Linux-audio-user September 2007

linux-audio-user@lists.linuxaudio.org

130 participants
158 discussions

by Kevin Cosgrove

I've been trying to figure out how to get a particular vocal effect using Linux tools, not that I even know how to get the sound with any tools. About 8-9 seconds in to the clip below a vocal is introduced which has the sound I'm looking for. http://www.doink.com/lau/spocks_beard-day_for_night-04-the_distance_to_the_… How do I get that sound? I have 5 very good singers at my disposal, if it turns out to be a vocal styling thing. But, I think there's more going on than vocal style. It sounds like there's some sort of envelope on that vocal track. Thanks.... FWIW, the entire CD from which I snipped this section is very very good in terms of musicianship & production. After that it's personal taste, and they're among my favorite bands. -- Kevin

16 years

Audio output from Flash plugin...?

by Michael TD Nelson

Hi all, Following Ken Restivo's recent post linking to some youtube videos, I'd like to get some audio output from Firefox's Flash plugin. I mostly use JACK; is there any hope of getting it to work with JACK? If not, maybe some sort of desktop sound mixer that uses ALSA? I have an up-to-date PlanetCCRMA based on FC6, and my main soundcard at the moment is an RME DIGI9636. The machine contains an MSI K8N nForce4 board, with an onboard soundcard, which appears to have an ALSA driver. Where should I start? Maybe an appropriate setup for me in the long-term would be to use the onboard soundcard for general desktop sound (using some sort of simple sound mixer?) and JACK with one (or more) of my RME cards for audio work. Is it possible to do this in a permanent and reliable fashion? How would I go about it? I am trying out various desktop environments and window managers at the moment, and haven't settled on anything yet. Maybe I will use KDE or GNOME when I want to use my machine as a general purpose workstation, and something like fluxbox for audio. Thanks, Michael

16 years

Finally found the outer limits of my hardware

by Ken Restivo

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 This atrocity represents an unscientific attempt to discover the absolute maximum my hardware can handle: http://restivo.nfshost.com/misc/gearwhore-0.2.ogg Enjoy. I'll have to take this track down after a while because of copyright issues-- I can't remember where I obtained the vocal sample and I doubt I'd be able to obtain the rights to it, but it is inspired by Fons's .sig, which aptly summarized the project in which I was engaging. What you are hearing is, all running at the same time: ams --poly 4, running a variation on Atte's "moving overtone" patch jconv phasex whysynth, running an async granular pad several fluidsynths specimen ardour pure data running the amazing "outer space" patch seq24 several LADSPA effects With all that, I was FINALLY able to get over 60% CPU on a 2.33Ghz Intel Core 2 Duo. Surprisingly few glitches, but it was acting rather flaky under those extreme conditions, and I was able to get JACK to disappear. I've noticed something strange. I have been running more than this many softsynths and effects, simultaneously, for months, and have never been able to get it past 30% CPU. But what I did here is add lots of tracks-- some of which weren't connected to anything-- and also lots of sends and busses (also which weren't connected to much). And that chewed up a lot of CPU. Everything you hear is synthesized "live" (looped from MIDI) except for the "outer space" patch, which was recorded to disk and then embedded in ardour and streamed back. The background hiss white noise that sounds like A/D aliasing noise, is coming from jconv. I've no idea why. It does that sometimes when I insert it into Ardour. What sounds like dust on an old analog LP record, was WhySynth losing its mind, trying to run an async granular patch on a max'ed-out CPU. The "Outer Space" PD patch is an excellent torture-test in and of itself. With 128 voices running, it was able to max out the CPU and cause crackles with nothing else running. OK, that's enough of that vain delirium. - -ken -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.6 (GNU/Linux) iD8DBQFG8jbTe8HF+6xeOIcRAnVFAKD1AehTCxnaeERVA5SmAP/35Tgh3wCg86rS c1MSLe/sWCVvzUKc6U3eDeU= =D+TQ -----END PGP SIGNATURE-----

16 years

How to edit OSC-Data manual with graphical editor ?

by Winfried Ritsch

Hello, I am thinking of having an graphical editor for OSC-streams natively. I doesnt have to support complex data types like lists of mixed types, but at least floating point controller. Is there an (free) application which can do that like sequenzer handle MIDI-controller or so ? mfg winfried -- -- - ritsch(a)iem.at - http://iem.at/ritsch - Institut fuer Elektronische Musik und Akustik - University of Music and Dramatic Art Graz - Tel. ++43-316-389-3510 (3170) Fax ++43-316-389-3171 - PGP-ID 69617A69 (see keyserver http://wwwkeys.eu.gpg.net/) --

16 years

Re: wine, vst, 64bit

by Dave Phillips

Marco Milanesi wrote: > By following LAD/LAU lists, I see that you are using pure 64bit >arch, could you confirm that VSTs with 32 bit wine works? > > Hi Marco, I've cc'd this response to the lists because I want to clarify the situation for all who are interested. The short answer is, "Yes". The longer answer is "Yes, with some very specific caveats." Wine can be compiled as a 32-bit binary for running in a pure 64-bit system. I'm running such an arrangement now, with backport packages of Wine and libwine 0.9.34 (Debian, 64 Studio). However, IIRC 64 Studio already has more recent packages available. The biggest problem is latency. In this scenario it's not possible to use the JACK/wineasio combination, so you'll end up using either the OSS or ALSA sound system for Wine (select in winecfg). It may be possible to lower latency with either of those drivers, but I've not worked at it. So far I've only tested Reaper with some VST plugs. It worked, but I wouldn't try using it as a production environment. I haven't yet tested any other hosts. And just to be clear: At this time I would advise anyone who wants to run VST plugins in production to do so with either JAD or 32-bit 64 Studio. They support the wineasio driver, thus greatly improving the latency numbers. Best, dp

16 years

Re: [LAU] [ANN] JAD 1.0 officially released

by david

Hartmut Noack wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > david schrieb: >> Dave Phillips wrote: >> >>> The JackLab Project announces its first public release >> Useless to me - no CD image available. I don't have a DVD burner. >> > You can upgrade a Standard-Suse to JAD by registering its package-repos > and installing the kernel and the App-packages. > > No unevitable need to use the DVD ;-) Yah, I could burn a SuSE CD and install that, then upgrade the packages. Thanks for the idea. Just what I need - (YALAD) Yet Another Linux Audio Distro throing it's hat into the ring. I wonder how it compares to pure:bolic, dyne:bolic, Musix, Ubuntu Studio, Studio64, or what other's I forgot to include? I do sometimes wish the developers would all get together and focus on one distro and finish making the applications work right ... ;-) -- David gnome(a)hawaii.rr.com authenticity, honesty, community

16 years

Re: [LAU] M-Audio Audiophile Line Level Inputs

by Bearcat M. Sandor

Loki, Because doing D/A on an internal soundcard is noisy. I can hear the hum of my fans coming through my tweeters. Having the D/A outside of the box is quiet. Granted in a year or so i'll be replacing this with something better like a RME ADI-8 QS or universe willing a Lexicon MC12. I do know how to use ebay, but i don't have much money at the moment. $100 is about my limit to spend on this. Firewire? Last time i looked almost all of it was proprietary and not useable on linux. I would love to use firewire, but you can't run firewire D/As on Linux (RME, MOTU are out for sure). Yes, USB is ugly, but it's better then the noise you get using internal D/A. My next build will be much, much quieter, but even a fanless system still creates hum that can be heard through the speakers, doesn't it? Bearcat On Sunday 16 September 2007 in an email titled "Re: [LAU] M-Audio Audiophile Line Level Inputs" Loki Davison wrote: >On 9/17/07, Bearcat M. Sandor <HomeTheater(a)feline-soul.com> wrote: >> (Also sent to the Mythtv-users list) >> >> Folks, >> >> I have a Creative SoundBlaster Extigy, and it's just not good enough. It >> sounds fairly veiled and distorts quickly. I'm thinking of moving up to a >> M-Audio Audiophile USB. >> >> I use the line in jack on the soundblaster as the input for Mythtv, >> running the RCA jacks from my Dish Network box to the SB. Given that the >> audiophile has separate, balanced (TAS), if i got a rca-tas cable >> (assuming they exist) and run my Dish Network box into the left and right >> tas jacks would this work >> in the same way? My question about the jacks themselves, not Mythtv. >> >> Thanks, >> >> Bearcat M. Sandor >> _______________________________________________ >> Linux-audio-user mailing list >> Linux-audio-user(a)lists.linuxaudio.org >> http://lists.linuxaudio.org/mailman/listinfo.cgi/linux-audio-user > >I klnow i have a said this 2 million times on the list but... is this >for a laptop? If not, why the hell would anyone in there right mind >want a usb soundcard? BAD. Get a pci card / firewire / cardbus card. >usb-audio isn't nice unless you are really short on cash and can't >work out how to use ebay... > >Loki

16 years

Re: [LAU] RME Digi96/8

by Michael TD Nelson

Hi Reuben and all, Thanks for your I hope you don't mind me posting this back to the list! I'm sorry for spamming you with my reply earlier - I was trying to send it to the list... Anyway, I've also added some extra text this time round... On Mon, 2007-09-17 at 14:09 -0500, Reuben Martin wrote: > I have the DIGI 96/8 PAD and it works fine. There are only a few > period sizes that work since the buffer sizes on this device are > fixed. Ok. That's kind of what I expected, from what Fernando said. > If you have it working, but get an absurd number of XRUNs when > starting Jack, it probably means you have a digital input selected, > but no actual digital devices connected. It's trying to sync with an > external device, but isn't finding a signal. Select an analog input, > or actually connect something to the digital input. I think that this is almost certainly exactly what is happening. At the moment I have my ADC/DACs connected to my Digi9636, and I didn't really consider that syncing would be a problem before I'm actually using the interface. That makes perfect sense, though. I'll test it tomorrow. > If you haven't found it yet, there is a tool for the DIGI 96/8 series > that is part of alsa tools that provides a little interface for > selecting which input you want to use, your clock source, pad > settings, DAC volume, sample rate, etc. Cool! It is called 'rmedigicontrol'. The alsatools website says that 'rmedigicontrol is a control tool for RME Digi32 and RME Digi96 sound cards. It provides a graphical frontend for all the sound card controls and switches.' So, does it also work on my Digi9636? When I run it, it just opens one window which appears to control the 96/8 PAD. Just out of interest, does anyone know a *useful* way that I could sync together two Digi9636 cards and one Digi96/8 PAD? Thanks again, Michael

16 years

Re: [Musix-usuarios] horgand 1.12 released (debian/etch package; Musix)

by Marcos Guglielmetti

El Martes, 18 de Septiembre de 2007 13:44, Carlos Pino escribió: | Suena muy bien , y funciona muy bien también . Me gusta , buenos | sonidos , y los rhodes están muy bien también . Enhorabuena , muy buen | trabajo el tuyo . | | | Marcos , ya está subido al servidor,pruebalo a ver que tal va. Thanks Carlos, now we can install from the Musix's repositories Gracias Carlos, ahora podemos instalarlo desde los repositorios de Musix sudo apt-get update sudo apt-get install horgand Leyendo lista de paquetes... Hecho Creando árbol de dependencias... Hecho Se actualizarán los siguientes paquetes: horgand 1 actualizados, 0 se instalarán, 0 para eliminar y 78 no actualizados. Necesito descargar 2407kB de archivos. Se liberarán 4096B después de desempaquetar. Des:1 ftp://musix.ourproject.org ./ horgand 1.12-1 [2407kB] -- `&' # Marcos Guglielmetti, co-director de # Musix GNU+Linux, 100% Software Libre para artistas _#_ http://www.musix.org.ar (#) / O \ + archivos: ftp://musix.ourproject.org/pub/musix ( === ) Ecología: http://autosus.wordpress.com `---' Personal: http://marcospcmusica.wordpress.com "Solo los idiotas estan seguros de lo que dicen.-está seguro profesor.-completamente." http://es.wikiquote.org/wiki/El_Chavo_del_Ocho

16 years

Help with fluidsynth and midi from drum pads

by Bengt Gördén

Hi, I just got a toy (at least it's been called that before) Millenium MD70 for my kid. Anyway its just about right for my 5 year old kid. It had midi out so I thought that I try it together with Hydrogen and Fluidsynth. One thing I first noticed was that Fluidsynth just responded about half of the hits made at the pads. Hydrogen responds ok but the velocity is low compared to what I get when I just play the sound in Hydrogen. I was puzzled to why Fluidsynth couldn't respond more than every second hit so I took amidi and dumped the output. It turns out that the MD70 sends a "Program Change" every time I hit a pad. In hex it is C9 00. Do anyone out there know why this happens? Or is it normal to send a program change at every hit? I can't get Fluidsynth to ignore the program change. Do anyone know if theres a way to do that? There might be a midi filter somewhere (suggestions welcome) but I thought that it would be nice to do it directly in Fluidsynth. Here is an example of amidi -d when I hit one pad C9 00 99 23 7F 99 23 00 As i interpret it (GM that is), it's: Program change to 0 Note on for Acoustic Bass Drum with velocity 127 Note on for Acoustic Bass Drum with velocity 0 /bengan

16 years

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Linux-audio-user September 2007