Linux-audio-user

linux-audio-user@lists.linuxaudio.org

3 participants
20662 discussions

Jack transport - was - Ardour/Muse Jack tempo lock

by Len Ovens

Tim E. Real wrote: > I believe it is crucial that all the apps (MusE included) use the complete > full feature set of the Jack Transport API for the best accuracy. Is there a record enable in there? It seems to me it would be worthwhile for recording audio in one application and MIDI in another or for punch in out. The application would choose to see or ignore such a signal. -- Len Ovens www.OvenWerks.net

11 years, 3 months

Trouble with Alsa MIDI connections

by Robert Persson

Hello, I have two machines running the latest UbuntuStudio. On one of them MIDI is working fine, but on the other I am having problems. On the problem machine I can't connect either of my external MIDI keyboards to another MIDI port while Jack is running. If I highlight the keyboard and the other port in the QJackCtl Alsa graph and click "connect", nothing happens. I can however connect them while Jack is inactive, and that connection will continue working even after Jack has started again. I am also having a problem with an application running in Wine (Reaper). Reaper's MIDI port does not appear in the QJackCtl Alsa graph, but Reaper shows the other MIDI ports available in Alsa in its own preferences. However if I try to connect it to either of my external keyboards it fails, and it does so regardless of whether Jack is running. It will however connect to MIDI Through (which gave me a reasonable workaround when my routing requirements were simpler than they are now). Does anyone know what I need to do to get MIDI connections working properly? Many thanks

11 years, 3 months

Ardour MIDI tracer

by Len Ovens

Is it just me? Has anyone else looked at pitch bend events on the Ardour MIDI Tracer? Quick test: ================================================ - edit->Preferences->Control surfaces. - select both enabled and feedback for generic MIDI. - double click on it and select bcf2000 with mackie protocol - open an external midi monitor (using qmidiroute here) and connect it to Ardours MIDI control out. - also open Ardours MIDI tracer window and connect it to the same output. (Now qmidiroute will be in decimal and MIDI Tracer is hex..) - use the mouse to move the gain up and down on channel one with an audio track. ================================================= I am seeing numbers that make sense on qmidiroute -8013 to 8177, but midi Tracer only shows two hex digits ever, shouldn't there be four (maybe three to handle 10bits)? It looks like I am seeing only the LSB (I am not sure about this as the lowest nyble should be 0). My about box says 3.5.308 so maybe this has been fixed since then. 3.5.380 seems to be were things are at now. I have looked through the change logs here: https://ardour.org/whatsnew.html for 3.5.380, (this doesn't seem to be included in the source package for 3.5.380) and https://ardour.org/news/3.5.older.html which is included with the source package. And I don't see any mention of a fix. BTW, I am having lots of fun with my "MIDI" controler project. Built around a $3 USB computer keyboard from dollar store. I use two columns for each strip for 10 strips. Strip one for example is 1->z and 2->x. 1 is channel select (does nothing right now), 2 is record, q&w are pan control, a&z are the fader, s is solo and x is mute. up and down changes banks Ah, banks. I have bank at 5 and am using 8 channels to test. It starts on channel 1 to 5, bank right does 6 to 8 (good so far), then another bank right does channel 8. I don't know if this is right or wrong, but my expectation is that once my controller has even one strip outside the range of channels that I would no longer be able to bank right. I would also expect the GUI to make sure all the channels my controler can control to be visible. My thought is that in the same way a selected track gets a red border (in the mixer) perhaps a green border around the group of tracks in the current bank. The BCF2000, for example, does not give much indication visually of where the banks are at. Mine of course gives none, but I could make a GUI across the bottom of the screen the controler sits in front of that does that... it just seems redundant when the mixer is already on the screen just above. I do realize that the controler code at least for the MCP stuff is not complete and will get done when it gets done. What I have already is a wonderful step up. The transport is on the numeric pad. All of this could easily change as I add some encoders for pan/faders -- Len Ovens www.ovenwerks.net

11 years, 3 months

Update of zita-at1

by Fons Adriaensen

Zita-at1-0.4.0 is now avaliable at the usual place: <http://kokkinizita.linuxaudio.org/linuxaudio/downloads/index.html> Bugfixes, MIDI channel selection added. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow)

11 years, 3 months

Slowing down audio while keeping same pitch ?

by jonetsu＠teksavvy.com

Is it possible to slow down an audio file (mp3 for instance) while keeping the same pitch ? Does it even make sense ? The software would have to insert more of the same audio 'bits' to make it last longer. Is it possible ? Cheers.

11 years, 3 months

Linux mixer

by Ralf Mardorf

Hi, regarding to some mixer issues with Qtractor, discussed at other lists, I searched the Internet for a sane virtual mixer, but didn't find one for Linux. I need stereo or mono channels, it doesn't matter when stereo channels only provide balance, it's ok, real panning would be better. An insert (audio out/in) isn't needed, I could add an "insert" before I connect to the mixer. Inserting LV2 and LADSPA effects would be useful, but aren't too important, with one exception, each channel needs an EQ (Fons' parametric EQ or a similar EQ) and I need at least one _post_ fader aux send to a sub group or at least to an aux return pot for the stereo master sum. Pre fader aux would be completely useless. A mute and/or solo switch would be useful too, not needed, but it's important, that the mixer settings can be stored and restored. Regards, Ralf

11 years, 3 months

Komplete Audio 6 and Jack

by David

I'm completely new to this, so be kind to me please. Ubuntu studio 14.04 Komplete Audio 6 Rode condensor mics. I've found this link which says it should "just work", http://lists.linuxaudio.org/pipermail/linux-audio-user/2013-October/094733.… but I can't seem to get response from the inputs (microphones). I did manage to make Yoshimi talk to the KA6 via Jack but that isn't what I need. Ultimately I want to be able to simply record stereo microphones into Ardour or Audacity. Preferably Audacity because I'm familiar with it. I suspect that I'm not configuring Jack connections correctly but don't know enough about it to see what the problem is. I've set the interface in qjackctl settings to the KA6. The KA6 never shows up in the volume control panel in the way that a USB headset would. The KA6 does show up in both input and output sides of the qjackctl ALSA tab. -- David McQuire 0418 310312

11 years, 3 months

Ubuntu Studio 14.04 Pulse removal question

by Kaza Kore

All. I have recently installed a fresh Ubuntu Studio 14.04, replacing my old 12.04 version I was running previously. In short; I believe I now have everything running directly through Jack, which is autostarted with a "pulseaudio -k" command and wonder whether it is worthwhile actually fully removing PA? And if so the best way to do this/exact packages to remove? For those that are interested the steps I took were: Compile and install libflashsupport-jack library following the instructions at http://jackaudio.org/faq/routing_flash.html (there was one more lib not listed I also had to install for <video.h>) Install jack plugin for vlc, gstreamer and any media players I use. Test audio is really going through Jack and not via the pulse-sync outputs. Test audio still present after running "pulseaudio -k".* Remove autostart options: PulseAudio Sound System & XFCE Volume Deamon. ** Add "pulseaudio -k" to the startup command in Jack. And qjackctl as an autostart program. Remap mediakeys to control level of amixer's Master. (eg (may be different on your setup) volume up 2dB: "amixer -c 0 sset 'Master',0 2dB+") *Seems to be working nicely and when running "pulseaudio -k" the first time I instantly noticed the percentage time used by Jack approximately dropped by a half! (4% -2 % with just Audacious playing.) Seems to let me go maybe one setting lower. Not really tested yet though... So thoughts on fully removing PA please :) **These steps didn't seem to make as much difference as I thought they should do! Even though I thought I had stopped PA from running the pulse/jack sync still showed up in jack's connections and running "pulseaudio -k" still killed it, giving me the extra overhead in jack. This is why I added the command to the jack startup option. Surely these options should have stopped it from running in the first place, no?? Dale.

11 years, 3 months

[ANN] MOD Duo's Kickstarter campaign to be launched in mid September

by Gianfranco Ceccolini

Greetings Linux Audio Users and Developers !!! I'm very happy to inform that we will launch the MOD Duo's Kickstarter campaign in mid September. The MOD Duo is our second model and we've been putting a lot of engineering in it based on the feedback we had from the MOD Quadra experience. We deeply hope it becomes a device that empowers the Linux Audio community, bringing together developers and musicians. A pre-campaign site was created to warm up the communication engines: http://stepontothefuture.com. Hope you all enjoy and spread the word Kind regards Gianfranco Ceccolini The MOD Team

11 years, 3 months

Why does PHASEX sound so damn good?

by Ken Restivo

Rolling by on randomize came a Me and My Cronies jam/joke from years ago: http://www.restivo.org/blog/podpress_trac/web/558/0/Not_OK_Computer.ogg And I was struck by how much PHASEX sounds like a real analog synth, like an ARP 2600 or similar, and so much more real than any other software synths I've used. It sounds so... raw, uncontrolled, well, ANALOG. Most software simulations sound more or less authentic, but all so much more "tame", for want of a better term. But PHASEX always sounded to me (and felt, as I was playing with it) that at any moment it could do something crazy like throw a DC offset, to into an uncontrollable oscillation, or blow up my speakers, etc. I don't like being at a loss for precise, engineering terms, or understanding WHY something is, so I'm asking any of the DSP'ers here who might also have looked at (and understood) PHASEX's source. Any ideas what is so different about PHASEX, and what might be this quality of it's sound I could be trying to describe? -ken

11 years, 3 months

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

Linux-audio-user