Linux-audio-dev

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[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-----

21 years, 2 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

21 years, 2 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

21 years, 2 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

21 years, 2 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

21 years, 2 months

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

by Ivica Ico Bukvic

> On Sunday 18 April 2004 13:44, Tim Blechmann wrote: > > out, the windows hdsp driver sets the latency timer of > > the cardbus bridge and (i am repeating what i wrote before) thomas added > > this to the linux driver as well ... which didn't work for no kernel i > > no. there was a change setting the lateny timer to 255, but only for the > HDSP, not for the bridge. that's why i ask. > I see. Not sure about that. What I do know, however, is that the hexdump I was looking into was the cardbus registers, *not* the hdsp ones (see below). > > used on my machine (nearly every kernel between 2.4.22 and 2.6.5 both > > vanilla and ck/gentoo) > > there is a latency problem, but it's NOT related to the settings of the > > latency registers of the hdsp or the cardbus bridge... > > > > > ico, i'm know it works for you with 0x04 at 0xc9, but please try > > > anyway. > > as i pointed out earlier, i've got the same behaviour on my machine with > > the same register settings on windows (works fine) and on linux (doesn't > > work). > > what we should do is figuring out, what this specific register on ico's > > machine is actually doing... > > could you actually show us the hexdump of the config space? > This is all I have currently. Will generate more as soon as I can (please let me know which particular device you are looking for). http://meowing.ccm.uc.edu/~ico/eMachines/ (jpegs are verbosely titled) Ico

21 years, 2 months

[linux-audio-dev] [ANN] Ecamegapedal 0.4.4 released

by Kai Vehmanen

1. A short summary of changes Minor bugs in JACK support have been fixed. Now Ecamegapedal makes sure it won't launch the JACK daemon by accident when probing for available devices on startup. The manual pages have been updated with some new sections. --- 2. What is Ecamegapedal? Ecamegapedal is a real-time effect processor software with a graphical user interface for controlling the effect parameters. It is meant to be used as a virtual guitar-fx or studio effect box. In addition to real-time operation, Ecamegapedal also supports reading from and writing to audio files. All audio object and effect plugin types provided by the Ecasound libraries are supported. This includes ALSA, JACK, OSS, aRts, over 20 file formats, over 30 effect types, LADSPA plugins and multi-operator effect presets. Ecamegapedal's implementation is based on Ecasound and Qt libraries. Ecamegapedal is licensed under the GPL. --- 3. Contributors Patches Kai Vehmanen (various) --- 4. Links and files http://www.eca.cx/ecamegapedal http://ecasound.seul.org/download/ecamegapedal-0.4.4.tar.gz --- http://www.eca.cx Audio software for Linux!

21 years, 2 months

[linux-audio-dev] ANNOUNCE: Q 5.3, Q-Midi 1.14, Q-Synth

by Dr.Graef＠t-online.de

Q 5.3 has been released, along with Q-Midi 1.14. (Mostly bugfixes this time.) Q is an equational programming language based on term rewriting. Q-Midi is an add-on module for the Q language which provides an interface to MidiShare, Grame's cross-platform MIDI library. If you want to try out programming computer music applications in a high-level functional programming language, then these might be for you. You can find more information about Q and Q-Midi at http://q-lang.sourceforge.net/. Also available now in CVS is a first snapshot of the upcoming Q-Synth package which provides interfaces to OSC and SuperCollider3. See http://sourceforge.net/cvs/?group_id=96881 for checkout instructions (the CVS module name is q-synth), and for a link which allows you to browse the CVS repository. Enjoy! Albert Graef -- Dr. Albert Gr"af Email: Dr.Graef(a)t-online.de, ag(a)muwiinfa.geschichte.uni-mainz.de WWW: http://www.musikwissenschaft.uni-mainz.de/~ag

21 years, 2 months

[linux-audio-dev] Re: Audio over Ethernet

by John Lazzaro

On Apr 14, 2004, Anders Torger wrote: > Thus, I think it is necessary to implement something operating on the > ethernet level to get best performance in terms of throughput and > latency. > > /Anders Torger If what you mean by "operating at the ethernet level" means "no Cobra-like hardware to help, but putting data directly into Etherframes w/o IP/RTP headers", then its unclear to me that working at the RTP/IP level is going to hurt you much. The simplest implementation would have RTP/IP header overhead, but there are nice header compression schemes that get rid of it: http://www.ietf.org/rfc/rfc2508.txt and its improved versions. By using RTP, you get a lot of protocol design you might otherwise need to do, within RTP (like RTP MIDI) and surrounding it (session management, etc). One big thing you need to worry about are clocks -- unlike a protocol like AES/EBU or SPIDF, packet-based media is not sending an implicit clock along with the data. So, the nominal "sender sampling rate" can't be precisely linked to the nominal "receiver sampling rate" in a simple way. The consequence is either too much data piles up at the receiver, or not enough. One solution to this problem is to continuously running a sample-rate converter at the receiver in software, to keep the two sampling rates locked. See: http://www1.ietf.org/mail-archive/working-groups/avt/current/ msg00569.html and use the "Thead Next" to cycle through the discussion, it goes on a ways and lots of interesting folks drop in with info. A separate issue for your "many streams" case is synchronizing the streams to each other, in the case where not all share the same nominal clock. RTP has tools for this, based on associating NTP timestamps from a common clock to each independent stream, that get used for audio/video lipsync, and can be repurposed here as well. --- John Lazzaro http://www.cs.berkeley.edu/~lazzaro lazzaro [at] cs [dot] berkeley [dot] edu ---

21 years, 2 months

[linux-audio-dev] minor update to Linux soundapps pages

by Dave Phillips

Greetings: Some URL errors fixed, Speech section completely updated (thanks to Antti Kaihola), LilyPond entry massively rewritten, and a few more apps in the New Additions... http://www.linuxsound.at (Europe) http://linuxsound.jp (Japan) http://linux-sound.org (US) Best, dp

21 years, 2 months

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Linux-audio-dev