TUTORIAL

For a complete description of distributed music we recommend reading at least chapter four of the author's Ph.D. thesis. Regarding artistic challenges and the effect of latency please read chapter five. With respect to the current state of the art and Soundjack in particular please read our IEEE-Access publication (mainly Section IV). Alternatively read this tutorial below as a "quick start".

Please take a look at the videos below - they show you all the details of how to adjust the relevant parameters.

Installation guide

  • Register on soundjack.eu
  • Once registered log in and click on Download tab
  • Download either windows or Mac OS depending on your system. Win users will need an ASIO sound device. If not existing please download the ASIO4ALL driver as well and install it: http://www.asio4all.com
  • Extract and open the downloaded file.
  • Run the soundjack application by doubleclicking on the binary file (Win: soundjack.exe / OSX: soundjack.app) - Important: A small XLR Icon in your dock will show up. This confirms the soundjack engine is running and working fine. On Windows you will see an additional DOS-type console.
  • Once the plugin is running on your machine return to the sound jack website and click the stage tab

Interface layout description

Soundjack is a browser-based low-latency communication system available at the stage tab. The left of the interface shows the local engine specific parameters, which determine the stream settings and in turn the bandwidth capacity of the outgoing stream. The right shows the user list which is to be understood as a mixing colole in horizontal layout. Each user corresponds to a channel strip. Enable the "play" button at the very right of the channel strip in order to connect with the specific user.

Configuration

  • Once you have successfully ran the application you will need to set your drivers/settings within the web browser. This may take some tweaking.
  • Select your the desired input and output interface of the existing ones 
  • Select the sample buffer (if you’re experiencing audio issues try raising the sample size to the next higher)
  • Select send channels. These should correlate to the inputs of your interface. Example inputs 1+2 would use 2 send channels. If you had a mic on in1 and a guitar on in3 you’d most likely need to select 4 channels here.
  • Lastly choose your network buffer and OPUS bit rate. The higher the OPUS bit rate the higher the quality but subsequently the more data will be transmitted which could cause issues with slower Internet connections.
  • Establish a test connection with the existing audio mirrors within the user list. You will see the jitter buffer turning green and/or red depending on the stability of the connection. Adjust this value according to your needs.

 

Lower network buffers will lower latency but could also create issues with streaming quality. Network buffers will correspond to peers’ sample buffer. For example, if they have a higher sample buffer (512 for instance) you may not need as large of a network buffer your end (perhaps 128). The opposite would also be true if the sender is sending smaller packets (say 128) the network buffer may need to be increased. Unfortunately, due to the nature of the internet and individual hardware setups, there is no straight forward way to determine the optimal settings for everyone so you are encouraged to tweak these settings to find out what works best for your session. OPUS bit rate in short the higher the bit rate the better quality the audio. Linear is uncompressed. 96 kbps is compressed by a factor of 8, 48 kbps by a factor of 16 etc. One thing to keep in mind is that this is the quality of each individual track so the more tracks you have sending the greater the speed requirements. Example one track at 96 kbps will be sending as much information as 4 tracks at 24 kbps.

Chat (time zone UTC-1)

andybrucenet - 01:40

no luck - if someone can try connecting to me i'd like to see if this can work.

andybrucenet - 01:38

just did - keeps saying "trying" for you. another guy could connect to me but he's on wifi and sound was bad.

JamTuner - 01:35

try logging out -restart client- re-log in

andybrucenet - 01:31

got the new focusrite interface setup properly

andybrucenet - 01:31

sample buffer is 64 (lowest)

andybrucenet - 01:31

network buffer is 128 (lowest)

andybrucenet - 01:30

codec is opus 96 k

andybrucenet - 01:30

hmm not working

JamTuner - 01:17

andy, try connecting to me

andybrucenet - 00:59

come on in and play...

pablogesell - 23:25

try to use 48 kbs codek and reduce some parameters

pablogesell - 23:25

hello golden

rorrr - 22:46

hi bana

JamTuner - 22:28

Reno Drums, and others : when latency spikes , it usually means you or your connected usr has traffic on their LAN. Someone in your household is streaming movies or gaming, etc.

JamTuner - 22:26

I am streaming audio for those needing a test. Just connect to me :)

Reno Drums - 20:59

What happened, latency is 950 ms or more?. Normal value in the past was around 10 15 ms

rorrr - 20:52

rorrr from Canada

BrimoneSick79 - 20:42

...und BassJogi?

MrGee - 20:42

huhuuu

BrimoneSick79 - 20:41

Mr.G?

Laubi - 20:27

test

tanersarf - 20:06

Strange it doesn nt reload ...

tanersarf - 20:06

Strange it doesn nt reload ... Strange really

tiborkiss - 20:00

JamTuner - I opened another account already. The UDP Port forwarding, other than default if exists I have to understand.

Drumah - 20:00

yeah.. just want to try band pracfice - 3 piece

JamTuner - 19:58

drumah- you can connect to as many users as you wish, just beware that everyone must have good bandwidth

Drumah - 19:53

how many people can join on you?

JamTuner - 19:52

tiborkiss - I believe you must open another, seperate account in order to use two pc's

tiborkiss - 19:50

Another issue. Behind the same firewall, with port forwarding 50000 is possible to enable only one. How I can test between two computers then?

Drumah - 19:50

haha.. just noticed typo in channels-4

tiborkiss - 19:49

MOTU Audio ASIO and 3 Voicemeeter Virtual ASIO, plus Realtek ASIO i see in input/output selection. VB-Cable, Bridge, etc can I use?

tiborkiss - 19:46

Hi Alex. In the audio device selection, is there a driver which route from virtual device and into virtual device? I see just soundcards.

DJR - 19:00

Great video, thank you Alex...

sr54 - 18:58

Allerhand morgen 18 Uhr

jazzalex - 18:40

Important new video online:https://www.youtube.com/watch?v=RydhJ0jMLB8&feature=youtu.be

djippy - 18:35

Forget it, on the FAQ, it is noted that it should be 48khz

djippy - 18:28

it seems that it creates problems (Cracks) when I try to listen to music...

djippy - 18:27

Is there a reason my focusrite Scarlett seems to always change to 48Khz when I do connect to the site?

JamTuner - 17:41

I will be afk, so I can not answer questions , sorry.

JamTuner - 17:40

I am streaming audio for those who need to test. Just connect to me :)

SHOUT_NONMEMBER