That is such good news. What(low cost) hardware would this development be
used on to support the developers with testing/debugging and maybe even
development ?
* MOTU LP32 (Preferred)
* MiniDSP https://www.minidsp.com/products/network-audio/avb-dg (I think
MOTU's switch uses midDSP switch hardware)
I hope someday it will be possible to connect 4 or more 8 channel ADAT
modules (32 channels) to a PC under Ubuntu via AVB with low latency. The
only option to get this done under Windows is a Focursrite DANTE based
Rednet 3 right now because Thunderbolt is not really available there as
well. Plan to get Rednet3, but that does not solve the Linux environment
which I prefer. Would love to be able to use the Rednet 3 under Linux but
since DANTE is proprietary , so unlikely.
My two wishes:
[a] Multi (16+) channel low latency audio I/O using ADAT audio AD/DA
[b[ Bitwig supporting LV2 plugins.
With those two, the Linux Audio environment would be perfect and the world
a better place.
*(Apology for the re-sends and ignore the previous edits. Web based Gmail
is such a annoyance and un-logically structured)*
Hi!
I have some questions about the minimum and maximum values of a port's latency
(i.e., its latency range). The documentation states the following:
"Both latencies might potentially have more than one value because there may
be multiple pathways to/from a given port and a terminal port."
Does this also apply to a setup with two output ports "A" and "B"
(representing two distinct physical devices) with different capture latency,
which are both connected to the same input port "C"?
Here are some details of an example setup (useful for demonstration purposes,
but probably not for doing any real recording work) as reported by jack_lsp:
"A" (Focusrite Saffire PRO 14 Firewire audio interface):
firewire_pcm:00130e0402403491_1394/Out:01 (Mic/Lin/Inst:01)_in
port latency = 337 frames
port playback latency = [ 0 1792 ] frames
port capture latency = [ 337 337 ] frames
properties: output,physical,terminal,
"B" (Line6 PODxt Live guitar effects processor synchronized by zita-ajbridge):
line6_in:capture_1
port latency = 867 frames
port playback latency = [ 1792 1792 ] frames
port capture latency = [ 867 867 ] frames
properties: output,physical,terminal,
"C" (Ardour track):
ardour:track 1/audio_in 1
port latency = 0 frames
port playback latency = [ 1792 1792 ] frames
port capture latency = [ 867 867 ] frames
properties: input,
Under the assumption that the physical inputs of "A" and "B" represent the
same time scale, the correct answer for the port capture latency at port "C"
would be [ 337 867 ] and not [ 867 867 ].
So my questions are:
*) What is the rationale behind jack reporting the larger of the two latencies
of "A" and "B" both as the minimum and maximum latency of "C", and ignoring
the smaller one?
*) Is it possible to provide additional information to jack to state that the
signals at "A" and "B" originate from the same physical source (or from
different physical sources sharing the same time scale), such that both
latencies of "A" and "B" are taken into account for the latency range of "C"?
Thanks & kind regards,
Markus
Hello
I am working on a Java program special for visual impaired persons
that needs to detect weather the headphone jack plugs on Windows-7.
can Jack detect it?
Jack can notify headphone plugging by a call back method?
I will be appreciate, if you would how to do that via Jack, if possible.
Shadyar Khodayari
