Linux-audio-dev

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7502 discussions

[linux-audio-dev] integrating qt apps with common plugin interface

by Patrick Stinson

has anyone gotten a qt-based gui to work with audiounits, vst, rtas, dxi, or another commercial plugin framework? cheers -- Patrick Kidd Stinson http://www.patrickkidd.com/ http://pkaudio.sourceforge.net/ http://pksampler.sourceforge.net/

18 years, 5 months

[linux-audio-dev] [ANN] jack_mixer version 2

by Nedko Arnaudov

jack_mixer version 2 released. jack_mixer is GTK (2.x) JACK audio mixer with look similar to it`s hardware counterparts. It has lot of useful features, apart from being able to mix multiple JACK audio streams. Changes since version 1: * Fix compilation issue for 64-bit platforms (-fPIC) * Add new meter scale - iec268, fewer marks * Add hints in documentation for compiling on Ubuntu * Fix compilation with offsetof macro definition Homepage with screenshots: http://home.gna.org/jackmixer/ Download: http://download.gna.org/jackmixer/ -- Nedko Arnaudov <GnuPG KeyID: DE1716B0>

18 years, 5 months

[linux-audio-dev] [ANN] EEL 0.1.11 released

by David Olofson

EEL 0.1.11 released ------------------- Here we go again! Been throwing a whole lot of new code and new problems at EEL since the last release, and as you can see, we've skipped two public releases. This was part due to lack of time, and part because there were stability issues I wanted to deal with before making a public release. Most interesting changes from 0.1.8: * All relevant VM instructions can now handle up to 65536 constants. * Compiling large files with line number debug info is now a lot faster. * for i=N, N-1 ... will no longer iterate once. * 64 bit pointer issue in relocate_limbo() fixed. * Fixed octal number escape parsing in strings. * A bunch of improvements to the EELBox GUI toolkit. (Sorry about the current lack of serious examples...) * Added a DSP module, with FFT, polynomial evaluation etc. * Minor audio API improvements (mono samples and vector support. * Vectors and arrays can now be sliced, and... * ...slice() has been renamed copy(), to avoid confusion with an upcoming language feature. * MIDI input support. (Very basic and ALSA only, for now.) * A bunch of new example/test scripts. (Hey, fftsynth.eel is almost interesting! ;-) * Various bug fixes and code cleanups. ChangeLog: http://eel.olofson.net/ChangeLog Home: http://eel.olofson.net/ Downloads: http://eel.olofson.net/download.html Direct download: http://eel.olofson.net/download/EEL-0.1.11.tar.gz Win32 binary package: http://eel.olofson.net/download/EEL-0.1.11-Win32-bin.zip //David Olofson - Programmer, Composer, Open Source Advocate .------- http://olofson.net - Games, SDL examples -------. | http://zeespace.net - 2.5D rendering engine | | http://audiality.org - Music/audio engine | | http://eel.olofson.net - Real time scripting | '-- http://www.reologica.se - Rheology instrumentation --'

18 years, 6 months

[linux-audio-dev] block i/o scheduler for JACK

by Andrew Gaydenko

(sorry for crossposting - have tried at LAU mailing list, but have not got any answer) Which kernel block i/o scheduler is most appropriate for JACK application? There are alternatives: Anticipatory I/O scheduler, Deadline I/O scheduler and CFQ I/O scheduler. I mean a kernel without RT-related patches.

18 years, 6 months

[linux-audio-dev] libsndfile, or audiofile?

by kind king knight

There is not much to say. Need to know, because learning two different interfaces would be non-economical :P.

18 years, 6 months

[linux-audio-dev] [ANN] Das_watchdog V0.3.1

by Kjetil S. Matheussen

Download from http://www.notam02.no/arkiv/src/ Das_watchdog ============ Whenever a program locks up the machine, das_watchdog will temporarily sets all realtime process to non-realtime for 8 seconds. You will get an xmessage window up on the screen whenever that happens. Das_watchdog is made to be run as a system daemon and does not interfere with normal operations (unless the lock-ups are supposed to happen). Das_Watchdog is inspired by the rt_watchdog program made by Florian Schmidt (http://tapas.affenbande.org/?page_id=38). Changes 0.2.5->0.3.1 -------------------- *Changed scheme for finding correct XAUTHORITY environment variable. (Now works with Fedora Core 6) Hopefully, theses changes should increase the chance of seeing the xmessage and avoid seeing multiple ones. (Theres no correct way to do this, so please send me the output of "uname -a" in case you don't see any window) *Added syslogging. *Added the --version argument.

18 years, 6 months

Re: [linux-audio-dev] Simple Jack mixing problem

by florentberthaut＠no-log.org

Hi thanks for answering, Ok i think i'll leave it to the user, because the goal is to make a simple sampler with kit/instrument creation and control. Flo David Olofson a écrit : > On Friday 05 January 2007 01:44, florentberthaut(a)no-log.org wrote: >> Hi everyone, >> >> I'm working on a sampler and i have a really silly problem. >> >> I simply want to mix sounds in a stereo buffer within the >> jack "process" function. >> So for each frame i sum the samples from each sound but then the >> values go above 1 and i get clipping. >> So my first thought was to divide by the number of samples playing, >> but it ends up with a decrease in volume. >> I tried to limit the values between -1 and 1 but it 's also >> clipping -> distortion. > > Well, that's what JACK is doing anyway - or there would be wrapping in > the float -> int conversion, which sounds a lot worse than clipping. > It's just not possible to represent larger amplitudes than [-1, 1] in > the integer domain (that is, [-32768, 32767] or whatever, depending > on resolution), which is what DACs deal with. > >> I looked into some sources and it seems i would actually just have >> to sum the samples. > > Yep, that's all there is to it, really - and it's the user's > responsibility to keep volumes low enough to avoid driving the output > into clipping. > > This is true on hardware synths and samplers too. The only reason some > of them appears not to have this problem is that they use various > tricks to hide it. > > For example, the old Roland JV-1080 had 18 bit DACs, where the two > extra bits were pretty much used for extra headroom above > the "normal" max amplitude of a single voice. Even so, it was pretty > easy to drive it into clipping by using resonant filters and/or lots > of loud voices. > > I wouldn't be surprised if some synths and samplers use 20 och 24 bit > DACs for even more headroom, waveshaping to make the last 12 dB or so > non-linear (soft saturation, sort of), dynamic output gain control > and stuff like that. > > Now, if you're dealing in digital, extra DAC bits and dynamic output > gain control is out, obviously. (Goes for plugins and JACK clients as > well as hardware devices with digital outputs.) So, either you set > your sampler's 0 dB level at -12 dB or something (allows mixing four > maximized samples at maximum volume without clipping), or you leave > that to the user. > > If you really want to make the impression sqeezing in more than > there's room for, you'll have to add some dynamics processing. There > is of course no way of doing this without some sort of distortion, > but even a simple waveshaper to "flatten out" peaks above 12 dB below > clipping sounds a great deal better than hard clipping. > > However, I'd rather see that kind of stuff left to the user. Those of > us who use 20+ bit sound cards would rather have it all linear and > undistorted right into the amplifier. There is always the option of > inserting some serious multiband compressor last thing in the JACK > chain. Might actually be a good idea anyway, if you like playing loud > and don't want to ruin your ears if some synth or effect freaks out. > > > //David Olofson - Programmer, Composer, Open Source Advocate > > .------- http://olofson.net - Games, SDL examples -------. > | http://zeespace.net - 2.5D rendering engine | > | http://audiality.org - Music/audio engine | > | http://eel.olofson.net - Real time scripting | > '-- http://www.reologica.se - Rheology instrumentation --' >

18 years, 6 months

[linux-audio-dev] Simple Jack mixing problem

by florentberthaut＠no-log.org

Hi everyone, I'm working on a sampler and i have a really silly problem. I simply want to mix sounds in a stereo buffer within the jack "process" function. So for each frame i sum the samples from each sound but then the values go above 1 and i get clipping. So my first thought was to divide by the number of samples playing, but it ends up with a decrease in volume. I tried to limit the values between -1 and 1 but it 's also clipping -> distortion. I looked into some sources and it seems i would actually just have to sum the samples. Can anyone help me with that ? Thanks Flo

18 years, 6 months

[linux-audio-dev] Simple music sequencer/sample player (drummmachine?)

by kind king knight

>libzzuvb is probably closest to what you describe. you could also hack together dino|seq24 with chionic|specimen, anything heftier (eg, wired, LMMS, hydrogen) is likely to be more annoynig to extract Those programs are to complicated. I need a toy to learn from it (it should be simple, but written using good techniques). I tough more about something like Hammerhead ( http://www.threechords.com/hammerhead/ ), or even something easier (no gui, one soucefile). >/dev/dsp represents a deprecated interface. >moreover, what happens when someone wants to route the output of your >cool, modified mod player through FreqTweak or jack_convolve? Like I said, I want to learn from it, so I don't need a complicated soundserver support (but I don't say that I won't ever use it). If there isn't such program, then maybe someone could write it for me :D.

18 years, 6 months

[linux-audio-dev] Simple music sequencer/sample player (drummmachine?)

by kind king knight

I need a framework of a sequenced sample player. This is becouse I want to start my own project, and don't want to invent everything from scratch. Ofcourse there is lot of this kind opensource applications, but I need the simplest. - It should use only /dev/dsp - the sequence should be in a form of an char array, where the bits decide what voice to play. For egzample: seq[]={1,0,0,0,2,0,1,3,0,3,1,0,2,0,1,0} seq[0]==1 means that in first step should be played only the first sample (bassdrum),seq[1]==0 means pause (no sample), seq[4]==2 second voice (snare), and seq[7]==3 means snare +bass drumm. - I need to understand how the sequence procesing, and sound mixing is done in real time, and how the gaps betwen sounds should be iplemented. - the samples should be held in raw arrays (16 of 8 bit), eventualy downloadable from hdd. - a simple dsp like lp-filter would be nice to see, becouse i am curios how the time needed to proces the signal should be counted into the latency (and buffer size). - there is also a need for bpm clock. - a simple interpolation would be nice. Ok. I think that this is all! :P Or wait!!!!! I have a better idea! I there a way to make those things using for example libmikmod? I would like to start a music program for live mixing of mod, and xm modules (plaing, and muting chanels, realtime patterns changing, looping sequences). If someoune is interested in helping, I am open for sugestions.

18 years, 6 months

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Linux-audio-dev