since recent kernels provide SCHED_DEADLINE, i'm porting my code to make
use of it.
* is there any plan to migrate jack1 and/or jack2 to use this scheduling
policy for the jack thread(s) of the application?
* what is the recommended way to set the scheduling policy of the jack
thread with the current API? afaict, the JackThreadInitCallback is
called before JackClient::AcquireSelfRealTime in jack2. (i cannot use
jack1 due to the old bug that jack corrupts the stack while trying to
pre-fault it)
thnx,
tim
Hi,
A new 64bit iso is up ;)
io GNU/Linux is a Live DVD/USB based on Debian Sid and focused on multimedia.
Kernel 3.14.4, Jack2 as default sound server, e18 as desktop environment and a
big collection of installed software... Full persistence for USB install (with
encryption) and more cool stuff... A great nomade studio :)
For more infos: manual, packages list, screenshots, video etc... Check:
-> http://manu.kebab.free.fr/iognulinux.html
-> https://sourceforge.net/projects/io-gnu-linux/
Feedbacks welcome, enjoy :)
MK
Hey Guys,
I'm new with LV2 and have some troubles: I'm familiar with LADSPA and I
normaly link my plugins with my .asoundrc to alsa. I want to apply this
plugin not only to one application, I want to use it for all aplikctaions
like aplay or anything else.
Is there a possibility for LV2 Plugins, to use the system-wide?
Does anyone have experience?
Thanks in advance,
Martin
Hey All... Again,
This has indeed caught my interest...
After doing some (most likely not enough) reading, I am left a little
curious for how some things happen.
In terms of connection management and SIP under unicast routing, I can see
a little bit how something like the Dante controller could happen. I am
guessing a third party (ie the Dante Controller like thing) could tell two
user agents to coneect with a REFER method. However when it comes to
instructing an existing session to stop, I am at a loss. Does anybody know
if there is a particular SIP method to get an existing SIP session to stop?
An interesting thing that cought my eye about the standard is that it seems
what discovery method is chosen is left open to the implementor.
Avahi/Bonjour seems to be a natural choice, but I am not quite sure how
"interoperable" all these Audio oIP standards would be if they all use
different discovery methods. Just rambling...
On Sun, May 25, 2014 at 4:11 PM, Jamie Jessup <jessup.jamie(a)gmail.com>
wrote:
>
> *There is a proprietary ALSA driver for (at least) some Dante Cards,
> butyou have to ask your reseller specifically for it and pay an additional
> fee.*
>
>
> Good to know, have you had any experience with the proprietary stuff?
> Whats the experience like in comparison to the Mac/Windows equivalents?
>
>
> On Fri, May 16, 2014 at 11:59 AM, Markus Seeber <
> markus.seeber(a)spectralbird.de> wrote:
>
>> On 05/16/2014 05:37 PM, Jamie Jessup wrote:
>> > I would love to see this happen in the world of LA. I was recently about
>> > to embark in a HW project for an AES50 implementation, however after a
>> > quick skim of the standard this sounds a little more exciting :).
>> >
>> > A Dante "Virtual Soundcard" style thing for ALSA/JACK would be amazing.
>>
>> There is a proprietary ALSA driver for (at least) some Dante Cards, but
>> you have to ask your reseller specifically for it and pay an additional
>> fee.
>>
>> -M
>> _______________________________________________
>> Linux-audio-dev mailing list
>> Linux-audio-dev(a)lists.linuxaudio.org
>> http://lists.linuxaudio.org/listinfo/linux-audio-dev
>>
>
>
>
> --
> Jamie Jessup
>
> http://jamiejessup.com
>
--
Jamie Jessup
http://jamiejessup.com
Goal of this tool is to allow to search for audio plugins and for other
resources types (sampler banks, instrument definitions, probably more)
independently of application, needing it. I'm still thinking about final name
for it, before to publish (since it should support not only plugins, but also
instruments), and unsure even where to create home page for it (will describe
in bottom).
As for plugins: currently each audio application has own plugin browser. In
some cases it is very handy (ingen, carla), sometimes awful (audacity, may be
more). My hope is to make it external with filemanager-like workflow.
Currently there is one implemented way to select plugins: dragging. Drag
object is some resource URL, which should be enough for drag receiver to
exactly find pointed resource. But unlike file managers, it is threated as
text - text editors just paste it instead of try to open and konsole - same,
instead of to ask, what to do, as it always does for dropped files. Of course,
applications are free, how to threat such resources, and apps could check for
content before to decide, what to do with it - i.e., check, is it url, etc.
Dragging to jack patchbay canvas may be used to load plugin in single host
(not sure, what about instrument, since there is usually separate sampler
engine, usually managing all instruments).
Proposals about drag object format and url types for each resources are
welcomed. I have idea, how to implement for ladspa/dssi, but unsure for lv2.
Currently there are following formats:
LADSPA: ladspa:/<path to library>/<label>
DSSI: same, but begining with "dssi:/"
LV2: lv2 plugin url (i don't know, is it possible to load plugin from specific
library, and they recommend to identify plugins only by URL - at least
remember URL's for database, used across sessions).
Not implemented:
Just have to learn VAMP and somehow try VST - for some completeness.
For soundfonts, gig banks, and instrument definitions there may be same way as
for ladspa/dssi. Sampler banks mode should be handy for
linuxsampler/fluidsynth gui, not providing any instrument browser (qsampler
doesn't have it, and linuxsampler database supports only gig format).
Screenshot: http://wstaw.org/m/2014/06/05/plasma-desktopau6238.png
About publishing. I have account on savannah, and want to publish it there.
But i'm going to add vst support, and i'm not sure, is it possible (even if
licensing is kept, how about fact itself, that it supports this feature)?
Dear musicians, programmers and normal people,
I plan to create a user group, a regular meeting, in my home town
Cologne, in Germany.
The first meeting ever will be already on June 18th, 19:00.
After that every two month or so. There are more dates on the website
(see below).
Here is a brief website with all the necessary information.
http://cologne.linuxaudio.org/ (Any language is welcome but the chances
are that most people will be from the area and therefore speak German.
So the page is in German)
If you intend to come you can put your name on this etherpad, but this
is not required. Anybody can show up.
http://yourpart.eu/p/linuxaudio-cologne
Topics will be unorganized Q&A, showing off programs and music, sharing
knowledge and tips and hopefully one day shorter or longer
presentations, tutorials, workshops etc.
I expect most people to use Linux but any OS is welcome, therefore I
named it just "Open Source Audio" and not Linux Audio.
So if you are in the area please join us! If you are not in the area but
know people in the area, please tell them.
Greetings,
Nils
http://cologne.linuxaudio.org/http://www.nilsgey.de
P.S.
Despite the domain saying linuxaudio.org this is an independently and
privately organized event. It is not intended to replace or get in
conflict with the Linux Audio Conference.
The FFADO project (www.ffado.org) is pleased to announce the release of
FFADO 2.2, the userspace framework supporting Firewire audio interfaces
under Linux. While the focus of this version has been bug fixes and
improvements to existing drivers, support for some additional devices has
been added.
FFADO is brought to you thanks to the work of Daniel Wagner, Pieter Palmers,
Philippe Carriere, Adrian Knoth, Arnold Krille, Takashi Sakamoto, Jonathan
Woithe and users who have tested FFADO against their devices, provided
patches and given suggestions.
Changes and additions in FFADO 2.2:
* Many bugs fixed
* Mixer, router and monitor support for Saffire Pro 14
* Mixer and router support for Presonus Firestudio Tube, Presonus Firestudio
Project, and M-Audio Profire 2626
* Support Echo Audio AudioFire12 with firmware versions 5.0 and later
(addresses ticket 360)
* Echo Audio device mixer additions:
- hide SPDIF mode switch on AudioFire12
- digital interface switch on AudioFire8 and Pre8
- phantom power switch for AudioFire4 (addresses ticket 364)
- playback routing for AudioFire2/4 (addresses ticket 335)
* Presonus device mixer additions:
- better support the FP10 (formerly known as the Firepod)
- implement support for the Firebox and Inspire1394
* M-Audio mixer additions:
- improve support for the Ozonic
- add support for Firewire Solo (ticket 336), Audiophile, 410,
1814 and ProjectMix I/O
* Device-specific mixer for Yamaha GO44 and GO46 interfaces added
* RME Fireface 400/800 improvements:
- mixer and device settings can now be saved to and restored from device
flash
- clock source selection made more consistent
* DICE EAP / RME Matrix mixer enhancements:
- "per output" view with mono/stereo control
- saving of mixer settings to file
- more consistent cooperation with jackd when sample rate is changed
* Audio streaming support added or refined for additional MOTU interfaces:
- Traveller mk 3
- Ultralite mk 3
- Ultralite hybrid (using firewire interface only)
- 4pre
Known issues:
* Saffire Pro 40 at 96 kHz fails to start most of the time (ticket 326)
* Saffire Pro 24 and Pro 40 MIDI problems (tickets 372 and 375)
* Saffire Pro 24 and Pro 40 lack an ADAT/SPDIF switch
* Only mixer control is supported on M-Audio 1814 and M-Audio, ProjectMix
* M-Audio Audiophile, 410 and 1814 require startup workaround. See
http://sourceforge.net/p/ffado/mailman/message/30807938
* M-Audio Audiophile, 410 and 1814 will only work with FFADO when loaded
with the latest firmware
Hi list,
I just discovered openframeworks
http://www.openframeworks.cc/
And it almost looks too good to be true.
For example, have you seen this?:
http://vimeo.com/59653952
Why is ofx not so prolific in lad land? I searched the lad mailing list and found just a handful of mentions over the years.
Who's using it?
Can you recommend it?
Would you have used it if it was available at the start of your project?
Best,
Bart.
studio magnetophon
Bart Brouns
Biesenwal 3
6211 GZ Maastricht
0031-6-22919561
bart(a)magnetophon.nl
www.magnetophon.nl
Hi,
Does anyone here know why I am seeing this error while loading a kernel
module:
Unknown symbol __stack_chk_guard (err 0)
Unknown symbol __stack_chk_fail (err 0)
I can't find anything online that explains the cause of the error in
relation to loading kernel modules.
--
Patrick Shirkey
Boost Hardware Ltd
As well as the jack ring buffer, I've looked at several others now, and their
example code. The most significant thing that seems to be different about them
(from a usage point of view) is the way they handle overflows. However, if the
buffer size is defined as an exact multiple of the data type/structure and only
complete structures are pushed or popped, would I be right in thinking that you
would only need to check on an all/none basis?
Have I missed something that could cause a partial data transfer?
--
Will J Godfrey
http://www.musically.me.uk
Say you have a poem and I have a tune.
Exchange them and we can both have a poem, a tune, and a song.
