Linux-audio-user February 2013

linux-audio-user@lists.linuxaudio.org

142 participants
178 discussions

by cunnilinux himself

hello community. well, i'm looking for good quality usb audio interface to use with linux. there are a couple reports on the net that Apogee Duet 2 does work with generic ALSA snd-usb-audio module. so, the question is: can anyone here on the list confirm this? thanx in advance.

13 years

Re: [LAU] Loop Composition

by Aurélien Leblond

My composition workflow is based on loops - starting by creating rythms in >> Hydrogen and recording a few drum loops in Ardour, creating a bass line >> with amSynth or ZynAddSubFX and recording a few bass loops in Ardour, >> recording some guitars in Ardour and creating some samples from that... >> >> The difficulty I'm having in Linux is to create composition from those >> loops. >> In Ardour is is time consuming to reorganise the loops to test new >> order/composition (because Ardour is not meant for that). >> >> My idea of a workflow would be to have a tool to try out the different >> loops in different order and "jam" with them to see what works and what >> doesn't. >> >> > Non-DAW has better features for dealing with looping regions/clips than > Ardour does. You can set a loop point on any region, after which adjusting > the region length adjusts the number of repetitions. > I quickly cheked the feature list for Non-DAW on their website. Although the feature you described is listed, it doesn't really say if it eases the composition part. It looks to me like you need to have a clear idea of the song structure. What if you would like to try out different arrangements quickly - can you confirm or infirm if that is the case?

13 years

Re: [LAU] Loop Composition

by Aurélien Leblond

On Mon, Feb 4, 2013 at 4:30 PM, Harry van Haaren <harryhaaren(a)gmail.com>wrote: > On Mon, Feb 4, 2013 at 4:09 PM, Aurélien Leblond <blablack(a)gmail.com>wrote: > >> My idea of a workflow would be to have a tool to try out the different >> loops in different order and "jam" with them to see what works and what >> doesn't. >> > > Hmm, although you describe things somewhat, its not totally clear to me > how you intend to choose which loop to play when. But yes, I kinda know > what you're getting at.. Loading all drum loops on one track, and all > guitar loops on the next track, and then triggering them should do. > One way would be to trigger them manually (with a midi controller for example), the other is to move clips on a grid. Ableton Live is kind of a mix between the two (as far as I know) and Bitwig seem to work similarly. the idea is to be able to try out different song structure based on the same loops. > > Do the loops need to be time-stretched? > Luppp2, as on https://github.com/harryhaaren/Luppp should do this. Note > that there is some issues with playback of different sample lenghts. I'm > re-writing the engine, so Im no longer familiar with the exact bug in that > version. I know it didn't work perfectly though.. YMMV :) > > In my case, there wouldn't be the need for time-stretching as I'm creating the loop myself. But yeah, the loops would be different side. > Then once that would be done - I would re-record everything properly in >> Ardour (drums with multiple tracks, breaks, etc... a more natural way of >> playing, guitar played and not looped, etc) >> > > Cool, perhaps using Luppp is feasible, if you trigger the loops again > they'll re-sync, so it'll give you a rough idea. > > In general this is exactly the intended workflow when designing Luppp, but > its just not there yet. I hear the future is a big place... -Harry > Can't wait for the future :)

13 years

[LAU] guitarix

by Jörn Nettingsmeier

hi *! today, i checked out guitarix for the first time, while working on a clean guitar track that needed some cojonification. built straight from git w/o issues (except for the fact that it does require a ladspa header even if explicitly asked not build ladspa plugins). after two hours of experimenting, i feel compelled to share my impression: FUCK YEAH! many thanks for sharing this awesome tool. best, jörn -- Jörn Nettingsmeier Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 Meister für Veranstaltungstechnik (Bühne/Studio) Tonmeister VDT http://stackingdwarves.net

13 years

Re: [LAU] algoscore PKGBUILD for archers

by Fero Kiraly

its my mistake, dependency to csound resist from the old package, which I have rewritten... try to remove csound (whatever ver.) from dependency array and try to build. If it goes fine I will remove csound from dependency. .. I am using that with pd, not with csound... but I have csound installed so cannot try.. -- fk

13 years

[LAU] algoscore PKGBUILD for archers

by Fero Kiraly

is ready.. if anybody wants to use it. https://aur.archlinux.org/packages/algoscore/ fk.

13 years

[LAU] ADAT tape transfer

by linux-audio＠to11.net

If I have two ADAT tapes that comprise a 16 track recording but I only have one ADAT recorder, can I transfer the tapes one at a time into Ardour on my computer via an ADAT cable and, in the end, be able to have all 16 tracks still synchronized somehow? I have heard that Broadcast Wave files have embedded timecodes that can be used for synchronization but I'm not really sure how to create them or, in this situation, if they'd even be applicable. Thanks in advance for any help/advice.

13 years

Re: [LAU] algoscore PKGBUILD for archers

by Fero Kiraly

hi, it is using old code. I only have reparired not working PKGBUILD. I have tried it with PureData throught OSC and it works. yes, it is a nice piece of code.. bye

13 years

[LAU] sample rate conversion question.......

by Aaron L.

I recorded a session at 48k in Pro Tools. I'd like to re-do some of the guitars in Ardour. The engineer gave me a 'reference mix' (just drums and bass) at 44100. (It's a stereo wav file.) The files that I'd like to edit are at 48k (the guitars......2 mono). I created a new session for these edits/overdubs at 44100 since that's what the reference mix was set to. I then imported the 48k files (which were sample rate converted into 44100 upon import) into the new session. Will there be any weirdness latency-wise (or worse?) if I do the edits to the 48k files (guitars) after sample rate conversion into a 44100 project (to match the reference mix) and then have them (the guitars) re-sampled back to 48 for final mix-down? I plan on just sending him the wav files. It this something to be concerned about? Would it be best to just keep everything at 48 until everything is finalized? I'd really rather not end up mixing down and thinking "does this need to be nudged here and there?" Any input is much appreciated. Thanks!

13 years

[LAU] Linux radio / streaming tips?

by Dan MacDonald

Yesterday I had a small local radio station get in touch with me as they're looking to switch from Windows to Linux for their streaming server. I'll be helping them out just as a favour and because it sounds like it could be fun but I don't really have any experience with the various Linux streaming solutions so I thought I'd post on here to see if anyone has any tips? I've not been down to check out their existing setup but I don't think they want much more than to be able to stream recordings and Skype. I think Skype lacks JACK support so maybe that will require a bit of messing to get it to play nicely (if we use a JACK based streaming solution). Did the OSM podcast guys ever do a write-up/ HOWTO I could read? Any other good resources for those setting up a small, Linux-based radio station?

13 years

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Linux-audio-user February 2013