Linux-audio-user September 2017

linux-audio-user@lists.linuxaudio.org

31 participants
32 discussions

[LAU] MOTU AVB series sample rate issue, JACK, and PulseAudio

by Be

tl;dr: Has anyone figured out a convenient, reliable way to setup MOTU AVB series interfaces? Hi, I recently got a MOTU Ultralite AVB to replace my RME Babyface Pro. RME advertise the Babyface Pro as working with Linux without proprietary drivers and because all essential functions can be controlled directly from the device, but that is not true. It is not possible to directly monitor a mono input on both sides of a stereo output. It is possible to hack around this with Y splitter cables on the inputs, but that's clunky and halves the number of available inputs. I contacted RME about this and pointed out an unused button on the device that could be used for toggling mono/stereo direct monitoring with a firmware update, but they did not care. It seems they only care about their own use cases, namely iOS. Apparently they can't be bothered to make minor changes for Linux users but are happy to misleadingly advertise Linux compatibility. So I got a MOTU Ultralite AVB to replace the RME Babyface Pro. The sound quality is on par with the Babyface Pro but the Ultralite AVB has more I/O and features for a lower cost (although if recording guitar is important for you, IMO the Babyface Pro's instrument inputs sound clearer than the Ultralite AVB -- but keep in mind the direct monitoring issue). The Babyface Pro has some quirks when using it with GNU/Linux (which I think are caused by the 2 channel mode it has for iOS), but the Ultralite AVB has different quirks. As discussed on this list previously, there are some oddities trying to set the Ultralite AVB's sample rate. This seems to cause some issues with PulseAudio. If I start PulseAudio with the Ultralite AVB plugged in, it works briefly but randomly drops out. When I plug in the Ultralite AVB after PulseAudio is started, it does not appear as an output for PulseAudio and I see this in PulseAudio's output: E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="1" name="usb-MOTU_UltraLite_AVB_0001f2fffe005a12-00" card_name="alsa_card.usb-MOTU_UltraLite_AVB_0001f2fffe005a12-00" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes card_properties="module-udev-detect.discovered=1""): initialization failed. Occasionally I also see this in the output when I start PulseAudio with it plugged in already. This seems to appear randomly and is not correlated with PulseAudio output dropping out: E: [alsa-sink-USB Audio] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. E: [alsa-sink-USB Audio] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. E: [alsa-sink-USB Audio] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Sometimes it looks like PulseAudio tries to set the Ultralite AVB to a different sample rate. I sometimes notice the LCD display on the Ultralite AVB showing 192 kHz when PulseAudio tries to use it. I tried setting this in /etc/pulse/daemon.conf: default-sample-rate = 44100 but it made no difference. So I have tried using JACK regularly with the PulseAudio to JACK bridge. Unfortunately this has some quirks too. JACK won't start unless I wait several seconds after plugging in the Ultralite AVB. If the Ultralite AVB is plugged in when I boot up and log into KDE, QJackCtl autostarting JACK doesn't work. I think this is because PulseAudio tries to grab the device before JACK starts. I have to wait a few seconds and manually start JACK. If the Ultralite AVB's firmware is set to a different sample rate than JACK is configured for, JACK won't start. Has anyone contacted MOTU about the sample rate switching issue? Ideally they could make it just work when the computer tells the audio interface to switch sample rates. Also, has anyone contacted them about possibly changing the firmware so the web interface is available over USB on Linux? I think if they used the RNDIS Ethernet-over-USB protocol that Android uses for tethering it could be plug-and-play with Linux. Or has anyone started working on a custom driver to make this work? In the meantime I got a cheap little GL-MT300A router ( http://www.gl-inet.com/mt300a/ ) to use the web control interface away from my home network. It's neat. :)

8 years, 3 months

[LAU] best way to start jack

by Robert Carter

Hi Everyone, I’m using qjackctl but it does not seem to be starting jackd and i’m getting errors in the log like: Cannot connect to server socket err = No such file or directory What is the best configuration for qjackctl so that it always reliably starts the audio server? Best Rob

8 years, 4 months

[LAU] trying to get pitch bend to work

by Bill Purvis

Hi, In a rash moment I splashed out load of dosh and bought an Artiprog Instrument 1. I play bass, and think it looks quite cool with this little thing playing nice low notes. I've not used by 5-string for months now! I have a couple of problems with it. The first is that Artiprog have not yet produced a Linux version of the editing program that is used to set up the device. It comes with four built-in devices, guitar, keyboard, violin and drums, and four programmable devices which can be programmed to behave like any of the built-ins, but with lots of variants. I want to use it as a fretless bass and have managed, by using by brother-in-law's Mac to program one of the devices, so that problem is resolved, though I wish they'd either bring out a Linux version or release the specs so I could write my own editor. The second problem is the one I'm really looking for help with: When it's programmed as a fretless bass, you press down the starting note and it sends a MIDI Note-on event, then as you move up and down it sends pitch bend events. I've checked using a midi monitor program and the events are all there, though they look a bit on the slim side - about -3500 - +3500 rather than -8192 - + 8191. The problem is they have no effect on the synth. I've been using Calf Fluidsynth, but tried the standard Fluidsynth and Yoshimi, but with no joy. I have tried patching in midi commands to set the pitch bend range to 12 (RPN 0,0, DATA 0,12) and tried the pitch_bend command in the standard Fluidsynth but nothing doing not even a semitone shift. Am I missing something here? Bill -- +----------------------------------------+ | Bill Purvis | | email: bill(a)billp.org | +----------------------------------------+

8 years, 4 months

[LAU] Modular software for Linux

by Jeanette C.

Hey hey, I just read about VCV rack and their modular software. From what I understand, it's a concept to mix hard- and software, but I might be mistaken, only had a glance yet. http://www.vcvrack.com Just thought some of you might be interested. Best wishes, Jeanette -------- * website: http://juliencoder.de - for summer is a state of sound * SoundCloud: https://soundcloud.com/jeanette_c 'Cause living in a dream of you and me Is not the way my life should be... <3 (Britney Spears)

8 years, 4 months

[LAU] I wonder if this hardware would be usable with Linux

by david

iZotope's Spire Studio, WIFI connected device with built-in mic and 2 inputs for other mics or guitars/synthesizers. https://www.fastcompany.com/40463399/this-little-wireless-gadget-turns-your… I guess it works only with iZotope's iPhone software, unfortunately. -- David W. Jones gnome(a)hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com

8 years, 4 months

[LAU] Soundcard volume problem

by ycollette.nospam＠free.fr

Hello I've got a focusrite 18i8 usb soundcard which works fine under Fedora 25. But, from time to time, the sound levels are not correct. For example, I can have a normal level on the right channel (the one saved dugin the preceding session) and 50% on the normal level on the left channel. When I start pavucontrol, or alsamixer, all the levels seems to be correct. To fix this, the only solution I found is to start qasmixer, to switch the problematic channel to off and then to switch it back to the last value. I wanted to fill a bug report, but I don't know where to start. Is it a kernel driver problem ? Is it due to an alsa script ? Best regards, YC

8 years, 4 months

[LAU] Berlin meeting 2017-09-06

by Daniel Swärd

Hi all. It's time for my monthly spam-mail to announce the Linux Audio Berlin meeting... As usual it's in the mainhall at c-base (Rungestraße 20). I'll be there from latest 20:00. I hope to see you there! Cheers /Daniel

8 years, 4 months

[LAU] Music: atmosjam

by jonetsu

Hello, This is a smooth and flowing short synth improv. All sounds apart from the drums are from the Atmos soundset published a few days ago for the Diva synth. https://soundcloud.com/nominal6/atmosjam Cheers.

8 years, 4 months

[LAU] Sourceforge Lists

by Will Godfrey

Anyone else having problems with these? Suddenly I seem to have been unsubscribed from all of them. No problems with any other lists. -- Will J Godfrey http://www.musically.me.uk Say you have a poem and I have a tune. Exchange them and we can both have a poem, a tune, and a song.

8 years, 4 months

[LAU] [ANN] Radium 5.0

by Kjetil Matheussen

Radium is a vertical music editor. Radium is inspired by trackers, but uses more graphics to show musical data. Radium also supports MIDI sequencing and hard disk recording. Radium has features like smooth scrolling, zooming, automation, piano roll, embedded Pure Data (Pd), and embedded Faust. * Homepage: http://users.notam02.no/~kjetism/radium/ * Screenshot: http://users.notam02.no/~kjetism/radium/pictures/ radium_4_9_20.png * Videos: http://www.youtube.com/user/KjetilMatheussen/videos * Demonstration video by Tobias Lutzenkirchen showing some of the features: https://www.youtube.com/watch?v=FhwmT0G5EwM (Radium 3.9.1) *Demonstration video showing developing Faust programs inside Radium: https://www.youtube.com/watch?v=LJm9Lox1WFA Changes between 4.8.2 and 5.0.10: * Lots of bug fixes and minor improvements. * Audio: Don't apply volume when bypassing. * Piano roll: Improved eraser functionality. * Editor: Move the swing sub tracks to the left of the piano roll. * Mixer strips: Show automation values. * Mixer strips: Quickly enable or disable plugins and sends. * GUI: Make F11 switch full screen for all windows. * Audio: Scan plugins in a separate process. * Audio: Apply correct initial automation values when starting to play. * Mixer: Configuration editor. * Documentation: Many improvements. * Audio: Always connect and disconnect several connections simultaneously, not one by one. * Mixer strips: Faster graphics. * Menus: Improve keyboard navigation. * Sequencer: Faster to move blocks while playing. * Faust Development: Fix various GUI issues. * Audio: Support Jack transport in order to sample-accurately synchronize to e.g. Ardour or Bitwig. * GUI: Program includes a new "Edit" tab in the main window. * GUI: Fix various keyboard focus issues * 594 other changes. (613 git commits).

8 years, 4 months

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Linux-audio-user September 2017