Linux-audio-dev July 2006

linux-audio-dev@lists.linuxaudio.org

76 participants
53 discussions

[linux-audio-dev] [ANN] New specimen list and home

by Eric Dantan Rzewnicki

Hello all, I've set up a mailing list for specimen: http://zhevny.com/mailman/listinfo/specimen Additionally the web site now lives at http://zhevny.com/specimen. This is a minimal re-working of Pete's old site. More improvements to follow. Let me know if anything is badly broken. (I know the guide is broken, but I need to re-do it with new screenshots anyway.) I just changed the dns record for zhevny.com to point to a new host earlier today. The TTL was set to 1 day, so some of you may get pointed to the old box. Let me know if you have any issues, or just try again a little later. Thanks, Eric Rz.

17 years, 9 months

[linux-audio-dev] Re: Nomad - Nord Modular Editor

by Juhana Sadeharju

[ in linux-audio-users, originally in nmedit-devel ] >our first release of Nomad - Nord Modular Editor is now available. It is at http://nmedit.sourceforge.net if that was not mentioned. The Nomad has also a UI builder for building UIs for modules. Check the screenshots and "nomad-ui-editor.jpg". Do we have anything similar? E.g. a collection of audio related widgets for Glade? How Nomad's UI builder could be re-used in other projects? By using it like VSTGUI? By building module UIs for Csound opcodes and writing Csound exporter? I have earlier suggested that Nord Modular modules could be written as Csound instruments, i.e., Nomad + Exporter + Csound == NM clone. One may find several thousands of NM patches from the web. Many of them are advanced and well documented. http://nm-archives.electro-music.com/010_NordModular/014_Interesting_Thread… Juhana -- http://music.columbia.edu/mailman/listinfo/linux-graphics-dev for developers of open source graphics software

17 years, 9 months

[linux-audio-dev] Re: diskstream and jack

by Juhana Sadeharju

>From: conrad berhrster <conrad.berhoerster(a)gmx.de> > >Now the question: >- The worker thread is faster then the jackthread. (Sure, it should be). What >is the usual way to pause the workerthread and wake up again, when the >ringbuffer needs more data. My alsashmrec uses kill((pid_t)diskpid,SIGSTOP); (in disk writer process) and kill((pid_t)diskpid,SIGCONT); (in A/D reader process) Perhaps that could be replaced with semaphore but what if multiple processes uses the same disk service process? That increases the number of SIGCONTs but otherwise the processes would be uncoupled. How it is with semaphores? >- How should the ringbuffer be initilized when starting to play? Fill it up if its size was selected optimally. If you have multiple songs and you don't know which the user would like to play next, then make one ringbuffer per song and fill them all up. Then the playing can start instantly when user presses the song button. Juhana -- http://music.columbia.edu/mailman/listinfo/linux-graphics-dev for developers of open source graphics software

17 years, 9 months

[linux-audio-dev] Improved decoder plugins for Linux

by Fons Adriaensen

Hello all, I just updated the small set of Ambisionics related LADSPA plugins available at <users.skynet.be/solaris/linuxaudio>. The three decoders now feature optional phase aligned shelf filters and independent control for LF and HF gain of the velocity components (from in-phase over max rE to max rV). The distance compensation is still there of course (and correctly calibrated !). -- FA Follie! Follie! Delirio vano e' questo!

17 years, 9 months

[linux-audio-dev] Re: Converting a 24bit sample to 16bit

by Juhana Sadeharju

>From: James Courtier-Dutton <James(a)superbug.co.uk> > >Is there a standard way of converting a 24bit sample to 16bit? How your 24bit audio was made? If your audio card gives out 24-bit, then you may loose bits because there are no true 24-bit converters. At top you could use a smooth hard-limiter (which processes each sample separately). It takes out random high peaks and makes the audio more analogue. At bottom the dither noise may not be necessary if the audio source's noise floor is high enough. All dither tests seems to test against the mathematical sine wave which you don't get if you do acoustic music. Noiseshaping dither is suspectible as well because one may then wonder where the high frequency noise introduced by the dither goes to. No amplifier near me have a dither noise remover. Juhana -- http://music.columbia.edu/mailman/listinfo/linux-graphics-dev for developers of open source graphics software

17 years, 9 months

[linux-audio-dev] MIDI... know how to do bank/patch changes on yamaha Motif Rack?

by Stephen Cameron

So, I've got a yamaha motif rack, and the manual (which is here, btw: http://www2.yamaha.co.jp/manual/pdf/emi/english/synth/MOTIFRACKE2.pdf ) On P. 46 of that manual describes how, and it seems like it should be brain dead simple. There's a basic recieve channel on the Motif Rack, (it claims that it is set to "1" -- I don't know if that is zero or one based, so I tried both. I think, if I send a byte sequence like: 0xb1 0xf3 0x03 0xc1 0x07 (or maybe substitute 0 for any 1 in the above, I tried them all) It ought to do something. I *can* change patches within a bank, that is the Sending 0xc0 0x25 will change the current preset. Just can't figure out the bank change thing. Well, I'm retarded, or the spec is wrong, or the motif is broken... I think I must be retarded. At the moment, I've reduced things to a few lines of C that just open up /dev/snd/midiC1D0 and write some bytes out -- I know that much works because note-on and patch change work. Just wondering if anybody else out there has a Motif and if they've ever gotten it to change banks. -- steve __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

17 years, 9 months

[linux-audio-dev] Hello and Python bindings for LADSPA

by Jono Bacon

Hi all, My name is Jono Bacon (www.jonobacon.org) and I am one of the developers working on Jokosher, a GStreamer based multi-track audio editor written in Python for the GNOME desktop. Jokosher is designed to be really simple to use, and we release our first 0.1 release on Saturday. You can find out more at www.jokosher.org. although the brand new, much improved site is released on Saturday too. I am also one of the members of LUGRadio. Anyway, for Jokosher 0.2 I want to include LADSPA support. From what I understand, LADSPA not only provides an audio layer, but also a GUI layer, so the widgets in the plug-in can be displayed in the host toolkit. I know there is an antiquated GStreamer bridge for LADSPA, but in terms of getting the GUI elements to display, I assume I need Python bindings. Do such bindings exist? Oh, and by the way, if you are in the UK, we are running an event called LUGRadio Live 2006 with over 40 speakers, a room full of exhibitors, lots of BOF sessions and a party. It is held in Wolverhampton on the 22nd and 23rd July (next weekend). It would be great to see a fraternity of Linux audio developers there. Anyone want to come along? See www.lugradio.org/live/2006 for more details. Thanks folks! Jono

17 years, 9 months

[linux-audio-dev] LV2 Turtle requirement

by carmen

i notice this file: http://lv2plug.in/spec/lv2.ttl is in the nicely readable turtle format. my main question is, whether this will be transformed to RDF-XML during 'make install' or perhaps by the developer themselves (Eg, similar to leaving a configure script around for those who dont have autoconf). mainly beacuse, it seems RDF-XML is the dominant serialization format, and having this additional dependency stand in the way means, at the very least one can't use Pyrple out of the box. i installed Redland and the Ruby and TCL bindings, and both conspicuously left out the "Redland::Parser.turtle" method, presumably beacuse i dont have it installed (and it's not in portage...) cheers

17 years, 9 months

[linux-audio-dev] [ANN] netjack-0.12 - Low Latency Network Audio Driver

by torbenh＠gmx.de

hi. netjack-0.12 is released. Netjack is a jack-driver which uses the network card. On the other end of the network there is a normal jack-client. So its possible, to share a single soundcard between several laptops. This release finally handles the packet disordering UDP does. Thus high channel counts can now be achieved. However a 24ch in/out link over 100Mbit gave me a major "net xrun" storm on vanilla 2.6.15 kernel. At a roundtrip latency of 2.9ms that is. It was reliable with 5.8ms. 16 channels gave me some "net xruns", which i could not hear though. i expect this performance to increase when using an rt-kernel with the network-irq set to rt-prio. So please report back. Additionally to the audio transport, netjack provides sample accurate transport syncronisation. The roundtrip latency is compensated for. get it while its hot at: http://sourceforge.net/project/showfiles.php?group_id=140191 there is no link on http://netjack.sf.net because the project shell servers are down. -- torben Hohn http://galan.sourceforge.net -- The graphical Audio language

17 years, 9 months

[linux-audio-dev] diskstream and jack

by conrad berhörster

Hello all, i try to implement the reading of audiofiles with ringbuffers. For this, i need two threads: a main process thread , which is triggered from jack every 1024 samples for example. This reads from a ringbuffer. And a worker thread which fills this ringbuffer. Now the question: - The worker thread is faster then the jackthread. (Sure, it should be). What is the usual way to pause the workerthread and wake up again, when the ringbuffer needs more data. -Which size should the ringbuffer have. - How should the ringbuffer be initilized when starting to play? - Any small program to have look into or a good inroduction into pthreads?. thanks c~

17 years, 9 months

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 July 2006