>>>it's easy for non-programing people to bring "visions" regarding
>>>interface design. (and i love do so :) as i know programers, it's quite
>>>hard to establish a new standard. but imho the interface standards
>>>(buttons, dropdown boxes, scrolling, menu-structure, etc.) are now a
>>>couple of years old, and there might be better solutions for specific
>>>tasks. audio seems to me like a good point to start.
> i wasn't talking about such rudimentary stuff. of course there are
> alternatives to these basic widgets and several audio applications (even
> free ones) have begun to support them.
> the point about a visual interface is that it acts as a "memory buffer"
> for the user: you do not have to remember much about the structure of
> the session because the structure is made visible on the screen. can't
> remember precisely where you put a certain sound? how many copies of the
> bridge riff did i put in? is the door slam before or after the creak?
> its all there on the screen, just waiting for you to look at it.
> as soon as you move away from a visual UI, you have to find some way to
> avoid requiring the user to remember everything about the session.
when i try to remember a poem my brain creates images and i walk trough
them, when i reproduce it. when i learn a piece of music it does other
stuff (i'm a pianist and singer) but in the end i have a very complex
thing in my mind, just think of a bach fugue. i have the fugue also in
"the fingers". different areas of the brain work together. i have the
same oppinion as you, we are very good in using a visual UI. we trained
it for a long time. but there could be other combinations that work
nearly as good as "mouse-to-eye".
> the visual interface offers another hard-to-replicate feature as well:
> trivially variable precision. if you try doing cut-n-paste based only on
> audio feedback, you will find it quite hard/laborious to be as precise
> as you might want to be. with the visual interface, its much easier to
> use visual information to get the rough location of an edit and then
> get to precisely where you want, without many steps. with audio feedback
> based approaches, i think you will find yourself needing many more
> iterations through the edit-play-edit-play cycle before you get the
> location correct.
i think it's all a matter of training. you do the
"display-keyboard-mouse-combination" for long years and you became
professional in speed and precision. watch a pro-gamer gaming with
mouse.. what's about data-gloves? whats with feet-controlers and other
(sorry for my clumsy english)
I am looking for suitable hardware to handle digital i/o between a Linux
system and an RME ADI-2 ad/da converter that I just bought. I don't need lots
of channels, but reliability of the data transfer is important, including
jitter reduction. An RME card would be excellent but it is somewhat outside my
budget. Also, connectivity to a laptop would be desirable, suggesting either a
USB interface or waiting until http://freebob.sourceforge.net/ (the Alsa
firewire project) matures. I don't plan to run any OS besides Linux with this
hardware, so Alsa support is crucial. As this is for home/personal use I'm not
in a hurry. M-Audio hardware is high on my list of possibilities at the
Now to the software question: does there exist any sound editor with a
non-graphical interface, i.e., one that can be operated from the Linux console
for inserting, deleting, copying and otherwise editing audio? Due to a
vision-related disability I can't use a graphical display and therefore need a
text-only solution - but all the sound editors appear to require X11. Surely
it should be possible to design an audio interface to a digital sound editor.
I've discussed hardware on this list once before, and the USB options weren't
highly regarded at the time.
hello all - got a question - I've only recently been stopping and taking a
look at my studio computer's performance and in the almost year since I
change from Red Hat 9 to gentoo, it's been more solid on some things, but I
notice a huge latency difference - ie: I have to run Jack at -p 8192 to get
anything done in Ardour
Anybody have any tips on what to look at to tweak it? Seems like it should
do better than that... I didn't see it as a problem until in the last few
days I started playing with playing softsynths live directly into Ardour -
you've gotta be running at -p 1024 or there's a latency that screws up your
playing - at 8192 it's a downright 8th note delay...
Here's some vitals that I can think of:
OS: gentoo 2.6.6-rc1 kernel (alsa built in)
jack command line:
jackd -R -d alsa -d hw:0 -r 48000 -p 8192 <------- (or whatever)
io support: 32 bit
use dma on
chip: 2ghz amd (I THINK - not at computer now)
thanks for any ideas! :)
>From: Neil Durant <lists(a)sphere3.co.uk>
>I'd be happy to sample the full lengths of all 35 notes for all six sounds,
>storing them as wavs. I don't have a lot of spare time these days, so I'll
>let someone else have the pleasure of cropping/editing!
Please do so. Make the files available from your site or
Note: If we end up to conclusion that the samples cannot
be put freely available, then I could make the samples
privately available for the following kind of project.
Research experts should analyse the sounds and come up
with synthesis method which generates as similar sounds
as possible. I'm aware of such research teams and I could
ask them to analyse the samples.
I also have a plenty of research papers on such analysis/synthesis
methods. I could place the papers privately available for anyone
who wish to write the analysis/synthesis software.
for developers of open source graphics software
jack_capture is a small simple program to capture whatever
sound is going out to your speakers into a file.
This is the program I always wanted to have for jack, but no
one made. So here it is.
jack_capture [-f filename] [ -b bitdepth ] [-c channels] [ -B bufsize ]
Filename is by default auotogenerated to something like "jack_capture_<date+exact_time>.wav"
Bitdepth is by default FLOAT.
Channels is by default 2.
Bufsize is by default 262144.
Mostly based on the jackrec program in the jack distribution
made by Paul Davies and Jack O'Quin. Automatic filename generation
code taken from the timemachine program by Steve Harries.
This is mi first post here, the reason of this post is a new proyect (
a very small proyect ) that I've started, I only have released a pair
of old demos sample-based, and some strange and electronic photos made
Unfortunatedly, the sessions are not too much amazing ( and they are
made with windows... :S ), but I hope u join them, I've discovered
Linux this year and I'm very nodvice on this world... All u are
invited to the site, the link is on http://perlssdj.blogspot.com , I
hope the site can grow with time ( and help ), but by the moment I see
too much windows users on my counter... and this doesn't likes me...
I'm interested on to run ProTools in Linx ( like a good sound
technician student.. ), if u know some interesting link, make me know
it, I need more informaton first... ( I'm like a robot... XDD )
I hope u enjoy the demos... see u there...
...visit allways http://perlssdj.blogspot.com... and b happy !!
I just got a Frontier Design Tranzport and after a little bit of work,
I finally got it working with Linux. I was wondering if anyone was
interested in knowing how it works and what not. If so, I will put
together some documentation and source code and post it somewhere.
Yahoo! Mail - PC Magazine Editors' Choice 2005
I used to work with Debian (for 2 years as desktop) and I purchased
amd64 3 weeks ago just to have top performance and possibility to use
some dssi software / syntezathors etc...
Just, I need some advise from Gentoo users - as I have to say, that I am
just a little bit disappointed with poor *deb repositories, problems
with dependencies and ... just I am thinking of "switch" by audio
station to Gentoo (I love Debian for philisophy of GNU && independence
and always will, however I WANT TO MAKE THE MUSIC ;) )
So, please AMD64 Gentoo guys, advise me - am I thinking right or not ?
Also, how REALTIME LSM is working with Ingo Molnars patched kernel
2.6.14* / is DSSI working properly / maybe any Audio manuals for Gentoo ?
and of course - I am interested in PURE64 architecture ;)
ecasound is way too complicated for me. All I want to do is open jack
stereo ins and outs, load the LADSPA amp_stereo plugin, and control
that plugin's gain parameter via alsa midi.
The -Md:alsaseq parameter is apparently ignored, and I get some error
about a plate reverb?!
[paul@localhost paul]$ ecasound -i jack -o jack
-G:jack,ecasound,notransport -el:amp_stereo,1 -Md:alsaseq
ecasound v2.4.2 (C) 1997-2005 Kai Vehmanen and others
[ Session created ]
Chainsetup created (cmdline) ]
Keyword plate doesn't match to regex ^plate$ for
... object 'Plate reverb' ().
- [ Connecting chainsetup ]
'rt' buffering mode selected.
(eca-chainsetup) Audio object "jack", mode "read".
(audio-io) Format: f32_le, channels 2, srate 44100, noninterleaved.
(eca-chainsetup) Audio object "jack", mode "write".
(audio-io) Format: f32_le, channels 2, srate 44100, noninterleaved.
ERROR: Connecting chainsetup failed: "Unable to open MIDI-device: rawmidi."
(audioio_jack_manager) Connection closed!