Sounds interesting. This is what I do for a living. But would any of
the linux audio developers actually follow it?
-Ben Loftis
> Maybe some kind of audio app interface design proposal is in order:
> linux audio interface design - basic application control keys: LAID
> BACK
>
> Gerard
>
No, I didn't dream it, I wrote it. I may have missed a slider or two but I think
I got most of them. Of course I am talking about the latest CVS not 0.8.0.
Jan
On Wed, 9 Jun 2004 07:30 , Richard Bown <richard.bown(a)ferventsoftware.com> sent:
>On Tuesday 08 June 2004 20:22, Chris Cannam wrote:
>> On Tuesday 08 Jun 2004 7:46 pm, eviltwin69(a)cableone.net wrote:
>> > Right click on any slider in JAMin and it immediately goes to
>> > the default position, whether center or zero.
>>
>> Ah, now I looked for that feature but didn't find it. In Rosegarden
>> you double-click to zero a fader. I didn't think of right-clicking.
>
>Actually no - you right click in RG to center a fader. Double click doesn't
>do anything AFAICT so maybe you dreamt that.
>
IMHO this is much bigger problem in UI design.
In ardour it is shift-click in JAMin it is right click, in Rosegarden it is
double-click.
That the sliders all look different is much less confusing than that the
behaviour is all different.
Maybe some kind of audio app interface design proposal is in order:
linux audio interface design - basic application control keys: LAID BACK
Gerard
On Tue, 2004-06-08 at 14:22, Chris Cannam wrote:
> On Tuesday 08 Jun 2004 7:46 pm, eviltwin69(a)cableone.net wrote:
> > Right click on any slider in JAMin and it immediately goes to
> > the default position, whether center or zero.
>
> Ah, now I looked for that feature but didn't find it. In Rosegarden
> you double-click to zero a fader. I didn't think of right-clicking.
>
--
electronic & acoustic musics-- http://www.xs4all.nl/~gml
Hey,
It seems to be a known issue that you cannot run JACK with the capture
period size lower than 512 with the SBLive ALSA driver. See this
thread:
http://ccrma-mail.stanford.edu/pipermail/planetccrma/2003-December/003764.h…
and this:
http://www.music.columbia.edu/pipermail/linux-audio-user/2003-April/004040.…
The above threads seem to indicate that this is a hardware limitation.
However it seems to me more like a driver issue. Using the kX drivers
(on Windows, http://www.kxproject.com) with the exact same card, an old
SBLive Platinum, Ableton Live is usable with the record and playback
period sizes (set via ASIO driver config) at 64 samples (2.33 ms
latency) with nothing else running, and rock solid at 128 (~5 ms) in the
face of basically anything you throw at it. Of course it crashes, as
it's Windows, using an alpha quality third-party driver, but is quite
usable in a live music setting. 512x2 is not really usable for my
purposes.
Is this assessment correct, and if so, can someone familiar with the
SBLive ALSA driver give me an idea as to how this could be fixed? Could
this be done via ALSA config files maybe?
Lee
Greetings all, a quick note to bring to your attention some exciting
audio summer courses happening this summer in Canada.
This summer is a special one for the annual CCRMA summer workshops,
the workshop series is expanded and is being held in a spectacular new
setting at the Banff Centre for the Arts in the Canadian Rocky
Mountains.
All of the workshops include significant hands-on lab components. The
labs will be done on Planet-CCRMA equipped
linux workstations - a great opportunity to get acquainted with linux
audio tools while learning volumes of useful theory and implementation
details.
New this summer is the Digital Audio Effects workshop taught by
Jonathan Abel and David Berners with special guest Julius Smith. The
course focuses on theory and practice of simulating / implementing a
wide range of classic analog audio effects (including compressors,
reverbs, equalizers ...) in the digital domain. Abel and Berners hail
from Universal Audio and are the driving force behind UA's range of
renowned and widely used digital audio effect plugins.
Detailed descriptions of the courses and registration information is
available here:
http://www.banffcentre.ca/ccrma/
For questions please do not hesitate to contact the faculty of the
courses you're interested in directly, myself, or the banff centre at
arts_info(a)banffcentre.ca (1.800.565.9989 or 403.762.6180).
Best Regards,
scott wilson
__________________________________________________________________
CCRMA@Banff Summer Workshops 2004
__________________________________________________________________
The Banff Centre and Stanford University welcome CCRMA (Centre for
Computer Research in Music and Acoustics) to Banff this summer for six
intensive programs where top educators and researchers from the fields
of music, engineering, and computer science will present a detailed
study of specialized subjects in an awe-inspiring setting.
The CCRMA@Banff Programs Include:
- Physical Interaction Design for Music (July 5 - July 16)
Faculty: Scott Wilson, Michael Gurevich
Guest: Bill Verplank
- Haptic Musical Devices (July 19 - 23)
Faculty: Charles Nichols
Guest: Perry Cook
- Digital Signal Processing I: Spectral & Physical Models (July 26-
August 6)
Faculty: Perry Cook, Xavier Serra
- Perceptual Audio Coding (August 9 - 13)
Faculty: Marina Bossi
Guest: Richard Goldberg
- ANET: High Quality Audio over Networks Summit
(August 20-22)
Faculty: Chris Chafe, Theresa Leonard
- Digital Signal Processing II: Digital Audio Effects (August 16 - 27)
Faculty: Jonathan Abel, Dave Berners
Guest: Julius O. Smith
About Music & Sound Programs at The Banff Centre:
Music & Sound programs are dedicated to supporting emerging and
mid-career artists and to providing personalized
artistic direction suited to each participant. The goal is to nurture
the creativity of musicians and audio engineers in a setting that
allows for maximum personal artistic development and interaction with
other musicians and artists in The Banff Centre community. Music &
Sound alumni are found on concert stages and in professional positions
nationally and internationally.
Register now to ensure space, as availability is limited.
For more information and to register, visit:
http://www.banffcentre.ca/ccrma
e-mail: arts_info(a)banffcentre.ca
call: 1.800.565.9989 or 403.762.6180
Hi there,
I am discovering python, having looked for a matlab-like environement.
I am wondering now if it is possible to do some small multimedia
applications with it; more precisely, I would like to develop a
scientific application for audio/video analysis. Basically, I need to
show an avi
video with a synchronised waveform view of the sound, and some other
features views, like the pitch of the film voices (the actual pitch
detection doesn't need to be computed on the fly).
Python seems really great for rapid developement, but I wonder if it is
possible to play different media synchronously (the media decoding
itself will be of course coded in C/C++) with it? Does anyone here have
any experience with multimedia and python ?
cheers,
David
i'm trying to create a program to create 6 mono wave files from n mono
waves files in a virtual room, to re-create dolby 5.1 sound.
have someone ideas about best algorithm to divide sound among diffusor?
if someone of you is interesting, pls reply me...thanks
Hi.
I am currently trying to write a player for the DAISY 2 and 3 Digital Talking
Book formats for UNIX machines. One of the big great features of the
hardware DAISY players available is to set ones own prefers playback
speed while retaining the original pitch of the voice. This only works
within a certain percentage of variation of course, but it does
actually help a lot if you're used to fast speech. I am wondering
if there are any existing libraries/command-line tools to do this.
DAISY uses mp3 as its main audio format, so it would help if this tool could
do it with mp3 directly, but I can probably integrate other solutions too.
Any suggestions would be appreciated. Note that I do not really need an algorithm
that works well with music in general, it only has to work well with
the typical spectrum of a human voice.
--
CYa,
Mario | Debian Developer <URL:http://debian.org/>
| Get my public key via finger mlang(a)db.debian.org
| 1024D/7FC1A0854909BCCDBE6C102DDFFC022A6B113E44
Finally, I would like to introduce the OpenJay Development Krew [OJDK],
which actually is only a mailing list with little mail-traffic.
The OJDK is the right place for OSS Dj oriented software developers: here
you should find the right audience for discussing, sharing and improving code,
take / give suggestions and similar.
The OSS software is actually a powerful alternative (and with always
greater occurrence a refferral point) in many fields. Although it is not the
djing OSS case. There are many reasons for that: little OSS compatible
hardware,
few and small projects... few users...
The closed ring of open dj software is based upon few users and few
developers. Crashing it is my intention. To do that my efforts are enclosed
in three projects (enough to cover the whole issue) :
- OpenJay.org : the user side of the opensource dj world ;
- OpenJay Development Krew [OJDK] : the developer side of the opensource dj
world ;
- Jay'O'Rama : my personal software solution which I'm developing since 1 year
and that will be only an alternative ;
You should think to OJDK not as a project factory, but mainly as an improving
factor for code and a discussing place. Projects will come if needed and
desired.
Please...if you are an interested developer, CATCH THE OPPORTUNITY, join
the OJDK list and mail it! More than money or hardware, I need more than ever
some community help in these directions...
There are already some project developers joined our little community.
See the homepage for more info:
http://www.openjay.org/ojdkhttp://www.ojdk.tk
--
          J_Zar
        Gianluca Romanin
        ----------------
   see you at OpenJay.Org