Linux-audio-dev

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3 participants
7439 discussions

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 ---

21 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

21 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

21 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 ---

21 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

21 years, 9 months

[linux-audio-dev] unsunscribe

by ShezZan

unsunscribe

21 years, 9 months

Re: [linux-audio-dev] Plugin APIs (again)

by John Lazzaro

> Steve Harris <S.W.Harris(a)ecs.soton.ac.uk> writes: > > Yeah, please do that would be damn useful. For rapid prototyping if > nothing else FYI, making sfront produce code suitable for .so's is at the top of the list of things to do these days, because AudioUnits support awaits it. But, that's the "sfront enhancements" list of things to do, which is kind of subordinate to the "get MWPP to Last Call in the IETF" list of things to do ... so it may take a while. Basically, many Logic users would like to use SAOL as a scripting language for their own plugins ... thus, AudioUnits support. This could actually be a catalyst for SAOL becoming more popular generally, if it works out ... ------------------------------------------------------------------------- John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley lazzaro [at] cs [dot] berkeley [dot] edu www.cs.berkeley.edu/~lazzaro -------------------------------------------------------------------------

21 years, 9 months

RE: [ardour-dev] Re: [linux-audio-dev] HDSP 9652 Users - Request for info

by Mark Knecht

Paul, I think that if the dmesg/var/log/messages output said there were buffers, then it was getting loaded by automatically. Most likely this is because it is in modules.conf. However, when I ran insmod snd-hdsp by hand, I got some error messages that were a bit different than the :init_module message that shows up at boot time. These messages (from memory) had something to do with memory buffer or memory addresses more memory names. I don't remember right now, but will check it out later and report back. Mostly Fernando and I wanted to know what configurations successful users were using. Since I'm on 7.3, I have this funky C compiler, and just wanted to know if others were successful. I'm still considering a RH8.0 upgrade simply to get rid of some of these issues with Ardour and Rosegarden. (And probably create new issues...who knows?) thanks, Mark -----Original Message----- From: Paul Davis [mailto:paul@linuxaudiosystems.com] Sent: Tuesday, December 10, 2002 11:36 AM To: Mark Knecht Cc: D R Holsbeck; linux-audio-dev(a)music.columbia.edu; PlanetCCRMA; Ardour; Ardour-User-List; alsa-user(a)lists.sourceforge.net Subject: Re: [ardour-dev] Re: [linux-audio-dev] HDSP 9652 Users - Request for info > Hi. Actually, I tried that last night, but I didn't load the >snd-hammerfall-mem. It complained about some sort of missing references. >I'll give this a try later today. Thanks. in the message you sent recently, snd-hammerfall-mem worked just fine, and reported allocating buffers for the card. if snd-hammerfall-mem doesn't load, then neither snd-hdsp nor snd-rme9652 (not relevant here) can do so. --p

21 years, 9 months

[linux-audio-dev] [admin] lau postings from lad'ers [was Fwd: [admin] list policy, non-member posting]

by Joern Nettingsmeier

hi guys ! as most of you will know, the linux-audio-user list is members-only. many of you post helpful and perfectly on-topic replies there, which is great, but unfortunately from a non-subscribed address, which is not :( i used to hand-approve them, but there are just too many of those messages now to do that, so i'm getting anal about the policy and reject everything regardless of sender or topic. please, everyone, do not cease to follow lau, and please do share your expertise, but also please do it from a subscribed account. it really hurts me to stash messages from bill s. or paul d. into the bit bucket, but i just can't keep up. :( unfortunately, there is no easy and reliable way to tell mailman to accept all members of another list as posters, and i do not have the necessary privileges on the list server to roll my own cross-authentication. check out the possibility described below if you need to post from several mail accounts. best, jörn forwarded from lau: Jörn Nettingsmeier wrote: > > hi everyone ! > > lately, the number of posting attempts from non-members has risen to a > point where i'm just rejecting them without reading each one and checking > if it's on-topic or has already been sent to the list from another > account. > > our current list policy is members-only posting, for the following > reasons: > * i believe people who want to profit from a community should take part in > it > * spam prevention. so far, it has saved everyone from about .5k spam msgs. > > a number of list subscribers seem to post from different addresses, so > some of their mails are being rejected. if this applies to you, there is > always the possibility of subscribing multiple addresses and disabling mail > delivery for all but one via the web interface. > > best, > > jo"rn -- Jörn Nettingsmeier Kurfürstenstr 49, 45138 Essen, Germany http://spunk.dnsalias.org (my server) http://www.linuxdj.com/audio/lad/ (Linux Audio Developers)

21 years, 9 months

[linux-audio-dev] Synth APIs, pitch control

by Sami P Perttu

Hi everybody. I've been reading this list for a week. Thought I'd "pitch" in here because I'm also writing a softstudio; it's pretty far already and the first public release is scheduled Q1/2003. First, I don't understand why you want to design a "synth API". If you want to play a note, why not instantiate a DSP network that does the job, connect it to the main network (where system audio outs reside), run it for a while and then destroy it? That is what events are in my system - timed modifications to the DSP network. On the issue of pitch: if I understood why you want a synth API I would prefer 1.0/octave because it carries less cultural connotations. In my system (it doesn't have a name yet but let's call it MONKEY so I won't have to use the refrain "my system") you just give the frequency in Hz, there is absolutely no concept of pitch. However, if you want, you can define functions like C x = exp((x - 9/12) * log(2)) * middleA, where middleA is another function that takes no parameters. Then you can give pitch as "C 4" (i.e. C in octave 4), for instance. The expression is evaluated and when the event (= modification to DSP network) is instantiated it becomes an input to it, constant if it is constant, linearly interpolated at a specified rate otherwise. I should explain more about MONKEY for this to make much sense but maybe later. Anyway, the question I'm most interested is: why a synth API? -- Sami Perttu "Flower chase the sunshine" Sami.Perttu(a)hiit.fi http://www.cs.helsinki.fi/u/perttu

21 years, 9 months

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Linux-audio-dev