Linux-audio-dev January 2006

linux-audio-dev@lists.linuxaudio.org

83 participants
66 discussions

by Paul Davis

Screenshot: http://www.kvraudio.com/forum/viewtopic.php? t=114488&postdays=0&postorder=asc&highlight=linux&start=105 i will be interested to see just how good its audio handling capabilities are. --p

18 years, 2 months

Re: [linux-audio-dev] xt2 coming to linux

by Chris Cannam

Lee Revell writes: > On Fri, 2006-01-27 at 15:57 +0100, Michael Bohle wrote: > > But anyway, VST on Linux is dead now, beause > > most of the user are not > > able to compile it for themself. > > Wrong. You just need to write a wrapper that > handles the compiling. Won't help if the code is to be part of a GPL'd application. Also, I think (?) you have to register to download the VSTSDK. Chris

18 years, 2 months

[linux-audio-dev] pthread_mutex_unlock

by Lee Revell

One Harold Chu on LKML is insisting that POSIX requires pthread_mutex_unlock to reschedule if other threads are waiting on the mutex, and that even if the calling thread immediately tries to lock the mutex again another thread must get it. I contend that both of these assertions are wrong - first, I just don't read the standard that way, and second, it would lead to obviously incorrect behavior - unlocking a mutex would no longer be an RT safe operation. What would be the point of trylock() in RT code if unlocking is going to cause a reschedule anyway? Can anyone back me up on this? Lee

18 years, 2 months

[linux-audio-dev] little question regarding Ardour

by zen95156＠zen.co.uk

Hello I started a session in Ardour - drag and dropped a .wav file, then recorded (sucessfully) from a Jack input. Checking the session folder, Ardour appears to record to disc as it's going along (on-the-fly). I presume Ardour can also route the incoming sound to its outputs, for monitoring. Does anyone know what mechanism Ardour uses to do this? (I'll walk blindly into speculation. Ardour uses a jack ringbuffer on its Jack input port, in the approved way - but that only leaves one read on the ringbuffer output...) Robert

18 years, 2 months

[linux-audio-dev] liblo method handler lookup logic

by David Olofson

Hacked some test code here and discovered something "interesting" with lo_server_add_method() and method handling. If I try to do it like in the examples and add the default/debug ("match all") method first, it gets called for every incoming message, before the real handler is called. That is, I see an error message - and then the correct method handler is invoked anyway. Looking quickly at the code, I'd actually expect this behavior, as lo_server_add_method() does indeed add methods at the end of the list. Trying the examples again, I realize that they demonstrate this behavior as well, so maybe it's not just my code doing something funny. :-) This is with liblo 0.22 on Gentoo/AMD64. //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, 2 months

[linux-audio-dev] libsamplerate question

by David

Hello everybody ! I've just added a resampling function to my code thanks to the excellent work of Erik de Castro Lopo (thanks a lot !). Combined with libsndfile (thanks again) it is really easy to load any sound file I want. But I'd like to make sure I'm using it correctly. I process my input data in one pass using src_simple() and I have to compute the length of the output data buffer beforehand. So I did somehting like this : out_len = (long int) ceil((double) in_len * ratio); It seems that my output buffer is always one frame too big (I checked this by reading the output_frames_gen field of the SRC_DATA structure after the processing is done). Is it safe to assume that using floor() instead of ceil() will not lead to a too short output buffer in some cases ? I can live with wasting a malloced float but I'd like to be sure it cannot be done in a prettier way. Thanks. -- David

18 years, 2 months

[linux-audio-dev] [ANN] WhySynth DSSI plugin 20060122 release

by Sean Bolton

Announcing the 20060122 release of WhySynth, a DSSI softsynth plugin. New since the last major release: * A new oscillator mode, based on Nasca O. Paul's gorgeous PADsynth algorithm. * A new filter mode, essentially the low-pass filter from amSynth. * A new dual delay effect. * Improved and extended wavetables. * More patches. * Lots of cleanups and bug fixes, including fixes for more stable operation especially under Rosegarden, and for compilation on Mac OS X 10.4 'Tiger'. Find WhySynth here: http://home.jps.net/~musound/whysynth.html More information on the DSSI plugin standard, available hosts and plugins can be found here: http://dssi.sourceforge.net/ WhySynth is written and copyright (c) 2006 by Sean Bolton, under the GNU General Public License, version 2.

18 years, 2 months

[linux-audio-dev] Additional chunks in WAV files with libsndfile ?

by fons adriaensen

Hi all, is there a recommended way to write / read additional chunks in WAV files, using libsndfile (assuming it's possible at all - I didn't find any hints to this in the docs) ? What I need in particular is some way to calibrate the time axis - i.e. to say frame #N corresponds to t = 0, and some other similar info, mostly sample indices. TIA, -- FA

18 years, 2 months

[linux-audio-dev] MIDI library

by Dominique Michel

Hi all, I am new to this list, living in Switzerland and working mainly with electronic. I have done it was some time ago a monophonic realtime note recognition software on an embeded dsp56k system and I want to adapt it to linux. It will take some time to me to do that because I know almost nothing about c and c++ programming, as about the alsa and jack libraries. But I know my algorytm, and it is working just fine on my dsp. The latency is very low, 1+(~1/4) period of the sound, the (~1/4) term depend of the harmonic content of the sound. I believe at it is worth to do a jack-alsa software with it. A gui will be needed, giving the possibility to change, save and recall some parameters of the recognition loop on an per instrument basis. Can you recommand me a good, and if possible simple to use and fast, MIDI library? I will use only basic functionalities as the possibility to send MIDI notes to the sound server. Ciao, Dominique

18 years, 2 months

[linux-audio-dev] Using UTF-8

by Richard Smith

A while ago the list moderator requested that all of us gmail users "fix" our mail by setting the UTF-8 option. Which I did. However, I'm now on another mailing list on Sourceforge where the utf-8 thing causes junk to be added after my signature. This is some sort of bug in sorceforge's base64 handleing. I've been asked by why I had it set that way. So I was just wondering what the issue with non- utf8 mail was on lad. -- Richard A. Smith

18 years, 3 months

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Linux-audio-dev January 2006