A while ago I initiated work on a Linux driver for Line6 devices (now under sound/usb/line6 in the Linux kernel tree). Some of these devices have weird sample rates (39062.5Hz, derived from the 10MHz USB clock divided by 256). The alsa_in and alsa_out jack clients are convenient tools to use such Line6 devices together with more standardized hardware operating at, e.g., 48kHz. However, alsa_in only supports integer sample rates, and even after a couple of minutes, alsa_in doesn't detect the correct resampling factor 1.2288 (48000/39062.5), but still reports the initial approximated value 1.228816 (48000/39062). Admittedly, the error isn't too big, but is larger than the error of high-quality crystal oscillators. And, after all, why use an approximation if we know better?
I modified alsa_in and alsa_out such that they accept a sample rate in floating point format (command line option "-r") and query the fractional sample rate from the ALSA driver to compute the initial estimation of the resampling factor (see attached patch).
What do you think about this patch?
Hello JACK community,
I’m planning my new audio editing setup (both hard- and software) and before investing in hardware, I’d like to ask for advice.
Here’s what I want to do:
– Both computers run Linux.
– On Computer A, I can browse the web and see Youtube videos, I can listen to MP3 files with Totem and I can do DAW stuff with Ardour.
– Computer B does nothing more than getting all the sound output from A and feed it to the speaker through it’s soundcard.
Here’s my theoretical approach:
– Both computers run JACK.
– A runs PulseAudio in top of JACK.
– A runs with the net backend using netJACK2.
– B has the Net Manager loaded which gets the audio from A and pipes it to the soundcard.
– Every time I boot the two computers, they are ready to work. No daemons I have to launch manually, etc. I want the configuration to be done one time for all.
I got my understanding of JACK through the network from here: https://github.com/jackaudio/jackaudio.github.com/wiki/WalkThrough_User_Net… And this site would also be my guide for setting up the thing.
But I’m not a JACK expert at all. It’s the first time I’m doing something like this. So I want to make shure I got the basics right. And, maybe you now even a better step-by-step tutorial which does exactly what I want.
So, I appreciate any comments on this.
I'm trying to get receive audio data from a USB software defined radio and
pipe it through jack to alsa out and also to another app dl-fldigi.
It's working (off a fashion) using this hacky script
starts the jack server
starts the 3 apps to link (mplayer --> alsa_out --> dl-fldigi)
links mplayer to alsa then also to dl-fldigi
I have 2 questions:
Some of the apps either crash occasionally (dl-fldigi) or their is a case
for restarting them (mplayer), at the moment I have to recreate the
connections each time an app closes. Is it possible to create the
connections first and have them persist, then connect the apps when they are
Secondly I have some audio issues, there are fractional pauses every so
often which make the audio had to decode, is there anything about piping
through jack that could cause this, on something which might mitigate it?
View this message in context: http://jack-audio.10948.n7.nabble.com/Linking-apps-together-from-script-tp1…
Sent from the Jackit mailing list archive at Nabble.com.
A Radium user can't get sound out of his firewire soundcard.
Here's the mail:
I seem to be too stupid to setup Jack OSX with Radium 4 in my OSX 10.9.4
I never hear anything. output 3+4 of radium are routed to system default
1+2, which should be my firewire audio interface.
Anyone knows what could be wrong?
i am working with some virtual systems.
i have a linux machine using pulse to output audio (i can't change this).
i have a windows machine without sound.
i followed this tutorial to stream audio from the windows to the linux
my problem is:
the linux machine is also virtual and has no alsa device.
all audio is send per pulse to the real audio output (as already
mentioned, i can't change this).
how can i configure jack to send its output to pulse? (i know, normally
people do this the other way around)
i am grateful for any hints.
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I have a Linux computer that is running a recording. I'd like to be able to
also stream what is recorded to another computer through the network and
listen to that with the speakers. I'm using Jack 1. I'm running jackd -d
net on the slave computer, and jack_netsource on the recording one. Then,
to play the sound through the speakers, I run alsa_out on the slave
computer and connect the appropriate ports.
Now I'd like to be also able to listen on a computer with OSX (the
recording computer stays the same). I've installed jack from Macports, but
there is no Alsa for OSX. What can I do to get the same result?
---------- Forwarded message ----------
From: "George Martínez" <latenciaalcontacto789(a)gmail.com>
Date: Jul 9, 2016 3:49 PM
Subject: MIDI in Jack
I will love to know how to use the Jack application with MIDI
Do I have to download virtual midi cables for that? Does it provides low
Thank you and great app by the way
Currently I am hacking in Jack transport support for a certain project.
And while doing that, I am wondering whether it is possible to get a
notficiation when the current timebase master has stepped down.
Bascially what I have in mind for my client:
* During startup, check if there is a timebase master and if not, be it
(which I know is possible with jack_set_timebase_callback(..,1,...)).
* In case there was another timebase master that resigns at a certain point
in time (leaves the graph, steps down for some reason,...),
take over the timebase.
I couldn't figure out a way to get a notification (callback etc.) that the
current timebase master just has stepped down.
Is there one? If not, is there a specific reason for that?
Your help would be highly appreciated.
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I've been happily using the pre-installed Jack2 that comes with Ubuntu
Now I require the ability to use more than 64 clients then tried to
compile Jack2 from the package JACK 1.9.10 found at jackaudio.org/downloads
I'm running on a 64 bits machine but would like to get support for 32
bits applications as well.
Configure runs fine with ./waf configure --prefix=/usr --dbus --profile
--mixed --clients=256 --ports-per-application=2048 --alsa
Then build fails:
/usr/bin/ld: cannot find -ldbus-1
I know it has something to do with the 32 bits library of dbus but need
help to find out how to get them properly.
I have already the following installed:
sudo apt-get install build-essential gcc-multilib g++-multilib
sudo apt-get build-dep jackd2
sudo apt-get install libdbus-1-3:i386
Am I still missing some libraries?
Is it about some environment variable?
Is it necessary to compile libdbus?
Thanks for your input.