JackAudio February 2018

jackaudio@lists.linuxaudio.org

21 participants
16 discussions

by Stefan Hellkvist

Hi list, (Sorry this question has already been discussed but the search function in the archive of the list does not work) Is anyone working on the problem of getting Jack up and running on MacOS again? I installed Jack with brew but I get these “could not handle external client request” errors which seems to be old problems from looking at search results and GitHub discussions - problems introduced when MacOS audio architecture was changed a while back. Is there any progress on these matters? Is someone working on it? I need Jack as it is a dependency to another library I would like to use but I realize that I might have to abandon MacOS as a platform in the process. Kindest regards, Stefan

5 years

Re: [Jack-Devel] AES67 / SMPTE ST 2110-30

by happy musicmaker

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)*

5 years, 9 months

[Jack-Devel] The playback device hw:0 is already in use

by Tim Janes

I am new to jack and trying to setup Jack2 on Ubuntu 17.10 on a Dell laptop 15-5570 that uses a Realtek ALC3246 sound device. It is basically working ( I am testing it with simple_client and latent_client) but I think that I am doing something wrong as the first time I use it after booting the laptop I get the error. "ATTENTION: The playback device "hw:0" is already in use. Please stop the application using it and run JACK again" followed by lots of "Cannot connect to server socket err = No such file or directory" "Cannot connect to server request channel" fuser -v /dev/snd/pcmC0D0p shows that it is use by pulseaudio. However the 2nd and subsequent runs of the jack example clients work just fine without any error. Thanks for any suggestions or help Tim.

6 years, 2 months

[Jack-Devel] Why can I only select 512 samples?

by lowkeyoutlaw

In Mixbus, I noticed that if I turn the latency up to 1024 samples, my DSP usage goes from 33% to 23% or so. The problem is, 512 samples is the only setting that doesn't produce nasty distorted audio, lower or higher are all bad. In the MPC software I can't even change it, only in Mixbus. I thought it was maybe due to a rate mismatch. Is there any way to increase the latency to 1024 and not get that nastiness? Thank you. -- Sent from: http://jack-audio.10948.n7.nabble.com/Jackit-f3.html

6 years, 2 months

Re: [Jack-Devel] Handling adaptive number of input samples required for adaptive resampler

by Chris Caudle

On Sun, February 25, 2018 5:53 am, Benny Alexandar wrote: > I have a requirement to do a re-sampling based on drift detected on jitter > buffer. zita-alsa bridge already does something similar, you could look at that source as an example. http://kokkinizita.linuxaudio.org/linuxaudio/zita-ajbridge-doc/quickguide.h… Uses zita-resample library, you will need that along with ajbridge source. Fons also wrote a paper explaining how it works: http://kokkinizita.linuxaudio.org/papers/adapt-resamp.pdf -- Chris Caudle

6 years, 2 months

[Jack-Devel] Handling adaptive number of input samples required for adaptive resampler

by Benny Alexandar

Hi, I have a requirement to do a re-sampling based on drift detected on jitter buffer. Suppose I have set jack process callback to 128 samples, and I detected that I have to do an resampling by 10%, ie n_in = 1.1 * nout then I need 12.8 samples more, which is not available and has to wait for next callback. This will delay my processing. If I make my buffer size more than 128, I may have to do a decimation based on drift in which direction. How to handle these cases of having varying buffer size requirements ? -ben

6 years, 2 months

Re: [Jack-Devel] Midi out no longer showing up in patch bay

by Chris Caudle

On Thu, February 8, 2018 11:12 am, lowkeyoutlaw wrote: > Mpc is a stand alone application. There is no way to route > audio and midi from it to any other application without > something like VAC or Jack. > Mixbus' routing doesn't see the MPC audio outs at all on its own. Then I don't understand what you wrote previously: "Mixbus can see the MPC's midi out, but Jack cannot." You did not show how you started jackd, are you using the -X winmme argument to start the MIDI driver? -- Chris Caudle

6 years, 2 months

[Jack-Devel] jack_iodelay ported C source code

by Benny Alexandar

Hi, I'm looking for the ported C source code of jack_iodelay. At the end of ma page it says "Originally written in C++ by Fons Adriaensen, ported to C by Torben Hohn." Please share the link for this C source code. -ben

6 years, 2 months

[Jack-Devel] Stupid question

by Judas

If I need help with Jack, do I post here? --Dean Who's trying to get Internet DJ Console running.

6 years, 2 months

[Jack-Devel] zita-a2j source code

by Benny Alexandar

Hi, Where can I get the zita-a2j/j2a ported C source code. Please send me the link. -ben

6 years, 2 months

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JackAudio February 2018