I've been having problems with connections between jack clients dropping
out.
I would like to have a program that monitors the state of connections.
I've worked out how to locate the end client ports, but I don't want to
have
poll the connections at frequent intervals. Is there any way I can get a
callback when either of the clients drops out?
I suppose I could set up links between by monitor and the two client ports
and activate those connections, but that seems like overkill. Is there
an easier way.
I'm actually using pipewire-jack but don't know enough about pipewire to
do it
directly.
Bill
--
+---------------------------------------------+
| Bill Purvis G8DIO |
|bill(a)billp.org |
+---------------------------------------------+
Bill Purvis:
>
> poll the connections at frequent intervals. Is there any way I can get a
> callback when either of the clients drops out?
>
The function jack_set_graph_order_callback does this. The callback is
called whenever the graph of connections is changed. Then you can use
other jack functions to find out if your client has dropped out. I
don't know if writing a c program is an option for you, but there are
probably higher-level languages that have support for jack.
https://jackaudio.org/api/group__ClientCallbacks.html#gacd0804ccef7c6891d82…
(Sorry if someone else has already replied this solution (I'm
subscribing to the mailing list in digest mode), but I could not find
any web archive of the linux audio mailing list working! Is there any
left?)
Hi,
I've just bought an ESI-Audio Gigaport-Ex to provide 8 output channels.
When I connect up to my Laptop which is running Debain 12 with Pipewire,
it appears on qpwgraph, but only shows two channels (Front-L + Front-R).
How can I persuade it to show all 8?
Thanks,
Bill
--
+---------------------------------------------+
| Bill Purvis G8DIO |
|bill(a)billp.org |
+---------------------------------------------+
Hi,
I am undertaking a new recording project of my spoken voice. I am
sorting through the many plugin options. I am writing to solicit signal
chain recommendations, please
I am happy with jackd and qtractor. Here are the plugins I have running
on my computer.
-DAW PLUGINS-
autotalent
calf-plugins
ir.lv2
jamin
lsp-plugins
pulseaudio-equalizer
pulseeffects
rev-plugins
swh-lv2
zita-at
zita-rev1
Please be so kind as to share your recommended signal chain for this
project.
Kindest Regards,
Stephen
Hi list,
excuse this question as being partially off-topic, as not sound is
involved, but since some time (possibly since the last big Debian
upgrade), mpv complains about the video backend only being available as
"software renderer" and not being able to use "vo=gpu".
I run this on a Lenovo X230 with an Intel card and the i915 kernel
driver. I have learned that for older intel hardware (than my card?)
users are recommended to use xserver-xorg-video-modesetting X11 driver
instead of the the xserver-xorg-video-intel one. And indeed, I can
switch back and forth between these two by (not) forcing the latter in
Xorg.conf as
Driver "intel"
With the modesetting driver, mpv plays flawlessly using the vo=gpu
output but Gem on PureData as well as glxgears stutter every few
seconds. Having researched the web further, I found discussions about
different performances of the two drivers.
However, I would simply like to have mpv play nicely with the intel
driver. Here is its output, that I can#t really make sense of despite
having searched the web for a long time. Perhaps anyone on this list is
able to help me further?
Thanks a lot! Peter
------------>
$ mpv --no-config test.mp4 2> /tmp/output.txt
● Video --vid=1 (h264 960x544 30.0001 fps) [default]
● Audio --aid=1 (aac 2ch 48000 Hz 192 kbps) [default]
[vo/gpu/opengl] Suspected software renderer or indirect context.
[vo/gpu/drm] VT_GETMODE failed: Inappropriate ioctl for device
[vo/gpu/drm] Failed to set up VT switcher. Terminal switching will be unavailable.
[vo/gpu/drm] Failed to acquire DRM master: Permission denied
[vo/gpu/drm] Failed to commit ModeSetting atomic request: Permission denied
[vo/gpu/opengl] Failed to set CRTC for connector 80: Permission denied
[vo/gpu/drm] Failed to commit ModeSetting atomic request: Permission denied
[vo/gpu/drm] Failed to restore previous mode
[vo/gpu/drm] Failed to drop DRM master: Permission denied
[vo/gpu-next/opengl] Suspected software renderer or indirect context.
[vo/gpu-next/drm] Can't handle VT release - signal already used
[vo/gpu-next/drm] Failed to set up VT switcher. Terminal switching will be unavailable.
[vo/gpu-next/drm] Failed to acquire DRM master: Permission denied
[vo/gpu-next/drm] Failed to commit ModeSetting atomic request: Permission denied
[vo/gpu-next/opengl] Failed to set CRTC for connector 80: Permission denied
[vo/gpu-next/drm] Failed to commit ModeSetting atomic request: Permission denied
[vo/gpu-next/drm] Failed to restore previous mode
[vo/gpu-next/drm] Failed to drop DRM master: Permission denied
[vo/vdpau] VDPAU is most likely emulated via VA-API.
[vo/vdpau] This is inefficient. Use --vo=gpu instead.
[vo/vdpau] Warning: this compatibility VO is low quality and may have issues with OSD, scaling, screenshots and more.
[vo/vdpau] vo=gpu is the preferred choice in any case and includes VDPAU support via hwdec=vdpau or vdpau-copy.
Hello all,
I found no solution for this in the Ardour manual:
- I import a MIDI file, using 'One track per channel'.
- This results in 3 tracks being created, apparently corresponding
to the two manual and one pedal parts of this organ work.
- I want to send the MIDI data from these to Aeolus, each track using a
different MIDI channel.
Problem: all three tracks output on channel 1, and the per track
channel selectors in the Edit window don't have any effect.
What am I doing wrong ?
TIA,
--
FA
Hi
The last day's I'm working on a little tool which allow me to generate a
SondFont 2 to be used with fluidsynth from a single audio file.
It allow now to clip out a part of a file to be used as OneShoot
instrument sample, select a part to be used as Looped Instrument. It
detect the dominant Frequency and set the RootKey and PitchCorrection to
be used in the generated SoundFont. Additional it allow to set the
values for Chorus and Reverb.
This allow to generate quickly a well tuned SoundFont from any sample
you've laying around, or grep out a sample from a audio file to use it
as Soundfont.
Note, it will use only the first channel of a audio file, regardless how
many channels it have.
Supported audio file format is what ever libsndfile supports.
Testers been welcome.
https://github.com/brummer10/sf2generate
regards
hermann
Hello everyone,
I need some recommendations for a portable hardware audio player (mp3, ogg, flac) that supports Linux, and is of reasonable quality. A quick search on Amazon pulls up everything from $18 - $200, and I'm afraid to pull the trigger without some kind of recommendation. My needs are simple, I just need to put tons of practice tracks on it and plug it up to my keyboard via some kind of audio jack (1/8"). Power on, select a track, play, experience musical disappointment, power off. :)
General brand recommendations are useful, pretty much any info is more than welcome - my trusty mp3 player that was recommended by this list many years ago has vanished, so I must replace it.
Thanks everyone!
Josh
