Linux-audio-dev

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7513 discussions

by David Olofson

What's going on with headers, docs, names and stuff? I've ripped the event system and the FX API (the one with the state() callback) from Audiality, and I'm shaping it up into my own XAP proposal. There are headers for plugins and hosts, as well as the beginnings of a host SDK lib. It's mostly the event system I'm dealing with so far. The modified event struct: typedef struct XAP_event { struct XAP_event *next; XAP_timestamp when; /* When to process */ XAP_ui32 action; /* What to do */ XAP_ui32 target; /* Target Cookie */ XAP_f32 value; /* (Begin) Value */ XAP_f32 value2; /* End Value */ XAP_ui32 count; /* Duration */ XAP_ui32 id; /* VVID */ } XAP_event; The "global" event pool has now moved into the host struct, and each event queue knows which host it belongs to. (So you don't have to pass *both* queue and host pointers to the macros. For host side code, that means you can't accidentally send events belonging to one host to ports belonging to another.) Oh, well. Time for some sleep... //David Olofson - Programmer, Composer, Open Source Advocate .- The Return of Audiality! --------------------------------. | Free/Open Source Audio Engine for use in Games or Studio. | | RT and off-line synth. Scripting. Sample accurate timing. | `---------------------------> http://olofson.net/audiality -' .- M A I A -------------------------------------------------. | The Multimedia Application Integration Architecture | `----------------------------> http://www.linuxdj.com/maia -' --- http://olofson.net --- http://www.reologica.se ---

22 years, 9 months

[linux-audio-dev] LADSPA and Softsynths

by Richard Furse

Just an observation about an alternative path on softsynths: a LADSPA plugin or network can be used easily enough as a softsynth using control-voltage (CV) approaches (a few already exist). It's just a matter of agreeing the conventions - implementation is trivial. I've been meaning to finish writing PNet for a while (I've mentioned it a few times) - essentially an environment where LADSPA plugins are strung together to form a "patch" and are wired up to "standard" CV controls for pitch, velocity, MIDI CC etc. These CV components and outputs can be provided by the host as "fake" plugins providing the CV signals based on MIDI input (or by using a non-LADSPA convention). This is trivial to implement and provides an extremely flexible way to build plugin-based softsynths from LADSPA components - or to wire existing self-contained LADSPA soft synths (e.g. the "analogue" synth by David Bartold in the CMT library, see http://www.ladspa.org/cmt/plugins.html) up to MIDI streams. All a question of time - if anyone wants to do the rest of the implementation then please let me know. The code required to do the above also provides a nice way to store patches of plugins for standard processing chains. Patches would probably be stored as XML representations of pure-LADSPA networks. BTW, is anyone doing this already? If so, 50% of the code is already done. ;-) I'm thinking in terms of defining a synth using two patches - one to define the per-note network required (e.g. CV->osc->filter->OUT) and another for any per-instrument post processing (e.g. IN->chorus->reverb->OUT). --Richard

22 years, 9 months

[linux-audio-dev] 2.4.20 low-latency patch

by Andrew Morton

At http://www.zip.com.au/~akpm/linux/2.4.20-low-latency.patch.gz Very much in sustaining mode. It includes a fix for a livelock problem in fsync() from Stephen Tweedie.

22 years, 9 months

[linux-audio-dev] XAP: Pitch control

by David Olofson

I was in this long thread about pitch control on the VST list, and I think I learned a few things. (For a change! ;-D) There are times when continous, linear pitch (what I have in Audiality) is perfectly fine - and in those cases, it's by far, the simplest possibly way you can control pitch of a synth. You get note pitch, pitch bend, continous pitch control over the whole range, whatever scales you like and all that, using *only a single pitch->frequency conversion* somewhere in your synth code. I will bet almost anything that there simply cannot be an easier way of dealing with this. *However*, in some cases, you may not be all that interested in the actual pitch, but rather just want to deal with the notes in whatever scale the user wants to deal with. One example would be a simple, basic arpeggiator. Sure, you *could* do that with linear pitch, but then the plugin would have to either assume that you want 12tET (or whatever), or you need a way to tell it what scale you want to use for the output. (Note that the kind of arpeggiator I'm thinking about here may be expected to generate a full, modulated chord from a single note, so it can't just look at a full input chord and pick the exact pitches from that.) In that case, you'd much rather have input more similar to integer MIDI pitch, and *possibly* pitch bend to go with that. This could indeed be expressed as "linear pitch" as well (float; 1.0 per octave), but with one very important difference: it would actually be 1.0 per *note* - where what a "note" is is not strictly defined or known to the plugin. The plugin just assumes that the user knows what 0-4-7 means, if he/she enters that for "arpeggio offsets". The plugin also assumes that the user will put a suitable note_pitch to linear_pitch pitch converter in between the output and the synth, or that the synth understands note_pitch events. Note that linear_pitch = note_pitch * (1.0/12.0) for 12teT, so these "conerters" (or note_pitch support) can be very trivial to implement. If you want "weird" scales, it gets slightly more complicated, but the *major* point here is that no synth plugin is required to do this - and still, every synth plugin can use any scale! (How many VSTi plugins actually support non-12tET scales? ;-) Hmm... As to having a synth support *both* note_pitch and linear_pitch controls, I suppose that would effectively just be a dual interface to a single internal control value. Send something to linear_pitch, and it goes directly into the internal pitch variable. Send it as note_pitch, and it gets multiplied by (1.0/12.0) or is passed through an interpolated "weird scale" table first. Makes sense? //David Olofson - Programmer, Composer, Open Source Advocate .- The Return of Audiality! --------------------------------. | Free/Open Source Audio Engine for use in Games or Studio. | | RT and off-line synth. Scripting. Sample accurate timing. | `---------------------------> http://olofson.net/audiality -' .- M A I A -------------------------------------------------. | The Multimedia Application Integration Architecture | `----------------------------> http://www.linuxdj.com/maia -' --- http://olofson.net --- http://www.reologica.se ---

22 years, 9 months

[linux-audio-dev] XAP and Event Outputs

by David Olofson

I did some thinking last night, and I have an interesting problem, as well as a nice solution - or so I think. Please tell me what you think. Consider the following situation: * You have a plugin "A", which has 1:4:7 (Bay:Channel:Control) (which is an Event Output) connected to some Event Input Port named "P". * Now, you want to connect output 1:2:9 (Bay:Channel:Control) to that same Event Input Port "P". So, what's the problem? Well, as I've mentioned before, having separate Event Input ports for Channels is probably an advantage in many most cases, since it avoids queue splitting overhead, and reduces event size. (No need for a "channel" field.) Regardless of the above, any reasonably complex synth will most probably have several "inner loops" working through the same number of sample frames. These two internal plugin designs have the same problem; you're not running *the whole plugin* one sample at a time, through the whole buffer. Instead, you're iterating through the buffer several times. Now, the *problem* is that whenever you send an event from inside one of these event and/or audio processing loops mentioned above, you risk sending events out of order, whenever two loops send to the same port! (Note that you can't know that without comparing a ton ofp ointers every time a connection is made. The host just tells you to connect some output, and gives you an Event Port pointer and a target Control Index to send to.) In theory, the problem is very easy to solve: Have the host throw in "shadow event ports", and then have it sort/merge the queues from those into a single, ordered queue that is passed to the actual target port. However, how on earth could the host know which outputs of a plugin can safely be connected to the same physical port, and which ones *cannot*? Easy: Output Context IDs. :-) Whenever the host wants to connect an output, it asks plugin->get_context_id(bay, channel, output), and gets an int. The actual values returned are irrelevant; they're only there so the host can compare them. How to use (plugin1 and plugin2 being the two plugins that have outputs to be connected to the same physical event port): struct XAP_cnx_descriptor { XAP_plugin *plugin; int bay; int channel; int output; }; /* * When you're about to make a connection to an input event port * that already has connections, use this to figure out whether * or not you need to do shadow + sort/merge. */ int must_shadow(XAP_cnx_descriptor *from1, XAP_cnx_descriptor *from2) { int ctxid1, ctxid2; if(from1->plugin != from2->plugin) return 1; /* Yes, *definitely* --> */ if(!from1->plugin->get_context_id) return 0; /* No, this plugin has only * one sending context. --> */ ctxid1 = from1->plugin->get_context_id(from1); ctxid2 = from2->plugin->get_context_id(from2); return (ctxid1 != ctxid2); /* Only if ctx IDs differ. */ } //David Olofson - Programmer, Composer, Open Source Advocate .- The Return of Audiality! --------------------------------. | Free/Open Source Audio Engine for use in Games or Studio. | | RT and off-line synth. Scripting. Sample accurate timing. | `---------------------------> http://olofson.net/audiality -' .- M A I A -------------------------------------------------. | The Multimedia Application Integration Architecture | `----------------------------> http://www.linuxdj.com/maia -' --- http://olofson.net --- http://www.reologica.se ---

22 years, 9 months

[linux-audio-dev] Re: [vst-plugins] Plugin server (Re: [vst-plugins] Re: [a bit OT] gcc as a cross compiler for VST?

by Paul Davis

Kjetil and i have been boring the VST crew to death. so we took it here :) >> because when running a real-time low-latency audio system, the cost of >> context switches is comparatively large. if you've got 1500usecs to >> process a chunk of audio data, and you spend 150usecs of it doing >> context switches (and the cost may be a lot greater if different tasks >> stomp over a lot of the cache), you've just reduced your effective >> processor power by 10%. >> >I dont believe you. I just did a simple context-switching/sockets >test after I sent the last mail. And for doing 2*1024*1024 context >syncronized switches between two programs, my old 750Mzh duron uses 2.78 >seconds. That should about 1.3usecs per switch or something. By you didn't touch much of the cache, did you? it doesn't matter how fast the actual switch is if each task wipes out the L1 and L2 cache, forcing a complete refill of the cache, reload of the TLB, etc. etc. the cost of a context switch is not just a register store+restore. the cost of it depends on what has happened since the last context switch. try your "simple context switch test" with a setup in which each task writes to about 256kB of memory. we measured this extensively on LAD a year or two ago. both myself and abramo and some others did lots of tests. we plotted lots of curves. the results were acceptable but not encouraging. yes, faster processors will decrease the time it takes to save and load the registers. but just as for much DSP code these days, other issues often dominate over raw CPU speed; the slow downs caused by the TLB being invalidated as a result of switching address spaces and the cache invalidation (for the same reason) are dramatic. >I'm not talking about jack tasks, I'm talking about doing a simple plug-in >task inside a standalone program, the way the vst server works. i don't understand how the vst server works. perhaps you can explain it. --p

22 years, 9 months

[linux-audio-dev] The beginnings of an ladcca manual

by Bob Ham

Hi again, I just completed a huge introduction to ladcca. I hope this will help generate some interest. It's part of a very incomplete manual, which I've put up on a webpage for the thing at http://pkl.net/~node/ladcca.html The introduction is here: http://pkl.net/~node/software/ladcca-manual-html-one-page/ladcca.html#SEC2 Bob

22 years, 9 months

Fwd: Re: [linux-audio-dev] XAP: Pitch control

by David Olofson

(Oops. Replied to the direct reply, rather than via the list. Please, don't CC me - I'm on the list! :-) ---------- Forwarded Message ---------- Subject: Re: [linux-audio-dev] XAP: Pitch control Date: Wed, 11 Dec 2002 18:05:57 +0100 From: David Olofson <david(a)olofson.net> To: Nathaniel Virgo <nathaniel.virgo(a)ntlworld.com> On Wednesday 11 December 2002 17.50, Nathaniel Virgo wrote: > On Wednesday 11 December 2002 3:41 pm, David Olofson wrote: > > On Wed, Dec 11, 2002 at 12:40:18 +0000, Nathaniel Virgo wrote: > > > I can't really say I can think of a better way though. > > > Personally I'd leave scales out of the API and let the host > > > deal with it, sticking to 1.0/octave throughout, but I can see > > > the advantages of this as well. > > > > Problem with letting the host worry about it is that the host > > would normally not understand anything of this whatsoever, since > > the normal case would be that a sequencer *plugin* controls the > > synths. It would be a hack. > > Oh. Well, when I said host I meant sequencer. I see. Well, either way, I still prefer thinking of scale converters as something I may just plugin, rather than waiting for my favourite sequencer to support the kind of scales I want. One multichannel event processor plugin more or less in the net won't be a disaster - and again, you *can* use 1.0/octave in the sequencer as well as an alternative. //David Olofson - Programmer, Composer, Open Source Advocate .- The Return of Audiality! --------------------------------. | Free/Open Source Audio Engine for use in Games or Studio. | | RT and off-line synth. Scripting. Sample accurate timing. | `---------------------------> http://olofson.net/audiality -' .- M A I A -------------------------------------------------. | The Multimedia Application Integration Architecture | `----------------------------> http://www.linuxdj.com/maia -' --- http://olofson.net --- http://www.reologica.se ---

22 years, 9 months

[linux-audio-dev] band limited interpolation

by Henry Gomersall

I'm trying to oversample to smooth the display on my software audio scope using band limited (sinc) interpolation. I have a quick question to ask. Which of the following implementations is liable to take more processing time: 1) Padding my data, and then convolving the original (non-zero) samples with a sinc function (from a LUT). I think it's only feasible to have a LUT for a sinc of unity amplitude. or 2) Doing an FFT on my original sample set. Padding, and multiplying by a rectangular window, then doing an iFFT to return to the time domain. The FFT library I would use would be FFTW. Cheers Henry

22 years, 9 months

[linux-audio-dev] unsunscribe

by ShezZan

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22 years, 9 months

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Linux-audio-dev