Sweep 0.5.9 Development Release
-------------------------------
Sweep is an audio editor and live playback tool for GNU/Linux, BSD and
compatible systems. It supports many music and voice formats including
WAV, AIFF, Ogg Vorbis, Speex and MP3, with multichannel editing and
LADSPA effects plugins. Inside lives a pesky little virtual stylus called
Scrubby who enjoys mixing around in your files.
This development release is available as a source tarball at:
http://prdownloads.sourceforge.net/sweep/sweep-0.5.9.tar.gz?download
Sweep now supports Speex, a special purpose speech codec designed for
efficient Voice over IP (VoIP) and file-based compression. Speex is free,
open and unpatented; more information is available at http://www.speex.org/.
This release also includes improved handling of the main volume and pitch
controls, contributed by Zenaan Harkness.
Screenshots:
http://www.metadecks.org/software/sweep/screenshots/
Audio demos:
http://www.metadecks.org/software/sweep/demos.html
Sweep is designed to be intuitive and to give you full control. It includes
almost everything you would expect in a sample editor, and then some:
* precise, vinyl like scrubbing
* looped, reverse, and pitch-controlled playback
* playback mixing of unlimited independent tracks
* looped and reverse recording
* internationalisation
* multichannel and 32 bit floating point PCM file support
* support for Ogg Vorbis, MP3 and Speex compressed audio files
* LADSPA 1.1 effects support
* multiple views, discontinuous selections
* easy keybindings, mouse wheel zooming
* unlimited undo/redo with fully revertible edit history
* multithreaded background processing
* shaded peak/mean waveform rendering, multiple colour schemes
Sweep is Free Software, available under the GNU General Public License.
More information is available at:
http://www.metadecks.org/software/sweep/
Thanks to Pixar Animation Studios and CSIRO Australia for supporting the
development of this project.
enjoy :)
Conrad.
Thanks Luis, I'll give it a try. And I'll check out the csound list too.
Stuart
> > Can anybody out there tell me what I (or csound :) did wrong?
>
> well, i won't try to tell you what might be wrong with your
> orchestra, but
> if you are interested in time-stretching using csound, try my
> instrument
> below. with it you can do time-stretch and/or pitch transpose
> independently.
[snip]
>
> good luck,
>
> lj
>
> (ps. aren't you interested in joining the csound mailing-list?)
Hi,
I *finally* got around to trying the csound method of changing the speed of
an audio file without changing the pitch, but it didn't work for me :(
I used the following "orc" file:
> sr = 44100
> kr = 4410
> ksmps = 10
> nchnls = 1
>
> instr 1
> kfreqscale = 1
> ispecwp = 0
> ktime line 1, p3, 0
> apvl pvoc ktime, kfreqscale, "file.pvc", ispecwp
> out apvl
> endin
and then the "sco" file:
> i1 0 12
> e
with a 6 second sample, so this should double the audio to 12 seconds. I did
the PV analysis like this:
> csound -U pvanal -n 1024 -w 4 file.wav file.pvc
and then ran csound via:
> csound -W -R -d file.orc file.sco -o slow.wav
and then I waited for a little while :)
When csound was done, the ourput file was, well, interesting...
Basically the input file was 6 seconds of single-note guitar playing, and
the output was a weird (though not unpleasant :) whooshing-sweeping-phasing
sound for 2 seconds, then a faint LFO-type sound to the end of the file. No
sign of the original clip, slowed or otherwise.
Can anybody out there tell me what I (or csound :) did wrong?
Thanks in advance,
Stuart
hi all
thanks for your help with compiling alsa, i've now got both usb-audio
and usb-midi compiled...
but i'm having problems getting midi running. the results of modprobe
snd-usb-midi is a load of unresolved symbols... (as listed below)...
i will also need to write a modules.conf which supports both my m-audio
quattro *and* the Evolution UC16 - which i am somewhat stumped by...
any help would be greatly appreciated,
thanks
m~
[root@hamish miriam]# modprobe snd-usb-midi
/lib/modules/2.4.18-6mdk/kernel/sound/acore/seq/snd-seq-virmidi.o:
unresolved sy
mbol snd_rawmidi_new_Rbc1a629f
/lib/modules/2.4.18-6mdk/kernel/sound/acore/seq/snd-seq-virmidi.o:
unresolved sy
mbol snd_rawmidi_transmit_peek_R2f3ef431
/lib/modules/2.4.18-6mdk/kernel/sound/acore/seq/snd-seq-virmidi.o:
unresolved sy
mbol snd_rawmidi_receive_R4b303a51
/lib/modules/2.4.18-6mdk/kernel/sound/acore/seq/snd-seq-virmidi.o:
unresolved sy
mbol snd_rawmidi_set_ops_R246c97c8
/lib/modules/2.4.18-6mdk/kernel/sound/acore/seq/snd-seq-virmidi.o:
unresolved sy
mbol snd_rawmidi_transmit_ack_Rb639865e
modprobe: insmod
/lib/modules/2.4.18-6mdk/kernel/sound/acore/seq/snd-seq-virmidi
.o failed
/lib/modules/2.4.18-6mdk/kernel/sound/acore/seq/snd-seq-virmidi.o:
unresolved sy
mbol snd_rawmidi_new_Rbc1a629f
/lib/modules/2.4.18-6mdk/kernel/sound/acore/seq/snd-seq-virmidi.o:
unresolved sy
mbol snd_rawmidi_transmit_peek_R2f3ef431
/lib/modules/2.4.18-6mdk/kernel/sound/acore/seq/snd-seq-virmidi.o:
unresolved sy
mbol snd_rawmidi_receive_R4b303a51
/lib/modules/2.4.18-6mdk/kernel/sound/acore/seq/snd-seq-virmidi.o:
unresolved sy
mbol snd_rawmidi_set_ops_R246c97c8
/lib/modules/2.4.18-6mdk/kernel/sound/acore/seq/snd-seq-virmidi.o:
unresolved sy
mbol snd_rawmidi_transmit_ack_Rb639865e
modprobe: insmod
/lib/modules/2.4.18-6mdk/kernel/sound/acore/seq/snd-seq-virmidi
.o failed
modprobe: insmod snd-usb-midi failed
--
iriXx
www.iriXx.org
copyleft: creativity, technology and freedom?
info(a)copyleftmedia.org.uk
www.copyleftmedia.org.uk
_
( ) ascii ribbon against html email
X
/ \ cat /dev/sda1 > /dev/dsp
*** stopping make sense ***
> What about using midi-thru boxes? they don't add latency to
> the midi signal,
> they just 'amplify' and 'split'.
If you run 14 MIDI devices through one MIDI Port using
MIDI Thru Boxes you get a worse timing compared
to using a multiple Output Interfaces like
the Steinberg Midex 8 oder the Midiman Midisport 8x8.
But I don't know if there is a 8x8 Interface with Linux drivers.
All of them implement a feature to reduce latency and I think
that's a problem for the driver developers because the companies
don't want to show the specifications for that in public.
Regards,
Joachim
PS: I added the Linux-Audio-User group to the cc because this
is linux specific question.
> ----- Original Message -----
> From: "Tjeerd Sietsma" <tsietsma(a)hotmail.com>
> To: <music-dsp(a)shoko.calarts.edu>
> Sent: Tuesday, October 22, 2002 9:03
> Subject: [music-dsp] linux and midi
>
>
> >
> >
> > I need to connect 14 midi devices (each using 1
> programmable midi channel)
> > to a PC runnig linux. Midi thru isn't an option, since the
> latency will be
> > too high. Does anyone has hardware recommendations?
> >
>
> dupswapdrop -- the music-dsp mailing list and website:
> subscription info,
> FAQ, source code archive, list archive, book reviews, dsp links
> http://shoko.calarts.edu/musicdsp/
>
I just searched the archives on this and the suggestions there
didn't help.
I'm trying to convert some mp3 files into wav files. First I
tried having Audacity export a wav file. Then I tried loading the
mp3 into Broadcast 2000 and having it render it to a wav. Finally
I tried the suggestion in the archives of mpg123 -w filename.wav
filename.mp3.
In each case, when I tried to burn a CD from the resulting wav
file, cdrecord reports "inappropriate audio coding in
[filename]." I'm using Eroaster 2.0.10 as a front end to
cdrecord.
Audacity and Broadcast 2000 both load all these wav files and
play them just fine.
Anybody know what's going on here?
Thanks.
Howard Sanner
flagstad(a)mindspring.com
Apparently my subscription to the list got dropped. I guess
that's why things have seemed so quiet lately in Linux audio
land.
I just searched the archives on this and the suggestions there
didn't help.
I'm trying to convert some mp3 files into wav files. First I
tried having Audacity export a wav file. Then I tried loading the
mp3 into Broadcast 2000 and having it render it to a wav. Finally
I tried the suggestion in the archives of mpg123 -w filename.wav
filename.mp3.
In each case, when I tried to burn a CD from the resulting wav
file, cdrecord reports "inappropriate audio coding in
[filename]." I'm using Eroaster 2.0.10 as a front end to
cdrecord.
Audacity and Broadcast 2000 both load all these wav files and
play them just fine.
Anybody know what's going on here?
Thanks.
Howard Sanner
flagstad(a)mindspring.com
Hi all,
If I use mpg123 -o oss it plays fine, but I can only use my first
soundcard. If I use mpg123 -o alsa09 it plays the same volume, but the
sound is distorted. mpg321 has the same problem. Programs like aplay do
not have the problem, so it's not for all alsa software.
what do mpg123 and mpg321 have in common? and what makes them distort when
using the alsa output instead of oss-emulation?
the system is a AthlonXP2000+ with Asus on-board cmipci and a Hoontech
ymfpci card. Both cards have the same problem with mpg123 and mpg321 and
both not with aplay.
anybody an idea?
thanks,
Olivier
For thos who are not following the alsa-devel list and want to know
about this card.
--
Patrick Shirkey - Boost Hardware Ltd.
For the discerning hardware connoisseur
Http://www.boosthardware.comHttp://www.djcj.org - The Linux Audio Users guide
========================================
"Um...symbol_get and symbol_put... They're
kindof like does anyone remember like get_symbol
and put_symbol I think we used to have..."
- Rusty Russell in his talk on the module subsystem
New in FreqTweak 0.4.4:
* Now deactivates processing while switching FFT sizes. This
should prevent JACK from kicking you out here.
* Added ability to reconnect to JACK with a different basename,
or just reconnect period if we are kicked out (I/O tab).
* Control-Alt left mouse draws arbitrary lines between
initial click and current cursor. Also smoothed regular drawing.
* More configure checks to ensure compiling against the float version
of FFTW.
http://freqtweak.sourceforge.net
jlc