7 Feb
2011
7 Feb
'11
5:52 a.m.
Folks,
I've been thinking about playing with sound effects. For a lot of my
life, i've been interested in the sounds that various things make and i
am wondering if there is a linux program that would allow me to take a
sound clip and manipulate it. Like taking the sound of a chair squeak
and distort it, stretch it, apply filters to it, alter the tone etc.
Is there a linux gui app for this sort of thing? Preferably in with the
gtk tool kit as i have that all loaded up?
I did some digging around, but i didn't see much. Of course there's
always the chance that i looked at something that would do it well and
didn't recognise it.
What is that kind of program called?
Thanks,
Bearcat
7 Feb
7 Feb
6:08 a.m.
On 07/02/11 16:54, Bearcat M. Şandor wrote:
Folks,
I've been thinking about playing with sound effects. For a lot of my
life, i've been interested in the sounds that various things make and i
am wondering if there is a linux program that would allow me to take a
sound clip and manipulate it. Like taking the sound of a chair squeak
and distort it, stretch it, apply filters to it, alter the tone etc.
Is there a linux gui app for this sort of thing? Preferably in with the
gtk tool kit as i have that all loaded up?
I did some digging around, but i didn't see much. Of course there's
always the chance that i looked at something that would do it well and
didn't recognise it.
I'm not sure you'll find something specifically tailored to producing
sound effects, but general audio editing tools should be able to do the
job. There are two types of apps and workflows to look at:
* off-line audio editors: these let you load a sound, process them with
effects, and play back the results. Audacity is the classic example,
though there are a few others around
* real-time audio manipulation: there are a few ways you could do this,
but it could be as simple as loading the sound on to a track in Ardour,
and adding effects to that track. Depending on how far down the
rabbit-hole you want to go, you could combine various real-time apps --
via JACK, you could run the audio out through Rakarrack,
AlsaModularSynth, PHASEX, or any number of other apps.
AlsaModularSynth and PHASEX are both synths, rather than dedicated audio
processors, but both can be set up to read audio from a JACK input and
then run that through their filters and other effects, in much the same
way that many hardware synths allow you to process audio from an
external line input.
I'd start with Audacity -- it has built-in effects for a lot of things,
including pitch and time manipulation, and it can use LADSPA plugins as
well, so you'll have a lot of scope to play with your sounds. The
real-time approach is definitely fun, though, especially once you start
plugging things together!
I recently made a short track that consisted of a looped sample in
Specimen (a standalone JACK sample), run in to Ardour, through some
effects, then through PHASEX, where I had the filter under realtime
control from a MIDI controller, and finally back in to Ardour for some
more effects. It's crazy, but it was a tonne of fun to make.
Thanks
Leigh
8:33 a.m.
On 02/07/2011 06:54 AM, Bearcat M. Şandor wrote:
Folks,
I've been thinking about playing with sound effects. For a lot of my
life, i've been interested in the sounds that various things make and i
am wondering if there is a linux program that would allow me to take a
sound clip and manipulate it. Like taking the sound of a chair squeak
and distort it, stretch it, apply filters to it, alter the tone etc.
Is there a linux gui app for this sort of thing? Preferably in with the
gtk tool kit as i have that all loaded up?
I did some digging around, but i didn't see much. Of course there's
always the chance that i looked at something that would do it well and
didn't recognise it.
What is that kind of program called?
Thanks,
Bearcat
Smasher. It's a beat slicer.
http://smasher.sourceforge.net/
Not real-time though.
Best,
Jeremy
9 Feb
9 Feb
10:13 a.m.
Excerpts from Jeremy Jongepier's message of 2011-02-07 09:33:38 +0100:
On 02/07/2011 06:54 AM, Bearcat M. Şandor wrote:
Do you happen to know how to change the audio interface or even better
how to tell it to use jack? Smasher seems to suffer from the gstreamer
curse, it's so simple and humble that it uses the laptops built in
soundcard & speakers instead of the nice interface & speakers.
Folks,
I've been thinking about playing with sound effects. For a lot of my
life, i've been interested in the sounds that various things make and i
am wondering if there is a linux program that would allow me to take a
sound clip and manipulate it. Like taking the sound of a chair squeak
and distort it, stretch it, apply filters to it, alter the tone etc.
Is there a linux gui app for this sort of thing? Preferably in with the
gtk tool kit as i have that all loaded up?
I did some digging around, but i didn't see much. Of course there's
always the chance that i looked at something that would do it well and
didn't recognise it.
What is that kind of program called?
Thanks,
Bearcat
Smasher. It's a beat slicer.
http://smasher.sourceforge.net/
Not real-time though.
Best,
Jeremy 
10:40 a.m.
On 02/09/2011 11:13 AM, Philipp Überbacher wrote:
Do you happen to know how to change the audio
interface or even better
how to tell it to use jack? Smasher seems to suffer from the gstreamer
curse, it's so simple and humble that it uses the laptops built in
soundcard& speakers instead of the nice interface& speakers.
Hello Philipp,
You have to run gstreamer-properties from a terminal, set Default Output
- Plugin to Custom and then enter jackaudiosink in the Pipeline field.
Best,
Jeremy
11:41 a.m.
Excerpts from Jeremy Jongepier's message of 2011-02-09 11:40:36 +0100:
On 02/09/2011 11:13 AM, Philipp Überbacher wrote:
Thanks Jeremy,
do you happen to know what gstreamer-properties is part of? It seems
like my gstreamer didn't come with this program.
I have the following installed:
gstreamer0.10 0.10.32
GStreamer Multimedia Framework
gstreamer0.10-base 0.10.32
GStreamer Multimedia Framework Base plugin libraries
gstreamer0.10-base-plugins 0.10.32
GStreamer Multimedia Framework Base Plugins
gstreamer0.10-good-slim 0.10.24
GStreamer Multimedia Framework Good plugin libraries with reduced and no GNOME
dependencies
gstreamer0.10-python 0.10.21
Python bindings for GStreamer 0.10
Could this program happen to be part of gstreamer0.10-good and a gnome
program?
Do you happen to know how to change the audio
interface or even better
how to tell it to use jack? Smasher seems to suffer from the gstreamer
curse, it's so simple and humble that it uses the laptops built in
soundcard& speakers instead of the nice interface& speakers.
Hello Philipp,
You have to run gstreamer-properties from a terminal, set Default Output
- Plugin to Custom and then enter jackaudiosink in the Pipeline field.
Best,
Jeremy
11:54 a.m.
On 02/09/2011 12:41 PM, Philipp Überbacher wrote:
do you happen to know what gstreamer-properties is
part of?
It's part of the gnome-media-common package for Ubuntu. But you can also
set it with gconf-editor or even gconftool:
gconftool -t string -s /system/gstreamer/0.10/default/audiosink
"jackaudiosink"
Best,
Jeremy
11:58 a.m.
On Wed, Feb 9, 2011 at 6:54 AM, Jeremy Jongepier <autostatic(a)gmail.com> wrote:
On 02/09/2011 12:41 PM, Philipp Überbacher wrote:
more comprehensive info at:
http://jackaudio.org/gstreamer_via_jack
it would be nice if more and more people read the FAQ :)
do you happen to know what gstreamer-properties is part of?
It's part of the gnome-media-common package for Ubuntu. But you can also set
it with gconf-editor or even gconftool:
gconftool -t string -s /system/gstreamer/0.10/default/audiosink
"jackaudiosink"
4:17 p.m.
Excerpts from Paul Davis's message of 2011-02-09 12:58:37 +0100:
On Wed, Feb 9, 2011 at 6:54 AM, Jeremy Jongepier
<autostatic(a)gmail.com> wrote:
Thanks Jeremy and Paul.
It doesn't seem to be up-to-date because gst-inspect-0.10 jackaudiosink
fails with the message "No such element or plugin 'jackaudiosink'" and
base/bad/ugly plugins installed. I don't want to install good-plugins
because it would pull a lot of otherwise unneeded gnome dependencies,
such as orbit2, polkit, gconf, libgnome-keyring and libsoup-gnome. I'll
try to build good-plugins without those things and see whether jack is
there.
Is there a way to configure gstreamer on a modern gnome-free system,
possibly using a common tool such as a text editor?
On 02/09/2011 12:41 PM, Philipp Überbacher
wrote:
more comprehensive info at:
http://jackaudio.org/gstreamer_via_jack
it would be nice if more and more people read the FAQ :)
do you happen to know what gstreamer-properties is part of?
It's part of the gnome-media-common package for Ubuntu. But you can also set
it with gconf-editor or even gconftool:
gconftool -t string -s /system/gstreamer/0.10/default/audiosink
"jackaudiosink"
4:31 p.m.
On Wed, Feb 9, 2011 at 11:17 AM, Philipp Überbacher
<hollunder(a)lavabit.com> wrote:
It doesn't seem to be up-to-date because
gst-inspect-0.10 jackaudiosink
fails with the message "No such element or plugin 'jackaudiosink'" and
base/bad/ugly plugins installed. I don't want to install good-plugins
because it would pull a lot of otherwise unneeded gnome dependencies,
you would have to check what package your distro has put the gstreamer
JACK plugin into. it varies from distro to disto.
note that gstreamer itself doesn't need most of the things you
mentioned - most of them are artifacts of the way its been packaged.
Is there a way to configure gstreamer on a modern
gnome-free system,
possibly using a common tool such as a text editor?
you are not configuring gstreamer as much as system properties which
gstreamer happens to look up. i know of no way to do this with a text
editor, and just as it used to be that you could screw up your system
by editing the windows registry with a text editor, i suspect you'd
run into the same issue if you tried that with the configuration
information normally controlled by gconftool and friends.
10 Feb
10 Feb
7:57 p.m.
Excerpts from Paul Davis's message of 2011-02-09 17:31:40 +0100:
It's an xml format so it should be possible to edit it, but I managed to
build and install a reasonably stripped down version of gconf.
gconftool-2 is also not easy to handle, the manpage is incomplete and
out of date. I'd appreciate it if you could add the following lines to
your jack FAQ:
gconftool-2 -s /system/gstreamer/0.10/audio/default/musicaudiosink
["jackaudiosink buffer-time=2000000"] --type=list --list-type=string
gconftool-2 -s /system/gstreamer/0.10/audio/default/audiosink
["jackaudiosink buffer-time=2000000"] --type=list --list-typ
e=string
Those lines seem to work in case of smasher at least, but sadly not for
firefox (ogg playback).
Thanks, this is a step closer to usable gstreamer.
On Wed, Feb 9, 2011 at 11:17 AM, Philipp Überbacher
<hollunder(a)lavabit.com> wrote:
It's in the good-plugins on Arch Linux,
It doesn't seem to be up-to-date because
gst-inspect-0.10 jackaudiosink
fails with the message "No such element or plugin 'jackaudiosink'" and
base/bad/ugly plugins installed. I don't want to install good-plugins
because it would pull a lot of otherwise unneeded gnome dependencies,
you would have to check what package your distro has put the gstreamer
JACK plugin into. it varies from distro to disto. note that gstreamer itself doesn't need most of
the things you
mentioned - most of them are artifacts of the way its been packaged.
After messing around for a while I managed to build gstreamer and gconf
without most gnome stuff (gconf being the obvious exception, and it
needs ORBIT for some reason).
Is there a way
to configure gstreamer on a modern gnome-free system,
possibly using a common tool such as a text editor?
you are not configuring gstreamer as much as system properties which
gstreamer happens to look up. i know of no way to do this with a text
editor, and just as it used to be that you could screw up your system
by editing the windows registry with a text editor, i suspect you'd
run into the same issue if you tried that with the configuration
information normally controlled by gconftool and friends. 
9 Feb
9 Feb
12:02 p.m.
On Wed, Feb 9, 2011 at 12:41 PM, Philipp Überbacher
<hollunder(a)lavabit.com>wrote;wrote:
Excerpts from Jeremy Jongepier's message of
2011-02-09 11:40:36 +0100:
You're on Archlinux, right? pkgfile is your friend :)
$ pkgfile gstreamer-properties
extra/gnome-media
extra/gnome-media-pulse
renato
On 02/09/2011 11:13 AM, Philipp Überbacher
wrote:
Thanks Jeremy,
do you happen to know what gstreamer-properties is part of? It seems
like my gstreamer didn't come with this program.
Do you happen to know how to change the audio
interface or even better
how to tell it to use jack? Smasher seems to suffer from the gstreamer
curse, it's so simple and humble that it uses the laptops built in
soundcard& speakers instead of the nice interface& speakers.
Hello Philipp,
You have to run gstreamer-properties from a terminal, set Default Output
- Plugin to Custom and then enter jackaudiosink in the Pipeline field.
Best,
Jeremy 
5:19 p.m.
On Wed, Feb 9, 2011 at 5:32 PM, Renato Budinich <rennabh(a)gmail.com> wrote:
You're on Archlinux, right? pkgfile is your friend :)
$ pkgfile gstreamer-properties
extra/gnome-media
extra/gnome-media-pulse
renato
On debian/ubuntu you can do
$ dpkg -S gstreamer-properties
to answer such questions
7 Feb
7 Feb
9:27 a.m.
On 02/07/2011 06:54 AM, Bearcat M. Şandor wrote:
Folks,
I've been thinking about playing with sound effects. For a lot of my
life, i've been interested in the sounds that various things make and i
am wondering if there is a linux program that would allow me to take a
sound clip and manipulate it. Like taking the sound of a chair squeak
and distort it, stretch it, apply filters to it, alter the tone etc.
Is there a linux gui app for this sort of thing? Preferably in with the
gtk tool kit as i have that all loaded up?
not for everything you mentioned. try freqtweak, tons of fun. does
spectral delay, gating, and eq.
for stretching, the rubberband tools come to mind. distortion is
available from numerous ladspa plugins, or try guitarix.
4:02 p.m.
New subject: [LAU] Linux programs for creatiing/manipulating sound effects
for stretching, the rubberband tools come to mind
Also there is PaulStretch. It don't work with jack and even may have problems
with some files. But it can write modified sound to fine.
It is notable for me also because has spectral blur function, which in this tool is called
"Spread".
8:12 p.m.
Folks,
Thanks for all of the replies. Ardour + plug-ins looks like my best best
so far, but i'll check out some of the other options as well. It seems
like a good excuse to learn ardour which i've been meaning to do anyhow.
When i'm playing with this, are there any guide lines that i can keep in
mind to avoid hurting my ears and stereo equipment when i'm playing with
stuff? Just adjust all knobs slowly till i know what they do?
Now all i need is to find some sort of battery powered mic with a little
sdhc recorder in it, that i could pull out of my pocket when ever i
heard something interesting. I'd like to start collecting sounds and
playing with them.
I love this mailing list.
Bearcat
10:03 p.m.
Am 07.02.2011 21:12, schrieb Bearcat M. Sandor:
Folks,
Thanks for all of the replies. Ardour + plug-ins looks like my best best
so far, but i'll check out some of the other options as well. It seems
like a good excuse to learn ardour which i've been meaning to do anyhow.
When i'm playing with this, are there any guide lines that i can keep in
mind to avoid hurting my ears and stereo equipment when i'm playing with
stuff? Just adjust all knobs slowly till i know what they do?
I think this will not be available.
Especially modulation-effects like flangers always have the potential to
produce extreme peaks. If you want to limit the danger to your
ears/hardware, put a capable limiter like foo lookahead limiter in
Ardours Master-track. To be even more secure connect Ardours master-outs
not to your soundcard but to jamin inputs. Then you connect jamin to
your hardware-ports. Jamin has got a brickwall-limiter that justifies
its name ;-)
Now all i need is to find some sort of battery powered mic with a little
sdhc recorder in it, that i could pull out of my pocket when ever i
heard something interesting. I'd like to start collecting sounds and
playing with them.
I love this mailing list.
you are most welcome :-)
Bearcat
_______________________________________________
Linux-audio-user mailing list
Linux-audio-user(a)lists.linuxaudio.org
http://lists.linuxaudio.org/listinfo/linux-audio-user
11 Feb
11 Feb
2 p.m.
On 02/07/2011 11:03 PM, Hartmut Noack wrote:
Am 07.02.2011 21:12, schrieb Bearcat M. Sandor:
not strictly necessary - the limiter is also availabe as a ladspa plugin
that can be put into ardour's master bus.
i find either a foolproof analog volume control or a Big Red Switch(tm)
in the immediate vicinity of the keyboard very reassuring.
Folks,
Thanks for all of the replies. Ardour + plug-ins looks like my best best
so far, but i'll check out some of the other options as well. It seems
like a good excuse to learn ardour which i've been meaning to do anyhow.
When i'm playing with this, are there any guide lines that i can keep in
mind to avoid hurting my ears and stereo equipment when i'm playing with
stuff? Just adjust all knobs slowly till i know what they do?
I think this will not be available.
Especially modulation-effects like flangers always have the potential to
produce extreme peaks. If you want to limit the danger to your
ears/hardware, put a capable limiter like foo lookahead limiter in
Ardours Master-track. To be even more secure connect Ardours master-outs
not to your soundcard but to jamin inputs. Then you connect jamin to
your hardware-ports. Jamin has got a brickwall-limiter that justifies
its name ;-) 
8:24 p.m.
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Jörn Nettingsmeier wrote:
On 02/07/2011 11:03 PM, Hartmut Noack wrote:
http://plessas.mur.at/IMG_5297.JPG_small.jpg
http://plessas.mur.at/IMG_5363.JPG_small.jpg
http://plessas.mur.at/IMG_5492.JPG_small.jpg
Sorry, I couldn't resist posting this :-)
The famous IEM Big Red Switch(tm) in use at the CUBE and MUMUTH concert
venues with their 24+ channel ambisonics PA system.
best, P.
Am 07.02.2011 21:12, schrieb Bearcat M. Sandor:
not strictly necessary - the limiter is also availabe as a ladspa plugin
that can be put into ardour's master bus.
i find either a foolproof analog volume control or a Big Red Switch(tm)
in the immediate vicinity of the keyboard very reassuring.
Folks,
Thanks for all of the replies. Ardour + plug-ins looks like my best best
so far, but i'll check out some of the other options as well. It seems
like a good excuse to learn ardour which i've been meaning to do anyhow.
When i'm playing with this, are there any guide lines that i can keep in
mind to avoid hurting my ears and stereo equipment when i'm playing with
stuff? Just adjust all knobs slowly till i know what they do?
I think this will not be available.
Especially modulation-effects like flangers always have the potential to
produce extreme peaks. If you want to limit the danger to your
ears/hardware, put a capable limiter like foo lookahead limiter in
Ardours Master-track. To be even more secure connect Ardours master-outs
not to your soundcard but to jamin inputs. Then you connect jamin to
your hardware-ports. Jamin has got a brickwall-limiter that justifies
its name ;-) 
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7 Feb
7 Feb
8:52 p.m.
On 02/07/2011 05:02 PM, Mike Cookson wrote:
FWIW. http://linuxaudio.org/mailarchive/lad/2010/9/30/174181
On 10/04/2010 08:18 PM, Paul N wrote:
"Thanks for JACK support, I will include it into the next version."
meanwhile:
git clone git://rg42.org/paulstretch
cd paulstretch
./compile_linux_fftw_jack.sh
for
stretching, the rubberband tools come to mind
Also there is PaulStretch. It
don't work with jack and even may have problems with some files.
But it can write modified sound to fine. It is notable for me also because has spectral blur
function, which in this tool is called "Spread".
Do you have a link to that?
BTW. I second the option to try 'freqtweak'
Here freqtweak and paulstretch are often the first tools of of choice to
alienate sounds.
--
Robin Gareus
web: http://gareus.org/ mail: robin(a)gareus.org
lab: http://citu.fr/ chat: xmpp:rgareus@ik.nu
Public Key at http://pgp.mit.edu/ http://gareus.org/public.asc
Fingerprint : 7107 840B 4DC9 C948 076D 6359 7955 24F1 4F95 2B42
10 Feb
10 Feb
11:36 a.m.
New subject: [LAU] Linux programs for creatiing/manipulating sound effects
On 10/04/2010 08:18 PM, Paul N wrote:
"Thanks for JACK support, I will include it into the next version."
meanwhile:
git clone git://rg42.org/paulstretch
cd paulstretch
./compile_linux_fftw_jack.sh
Cool, thanks. Btw before jack was available, i some
files could not be played back due to mismatch of some file format details.
It is notable
for me also because has spectral blur function, which in this tool is called
"Spread".
Do you have a link to that? 
11 Feb
11 Feb
3:56 p.m.
On 02/10/2011 12:36 PM, Mike Cookson wrote:
Yesterday paulstretch 2.1 came out:
http://hypermammut.sourceforge.net/paulstretch/
It is on "Process" tab.
Aw right. I though you might mean that, but you made it sound as if it
were a different application.
On 10/04/2010
08:18 PM, Paul N wrote:
"Thanks for JACK support, I will include it into the next version."
meanwhile:
git clone git://rg42.org/paulstretch
cd paulstretch
./compile_linux_fftw_jack.sh
Cool, thanks. Btw before jack was available, i some
files could not be played back due to mismatch of some file format details. It is notable for me also because has spectral blur
function, which in this tool is called "Spread".
Do you have a link to that? Btw, sometimes i'm thinking about to extract it to
some plugin or standalone jack app.
It's already a standalone jack-app and it is hard to wrap it into a
plugin because it changes the duration of the sound: i.e. you feed it
with 1 second of audio and it returns 10 seconds.
I suppose you could do some tricks or require constraints.. The
file-reader and playback code is nicely separated from the engine in
paulstretch's code. YMMV
2c,
robin
13 Feb
13 Feb
10:32 a.m.
Excerpts from Robin Gareus's message of 2011-02-11 16:56:42 +0100:
On 02/10/2011 12:36 PM, Mike Cookson wrote:
Yesterday paulstretch 2.1 came out:
http://hypermammut.sourceforge.net/paulstretch/
Can you tell me how to get sound out of it? Version 2.1 (the
'about' says 2.2). I tried with and without jack, mp3, ogg and wav
files, without any success. I never see jack ports appear. With wav
files I get the output:
AInputS::seek 0 0 54110260 - possible audiofile
bug (it always seeks to the end of the file)
Portaudio seems to work, tested with audacity. Any ideas?
On
10/04/2010 08:18 PM, Paul N wrote:
"Thanks for JACK support, I will include it into the next version."
meanwhile:
git clone git://rg42.org/paulstretch
cd paulstretch
./compile_linux_fftw_jack.sh
Cool, thanks. Btw before jack was available, i some
files could not be played back due to mismatch of some file format details.
12:12 p.m.
New subject: paulstretch - was Linux programs for creatiing/manipulating sound effects
On 02/13/2011 11:31 AM, Philipp Überbacher wrote:
Excerpts from Robin Gareus's message of 2011-02-11
16:56:42 +0100:
Did you compile it with './compile_linux_fftw_jack.sh' ?
I start JACK, start paulstretch, load some mp3 and just press play.
works just fine with .wav files too and the samplerate does not need to
match JACK's samplerate..
FWIW: you can set the environment variable JACK_AUTOCONNECT
to decide the port-connection behaviour. The default is
'system:playback_' - setting it to "DISABLE" will leave the output
ports
unconnected.
robin
On 02/10/2011 12:36 PM, Mike Cookson wrote:
Can you tell me how to get sound out of it? Version 2.1 (the 'about'
says 2.2). I tried with and without jack, mp3, ogg and wav files,
without any success. I never see jack ports appear. With wav files I
get the output: AInputS::seek 0 0 54110260 - possible audiofile bug
(it always seeks to the end of the file)
Portaudio seems to work, tested with audacity. Any ideas? Cool, thanks. Btw before jack was available, i
some files could
not be played back due to mismatch of some file format details.
Yesterday paulstretch 2.1 came out:
http://hypermammut.sourceforge.net/paulstretch/ _______________________________________________
Linux-audio-user
mailing list Linux-audio-user(a)lists.linuxaudio.org
http://lists.linuxaudio.org/listinfo/linux-audio-user
6:32 p.m.
New subject: paulstretch - was Linux programs for creatiing/manipulating sound effects
On 2/13/11, Robin Gareus wrote:
OTOH JACK port appears and gets automatically connected only after you
press Play button. Then it disappears.
Alexandre Prokoudine
http://libregraphicsworld.org
Can you tell
me how to get sound out of it? Version 2.1 (the 'about'
says 2.2). I tried with and without jack, mp3, ogg and wav files,
without any success. I never see jack ports appear. With wav files I
get the output: AInputS::seek 0 0 54110260 - possible audiofile bug
(it always seeks to the end of the file)
Portaudio seems to work, tested with audacity. Any ideas?
Did you compile it with './compile_linux_fftw_jack.sh' ?
I start JACK, start paulstretch, load some mp3 and just press play.
6:41 p.m.
New subject: paulstretch - was Linux programs for creatiing/manipulating sound effects
Excerpts from Alexandre Prokoudine's message of 2011-02-13 19:32:02 +0100:
On 2/13/11, Robin Gareus wrote:
OTOH JACK port appears and gets automatically connected only after you
press Play button. Then it disappears.
Alexandre Prokoudine
http://libregraphicsworld.org
That seems to be typical portaudio behavior. It's exactly the same with
audacity. I wish they'd change it to persistent ports someday.
Can you
tell me how to get sound out of it? Version 2.1 (the 'about'
says 2.2). I tried with and without jack, mp3, ogg and wav files,
without any success. I never see jack ports appear. With wav files I
get the output: AInputS::seek 0 0 54110260 - possible audiofile bug
(it always seeks to the end of the file)
Portaudio seems to work, tested with audacity. Any ideas?
Did you compile it with './compile_linux_fftw_jack.sh' ?
I start JACK, start paulstretch, load some mp3 and just press play.
14 Feb
14 Feb
10:21 a.m.
New subject: paulstretch - was Linux programs for creatiing/manipulating sound effects
On 02/13/2011 07:41 PM, Philipp Überbacher wrote:
That seems to be typical portaudio behavior. It's
exactly the same with
audacity. I wish they'd change it to persistent ports someday.
For Audacity to get rid of this behaviour I patched the source to
prevent PortAudio from connecting automatically. And with the help from
some regex magic in QjackCtl ports are pretty persistent now. But it's
still not 100% reliable I'd love to see Audacity with proper JACK support.
Best,
Jeremy
22 Feb
22 Feb
1:07 p.m.
New subject: paulstretch - was Linux programs for creatiing/manipulating sound effects
2011/2/13 Philipp Überbacher <hollunder(a)lavabit.com>om>:
Excerpts from Alexandre Prokoudine's message of
2011-02-13 19:32:02 +0100:
+1
BTW VLC jack plugin behave same, I am using VLC player via jack for
live performances
quite a lot and there is always audible click when I start playback
and VLC connect to jack.
Pretty annoying. :(
regards
mira
On 2/13/11, Robin Gareus wrote:
OTOH JACK port appears and gets automatically connected only after you
press Play button. Then it disappears.
Alexandre Prokoudine
http://libregraphicsworld.org
That seems to be typical portaudio behavior. It's exactly the same with
audacity. I wish they'd change it to persistent ports someday. Can you
tell me how to get sound out of it? Version 2.1 (the 'about'
says 2.2). I tried with and without jack, mp3, ogg and wav files,
without any success. I never see jack ports appear. With wav files I
get the output: AInputS::seek 0 0 54110260 - possible audiofile bug
(it always seeks to the end of the file)
Portaudio seems to work, tested with audacity. Any ideas?
Did you compile it with './compile_linux_fftw_jack.sh' ?
I start JACK, start paulstretch, load some mp3 and just press play.
13 Feb
13 Feb
6:44 p.m.
New subject: paulstretch - was Linux programs for creatiing/manipulating sound effects
On 02/13/2011 07:32 PM, Alexandre Prokoudine wrote:
On 2/13/11, Robin Gareus wrote:
OTOH JACK port appears and gets automatically connected only after you
press Play button. Then it disappears.
Alexandre Prokoudine
http://libregraphicsworld.org
yes, however pausing playback keeps the JACK port.
Retaining the JACK-client port for the whole session would require much
more intrusive changes and is not trivial with the current architecture
of paulstretch.
I don't really see a use-case where this is needed for paulstretch;
besides: either auto-connect to some jack-patchbay can do the job.
robin
--
Robin Gareus
web: http://gareus.org/ mail: robin(a)gareus.org
lab: http://citu.fr/ chat: xmpp:rgareus@ik.nu
Public Key at http://pgp.mit.edu/ http://gareus.org/public.asc
Fingerprint : 7107 840B 4DC9 C948 076D 6359 7955 24F1 4F95 2B42
Can you
tell me how to get sound out of it? Version 2.1 (the 'about'
says 2.2). I tried with and without jack, mp3, ogg and wav files,
without any success. I never see jack ports appear. With wav files I
get the output: AInputS::seek 0 0 54110260 - possible audiofile bug
(it always seeks to the end of the file)
Portaudio seems to work, tested with audacity. Any ideas?
Did you compile it with './compile_linux_fftw_jack.sh' ?
I start JACK, start paulstretch, load some mp3 and just press play.
11:49 p.m.
New subject: paulstretch - was Linux programs for creatiing/manipulating sound effects
On 2/13/11, Robin Gareus wrote:
Retaining the JACK-client port for the whole session
would require much
more intrusive changes and is not trivial with the current architecture
of paulstretch.
I don't really see a use-case where this is needed for paulstretch;
besides: either auto-connect to some jack-patchbay can do the job.
There are other things, but I'm not quite sure about the best channel
to report them. Things like seeking not working properly, or rendering
to a 4 minutes long sound file taking an hour on a CoreQuad. Should
that be reported to Paul directly?
Alexandre Prokoudine
http://libregraphicsworld.org
14 Feb
14 Feb
1:26 a.m.
New subject: paulstretch - was Linux programs for creatiing/manipulating sound effects
On 02/14/2011 12:49 AM, Alexandre Prokoudine wrote:
On 2/13/11, Robin Gareus wrote:
seeking has quite some latency until both GUI and sound react, but other
than that it works. - I suppose this is due to computational overhead
and there's not much that can be done about. maybe the GUI behaviour can
be changed to be less "jumpy" ?!
Retaining the JACK-client port for the whole
session would require much
more intrusive changes and is not trivial with the current architecture
of paulstretch.
I don't really see a use-case where this is needed for paulstretch;
besides: either auto-connect to some jack-patchbay can do the job.
There are other things, but I'm not quite sure about the best channel
to report them. Things like seeking not working properly, or rendering
to a 4 minutes long sound file taking an hour on a CoreQuad. Should
that be reported to Paul directly?
yes. If it's related to the JACK output, keep me in the loop.
Cheers!
robin
Alexandre Prokoudine
http://libregraphicsworld.org
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5:18 p.m.
New subject: [LAU] paulstretch - was Linux programs for creatiing/manipulating sound effects
OTOH JACK port
appears and gets automatically connected only after you
press Play button. Then it disappears.
Alexandre Prokoudine
http://libregraphicsworld.org
yes, however pausing playback keeps the JACK port.
5:26 p.m.
New subject: [LAU] Linux programs for creatiing/manipulating sound effects
It's already a standalone jack-app and it is hard
to wrap it into a
plugin because it changes the duration of the sound: i.e. you feed it
with 1 second of audio and it returns 10 seconds.
I suppose you could do some tricks or require constraints.. The
file-reader and playback code is nicely separated from the engine in
paulstretch's code. YMMV
I meand - move only that blur module, not entire tool
))). As for stretch module - it is nice candidate for audacity modules (i'm not sure,
is there any open standard for non-linear audio plugins).
5:32 p.m.
New subject: [LAU] Linux programs for creatiing/manipulating sound effects
Oops, forgot about new theme :)
7 Feb
7 Feb
10:55 a.m.
Am 07.02.2011 06:54, schrieb Bearcat M. Şandor:
Folks,
I've been thinking about playing with sound effects. For a lot of my
life, i've been interested in the sounds that various things make and i
am wondering if there is a linux program that would allow me to take a
sound clip and manipulate it.
Ardour.
While Ardour may not be the typical Waveeditor for Soundeffects it still
offers the most advanced features for such tasks:
1.) completely realtime -- adjust the parameters of effect-software
while you actually listening to what you do.
2.) non-destructive, sample-accurate editing
3.) complete automatisation -- draw a curve to change parameters in
effects and filters in the timeline.
4.) acceptable offline time-streching.
5.) complete Jack-integration -- not satisfied with what you can do with
you chair-sqeak using plug ins? Send the output of Ardour to standalone
FX-software like Guitarix or Autotalent and record the results to a new
track in Ardour.
6.) export to professional formats in a professional manner.
The only program I know, that is up to date and can do some stuff not
easily possible with Ardour would be SND. Another very interesting
Application for your Task is Sonic Visualizer.
Combining these 3 should give you access to near-unlimited possibilities
for your task.
Like taking the sound of a chair squeak
and distort it, stretch it, apply filters to it, alter the tone etc.
Is there a linux gui app for this sort of thing? Preferably in with the
gtk tool kit as i have that all loaded up?
I did some digging around, but i didn't see much. Of course there's
always the chance that i looked at something that would do it well and
didn't recognise it.
What is that kind of program called?
Thanks,
Bearcat
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11:16 a.m.
Am Montag 07 Februar 2011, 06:54:56 schrieb Bearcat M. Şandor:
Folks,
I've been thinking about playing with sound effects. For a lot of my
life, i've been interested in the sounds that various things make and i
am wondering if there is a linux program that would allow me to take a
sound clip and manipulate it. Like taking the sound of a chair squeak
and distort it, stretch it, apply filters to it, alter the tone etc.
Is there a linux gui app for this sort of thing? Preferably in with the
gtk tool kit as i have that all loaded up?
I did some digging around, but i didn't see much. Of course there's
always the chance that i looked at something that would do it well and
didn't recognise it.
Along with what other posters answered, i'd say have a look at mammut:
http://archive.notam02.no/arkiv/doc/mammut/
it can mangle your soundfiles in astounding ways, with algos you never
heard about before, and rendering completely unexpected results.
Then there's paulstretch, by the author of ZynAddSubFX, being somewhat
more specialized as it only does time stretching, and it does that very well
for rather big timestretch ratios:
http://hypermammut.sourceforge.net/paulstretch/
Just one more idea: I got quite interesting results by abusing dsp algorithms,
like e.g.audacity's denoiser module: I took some hissy/noisy garbage from old
recordings i had lying around, and experimented with the denoiser parameters,
normalized after processing, applied it several times in a row etc. The artifacts it
produces tend to have a "resonating" quality so that noise starts to have a
tonal
component...
What is that kind of program called?
Thanks,
Bearcat
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Edgar
3:40 p.m.
New subject: [LAU] Linux programs for creatiing/manipulating sound effects
For non-realtime (including non-linear, like montage) processing you need only plugins
(ladspa, lv2, vamp) and some editor like Audacity, mhWaveEdit or something other.
For realtime (also called non-destructive editing... hm, probably, they are right :) you
need set of various software, that could be used at one time and be connected each to
other).
It would be better for you to begin with special distros, where all necessary staff is
installed and has good settings out of box. My favorite is kxstudio, as (imho), best
variant vor newbies.
Lookup for some (or more) articles and videos.
9:22 p.m.
Hi Mike,
On 02/07/2011 04:40 PM, Mike Cookson wrote:
For non-realtime (including non-linear, like montage)
processing you
need only plugins (ladspa, lv2, vamp) and some editor like Audacity,
mhWaveEdit or something other.
For realtime (also called
non-destructive editing... hm, probably, they are right :) you need
set of various software, that could be used at one time and be
connected each to other).
real-time effects processing and non-destructive editing often go hand
in hand, but note that
"non-destructive" means that the original [audio] data will never be
modified. Any edit/effect/modifications are saved as new files (or
remebered as application-settings operating on the original data).
audio-editors (rezound, audacity, sweep, etc) are usually destructive:
load file, apply effect, save file -> original file is gone.
DAWs (ardour, qtractor, ecasound,..) are non-destructive:
reference file, apply effect or modifications to data
but to save the resulting audio, you need to bounce the track or export
the session.
It would be better for you to begin with special
distros, where all
necessary staff is installed and has good settings out of box. My
favorite is kxstudio, as (imho), best variant vor newbies.
Lookup for some (or more) articles and videos.
11:02 p.m.
I second SND which seems to be the editor but it comes with a learncurve.
Other candidates for extreme sound processing are Csound and Pure Data,
they even let you create your own processing patches.
Cheers,
Malte
--
----
media art + development
http://www.block4.com
new on iTunes: Notstandskomitee Automatenmusik
http://itunes.apple.com/us/album/automatenmusik/id383400418
8 Feb
8 Feb
12:22 p.m.
SuperCollider, PureData and CSound can do anything you want, you just
need how to ask. :-)
http://supercollider.sourceforge.net/
9 Feb
9 Feb
10:05 a.m.
Excerpts from Malte Steiner's message of 2011-02-07 23:04:27 +0100:
I second SND which seems to be the editor but it comes
with a learncurve.
Other candidates for extreme sound processing are Csound and Pure Data,
they even let you create your own processing patches.
Cheers,
Malte
SND seems to be extremely powerful and flexible but its learning curve
is insane. I just found out that it has a built-in extensive,
half-working, possibly useful help system.
There's SND-ls that aims to make SND usable by mere mortals, but I'm not
sure it achieved this goal yet:
http://archive.notam02.no/arkiv/src/snd-ls/
7:49 p.m.
On 2/9/2011 3:05 AM, Philipp Überbacher wrote:
SND seems to be extremely powerful and flexible but
its learning curve
is insane. I just found out that it has a built-in extensive,
half-working, possibly useful help system.
There's SND-ls that aims to make SND usable by mere mortals, but I'm not
sure it achieved this goal yet:
http://archive.notam02.no/arkiv/src/snd-ls/
I've understood (though not used)
ecasound to have a lower learning
curve and be just as able as snd. Is that correct?
Bearcat M. Sandor
10:16 p.m.
Excerpts from Bearcat M. Sandor's message of 2011-02-09 20:49:07 +0100:
On 2/9/2011 3:05 AM, Philipp Überbacher wrote:
It's hard to say because it's hard to compare those two programs. SND is
a completely scriptable graphical audio editor. Ecasound is a DAW
with a commandline interface and bindings for many programming
languages. I'm sure both can be used for pretty similar things but their
goals are different and they both have a rather steep learning curve.
SND seems to be extremely powerful and flexible
but its learning curve
is insane. I just found out that it has a built-in extensive,
half-working, possibly useful help system.
There's SND-ls that aims to make SND usable by mere mortals, but I'm not
sure it achieved this goal yet:
http://archive.notam02.no/arkiv/src/snd-ls/
I've understood (though not used)
ecasound to have a lower learning
curve and be just as able as snd. Is that correct?
Bearcat M. Sandor
10 Feb
10 Feb
9:52 a.m.
Am 09.02.2011 23:16, schrieb Philipp Überbacher:
Excerpts from Bearcat M. Sandor's message of
2011-02-09 20:49:07 +0100:
I think, a beginner will find some useful, interesting things in SND,
before the learning curve starts to become very steep. In Fact SND can
be used for many things like granular synthesis, basic filtering or
dynamics before you learn a single command for its scripting language.
But if you really want to control what is going on in SND you will have
to learn the basics of its command line interface.
On 2/9/2011 3:05 AM, Philipp Überbacher wrote:
It's hard to say because it's hard to compare those two programs. SND is
a completely scriptable graphical audio editor. Ecasound is a DAW
with a commandline interface and bindings for many programming
languages. I'm sure both can be used for pretty similar things but their
goals are different and they both have a rather steep learning curve. SND seems to be extremely powerful and flexible
but its learning curve
is insane. I just found out that it has a built-in extensive,
half-working, possibly useful help system.
There's SND-ls that aims to make SND usable by mere mortals, but I'm not
sure it achieved this goal yet:
http://archive.notam02.no/arkiv/src/snd-ls/
I've understood (though not used)
ecasound to have a lower learning
curve and be just as able as snd. Is that correct?
Bearcat M. Sandor
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8 Feb
8 Feb
7:35 a.m.
Robin Gareus wrote:
Hi Mike,
On 02/07/2011 04:40 PM, Mike Cookson wrote:
Audacity is import audio file, apply effect, save project (optional),
export in chosen format. It never replaces the original file.
It IS destructive in that it applies the effect to its imported copy of
the original audio. But that doesn't effect the original file unless you
chose to export to the same location in the same format with the same
filename.
--
David
gnome(a)hawaii.rr.com
authenticity, honesty, community
For non-realtime (including non-linear, like
montage) processing you
need only plugins (ladspa, lv2, vamp) and some editor like Audacity,
mhWaveEdit or something other.
For realtime (also called
non-destructive editing... hm, probably, they are right :) you need
set of various software, that could be used at one time and be
connected each to other).
real-time effects processing and non-destructive editing often go hand
in hand, but note that
"non-destructive" means that the original [audio] data will never be
modified. Any edit/effect/modifications are saved as new files (or
remebered as application-settings operating on the original data).
audio-editors (rezound, audacity, sweep, etc) are usually destructive:
load file, apply effect, save file -> original file is gone.
7:50 a.m.
Am 08.02.2011 08:35, schrieb david:
Robin Gareus wrote:
So there is a major dfference between audiofiles, you have imported and
audiofiles, you have recorded with audacity -- correct?
Hi Mike,
On 02/07/2011 04:40 PM, Mike Cookson wrote:
Audacity is import audio file, apply effect, save project (optional),
export in chosen format. It never replaces the original file. For non-realtime (including non-linear, like
montage) processing you
need only plugins (ladspa, lv2, vamp) and some editor like Audacity,
mhWaveEdit or something other.
For realtime (also called
non-destructive editing... hm, probably, they are right :) you need
set of various software, that could be used at one time and be
connected each to other).
real-time effects processing and non-destructive editing often go hand
in hand, but note that
"non-destructive" means that the original [audio] data will never be
modified. Any edit/effect/modifications are saved as new files (or
remebered as application-settings operating on the original data).
audio-editors (rezound, audacity, sweep, etc) are usually destructive:
load file, apply effect, save file -> original file is gone.
It IS destructive in that it applies the effect to its imported copy of
the original audio. But that doesn't effect the original file unless you
chose to export to the same location in the same format with the same
filename.
8:15 a.m.
Hartmut Noack wrote:
Am 08.02.2011 08:35, schrieb david:
No, I never have. I usually use JACK, and have never been able to make
Audacity work with JACK. I've only done the following things with Audacity:
1. Import 16 tracks of 32-bit WAV files (recorded on another machine
from my church band's Firewire interface using some Windows software)
and do basic mixing.
2. Trim and cleanup voice audio recordings made on my PDA.
3. Trim and convert wave files recorded using jack_capture.
4. Pitch shift prerecorded MP3s if needed for band members who play
solely by ear (if the recording's in Eb and we're playing it in D,
they're lost).
If you have the time and brains needed to learn Ardour, go for it!
--
David
gnome(a)hawaii.rr.com
authenticity, honesty, community
Robin Gareus wrote:
So there is a major dfference between audiofiles, you have imported and
audiofiles, you have recorded with audacity -- correct? Hi Mike,
On 02/07/2011 04:40 PM, Mike Cookson wrote:
Audacity is import audio file, apply effect, save project (optional),
export in chosen format. It never replaces the original file. For non-realtime (including non-linear, like
montage) processing you
need only plugins (ladspa, lv2, vamp) and some editor like Audacity,
mhWaveEdit or something other.
For realtime (also called
non-destructive editing... hm, probably, they are right :) you need
set of various software, that could be used at one time and be
connected each to other).
real-time effects processing and non-destructive editing often go hand
in hand, but note that
"non-destructive" means that the original [audio] data will never be
modified. Any edit/effect/modifications are saved as new files (or
remebered as application-settings operating on the original data).
audio-editors (rezound, audacity, sweep, etc) are usually destructive:
load file, apply effect, save file -> original file is gone.
12:50 p.m.
Am 08.02.2011 09:15, schrieb david:
Hartmut Noack wrote:
I think it is a myth, that Ardour is too complicated to learn for a
beginner.
The Interface may look somewhat intimidating at first glance but with
some basic rules in mind it is not too hard to be handeled.
1.) learn to ignore, what you cannot understand at first sight. about
99% of the default-settings are sane and usefull and if you need to
change something, you will find out how as soon as you need it.
2.) understand the interaction between the tracks in the editor and the
mixer. Tracks are sources, the mixer is manipulation both are connected
by a signalflow that is setup by Ardour automagically. In the beginning
you will not need to change anything in the routing later you will
seldom need to change some details.
With this in mind you can start using Ardour like this:
1.) start Jack
2.) start Ardour, type a name for your project and click OK
3.) rightklick under the master-track and choose "import audio" or
drag/drop WAV-files from a GTK-Filemanager like Nautilus.
4.) Hit Space an hear your files play.
Next step would be to insert some plugins in your tracks and start
fiddeling with them.
All the advanced stuff like set up recording from internal jack-clients,
automatisation of faders and plugins, special mouse-modes or usage of
the things in the right-click menues can be learned step by step after that.
I really believe that an absolute beginner who does not allow
him/herself to be intimidated by all the advanced features can begin to
use Ardour with fun and good results within 1 h after starting such a
program for the first time.
best regards
HZN
Am 08.02.2011 08:35, schrieb david:
No, I never have. I usually use JACK, and have never been able to make
Audacity work with JACK. I've only done the following things with Audacity:
1. Import 16 tracks of 32-bit WAV files (recorded on another machine
from my church band's Firewire interface using some Windows software)
and do basic mixing.
2. Trim and cleanup voice audio recordings made on my PDA.
3. Trim and convert wave files recorded using jack_capture.
4. Pitch shift prerecorded MP3s if needed for band members who play
solely by ear (if the recording's in Eb and we're playing it in D,
they're lost).
If you have the time and brains needed to learn Ardour, go for it!
Robin Gareus wrote:
So there is a major dfference between audiofiles, you have imported
and audiofiles, you have recorded with audacity -- correct? Hi Mike,
On 02/07/2011 04:40 PM, Mike Cookson wrote:
> For non-realtime (including non-linear, like montage) processing you
> need only plugins (ladspa, lv2, vamp) and some editor like Audacity,
> mhWaveEdit or something other.
>
> For realtime (also called
> non-destructive editing... hm, probably, they are right :) you need
> set of various software, that could be used at one time and be
> connected each to other).
real-time effects processing and non-destructive editing often go hand
in hand, but note that
"non-destructive" means that the original [audio] data will never be
modified. Any edit/effect/modifications are saved as new files (or
remebered as application-settings operating on the original data).
audio-editors (rezound, audacity, sweep, etc) are usually destructive:
load file, apply effect, save file -> original file is gone.
Audacity is import audio file, apply effect, save project (optional),
export in chosen format. It never replaces the original file.
4:33 p.m.
Hartmut Noack wrote:
Am 08.02.2011 09:15, schrieb david:
Perhaps it's a myth for others. I responded only from my own experience.
There are plenty of folk here who use Ardour and do wonderful things
with it. Advanced features? I couldn't tell an advanced feature from a
basic feature. I couldn't even figure out to simply record anything with
it, and Ardour's "automagic" setup didn't seem to include that
connection.
No insult to Ardour, I'm no audio techno whiz. I figured I'd wait til
Ardour 3 is released and I have the time to learn it.
--
David
gnome(a)hawaii.rr.com
authenticity, honesty, community
Hartmut Noack wrote:
I think it is a myth, that Ardour is too complicated to learn for a
beginner. Am 08.02.2011 08:35, schrieb david:
No, I never have. I usually use JACK, and have never been able to make
Audacity work with JACK. I've only done the following things with
Audacity:
1. Import 16 tracks of 32-bit WAV files (recorded on another machine
from my church band's Firewire interface using some Windows software)
and do basic mixing.
2. Trim and cleanup voice audio recordings made on my PDA.
3. Trim and convert wave files recorded using jack_capture.
4. Pitch shift prerecorded MP3s if needed for band members who play
solely by ear (if the recording's in Eb and we're playing it in D,
they're lost).
If you have the time and brains needed to learn Ardour, go for it!
Robin Gareus wrote:
> Hi Mike,
>
> On 02/07/2011 04:40 PM, Mike Cookson wrote:
>> For non-realtime (including non-linear, like montage) processing you
>> need only plugins (ladspa, lv2, vamp) and some editor like Audacity,
>> mhWaveEdit or something other.
>>
>> For realtime (also called
>> non-destructive editing... hm, probably, they are right :) you need
>> set of various software, that could be used at one time and be
>> connected each to other).
>
> real-time effects processing and non-destructive editing often go hand
> in hand, but note that
>
> "non-destructive" means that the original [audio] data will never be
> modified. Any edit/effect/modifications are saved as new files (or
> remebered as application-settings operating on the original data).
>
> audio-editors (rezound, audacity, sweep, etc) are usually destructive:
> load file, apply effect, save file -> original file is gone.
Audacity is import audio file, apply effect, save project (optional),
export in chosen format. It never replaces the original file.
So there is a major dfference between audiofiles, you have imported
and audiofiles, you have recorded with audacity -- correct? 
6:07 p.m.
On Tue, 08 Feb 2011 06:33:03 -1000
david <gnome(a)hawaii.rr.com> wrote:
Hartmut Noack wrote:
I had exactly the same experience. Twice over a two year period!
These days I either record with timemachine, then process later in audacity, or
record directly into Rosegarden, where I usually have associated MIDI tracks.
--
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.
Am 08.02.2011 09:15, schrieb david:
Perhaps it's a myth for others. I responded only from my own experience.
There are plenty of folk here who use Ardour and do wonderful things
with it. Advanced features? I couldn't tell an advanced feature from a
basic feature. I couldn't even figure out to simply record anything with
it, and Ardour's "automagic" setup didn't seem to include that
connection.
No insult to Ardour, I'm no audio techno whiz. I figured I'd wait til
Ardour 3 is released and I have the time to learn it. Hartmut Noack wrote:
I think it is a myth, that Ardour is too complicated to learn for a
beginner. Am 08.02.2011 08:35, schrieb david:
> Robin Gareus wrote:
>> Hi Mike,
>>
>> On 02/07/2011 04:40 PM, Mike Cookson wrote:
>>> For non-realtime (including non-linear, like montage) processing you
>>> need only plugins (ladspa, lv2, vamp) and some editor like Audacity,
>>> mhWaveEdit or something other.
>>>
>>> For realtime (also called
>>> non-destructive editing... hm, probably, they are right :) you need
>>> set of various software, that could be used at one time and be
>>> connected each to other).
>>
>> real-time effects processing and non-destructive editing often go hand
>> in hand, but note that
>>
>> "non-destructive" means that the original [audio] data will never be
>> modified. Any edit/effect/modifications are saved as new files (or
>> remebered as application-settings operating on the original data).
>>
>> audio-editors (rezound, audacity, sweep, etc) are usually destructive:
>> load file, apply effect, save file -> original file is gone.
>
> Audacity is import audio file, apply effect, save project (optional),
> export in chosen format. It never replaces the original file.
So there is a major dfference between audiofiles, you have imported
and audiofiles, you have recorded with audacity -- correct?
No, I never have. I usually use JACK, and have never been able to make
Audacity work with JACK. I've only done the following things with
Audacity:
1. Import 16 tracks of 32-bit WAV files (recorded on another machine
from my church band's Firewire interface using some Windows software)
and do basic mixing.
2. Trim and cleanup voice audio recordings made on my PDA.
3. Trim and convert wave files recorded using jack_capture.
4. Pitch shift prerecorded MP3s if needed for band members who play
solely by ear (if the recording's in Eb and we're playing it in D,
they're lost).
If you have the time and brains needed to learn Ardour, go for it!
9 Feb
9 Feb
10:28 a.m.
Folderol wrote:
On Tue, 08 Feb 2011 06:33:03 -1000
david <gnome(a)hawaii.rr.com> wrote:
I'm starting to record directly into Rosegarden more now.
Sometimes I just start up jack_capture, then clean up and such in Audacity.
--
David
gnome(a)hawaii.rr.com
authenticity, honesty, community
Hartmut Noack wrote:
I had exactly the same experience. Twice over a two year period!
These days I either record with timemachine, then process later in audacity, or
record directly into Rosegarden, where I usually have associated MIDI tracks. Am 08.02.2011 09:15, schrieb david:
Perhaps it's a myth for others. I responded only from my own
experience.
There are plenty of folk here who use Ardour and do wonderful things
with it. Advanced features? I couldn't tell an advanced feature from a
basic feature. I couldn't even figure out to simply record anything with
it, and Ardour's "automagic" setup didn't seem to include that
connection.
No insult to Ardour, I'm no audio techno whiz. I figured I'd wait til
Ardour 3 is released and I have the time to learn it. Hartmut Noack wrote:
> Am 08.02.2011 08:35, schrieb david:
>> Robin Gareus wrote:
>>> Hi Mike,
>>>
>>> On 02/07/2011 04:40 PM, Mike Cookson wrote:
>>>> For non-realtime (including non-linear, like montage) processing you
>>>> need only plugins (ladspa, lv2, vamp) and some editor like Audacity,
>>>> mhWaveEdit or something other.
>>>>
>>>> For realtime (also called
>>>> non-destructive editing... hm, probably, they are right :) you need
>>>> set of various software, that could be used at one time and be
>>>> connected each to other).
>>> real-time effects processing and non-destructive editing often go hand
>>> in hand, but note that
>>>
>>> "non-destructive" means that the original [audio] data will never
be
>>> modified. Any edit/effect/modifications are saved as new files (or
>>> remebered as application-settings operating on the original data).
>>>
>>> audio-editors (rezound, audacity, sweep, etc) are usually destructive:
>>> load file, apply effect, save file -> original file is gone.
>> Audacity is import audio file, apply effect, save project (optional),
>> export in chosen format. It never replaces the original file.
> So there is a major dfference between audiofiles, you have imported
> and audiofiles, you have recorded with audacity -- correct?
No, I never have. I usually use JACK, and have never been able to make
Audacity work with JACK. I've only done the following things with
Audacity:
1. Import 16 tracks of 32-bit WAV files (recorded on another machine
from my church band's Firewire interface using some Windows software)
and do basic mixing.
2. Trim and cleanup voice audio recordings made on my PDA.
3. Trim and convert wave files recorded using jack_capture.
4. Pitch shift prerecorded MP3s if needed for band members who play
solely by ear (if the recording's in Eb and we're playing it in D,
they're lost).
If you have the time and brains needed to learn Ardour, go for it!
I think it is a myth, that Ardour is too complicated to learn for a
beginner.
10:50 a.m.
Excerpts from david's message of 2011-02-09 11:28:10 +0100:
Folderol wrote:
For recording I'm also a fan of timemachine because of its simplicity.
You push the big button or press space and it records. You want another
take, press space twice. A simple peak meter is built in, the 'record
the past' feature is nice but I don't really need it. Most importantly
it simply works, jack_capture gives me problems with jack connections
and jack_capture_gui2 has some benefits but seems to have some problems
as well. If you use a vastly more complex program such as ardour or
rosegarden the potential for problems is much larger as well. I'm not
only talking about fatal errors but about all those tiny annoyances as
well. If there are fatal problems you use another program until those
are resolved, but those tiny annoyances are what distract you, ruin your
creativity and productivity. Most of us are the musicians and recording
engineers at the same time, so the process of recording must not be
distracting, it has to just work so you can keep on your muso-hat and be
creative.
On Tue, 08 Feb 2011 06:33:03 -1000
david <gnome(a)hawaii.rr.com> wrote:
I'm starting to record directly into Rosegarden more now.
Sometimes I just start up jack_capture, then clean up and such in Audacity. Hartmut Noack wrote:
I had exactly the same experience. Twice over a two year period!
These days I either record with timemachine, then process later in audacity, or
record directly into Rosegarden, where I usually have associated MIDI tracks. Am 08.02.2011 09:15, schrieb david:
> Hartmut Noack wrote:
>> Am 08.02.2011 08:35, schrieb david:
>>> Robin Gareus wrote:
>>>> Hi Mike,
>>>>
>>>> On 02/07/2011 04:40 PM, Mike Cookson wrote:
>>>>> For non-realtime (including non-linear, like montage) processing you
>>>>> need only plugins (ladspa, lv2, vamp) and some editor like Audacity,
>>>>> mhWaveEdit or something other.
>>>>>
>>>>> For realtime (also called
>>>>> non-destructive editing... hm, probably, they are right :) you need
>>>>> set of various software, that could be used at one time and be
>>>>> connected each to other).
>>>> real-time effects processing and non-destructive editing often go hand
>>>> in hand, but note that
>>>>
>>>> "non-destructive" means that the original [audio] data will
never be
>>>> modified. Any edit/effect/modifications are saved as new files (or
>>>> remebered as application-settings operating on the original data).
>>>>
>>>> audio-editors (rezound, audacity, sweep, etc) are usually destructive:
>>>> load file, apply effect, save file -> original file is gone.
>>> Audacity is import audio file, apply effect, save project (optional),
>>> export in chosen format. It never replaces the original file.
>> So there is a major dfference between audiofiles, you have imported
>> and audiofiles, you have recorded with audacity -- correct?
> No, I never have. I usually use JACK, and have never been able to make
> Audacity work with JACK. I've only done the following things with
> Audacity:
>
> 1. Import 16 tracks of 32-bit WAV files (recorded on another machine
> from my church band's Firewire interface using some Windows software)
> and do basic mixing.
>
> 2. Trim and cleanup voice audio recordings made on my PDA.
>
> 3. Trim and convert wave files recorded using jack_capture.
>
> 4. Pitch shift prerecorded MP3s if needed for band members who play
> solely by ear (if the recording's in Eb and we're playing it in D,
> they're lost).
>
> If you have the time and brains needed to learn Ardour, go for it!
>
I think it is a myth, that Ardour is too complicated to learn for a
beginner.
Perhaps it's a myth for others. I responded only from my own
experience.
There are plenty of folk here who use Ardour and do wonderful things
with it. Advanced features? I couldn't tell an advanced feature from a
basic feature. I couldn't even figure out to simply record anything with
it, and Ardour's "automagic" setup didn't seem to include that
connection.
No insult to Ardour, I'm no audio techno whiz. I figured I'd wait til
Ardour 3 is released and I have the time to learn it.
11:12 a.m.
Philipp U"berbacher wrote:
Excerpts from david's message of 2011-02-09
11:28:10 +0100:
Never had any problems with jack_capture.
Folderol wrote:
For recording I'm also a fan of timemachine because of its simplicity.
You push the big button or press space and it records. You want another
take, press space twice. A simple peak meter is built in, the 'record
the past' feature is nice but I don't really need it. Most importantly
it simply works, jack_capture gives me problems with jack connections On Tue, 08 Feb 2011 06:33:03 -1000
david <gnome(a)hawaii.rr.com> wrote:
I'm starting to record directly into Rosegarden more now.
Sometimes I just start up jack_capture, then clean up and such in Audacity. Hartmut Noack wrote:
> Am 08.02.2011 09:15, schrieb david:
>> Hartmut Noack wrote:
>>> Am 08.02.2011 08:35, schrieb david:
>>>> Robin Gareus wrote:
>>>>> Hi Mike,
>>>>>
>>>>> On 02/07/2011 04:40 PM, Mike Cookson wrote:
>>>>>> For non-realtime (including non-linear, like montage) processing
you
>>>>>> need only plugins (ladspa, lv2, vamp) and some editor like
Audacity,
>>>>>> mhWaveEdit or something other.
>>>>>>
>>>>>> For realtime (also called
>>>>>> non-destructive editing... hm, probably, they are right :) you
need
>>>>>> set of various software, that could be used at one time and be
>>>>>> connected each to other).
>>>>> real-time effects processing and non-destructive editing often go
hand
>>>>> in hand, but note that
>>>>>
>>>>> "non-destructive" means that the original [audio] data will
never be
>>>>> modified. Any edit/effect/modifications are saved as new files (or
>>>>> remebered as application-settings operating on the original data).
>>>>>
>>>>> audio-editors (rezound, audacity, sweep, etc) are usually
destructive:
>>>>> load file, apply effect, save file -> original file is gone.
>>>> Audacity is import audio file, apply effect, save project (optional),
>>>> export in chosen format. It never replaces the original file.
>>> So there is a major dfference between audiofiles, you have imported
>>> and audiofiles, you have recorded with audacity -- correct?
>> No, I never have. I usually use JACK, and have never been able to make
>> Audacity work with JACK. I've only done the following things with
>> Audacity:
>>
>> 1. Import 16 tracks of 32-bit WAV files (recorded on another machine
>> from my church band's Firewire interface using some Windows software)
>> and do basic mixing.
>>
>> 2. Trim and cleanup voice audio recordings made on my PDA.
>>
>> 3. Trim and convert wave files recorded using jack_capture.
>>
>> 4. Pitch shift prerecorded MP3s if needed for band members who play
>> solely by ear (if the recording's in Eb and we're playing it in D,
>> they're lost).
>>
>> If you have the time and brains needed to learn Ardour, go for it!
>>
> I think it is a myth, that Ardour is too complicated to learn for a
> beginner.
Perhaps it's a myth for others. I responded only from my own experience.
There are plenty of folk here who use Ardour and do wonderful things
with it. Advanced features? I couldn't tell an advanced feature from a
basic feature. I couldn't even figure out to simply record anything with
it, and Ardour's "automagic" setup didn't seem to include that
connection.
No insult to Ardour, I'm no audio techno whiz. I figured I'd wait til
Ardour 3 is released and I have the time to learn it.
I had exactly the same
experience. Twice over a two year period!
These days I either record with timemachine, then process later in audacity, or
record directly into Rosegarden, where I usually have associated MIDI tracks. and jack_capture_gui2
Didn't know there was such a thing!
has some benefits but seems to have some problems
as well. If you use a vastly more complex program such as ardour or
rosegarden the potential for problems is much larger as well. I'm not
only talking about fatal errors but about all those tiny annoyances as
well. If there are fatal problems you use another program until those
are resolved, but those tiny annoyances are what distract you, ruin your
creativity and productivity. Most of us are the musicians and recording
engineers at the same time, so the process of recording must not be
distracting, it has to just work so you can keep on your muso-hat and be
creative.
When it comes to annoying distractions, having to re-setup and
reconnnect things all the time is annoying. Creating and maintaining a
standard "studio" setup of connections across multiple programs and even
my very simple music set up seems to be tough.
--
David
gnome(a)hawaii.rr.com
authenticity, honesty, community
11:20 a.m.
Am 09.02.2011 11:50, schrieb Philipp Überbacher:
Excerpts from david's message of 2011-02-09
11:28:10 +0100:
I do so quite similar using qarecord (timemachine did not worked
perfectly enough for me for such a simple task...)
Whenever I just want to record a stream from a synth I am testing or a
short idea -- whenever I do not plan to use the powers of Ardour as a suite.
But the thread starter asked for solutions to manipulate and edit audio
and for that Ardour and SND are the gold-standard in Linux the latter
not the most obvious choice for beginners so I recommend Ardour.
Folderol wrote:
For recording I'm also a fan of timemachine because of its simplicity.
You push the big button or press space and it records. On Tue, 08 Feb 2011 06:33:03 -1000
david<gnome(a)hawaii.rr.com> wrote:
I'm starting to record directly into Rosegarden more now.
Sometimes I just start up jack_capture, then clean up and such in Audacity. Hartmut Noack wrote:
> Am 08.02.2011 09:15, schrieb david:
>> Hartmut Noack wrote:
>>> Am 08.02.2011 08:35, schrieb david:
>>>> Robin Gareus wrote:
>>>>> Hi Mike,
>>>>>
>>>>> On 02/07/2011 04:40 PM, Mike Cookson wrote:
>>>>>> For non-realtime (including non-linear, like montage) processing
you
>>>>>> need only plugins (ladspa, lv2, vamp) and some editor like
Audacity,
>>>>>> mhWaveEdit or something other.
>>>>>>
>>>>>> For realtime (also called
>>>>>> non-destructive editing... hm, probably, they are right :) you
need
>>>>>> set of various software, that could be used at one time and be
>>>>>> connected each to other).
>>>>> real-time effects processing and non-destructive editing often go
hand
>>>>> in hand, but note that
>>>>>
>>>>> "non-destructive" means that the original [audio] data will
never be
>>>>> modified. Any edit/effect/modifications are saved as new files (or
>>>>> remebered as application-settings operating on the original data).
>>>>>
>>>>> audio-editors (rezound, audacity, sweep, etc) are usually
destructive:
>>>>> load file, apply effect, save file -> original file is gone.
>>>> Audacity is import audio file, apply effect, save project (optional),
>>>> export in chosen format. It never replaces the original file.
>>> So there is a major dfference between audiofiles, you have imported
>>> and audiofiles, you have recorded with audacity -- correct?
>> No, I never have. I usually use JACK, and have never been able to make
>> Audacity work with JACK. I've only done the following things with
>> Audacity:
>>
>> 1. Import 16 tracks of 32-bit WAV files (recorded on another machine
>> from my church band's Firewire interface using some Windows software)
>> and do basic mixing.
>>
>> 2. Trim and cleanup voice audio recordings made on my PDA.
>>
>> 3. Trim and convert wave files recorded using jack_capture.
>>
>> 4. Pitch shift prerecorded MP3s if needed for band members who play
>> solely by ear (if the recording's in Eb and we're playing it in D,
>> they're lost).
>>
>> If you have the time and brains needed to learn Ardour, go for it!
>>
> I think it is a myth, that Ardour is too complicated to learn for a
> beginner.
Perhaps it's a myth for others. I responded only from my own experience.
There are plenty of folk here who use Ardour and do wonderful things
with it. Advanced features? I couldn't tell an advanced feature from a
basic feature. I couldn't even figure out to simply record anything with
it, and Ardour's "automagic" setup didn't seem to include that
connection.
No insult to Ardour, I'm no audio techno whiz. I figured I'd wait til
Ardour 3 is released and I have the time to learn it.
I had exactly the same experience. Twice over a two year period!
These days I either record with timemachine, then process later in audacity, or
record directly into Rosegarden, where I usually have associated MIDI tracks. You want another
take, press space twice. A simple peak meter is built in, the 'record
the past' feature is nice but I don't really need it. Most importantly
it simply works, jack_capture gives me problems with jack connections
and jack_capture_gui2 has some benefits but seems to have some problems
as well. If you use a vastly more complex program such as ardour or
rosegarden the potential for problems is much larger as well. I'm not
only talking about fatal errors but about all those tiny annoyances as
well. If there are fatal problems you use another program until those
are resolved, but those tiny annoyances are what distract you, ruin your
creativity and productivity. Most of us are the musicians and recording
engineers at the same time, so the process of recording must not be
distracting, it has to just work so you can keep on your muso-hat and be
creative.
_______________________________________________
Linux-audio-user mailing list
Linux-audio-user(a)lists.linuxaudio.org
http://lists.linuxaudio.org/listinfo/linux-audio-user
12:21 p.m.
On Wednesday 09 February 2011 11:50:16 Philipp Überbacher wrote:
For recording I'm also a fan of timemachine
because of its simplicity.
You push the big button or press space and it records. You want another
take, press space twice. A simple peak meter is built in, the 'record
the past' feature is nice but I don't really need it.
Recording the past is nice when you are the foh-tech and can only start the
recording once the song is on its way. Setting a buffer of like 60secs gives
you enough time to start up and check the live-sound and only after that
"start" the recording. :-)
Have fun,
Arnold
3:43 p.m.
Excerpts from Arnold Krille's message of 2011-02-09 13:21:45 +0100:
On Wednesday 09 February 2011 11:50:16 Philipp
Überbacher wrote:
Does this really help? You could just let your recording roll all the
way. The way the timemachine feature works right now you'll likely have
weird issues when you record song after song since the current
implementation doesn't seem to clear the ringbuffer or whatever it uses.
Try recording a couple of seconds, stop recording and start the next
recording within the pre-recording buffer length (10 sec. default) to
see what I mean. It can be a bit confusing and requires editing, or
worse.
For recording I'm also a fan of timemachine
because of its simplicity.
You push the big button or press space and it records. You want another
take, press space twice. A simple peak meter is built in, the 'record
the past' feature is nice but I don't really need it.
Recording the past is nice when you are the foh-tech and can only start the
recording once the song is on its way. Setting a buffer of like 60secs gives
you enough time to start up and check the live-sound and only after that
"start" the recording. :-)
Have fun,
Arnold
7:17 p.m.
On Wednesday 09 February 2011 16:43:28 Philipp Überbacher wrote:
Excerpts from Arnold Krille's message of
2011-02-09 13:21:45 +0100:
Editing is required in either way. Normally my recordings are not concerts
where one songs follows the next (this I record in one single take but also
with timemachine), but services and the like where its 3min music followed by
5min talking followed by 6min music followed by 30min talking and so on. And
when I am only interested in the music, I only record the music.
Last time I used it and actually did a stop-restart within the timemachine-
ringuffer, it seemed to just have a smaller pre-roll, basically the stuff since
the last stop. But my memory is failing there...
Have fun,
Arnold
On Wednesday 09 February 2011 11:50:16 Philipp
Überbacher wrote:
Does this really help? You could just let your recording roll all the
way. The way the timemachine feature works right now you'll likely have
weird issues when you record song after song since the current
implementation doesn't seem to clear the ringbuffer or whatever it uses.
Try recording a couple of seconds, stop recording and start the next
recording within the pre-recording buffer length (10 sec. default) to
see what I mean. It can be a bit confusing and requires editing, or
worse. For recording I'm also a fan of timemachine
because of its simplicity.
You push the big button or press space and it records. You want another
take, press space twice. A simple peak meter is built in, the 'record
the past' feature is nice but I don't really need it.
Recording the past is nice when you are the foh-tech and can only start
the recording once the song is on its way. Setting a buffer of like
60secs gives you enough time to start up and check the live-sound and
only after that "start" the recording. :-)
8 Feb
8 Feb
10:45 p.m.
Am 08.02.2011 17:33, schrieb david:
Hartmut Noack wrote:
New Project -> add stereo-track -> arm it for recording -> klick the red
record-button in the upper left play-control.
Here is another little trick that may be helpfull for beginners:
SHIFT+E pops up the Mixer-channel of the active track. Klick around in
it to solve most of the problems and answer most of the questions that
may arise for a beginner in Ardour. Such as from top to bottom:
- where is my sound-card input signal going to?
- where are these plug-ins and how and where can I insert them?
Do not get me wrong: Ardour IS in fact not the most simple of the
beasts. But the comparably little effort to get it under controll brings
a lot of benefits that are must haves for serious audio-work. The most
prominent are realtime-mix/FX and automation.
What good is a simple interface if it does not give you the possibility
to set the parameters of an EQ *while* you hear, what you do.
As I switched from Windows 98 and Samplitude to Linux 2.4 I tried all
audio-editors available for Suse 7.3 and the only one that I found to be
worthwhile was SND. Shortly after that I heared word from Ardour that
was a legend that time. Only installable by wizards that know how to
compile komplex software from source and only runable with the magickal
jack, who needed an even more magical rt-kernel.
I worked for months until I had this Ardour-thing running and from that
very point I knew, that I would never again install a dual-boot system.
Am 08.02.2011 09:15, schrieb david:
Perhaps it's a myth for others. I responded only from my own experience.
There are plenty of folk here who use Ardour and do wonderful things
with it. Advanced features? I couldn't tell an advanced feature from a
basic feature. I couldn't even figure out to simply record anything with
it, and Ardour's "automagic" setup didn't seem to include that
connection. Hartmut Noack wrote:
I think it is a myth, that Ardour is too complicated to learn for a
beginner. Am 08.02.2011 08:35, schrieb david:
> Robin Gareus wrote:
>> Hi Mike,
>>
>> On 02/07/2011 04:40 PM, Mike Cookson wrote:
>>> For non-realtime (including non-linear, like montage) processing you
>>> need only plugins (ladspa, lv2, vamp) and some editor like Audacity,
>>> mhWaveEdit or something other.
>>>
>>> For realtime (also called
>>> non-destructive editing... hm, probably, they are right :) you need
>>> set of various software, that could be used at one time and be
>>> connected each to other).
>>
>> real-time effects processing and non-destructive editing often go
>> hand
>> in hand, but note that
>>
>> "non-destructive" means that the original [audio] data will never be
>> modified. Any edit/effect/modifications are saved as new files (or
>> remebered as application-settings operating on the original data).
>>
>> audio-editors (rezound, audacity, sweep, etc) are usually
>> destructive:
>> load file, apply effect, save file -> original file is gone.
>
> Audacity is import audio file, apply effect, save project (optional),
> export in chosen format. It never replaces the original file.
So there is a major dfference between audiofiles, you have imported
and audiofiles, you have recorded with audacity -- correct?
No, I never have. I usually use JACK, and have never been able to make
Audacity work with JACK. I've only done the following things with
Audacity:
1. Import 16 tracks of 32-bit WAV files (recorded on another machine
from my church band's Firewire interface using some Windows software)
and do basic mixing.
2. Trim and cleanup voice audio recordings made on my PDA.
3. Trim and convert wave files recorded using jack_capture.
4. Pitch shift prerecorded MP3s if needed for band members who play
solely by ear (if the recording's in Eb and we're playing it in D,
they're lost).
If you have the time and brains needed to learn Ardour, go for it!
No insult to Ardour, I'm no audio techno whiz. I
figured I'd wait til
Ardour 3 is released and I have the time to learn it.
9 Feb
9 Feb
10:32 a.m.
Hartmut Noack wrote:
Am 08.02.2011 17:33, schrieb david:
It helps when you already know what to do.
Hartmut Noack wrote:
New Project -> add stereo-track -> arm it for recording -> klick the red
record-button in the upper left play-control. Am 08.02.2011 09:15, schrieb david:
Perhaps it's a myth for others. I responded only from my own experience.
There are plenty of folk here who use Ardour and do wonderful things
with it. Advanced features? I couldn't tell an advanced feature from a
basic feature. I couldn't even figure out to simply record anything with
it, and Ardour's "automagic" setup didn't seem to include that
connection. Hartmut Noack wrote:
> Am 08.02.2011 08:35, schrieb david:
>> Robin Gareus wrote:
>>> Hi Mike,
>>>
>>> On 02/07/2011 04:40 PM, Mike Cookson wrote:
>>>> For non-realtime (including non-linear, like montage) processing
>>>> you
>>>> need only plugins (ladspa, lv2, vamp) and some editor like
>>>> Audacity,
>>>> mhWaveEdit or something other.
>>>>
>>>> For realtime (also called
>>>> non-destructive editing... hm, probably, they are right :) you need
>>>> set of various software, that could be used at one time and be
>>>> connected each to other).
>>>
>>> real-time effects processing and non-destructive editing often go
>>> hand
>>> in hand, but note that
>>>
>>> "non-destructive" means that the original [audio] data will never
be
>>> modified. Any edit/effect/modifications are saved as new files (or
>>> remebered as application-settings operating on the original data).
>>>
>>> audio-editors (rezound, audacity, sweep, etc) are usually
>>> destructive:
>>> load file, apply effect, save file -> original file is gone.
>>
>> Audacity is import audio file, apply effect, save project (optional),
>> export in chosen format. It never replaces the original file.
>
> So there is a major dfference between audiofiles, you have imported
> and audiofiles, you have recorded with audacity -- correct?
No, I never have. I usually use JACK, and have never been able to make
Audacity work with JACK. I've only done the following things with
Audacity:
1. Import 16 tracks of 32-bit WAV files (recorded on another machine
from my church band's Firewire interface using some Windows software)
and do basic mixing.
2. Trim and cleanup voice audio recordings made on my PDA.
3. Trim and convert wave files recorded using jack_capture.
4. Pitch shift prerecorded MP3s if needed for band members who play
solely by ear (if the recording's in Eb and we're playing it in D,
they're lost).
If you have the time and brains needed to learn Ardour, go for it!
I think it is a myth, that Ardour is too complicated to learn for a
beginner. Here is another little trick that may be helpfull for
beginners:
SHIFT+E pops up the Mixer-channel of the active track. Klick around in
it to solve most of the problems and answer most of the questions that
may arise for a beginner in Ardour. Such as from top to bottom:
- where is my sound-card input signal going to?
- where are these plug-ins and how and where can I insert them?
Thanks, will keep that in mind when I tackle Ardour 3.
Do not get me wrong: Ardour IS in fact not the most
simple of the
beasts. But the comparably little effort to get it under controll brings
a lot of benefits that are must haves for serious audio-work. The most
prominent are realtime-mix/FX and automation.
What good is a simple interface if it does not give you the possibility
to set the parameters of an EQ *while* you hear, what you do.
As I switched from Windows 98 and Samplitude to Linux 2.4 I tried all
audio-editors available for Suse 7.3 and the only one that I found to be
worthwhile was SND. Shortly after that I heared word from Ardour that
was a legend that time. Only installable by wizards that know how to
compile komplex software from source and only runable with the magickal
jack, who needed an even more magical rt-kernel.
I worked for months until I had this Ardour-thing running and from that
very point I knew, that I would never again install a dual-boot system.
Won't argue with any of that.
--
David
gnome(a)hawaii.rr.com
authenticity, honesty, community
8 Feb
8 Feb
7:33 p.m.
On 2/8/2011 5:50 AM, Hartmut Noack wrote:
Am 08.02.2011 09:15, schrieb david:
Thank you for all that!! Awesome!
I'll start playing tonight. I might try to grab some sounds from my
friends PC video fame collection and start playing around.
Bearcat
Hartmut Noack wrote:
I think it is a myth, that Ardour is too complicated to learn for a
beginner.
The Interface may look somewhat intimidating at first glance but with
some basic rules in mind it is not too hard to be handeled.
1.) learn to ignore, what you cannot understand at first sight. about
99% of the default-settings are sane and usefull and if you need to
change something, you will find out how as soon as you need it.
2.) understand the interaction between the tracks in the editor and
the mixer. Tracks are sources, the mixer is manipulation both are
connected by a signalflow that is setup by Ardour automagically. In
the beginning you will not need to change anything in the routing
later you will seldom need to change some details.
With this in mind you can start using Ardour like this:
1.) start Jack
2.) start Ardour, type a name for your project and click OK
3.) rightklick under the master-track and choose "import audio" or
drag/drop WAV-files from a GTK-Filemanager like Nautilus.
4.) Hit Space an hear your files play.
Next step would be to insert some plugins in your tracks and start
fiddeling with them.
All the advanced stuff like set up recording from internal
jack-clients, automatisation of faders and plugins, special
mouse-modes or usage of the things in the right-click menues can be
learned step by step after that.
I really believe that an absolute beginner who does not allow
him/herself to be intimidated by all the advanced features can begin
to use Ardour with fun and good results within 1 h after starting such
a program for the first time.
best regards
HZN Am 08.02.2011 08:35, schrieb david:
No, I never have. I usually use JACK, and have never been able to make
Audacity work with JACK. I've only done the following things with
Audacity:
1. Import 16 tracks of 32-bit WAV files (recorded on another machine
from my church band's Firewire interface using some Windows software)
and do basic mixing.
2. Trim and cleanup voice audio recordings made on my PDA.
3. Trim and convert wave files recorded using jack_capture.
4. Pitch shift prerecorded MP3s if needed for band members who play
solely by ear (if the recording's in Eb and we're playing it in D,
they're lost).
If you have the time and brains needed to learn Ardour, go for it!
Robin Gareus wrote:
> Hi Mike,
>
> On 02/07/2011 04:40 PM, Mike Cookson wrote:
>> For non-realtime (including non-linear, like montage) processing you
>> need only plugins (ladspa, lv2, vamp) and some editor like Audacity,
>> mhWaveEdit or something other.
>>
>> For realtime (also called
>> non-destructive editing... hm, probably, they are right :) you need
>> set of various software, that could be used at one time and be
>> connected each to other).
>
> real-time effects processing and non-destructive editing often go
> hand
> in hand, but note that
>
> "non-destructive" means that the original [audio] data will never be
> modified. Any edit/effect/modifications are saved as new files (or
> remebered as application-settings operating on the original data).
>
> audio-editors (rezound, audacity, sweep, etc) are usually
> destructive:
> load file, apply effect, save file -> original file is gone.
Audacity is import audio file, apply effect, save project (optional),
export in chosen format. It never replaces the original file.
So there is a major dfference between audiofiles, you have imported
and audiofiles, you have recorded with audacity -- correct?
10 Feb
10 Feb
11:27 a.m.
New subject: [LAU] Linux programs for creatiing/manipulating sound effects
Mon, 07 Feb 2011 22:15:27 -1000 письмо от david <gnome(a)hawaii.rr.com>om>:
No, I never have. I usually use JACK, and have never been able to make
Audacity work with JACK. I've only done the following things with
Audacity:
1. Import 16 tracks of 32-bit WAV files (recorded on another machine
from my church band's Firewire interface using some Windows software)
and do basic mixing.
2. Trim and cleanup voice audio recordings made on my PDA.
3. Trim and convert wave files recorded using jack_capture.
4. Pitch shift prerecorded MP3s if needed for band members who play
solely by ear (if the recording's in Eb and we're playing it in D,
they're lost).
If you have the time and brains needed to learn Ardour, go for it!
--
David
gnome(a)hawaii.rr.com
authenticity, honesty, community
_______________________________________________
Linux-audio-user mailing list
Linux-audio-user(a)lists.linuxaudio.org
http://lists.linuxaudio.org/listinfo/linux-audio-user
It is possible even, to make
Audacity a part of some processing chain. The only problem, that you get, is
synchronisation. Some example:
1. First of all - you need to enable devices toolbar, to switch recording and playback
source. Not, however, when you finished work, you should switch them back to what you have
always (most possible - "System", which is default in most cases), otherwise you
will need to get the same ports again before to start Audacity again, otherwise it will
fail to start.
2. Run jack-rack. Select it in Audacity both for input and output. Load some track.
3. Try to record something (i think, you already should understand, what is expected,
right?). You should get new track with data from first track, processed by JR.
More real example. When i tried to add one MP3 file to audacity by dragging, sometimes it
was loaded incorrectly - some parts was lost or modified in strange (for me) way.
Solution, i found, is to record it from audacious.
Anyway, in all these cases - recording in audacity should be started before to run
playback in other staff (i.e., before to start playback in audacious).
11:14 a.m.
New subject: [LAU] Linux programs for creatiing/manipulating sound effects
Tue, 08 Feb 2011 08:50:47 +0100 письмо от Hartmut Noack <zettberlin(a)linuxuse.de>de>:
Am 08.02.2011 08:35, schrieb david:
So there is a major dfference between audiofiles, you have imported and
audiofiles, you have recorded with audacity -- correct?
When they say about
desctructive / non-destructive editing, they usually mean way of audio data editing, not
what happens with files, where processed data are stored.
About differences between loaded and saved data - with Audacity you can export to the same
file, as loaded from. And in mhWaveEdit (as much as all other) - "Save As"
action is still there.
Robin Gareus wrote:
> Hi Mike,
>
> On 02/07/2011 04:40 PM, Mike Cookson wrote:
>> For non-realtime (including non-linear, like montage) processing
you
>> need only plugins (ladspa, lv2, vamp) and
some editor like
Audacity,
>> mhWaveEdit or something other.
>>
>> For realtime (also called
>> non-destructive editing... hm, probably, they are right :) you
need
>> set of various software, that could be
used at one time and be
>> connected each to other).
>
> real-time effects processing and non-destructive editing often go
hand
> in hand, but note that
>
> "non-destructive" means that the original [audio] data will
never
be
> modified. Any edit/effect/modifications are
saved as new files (or
> remebered as application-settings operating on the original data).
>
> audio-editors (rezound, audacity, sweep, etc) are usually
destructive:
load
file, apply effect, save file -> original file is gone.
Audacity is import audio file, apply effect, save project (optional),
export in chosen format. It never replaces the original file.
It IS destructive in that it applies the effect to its imported copy of
the original audio. But that doesn't effect the original file unless
you
chose to export to the same location in the same
format with the same
filename.
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61 comments
22 participants
participants (22)
-
Alexandre Prokoudine -
Arnold Krille -
Bearcat M. Şandor -
Bernardo Barros -
david -
Edgar Aichinger -
Folderol -
Hartmut Noack -
Jaromír Mikeš -
Jeremy Jongepier -
Jeremy Jongepier -
Jörn Nettingsmeier -
Leigh Dyer -
Malte Steiner -
Mike Cookson -
Paul Davis -
Peter Plessas -
Philipp Überbacher -
Renato Budinich -
Robin Gareus -
Robin Gareus -
Rustom Mody