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.
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?
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 :)
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
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
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.
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
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!
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?