Hi
I'm a bit confused about midi note numbers:
1) am I right that the midi note number goes from 0 to 127?
2) which note name corresponds to 0? C0? C-1?
3) which would be the correct range (in note numbers) from a 88 note
keyboard?
--
peace, love & harmony
Atte
http://www.atte.dk
Hi all,
It seems that I am unable to reach Patrick (getting weird 550 relaying
denied issue even though I am not relaying on my end). So, Patrick, if you
can get this e-mail, could you please contact me asap?
Many thanks!
Best wishes,
Ivica Ico Bukvic, D.M.A. Composition
Virginia Tech
Dept. of Music - 0240
Blacksburg, VA 24061
(540) 231-7047
(540) 231-5034 (fax)
ico(a)vt.edu
http://www.music.vt.edu/people/faculty/bukvic/
hey..another post here... this one to tell u about new stuff...
these days I've been very busy, so, I have not too many time to play
or record, but the other day I've recorded a strange soundcheck with a
small groovebox "a capella"... it has been terrible, but I've decided
to upload it...
4 all those electronica lovers... a very minimal session on my proyect:
http://perlssdj.blogspot.com
enjoy it...
( any kind of comment is good, thanks... ) .... :)
--
... visit always http://perlssdj.blogspot.com 4 cool stuff !!...
I am trying to connect to the internet using a debian distro but I can not
even connect to my router.
I have 3 other debian based distros on the same machine that are all able to
connect.
I tried to go into the router via telnet but when I type in the router
address 198.168.1.1 I get the reply that is was unable to establish a
connection.
I also went into my /etc/resolv.conf file, which I had to create, and
entered the primary and secondary DNS addresses.
and set nameserver to the above router address.
I have had a look at Linux.org beginers guide on how to connect but I do not
get further than telnet connecting to the router.
Can some one please guide me through what I need to do to connect this
distro to the Internet.
I have looked at the /etc/resolv.conf and /etc/network/interfaces files in
distros that work and made changes to these files in demudi but I still can
not access the internet from it. When I go to PPPoE setup it says that it
has found one device, eth1, and when I tell it to set it up it says that it
fails.
I have been struggling for some time on this now. I have googled and read
the demudi FAQ but just can not find anything that enables me to get this
distro to connect to the internet.
If someone can at least point me in the right direction that would be of
some help.
Many thanks.
--
View this message in context: http://www.nabble.com/Demudi-can-connect-to-the-internet-t1380902.html#a370…
Sent from the linux-audio-user forum at Nabble.com.
My name il Alessio,
I'm writing from Italy and I'm aproaching to linux audio recording.
We've bought a ice1712 card and we use a P4 2.8Ghz (non HT) with 1Gb
of Ram, sata disks and a SiS MoBo.
Can someone give me an idea about the limits of this system?
Thansk in advance!
--
l'Alessio
"I computer sono incredibilmente veloci, accurati e stupidi. Gli
uomini sono incredibilmente lenti, inaccurati e intelligenti. Insieme
sono una potenza che supera l'immaginazione." Albert Einstein
I'm experiencing problems with plug:jack. Playing a 96k file through
the device with aplay works nice, but when playing a file with
different sampling rate than the device is set to, it just stutters a
little and then exits.
My understanding is that it's supposed to resample the file with
ALSAs' own resample code.
I won't mention any specific hardware, cause this is just a question
to hear if it's a fact that plug:jack resamples. I already have a more
thorough report filed on alsa-user:
http://www.mail-archive.com/alsa-user@lists.sourceforge.net/msg15735.html
--
Esben Stien is b0ef@e s a
http://www. s t n m
irc://irc. b - i . e/%23contact
[sip|iax]: e e
jid:b0ef@ n n
hi there,
hope this mail is on topic on this list.
for an FM and internet radio project during the
worldcup 2006 in Berlin, we're looking for support on
the linux audio developer side. it's a cultural project
and we'll recording audio events at different
places, from clubs to universitities. there's a small budget
included and some preliminary code is available from
an older streaming project.
of course the project will be released under GNU.
if you are interested to join in and help us with your
work, please contact us quickly at:
kontakt->radioeinszueins.de
here's the basic setup we're working on,
it derives from years of practical d.i.y.
experiences with recordings and streaming
of club events.
your questions and comments are welcome.
.....................................................................................................................
>> automatic event recorder:
questions/tasks:
- how to profit from 24bit to 16bit dithering and 88.2 khz to 44.1 kz resampling
for dynamic limiting/compression aka "Automatic Gain Control"?
- silence detection (pause of recording or auto cutting)
- premastering to have recordings broadcast-ready and normalized
(up to the much hated brickwalled optimod FM sound..)
- tradeoff between sound quality and throughput
- presets for speech, music (jazz, classic, rock, electro), room micros
+ making podcast "secure"
+ usability, maintainablity, error control + watchdogs.
hardware: terratec phase 22, amd 1400 mhz,
512 RAM, 160 HDD, 2HE case connected to lan/dsl
router and balanced audio signal from mixer / pa.
input:
scheduler data via ical and/or dublin core metadata.
analog balanced stereo audio in, 0 - 17 db
output
podcast xml, mp3 lame 128kbps
archive quality luxury version: ogg, flac
including appropriate id3 tags with event metadata.
file repository (like apache-modmp3) with secure access
logs and error reports
control:
via web interface
proposal:
when you do not have the time to master large amounts
of recorded audio material by hand, an automatic gain control
at the time of the recording could help a lot.
aiming at a good tradeoff between dynamic compression and
sound quality, the main issue is to get a good leveled signal
in the digital domain. it could be done by auto-readjusting the input
gain, or by using using the headroom of 8 bit,
before the 24bit to 16 bit conversion for some smart compressor/limiter
magic in real time. the mechanism of dithering/downsampling is known from
mastering at the end of the chain (e.g. the waves l-1 maximizer).
so why not using it for an unmaintained non-annoying AGC?
maybe appropriate algorithms are available from voip projects,
or by directly using vst plugins, or tuning of the jamin multiband
compressor. a window manager like gnome is not obligatory.
we're experimenting with a plugin-chain for
alsa, jackd, ecasound based on ladspa/fst.
by now we can get a basic setup, but we're not even
sure if the AGC is realizeable in this short time.. some
basic code is available but we'd need help with
making it really run on a more advanced level.
This week's subjects will be:
* Live Internet Broadcasting.
* Icecast. The Broadcast Server.
* Darkice. The Broadcast Encoder.
* How will your stream be delivered to your audience?
* Jack Review. Basic Concepts. Stream Monitoring.
* Configuring Darkice for mp3, vorbis and jack.
* Configuring Icecast.
Here's the link
http://radio.socialtechnology.net/?q=node/27
-lee
----- Brian Dunn <job17and9(a)sbcglobal.net> wrote:
> I'm trying to un-ipod some songs.
> Does anyone know why the following command gives me an mp3 with a
> horid
> hiss?
>
> #!/bin/bash
> for i in *.m4a; do
> faad -o - "$i" | lame -m s - -o "${i%m4a}mp3"
What happens if you decode it to a wav file and play that? If it's not hissing, read the lame manual page and look for the -x option.
-lee
