Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

3 participants
7520 discussions

by Phil CM

Hello LADs I'm trying to make a GUI for a LV2 synth based on so-404. In fact, I'm trying to learn C / C++. I got everything to build OK, but I'm stuck on a hard to read execution error ; when I load the plugin in jalv, it loads it OK, but fails to load the UI and says : suil error: Unable to open UI library /usr/local/lib/lv2/kis.lv2/kis_gui.so (/usr/local/lib/lv2/kis.lv2/kis_gui.so: undefined symbol: _Z17instantiateSO_404PK15_LV2_DescriptordPKcPKPK12_LV2_Feature) (The architecture is based on https://github.com/harryhaaren/lv2/tree/master/plugins/eg-sinsynth.lv2 in fact it is exactly the same : synth.c/h, synth_gui.cpp, widget.cpp/h, built as synth.so synth_gui.so synth.ttl and manifest.ttl using waf). I must say I'm still to fully understand the concept of descriptors... Can somebody see what I'm doing wrong? Thanks --Philippe "xaccrocheur" Coatmeur

11 years, 2 months

Re: [LAD] suil error: Unable to open UI library

by Phil CM

I'm almost there: ⚡ jalv.gtk https://bitbucket.org/xaccrocheur/kis Plugin: https://bitbucket.org/xaccrocheur/kis UI: https://bitbucket.org/xaccrocheur/kis#gui UI Type: http://lv2plug.in/ns/extensions/ui#X11UI JACK Name: Kis Block length: 512 frames MIDI buffers: 32768 bytes Comm buffers: 524288 bytes Update rate: 25.0 Hz SO-404 v.1.2 by 50m30n3 2009-2011 controlmode = 1.000000 volume = 50.000000 cutoff = 50.000000 resonance = 100.000000 envelope = 80.000000 portamento = 64.000000 release = 100.000000 channel = 1.000000 frequency = 440.000000 suil error: Failed to find descriptor for <https://bitbucket.org/xaccrocheur/kis#gui> in /usr/local/lib/lv2/kis.lv2/kis_gui.so Please, give me a pointer to a basic implementation of that descriptor thing, somebody, please end my misery. --Phil On 15/10/14 20:17, Paul Davis wrote: > > > On Wed, Oct 15, 2014 at 2:24 PM, Phil CM <philcm(a)gnu.org > <mailto:philcm@gnu.org>> wrote: > > Hello LADs > > I'm trying to make a GUI for a LV2 synth based on so-404. In fact, > I'm trying to learn C / C++. > > I got everything to build OK, but I'm stuck on a hard to read > execution error ; when I load the plugin in jalv, it loads it OK, > but fails to load the UI and says : > > suil error: Unable to open UI library > /usr/local/lib/lv2/kis.lv2/kis_gui.so > (/usr/local/lib/lv2/kis.lv2/kis_gui.so: undefined symbol: > _Z17instantiateSO_404PK15_LV2_DescriptordPKcPKPK12_LV2_Feature) > > > > % echo _Z17instantiateSO_404PK15_LV2_DescriptordPKcPKPK12_LV2_Feature > | c++filt > instantiateSO_404(_LV2_Descriptor const*, double, char const*, > _LV2_Feature const* const*) > % > > your plugin is missing a symbol (instantiateSO_404(...)), apparently. > > > > _______________________________________________ > Linux-audio-dev mailing list > Linux-audio-dev(a)lists.linuxaudio.org > http://lists.linuxaudio.org/listinfo/linux-audio-dev

11 years, 2 months

[OT] Duplicate messages originating from dspam@snarchi.io

by Fons Adriaensen

Hello all, Since some time I do get duplicate messages on this list for example the last one just a few minutes ago: > Date: Sun, 5 Oct 2014 23:35:21 +0200 > From dspam(a)snarchi.io Sun Oct 5 21:35:55 2014 > From: tom(a)trellis.ch > To: Fons Adriaensen <fons(a)linuxaudio.org> > Cc: linux-audio-dev(a)lists.linuxaudio.org This seems to happen only to messages which are a reply to one I sent. The 'dspam(a)snarchi.io' address seems to be related to 'marcochapeau', a name I've seen before on this list. I just wonder if I'm the only one getting these. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow)

11 years, 2 months

build problem with ambix plugins + LV2

by Jörn Nettingsmeier

Hi Matthias, I'm trying to build your ambix plugin suite on Linux using the LV2 wrapper, which fails with the following problem during makefile generation: CMake Error at CMakeLists_subprojects.txt.inc:104 (ADD_LIBRARY): Cannot find source file: /local/build/ambix/JUCE/modules/juce_audio_plugin_client/LV2/juce_LV2_Wrapper.cpp Tried extensions .c .C .c++ .cc .cpp .cxx .m .M .mm .h .hh .h++ .hm .hpp .hxx .in .txx Call Stack (most recent call first): ambix_binaural/ambix_binaural/CMakeLists.txt:20 (INCLUDE) It seems that the JUCE tree in your repo does not have an LV2 directory at all, and neither has the official JUCE repo - has it been dropped? Any hints much appreciated, best greetings from Essen to Vilnius, Jörn -- Jörn Nettingsmeier Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 Meister für Veranstaltungstechnik (Bühne/Studio) Tonmeister VDT http://stackingdwarves.net

11 years, 2 months

Advanced Gtk+ Sequencer API online

by Joël Krähemann

Hi all Please take a look at http://ags.sf.net/api/ags and let me know what you're missing. There may follow some corrections if you find mistakes let me know, too. At the moment there probably plenty of them. kind regards Joël

11 years, 2 months

Linux audio jobs/companies?

by Bill Gribble

It's job hunting time, and while I am skeptical that there are very many paying jobs for Linux audio developers out there, I thought I'd at least ask on this list. What companies are doing work in the Linux audio universe? Or, on the other side, anybody know of interesting music-instrument/audio companies located in or around NYC (where I live) who might need Linux folks for their web presence or other online stuff? Any tips appreciated! Thanks, Bill Gribble

11 years, 2 months

media clock from wall clock

by Len Ovens

I have been playing with numbers (a little) for AoIP and AES67 (because I can actually read the spec). The media clock (AKA wordclock) is derived from the wall clock which is synced via PTP. There are 3 sample rates supported 44.1k (not sure if any physical devices really do), 48k and 96k. The first thing I find is that it is not possible to get even word clock via simple math. The wall clock moves one tick per usec which at 48K is 20.833rep. (44.1k is a mess) I would suggest this is why AVB and AES67 at lowest latency already uses 6 sample frames which is a nice even 125 usec. It is obvious to me from this, that the true media clock is derived by external circuitry. The one "must support format" is 48k at 48 samples/packet. (both 16 and 24bit on the receive end of a stream) This is 1ms packet time. (there are AoIP devices that only support AES67 with this format) So I am guessing that it must be easy to derive a media clock from a 1ms clock input. The computer (CPU board) would have a GPIO with a 1ms clock out derived from the cpu wall clock which in turn is kept in sync with the network via PTP. I am guessing there is already a chip in production that does this with almost no support circuitry. The side effect of the 1ms/packet standard is that streams are limited to 8 channels at 48K and 4 channels at 96k. So for more channels more streams are required. This maximum comes from the limit that the full packet must fit into one standard ethernet packet with the MTU at 1500 (minus headers etc.). The transport does not do packet reassembly of fragmented packets and basically drops them. 1 ms latency in the jack world is the same as 16/3. This may explain why /3 works better for other formats such as USB or FW. It may also be the bridge that works best with OPUS at 5 ms. -- Len Ovens www.ovenwerks.net

11 years, 2 months

Lars Luthman

by Will Godfrey

Is anyone in contact with Lars? I tried to send him an email but it immediately bounced, so i think I've got an address that's no longer valid. -- Will J Godfrey http://www.musically.me.uk Say you have a poem and I have a tune. Exchange them and we can both have a poem, a tune, and a song.

11 years, 2 months

zitaretuner

by Fons Adriaensen

Hello all, Today I got an email of a user asking me to help him make a plugin called 'zitaretuner' work. I never wrote such a plugin, and I didn't even know it existed. So I can't help this user. Of course this made me curious, and I managed to get a copy of the source code of this lv2 plugin. And I wasn't very amused. As expected it's based on zita-at1, and again a complete disaster. The DSP code of zita-at1 is written as a neat self-contained C++ class with a very clean interface, and this is done explicitly to make it re-usable. But instead of re-using it, the author of the plugin decided to rewrite it in C, and combine it in the same source file with parts of libzitaresampler (instead of using that as a library as it is meant to), and with whatever is required to turn it into an lv2. The whole thing is just a single source file. The same author (who is know only as 'jg') didn't bother to add a decent GUI, relying on the plugin host to create one. That means for example that the note selection buttons (which also double as 'current note' indicators in zita-at1), are replaced by faders. Only $GOD knows what they are supposed to control. And as a final topping on the cake, that whole crappy thing is presented as if I were the author of it all. No mention at all that things have been modified, and by whom or why. This alone is a clear violation of the license under which zita-at1 was released. And whoever did it doesn't even have the courage to identify him/herself. I've complained about this sort of thing before, and this time I'm really pissed. So let one thing be clear: I will never again release any code under a license that allows this sort of thing to happen. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow)

11 years, 2 months

AoIP question

by Len Ovens

I have been doing some reading on audio over IP (or networking of any kind) and one of the things that comes up from time to time is collisions. Anything I read about ethernet talks about collisions and how to deal with them. When I was thinking of a point to point layer two setup, my first thought was there should be no collisions. Having read all the AES67 and other layer 3 protocols there does not seem to be mention of collisions really. My thought is that on a modern wired network there should be no collisions at all. The closest thing would be a delay at a switch while it transmitted another packet that in a hub would have been a collision. So my thought is that AoIP at low latencies depends on a local net with no collision possible. Am I making sense? or am I missing something? The various documents do talk about three kinds of switches, home, enterprise and AVB. It is quite clear what makes an AVB switch, but what does an "enterprise" switch have over any other switch aside from speed? I am sure I am being small minded in my thoughts here. For example, I am expecting very little non-audio bandwidth and I am guessing that the average home switch does not prioritize any style of packet over another. I guess I am asking what parts of a switcher are important for audio? I am guessing that HW encription (offloading the SSL from the server to the switch) is not something that helps audio. Even with proprietary protocols, it would not make sense to use encription. There seems to be a real plug and play emphasis where the audio enabled LAN is firewalled from the rest of the world. The streams are multicast so any box that recognizes the packets can use those channels and offer it's own audio channels. Any box can send control messages to reroute audio... there must be some kind of authentication for that.... (light bulb) this is why AES67 does not include controls and perhaps discovery either. I can just see someone walking into a concert with a notepad and deciding they want their own mix... I am guessing the promoter would not want people walking away with their own "direct from the mixer" audio track of the event either. Not quite plug and play then. The wireless router wants to limit what traffic it deals with. So AES67 allows the control to be offloaded to a web interface with access control (switch SSL offloading would not help here anyway). Discovery ends up being manual at first glance. I am thinking it will not be long before the control and setup will automate loging into the web IF and setting things up. A system wide user/password and an interface that defaults to streaming it's own inputs as multicast is already discoverable. -- Len Ovens www.ovenwerks.net

11 years, 2 months

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Linux-audio-dev