Linux-audio-user January 2020

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Re: [LAU] sound with Scarlett 3rd Gen device?

by Christoph Kuhr

6 years

[LAU] sound with Scarlett 3rd Gen device?

by Iain Mott

Dear Linux audio people, I have a Scarlett 18i20 3rd Gen interface which I'm trying to use on a recent Samsung laptop with Ubuntu 19.04. The 3rd gen devices are apparently shipped in non-functional "MSD" mode, so I registered and activated the device using Windows 10 following the instructions on the Focusrite website. Tested it there with Ableton Live Lite (provided via registration of the device) and also Ardour. It works fine. To get it to work on Ardour it was necessary to use the ASIO driver in the Ardour audio configuration. I had removed ASIO4ALL before installing the Focusrite/Scarlett software as suggested, so perhaps this driver came with Focusrite install. Now with the device apparently configured and the routing done in Windows using the Focusrite Control software (i thinks that's the name, I had selected output channels 1 and 2 to connect with the monitors/headphone jack), I returned to Linux. The Scarlett device is detected by lsusb, qjackctl and pavucontrol. Jack runs with no errors when the Scarlett is selected as the i/o device and all the correct channels appear in the Connect window. No sound however is heard through the phone jack, nor if i plug the phones directly in to the output jacks at the back. I tested using VLC configured for Jack output and also with pd -jack. With jack turned off and with VLC audio out set to pulsaudio, I set the Scarlett 18i20 as the preferred device in pavucontrol. With VLC playing the output VU meter for the Scarlett is active in pavucontrol but there is still no sound coming out of the device itself. Has anyone managed to get a 3rd generation Scarlett to work on Linux? Can anyone suggest please what I can do? Thanks! Iain PS. i did see an unresolved thread on this list last year about a 3rd gen Scarlett. I'll send a message to the author to see if the problem was fixed.

6 years

[LAU] [music and video] a773: diffusion

by Atte

Hi I had a blast generating alot of the sounds on this track with csound (my first love): https://youtu.be/mD762X1wxdI It's recorded in one take, mixed in linux native reaper, video edited in kdenlive. Hope you enjoy it! -- Atte http://latestyoutube.a773.dk | http://a773.dk

6 years

[LAU] Motu M64 Madi via USB (class compliant)

by Peter P.

Hi list, does anyone have experience using the Motu M64 https://motu.com/products/avb/m64/specs.html as class compliant interface under linux and can confirm that all 64 channels are working in duplex mode? The list archive didn't give a positive result. Thanks, P

6 years

[LAU] Linux Audio Conference 2020 : deadline extension

by Linux Audio Conference

The 18th Linux Audio Conference call for participation is being extended for papers, posters and workshops, to the 9th of February 2020. Reminder : The conference will take place May 11-13, 2020, at the SCRIME, part of the University of Bordeaux in France. Up-to-date information is available at the following address : https://lac2020.sciencesconf.org/ Linux Audio Conference is the premier meeting of developers, researchers, and artists developing and using free, libre and open-source audio software, mainly on the GNU/Linux operating system. The conference is part of the Linux Audio consortium. With our best regards, The Linux Audio Conference Team

6 years

[LAU] ALSA sequencer: buffered events (clearing them?)

by Jeanette C.

Hey hey, I wonder if the ALSA sequencer system buffers MIDI events or if it is my hardware. When I start a synth and play something or turn a knob and then connect it to another hardware synth, I often hear a burst of notes or find that some CC(s) have been sent. If that is a feature of the ALSA SEQ, can it be cleared? Best wishes and thanks, Jeanette -- * Website: http://juliencoder.de - for summer is a state of sound * Youtube: https://www.youtube.com/channel/UCMS4rfGrTwz8W7jhC1Jnv7g * SoundCloud: https://soundcloud.com/jeanette_c * Twitter: https://twitter.com/jeanette_c_s * Audiobombs: https://www.audiobombs.com/users/jeanette_c * GitHub: https://github.com/jeanette-c Top down, on the strip Lookin' in the mirror I'm checkin' out my lipstick <3 (Britney Spears)

6 years

[LAU] Call: Audio Mostly Sep 15-17 Graz 2020 submit paper/workshop/music contribution until May 1st.

by IOhannes m zmoelnig

Audio Mostly 2020 (AM'20) calls for Papers, Demos, Workshops, and Music & Installations on the topic of audio mostly in extended realities. Audio Mostly is an audio-focused interdisciplinary conference on design, technology, and interaction with sound that embraces applied theory and practice-oriented research. As in previous years, the conference will be in-cooperation with ACM/SIGCHI (tbc). The Institute of Electronic Music and Acoustics (IEM, https://iem.at/) is hosting AM'20 in Graz/ Austria: we are looking forward to an interdisciplinary event extending mostly -but not exclusively- the audio reality! Details can be found below and on https://audiomostly.com/2020/call/ The Conference Chairs Katharina Groß-Vogt and Robert Höldrich am20(a)iem.at # Virtual Participation AM'20 offers virtual participation for oral paper presentations. Accepted papers may be presented and discussed via video live stream and are part of the proceedings. Passive virtual participation will be possible for a small fee and includes a live stream of all paper sessions and concerts and other virtual meeting formats. # Call for Papers Please submit your research as full papers (5-8 pages) or short papers (4 pages). Both will be fully peer-reviewed in a double blind process. Accepted papers will be published in the ACM Digital Library (tbc). # Call for Demos We encourage authors to contribute demonstrations to be displayed at the conference. Demos require a paper submission. # Call for Workshops We invite proposals for workshops to be held at AM'20. Selected workshops will take place the day before the main conference (September 14th). We encourage hands-on/interactive workshops and regular workshops that last from 2 hours to a full day. Workshops are proposed in the form of a submitted paper. # Call for Music & Installations We call for Music for auditory virtual environments and sound art to be performed in a concert hall and as sound installations around the conference site. # Important Dates Submission Deadline: May 01, 2020 Notification of Acceptance: June 15, 2020 Workshops: September 14, 2020 Conference: September 15-17, 2020 Registration opens in June, with fees in the range of 30 EUR (virtual) to 150-400 EUR (physical).

6 years

[LAU] systemd woes with jackd and its permissions (raspbian)

by Peter P.

Hi list, please help with a realtime permissions issue trying to start jackd via my systemd service script on raspbian. As root or as user peter (which is part of the audio group) I can start jackd fine using either jackd -v -R -d alsa -r 44100 -P or jackd -v -d alsa -r 44100 -P No error message mention that realtime was (or wasn't) enabled. Is this normal and intended? Checking with htop, there are three jackd processes with a priority of 20 and on with -11. Does this mean that realtime is enabled? Using mplayer as user peter I can play back audio via jack. Now I try to start jackd automatically from a systemd service file created as /etc/systemd/system/jackd.service with the following contents: [Unit] Description=jackd After=sound.target [Service] User=peter ExecStart=/usr/bin/jackd -d alsa -r 44100 -P [Install] WantedBy=multi-user.target I install that service with sudo systemctl enable jackd.service and run sudo systemctl daemon-reload after having changed the above service file. When I start that service it fails with errors in the journal: "JACK is running in realtime mode, but you are not allowed to use realtime scheduling." and "Please check your /etc/security/limits.conf for the following line" and so on. Messages that are not present when starting jackd as user from the command line. Now I am puzzled because I thought that it is enough to specify user "peter" in the systemd service. Well if I comment the line User=peter out and start the service again, jackd runs. It is five processes with priority 20 and one with -11. The stupid thing now is that the normal user peter can use that jackd: Mplayer tells me that "connect(2) call to /tmp/jack-1000/default/jack_0 failed (err=No such file or directory)" So I reckon that a user can't use a jackd started by another user, or root? Now I removed above service and installed it as user service ~/.local/share/systemd/user/jackd.service and contents: [Unit] Description=jackd After=sound.target [Service] User=peter ExecStart=/usr/bin/jackd -d alsa -r 44100 -P [Install] WantedBy=default.target It gets installed with systemctl --user enable jackd.service and started with systemctl --user start jackd.service Jackd is now started and can be used with mplayer. But the systemd service is no longer started at boot not at login of user peter. I have to run: systemctl --user start jackd.service Also trying loginctl enable-linger peter does not help. The general question remains: How can I run a systemd user service without that user being logged in? Another way recommended at https://www.linux.org.ru/forum/admin/10795275 or https://ask.fedoraproject.org/en/question/131584/jack-fluidsynth-as-user-sy… is to add some lines specifying Limits. [Unit] Description=jackd After=sound.target [Service] User=peter LimitRTPRIO=infinity LimitMEMLOCK=infinity ExecStart=/usr/bin/jackd -d alsa -r 44100 -P [Install] WantedBy=multi-user.target It does not start jackd with the same error as above. "JACK is running in realtime mode, but you are not allowed to use realtime scheduling." and "Please check your /etc/security/limits.conf for the following line" Now this took me four hours so far without success and I wonder what is a functional way to use jackd via systemd on a headless computer in 2019. All comments are very welcome! Peter

6 years

[LAU] Creating a personalised HRTF?

by Jeanette C.

Hey hey, is there any (convenient) way to create a personal HRTF (head related transfer function)? I've been wondering about this for a while. I knew only one way: go to a very tech savvy place - a uni maybe - and let them do it. Now, I've just listened to a NAMM video ( http://www.youtube.com/watch?v=FWotz3oz2gs ) and wondered if there is any open source solution, be it along their lines or any other, to create a personal HRTF. I am particularly interested in using such a model in Csound, which offers very nifty modules to work in three dimensions with HRTF. Any tips are welcome. Best wishes and thanks, Jeanette -- * Website: http://juliencoder.de - for summer is a state of sound * Youtube: https://www.youtube.com/channel/UCMS4rfGrTwz8W7jhC1Jnv7g * SoundCloud: https://soundcloud.com/jeanette_c * Twitter: https://twitter.com/jeanette_c_s * Audiobombs: https://www.audiobombs.com/users/jeanette_c * GitHub: https://github.com/jeanette-c But tell me what happens when it stops? <3 (Britney Spears)

6 years

[LAU] Find all "beat offsets" from .mscx (MuseScore 3 uncompressed) or exported MusicXML?

by Kevin Cole

Hi, I'm not terribly well-versed in either music theory or the capabilities of all the various musical software tools out there in the happy land of Linux (in my case, Arch Linux). I'm looking to "shortcut" a process: I want to generate two sets of WebVTT captions / subtitles for an MP3 that was produced from a MuseScore 3 file. One set will be chord notations like "F#m/A". (The other will be lyrics. Sort of a karaoke thing with chords.) Since the chord changes fall on a beat, and there are no alterations in the time of this piece -- i.e. no "slower" or "faster" parts, but there are time signature changes, it should be a programmatically solvable problem. The simplest approach seemed to be to generate a starting and ending time for each beat, and go through a parseable file and parse out which chord falls where and attach it to the appropriate element in the list of durations. The thought occurred that, if I have the duration of the file, and either the uncompressed MuseScore or uncompressed MusicXML, there might be a clever tool or Python module to parse the file and come up with the total number of beats, and then divide the duration by that. Rather than invent the wheel from scratch, I thought I'd ask if there was already a wheel -- or if the wheel I'm building ultimately won't roll. ;-) Is this making any sense, and are there such handy tools? Thanx.

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Linux-audio-user January 2020