dll DeeLay_sm.dll Delay Lama.dll deloizer095.dll DFX Transverb.dll dominion v1.2.dll Drumatic_22.dll endorphin.dll E-Phonic XPressor.dll FLOORFISH.dll Frohmage.dll GoldenGate.dll H2O.dll JS Vibrato V1.0.dll LoopaZoid.dll mabento.dll MadShifta.dll mda DX10.dll mda ePiano.dll mda Piano.dll MjMultibandCompressor.dll MjRotoDelay.dll ParisEQ.dll relofter.dll resolator_1.03_demo.dll RetroDelay.dll SIR.dll sloper.dll SoloString_v10.dll SPITFISH.dll Stretch & Squash.dll SupaPhaser.dll Syntar.dll Tape Delay.dll THD.dll themodulator2.dll Trancemitter.dll

20 years, 8 months

[linux-audio-dev] Re: linux-audio-dev Digest, Vol 7, Issue 42

by Kjetil Svalastog Matheussen

Paul Davis: > > >Can you elaborate in terms of which version of wine you have used > >successfully? (ie: wine = x.y.z or wine >= x.y.z?) > > i am using wine 20040309. i think torben has a slightly earlier > version than this which has worked for him to the same extent. > This is the version I'm using with the vstserver now as well. It seems to work better than the december 2003 version (finally!). (Vsterver won't compile with it yet though.) --

20 years, 8 months

[linux-audio-dev] Re: initial release of libfst - in-process VST support

by Kjetil Svalastog Matheussen

Paul Davis: > > Torben Hohn and I are pleased to release an initial version of libfst, > a small GPL'ed C library that provides support for using win32/x86 VST > plugins (FX and VST/i) within native Linux applications, with the > assistance of the Wine project's libwine. > > ========================================================== > We expect there will be several minor problems with this > initial release. Please help us fix them! > ========================================================== > > How is this different from VSTserver? > ------------------------------------- > > This work differs from Kjetil Mattheusen's VSTserver because the > plugin is loaded into the address space of the program using the > library, not into a server. As a result, it is a more appropriate > solution for audio applications that might support many VST plugins, > where the context switches required in the VSTserver case do not scale > well. The downside, of course, is that a misbehaving plugin will crash > the host application, which VSTserver avoids. > Another small point, I might add, is that the API against vstserver is far far simpler, at least for adding native plugin-gfx support. But regarding the performance and simplicity for the user, this solution is of course better. --

20 years, 8 months

[linux-audio-dev] Does ALSA offer built-in PLUG for JACK and non-interleaved cards?

by ico＠fuse.net

Hi all, 2 quick questions: 1) A while ago I heard that the JACK plug has been included into the ALSA's default array of plugs. Also today I found this on the Web (for the asoundrc): pcm.jackplug { type plug slave { pcm "jack" } } pcm.jack { type jack playback_ports { 0 alsa_pcm:playback_1 1 alsa_pcm:playback_2 } capture_ports { 0 alsa_pcm:capture_1 1 alsa_pcm:capture_2 } } Does this mean that now one can channel ALSA-only aware apps directly to JACK and if so, are there any penalties of doing it this way as oposed to using JACK-aware apps (i.e. Sample-sync?)? 2) Is there also a plug in ALSA that allows for non-interleaved cards (i.e. hdsp) to be directly talked to (i.e. using aplay) since by default aplay complains how the soundcard is not interleaved and hence it fails. Apologies if these questions have been answered before. I did a bit of looking through the archives but was unable to find anything of relevance. Many thanks for your help! Best wishes, Ico

20 years, 8 months

[linux-audio-dev] laconf2 call for help w/ irc and streaming

by Joern Nettingsmeier

hi everyone! as you will certainly have heard, we are getting closer to the 2nd linux audio conference at zkm (http://www.zkm.de/lad). as you might also have heard, we will have live streams of all talks and lectures in ogg format for those who can't make it there in person. last year, we had an irc "feedback channel", meaning i would sit in the lecture room with a notebook, answer questions from people who were listening to the stream and relay their questions to the audience if necessary. i want to do the same this year. problem is: there are three lecture rooms now. so i'm looking for a couple of people who'd be willing to help out. specifically, i need people who * are at the conference :) preferably the whole time * bring their own laptop with working sound output and headphones * have some experience with networking * have or can operate a webcam on their machine and are willing to install xawtv * like to chat your tasks will be: * to attend talks * to have an irc window open and answer and relay questions * to set up a webcam that will provide the listeners with stills so that they'll know which slide is up atm etc. * to monitor the outgoing stream and to ping the relays from time to time. the more helpers, the less work for all of us. last year, it has been great fun, although chatting, monitoring (with some 5 seconds lag) and listening at the same time can make your head spin in funny ways. please cc: me privately on any replies. thanks. regards, jörn -- Jörn Nettingsmeier Lortzingstr. 11, 45128 Essen, Germany http://spunk.dnsalias.org (my server) http://www.linuxaudiodev.org (Linux Audio Developers)

20 years, 8 months

[linux-audio-dev] MuSE 0.9 codename "COTURNIX" - out now with documented API!

by jaromil

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 re all, i forward you here the announce as somebody could be interested also in exploring the doxyfied MuSE API, which is getting pretty stable now. see http://muse.dyne.org/codedoc ciao! - ----- Forwarded message from jaromil <jaromil(a)dyne.org> ----- Date: Sat, 17 Apr 2004 20:09:46 +0200 From: jaromil <jaromil(a)dyne.org> To: muse(a)dyne.org annunciazio'! annunciazio'! dyne.org autoproduzioni & the FreakNet Medialab proudly present: __ __ ____ _____ ___ ___ | \/ |_ _/ ___|| ____| / _ \ / _ \ | |\/| | | | \___ \| _| | | | | (_) | | | | | |_| |___) | |___ | |_| |\__, | |_| |_|\__,_|____/|_____| \___(_) /_/ codename "COTURNIX" this is RASTA SOFTWARE, Jah Rastafari Livity bless OUR Freedom! :: the Multiple Streaming Engine :: free software for free radios! since 2001, getting better & better download the sourcecode from: http://muse.dyne.org http://muse.dyne.org http://muse.dyne.org http://muse.dyne.org http://muse.dyne.org This application is being developed in the hope to provide the Free Software community a user friendly tool for network audio streaming, making life easier for indypendent free speech radios wanting to stream via http on icecast servers. MuSE is a software for the mixing, encoding, and network streaming of sound: it can transmit an audio signal by mixing together sound taken from files or also network, recursively remixing more MuSE streams. MuSE can simultaneously mix up to 6 encoded audio bitstreams (from files or network, ogg, mp3, wav and other common sound formats), plus an input signal from microphone. MuSE offers an intuitive interface to be operated in realtime, while it can also run slick from the Unix commandline. _ _ _ ___ __ __ _| |_ __ _| |_( )___ _ _ _____ __ _|__ \ \ V V / ' \/ _` | _|/(_-< | ' \/ -_) V V / /_/ \_/\_/|_||_\__,_|\__| /__/ |_||_\___|\_/\_/ (_) There are a LOT of new things in this version of MuSE: it passed more than a year since the previous release and development never stopped, just kept going on silently. So now we have a lot of new features! Spotlights on the large rewrite of Engine parts, especially the Input and Encoder channels, the tightening of the FIFO Pipe mechanism, many speed improvements and full documentation of the API. There is a revamped GTK-2 interface now featuring drag&drop capability, Language translations and much better Profile and Playlist handling, a new libSnd (wav player) input channel during the hackmeeting, full support of Icecast2 streaming both with OGG and MP3 and yet more CLI flexibility. So as spring is coming is time to throw the peach and say COTURNIX! here are the ChangeLog entries: - - important engine optimization and speed improvements - - new GTK-2 graphical interface with improved usability (nightolo) - - new resampling for ogg/vorbis encoding (the secret rabbit code) - - new libsndfile input channel (pallotron) - - new xml profiles (nightolo) - - full range of login types for different servers - - revamped commandline interface with more switches - - some relevant bugfixes improving overall stability - - rewrite of decoder channels - - GTK-2 GUI localization - - updates and more info in the unix manual page - - doxygen documentation of the engine's API ***** Supported servers: You can use MuSE to stream both MP3 or Ogg/Vorbis sound format to a broadcast server, which means that for doing an online radio you still need to setup yours, or find one that let you stream. Such servers can be seen like antennas which amplify your signal and redistribute it to listeners. there are free software implementations of such technology! the ones supported by MuSE are: - - Icecast2 - http://icecast.org - can stream OGG & MP3 - - Litestream - http://litestream.org - can stream MP3 - - Darwin - http://developer.apple.com/darwin/projects/streaming - - Shoutcast - http://shoutcast.com - non free, runs on win32 ***** Supported players: MuSE streams via http, it doesn't uses multicast technology, nor RTP/RTSP, to have the widest possible range of compatible players. You can listen audio produced by MuSE from almost every personal computer and operating system, using one or more of the following: - - xmms - http://www.xmms.org - for GNU/Linux, *BSD, Sun, etc. - - mpg123 - http://www.mpg123.de - for various UNIX systems - - itunes - http://www.apple.com/itunes - on MacOSX - - zinf - http://www.zinf.org - on GNU/Linux and Win32 - - winamp - http://www.winamp.com - on all Win platforms and even more players: something should be also allready available for handeld devices and embedded systems as well. ***** Internals: MuSE is written in C++ and is a multithreaded application. It reads streams using the included libmpeg library (mp3 format) and it can optionally link the OggVorbis library to read ogg files. For encoding the sound into mp3 or ogg it can link either or both LAME and OggVorbis as shared libraries. MuSE also features a user interface using the GTK+ widget library and a console interface using ncurses. Resampling of any input is done with bicubic interpolation to 44khz stereo format, then channels get mixed together and encoded to the desired quality by the selected codec. Separate threads are running for each decoder, the mixer and the encoders, while the flow is synced thru FIFO pipes which implement mutex locking and avoid well race conditions. All the functionalities of MuSE are quite well exposed thru a reusable API, which in fact was the one used to build the GUIs on top. Documentation for it is available on http://muse.dyne.org/codedoc If you are interested, you are very welcome to build new MuSE interfaces, there are still a lot of unexplored possibilities and this engine can be a realiable backend for radio automation interfaces and more. Get in touch with developers! join the mailinglist on http://lists.dyne.org/muse or peek into irc.freenode.net channel #dyne ***** Libraries: - - LAME (optional) Lame can be installed but is no more needed, in particular libmp3lame and the header lame.h must be properly installed. - - OGG VORBIS (optional) You can compile and install libogg and libvorbis on your machine before compiling MuSE; the configure script will recognize them and include support for decoding and mixing of .ogg files. - - GTK+ (optional) if libgtk and all the related libraries are present, MuSE will compile the GTK+ graphic user interface for interactive use and additional fun. - - NCURSES (optional) if libncurses is present, MuSE will compile a text console interactive interface to be used into ASCII terminals. - - SNDFILE (optional) if libsndfile is present then you'll be able to play uncompressed sound files like wav, aiff, snd, voc, pvf, mat, au, sf etc. ***** Compile and install: - - you can get latest version of lame: (if you already have lame and libmp3lame jump to point 2, after checking that version is >3.89) with 'lame --version' or download it from www.mp3dev.org/mp3 and follow the simple instructions to get installed this wonderful GPL mp3 encoder. - - install libogg and libvorbis: http://www.xiph.org/ogg/vorbis/index.html if you don't install any of the above, MuSE will be a simple player. - - compile: cd MuSE-x.x.x ; ./configure (or try ./configure --help to have listed some compile options) ; make ; make install you can also make a debian package and debian aided compilation with: fakeroot debian/rules binary this command will build debian packages for you! to KNOW MORE go to the website on http://muse.dyne.org ***** Report bugs BUG REPORTING is REALLY APPRECIATED! BUT TAKE CARE TO DO IT WELL: please ALWAYS REPORT the muse --version you are running and possibly run the binary with gdb giving us the backtrace of the error. if you understand the above, please go to http://bugs.dyne.org and submit a BUG!!! ***** AUTHORS MuSE is copyleft (c) 2000-2004 by Denis "jaromil" Rojo MuSE's GTK+ GUI is copyleft (c) 2001-2004 by Antonino "nightolo" Radici MuSE's NCURSES GUI is copyleft (c) 2002-2004 by Luca "rubik" Profico MuSE's first GUI is copyleft (c) 2000-2001 by August Black included resampling algos are copyleft (c) 2002 by Matteo "MoP" Nastasi part of the redistributed code is copyright by the respective authors, please refer to the AUTHORS file and to the supplied sourcecode for further information and to COPYING for the full license. for MuSE's development, the author and mantainer has been supported by PUBLIC VOICE Lab ........ [ http://www.pvl.at ] SERVUS.AT ............... [ http://www.servus.at ] ***** PLEASE DONATE! This whole software was built by autonomous efforts and occasionally supported by non-profit organizations, while the development currently relies on the political believing of MuSE's authors, that there should be such a tool and it should be freely available to people willing to do online radio. If you can afford to donate us some money let us know, we also need new and old working hardware. The main coder is a nomad and has no fixed job, you can help him survive with a small donation via paypal.com to jaromil \@\ dyne.org this would be allready much if most users do! and you can be sure this way MuSE will keep on existing. The MuSE development team is also avaliable to run formation courses about usage, customization and reuse/abuse of free and open source software - it offers as well support and warranty, development of software solutions and consulting. There were a number of workshops allready done around Europe on free and opensource network radio streaming, by the hackers @ dyne.org if you want to organize one you're very welcome to contact us! this way you can also help us develop more free software. THANKS, a thousand flowers will blossom! == DISCLAIMER This source code is free software; you can redistribute it and/or modify it under the terms of the GNU Public License as published by the Free Software Foundation; either version 2 of the License, or (at your option) any later version. This source code is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. Please refer to the GNU Public License for more details. You should have received a copy of the GNU Public License along with this source code; if not, write to: Free Software Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. - ----- End forwarded message ----- - -- jaromil, dyne.org rasta coder, http://rastasoft.org I'd be happy to be kidnapped and join the fight against my own gov't - Jo van der Spek -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.4 (GNU/Linux) Comment: http://gnupg.org iD8DBQFAgmRVWLCC1ltubZcRAuidAJ9QVFBToY97CBGKTtEzMNpNtYrBiACcCWbT 5TroG5pPXf9LDvmbxfaDORk= =+xVY -----END PGP SIGNATURE-----

20 years, 8 months

[linux-audio-dev] Some more Muzak

by james＠dis-dot-dat.net

Jon, Andy, LAD people, http://www.dis-dot-dat.net/dasub.mp3 Playing about with a classic sampled beat - something I normally try to avoid. Still, I think I broke it enough that it doesn't sound repetetive. In case your interested, there is only one beat sample, I use cheesetracker's offset copmmand (Oxy) to pick out the bits I want and munge the beat, kind of like recycle, but all done by hand. You can probably guess, but the vocals are taken from Thundercats. The voice of Mumm-Ra and a snipet explaining the eye of Thundera (the one that now goes: THE EYE, The Eye, the eye...). All chopping and munging was done in cheesetracker, even the crunchy bass was made from the smooth one by over amplifying and then filtering out the nasties. Lots of ladspa effects helped, too, although the ones I most want to use (compressors, etc) are too slow to be used "inline" like that. Are there any VERY fast LADSPA compressors about? James

20 years, 8 months

[linux-audio-dev] RE: [linux-audio-user] snd-hdsp+cardbus+M6807 notebook=distortion -- FIXED!

by Ivica Ico Bukvic

It may be very well conceivable that the sources of the problem are different. Hence my fix should only work on the CB1410 cardbuses (or perhaps the fix is even more specific to the eMachines m680x notebooks). Ivica Ico Bukvic, composer & multimedia sculptor http://meowing.ccm.uc.edu/~ico/ > -----Original Message----- > From: Russell King [mailto:rmk@arm.linux.org.uk] On Behalf Of Russell King > Sent: Monday, April 19, 2004 4:15 AM > To: Ivica Ico Bukvic > Cc: daniel.ritz(a)gmx.ch; 'Tim Blechmann'; ccheney(a)debian.org; linux- > pcmcia(a)lists.infradead.org; 'Thomas Charbonnel'; linux-audio- > dev(a)music.columbia.edu > Subject: Re: [linux-audio-user] snd-hdsp+cardbus+M6807 notebook=distortion > -- FIXED! > > On Sun, Apr 18, 2004 at 11:12:59PM -0400, Ivica Ico Bukvic wrote: > > > ico: > > > would it be possible that you send me an lscpi -vv output before and > > > after you changed the registers? maybe we see any differences there > ... > > > and could you ask the manufacturer of your notebook, if he knows the > > > purpose of these registers? > > > > Last time I checked, he was trying to figure it out as well :-( > > > > I am also including the entire PDF (URL) on the CB1410. Perhaps that > will > > help (I got this from the notebook manufacturer). > > The point about Tim's problem is that he has a different Cardbus bridge, > so the documentation and fixes for your bridge won't work. > > -- > Russell King > Linux kernel 2.6 ARM Linux - http://www.arm.linux.org.uk/ > maintainer of: 2.6 PCMCIA - http://pcmcia.arm.linux.org.uk/ > 2.6 Serial core

20 years, 8 months

[linux-audio-dev] Can software authors make allowances for packagers?

by Luke Yelavich

Hi all I am busy packaging software for Slackware and trying to compile for a specific architecture. The one problem I am finding with several packages, is that the configure scripts obtain the architecture of the host system, and ignore the CFLAGS variable set that contains arguments for the compiler to compile for a specific architecture. I am wondering whether anybody has some work-work-arounds that might work on all packages, or could developers possibly provide a configure command-line argument to allow a packager to compile for a specific architecture, other than their own? For example I am running an i686 system, whereas I want to compile for an i586 based system. Thanks Luke -- Luke Yelavich http://www.audioslack.com luke(a)audioslack.com

20 years, 8 months

[linux-audio-dev] RE: [linux-audio-user] snd-hdsp+cardbus+M6807 notebook=distortion -- FIXED!

by Ivica Ico Bukvic

> ico: > would it be possible that you send me an lscpi -vv output before and > after you changed the registers? maybe we see any differences there ... > and could you ask the manufacturer of your notebook, if he knows the > purpose of these registers? Last time I checked, he was trying to figure it out as well :-( I am also including the entire PDF (URL) on the CB1410. Perhaps that will help (I got this from the notebook manufacturer). All the stuff can be found here: (including various states of the Hexdump in Windows as well as the lspci stuff) http://meowing.ccm.uc.edu/~ico/eMachines/ Hope this helps for the time being! Best wishes, Ico

20 years, 8 months

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Linux-audio-dev April 2004