Yes, as long as you (or an init script) runs "alsactl restore" on the
next boot.
Save/restore mixer settings should work OOTB with any decent distro,
normal users should never have to use alsactl.
So how and where do I put this in an init script?
i happened to have a short glimpse on your posting,
david. i don't want to insult anyone, but there is a
bit more to evaluating accuracy than comparing sample
values at fixed locations ;--)))))))
consider a few of these things :
a) non-causal algorithms introduce a natural delay.
for windowed sync interpolators this is half of the
filter kernel. ok, the algorithm could skip those
samples of course (that's what fscape is doing by the
way, if i remember correct).
b) the algorithm could introduce a gain factor
different from 1.0, this is in fact very likely,
consider resampling 44.1 khz to 22.05 khz, a 0 dB
headroom file would be distorted if there is no gain
compensation.
c) the crucial point about windowed sync is that it's
naturally also a lowpass filter (for upsampling which
you were doing in your example). so, those filters are
windowed sync filters, they are linear phase and show
an effect called gibbs effect which means "edges" show
pre and post ringing phenomena. this is not
necessarily "bad" and with not extremely long kernels
it's not audible, too. in fact, when you run a
software synthesizer you will most likely want band
limited oscillators, i.e. those with gibbs effect.
now, no realizable filter is infinitely steep, so
you'll have a passband etc.
summa summarum it's very likely that for an arbitrary
input sound, resampling with an integer factor will
*not* produce equal sample values with the factor
spacing in the output.
without blowing up this stupid discussion, i suggest
to do two things :
a) *listen* to the results for an aural comparison.
this is in fact the most important point, because for
different ears different algorithms are better or
worse. i, for example, use bandlimited resampling in
most cases and in real resamping cases (e.g. 48 kHz ->
44.1 kHz), but sometimes i prefer the additional
colouring and brightening (= distortion) of say cubic
interpolations or even linear interpolations which
produce the hardware-sampler-kind-of-timbre.
b) read a book about audio analysis, and perform some
more appropriate tests, for example calculate the
harmonic distortion (try your example with a sine
tone!).
best, -sciss-
___________________________________________________________
Gesendet von Yahoo! Mail - Jetzt mit 1GB Speicher kostenlos - Hier anmelden: http://mail.yahoo.de
All:
I would like to announce the Mondrian Project. Mondrian is an attempt
to create a text-based setting for writing and performing music. It
currently consists of the following parts:
- a simple music description language intended to be small and
keystroke efficient
- a Mondrian-to-MIDI converter
- interactive sequencers intended to play Mondrian code in a
live performance
- sequencer plugins for vi and emacs that turn text editors into
musical instruments
You can find Mondrian at
http://www.math.tu-berlin.de/~brinkman/software/mondrian/
There's an online demo that'll let you try the language without
the need to install anything.
Any feedback would be appreciated!
Peter
Hello,
I have a purely academic question stemming from an accidental install on
my part:
Will my AMD64 laptop run PD and other sound apps (using mingo's RT
patched kernels) faster in native 64-bit linux or faster as a
conventional 32-bit linux?
I ask this because I am about to install the "correct" 64-bit linux but
I will then have array problems and other PD errors that have not yet
been solved... So if my machine is faster in 64 -bit linux the
shortcommings of 64-bit PD are outweighed by the speed gain.
Thanks all!
-thewade
> Just to be sure: this is the MIDI/gameport on the mainboard?
Yep.
> What ALSA version are you using?
Advanced Linux Sound Architecture Driver Version 1.0.9rc2 (Thu Mar 24
10:33:39 2005 UTC).
> What kernel version?
Linux demudi 2.6.12-3-multimedia-k7 #1 Thu Jun 23 12:52:11 CEST 2005
i686 GNU/Linux
I just installed the latest version of Demudi, and I haven't really done
much to the default install yet.
I've got the following etc/modules:
snd-seq-midi
ide-cd
ide-disk
ide-generic
psmouse
And I'm not sure what options are set for snd-mpu401, and don't know how
to find out.
Thanks,
Kevin
HI All
Hopefully the right link now , sorry :
You should just be able to play/download the mp3.
http://www.soundclick.com/bands/songInfo.cfm?bandID=390090&songID=2785733
This is causing quite a stir on some of the more windows sites, to say
the least!
Mostly I get:
Whats Ardour and can I where can I get it?
Must be that taste of Freedom they hear!
Cheers
Bob
bearmusic
First: sorry for crossposting on two lists (inux-audio-user; pd-list),
hope it doesn't bother anyone...
Second thing: hello!
Third thing: the facts:
I now own a Wacom Intuos 2 tablet which I'd like to use with Pure Data
just like a MIDI controller. I plugged the thing in my PC equipped with
Ubuntu Hoary (kernel 2.6.10-5 included in the distro) and it works
plugnplay out of the box as a mouse would ("pointing device" seems to be
the scientific name for that stuff).
The two things I'd like to be able to do are:
- Stop using the tablet as a pointing device. To make it simple: I'd
like it to stop moving my cursor over the screen!
- Be able anyway to input data into Pd (x, y, pressure would be an
appreciated plus...)
If it is possible to do, I'm sure someone has done it already. If I'm
lucky he reads the lists...
How should I procede?
Thanks in advance...
++
Jé
Part 1 of (maybe) 3
Some time ago, a question was asked on LAU regarding resampling of an
audio library.
I strongly recommend the following, assuming that you decide to
resample (and that's your call, not anyone else's):
If you can at all avoid it, *don't* use sinc resamplers (for example,
sndfile-resample and resample-1.7) for bulk rate conversions at
constant rates, especially for resampling something of great value
such as an audio library which you plan to use over and over again on
future projects.
To be most certain you've got a good library at the new sample rate,
re-record the original sources or purchase a competently produced
new library.
------------------------
In answering this question about resampling an entire audio library,
Steve Harris (prolific author of LADSPA plugins for which we are all
grateful) posted: "I would say that a high quality sinc resampler will
be the best quality."
Upon contacting Steve and exchanging several emails, it became clear
that Steve was actually talking about the theoretically ideal case,
not about the algorithm developed and described by Julius O. Smith III
of CCRMA and implemented in such codes as sndfile-resample and
resample-1.7. Steve says he doesn't have enough experience with these
programs to either agree or disagree with claims made about them.
For many readers, "sinc resamplers" probably does refer to these
programs, so I think it's important to attempt once again to put them
into perspective:
Julius O. Smith III of CCRMA, in his PDF document describing a sinc-
function based algorithm in detail, himself mentioned the suitability
of this particular algorithm for situations wherein the sampling rate
was *changing* and *arbitrary* such as audio scrubbing, a real-time
application of resampling. Resampling an audio library is an
application of *bulk conversion* to a *fixed sample rate* and is very
different, therefore, as is easily arguable, an unsuitable application
for Smith's algorithm, therefore unsuitable for sndfile-resample and
resample-1.7.
I may demonstrate some problems with bulk conversions later such as I
did for Steve, but first should point out that anyone who is familiar
with Fourier analysis and synthesis should appreciate immediately why
this is the case upon reading Smith's description of the algorithm: As
described in [1] it utilizes very small, comparatively infinitesimal
windows (compared to the infinite signals of the Sampling Theorem) and
is of order n squared. For someone with the proper education and/or
experience, this is really
*all that is necessary to know*
to predict problems with speed or accuracy --- or both of these.
------------------------
Reference
[1] Julius Orion Smith III. Digital Audio Resampling Home Page.
http://www-ccrma.stanford.edu/~jos/resample/. This reference cites
(after reordering author/title): J. O. Smith and P. Gossett. Flexible
Sampling-Rate Conversion Method. In Proceedings of the IEEE
International Conference on Acoustics, Speech, and Signal Processing,
March 1984 (ICASSP-84), Volume II, pp. 19.4.1-19.4.2, IEEE Press.
Note the word "flexible."
Something miserable for a miserable day.
http://www.dis-dot-dat.net/content/music/socold.ogg
Also available in mp3 flavour.
--
"I'd crawl over an acre of 'Visual This++' and 'Integrated Development
That' to get to gcc, Emacs, and gdb. Thank you."
(By Vance Petree, Virginia Power)