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
Greetings:
I've made some minor updates and URL corrections for the Linux
soundapps site, but I've also discovered a problem with the European
mirror. The site at www.linuxsound.at now presents an advertisement for
ATNET, and the advert includes a link to http://linuxsound.atnet.at.
Alas, that link doesn't work correctly either. I've written to ATNET
twice already and have heard nothing from them. If the problem persists
for another week I'll remove that link. The Japanese mirror is also
experiencing a problem: apparently it isn't updating the top and TOC
pages, which is uncool because I've added material to both those parts.
Hopefully the Japanese mirror will update completely by this weekend.
For the time, the only completely current site for the Linux soundapps
pages is now:
http://linux-sound.org
And you all know the rest...
Best regards,
dp
[ Sorry for cross-posting - feel free to forward around ]
======================================================================
FSMSI 2004
First european seminar
on
Free software for multimedia streaming over internet
June 23/24, 2004
IRCAM, Paris, France
http://freesoftware.ircam.fr/fsmsi2004
======================================================================
SCOPE OF THE SEMINAR
The exponential growth of network speed makes it possible to
exchange high quality audio and video contents over the Internet,
often with real time capabilities. However, multimedia streaming
still presents several issues with regards to Free Software and open
standards: patents on software and data formats, free codecs
availability, standards definitions, etc, etc. The goal of this
seminar is to provide an overview of existing Free Software for
multimedia streaming over Internet, to envisage future developments
and collaborations in this area and to examine the possible ways to
fund these developments.
TOPICS OF INTEREST
* Standards and open protocols for multimedia streaming
* Free implementations of open protocols
* Free players for multimedia contents
* Free servers for multimedia streaming
* Server architectures for multimedia streaming
* Open multimedia formats
* Free codec implementations
* Real time multimedia streaming
* Interactive multimedia applications
INVITED SPEAKERS
Carlo Calabrò (Videolan)
François Déchelle (Ircam)
Dominique Fober (Grame)
Andrea Glorioso (MIU-FT)
Eric Gressier (Cnam)
Olivier Lescurieux (Ircam)
Hans-Nikolas Locher (Cnam)
Juan Carlos De Martin (IEIIT-CNR/Politecnico di Torino)
Yann Orlarey (Grame)
Patrice Tisserand (Ircam)
SEMINAR PROGRAM
Thursday, june 23:
* presentations by the invited speakers
* round table discussions for future projects and collaborations
Friday, june 24:
* presentations by the invited speakers
* round table discussions for future projects and collaborations
* "Bring your own laptop" hacking sessions
ORGANIZATION COMITTEE
Francois Déchelle, IRCAM (dechelle(a)ircam.fr)
Andrea Glorioso, Media Innovation Unit - Firenze Tecnologia (sama(a)miu-ft.org)
SEMINAR STREAMING
The presentations will be streamed in OGG/VORBIS format.
CONTACTS
Web site: http://freesoftware.ircam.fr/fsmsi2004/
Mailing list: freestreaming-paris2004(a)lists.miu-ft.org
http://lists.miu-ft.org/mailman/listinfo/freestreaming-paris2004
Hello,
JJack 0.1 - Java bridge API for JACK has initially been released.
http://jjack.berlios.de/
JJack is a framework for the Java programming language that allows
creating and running portable audio processor clients for the JACK Audio
Connection Kit.
There are 3 alternative ways to run JJack clients:
- using the JJack shell application
- as JavaBeans
- as standalone application
Please let me know if you have created a Java audio application with
JJack, it can be made available for download on the JJack site if you like.
If someone knows how to compile the native bridge library libjjack.c for
other OSs than Linux, please send the resulting binary to me.
Co-developers are always welcome.
Make noise.
Jens