Apologies for those with an allergy to soundcloud, but I don't have a
better way of distribution at the moment.
http://soundcloud.com/jmstone/never-again
I recorded this a few months ago now, planning to make some further
changes, but I'm not sure I will get round to it in the near future,
so.. RERO.
The extreme distortion, and vocals back in the mix is on purpose - and
hopefully reasonably in keeping with the genre. The piece could
probably do with a better ending!
This track is composed of a load of guitar and bass loops recorded
into Renoise. Effects are mostly the built-in Renoise plugins IIRC.
The guitar (Aria TA50) was recorded thru a Behringer V-AMP 2 and the
bass (Ibanez Roadster RS800) through a BDI 21. Vocals were through 2
different cheapo Behringer mic preamps (minimic 800 replaced by a mic
100 after I fried the former accidentally!) Samson C01 condenser.
Drums were laid down in 10 seconds! :)
J
Has anyone compiled Maarten de Boer's "polarbear" on recent compilers?
I just downloaded polarbear-0.5.3 from
http://www.resorama.com/maarten/files/polarbear-0.5.3.tgz. While
compiling it complains about some things i beleive are related to the
version of the compiler. Hacking it slightly, the app seems to run ok,
adressing jack correctly.
Ive got some questions about the edit-menu - select, cut, copy... - but
am unsure whether ive got a fully functional copy up, and dont know how
they are meant to behave.
Maarten de Boer, maybe you are out there?
Polarbear functions similarly to a sorely missed app from the old
NeXT's, an interactive z-plane editor (ive forgotten the name, but i
beleive it was setup by someone at ccrma) where coefficients were made
available in the clipboard after manipulating the filter in the gui.
Maybe there are other similar tools around?
Thanks,
-anders
Hi,
VMPK & FluidSynth is a MeeGo Harmattan application for Nokia N9/N950
smartphones. It contains a QML based VMPK user interface bundled with
FluidSynth for sound generation.
Several enhancements have been included since the beta announced in August.
* Controllers, Bender, and Velocity values can be optionally controlled by the
accelerometer.
* Internationalization. This version includes translations to Spanish, Russian
(thanks to Serguey Basalaev) and Czech (thanks to Pavel Fric).
* Inverted color theme. This dark color combination consumes less power,
enabling longer battery life.
* Latest FluidSynth included.
Binary packages available at the OVI Store
http://store.ovi.com/content/210572
Sources at SourceForge.nethttp://sourceforge.net/projects/vmpk/files/vmpkn9/
Screenshots in this blog post
http://midi-clorianos.blogspot.com/2011/10/vmpk-fluidsynth-010-released.html
Regards,
Pedro
LAC 2012: the Linux Audio Conference - Call for Participation
April 12-15, 2012 @ CCRMA, Stanford University
http://lac.linuxaudio.org/2012/
[Apologies for cross-postings] [Please distribute]
Online submission of papers, music, installations and workshops is now
open! On the website you will find up-to-date instructions, as well as
important information about deadlines, travel, lodging, and so on. Read
on for more details!
We invite submissions of papers addressing all areas of audio processing
based on Linux and open source software. Papers can focus on technical,
artistic or scientific issues and can target developers or users. We are
also looking for music that has been produced or composed entirely or
mostly using Linux and other Open Source music software.
The Deadline for all submissions is January 11th, 2012
The Linux Audio Conference (LAC) is an international conference that
brings together musicians, sound artists, software developers and
researchers, working with Linux as an open, stable, professional
platform for audio and media research and music production. LAC includes
paper sessions, workshops, and a diverse program of electronic music.
The upcoming 2012 conference will be hosted at CCRMA, Stanford
University, on April 12-15. The Center for Computer Research in Music
and Acoustics (CCRMA) at Stanford University is a multi-disciplinary
facility where composers and researchers work together using
computer-based technology both as an artistic medium and as a research
tool. CCRMA has been using and developing Linux as an audio platform
since 1997.
http://ccrma.stanford.edu
Stanford University is located in the heart of Silicon Valley, about one
hour south of San Francisco, California. This is the first time LAC will
take place in the United States.
http://www.stanford.edu
We look forward to seeing you at Stanford in April!
Sincerely,
The LAC 2012 Organizing Team
Hello all,
New releases on <http://kokkinizita.linuxaudio.org:/linuxaudio/downloads>:
zita-convolver-3.0.2
--------------------
* General code cleanup, will now allow bugfixes and
minor changes without breaking binary compatibility.
* Will work correctly in Jack's 'freewheeling' mode.
* Optimised partition size sequence in function of number
of inputs and outputs, size and matrix density.
* Should compile and work on OSX. The Makefile is untested.
This release is NOT binary compatible with 2.0.0.
This release is NOT API compatible with 2.0.0. The
required changes are small but essential, and are
documented in the README file. Full API documentation
will follow.
jconvolver/fconvolver-0.9.1
---------------------------
* Spaces allowed in filenames and jack port names. Use
quotes or escape the spaces.
* Hilbert transform IR built-in, allows creation of
arbitrary complex matrices without requiring any
external audio files. See README_CONFIG.
* Should compile and work on OSX. The Makefile is untested.
This release requires zita-convolver-3.0.x
For an intersting demo of what convolution can do, try
'weird.conf'. Instructions are in the file.
Ciao,
--
FA
Ralf wrote:"Again, the usage Fons was suggesting is uncritical, but using a limiter
instead of a compressor, for often repeating peaks isn't good, since the
limiter would cause distortion."
I always use the SC4 compressor on master bus for a little compression followed by the fastlookaheadlimiter to protect against any unpredictable inevitable spikes arising from a bunch of waveforms phases coinciding with each other.
Also, you might find these links interesting:
https://wiki.archlinux.org/index.php/Forum_Etiquette#No_Power-Posting.2FEmp…http://en.m.wikipedia.org/wiki/My_two_cents_(idiom)
James
Sent using BlackBerry® from Orange
On Thu, 2011-10-20 at 01:40 +0000,
linux-audio-user-request(a)lists.linuxaudio.org wrote:
> Message: 22
> Date: Wed, 19 Oct 2011 22:42:12 +0000
> From: Fons Adriaensen <fons(a)linuxaudio.org>
> Subject: Re: [LAU] Top DSP plugins?
> To: linux-audio-user(a)lists.linuxaudio.org
> Message-ID: <20111019224212.GC10008(a)linuxaudio.org>
> Content-Type: text/plain; charset=us-ascii
>
> On Thu, Oct 20, 2011 at 12:16:45AM +0200, Ralf Mardorf wrote:
>
> > I experienced this 0.01% of the time margin is overstepped as not
> being
> > audible. YMMV Overstepping 0 dBFS not always cause audible results.
>
> I recently made a live recording of Berio's 'A-Ronne', in this case
> performed by six singers. It has a *very* wide dynamic range (which
> has to be reduced for e.g. broadcasting, as was the case) and takes
> about 26 minutes. 0.01% of 26 minutes is around 1.5 seconds, and I
> can assure you that a female voice at +9dB and being clipped for
> 0.5 seconds three times is quite a nasty effect.
Yes, 9dB above 0 dBFS is audible :(. I didn'd use a limiter for this
amount, but I suspect the limiter will cause unwanted effects too. I
might be wrong. If such a +9dB issue should happen for popular music in
the studio, than we still record it again. Hm? Really an issue when
recording live played classic. So again, for your usage you are right
regarding to the usage of a limiter, but a limiter isn't a tool to
minimise dynamic. Even for loudness war mixes the EQ and compressor are
important, a limiter is the last tool that should be used to manipulate
the dynamic, in other words it isn't a tool to manipulate dynamic.
To reduce the dynamic for popular music, do a good EQed mix and than use
a multi-band-compressor with what ever high ratio you prefer, but avoid
the usage of a limiter. Here the limiter only sometimes is useful!
On Thu, 2011-10-20 at 01:40 +0000,
linux-audio-user-request(a)lists.linuxaudio.org wrote:
>
> Message: 5
> Date: Wed, 19 Oct 2011 22:02:44 +0200
> From: Philipp ?berbacher <hollunder(a)lavabit.com>
> Subject: Re: [LAU] Top DSP plugins?
> To: linux-audio-user <linux-audio-user(a)lists.linuxaudio.org>
> Message-ID: <1319053780-sup-6665@eris>
> Content-Type: text/plain; charset=UTF-8
>
> Excerpts from S. Massy's message of 2011-10-19 21:09:36 +0200:
> > On Wed, Oct 19, 2011 at 10:31:48AM +0200, Jostein Chr. Andersen
> wrote:
> > >
> > > It's interesting that in the 70's (the last half for me) and 80's,
> we did not
> > > have much possibilities compared to what we have today, we had to
> stick to
> > > what we had. But the sound itself and the mixes could sound very
> good. Much of
> > > what's typical of the sound from the 70's and 80's was not about
> quality but
> > > sound preferences, well expect for noise, echo and other tape and
> HW
> > > artifacts.
> > It's also interesting to note that, with the plethora of devices,
> > plugins and general techniques available today, mmusic production
> seems
> > to be sonically convergent. IOW, back when people had less freedom
> in
> > terms of choice, they fought harder to create a production sound
> > specific to them, while today, that self-same freedom seems to be
> used
> > to mimic one another as much as possible. (This is speaking broadly,
> of
> > course.)
> >
> > Cheers,
> > S.M.
>
> The question is, what can you do except mimic each other? Where can
> you
> go where no-one has gone before?
> I think I heard an anecdote once that Mozart (or another big classical
> composers) wondered about the same thing, yet a lot stuff happened
> since. The only possible way to go is into experimental music, and
> even
> there it's very hard to do something no-one has done before. The added
> downside is that nearly no-one will want to listen to experimental
> music (based on the principle of pop music, people like what's similar
> to what they know and like). I think many people have run into this
> problem and I believe that's where the (from my observation)
> increasing
> trend of combining different kinds media comes from.
>
> In at least the time I could observe them many musicians fell into two
> categories: Those who are happy with playing music in the style of
> popular musicians from years past (Classical era, Beatles era, Rock
> era)
> and those who are happy to play in the popular style of the present
> (Pop). I believe those who tried to find new grounds have always been
> a
> minority, and it likely always was hard, for a variety of reasons.
>
> Regards,
> Philipp
Do we need e.g. guys like Mark Guilian, Mike Patton, etc. those spoiled
kids who bedder should drive the Porsche spend by mum and dad? They
steal the music of my generation independence musicians and make a lot
of money. We are hardly able to event some new kind of music, but we can
try to do it and we can keep our roots. Those roots might be Schubert,
McCoy Tyner, Radio Birdman, The Acüssed etc. but never ever some of that
commercial mainstream.
The other thread was about what? Analog mixing vs "how much plugins can
I use to kill a good composition?" when doing digital recording. Nobody
force us to do over-processing when doing HDD and MIDI-sequencer
recording. We are free to use the computer in a sane way.
On Wed, 2011-10-19 at 12:00 +0000,
linux-audio-user-request(a)lists.linuxaudio.org wrote:
> Message: 18
> Date: Wed, 19 Oct 2011 11:02:00 +0200
> From: Jeremy Jongepier <jeremy(a)autostatic.com>
> Subject: Re: [LAU] Top DSP plugins?
> To: linux-audio-user(a)lists.linuxaudio.org
> Message-ID: <4E9E9208.5080902(a)autostatic.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> On 10/19/2011 10:46 AM, Ralf Mardorf wrote:
> > Do wish to get rid of any dynamic?
>
> For my indie-electro solo stuff yes. Dynamics don't work on the
> dancefloor ;)
>
> Best,
>
> Jeremy
Jeremy :)
I disagree! http://www.youtube.com/watch?v=qrrosELRyss
It's compressed but still has some dynamic and it's hardly possible to
do better dancefloor music :p.
- Ralf
On Wed, 2011-10-19 at 19:00 +0000,
linux-audio-user-request(a)lists.linuxaudio.org wrote:
> Almost all of my recording is 'classical' music which does not
> require high average level or invasive compression. But even
> using conservative average levels (mostly EBU loudness based
> these days), that doean't mean that you can't get a signal that
> peaks above 0 dB. It may happen just 0.01% of the time, but it
> happens. And such short peaks are no good reason to lower the
> average level, and limiting them to avoid clipping is completely
> transparent, so I always use a limiter in the master bus.
I experienced this 0.01% of the time margin is overstepped as not being
audible. YMMV Overstepping 0 dBFS not always cause audible results.
Most times I'm not using a limiter, sometimes I'm using a limiter for
the master bus too, but I don't think that it's a good thing doing this,
usually a limiter does destroy the sound.
Anyway, the context was about using the limiter as a very active
plug-in, while you're talking about the limiter as something that will
be active in the worst case szenario, 0.01% of the times ;).
> but none of them would produce such
> gratituous nonsense as what you write above.
It's hard for you to understand the context of speech? :D
2 Cents,
Ralf