Linux-audio-dev

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7520 discussions

Re: [LAD] Plugin buffer size restrictions

by Jeff McClintock

> From: David Robillard <d(a)drobilla.net> > > I'm a modular head, I remain convinced that control ports are nothing > but a pain in the ass and CV for everything would be a wonderful > fantasy land :) It's called "SynthEdit land" *everything* is CV ;) (not on Linux sorry). > As it happens, I am currently porting the blop plugins to LV2, and > making a new extension in order to drop the many plugin variants (which > are a nightmare from the user POV). This simple extension lets you > switch a port from its default type (e.g. Control) to another type > (e.g. > CV). The pattern looks something like this: > > /* plugin->frequency_is_cv is 1 if a CV buffer, 0 if a single float */ > for (uint32_t i = 0; i < sample_count; ++i) { > const float freq = frequency[s * plugin->frequency_is_cv]; > if (freq != plugin->last_frequency) { > recalculate_something(freq); > plugin->last_frequency = freq; > } > > /* Do stuff */ > } That's smart. In a simple example this doesn't seem like much of a win. Because A 1 port plugin has only two possible variants (frequency as single-float/ buffer). But.. * A 2-port plugin has 4 varients. * A 3-port plugin has 8 varients. * A 10 port plugin has 1024 varients! So you're avoiding that combinatorial nightmare. I do something similar. The port is flagged as either 'streaming' (use the entire buffer) or 'static' use a single float. My point of difference is - the entire buffer is provided either way. So you have the option of writing the plugin like.. const float freq = frequency[s]; ..OR... const float freq = frequency[s * plugin->frequency_is_cv]; .. and it works transparently either way. So the extension is backward compatible with 'dumb' plugins, or 'dumb' plugin standards like VST (I can interface VST plugins with modular components). > Doing those comparisons to see if the value actually changed since the > last sample in order to recalculate is not so great (branching). I don't know if you can implement what I do. Once I know which ports are single floats I 'switch' processing functions. i.e. use a function pointer to select 1 of several optimised functions. So you write a general purpose loop like the one above, this is your fallback. Then you write an optimised one that assumes 'frequency' is a single float - This one has no branching and no extra multiplication, it's super efficient. You get the best of both worlds. Note I don't write loops optimised for every possible combination, just pick a few key ones. The function pointer is one extra level of indirection, but it's much faster than branching, esp when there's several ports involved in the decision. > personally my interest in a solution here is very real. More people > care about normal high level parameters and being able to interpolate > than low-level modular synth CV stuff, but to me it's telling that (it > seems...) one solution can solve both problems nicely. <high five> ;) Best Regards, Jeff

13 years, 7 months

Should LV2 Activate() clear MIDI CC parameters?

by Jeremy Salwen

Hi all, The LV2 spec says that on a call to activate(), "the plugin instance MUST reset all state information dependent on the history of the plugin instance except for any data locations provided by connect_port()" I am not certain whether MIDI CC parameters are included in this category of "data locations provided by connect_port()". The CC parameters are sent through port buffers provided by connect_port(), but because they are *event * buffers, all information passed through them is necessarily part of the * history* of the plugin instance. I could imagine cases where you would want to reset all internal state of the plugin, but since CC values are very much like port values, they would be kept. On the other hand, I could also imagine cases where you would want to reset all internal data including the CC parameters. I'm assuming MIDI note on/off status certainly should be reset. Thanks, Jeremy Salwen

13 years, 7 months

: Plugin buffer size restrictions

by Jeff McClintock

> I think providing synchronous control events, with 'future' values (at > least some distance L in the future) is the way to get that. Let's > pretend that the Ultimate Plugin Interface (UPI) 1.0 exists, works this > way, is stable and unmalleable, and all you have to work with to > deliver > your product (a plugin). > > >From the plugin author perspective: is there anything that is > *impossible* to do correctly? > > -dr I believe it simply impossible to reliable deliver 'future' parameter values. Even when the automation is pre-recorded. E.g. smoothly ramping up a parameter over 1 second. You can't say to the plugin 'ramp this parameter over 1 second' - because partway through the 'ramp' the user can reposition the playback to another part of the song, or loop a section, or change the tempo, or hit 'Stop'. Any attempt to predict the future like that leads to kludgy hacks. Now you can say to the plugin 'process 100 samples' while specifying the parameter value at sample# 0 and also at sample# 99. That is how to specify a precise ramp (or section of a longer ramp) without providing future parameter values. Apologies if that's what you meant. (I classify 'future' values in this example as being later than sample #99 ). Best Regards, Jeff

13 years, 7 months

Re: [LAD] Plugin buffer size restrictions

by Jeff McClintock

> From: Fons Adriaensen <fons(a)linuxaudio.org> > Subject: Re: [LAD] Plugin buffer size restrictions > On Mon, May 28, 2012 at 06:05:20PM +0300, Stefano D'Angelo wrote: > > > IMO it's easily said: if control rate < audio rate it's plugin's > > responsibility, otherwise the host feeds upsampled/filtered control > > signals at audio rate to the plugin and all problems evaporate... > > The don't evaporate, they explode. > > Take a filter plugin. Calculating the actual filter coefficients from > the 'user' parameters (frequency, gain, etc..) can easily be > 10..1000 times more complex than actually using those coefficients to > process one sample. So you really don't want to do that at the audio > sample rate. True, But with synths at least, recalculating the filter at audio rate is routine these days. Admittedly we are pre-calculating the full range of coefficients in advance. You do get *very* nice filter sweeps and blips at full audio-rate modulation. Best Regards, Jeff

13 years, 7 months

Fwd: Bug#674651: Please disable celt support in jack

by Adrian Knoth

Hi! I'm CC'ing this to LAD and jack-devel with reply-to set to jack-devel, hope the listservers keep the header fields intact. Though it reads jackd1 there, it also affects jackd2. No action has been taken, yet, but I tend to agree with the bug report and (temporarily) disable celt support in both packages. The older celt versions are about to vanish, so neither Debian nor Ubuntu will provide it within one year from now or so. I haven't followed the Opus discussion recently, but it seems the CELT support in both jackds needs to be replaced by Opus. The Wheezy freeze is scheduled for "mid June", so removing CELT-0.7 is likely to happen within the next weeks. So as a warning to all users, it looks like Debian and Debian-based distros are losing CELT support in netjack until somebody steps up and ports the code to Opus. Cheers -------- Original Message -------- Subject: Bug#674651: Please disable celt support in jack Resent-Date: Sat, 26 May 2012 12:36:50 +0000 Resent-From: Ron <ron(a)debian.org> Resent-To: debian-bugs-dist(a)lists.debian.org Resent-CC: Debian Multimedia Maintainers <pkg-multimedia-maintainers(a)lists.alioth.debian.org> Date: Sat, 26 May 2012 21:28:19 +0930 From: Ron <ron(a)debian.org> Reply-To: Ron <ron(a)debian.org>, 674651(a)bugs.debian.org To: Debian Bug Tracking System <submit(a)bugs.debian.org> Package: jackd1 Version: 1:0.121.3+20120418git75e3e20b-1 Severity: normal Hi, We're planning on removing the celt package from Wheezy, since we now have a stable release of Opus that people can really use. Please disable celt in jack so that we can move ahead with doing that before the freeze. If you can and wish to enable Opus support, that would be great, but for now we're mostly concerned with not shipping an obsolete celt version for another whole release cycle. Thanks! Ron _______________________________________________ pkg-multimedia-maintainers mailing list pkg-multimedia-maintainers(a)lists.alioth.debian.org http://lists.alioth.debian.org/cgi-bin/mailman/listinfo/pkg-multimedia-main…

13 years, 7 months

Re: [LAD] Plugin buffer size restrictions

by Jeff McClintock

> I assume that computing parameter trajectories basically means > interpolating, and that inevitably introduces some latency. That's a key point. Interpolating any sampled signal introduces latency. > let the host pass a limited number of future parameter samples at each > run() (could be negotiated at instantiation time), so that the plugin > doesn't have to add latency to the audio streams in any case. Would be > only supported by "offline hosts". If the block sizes are variable, > future block sizes should be passed as well (argh?). But I don't know > if this really makes sense or has downsides... ideas, folks? I *really* hate this idea. I play my MIDI keyboard into my DAW, perhaps while using my Mod-Wheel, or artistically using the filter-cuttoff parameter...I hit record...stop.. Then I push 'offline render'. You would say - shift all my parameter events earlier in time and render the result to disk? It's going to sound different. The timing will be wrong. A DAW is like a tape recorder. Playback or offline rendering should result in an identical performance surely?. Why are you selectively shifting some musical events in time but not others, why not note-ons too? You can't provide live MIDI playing 'in advance', you can't prove parameter updates in advance, just like you can't provide live audio in advance. If the plugin wants 'future' data to interpolate stuff, it needs to introduce latency. A good host will compensate for latency if the plugin API supports that. Parameters aren't special, they don't require any different handling than MIDI. What's the difference between a MIDI controller tweaking the filter cuttoff, or directly tweaking the parameter? Nothing. They both need smoothing, they both need interpolating, they both will have latency. Don't overcomplicate it. Jeff

13 years, 7 months

[OT] need help from a Qt guru re: slow graphics

by Dave Phillips

Greetings: The problem: When I build Kdenlive for my Arch 64 system it compiles without problems and starts up okay. After that the sluggishness of its response is almost unbearable. For example, I've timed up to 10 seconds between a mouse click and the resulting action, e.g. right-click to Add Clip, and everything stalls badly when the program is calculating audio or video thumbnails. Response time gets better after running the program for a while, but getting it going is rather frustrating. I've already searched Google and asked about the problem on the Kdenlive forum and got no useful replies, hence my request here. Some of you use Arch systems, and some of you know Qt well enough that maybe you can advise me on this problem. (I'm hoping for a relatively simple solution, of course). Incidentally, the problem occurs with Arch's packaged Kdenlive and the binary I compile with the build script. I just upgraded both, the problem is still there. Other large Qt apps don't have the problem, e.g. QTractor. The problem is also absent with Kdenlive on my 32-bit machines. Btw, video is an nVidia GeForce 7600 GS, with nVidia's driver. Best, dp

13 years, 7 months

Virtual Analog Filters | Fwd: [music-dsp] ANN: Book: The Art of VA Filter Design

by Florian Paul Schmidt

Maybe this is of interest to one or the other :D Flo -------- Original Message -------- Subject: [music-dsp] ANN: Book: The Art of VA Filter Design Date: Fri, 25 May 2012 10:54:25 +0200 From: Vadim Zavalishin <vadim.zavalishin(a)native-instruments.de> Reply-To: A discussion list for music-related DSP <music-dsp(a)music.columbia.edu> To: A discussion list for music-related DSP <music-dsp(a)music.columbia.edu> Hi all This is kind of a cross-announcement from KVRAudio, but since there are probably a number of different people on this list, I thought I'd announce it here as well. Get it here: http://ay-kedi.narod2.ru/VAFilterDesign.pdf http://images-l3.native-instruments.com/fileadmin/ni_media/downloads/pdf/VA… http://www.discodsp.net/VAFilterDesign.pdf (thanks to "george" for mirroring) There is a discussion thread at http://www.kvraudio.com/forum/viewtopic.php?t=350246 Regards, Vadim -- Vadim Zavalishin Software Integration Architect | R&D Tel +49-30-611035-0 Fax +49-30-611035-2600 NATIVE INSTRUMENTS GmbH Schlesische Str. 29-30 10997 Berlin, Germany http://www.native-instruments.com Registergericht: Amtsgericht Charlottenburg Registernummer: HRB 72458 UST.-ID.-Nr. DE 20 374 7747 Geschaeftsfuehrung: Daniel Haver (CEO), Mate Galic -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp

13 years, 7 months

simple FAUST OSC example

by Kaspar Bumke

Hi all, I am trying to wrap my head around the OSC interface to FAUST and am trying to get the noise.dsp from the examples to respond to OSC messages. I have tried sending it things like "/hello" and "/1/slider1 s 'hello'" and "/hello f 1.0" on port 5010 while listening on 5011 and 5012 to no avail. I must be completely missing something here but what exactly am I supposed to send when it says "a hello message" in the docs? Could someone just go through all the simple steps of an OSC "hello world" in FAUST? Kind Regards, Kaspar

13 years, 7 months

Proposals for JACK

by Nikita Zlobin

Hi all. Just returned from military services (mandatory period). It is likely, that i was somehow unsubscribed from list. In summary, proposed changes are joined in two parts: 1. A few advances for freewheel mode: waiting wheel and per client free/waiting wheel mode. 2. Get support for any count of "rooms", as they are called in LADISH therminology, but IMHO, a bit more flexible. Waiting wheel. Like freewheel, but doesn't speed up system. When performance is enough, behave like without any mode (produce sound). And only when it is not enough, automatically free wheel, until performance is enough again. Per client time wheel management. Clients could be joined into groups, where they have own time flow and may interact as they are not in free/waiting wheel mode (pass sound/MIDI each to others). Other proposals, in two words: - Ability to change master client (for now it is audio card) on the fly. - Attaching of audio devices as clients with ability to change driver and its settings for each (i mean - for all devices; this feature should go with previous item). Now about sctucture. Some components are plugins, which are like LADSPA / LV2 / Other kinds of audio plugins, but are optimized for use only on JACK (in short, they are JACK modules). JACK components: * single plugin shell (types: plugin host, jack / alsa app) - just to load other plugins as audio applications, connecting them to other audio systems, like alsa or other jack instance, like jack app. * Patchbay (types: plugin host, plugin). Master instance should be ran by already called plugin shell. Other instances may be loaded as plugins (making more rooms), or as jack applications. Time wheel mode changes affect all dependent clients, inluding other patchbays. * JACK app interface (plugin) * Adapter for other types of audio plugins (LADSPA, LV2, and so on). As you can see, such jack implementation may be jack app itself, and so - it is good to use existing work just as plugin host (like AMS, Ingen, Jost). Such jack structure should allow easy implementation of both multiroom session managers, like LADISH (which is forced for now to implement it by self) and even allow nested sessions: i'm thinking about universal modular session manager, which could be used as user session manager, but also allow nested desktop sessions, and could be expanded by plugins, adding integration to various systems, like X11 (to manage windows), jack / alsa (no comments needed) and so on.

13 years, 7 months

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Linux-audio-dev