Linux-audio-user

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20676 discussions

[linux-audio-user] now streaming: linux audio developers meeting at zkm in karlsruhe

by J�rn Nettingsmeier

hi everyone ! i'd like to invite you all to tune in to our live streaming program from the linux audio developers meeting at the zkm karlsruhe. please visit http://www.linuxdj.com/audio/lad/eventszkm2003.php3 for a list of streaming mirrors and additional material (slides). dust off your mp3 players, it's only 32 kbit/sec ! currently speaking is alsa core developer takashi iwai, further speakers will include paul davis and dave philips. best, joern J�rn Nettingsmeier Kurf�rstenstr 49, 45138 Essen, Germany http://spunk.dnsalias.org (my server) http://www.linuxdj.com/audio/lad/ (Linux Audio Developers)

22 years, 9 months

[linux-audio-user] GWC quits

by Reid Vail

I'm running GWC and have run into a problem with the app just aborting. Quits in the midst of the Pop/clicks removal routine. It's consistant. The hardware I'm running is an Athlon 850 with 256k RAM. The sound card is an SB16 ISA. I know it's not much, but it was free, back a few years ago. Now I wonder if the routines that GWC is performing are just too much for that little engine to execute, so the program aborts. Any advice would be appreciated. RV

22 years, 9 months

[linux-audio-user] Dirty, Nasty Raunchy Guitars

by Levi Burton

Greetings, Searched the list archives but found nothing appropriate, so I thought I would just ask. I am a metal guitar player. I like a heavily saturated guitar tone. Could someone point me in the right direction for processing a clean guitar signal in real-time such that it can (even partially - i like weird sounding stuff) emulate tube circuitry. I am thinking a chain of LADSPA plug-ins would do the job. The sound im looking for doesn't have to emulate anything to a T, it just has to be raunchy (think Nine Inch Nails - Broken, or any Nine Inch Nails album for that matter). Thanks! -- Levi Burton http://www.puresimplicity.net/~ldb/

22 years, 9 months

[linux-audio-user] vsound

by jdennett＠science.uva.nl

>> cygnus:~/mars> vsound --timing -f output.wav realplay >> _1790918_blur_beagle_vi.ram >> About to start the application. The output will not be available until the >> application exits. >> /usr/bin/vsound: line 163: 1366 Aborted >> LD_PRELOAD="$pkglibdir/libvsound.so" "$@" >I find I have to use vsound -t hi, vsound -t and vsound --timing are meant to be synonyms, and indeed both give me the same problem. has anyone else had this problem? i don't even know what it means. why does it abort on this line? cheers jane

22 years, 9 months

[linux-audio-user] simple apps

by jdennett＠science.uva.nl

Thanks for all the responses, especially the curses based vu meters... off to try them, that's exactly what i meant :) > >>Personally I use arecord and then adjust the volume level with a sound > >>editor later. > > > > Thats' all very well but when recording you don't want to (a) go over > > full scale or (b) have the incoming level so low that you lose resolution. > > > > Adjusting the level digitally afterwards won't help with either of these > > problems. > > You are completely correct but if you know your system well then this > should not be a problem. If it is then do a retake or ditch the parts > that don't work. right, well, exactly. that's what i've been doing and i've been getting very very bored with it. consequently now delighted with the meter suggestions :) thanks! jane

22 years, 9 months

[linux-audio-user] Problem with jackd

by Raphaël Haberer-Proust

Hello, I would like to use ardour. I downloaded it from CVS and finally got it compiled, but I can't start jackd: titanium:~# jackd -d alsa -d hw:0 jackd 0.50.0 Copyright 2001-2002 Paul Davis and others. jackd comes with ABSOLUTELY NO WARRANTY This is free software, and you are welcome to redistribute it under certain conditions; see the file COPYING for details loading driver .. creating alsa driver ... hw:0|1024|2|48000|swmon|rt ALSA lib pcm_hw.c:866:(snd_pcm_hw_open) open /dev/snd/pcmC0D0c failed: No such file or directory ALSA: Cannot open PCM device alsa_pcm cannot load driver module alsa jack main caught signal 15 here is my ~/.asoundrc pcm.myname { type hw card 0 } ctl.myname { type hw card 0 } (it is a copy from the one in /usr/share/doc/jackd/examples/) I could not find more details in the documentation or on http://jackit.sf.net Here is what I have in /dev/snd/ controlC0 pcmC0D0p seq timer There is really no /dev/snd/pcmC0D0c What am I doing wrong? I tested it with alsa started and stoped. Did I misunderstood anything essential? I'm using debian woody on a Mac PowerBook G4 with alsa 0.90rc7 from sid and with a special mac kernel 2.4.20 (from benh). Thanks very much for your help raph <---------> clef publique : http://www.informatik.uni-freiburg.de/~proust/raph-pub-key.txt S'il vous plaît, évitez de m'envoyer des attachements au format Word ou PowerPoint. Voir http://www.fsf.org/philosophy/no-word-attachments.fr.html

22 years, 9 months

[linux-audio-user] simple apps

by Jane DT

Hi thanks for your suggestions on simple apps for simple people... >From Julien: > For just recording and checking mixer-settings, I think ecasound or > qtecasound, if you like GUIs, might be ok. Otherwise there is still ardour, > muse and the other big ones, but they contain a lot of functionality. um, sorry to be silly, but how does ecasound tell me whether my mixer levels are suitable for recording the input? don't i need something like a VU meter, ..or? > Recording mp3s: Why not just record audio-data to a raw (.raw or .cdr) file > or .wav-file and convert them afterwards? ? gosh i really wasn't clear. i'm not sure what you are suggesting, but, well, i think i'm doing that. but i need to do a bit of editing along the way. Matthijs: > Sweep is quite a nice audio editor, if not that great at working with > large sound files: thanks! it has the functionality of x-fades which is really nice, and audacity doesn't. but... hmmm... whats this about large sound files... what happens?? <worries about spending hours to have it swept and mangled> Alexandre: > > But, oh god, I installed ReZound 0.7 yesterday... It has VU meters > for both recording and playing and an EQ analyser as well. > Give it a try! i tried. it sounds perfect, but... hmmm can't yet get it to compile. it depends on a lot a lot of libs that are >versions than mine. i am still struggling (debian woody/unstable) so i'm still open for suggestions for a VU meter, or even some smart way that makes my recording levels easy to assess... Now this is a vsound (0.5) problem I've had for a while. It occurs on realmedia files. Somewhen during the stream realplayer just closes. This is what it says: cygnus:~/mars> vsound --timing -f output.wav realplay _1790918_blur_beagle_vi.ram About to start the application. The output will not be available until the application exits. /usr/bin/vsound: line 163: 1366 Aborted LD_PRELOAD="$pkglibdir/libvsound.so" "$@" Any suggestions? thanks! jane

22 years, 9 months

[linux-audio-user] Sounds play too fast!!

by toby＠tobiah.org

Hello, After installing Gentoo Linux over RedHat, I find that my Delta44 playes sounds a about 1.5 semitones too fast!. Boot into windows, and it goes back to normal. Not sure if I suspect ALSA drivers, or the kernel, or what. It is not user error, like sample rate probs or anything. XMMS, aplay etc... all play content two fast, no matter what the soundfile, or mp3. Thanks for any help! Tobiah

22 years, 9 months

RE: [linux-audio-user] ANN: swh-plugins 0.3.7

by Matthew Sargent

> > > Yes, thats the main point: > > > http://www.ecs.soton.ac.uk/~swh/jrss.png > > > > This link is broken :-( > > Works for me. > > - Steve Fine from here, too. Matt

22 years, 9 months

[linux-audio-user] ANN: swh-plugins 0.3.7

by Steve Harris

http://plugin.org.uk/releases/0.3.7/ I've done a major code audit (with the help of valgrind :), and things are a lot less crufty now. There are still some outstanding known sound quality/noise/aliasing bugs, I'l tackle them in the next release, but I'd appreciate more reports. ChangeLog: 2003-02-23 Steve Harris <steve(a)plugin.org.uk> * Fixed memory leak in gate * Fixed filter implementation in gate * Fixed key defaults in gate * Made passes=0 work in GSM * Added bandlimiting filter to GSM (less cruchy sounds) 2003-02-24 Steve Harris <steve(a)plugin.org.uk> * Removed stale code from surround encoder * Fixed memory leak in surround encoder 2003-02-24 Steve Harris <steve(a)plugin.org.uk> * Fixed maths error in multiplexer * Fixed buffer overrun in sifter * Efficiency improvements to FAD delay * Fixed infinite loop in FAD delay. * Fixed (another) buffer overrun in FM oscillator * Performance improvement for FM oscillator * Fixed buffer overrun in multiband EQ * Fixed aliasing in Hermes * Fixed memory leaks in: AM pitchshift Analogue osc Bode sifters Comb Comb splitter Delayorama Dyson compressor FM oscilator Giant flange Gong GVerb Hermes filter L/C/R delay Multiband EQ Plate reverb Rate shifter Retro flanger Satan maximiser SC* Sifter Single band parametric Multiplexer Tape delay There are still known leaks in imp and the multiband EQ

22 years, 9 months

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Linux-audio-user