Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

3 participants
7535 discussions

by federico lopez

Hi, I am trying to compile Aliki 0.1 in Ubuntu 10.10 I Installed the required libraries libclthreads (>= 2.4.0) and libclxclient (>= 3.6.1) and libclalsadrv (>= 2.0.0). but when I try to compile I get this error: In file included from aliki.cc:26: styles.h:26: fatal error: clxclient.h: No such file or directory compilation terminated. Note: I found and fixed a problem installing libclxclient, but maybe I broke something... when I tried to install clxclient show this error: In file included from xdisplay.cc:22: clxclient.h:31: fatal error: X11/Xft/Xft.h: No such file or directory compilation terminated. I found a solution in a website and was solved with: |sudo apt-get install libxft-dev| thanks, federico lopez

14 years, 2 months

[LAD] "bleeding edge html5" has interesting Audio APIs

by Niels Mayer

http://kinlan-presentations.appspot.com/bleeding/index.html#42 ............ Don't we already have HTML5 <audio>? Yes :)...but <audio> can only take us so far Simple low-latency, glitch-free, audio playback and scheduling Real-time processing and analysis Low-level audio manipulation Effects: spatial panning, low/high pass filters, convolution, gain, ... ........... Judging by all the google-chrome symbols, this appears to be google-chrome-specific. Is any of this FOSS and available in Chromium? -- Niels http://nielsmayer.com

14 years, 2 months

[LAD] jack position/timebase tutorial/starting points?

by Iain Duncan

Just because everyone's tips here were so helpful for the ringbuffer conversation, does anyone have any pointers for where to start understanding jack transport and clocking, other than the transport client example? thanks! Iain

14 years, 2 months

[LAD] 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

14 years, 2 months

[LAD] 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

14 years, 2 months

[LAD] 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

14 years, 3 months

[LAD] [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

14 years, 3 months

[LAD] 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

14 years, 3 months

[LAD] 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

14 years, 3 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

14 years, 3 months

Jump to page:

2026

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

Linux-audio-dev