Hey Linux Audio *,
After some time away, OpenAV is back with some updated releases! Much of
the work done in these releases
was in collaboration with the Linux Audio Developers and Uesrs community,
thanks for your contributions!
This release announce email includes ArtyFX 1.3.1, a dot release
Hey All,
Its been a long time, but there's a new Luppp release out now, version 1.2
https://github.com/openAVproductions/openAV-Luppp/releases/tag/release-1.2.0
Huge thanks to all the contributors, this release was largely driven by
contributions,
and community interaction and interest in the
On Fri, Sep 8, 2017 at 6:28 PM, benravin wrote:
> Hi Paul,
>
> I feel its better to add these support as well in JACK framework, like
> sending control signals, notifications etc, rather than user has to
> implement these.
>
Hi Ben,
Please be aware of "top posting" - it
On Sun, Sep 3, 2017 at 3:02 PM, Will J Godfrey
wrote:
>
> On Sun, 3 Sep 2017 13:35:47 +
> Fons Adriaensen wrote:
>
> >On Sun, Sep 03, 2017 at 08:50:08AM +0100, Will Godfrey wrote:
> >
> >> Does anyone know of software that can log these
On Mon, May 22, 2017 at 2:04 PM, Thomas Cipierre wrote:
> Dear LAC2017,
>
> I’m part of the LAC2017 committee, and was nearly all the time involved in
> the IRC #LAC2017 freenode channel as tcUJM (except one chairing session).
>
Hi Thomas,
Thanks for taking
On 3 May 2017 22:53, "Simon van der Veldt"
wrote:
Well, this will be my first LAC, so I'm pretty new to all of this, but I'd
very much like to join if that's OK.
As long as they have something vegetarian that's half way decent I'll be
fine :)
+1 I'm in for pre-conf
Hey All!
OpenAV has just pushed releases for Luppp, ArtyFX, Fabla, and Sorcer!
Release Video: https://youtu.be/s07wNn9Xc7s
But what's more, there's now an OpenAV Manual: detailed and useful
information
about all of the OpenAV projects, and what features they can do for you:
On Thu, Sep 8, 2016 at 8:54 AM, Christopher Arndt
wrote:
> Am 08.09.2016 um 09:25 schrieb Bengt Gördén:
>> Is Portaudio the best choice for portability or
>> is there some other option that might be better?
>
> You might want to have a look at RtAudio or even JUCE too.
On Mon, Feb 29, 2016 at 7:52 PM, Spencer Jackson
wrote:
> > The generic solution for cases like this is a lock-free ringbuffer.
> I've also used the jack ringbuffer for this and it was easy enough.
Simple tutorial on using JACK ringbuffer and C++ event class here:
Hi All,
Just a note that today(Tuesday 23rd) OpenAV Productions[1] and MOD[2]
will be interviewed by the FLOSS Weekly show.
Live link, 5:30 pm Irish / UK time: https://twit.tv/live
Use this link to figure out what time that is in your time-zone :)
On Mon, Feb 8, 2016 at 10:07 AM, Fokke de Jong
wrote:
> And the result is super-stable audio with a period of 32 samples @48Khz
> (for a total roundtrip latency of 3.5ms).
>
Congrats, that's pretty solid! Would you share some details on what config
options you used, and
On Thu, Feb 4, 2016 at 11:22 AM, Fokke de Jong
wrote:
> So now i am a but confused how to get the same kind of latency in my own
code (using alsa directly rather than through jack)
Are you coding ALSA yourself, or using a helper library? I can recommend
Fons'
On Wed, Jan 27, 2016 at 6:22 PM, Ivica Ico Bukvic
wrote:
> The migration should be now complete. Please let us know if you encounter
> any problems.
>
I clicked a heap of links, and tried breaking it - did not manage :)
Thanks for your effort - appreciated!
Cheers, -Harry
On Sun, Jan 24, 2016 at 2:03 PM, Fokke de Jong
wrote:
> Currently working on some realtime convolution with lots of channels and
> low latency requirements, but I am running into some unexpected cpu-spikes
> and hope some of you might have an idea of possible causes.
>
On Mon, Jun 8, 2015 at 10:18 PM, Gerald gerald.mwa...@gmx.de wrote:
Just out of interest: why are you trying to run jack as root?
$ whoami
root
I am root, a user who likes to run JACK for audio :)
Cheers, -Harry
PS: Yes I'm aware of security concerns etc
--
On Tue, Apr 28, 2015 at 12:57 AM, Tim E. Real termt...@rogers.com wrote:
The effect is striking. You can hear it without even plugging the guitar in.
As you adjust the pickup ever higher, and pluck the strings, you can
hear the horrible overtones from the frequency splitting.
Wow really? I
On Fri, Apr 24, 2015 at 2:26 PM, Len Ovens l...@ovenwerks.net wrote:
In my opinion the best slider will allow the pointing device (finger or
mouse) to be placed anywhere on the slider and moving the mouse will move
the value from where it was in the direction the finger moves. (Ardour fader
Hi *,
Some good points, and interesting points of view. @Thijs van Severen:
I completed various UX / UI modules during my undergrad - that,
together with the most obvious lacks (IMO) of Linux Audio UX is what
prompted my composing of that exact list.
Tracey Hytry wrote:
a brief splash screen
Hi All,
As promised just at the closing ceremony of LAC, an email opening the
discussion of User Experience on Linux Audio. To all Developers,
please use this as a checklist and consider supporting each item. It
will improve the user experience.
1: Splash Screen
If an app takes more than one
Hi All,
As OpenAV it is my pleasure to call for testing of ArtyFX 1.3.
The code is available online right now! [1]
New plugins include:
- Driva (Guitar Distortion)
- Whaaa (Wah pedal)
Credits:
- Whaaa DSP from WAH Plugins by Fons
- Driva distortion algorithms ported from Rakarack project
Hi All,
As root, starting JACK1 0.124.1 on an almost totally vanilla
3.19.2-1-ARCH kernel fails. Output of kernel with JACK1 issue is
pasted below[1]. The same system works fine with the 3.18.9-rt5-1-rt
kernel.
Running groups tells me there is no group other than root, so JACK
seems to
On Tue, Mar 31, 2015 at 2:02 PM, Vaclav Mach vaclav.m...@artisys.aero wrote:
What am I doing wrong? Or is it a bad way of using JACK2?
Hi,
You seem to want to write JACK clients - using C++. To do this you use
the JACK C API, which is the same for JACK 1 and JACK 2. Details of
the difference
On Tue, Oct 21, 2014 at 2:56 PM, Philippe Coatmeur phi...@gnu.org wrote:
Sends a sine wave to output (this is the standard sin
http://www.cplusplus.com/reference/cmath/sin/ math function, right?)
then what would send a square wave? What would send noise?
If only it were so simple... Sine
On Mon, Oct 20, 2014 at 1:39 AM, Philippe Coatmeur phi...@gnu.org wrote:
lots of emails
Hi Philippe,
I can't follow the issue you're having here: you've sent 8 emails, most
with different questions with a layout that is hard to follow: please be
more careful in what text you reply to, and remove
On Mon, Oct 20, 2014 at 12:26 PM, Philippe Coatmeur phi...@gnu.org wrote:
Now for the direct questions:
- Is it a proper NTK widget class declaration, that would expose a
value() method?
- Are those proper NTK widget class instances?
You must derive from an NTK widget in order for
On Fri, Oct 17, 2014 at 1:30 PM, Phil CM phi...@gnu.org wrote:
Also each time the UI is opened, the widgets are reset at maximum (not the
values).
Hi Phil,
Glad to see you're excited and working hard on learning LV2! Checkout the
example plugins: they show how things work.
In the UI, there
On Thu, Oct 2, 2014 at 1:14 PM, Phil CM phi...@gnu.org wrote:
I need a synth with a pitch control. A synth that can be controlled to
produce all tones between one note and another, say from a2 to c3 in an
anti-aliased way, like what you would get in pd by altering the pitch of a
sine wave.
Hi
Hi,
When replying, please be careful to reply to the list, as well as the
person: otherwise 1/2 a conversation gets lost ;) I've included
linux-audio-dev@lists.linuxaudio.org again.
On Thu, Oct 2, 2014 at 4:13 PM, Phil CM phi...@gnu.org wrote:
On 02/10/2014 14:13, Harry van Haaren wrote
On Thu, Oct 2, 2014 at 7:02 PM, Phil CM phi...@gnu.org wrote:
Funny, I was just reading your blog post about it :)
Its pretty outdated by now, lv2plug.in/book is the new resource IMO.
But I'm leaving it there for the sake of it :)
But I cannot get it to build because of a gtkmm error:
Fons wrote:
The many times I've had to set a delay time the most convenient unit could
have been samples, millisecs or meters (at the speed of sound), depending on
the context
Sure, there are uses (particularly your fields of WFS, Ambisonics, and
related) where such uses are prevailent. Aka,
Hi all,
There's a lot of interesting points brought up: I've only just had the
time to read them, and reply to the points that most speak to me (the
individual) and me (as OpenAV).
Charles Henry wrote:
There does not currently exist a company that is credibly making a complete,
whole-system
On Wed, Oct 1, 2014 at 12:26 PM, Joël Krähemann weedli...@gmail.com wrote:
Why doesn't Harry learn howto do a dial. I just take a look at his code
and it just sucks.
Hi Joel,
The topic isn't the quality of my code: I'm sorry to hear you feel it
just sucks. If you wish to discuss the quality of
Hi Linux Audio Developers,
TL;DR; Discussing experience driven design for linux audio.
I'd like to discuss the age of experiences. Allow me 10 minutes of
your time, to watch a video by Aral Balkan talk about development of
technology, FLOSS, design, and the future.
To start, please watch the
On Sat, Sep 27, 2014 at 6:11 AM, Joël Krähemann weedli...@gmail.com wrote:
Am I correct assuming 1 sample of stereo frames will be written 44100
times by a rate of 44100 Hz audio sampled data?
Yes. Just to be 100% clear, a sample is a *single* point of data.
In a mono situation: a frame refers
On Thu, Sep 25, 2014 at 8:50 PM, Joël Krähemann weedli...@gmail.com wrote:
Does someone know about writing WAV using libsndfile?
My file becomes 174M big as using sf_writef_short() within 8s
Are you sure you're only writing each sample once? Sounds like there's
an error in writing nframe
On Fri, Aug 22, 2014 at 3:12 PM, t...@trellis.ch wrote:
is it correct that the following two scenarios give the exact same result?
(digital audio signal) - (record) - (playback) - (apply fx) - (result)
(digital audio signal) - (apply fx) - (record) - (playback) - (result)
I'll add a note that
On Mon, Jul 7, 2014 at 5:00 PM, Flávio Schiavoni f...@rendera.com.br wrote:
I'm asking it to try to help with your other doubt involving Jack MIDI or
ALSA MIDI.
Here goes my 3 cents.
If you need to run it with sudo, maybe the best approach is to divide the
system in a client / server
Hey Will,
This really depends: are you asing about refactoring the code-structure
(classes, inheritance, is-a vs. has-a owenership etc) or actual code in
the sense of stl vectors and refactoring actual useful code as I see it :)
For basics of C++ std libs etc, I generally refer to these:
Hey *,
I'd like to inform the linux-audio-developers / community that I offer my
assistance to developers in supporting the NSM session-management system.
Its a low-overhead (OSC library environment variable only) session
management system, and its working well for a large majority of users.
Rt audio and poet audio are different projects.
For RtAudio, there are 2 CPP files that you compile into your code : no
linking library just yet.
When you want jack / salsa support, you add the appropriate #define, and
link to the libs needed by that output framework. Check the RtAudio page
for
Hi Thiago,
Slightly other question: are you *sure* you want to write directly using
the ALSA API?
Generally, writing a JACK[1] client (for linux-pro-audio use-cases), or
RtAudio[2] is a better idea, and easier in my opinion...
They're easier to write to, and abstract away some of the details of
Hey All,
It's my pleasure announce that ArtyFX 1.1 is released!
See the ArtyFX page for details on the 3 new plugins:
http://openavproductions.com/artyfx/
Source available from github:
https://github.com/harryhaaren/openAV-ArtyFX/releases
And many thanks to the contributors! -Harry
On Tue, Jan 21, 2014 at 3:34 PM, Filipe Coelho fal...@gmail.com wrote:
On 01/21/2014 12:40 PM, Fons Adriaensen wrote:
They can learn to do it [compile software, -Harry]. It's not rocket
science.
I think it's not up to the users to understand how software compilation
works.
I feel this is an
On Mon, Dec 23, 2013 at 5:05 PM, Gabbe Nord gabbe.n...@gmail.com wrote:
I'd greatly appreciate any participation from developers.
On behalf of OpenAV I've completed it.
Thanks for taking the initiative and making this happen.
To an interesting and productive discussion! -Harry
I've created a github issue for this discussion, to avoid software-specific
discussion to LAD:
https://github.com/harryhaaren/openAV-Luppp/issues/67
Lets continue the discussion there :)
Cheers, that's a nice piece of software !
Thanks, -Harry
___
Hey All,
I'm delighted to say that 5 days after announcing, Luppp has reached its
target donation!
That means that its now available under the GPLv3+ license, to everybody!
Grab the details source from here: http://openavproductions.com/luppp
I'd like to thanks the community, for the awesome
Hey!
Its my pleasure to announce Luppp, the flagship project of OpenAV!
Announce video: https://www.youtube.com/watch?v=AcIuKktCaLg
Its a live-looping program with a grid based workflow. This the Luppp
has features like NSM integration, ArtyFX integration, JACK integration,
and powerful MIDI
On Sun, Oct 13, 2013 at 9:49 PM, Philipp Überbacher mu...@tuxfamily.org
wrote:
I'm looking for a library that I could use to categorise audio.
Do you know or even have experience with such a library?
Aubio, and the VAMP plugins would be where I'd start.
Aubio has beat-onset, and BPM etc, some
Hey All,
ArtyFX is released after 5 days, cheers to all the contributors!
Checkout details here: http://www.openavproductions.com/artyfx
Cheers, -Harry
___
Linux-audio-dev mailing list
Linux-audio-dev@lists.linuxaudio.org
On Thu, Oct 10, 2013 at 6:29 PM, Aurélien Leblond blabl...@gmail.comwrote:
- what is the cleanest way to access input port data from the GUI?
In short: the DSP part should write an Atom event to notify the GUI.
The long: Ardour3 deletes UI instances when the window is closed: that
means the
On Thu, Oct 10, 2013 at 8:07 PM, Paul Davis p...@linuxaudiosystems.comwrote:
If a plugin uses one of them (the original) then Ardour will NOT delete
the UI instance when it is closed.
If a plugin uses the other (the version forked/copied by falktx) then
Ardour WILL delete the UI instance
Hello all Users Devs of linux-audio-land,
Moving forward from the topic on Aeolus and forking projects, perhaps it is
wise to look at how the community as a whole can grow from this situation:
1) It seems the frustration of forks is mainly due to lack of communication.
2) Had appropriate
On Fri, Sep 20, 2013 at 11:38 AM, Ralf Mardorf
ralf.mard...@alice-dsl.netwrote:
Harry, could you please post some links, when you have seen the
frustration you're talking about?
I'd much prefer focus on improving from where we are: not highlighting
where communication may have broken down.
On Fri, Sep 20, 2013 at 1:26 PM, Ralf Mardorf ralf.mard...@alice-dsl.netwrote:
The way things are saved and restored for Qtracor is a real PITA and that
it seems to be impossible to cancel changes, when leaving an editor.
I see a QTractor feature request / bug report coming...
On Fri, Sep 20, 2013 at 1:25 PM, Dan Muresan danm...@gmail.com wrote:
There's also the fact that you can't attach patches to github issues.
https://gist.github.com/ can be used for this: see
https://github.com/ned14/Easyshop/issues/1
-Harry
PS: Apologies for the blank mail
Hey everybody,
OpenAV is releasing Sorcer 1.1! Upgraded features such as a compressor and
output level provide better feedback in an all new NTK based UI! Existing
presets are remain unchanged: the compressor is off by default.
Packagers will be happy to know that Sorcer now uses Make, so
Hey All,
I've written up a blog post on allocating (audio) buffers in real-time.
The trigger was designing the record functionality for Luppp.
http://openavproductions.com/realtime-buffer-allocation
Hope its of interest to some, -Harry
___
On Fri, Aug 9, 2013 at 2:54 AM, J. Liles malnour...@gmail.com wrote:
Well, it took a couple of weeks of hair pulling and many, many hours of
testing, but I finally arrived at a solution.
Brilliant stuff, most definatly will test after my mini-holiday till
Monday. Expect feedback of some sort :)
Hi,
Its my pleasure to announce the release of Fabla!
After 8 days we have reached the target donation amount, many thanks to all
those who contributed!
Available here: http://openavproductions.com/fabla
Cheers! -Harry
___
Linux-audio-dev mailing
Hi all,
I'm working on a wavetable oscillator class, and I'm wondering about how to
best go about bandlimiting. I see two ways to achieve bandlimiting, i'll
detail as A and B.
A) Create different wavetables for each octave. Base octave includes all
harmonics. Octave 1 has the top half of the
On Sat, Jul 13, 2013 at 5:32 PM, Fons Adriaensen f...@linuxaudio.org
wrote:
This could be done very efficiently, in particular if you accept some
compromises for the lowpass filter, and even more if the waveforms can
be summed before downsampling.
Good idea, I wouldn't have thought of summing
Hey everybody,
Its my pleasure to announce that the next OpenAV Productions LV2 plugin is
finished!
Its called Fabla, and its a performance sampler.
Page: http://openavproductions.com/fabla
Demo reel: https://vimeo.com/70122957
One year from today Fabla will be released, and each donation
On Wed, Jun 12, 2013 at 8:10 PM, ssm salmin01...@gmail.com wrote:
I know it's a long post, but even if you can help with a part of it, I'll
be greatly benefited.
Hi Salmin,
You are using the ALSA API directly: if you're *sure* you want to do that,
disregard the next bit.
Most pro-audio
Hi everybody!
Its my pleasure to announce, that after 9 days the Sorcer wavetable synth's
donation target has been reached. Not only that, but as I prepared to push
the code to github, not one but two more donations arrived.
Wauw. Thank you for kicking off OpenAV productions.
Now for the best
Hey Everybody,
I'm happy to announce OpenAV productions: http://openavproductions.com
OpenAV productions is a label under which I intend to release my
linux-audio software projects. The focus of the software is on the workflow
of creating live-electronic music and video.
The release system for
On Mon, Apr 15, 2013 at 1:23 AM, J. Liles malnour...@gmail.com wrote:
To whom it may concern, http://faust.grame.fr is down.
And its back up :)
Announcements of server up-down time are made on the faudiostream-users
mailing list:
On Mon, Apr 8, 2013 at 12:45 PM, Raphaël BOLLEN raphael.bol...@mobistar.be
wrote:
error: invalid conversion from 'void*' to 'char*' [-fpermissive]
jack_ringbuffer_read() expects the buffer pointer to be of type char* not
void*.
The char* should just be interpreted as pointer, as the data is
On Mon, Apr 8, 2013 at 1:01 PM, Raphaël BOLLEN
raphael.bol...@mobistar.bewrote:
Hi Harry, thanks for the information.
Your welcome!
I'm intrested in what kind of JACK program you're working on..?
Also I'm collecting example / tutorial code for JACK programming, perhaps
its of intrest to you:
Ah very cool. I'm not using python, but might implement a class / some C++
code when I get time for the same purpose. Hooks on OSC commands to do /
tell to do the NSM stuff...
On Thu, Apr 4, 2013 at 1:22 AM, Nils Gey i...@nilsgey.de wrote:
Dear Developers,
I've written a convenience
On Thu, Apr 4, 2013 at 7:19 PM, J. Liles malnour...@gmail.com wrote:
Harry, this already exists in C++ class form and also in a single C header
file. nonlib/NSM/Client.{CH} and nonlib/nsm.h, respectively.
Brilliant, thanks for the pointer. I've not played with implementing NSM
yet, but its
On Tue, Mar 19, 2013 at 1:02 PM, Fons Adriaensen f...@linuxaudio.orgwrote:
The code below will do the trick
Brilliant, thanks for sharing. Will be implementing learning from this
later, appreciated!
-Harry
___
Linux-audio-dev mailing list
On Mon, Mar 18, 2013 at 11:45 PM, Fons Adriaensen f...@linuxaudio.orgwrote:
A critically damped
second order lowpass with a rise time of 30 ms or so will eliminate all
audible artefacts. It's very low on CPU and you only need to run it while
the gain is changing.
Although I understand the
On Tue, Mar 19, 2013 at 3:26 AM, Tim Goetze t...@quitte.de wrote:
tradition (like RBJ's lovely cookbook at musicdsp) will ask for filter
Q, which is 0.5 for critical damping.
Nice mention, checked out the code there, will play around with it a bit :)
On Wed, Mar 6, 2013 at 12:05 PM, Dave Phillips dlphill...@woh.rr.comwrote:
http://www.linux-community.de/Internal/Artikel/Print-Artikel/LinuxUser/2013/01/Bitwig-Professionelle-Musik-Workstation-fuer-Linux
Excellent review, in German only, with some enticing screenshots.
Thanks for the link,
Hi everybody!
Rui: nice graphs, definatly using that one :D
Dominique: I've not tried your suggestion yet. Perhaps later if I have time.
Adrian: Hah, the brute force method. Will give it a go and see what happens
Robin: I'll have a look at the scripts, and see if inspiration strikes ;)
Thanks
Hey all,
I'm currently attempting to stress test my setup of -rt kernel, rtirq
scripts, and a Jack client program I've been working on.
So my idea is to create a script that runs the programs, and also a
cpu-load generating program (cpuburn or alternative).
Then collecting stats based on Xruns,
On Sun, Feb 17, 2013 at 3:57 AM, M Donalies ingeniousnebb...@cox.netwrote:
No locks or mutexes in a callback function. I need to think about that one.
This is indeed a lovely topic for debate. I'm bound to say that, I'm
currently doing a final-year project for college on the topic.
I've came up
On Sun, Feb 17, 2013 at 9:20 PM, Paul Coccoli pcocc...@gmail.com wrote:
This scheme sounds error prone. In general, copying C++ objects via
memcpy (or writing them 1 byte at a time into the ringbuffer, which is
what I think you're proposing) is a bad idea.
Nope, write them one sizeof(
On Sun, Feb 17, 2013 at 9:20 PM, Paul Coccoli pcocc...@gmail.com wrote:
JACK ringbuffers are
ideally suited to passing simple types (like floats), and not vairable
sized things (like different derived Event classes). Your enum for
event types is a bit of a red flag, too. While its perfectly
On Fri, Feb 15, 2013 at 11:41 PM, M Donalies ingeniousnebb...@cox.netwrote:
There's no here's how you use these guys sections.
Hi M!
As a relative new-comer to linux-audio, I struggled with the same issues
you're having.
I had intended to document my own learning of JACK MIDI, and have a repo
On Sat, Feb 9, 2013 at 11:16 AM, John Rigg lad...@jrigg.co.uk wrote:
I will stress that I'm talking about audio engineering tools, not music
creation software here. I do appreciate that users of the latter have very
different requirements.
I was reading your post, about to vocalize the
Hey all!
I have a simple enough question, but I don't know the best practice for
solving it, so figured I'd ask.
There's an LV2 synth running in a LV2 host. The synth exposes its operation
trough control ports.
Option 1:
The plugin can bind incoming MIDI events to these control ports values,
On Wed, Nov 21, 2012 at 11:54 AM, Shani Hadiyanto Pribadi
shaniprib...@gmx.net wrote:
jack_midi_event_get(in_event, midi_buf, event_index);
if ((*(in_event.buffer) 0xf0) == 0x90) // Segfault happens here, on
boolean mask operation
{
_note = *(in_event.buffer + 1);
}
Extra context
On Fri, Nov 9, 2012 at 2:26 PM, Orlarey Yann orla...@grame.fr wrote:
snip is happy to announce the release of FAUST 0.9.54 /snip
Great news! I noticed the LV2 output option on the online compiler, the
rest of the news is great too!
___
On Fri, Oct 19, 2012 at 9:31 AM, 新月如钩 46620...@qq.com wrote:
**
Hi All,
Say hello to everyone!
Hi Spring!
I'm a new member to learn how to write a ALSA application.
Is there a reason that you want to learn ALSA specifically? Or do you want
to start audio programming in
Replying to nobody in particular but perhaps bringing some new things to
the table:
I feel there's a lot going on just-under-the-surface of what most of us
know about. I presume not everybody here is aware of the advances FAUST has
recently made in DomainSpecificLanguage technology. Similary I'm
On Sun, Sep 2, 2012 at 7:32 PM, Edward Diehl di...@umich.edu wrote:
However, if you append _2chan to the plugin library
name (e.g. Pianoteq.so to Pianoteq_2chan.so) , the plugin will output
only 2 channels and work happily in Ardour.
Neat trick, I have Pianoteq and didn't know about it. Of
On Sat, Sep 1, 2012 at 1:34 PM, Emanuel Rumpf xb...@web.de wrote:
conclusion:
1. don't use the kontakt sampler (or other proprietary sw)
2. don't buy kontakt samples
3. support free, libre sample libraries
4. create free, libre sample libraries
this appears as the most effective stategie
? Standalone? DSSI? LV2?
What is necessary to cover 95% of the users?
Thanks,
Florian
On 31.08.2012, at 15:43, Harry van Haaren harryhaa...@gmail.com wrote:
On Fri, Aug 31, 2012 at 1:03 PM, Nils l...@nilsgey.de wrote:
The direct and naive solution would be a reversed engineered kontakt
On Fri, Aug 31, 2012 at 1:03 PM, Nils l...@nilsgey.de wrote:
The direct and naive solution would be a reversed engineered kontakt
sample engine, yes.
Very naive.
The community could approach NI and ask if they're intrested in supporting
a Linux version of Kontact? I volunteer to write the
On Fri, Aug 31, 2012 at 4:27 PM, Paul Davis p...@linuxaudiosystems.comwrote:
A lot of people (even on this list) don't understand the extent to which
*supporting* a piece of software is often a far bigger cost than the
initial development, and providing support for a platform with very few
I use Fons' JAAA for this. It has a freeze button, so when you hear a low
note, you'll see it, then hit freeze, then there's peak analysers that you
can place on the display, and it'll tell you its Hz (and estimate a note).
dB can be read right off the Y axis.
It doesn't analyse the whole song as
On Fri, Aug 24, 2012 at 11:18 PM, Robin Gareus ro...@gareus.org wrote:
Even if you implemented it correctly, overtones or undertones can
mislead ones perception quite easily.
Indeed: I've not taken a spectrum analyzer to the signal yet: but something
feels wierd with it.
Will look at the
On Fri, Aug 24, 2012 at 11:48 PM, Fons Adriaensen f...@linuxaudio.orgwrote:
The easiest way in the case of wavetable synthesis is to upsample
your waves by a factor of say 8, then use linear interpolation.
So the preparation process is:
-record the sounds
-upsample x8
Live playing:
On Thu, Aug 2, 2012 at 10:16 PM, Robin Gareus ro...@gareus.org wrote:
1e-20 ( ~= signal at -400db ) should be inaudible enough.
Done! Works well.
Thanks, it seems that this is the easiest cross-CPU / architecture etc fix,
so I'll go with it.
Thanks all! Learning bout CPU mechanics is
Hi all,
I've working on a LV2 instrument plugin, and it consumes about 1-2% CPU on
idle. When I leave it for about 20 seconds, the CPU usage jumps to 38 / 40
% of a core, and JACK xruns. The code contains IIR's for a reverb effect,
so I'm going to blame this CPU burning on denormal values.
I'm
Thanks all for the replies, I've certainly learnt a lot.
On Thu, Aug 2, 2012 at 9:11 PM, Tim Goetze t...@quitte.de wrote:
I think it's almost always a better idea to add an inaudible DC offset
or a square wave at the block interval or at Nyquist
How small is inaudible? Or better yet, how
I'm pretty intrested in this device:
http://www.acer.co.uk/ac/en/GB/content/iconia-tab-w500
It runs the C50, and is basically a full 32bit computer, this guy installed
Arch on it:
http://ant1antuan.wordpress.com/2011/11/10/tablet-acer-iconia-tab-w500-arch-linux-setup-e-first-configuration/
Hey All,
After some very inspiring conversations at the LAC, I have decided to renew
the efforts to document linux audio programming for beginners. I feel that
although there's a lot of really useful tutorials out there, but there's
still a lack of easy accessible introductory audio programming.
On Fri, May 4, 2012 at 6:07 PM, Harry van Haaren harryhaa...@gmail.comwrote:
Announcing: Open Audio Programming Tutorials!
https://github.com/harryhaaren/openAudioProgrammingTutorials
And the link too this time :) -Harry
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