Re: using Jack an interface to ecasound

Tweed <tweed@xxxxxxxxxxxxxxxxxxx> · Tue, 17 Jan 2017 10:41:48 -0500

    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:

                          ...

                             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.

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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.

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