>>>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
"non-standard" devices?
greets,
gabriel
(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
moment.
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.
Suggestions welcome.
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: 0.99.0
ardour: beta28
jack command line:
jackd -R -d alsa -d hw:0 -r 48000 -p 8192 <------- (or whatever)
harddrive:
multicount on
io support: 32 bit
unmaskirq on
use dma on
keepsettings off
readonly off
readahead on
chip: 2ghz amd (I THINK - not at computer now)
ram: 512MB
thanks for any ideas! :)
---------------------
Aaron Trumm
www.nquit.com
-----------------------
hi,
i would like to tell you all that my first (and non finished) howto on
how to configure a mandriva linux system for audio works is online at
www.rumoridifondo.com/progetto_mdaw
it's in italian only at the moment.
bye
emanuele
> 'm struggling with Scons, trying to build from source. Has anyone seen
> debian packages for this yet?
>
> Thanks in advance!
Make sure that PKG_CONFIG_PATH is correctly set in you /etc/profile or do it
before in the session doing the scons configure.
Then the thing will find the libraries it needs to compile. If it does not
find stuff it can live without but you want, install them first.
Greetings all;
Has anyone a url where I might be able to purchase the expansion
interface gizmo for an Audigy 2 value card?
Thanks.
--
Cheers, Gene
People having trouble with vz bouncing email to me should add the word
'online' between the 'verizon', and the dot which bypasses vz's
stupid bounce rules. I do use spamassassin too. :-)
Yahoo.com and AOL/TW attorneys please note, additions to the above
message by Gene Heskett are:
Copyright 2006 by Maurice Eugene Heskett, all rights reserved.
( LAU folk: this is an initial outline of an email I want to dispatch to
the desktop-architects list in the very near future. Your comments
are eagerly sought. Note that this section specifically seeks to
avoid any discussion of implementations or specific approachs. I
would like to fully flesh out the list of tasks ASAP )
Making Sound Just Work
------------------------
One of the "second tier" of requirements mentioned several times at
the OSDL Portland Linux Desktop Architects workshop was "making audio
on Linux just work". Many people find it easy to leave this
requirement lying around in various lists of goals and requirements,
but before we can make any progress on defining a plan to implement
the goal, we first need to define it rather more precisely.
DEFINING THE GOAL
=================
The list below is a set of tasks that a user could reasonably expect
to perform on a computer running Linux that has access to zero, one
or more audio interfaces.
The desired task should either work, or produce a sensible and
comprehensible error message explaining why it failed. For example,
attempting to control input gain on a device that has no hardware
mixer should explain that the device has no controls for input gain.
PLAYBACK
- play a compressed audio file
* user driven (e.g. play(1))
* app driven (e.g. {kde,gnome_play}_audiofile())
- play a PCM encoded audio file (specifics as above)
- hear system sounds
- VOIP
- game audio
- music composition
- music editing
- video post production
RECORDING
- record from hardware inputs
* use default audio interface
* use other audio interface
* specify which h/w input to use
* control input gain
- record from other application(s)
- record from live (network-delivered) compressed audio
streams
MIXING
- control h/w mixer device (if any)
* allow use of a generic app for this
* NOTE to non-audio-focused readers: the h/w mixer
is part of the audio interface that is used
to control signal levels, input selection
for recording, and other h/w specific features.
Some pro-audio interfaces do not have a h/w mixer,
most consumer ones do. It has almost nothing
to do with "hardware mixing" which describes
the ability of the h/w to mix together multiple
software-delivered audio data streams.
- multiple applications using soundcard simultaneously
- control application volumes independently
- provide necessary apps for controlling specialized
hardware (e.g. RME HDSP, ice1712, ice1724, liveFX)
ROUTING
- route audio to specific h/w among several installed devices
- route audio between applications
- route audio across network
MULTIUSER
- which of the above should work in a multi-user scenario?
MISC
- use multiple soundcards as a single logical device
Download from http://ccrma.stanford.edu/~kjetil/src/
Snd-ls v0.9.5.4
================
Contains
--------
Snd v7.15 from 17.8.2005
About
-----
Snd-ls is a distribution of the sound editor Snd. Its target is
people that don't know scheme very well, and don't want
to spend too much time configuring Snd. It can also serve
as a quick introduction to Snd and how it can be set up.
Changes 0.9.5.3 -> 0.9.5.4
--------------------------
-Changed default resampling quality to SRC_SINC_BEST_QUALITY
-Added workaround for shift-handling across various keyboard settings.
(shortcuts for zoom in and undo works with american keyboards now.)
-Added check for Guile 1.8. Snd-ls crashes with guile 1.8.
(all versions I have tried of 1.7 seems to work though...)
-Use JackPortIsPhysical instead of "alsa_pcm" when finding jack ports.
-Updated the rt stuff to latest versions.
***************************************
***************************************
Das_Watchdog V0.2.1
===================
About
-----
Das_Watchdog is a general watchdog for the linux operating system that
should be runned in the background at all times to ensure a realtime
process won't hang the machine.
Changes 0.2.0->0.2.1
--------------------
*Cleaned up source a bit.
*Properly find number of timer processes.
*Added shortcuts for optargs.
Hi all,
A couple days ago I sent an e-mail on this topic due to some initial positive
feedback, yet not much has happened since... (it may very well be that a lot
of people are very busy with LAC preparations (-: ). At any rate, please allow
me to reiterate what I've mentioned before in hope that this time it may
elicit some fruitful discussion on this IMHO very important topic.
Best,
Ico
BEGIN-OLD-MESSAGE
It appears to me that there were at least a few members of the LAD/LAU list
who have expressed interest in having LAD site somehow integrated in the
Linuxaudio.org. I believe that this would be a very encouraging step towards
consolidating online LA resources into one site which would IMHO ultimately
make LA users' lives a lot easier as well as make the overall LA scene look
more professional to the outsiders/potential adopters. I see this kind of an
idea as a first step towards a much more demanding goal--integration of other
online resources, i.e. Dave's LA software page. I could see this integration
happening via a single Wiki page that would contain detailed
info/screenshots/documentation/mailing-list and other pertinent info for every
LA software available out there. Naturally, linking these lists is also a
possibility, yet the very thought of having one place with unified appearance
that would provide all the necessary info, including documentation,
application-specific mailing lists etc. seems IMHO truly inspiring.
Such a project obviously bears a huge overhead. I can also see devs objecting
to the redundancy of information that may be already available on their
software's dedicated website. The solution to both problems would be asking
devs and/or their project maintainers/helpers to assist with the generation of
their software's Wiki page which should adhere to certain predetermined
standards and then also providing a link to their original project's page.
Yes, there would be some redundancy, but a vast number of projects could
greatly benefit from such a consolidation, including one of the most
important, yet often neglected aspects--proper documentation.
For this reason, I would like to use this opportunity to possibly elicit a
discussion on this matter and hopefully get the ball rolling :-).
END-OLD-MESSAGE
>Because no converter can reach even the 24bit resolution. In fact the best
>resolution you can reach is about 21 bits and the rest three bit contains
>only a random thermal noise.
>
>regards,
>Ctirad
I did not know that - but am not really surprised.
>There are not any true 24bit a/d converters yet.
>There are not even any true 22 bit a/d converters, perhaps in some labs.
>
>I think the best converters in the world manage about 20bit. (120db
>dynamic range.)
>
>If a converter had a 24bit dynamic range (144db) and full scale was
>+7db, then it would have to be able to resolve differences of 10
>nano-volts (10 one billionths of a volt). That's perhaps possible with
>cryogenics. Remember each extra bit *doubles* the dynamic range!
>
>Real 24 bit recording should resolve from below the brownian noise floor
>of air molecules hitting your ear drums to beyond the threshold of pain.
>
>That's why we are stuck at 24bit
Well thank you for a scientific explanation of this ceiling.
I guess, then, that *real* 24-bit resolution, or something very close to
it, would yield what I am looking for - if it can be achieved.
>Recording is about creating illusions, not fidelity. If you record an
>acoustic guitar in a totally dead room with the flattest most accurate
>mic and pre, in to best a/ds in the world, it sounds... ok.
>Put some reverb and top end on it, a little compression, perhaps add a
>little distortion with an aural exiter, or recording to tape, and people
>will say 'wow, what an amazing fidelity guitar recording!' :)
I agree with this to a certain extent, but the quality of the effects - or
the final signal after the effects are added, is affected by the fidelity
of the original signal.
There is a huge difference in our guitar sound put through an 8-bit Zoom
processer, an 18-bit Alesis Q2, a 20-bit Alesis Q20, and a Behringer
"24"-bit V-Verb.
I think it is about both - using a high-fidelity acoustic signal blended
with creative, high-quality effects to create a beautiful auditory
experience.
>Bullshit. If you can hear the difference between a 20 bit converter
>and a >20 bit one, what you hear is the difference between two
>converters, regardless of the number of bits they use.
And you can prove this?
I would assume, that if "24-bit" converters are really only 20-21 bits,
then a so-called "20-bit" converter is likely <<20 bit.
I maintain that I *can* hear bit-depth difference.
Are you perhaps suggesting that there exists some bit-depth threshold w/re
to human hearing?
What do you base your comment on?
>Even 16 bits correctly dithered is better than 24 tracks on a 2 inch tape.
Again, what do you base this on?
Recording what?
"Correctly dithered" - and you would maintain that there is some objective
standard as to what constitutes this?
I can hear the distortion of the audio signal created by dithering, just as
I can hear the distortion of the audio signal created by Dolby - and I
don't like it.
If you think existing digital technology can already match or exceed the
audio fidelity of a 24-track reel-to-reel recorder, I would very much like
to know what it is, and where it is available - and I would like to hear
it.
-Maluvia