First off, Happy New Year to all! :)
Can anyone recommend a quick way of seeing the differences highlighted
visually between two channels in a stereo sample...sort of like a
graphical 'diff' for waveforms?
I think I may have some Akai MPC2000 samples that somehow were saved by
the Akai as mono samples even though they actually contain two channels
and are really stereo:
Using 'sndfile-info' on them reports them as mono, but it's just looking
at the 22nd byte in the file, which should contain 0x00 or 0x01 to say
if it is mono or stereo. The file contains 0x00 in that location, so
sndfile-info believes that it's mono because the file header says so.
However, if I flip the bit in a hex editor and resave the file, now I
can load the file in a sample editor and it seems to have two channels
which play at double speed. I'm still not positive I'm hearing stereo
(if I am, it's very subtle), but I'm definitely seeing two channels
visually, and I know this sample was sampled on the Akai in stereo.
Any way to do a graphical comparison that will highlight even small
channel waveform differences?
--
+ Brent A. Busby + "We've all heard that a million monkeys
+ Sr. UNIX Systems Admin + banging on a million typewriters will
+ University of Chicago + eventually reproduce the entire works of
+ James Franck Institute + Shakespeare. Now, thanks to the Internet,
+ Materials Research Ctr + we know this is not true." -Robert Wilensky
On 01/09/2013 02:27 AM, Burkhard Woelfel wrote:
> What size is that buffer?
Not much. In the Tascam DR-40 it is two seconds. In the Zoom H4N it is
two seconds unless recording 96KHz 4 channel, then it's only one second.
wes
Hi all,
I'm not quite sure how to interpret this, but it sounds like
linuxaudio.org will be offline for a short time (failover period) coming
Saturday January 12th 8am EST.
-------- Original Message --------
Just wanted to make sure you were aware of the upcoming maintenance this
Saturday.
Maintenance for the central storage system known as minnow.cc.vt.edu is
schedule for Saturday, January 12 starting at 8am. Several hardware
components in the filer head needs to be replaced. Since this is a
cluster storage system, the maintenance will be performed in failover
mode, so minnow services will continue to be available. During the
failover, services will pause until the failover is complete.
Hi William,
Building the latest git pull for 0.14.97 on a completely updated Arch 64
system:
[dlphilp@bigblack phasex]$ aclocal
configure.ac:695: error: 'AM_CONFIG_HEADER': this macro is obsolete.
You should use the 'AC_CONFIG_HEADERS' macro instead.
/usr/share/aclocal-1.13/obsolete-err.m4:12: AM_CONFIG_HEADER is expanded
from...
configure.ac:695: the top level
autom4te: /usr/bin/m4 failed with exit status: 1
aclocal: error: echo failed with exit status: 1
I followed the advice, updated the macro declaration, and aclocal ran to
its end, after which the rest of the autotools ran without complaint.
I used the following build configuration:
./configure --enable-cpu-power --enable-arch=native --prefix=/usr/local
No reported errors during the build config, Phasex compiled without
problems.
CPU is an AMD64 3200+, single-core, with 4G RAM, GCC 4.7.2.
From uname :
Linux 3.7.1-2-ARCH #1 SMP PREEMPT x86_64 GNU/Linux
Alas, when I start Phasex it segfaults immediately:
[dlphilp@bigblack ~]$ phasex
Not sending deprecated LASH_Client_Name event
LASH client initialized. (LASH_Client_Name='phasex').
Main: LASH client started.
(phasex:1239): Gtk-WARNING **: Unable to locate theme engine in
module_path: "nodoka",
(phasex:1239): Gtk-WARNING **: Unable to locate theme engine in
module_path: "nodoka",
Segmentation fault (core dumped)
Bummer. Also, the splash indicates 0.14.96 but the git pull specified
0.14.97. There is no previous version of Phasex on my system.
Suggestions are welcome. :)
Best,
dp
New in CAPS 0.9.4:
* selectable oversampling ratios for AmpVTS (2x,4x,8x)
* selectable sounds for Click (box, stick, beep; the second being
very close to the sound of the unit in 0.4.x)
* further smoothening of ChorusII modulation
* selectable oversampling ratios for Compress (2x,4x)
Also, a serious issue with Compress has been fixed: gain had been
applied before saturation, greatly reducing the effectiveness of the
unit. It's squashing smoothly now :)
The 10-band Eq has been fixed and now has a reasonably flat frequency
response.
http://quitte.de/dsp/caps.htmlhttp://quitte.de/dsp/caps.html#AmpVTShttp://quitte.de/dsp/caps.html#Compresshttp://quitte.de/dsp/caps.html#Clickhttp://quitte.de/dsp/caps.html#Eq
check this out:
http://www.instructables.com/id/3D-Printed-Record/
the sound quality is quite bad, but it turns out the author neglected to
apply the RIAA curve to the master before rendering the groove, so it
could be made to sound much better. but that's not really the point :)
reminds me of a band i heard of in the 80s (forgot the name) who
released a single which, if recorded to some data cassette format, would
yield a computer game...
best,
jörn
--
Jörn Nettingsmeier
Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487
Meister für Veranstaltungstechnik (Bühne/Studio)
Tonmeister VDT
http://stackingdwarves.net
As a newcomer to the "Pro Audio" industry and a long time *nix user
(somewhat mitigating this outburst here) I must say I'm not impressed.
I've spent thousands recently and found suppliers either just don't
'know' the products they sell, or purposely misrepresent them. In one
instance it took nearly a month to obtain a refund!
For RME products; the 'bundled' word seems to be the one to watch out for.
If it's a Multiface II bundled with a card, then double check. Same for
UCX bundled with remote controller.
I'd be getting on with building a new rack right now, but I've just
received half a pair of brackets from a UK supplier and there's
hardly anything more useless than half a pair of rack brackets. Grr.
Why must I have so many microphone inputs? I'm a solo poet/musician
with one mic. and enough places to plug it in to sink a battleship.
Synth manufacturers: Why no ADAT out?
But what really annoys me are the presumptions I had about my best
synthesiser/drum machine's MIDI capabilities. You'd probably similarly
presume an state-of-the-art instrument designed by Dave Smith and Roger
Linn would be spot on in the MIDI dept, but it isn't. Six stereo 'voice'
outs, but FIXED PITCH! To be fair(er) the OS is still in development...
I don't resent Dave Smith getting a Technical Grammy though. Not at all.
I've been having fun with MIDI for an awful long time.
http://www.factmag.com/2012/12/13/dave-smith-father-of-midi-to-receive-spec…
--
John.
(Definitely feeling better for having shared my frustration.)
MusE 2.1 Jan 06, 2013
Happy new year everybody and happy new MusE release!
We've tried hard to break tradition and release somewhat quicker than
what we usually been able to.
So without further ado, here we are with a shiny 2.1 release.
Assortment of topics worked on this time around are:
* Native VST support, first drop: effect rack isn't supported yet and
some synths are misbehaving a bit but it should be usable.
* New type of drum track with easier instrument handling
* Improved Wave Editor.
* Improved midi import/export
* Longtime wierdness in MusE, song-type is finally removed
* Improved: Deicsonze soft synthesizer: Fixed crash, bugs, added ladspa plugins
* Added: Request from forums: Auto-start Jack upon MusE startup.
Command-line disable switch -J added
* Added aftertouch controller handling
* Piano KB and drum list show coloured dots when per-pitch controllers
exist or have data.
* Improved: Instrument Editor (controller tab): Redesigned. Fixed MANY bugs
* Improved: Midi controller graph 'Ctrl' popup menus now unified (cascading).
* Improved: built-in templates and removed lots of unneeded definitions
* Improved: Velocity graphs. Icon for showing per-note or all velocities
* Improved: Piano KB has current selected note (yellow). For
velocity/polyaftertouch/other per-note ctrls
* Added 'speaker' icon to drum edit. And drum list and piano keyboard
now obey the 'speaker' icon
* Improved: Multi-port (aka multi-channel) midi import and export
* Added Gain knob on channel strip for audio tracks
* Changed WaveTrack and AudioInput to create mono tracks instead of stereo
* Added: Informative text on Undo/Redo buttons/menu text/tooltips
("Undo AddTrack" etc).
* Fixed CR:3567893 Play Event on note creation
* Bug 3555569, 3555572: New informative ERROR, WARNING and HINT
messages in cmake build script.
* Bug 3555581: No synths listed in Edit menu: main.cpp: Move
initMidiSynth() ahead of MusE.
* Feature request 3565102: FLAC audio file import.
* Fix for forum topic "Ordering of simultaneous events".
* Summer sleep is over! plugins can now be grouped
* Finished the Sysex Editor in the Instrument Editor.
* increased zoom range in arranger, reversed zoom in wave editor
* Internal fluidsynth and simpledrums search project dir for missing
sounds to ease portability, and if no sounds are found a load file
dialog is shown
For more information and additional changes see the changelog:
http://lmuse.svn.sourceforge.net/viewvc/lmuse/trunk/muse2/ChangeLog?revisio…
Find the download at:
https://sourceforge.net/projects/lmuse/files/
MusE on!
The MusE Team
>From earlier talk about sample rates, it appears that for recording audio,
48k is better than 44.1. Having decided that, what other issues cross my
path if I default my audio rate to 48k? All of my softsynths seem to
work... I do wonder about samplers though. Are the samples mostly in 44.1k
if they come on a CD? Would that make them off-key when used? Would they
get rate-changed on the fly? Would that use more CPU? (questions,
questions, questions)
It does not seem to affect my desktop, flash, ogg, mp3, ac3 etc. all
playback fine. Pulse seems to be doing more work though at least when
playing a CD. I'm pushing it pretty hard though, I have it bridged to
jackdbus with -p64 so Pulse has to keep up.
I'm just wondering why Ubuntu (which presumably means Debian too),
qjackctl, Pulse and lots of other audio apps all seem to default to 44.1k.
Speaking as one of the devs for the Ubuntu Studio distro, I'm wondering
how much trouble I'm going to get for asking for 48k default sample rate
in audio.
--
Len Ovens
www.OvenWerks.net
Hi All,
A slightly-belated Happy New Year to you all, and notification of a
new release of Praxis LIVE (build:121231).
Praxis LIVE is an open-source, graphical environment for rapid
development of intermedia performance tools, projections and
interactive spaces. This release brings a range of performance
improvements to the video pipeline, some useful UI improvements, and
support for up to 16 channels of audio IO. There is also some early
support for physical computing using the excellent TinkerForge
(http://www.tinkerforge.com/).
Praxis LIVE is now available as a .deb pacakge or as a ZIP (for
un-installed usage).
Website (with downloads and manual) - http://code.google.com/p/praxis
Release notes - http://code.google.com/p/praxis/wiki/ReleaseNotes
Blog post (with TinkerForge vid) -
http://praxisintermedia.wordpress.com/2013/01/07/new-year-new-praxis-live/
Thanks and best wishes,
Neil
--
Neil C Smith
Artist : Technologist : Adviser
http://neilcsmith.net
Praxis LIVE - open-source, graphical environment for rapid development
of intermedia performance tools, projections and interactive spaces -
http://code.google.com/p/praxis
OpenEye - specialist web solutions for the cultural, education,
charitable and local government sectors - http://openeye.info