Thanks!! I can answer some of these questions, though am at work w/o access to my system. I think Tweed is correct that I'm probably running jack1. I *am* running a recent copy of Av-Linux, don't know where I got AVS from, sorry. And I agree with Ralf about user error... am definitely pretty clueless, but trying to learn... I won't complicate things by trying to switch to jack2, not yet anyway. I do seem to be able to stop and start Jack via qjackctl. I *have* configured Setup in qjackctl for the desired 128 sample buffer. qjackctl starts jackd, not jackdbus. I will send the version later when I get home. One weird thing is that even after stopping jack via qjackctl, if I then try to start it back from the command line, it says "default jack server" is already running.
Seems to me my main confusion/going wrong has to do with trying to insert jack BEFORE ecasound, i.e. using "dummy" backend instead of alsa. When I don't do this, a problem occurs but just now I can't remember exactly how to describe the problem. When I use "dummy" as backend, everything seems to work almost perfectly; I see pianoteq in qjackctl, I send 2 channels of pianoteq to the jack outputs, and ecasound picks them up and changes them to 6 channels correctly (I hear correct signals on low/mid/high analog outputs). Everything works, EXCEPT, in the pianoteq output dialog where I select "jack", I see it's using 1024 samples instead of the desired 128, and 1024 is the only choice... And I also see, in the message log of qjackctl after I start jack, where it says something like "overriding period size to 1024". So even though I set the sample size to 128 in qjackctl setup for dummy backend, and I set -i jack -b 128 in ecasound as well, looks like I end up with a 1024 sample buffer going into Jack....
Thanks again,
John
On Tue, Jan 17, 2017 at 10:41 AM, Tweed <tweed@xxxxxxxxxxxxxxxxxxx> 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.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@xxxxxxxxxxxxxxxxxxx> wrote:
______________________________On 01/17/2017 04:35 AM, john gibby wrote:
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@xxxxxxxxxxx> wrote:
Jan 17 2017, john gibby has written:
...
Hi John,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?
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
_______________________________________________ Linux-audio-user mailing list Linux-audio-user@lists.linuxau dio.org http://lists.linuxaudio.org/listinfo/linux-audio-user 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.
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