Linux-audio-dev June 2005

linux-audio-dev@lists.linuxaudio.org

120 participants
101 discussions

by Brian Black

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 hi, This might be a a newbie question, but i have openal installed and i am running into problems. i have also posted to openal-dev list about my concerns. I have created an ~/.asoundrc file for help with dmix, because i have had problems playing multiple sound programs at the same time. This .asoundrc file helped me use xmms and amarok at the same time. 1. My problem is when i am trying to run an openal program when i am using one of the other sound programs, the program ran last will block untill the first program that i started had finished. This seems strange to me that some programs will "dmix" and some will not. Any ideas on this? Also, I am thinking about developing and audio app using the openal api, and i had some questions that maybe someone could clear up for me. 1. I am assuming this api wraps around alsa or oss, i am going to be using an RME hammerfall dsp and i need to attach some inputs to some outputs to allow the input from a mic to go to the output which will go to their headset, (low latency Monitoring). How is this done, as i have not found any docs talking about how to send sounds out certain channels? 2. Some examples i have found do not compile because they are including the file #include <strstring.h> , which i do not have. (i am running gentoo 2005.0, kernel-2.6.11)? is this a known issue? thanx in advanced bj -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFCwEfQI9Y/1VIS+jgRAqDRAKCGQNQFa9cJ55qFHO4hYnVfJzsNVgCeJQNb 9aSjBYkHBeLaagkjI9h65BQ= =98ga -----END PGP SIGNATURE-----

19 years, 6 months

[linux-audio-dev] jack_convolve-0.0.10, libconvolve-0.0.3 released

by Florian Schmidt

Hi, i added an experimental feature to libconvolve/jack_convolve which might be able to preserve some of those precious cpu cycles from being burnt. jack_convolve now has commandline switches --min_bin=bin_no and --max_bin=bin_no which can be used to specify which bins of the fourier transformed signal and response to multiply. The range for both is always 0..periodsize+1, but min_bin must always be < than max_bin.. An example: assuming jack periodsize of 2048 and samplerate of 48khz: jack_convolve response_file.wav --max_bin=1200 This leaves out the top 849 bins out of the multiplication and thus reduces cpu load by ca. 2/5. The cost is that the high frequency spectrum is cut off. And it's not even a really clean cutoff which jack_convolve response_file.wav --max_bin=40 would show (you get crackles due to edge effects, etc). This feature is probably most useful to get some preview with degraded quality but less cpu consumption. Although at some samplerates it even makes sense. Assume a samplerate of 96khz, then there's quite a bit of signal which doesn't need to be processed since it's far out of the range of human perception. the min_bin parameter is not really useful and only included for completeness sake :) It cuts out low frequency bins out of the equation. Due to the crackle effects when throwing away bins that are audible one should not use these settings for filtering of the convolution in the audible frequency range.. In one of the previous releases (which went unannounced i think) i also added a gain argument which can be used to boost the level of the convolution output. Example: jack_convolve response_file.wav --gain=4.0 to get a 300% increase in level.. Regards, Flo P.S.: I am also currently working on a qt app which uses libconvolve, but it might still take a while until initial release (due to my time constraints induced by studying). Preliminary screenshot here: http://www.affenbande.org/~tapas/wordpress/?page_id=27 Help would, of course, be appreciated :) Anyone know of waveform or vu meter widgets for qt which are reusable? Flo -- Palimm Palimm! http://affenbande.org/~tapas/

19 years, 6 months

Re: [linux-audio-dev] jack_convolve-0.0.10, libconvolve-0.0.3 released

by Ben Loftis

> (*) Recent experiments by prof. Angelo Farina (Univ. of Parma, Italy) > suggest strongly that when the DA conversion is done properly, there is > no audible difference between a sample rate of 48 kHz and any higher value. > OTOH, he only found one type of DAC that was good enough in order to be > completely free of audible artefacts at 48 kHz (by Apogee, and they are > quite expensive). > Is there an online publication of Mr Farina's findings? Google is not turning up anything definitive for me. -Ben

19 years, 6 months

[linux-audio-dev] jack.el -- Run and monitor JACK from within Emacs

by Mario Lang

Hi. I am sure this sounds a bit crazy to some of you, but be ensured, I am not (yet) completely out of my mind and there are actually real reasons for doing what I do :-). I am now running jackd from within Emacs, using jack.el. Find it here: http://delysid.org/emacs/jack.el Basic usage: * Install the elisp file and put (require 'jack) in your .emacs. * Customize startup options: M-x customize-group RET jack RET * I.e. jack-sample-rate and jack-period-size are two important candidates. * Start jack: M-x jack-start You will see the verbose output in a special buffer called *JACK output*. * If you want/need to get rid of your jack instance, simply do M-x jack-kill RET * If you changed sample rate or any other startup parameter, do M-x jack-restart RET to make them effective. Plans: * Do something useful with xrun output, maybe some stats (suggestions?). * Add support for dummy and oss drivers, and add remaining ALSA params. * Write a patchbay based on jack_lsp and jack_{dis,}connect. Why am I doing this? You might have noticed I am recently seriously working on Emacs extensions related to Sound and audio. osc.el is working nicely and om.el, a client to drobillas om-synth is just waiting for a polish up. midi.el is parsing binary data already and needs a bit more careful planning before it can really grow. The idea behind jack.el was that I basically got fed up by jack wasting either a virtual console, or a screen window. Both dont really offer confortable scrollback, so the idea to make jack run from within Emacs, and collect all its output into a Emacs buffer was born. Some nice sideeffects: Load statistic is collected in a variable, so if you are into elisp, you can do fancy stuff like: (with-current-buffer (jack-output-buffer) jack-load) to retrieve the current load :-) Combine all this with things like csound-x (Csound support for Emacs) and the SuperCollider Emacs frontend, and you get a truly powerful composition and experimentation environment. One day I'll learn CLM and CM, and add it to the mix as well :-) -- CYa, Mario

19 years, 6 months

[linux-audio-dev] Linuxtag 2005

by Christoph Eckert

Hi all, Linuxtag[1] 2005 is over. It took place from 20050622 to 20050625 in Karlsruhe, Germany. We had a booth there populated by the following people: Arnold Krille Christoph Eckert Frank Neumann Gerd Flaig Joachim Schiele Moritz Dressler Reinhard Katzmann We had 4 presentation places with a keyboard and a notebook each, additionally an acoustic guitar and an electric violin so we have been able to show soft synths as well as realtime audio processing. Needless to say that there have been lots of USB audio devices :) . There was a lot of interest of the visitors in what we can do with free audio software, and my observation is that most people have been simply surprised by the amount of available applications as well as the quality of the apps and the processed audio material. Furthermore I had the occasion to hold two talks about linux audio, one for musicians[2] (en) and one for desktop users[3] (de). Because it has been a successful show with really great software, I'd like to thank all who helped and especially those who do the hard background work on free audio software. Best regards ce [1] http://www.linuxtag.org/typo3site/8.0.html?L=1 [2] http://www.linuxtag.org/typo3site/ freecongress-details.html?&L=0&talkid=145 [3] http://www.linuxtag.org/typo3site/ freecongress-details.html?&L=0&talkid=300

19 years, 6 months

[linux-audio-dev] What parts of Linux audio simply suck ?

by Dave Phillips

Greetings: The subject says it all. My own "Linux audio sucks" hobbyhorse: No way to recall a complex configuration of apps and plugins with all settings intact. If I use a complicated setup with multiple synths and plugins I have no way to recall these applications to their previous settings. LASH/LADCCA was supposed to address this situation but I don't know where that project stands at this point. And your favorite is... ? Best, dp

19 years, 6 months

[linux-audio-dev] La Marmite

by nico

Hi, I started last year the project of an small framework for audio apps based on JACK and LADSPA. I published it quite a while ago on : http://lamarmite.poivron.org But I'm now really doubting of the interest of this project and this should be THE place to find answers ... --*/--*/--*/--*/ Here is a short intro La Marmite is a C++ framework cooked for the development of audio applications for GNU/Linux which is based on the will of simplifying the combined use of the standard tools JACK and LADSPA. La Marmite is free software and published under the GNU Public License. JACK and LADSPA are two major ingredients from the Linux audio world but mixing them together often bury me in long hours of C coding which rapidly oxidize my placid humour... My idea was to construct a object-oriented environment providing a higher level of abstraction and which allows me to concentrate on the logic of the application and to achieve my goal faster without having neither to bury myself in the inmost depths of the interfaces nor to recode over and over the same routines. Do you feel like ? La Marmite was designed in order to create same applications, audio toys, or autonomous audio modules which be further mixed or combined through JACK. But only few things have been realised so far. And it might not be well suited for larger scale applications.

19 years, 6 months

[linux-audio-dev] jackd ubuntu oddities

by Ivica Ico Bukvic

Hi all, I am quite flabbergasted by this particular problem as it never was a problem before. Therefore, I would greatly appreciate any help I can get in this matter. I've installed Ubuntu Hoary and then installed realtime-lsm module from the Hoary deb repository. The realtime-lsm definitely works without a problem and has been thoroughly tested. If I install any audio jackd-capable app from Ubuntu Hoary deb repository, it works just fine with jackd (which was also installed from the repository) both real-time and non-realtime, except for the Supercollider. Jackd uses in /tmp folder which has been mounted as tmpfs to improve performance. I tried installing Supercollider from source and it exhibits the same behavior as the .deb. I did install Alsa 1.0.9 as 1.0.8 that ships with Ubuntu has some bugs in respect to hdsp driver (although Ubuntu may have patched these, haven't bothered to check). When trying to connect to non-rt jackd, Supercollider works, but when connecting to real-time enabled jackd it connects and immediately gets disconnected reporting error that it has been "zombified" by jackd. I tried even enabling realtime-lsm for all apps and that did not help. Starting jackd in softmode does not help. Now, I know that it worked just fine on Mandrake with realtime-lsm module so I have no idea why jackd is being so picky. Considering all the problems, I tried installing jackd from source (Hoary has 0.99.0, so I downloaded the same) and now here is where the rub starts. Once I install the same version jackd from source, no app from Hoary's .deb repository works any more with it, including qjackctl. Error reported is "could not tie an input port" or something along those lines. So, the only thing I can do is then install all the audio apps from source to make them work with the jackd. So here are my questions which I would greatly appreciate your help in answering: 1. Why is this happening when both the .deb and source versions of jackd are the same? 2. So what can I do at this point and why some apps (pd sometimes also gets zombified) get so easily kicked from jackd while others don't? 3. Is there anything I can do to retain compatibility with the jackd-enabled apps from .deb repository when compiling newer version of jack from the source? 4. What actual version of jackd is Ubuntu using? How is it altered? 5. What can I do to make jackd tolerate initial start-up delays (ironically this has not been the problem in the past with supercollider, so at this point I am not sure whether it is a problem with supercollider, jackd, or Ubuntu)? 6. Should recompiling kernel with low-lat and realtime patches (i.e. ck) help solve some of these problems? 7. Any other advice that may help solve these problems would be most appreciated! Sorry for cross-posting, I am hoping that Ubuntu devs/users may be able to answer some of these. Many thanks! Best wishes, Ivica Ico Bukvic, composer & multimedia sculptor http://meowing.ccm.uc.edu/~ico/

19 years, 6 months

Re: [linux-audio-dev] [ot] [rant] gcc, you let me down one time toomany

by Michael.Gogins＠sungard.com

I did test it, and it was just as fast either way (g++ 3.3/MSVC++ 7.0 on Windows 2000 a year or so ago). Regards, MIke

19 years, 6 months

[linux-audio-dev] Author on LADSPA/DSSI wanted

by Oliver Frommel

Hello altogether, the German language Linux-Magazin is going to produce an issue on audio. Therefore we are looking for someone who is willing and able to write on LADSPA and/or DSSI, possibly focusing on just one of both technologies. If you are interested please contact me. Thanks Oliver -- Linux New Media AG Süskindstraße 4 D-81929 München Tel: +49 (89) 99 34 11 42 Fax: +49 (89) 99 34 11 99

19 years, 6 months

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Linux-audio-dev June 2005