Linux-audio-user March 2003

linux-audio-user@lists.linuxaudio.org

134 participants
134 discussions

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

21 years, 8 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

21 years, 8 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

21 years, 8 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

21 years, 8 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

21 years, 8 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

21 years, 8 months

[linux-audio-user] LPs to CD - help with filters

by Reid Vail

I need some help understanding the basics of audio editing/filters using the linux tools. I've successfully used Broadcast2000 and SND to capture music input from cassette tape and LP, and have written some to CD. Next I want to do some work on the files, filter out what nasty stuff I can, but don't know what filters have what effect. I'm also not sure of the steps to think the process through; that is, if you see/hear symptom A, then X process, or X filter. Actually, I know how to apply the "plugins" in Broadcast 2000, but don't know what to pick or why. In SND I haven't gotten that far, but know it's possible. Any hints, howto suggestion or pointers would be GREATLY appreciated. My audio equipment is pretty modest; a SB16 ISA...and I know I might have to change that. Thanks in advance Reid

21 years, 8 months

[linux-audio-user] various questions (long)

by Levi Burton

Greetings list, Thanks again for your responses to my questions about iiwusynth, zynaddsubfx and jack; you were all most helpfull. And now for something completely different... When creating music, there are various issues I run into. First, the way I would like to work is to build tracks within something such as Rosegarden. I use a hardware synth (alesis qs6), and other software tools. Since I have a relatively slow computer, I would like to build the tracks, then record each one separately, then composite them with something like ecasound. My first problem is the recording of the various tracks. Right now, I use snd because it has this record feature called "trigger". This feature is extremely usefull. My question is do any other software packages support this feature? For those who are not familiar with it (or cannot infer its meaning), it essentially just waits until the audio level reaches a specified threshold, then starts recording. I would love to know if ecasound can do this. My next question is one concerning synchronicity of my composited tracks, and also of sample loops. I am not mathematically inclined, and i do in fact suck at math. What I would like to know is if there is (im sure there is) a formula for calculating audio segment lengths according to a specified BPM. For example, say I have a loop of some recorded drums at 120 BPM. I would like to know exactly how long the audio sample must be to match 120BPM, rather then trial and error. I'm sure I had more questions but I seem to have forgotten them at the moment, but im sure this is enough for this particular post! Thanks! -- Levi Burton http://www.puresimplicity.net/~ldb/

21 years, 8 months

[linux-audio-user] [ANN] RTMix software interactive multimedia interface version 0.7 released (3/10/03)

by Ivica Bukvic

Hello fellow musicians, composers, multimedia sculptors, and developers! It is my pleasure to announce to you the newest release of RTMix version 0.7. New improvements include: *Internal error widget has been removed. *External Error Log now became a general purpose Console. *Added visuals for monitoring of data flow at the bottom of the external Console. *Sped-up the start-up time by 300-400% (literally :-) *Fixed bug where global transport events did not execute on a local client causing many scripts to fail. *Reordered Tabs on the main widget. *Made External resizable widgets resizable only when performance was not in session (in order to minimize cpu-utilization by RTMix during the performance settings). *Fixed Metronome weird resizing bug. Now when the meter is changed, the widget resizes appropriately. *Fixed color coding errors for the notification interface. *Re-arranged the settings saving routines. *Added new parameters to the config file. *Added networking authentication code and made it configurable in the settings tab. *Added filter for events in order to disable potentially malicious sys-calls to be executed. *Fixed gazillion (literally :-) bugs in the parsing engine. *Standardized error and logging output messages. *Color-coded Console messages. *Added "go-to-error" feature. *Implemented MIDI protocol as a separate thread. Users can now use MIDI for real-time events, as well as MIDI routing. *Provided new tabs in the settings menu that enable user to specify the appropriate MIDI port. *Implemented OSC (Open Sound Control) for inter-app communication. *Implemented generic OSC network communication. *Implemented OSC routing for the purpose of sharing the MIDI port. *Enabled variables to be included in notification interface messages. *Enabled multiple instance of variables and MIDI parameters to be included in functions, assigns, and events (sys calls and others). *Fixed metronome's inconsistent resizing. *Fixed bug where BPM's on the metronomes 2-4 were corrupt. *Jump events now interpret events they jump to in a proper fashion. *Added full-fledged HTML documentation (Yay!) *Included more tutorials and provided better annotations for the older ones. *Made apply button disabled in the settings menu, unless something was changed. *Made MIDI monitoring and MIDI logging buttons disabled by default, unless the real-time monitoring is enabled. *Added color-coding and more verbose descriptions of the real-time events in the table. *Enabled differentiation between keyboard presses and releases and their mappings to the real-time events. *Added line-number tool for the editor. *Fixed behavior of the probability parameters. *Annotated more parser's warnings. *Implemented protection against infinite recursion scripts. *Other stuff that I cannot think of at this moment. --------------------------------------------------------------- RTMix is downloadable here: http://meowing.ccm.uc.edu/~ico/rtmix-latest.tar.gz (approx. file size is 4.8MB). For more info, see the included HTML docs, or visit the author's webpage at http://meowing.ccm.uc.edu/~ico/ RTMix currently runs only on Linux, although the transparency of code should make it easily portable to other Unix platforms supporting Qt toolkit. --------------------------------------------------------------- If you are not familiar with RTMix, here's a quick overview: What is RTMix? RTMix is an open-source (GPL-licensed) software application designed to provide stable, user-friendly, standardized, and efficient performance interface that enables performer(s) to interact with both the computer and each other in the least obtrusive fashion. What this means is that RTMix offers an array of visual stimuli that can be utilized on-stage in order to coordinate various performing forces utilizing diverse media. What do I need it for? How many times have you witnessed an interactive work that requires coordination between the composer and performer, composer usually being off-stage and posing as an aircraft navigator sending out all kinds of signals with waving hands and other distracting (perhaps even comical) physical gestures? Have you ever questioned computer's off-stage presence when it has an important role in generating the resulting sonic landscape (or even a multimedia setting)? Did you ever wish to have an elegant on-stage interface that is easy to use and furthermore provides the least amount of distraction for the performer(s) -- an interface that offers standardization, transportability, and most importantly low cpu-footprint, therefore enabling user to utilize majority of processor cycles for the stuff that matters the most -- the content-generation, processing and reproduction? Have you ever wished to have your work more "transportable", to have it more accessible and more easily performable in settings where you were not physically available to provide technical support to the not-so-computer-literate performer? Did you ever write a chamber acoustic work that required considerate amount of coordination but you did not want to use a conductor? How about a work for a large performing groups? Do you use powerful Music-N languages for real-time work but do not have an elegant interface for real-time performance settings? Are you a PD/Max/MSP/jMax object-oriented multimedia composer, but do not want to deal with designing the user-interface for your contraptions, nor with the lack of standardization such interfaces impose on end-users (i.e. performers other than the composer themselves)? Did you ever feel like using only one multimedia tool at a time was limiting your creativity (i.e. Csound, RTcmix, Supercollider, Pd, Max/MSP, etc.) and that you always wanted to have multiple audio applications to coexist in your work? If you have answered any of these questions positively, then RTMix just might be the answer to your needs :-). Ivica Ico Bukvic, multimedia sculptor http://meowing.ccm.uc.edu/~ico P.S. Apologies for cross-posting! Some mailing lists ate up my original post :-(

21 years, 8 months

[linux-audio-user] Audio routing software

by Kristof Pelckmans

Hi all, I would like to route audio streams coming from different (software) sources to the outputs of my different sound cards (1 card multichannel and 2 cards 2-channel). The outputs are connected to different rooms in my house. For the moment, I'm using Alsa with different 'routing-profiles' in my .asoundrc settings, for instance one profile that will address all my different channels of my multichannel card (this would cover the living room, kitchen, bathroom and study, but not the bedroom). I can use this profile with aplay to play a soundfile in all these rooms. However I would now like to switch to different rooms _while_ playing a soundfile with aplay. I am dreaming of some sort of interface that has different audio sources on one end (mp3, television, radio, ...) and outputs on the other side (kitchen, living room, bathroom, ...). Using this interface, one would be able to switch any source to any output. Preferable, also with different volume-settings per channel, per source. I allready looked at Jack and aRtsd, but I can't seem to figure out if they are the holy grail. Has anyone encountered such routing issues ? Is there any program out there that already does more or less what I described ? Any thoughts or comments are greatly appreciated ! Ciaos, Kristof

21 years, 8 months

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Linux-audio-user March 2003