Hello JACK community,
I’m planning my new audio editing setup (both hard- and software) and before investing in hardware, I’d like to ask for advice.
Here’s what I want to do:
– Both computers run Linux.
– On Computer A, I can browse the web and see Youtube videos, I can listen to MP3 files with Totem and I can do DAW stuff with Ardour.
– Computer B does nothing more than getting all the sound output from A and feed it to the speaker through it’s soundcard.
Here’s my theoretical approach:
– Both computers run JACK.
– A runs PulseAudio in top of JACK.
– A runs with the net backend using netJACK2.
– B has the Net Manager loaded which gets the audio from A and pipes it to the soundcard.
– Every time I boot the two computers, they are ready to work. No daemons I have to launch manually, etc. I want the configuration to be done one time for all.
I got my understanding of JACK through the network from here: https://github.com/jackaudio/jackaudio.github.com/wiki/WalkThrough_User_Net… And this site would also be my guide for setting up the thing.
But I’m not a JACK expert at all. It’s the first time I’m doing something like this. So I want to make shure I got the basics right. And, maybe you now even a better step-by-step tutorial which does exactly what I want.
So, I appreciate any comments on this.
Bye,
xan
Hello,
I am compiling JACK2 from commit 7de15e7a (the same as shipped with Ubuntu
14.04). I need to set --clients to quite high value, and I have found out
that setting --clients=512 causes jackd to segfault while --clients=500 is
fine.
Is that a bug or limitation by design?
Marcin
Hi,
I'm writing a jack midi port for incudine, a common lisp based
realtime audio system. Midi in is already working, but for midi out I
couldn't find information about what kind of time should be supplied
to jack_midi_event_write. Is it in absolute or relative frames?
If it is relative to the current frame/block boundary:
I tested the received time values (using jack_midi_event_get) with
randomly generated midievents sent from pd over a2jmidi and got time
values between 0 and 42 with a jackd setting of 1024 frames/period and
2 periods. I would expect values between 0 and 1023, or is there some
misunderstanding on my side? Could this be due to a2jmidid?
--
Orm
Hello,
I need to make custom build of JACK that will reside in /opt/myjack
Now I have to use LD_LIBRARY_PATH because without one I get errors like this
/opt/myjack/bin/jack_lsp: error while loading shared libraries:
libjack.so.0: cannot open shared object file: No such file or directory
I would like to use -rpath linker flag, but I found no clean way to use it.
I tried to set LDFLAGS and other env vars but waf keeps ignoring it.
Any advice?
Marcin
Hi.
I've been sinking lots of time in trying to get ALSA to work with a
SGTL5000 (Teensy Audio Adapter) in Raspbian on a Raspberry Pi 3. I have the
complete documentation for how to manually set up the audio chip, and
sufficient documentation on how to manually set up the Pi's ARM I2S
peripheral. The ALSA solution is generalized for any kind of hardware you
could want, is something I don't really want to know, at all, in my life,
and seems just far too complex when I have detailed guides for both the
audio chip and the CPU. The one combination of ALSA platform, machine and
codec drivers that I've managed to get sound from plays more than 2x too
fast and will not open with JACK.
So, I'm reading about the JACK API, wondering about simply writing a C
program to configure and write to the Pi's built-in I2S FIFOs, without even
telling ALSA that I'm doing it; but I see that all JACK clients should use
callback functions. Still, audio input from JACK into a program using the
API is a regular thing, right? So how complicated might it be to set up
such a client that gets audio input from JACK and writes that to the chip's
I2S FIFO? My timing will be controlled by the audio chip, the Pi chip will
be a slave to it. So as long as my program writes to the transmit FIFO fast
enough, and waits if the FIFO is full, I think I don't need to worry about
synchronizing JACK exactly to the audio chip's clock... right?
I'm just a little fuzzy on this timing part.
Thanks for any clarification.
-Chuckk
--
http://www.badmuthahubbard.com
Anyone have an idea why jack1 displays xruns like this on my system
(for a long time already, not a recent issue)?
**** alsa_pcm: xrun of at least -1386461.056 msecs
The system is Archlinux with jack 0.124.1.
--
Joakim
What about aliases ?
Dnia Czwartek, 12 Maja 2016 08:22 Hanspeter Portner <ventosus(a)airpost.net> napisał(a)
> On 12.05.2016 06:27, aledez0 wrote:
> > is there any way to list the connections by how there are in the project file
> > instead of by alphabetical order?
>
> YES (theoretically)
>
> There is a way, by means of the metadata api [1] via the metadata key
> 'http://jackaudio.org/metadata/order' [2], but the trio of JACK client,
> server and patchbay must support this for it to work.
>
> * the client sets the order of its ports via 'jack_set_property'
> * the server stores the metadata in a database
> * the patchbay reads the metadata to order ports accordingingly
>
> NO (practically)
>
> * only few clients and patchbays support this yet
> * JACK1 implements the metadata API, JACK2 still does NOT
>
> > in mixbus 3 once you have named the tracks
> > they don't show up in the connection window in the order they are in the
> > project window.
>
> [1] http://jackaudio.org/api/group__Metadata.html
> [2] https://github.com/drobilla/jackey/blob/master/jackey.h
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