Linux-audio-dev

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4 participants
7502 discussions

RAUL or other libraries for real this time? ;-)

by Iain Duncan

Still curious about RAUL. As i have no immediate plans beyond learning how to write proper audio app, even if license restrictions prevent from using RAUL in a hypothetical commercial product years down the road, it may well be worth me using for my own personal needs in the meantime. Would love to hear feedback on technical merits of RAUL, minus license conversations which now have my other thread. ;-) heck I'd like to hear about any other libraries worth looking into too. I'm trending at the moment to STK, embedded csound, Jack, and QT for gui. The part I need library help with is likely synchronization and interprocess/interthread communication. ( ie do I use the jack ringbuffer? Do I look at boost queue implementations? does RAUL have a higher level convenience ring buffer? thanks! iain

13 years, 7 months

RAUL?

by Iain Duncan

I found it on Dave's site, but other than that, couldn't find find much mention of it. Do many people use it? Would it be wise to dig into RAUL for writing a real time jack app? Dave, any comments on it? http://drobilla.net/software/raul/ thanks iain

13 years, 7 months

tutorial or example for using jack ring buffer between threads?

by Iain Duncan

Can anyone point me at what they consider the best thing to look at for an introduction to communication between threads in a jack app using the ringbuffer? I found some, but as docs appear a bit scattered, wondered if there was a known best-first-reference type thing. thanks iain

13 years, 7 months

[ANN] ALPHA: jiss - Jack Interactice Sequencing Software

by Florian Paul Schmidt

this is totally prealpha-OMG-IT'S-FULL-OFF-FAIL-state.. https://github.com/fps/jiss requirements: SWIG, libjack-dev and liblua5.1-dev on ubuntu.. compile with make (if it fails, you're on your own. it's a simple makefile though). then run in the build dir: lua wicked.lua if you have jass running with a piano sample on midi channel 0, a bass drum on channel 1 and a hihat on 2 you should get a rather weird interpretation of "stella by starlight", a jazz standard.. something like this (some effects added with jack-rack): http://shirkhan.dyndns.org/~tapas/stella.ogg (wicked.lua code here with some omission of some chords at the start and some remarks in comments added): -- some stuff :D require "jiss" require "jissing" -- create engine in stopped state e = jiss.engine() -- setup some state that the sequences later use -- e:run can only be used when the engine is stopped.. -- as this is executed in non-RT context it's ok to -- create some variables and tables here.. e:run([[ bar = 0; min = 20; max = 80; stella = { range(min, 80, min7b5(E(4))), range(min, 80, min7b5(E(4))), -- cut away quite a bit here (see wicked.lua in git clone) :D range(min, 80, maj7s11(B(4)-1)), range(min, 80, maj7s11(B(4)-1)) } ]]) -- this sequence can control the others since it's processed before -- the others in the engine -- events string is newline sensitive. in this case the events -- on consecutive lines are spaced 1 second apart.. -- also: loop back to 0 at time t = 8 sec tune = seq(e, "tune", loop_events(8, events_string(1, [[ drums1:relocate(0.0); drums1:start_(); notes:relocate(0.0); notes:start_() drums1:stop_(); ]]))) -- manually start this sequence and add to the engine tune:start() -- note that a copy is appended to the engine e:append(tune) -- a sequence that controls the global variable bar to advance through the song play(e, seq(e, "control", loop_events(1, events_string(1, [[ bar = bar + 1; bar = (bar % #stella); ]])))) -- events at fixed times. loop at t = 0.75 sec play(e, seq(e, "notes", loop_events(0.75, { { 0.125, [[ for i = 1,4 do note_on(0, 24 + stella[bar][math.random(#stella[bar])], 30 + math.random()*64) end ]] }, { 0.5, [[ for i = 1,2 do note_on(0, 24 + stella[bar][math.random(#stella[bar])], 10 + math.random()*34) end ]] } }))) -- a drum pattern drums = [[ note_on(1, 64, 127); note_on(2, 64, 127) note_on(2, 64, 127) note_on(2, 64, math.random(127)) note_on(2, 64, math.random(127)) note_on(2, 42, 110) note_on(2, 64, 127) note_on(2, 64, math.random(127)) note_on(1, 64, 127); note_on(2, 64, 127) note_on(2, 64, math.random(127)) ]] play(e, seq(e, "drums1", loop_events(1, events_string(0.125/2, drums)))) -- connect all sequence outputs to jass:in connect(e,"jass:in") -- run the whole thing e:start() -- wait for the user to press enter io.stdin:read'*l' Have fun, Flo

13 years, 7 months

platform choices, jack for sequencing?

by Iain Duncan

Thanks everyone for all the help on my architecture questions. It seems like a lot of the best practise functionality has tools/components for it already in Jack. I *was* planning on using rtaudio in order to be cross platform, but if it's a lot easier to get things done in Jack, i could live with being limited to linux and OS X. Just wondered if I could poll opinions, for a real time step sequencer meant to do super tight timing and by syncable with other apps, is Jack going to be a lot easier to work with? Should I just lay into the jack tutorials? And is it straightforward to use the perry cook stk in a jack app? thanks everyone iain

13 years, 7 months

Conciderations of Design

by harryhaaren＠gmail.com

Hi All, Recent thoughts of mine include changing the "direction" of operations in real-time program design: Eg: Don't call a set() on a Filter class to tell it its cutoff, every time there's a change, but make it ask a State class what its cutoff should be every time it runs. I'm now considering implementing this pattern, but I'm wondering have others done this before? Along the same lines, say I have a single linked list of AudioElements, and the 3rd element needs info, should it request it, be told it, or have some other system to inform it of events? I'm seeing downsides to each approach: 1: Tell it on every change -> performance hit 2: Request it every time it runs -> Keeping control over the many values & unique ID's of class instances Experience & Opinions all welcomed, -Harry

13 years, 7 months

Re: [LAD] Conciderations of Design

by Jeff McClintock

David's advice is right on. > why not use audio rate control ports? Or some sort of hybrid, allowing > you to > switch as needed - but that quickly becomes a complexity explosion... I use a hybrid where ports are audio-rate AND each port has two states: 'streaming' or 'static'. So if you don't want audio-rate modulation, you pass a buffer of identical values and set the port state 'static' (not changing). The advantage is: 'Dumb' plugins need no special-case code, just write them as though the port was always audio-rate, one single simple code-base. 'Smart' plugins can query the port state and switch to more efficient code when the port is 'static', e.g. read only the first sample, treat it like a block-accurate parameter (i.e. Far more efficient). Best Regards, Jeff

13 years, 7 months

[ANN] Jass-0.9 - Jack Simple Sampler

by Florian Paul Schmidt

Hi, I spent a few hours here and there to work on jass. Thus I give you release 0.9 which is fairly feature complete. But it might still have a gazillion of bugs. So please test, before I go 1.0.. http://shirkhan.dyndns.org/~tapas/Jass-0.9.tar.bz2 Jass - A Jack Simple Sampler Qt4-, libsamplerate-, libsndfile-, jack_midi-, jack_session-, ladish-L1-enabled sampler.. Changes (AFAICT): * graphical editors for most parameters (those can be resized to suite your needs) * waveform display to set sample start/end, loop start/end * keyboard widget to set note, min note and max note * a retarded dial_widget that is barely usable :D * global voice allocation (you can set the global polyphony per setup in the xml file * ADSR envelope that actually works * show/hide some parameter sections Screenshot (showing all parameter editors): http://i.imgur.com/Ssc4F.png Regards, Flo

13 years, 7 months

audio architecture wiki/docs/e-book idea?

by Iain Duncan

Hey everyone, especially those who have been helping me with me architecture questions. I'm wondering whether some of you would be interested in helping in a simple advisory/editorial capacity if I were to try to write up a wiki or e-book of some kind on real time audio architecture. It seems to me like there is a lot of knowledge here on it, and not very many (if any) good starting points for such things. I've found decent references on the dsp side of audio coding, but not seen anything on 'how to put together a real time audio app from a-z' kind of thing. I find writing docs helps me clarify things in my head, I'd be interested in doing some writing if I know that people who know what they are doing would be interested in advising and correcting. I figured if I put it online it might be a good source of publicity for your work and we could link back to projects ( ardour, etc, ) It would take a while of course, but might also help people new to these lists and give us all some thing to point at and say: there's a good write up on that here -> thoughts? iain

13 years, 7 months

Blog post on Polyphonic Faust synth

by Timo Westkämper

Hi. I wrote this blog post on a Faust based polyphonic virtual analog synth prototype : http://westkamper.wordpress.com/2011/11/09/polyphonic-faust/ This might be of interested to some of you Cheers, Timo

13 years, 7 months

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Linux-audio-dev