Linux-audio-dev February 2005

linux-audio-dev@lists.linuxaudio.org

78 participants
65 discussions

by Jens M Andreasen

Hi! I Noticed that the Aureal drivers are now in Alsa. I used to be an early adopter (with bad experience ...) Anybody around here care to share some modern succes stories? -- ( ) c[] // Jens M Andreasen

19 years, 10 months

[linux-audio-dev] [ANN] QjackCtl 0.2.15a (fix) released!

by Rui Nuno Capela

Hi, QjackCtl 0.2.15a has been released! This is just a fix release, update highly recommended. As from the change log: - Regression from 0.2.13, of the not so stupid pseudo-mutex guards on the connections management framework, after fixing some crash reports from Fernando Pablo Lopez-Lezcano and Dave Phillips (thanks!); it pays to be such a paranoid after all :). Check it out from the usual place: http://qjackctl.sourceforge.net Enjoy. -- rncbc aka Rui Nuno Capela rncbc(a)rncbc.org

19 years, 10 months

[linux-audio-dev] [OT] How to attend LAC 2005?

by Bengt Gördén

This is sort of really of topic. I'm not a professional developer of music software or something (I do have some ideas though) but I would very much like to attend the conference. The problem is that private I can't afford the trip and stay. Now, I have a nice boss that says that if I could find something in the neighborhood that is close to my normal day job (network engineer) he is willing to pay for the trip as long as he gets something out of the trip. Is there anyone here that knows anything close to networking at LAC 2005 or is there something in Karlsruhe that I could have as a reason for traveling there. regards, /bengan

19 years, 10 months

[linux-audio-dev] about LAC 2005 paper template

by Juhana Sadeharju

I compared the LaTeX paper templates of LAC 2005 and DAFX 2004, and the output screenshots are available at: ftp://ftp.funet.fi/pub/sci/audio/devel/dsp/papertemplates/ It looks to me that DAFX template is better, both in the text quality and the example text. Should we switch? They say DAFX templates are the same what are used in IEEE audio conferences. The text is smaller but I'm sure it is tuned for readability. http://dafx04.na.infn.it/Tmpl_LaTeX.zip http://dafx04.na.infn.it/Tmpl_Word.zip http://www.zkm.de/lac/templates-lac2005.tgz Juhana -- http://music.columbia.edu/mailman/listinfo/linux-graphics-dev for developers of open source graphics software

19 years, 10 months

[linux-audio-dev] Re: [OT] RFA

by Juhana Sadeharju

>> technical operations website: http://techweb.rfa.org/. techweb also [ ... ] >> I see an opportunity here for future use of numerous >> linux audio applications in a working digital audio production facility >> that puts out 36+ hours of programming each day. I would be honored and >> excited if my new boss were to come from the linux audio community. OK. Here is a project suggestion: Remember the old days when Spectrum and Commodore computer software was broadcasted through radio to computer users? This system could be brought to this day using latest tech in coding. This would make it possible to make forbidden webpages available to countries who restrict the access. People would need a simple and cheap radio receiver, and plug it to the audio card. The signal coding would differ from communication coding systems in that sample rate is limited to 44100 Hz and that audio has more signal level resolution. I invented this "Radio Internet" several years ago because Internet costed (and still costs) too much and because the transfer direction is mostly from Internet to home for personal users. The idea of using this tech to broadcast Internet to China became a few months ago after BBC World's Click Online show reported about the Internet retrictions in China. With non-cheap radio tech the transfer rate would not be limited to the audio card properties, but simple and cheap might be better when it comes to the freedom spreading. Juhana -- http://music.columbia.edu/mailman/listinfo/linux-graphics-dev for developers of open source graphics software

19 years, 10 months

[linux-audio-dev] the bandwidth of the each harmonic

by Paul

...a simple, but powerfull paradigm that helps to make simple algorithms that produces very beautiful sounds. Hi. I want to open a topic about what I call "bandwidth of each harmonic", to say some ideas that helped me to make beautifull sounds. I am Paul, the author of ZynAddSubFX software synthsizer ( http://zynaddsubfx.sourceforge.net ) and I wish to share some ideas ;) Referring to musical sounds,usually the harmonics are considered simple sine function. Of course, the reality shows different thing, and so, the harmonics was started to be considered as sine functions modulated by a lower frequencies (I saw some time ago a link that describes that the choir sound is beautifull because of the "micromodulations",etc...). A very good thing is to "have a look" in a frequency domains. Let's take for example a choir that sing same note (like A=440Hz). Because all are human, even they are very well trained singer, they will not sing exactely the same note; for example one will sing at 435 Hz or other, at 443 Hz, and so on. Now, the first harmonic (the fundamental note) is not longer a sine of 440 Hz, but a narrow band signal that has a certain bandwith. For example, let's take a very simple case: if all sing at same loudness, and they sing from 435 Hz and the 445 Hz. In this case, the bandwidth will be 10 Hz. Of course, in real choir the frequency distribution of the harmonic will be not flat, but usually will be a curve that looks like a normal (gaussian) curve(found this after I did some research= very fine frequency analysis with very long FFTs). Now, let's go to the second harmonic: if you multiply 435 and 445 with 2, the difference will be 20 Hz. So, the bandwidth of the second harmonic is 20 Hz. So here a important rule of real instruments (esp. ensembles): the bandwith of each harmonic is proportional to the it's frequency; so in this example the first harmonic has bandwidth 10Hz, the second 20Hz, the third 30Hz, and so on. Here I made a very fine frequency analisys of a synthsized sound(a real orchestra sound will give similar results): [img]http://zynaddsubfx.sourceforge.net/doc/paul1.png[/img] You see, that the harmonic's bandwith increase according to their frequency. If you don't increase the bandwidth of higher harmonics, the resulting sound will be unpleasant especially on higher bandwiths of the first harmonic. So, not all quasi-periodic sounds are good, usually only if you increase the bandwidth of the harmonics. Now, what's happens if there are a lot of harmonics or the pitch of the sound is enough low? Let's see: [img]http://zynaddsubfx.sourceforge.net/doc/paul2.png[/img] The upper harmonics will merge to a single frequency band that will sound like a hiss, that is pleasent to the ear(eg. choir). I found that bandwidth of each harmonics can go even greater than 50 cents (quater of a tone) and still sound musical - I can give you some example wavs. Please note that this a different situation than 2 notes detuned by 50 cents (that sound very dissonand). Also, the best is that the phases are random. In real life this is ensured by the reverberation of the hall, for example if a instrument plays with a vibratto (a flute). Perhaps you noticed that always if more instruments plays the same note(as a enseble), the sound is very pleasant. I consider that this is the cause: the bandwidth of each harmonic. Unfortunately, there is very little on the internet about this stuff, because, I think, that are used more complicated paradigms (like some statistics on how sine harmonics modulates, or so). Now I want to give examples on how you can synthesize sounds, using this ideea. You can use zynaddsubfx or another synthesizer in most cases. Of course: I don't claim that I invented the bandwidth of each harmonic (or frequency distribution of each harmonic), because the ensembles, the choirs, the reverberations, exists since thousands of years (or more :) ). I just consider this a very important fact on how sounds are beautifull and it helped me to synthsize good sounds even since I was on highschool :D. You can make bandwidth of each harmonics by: 1) make more oscillators and detune them a bit (and a slight vibratto helps alot). Most synth use this and allows this, and perhaps it is known by you. This is one of the simplest method. This is implemented in ZynAddSubFX as "ADDsynth" module 2) generate white noise and filter each harmonics with a bandpass filter and mix the results. Be carefull to make higher harmonics to have higher bandwidth. This is implememnted in zynaddsubfx as "SUBsynth" module. 3) You see that above graphs. You can represent them as numbers in the frequency domain that represent the amplitudes of the frequencies, add random phases and do a SINGLE IFFT and voila! A very beautifull sound will "born". This new ideea is implemented in ZynAddSubFX as "PADsynth" 4) you can do other things, like a vibratto on a periodic oscillator and do a FFT to all sound, put random phases, and do a IFFT. For more information, I made few years ago a page that describes above things at: http://zynaddsubfx.sourceforge.net/doc_0.html Good luck. Paul __________________________________ Do you Yahoo!? The all-new My Yahoo! - What will yours do? http://my.yahoo.com

19 years, 10 months

Re: [linux-audio-dev] LADSPA stereo-expander

by Stefan Turner

> Hi! > > Is there some kind of LADSPA plugin with stereo base > expansion > capabilities? > > Thanks! > Andrew > Is this what you mean: http://www.plugin.org.uk/ladspa-swh/docs/ladspa-swh.html#id1422 Stefan Turner ___________________________________________________________ ALL-NEW Yahoo! Messenger - all new features - even more fun! http://uk.messenger.yahoo.com

19 years, 11 months

[linux-audio-dev] [ANN] QjackCtl 0.2.15 released!

by Rui Nuno Capela

Hi, QjackCtl 0.2.15 has been released. As a major new feature you are now allowed to rename (alias) the JACK/ALSA connections client/port names to something intelligible. Another nice one is about actual ALSA hardware device names which are now presented for selection as a pull-down menu on the setup dialog. Grab it from: http://qjackctl.sourceforge.net As taken from the change log: - JACK/ALSA client and port name aliasing (renaming) is now an optional feature for the connections window; all client/port aliases are saved on a per preset basis (as proposed for Lionstracs' Mediastation). - Server state now shown (back gain) on the system tray icon tooltip; speaking of which, tooltips are now also featured on connections, status and patchbay windows. - New actual hardware device selection menu featured on setup dialog; these new button menus are only available for the ALSA driver settings. - Server path factory default to jackd instead of jackstart; preset setup button icons are back. - Fixed rare connection port item removal/disconnection dangling pointer bug. Have fun. -- rncbc aka Rui Nuno Capela rncbc(a)rncbc.org

19 years, 11 months

[linux-audio-dev] LADSPA stereo-expander

by Andrew Gaydenko

Hi! Is there some kind of LADSPA plugin with stereo base expansion capabilities? Thanks! Andrew

19 years, 11 months

[linux-audio-dev] Trying to get midi input using OSS

by airplays55＠yahoo.com

I've been trying for months to get my midi input (using OSS/free) on a knoppix liveCD so that I can record midi in the included rosegarden2.1, and here is where I am now: I found a very short program that all it does is simply print out the bytes recieved at the midi input - its at http://ccrma.stanford.edu/~craig/articles/linuxmidi/input/section1.html My /dev/sequencer is working fine for output (verified with playmidi -e) however this program does not display any midi input. Actually what happens is BEFORE running the program, I verify that my Roland sound module (connected to the midi output of the midi cable plugged into the soundcard) DOES recieve input from the midi keyboard plugged into the midi input, BUT then when I run the program, the midi input is no longer passed through to the midi output. I've verified that the test program displays no midi input using /dev/sequencer, or /dev/midi, /dev/midi00 (I changed the device in the code) Here are the only steps I followed which succeeded in getting the ESS1868 midi output working. modprobe sb io=0x220 irq=5 dma=1 dma16=5 mpu_io=0x330 # OR dma16=-1 modprobe -v mpu401 io=0x330 irq=7 modprobe -v opl3 io=0x388 I guess that if I just get this sample program to display midi input, then I will be able to start using rosegarden2.1 to record midi? But why would the test program use /dev/sequencer for input (I thought /dev/midi was the input?) Why would my midi output (/dev/sequencer) but my midi input wouldn't work? here is the relevant output of sndstat: Synth devices: 0: Yamaha OPL3 MIDI devices: 0: MPU-401 0.0 Midi interface #1 Any advice would be VERY appreciated! __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

19 years, 11 months

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

Linux-audio-dev February 2005