17 Jan
2017
17 Jan
'17
4:42 p.m.
On 01/17/2017 09:31 AM, john gibby wrote:
I think my AVS implementation is not using Jack2. (I
say this b/c I
tried to use the -S option for synchronous running, and jackd didn't
recognize it.) From what I read, seems worth it to install Jack2, so
I plan to do that... unless you think a bad idea :)
That zita-Irx software looks, well, incredibly rich. I guess I may
try to switch to it instead of ecasound... Amazing, all these great
resources in the Linux world...
Thanks again,
John
On Tue, Jan 17, 2017 at 6:17 AM, Tweed <tweed(a)lollipopfactory.com
<mailto:tweed@lollipopfactory.com>> wrote:
On 01/17/2017 04:35 AM, john gibby wrote:
If jackd doesn't recognize "-S" then its jack1. whats jackd -v show.
0.12x is jack1, 1.9.x is jack2. If you have jack1 its not a dbus
issue. maybe another running audio process (alsa-loop daemon?
pulse-jack whatever its called?) not allowing jackd to stop(not sure if
thats right tho seems like I've seen this with aj-snapshot). I use
jack2 (unrelated reasons). I don't know anything about avs linux (I
first thought you meant Av-Linux - a great audio distro), be careful
when changing out one JACK for another as your package manager may try
to remove things you don't want removed. Anyway, there shouldn't be any
reason related to your problem that you need to switch JACK1 <> JACK2.
Try to close any jack clients then stop the jack server. at this point
you should be able to stop jack with qjackctl if not, killall jackd on
command line. "top" or "htop" or "ps -ef | grep jackd" to
see if
jackd has stopped or not. once you confirm jackd has stopped, try to run
your ecasound command.
--
www.the-temp-agency.com/lollipop-factory
Sound is via ALC 1150 chipset; I don't
think that's the problem.
When I go directly from pianoteq to alsa there's no problem; can
use even a 64 sample buffer. Maybe I need a little help in
killing the default jack server and starting it back (with dummy
back end ) using direct jackd command line instead of using
qjackctl? Then I think it may keep my specified buffer size. Am
Linux newby, takes a little work! :)
On Jan 17, 2017 4:23 AM, "Jeanette C." <julien(a)mail.upb.de
<mailto:julien@mail.upb.de>> wrote:
Jan 17 2017, john gibby has written:
...
When qjackctl brings up
the jack server, the buffer size gets overridden to
1024; I see the message
in the log. What am I doing wrong? Is Jack the wrong
approach, when it is
ecasound, not jack, that writes to alsa?
Hi John,
it appears that your soundcard is the problem. I've only
started JACK on
the commandline or through a dedicated start script, not
using qjackctl
or other JACK-supplied tools. But if you give a buffersize to
JACK it
will honour that buffersize, if the soundcard can stand it. I
haven't
seen an application before that couldn't honour JACK's
buffersize,
whatever it is. Especially Ecasound can certainly go down to
64 samples.
What soundcard do you have? Have you tried starting JACK for your
soundcard on the commandline and see what happens?
jackd --timeout 4500 -R -d alsa -d hw:0 -p 128
Assuming that your soundcard is the first one (hw:0).
I have no experience with Pianoteq, but since it is meant as
a realtime
app, it should make sure that its sounds are played back
without delay
or with minimal delay. 128 and even 64 samples aren't that
uncommon.
...
Best wishes,
Jeanette
--------
When you need someone, you just turn around and I will be
there <3
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maybe a jackdbus thing? if you're using jack2, what does
"jack_control status" show?
if it says "started", do "jack_control stop" then try your jack
command/qjackctl.
--
www.the-temp-agency.com/lollipop-factory
<http://www.the-temp-agency.com/lollipop-factory>
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