Linux-audio-dev

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7534 discussions

[linux-audio-dev] Anyone else ever run a THD test on Jamin?

by Dan Mills

Hi all, While hacking around with aliasing effects in digital compressors (Yes it is real, yes you can hear it!), I happened to run a 10Khz sine wave into jamin with an instance of Jaaa hooked up to the output. The results were 'interesting' as it appears that jamin introduces easily measurable harmonic distortion even with all compressors and eq bypassed! Switching the master bypass in jamin however does make the effect go away. This was at a level of -26dbFS with no boost and with the limiter bypassed. I say 'harmonic distortion', but it really is not quite that as it appears as a series of narrow spikes every few hundred Hz. Further checks show that the same effect appears with the test tone turned down to 2.5Khz and that eq bypass has no effect. Now with a single tone we are looking at per 'harmonic' energy almost a hundred db down on the test tone, but there are a lot of these spikes and they increase in number as additional inputs are added, so there might be quite a lot of energy here all told. Anyone have any ideas? This thing should be linear under these conditions! Regards, Dan. Send instant messages to your online friends http://uk.messenger.yahoo.com

19 years, 3 months

[linux-audio-dev] alsa, oss , efficiency?

by lemmel

Thanks for all of your mails, I collected a lot of information :-) Bye !

19 years, 3 months

[linux-audio-dev] Multiplexing 4 channels on SPDIF

by Fons Adriaensen

Hello all, Core Sound <http://www.core-sound.com/default.php> willsoon be offering a tetrahedral (Ambisonic) microphone at a very reasonable price. They are also working on a combined preamp + AD converter unit for this mic. This will be able to multiplex the 4 channels over a single SPDIF link, by using it a the double sample frequency. I'm currently working on a software controller unit for this microphone. It will perform A-format to B-format conversion, and allow measured impulse responses to be used for calibrating the four mics. The result should be a very high quality portable surround recording system at a reasonable price (compared to other solutions which cost easily five times as much). The remaining problem is the demultiplexing of the two double speed SPDIF channels to four channels. It could either be done within the ALSA layer, or in JACK's ALSA backend. Doing this in a JACK client will not work unless it would be the only client - all others would get the wrong idea of the sample frequency and buffer size. So here's my question to both the ALSA and JACK teams: what would be your idea of a solution for this ? -- FA Lascia la spina, cogli la rosa.

19 years, 3 months

[linux-audio-dev] [ANN] Faust Online

by Orlarey Yann

Dear All, A completely rewritten Faust website is available at http://faudiostream.sourceforge.net or at http://faust.grame.fr . Its main feature is the possibility to use the Faust compiler online, via the web pages, without having to install it on your machine. The website contains a small catalog of softwares written in Faust. Each software is available to download in various binary formats (Linux/i386) : * standalone applications : alsa-gtk, jack-gtk, oss-gtk, libsndfile command line * plugins : ladspa, puredata, Q, supercollider The SVG block-diagrams of each software are also available ('->svg' link). These block-diagrams were generated by the Faust compiler using the -svg option. You can navigate down the hierarchy of block diagrams by clicking on the blue boxes. A click on the white background allows to move up. Another link, the '->code' link, allows to view and edit the Faust code of the application. It is useful to learn how a specific application is written in Faust. But the funny part is to modify the Faust code to your needs. You can then recompile it and look at the generated C++ code or at the corresponding SVG block-diagrams. Once your Faust program successfully compile, you can choose the appropriate architecture and download a 'source' tar.gz file that contains the C++ code and a Makefile, or directly a binary file for Linux/i386. Please note that for puredata, the binary file is a tar.gz archive that packs together the plugin and a user interface patch, thanks to Albert Graef 'faust2pd' patch generator. For supercollider it is also a tar.gz archive that packs together the plugin and a class description, thanks to Stefan Kersten 'faust2sc' class generator. Please note also that the website is still experimental and probably not very robust. To navigate the website use the navigation buttons on the left, the back and forward buttons will not work very well. Yann

19 years, 3 months

[linux-audio-dev] Re: Re: alsa, oss , efficiency?

by Kjetil S. Matheussen

lemmel: > > Le mercredi 01 novembre 2006 16:51, Kjetil S. Matheussen a écrit : >> Only the oss modules in the linux kernel are deprecated. Programs using >> the OSS api will still continue to work, currently most importantly >> because of the oss emulation module in alsa. > I knew that the oss module was deprecated, and I infered that using the OSS > mecanism was too deprecated :-P. Are sure [1]? Hmm, well, who decides... I think its pretty safe to write code for OSS and be sure it will work for a long time. Maybe longer than code written for ALSA as well, who knows. >> If you should choose between alsa and oss, and the oss version works >> just as well, or better than the alsa version, choose oss, because its a >> more portable API than alsa. > What do you mean by "the oss version works just as well, or better than the > alsa version" ? Thats a liberal quote... But sorry, I don't think I can write what I ment any clearer... (well, you can remove the comma after "well", but thats as clear as I can make it, english is not my native language either. :-) ) >> However, if I were you, I would use sndlib, portaudio, jack, or some other >> higher level audio input/output library instead of oss or alsa. > Well, all the files, that I will play, will have the same charactistics, do I > really need to bother with an hi-level API [2] ? Why not? > > [1] I noticed that a lot of applications still use OSS, and I thought it was > because the migration to ALSA was too complex. That might be one reason. But if it works well with OSS, thats often good enough. Its a shame it sometimes is hard to make OSS programs work with jack though, but thats a problem that goes for alsa programs as well. (Actually, its worse with alsa programs.) > [2] Futhermore I am afraid that an hi-level API will produce time drifts in > the playing. > No, I don't think so. Many people use portaudio, it could be your best choise.

19 years, 3 months

[linux-audio-dev] Pd Q and Faust interfaces / ICMC 2006 demo

by Albert Graef

Hi all, (Sorry for crossposting.) This should be interesting for all who want to extend their Pd with programs written in languages other than C/C++. I have created Pd plugin interfaces for these two languages: - Q, a functional programming language based on term rewriting (my own creation; see http://q-lang.sf.net). Q is a modern-style functional programming language in which functions are defined by equations. It also has an extensive collection of modules for doing graphics and multimedia. My Pd/Q external allows you to execute Q functions in order to do complicated control stuff in Pd, and it also provides a way to access Q's multimedia interfaces inside Pd. This is available as a source tarball (pd-qext-0.1.tar.gz); RPMs for Linux systems are also available. - Faust, Yann Orlarey's functional DSP programming language (http://faudiostream.sf.net). Faust's programming model combines two approaches: functional programming and block diagram composition. You can think of Faust as a structured block diagram language with a textual syntax. The resulting C++ code is heavily optimized and can compete in speed with handwritten C code. My Faust architecture file allows Faust programs to be translated to Pd externals using the Faust compiler. This makes it very easy to create new audio externals for Pd, and a bunch of examples are readily available. I have also written a Q script which generates complete Pd patches with graph-on-parent GUIs from Faust programs. This stuff can be found in the faust2pd-1.0.tar.gz package. You can find all the good stuff, including documentation and a lot of examples on the Q website at: http://q-lang.sourceforge.net/examples.html#Multimedia (see the bottom of this page). The Faust interface will also soon be available as a part of the mainstream Faust distribution. Yann and me will show Faust, Q and their Pd and SuperCollider interfaces at the International Computer Music Conference (ICMC) next week in New Orleans, so if you have an opportunity to come we hope to meet you there. (The presentation is on the very last day of the conference, Sat Nov 11th, 3:30 p.m., see http://www.icmc2006.org.) For more information please see http://faudiostream.sf.net and http://q-lang.sf.net. Enjoy. :) Albert -- Dr. Albert Gr"af Dept. of Music-Informatics, University of Mainz, Germany Email: Dr.Graef(a)t-online.de, ag(a)muwiinfa.geschichte.uni-mainz.de WWW: http://www.musikinformatik.uni-mainz.de/ag

19 years, 3 months

[linux-audio-dev] alsa, oss , efficiency?

by lemmel

Hi everyone. for a project, we need to be able to play sound (at first look wav file), and we made several tests ; with a created stereo sound, we try to use alsa but the results doesn't fullfill our needs : sound played at the time, T, we want, and finished at the date, D=T+sound duration. (thi is a software with strict time constraints) The sound was always troncated (even with finished software such as xmms, amarok), and even randomly truncated, (sound created with audacity, and exported as WAV 16/32 bit etc). When we use OSS, all seems to be perfect. But, it seems that OSS is nowadays "deprecated", and consequently we shouldn't use OSS. What we can do ? Are our alsa results due to misconfigurations ?

19 years, 3 months

[linux-audio-dev] Alpha release of Aliki

by Fons Adriaensen

For the brave: an alpha release of Aliki (Room Impulse Response Measurement) is now available on: http://users.skynet.be/solaris/linuxaudio along with a manual that should get you started. This is basically the code used at the LAC2006 workshop, cleaned up a but. As said, ALPHA, incomplete, probably lots of bugs. -- FA Lascia la spina, cogli la rosa.

19 years, 3 months

[linux-audio-dev] [ANN] Vmwaredspjack 1.3, Snd-ls 0.9.7.6 and Jack_capture V0.3.8

by Kjetil S. Matheussen

Vmwaredspjack ============= This is the vmwaredsp program, made by Petr Vandrovec, which makes vmware work with esd or arts. This version adds jack support as well. (Unfortunately, jacklaunch (which is a similar program) doesn't work with vmware, but I think Gunter is working on it... :-) .) The program isn't always working that well, but if used with care (don't trust the output too much) and proper tuning, you can use professional windows audio software in vmware using jack for audio communication. Download from http://www.notam02.no/arkiv/src/ Snd-ls v0.9.7.6 =============== Snd-ls is a distribution of Bill Schottstaedt's sound editor SND. Its target is people that don't know scheme very well, and don't want to spend too much time configuring Snd. It can also serve as a quick introduction to Snd and how it can be set up. Changes 0.9.7.1 -> 0.9.7.6: --------------------------- -Proper debug output in case startup fails. -Fixed bug in jack audio. -Temporary remove the fft menu because its not working with the 26.9.2006 version of Snd. Bug found by Dragan Noveski. -Check for the existence of the sndfile.h header file before compiling. If it doesn't exist, snd-ls will refuse to run. Problem reported by Krzusztof Gawlas. -Make sure snd starts up even if no file was loaded during startup. Bug found by Dragan Noveski. -Really apply the workaround for the menu problem. Download from http://www.notam02.no/arkiv/src/snd/ Jack_capture V0.3.8 =================== jack_capture is a small program to capture whatever sound is going out to your speakers into a file. This is the program I always wanted to have for jack, but no one made. So here it is. Changes 0.3.7 -> 0.3.8: ----------------------- *Added the --recording-time option to stop recording after a certain number of seconds. *Quitting with CTRL-C/SIGINT writes remaining buffer to disk before ending program. Download from http://www.notam02.no/arkiv/src/

19 years, 3 months

[linux-audio-dev] Re: Alpha release of Aliki

by Ben Loftis

Hi Fons. Can you please tell us more about Aliki? I am using Denis Sbragion's DRC program to generate an impulse response file, and his suite of graphing tools to generate various views of the measurement. What is the output of Aliki? Just an impulse response file? In what way is your method different than DRC? Does Aliki use ALSA or JACK for I/O? I'll have a look at the code tonight but I was hoping you could fill me in on the "big picture". Thanks, Ben Loftis

19 years, 3 months

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