Mark Knecht:
> > http://www.notam02.no/radium/
>
>
> Kjetil,
> Hi. Radium looks interesting, but I'm not sure I understand it from
> just a quick look at screen shots. Is it along the lines of a tracker?
Its a bit like a tracker, yes. But without a lot of the limitations
found in trackers. It takes the best from the tracker, which is fast
editing with lots of information using small space; less time used for
navigating. The screenshot is a bit outdated though, in >v0.60
you can have a total graphical layout looking more like a vertical
pianoroll as an alternative. There is also a new track showing
the tempo graphical with gradiently changing colors.
> Time seems to be progressing downward.
Yes.
> What is placed in each column?
Its sort of explained here:
http://www.notam02.no/radium/docs/radium_Sections/visibleoverview.HTML
> Wave files? Loops?
Unfortunately just midi data for now. The code is modular build up,
though, so its not ~that hard to add sample-support.
> I'm guessing the numbers in the left column (headed
> with the number 15) indicate how long you play a loop at a given
> vertical slot?
The track to the right for the track headed with the number 15
shows the sublevel, and the track headed with the number 15
shows the linenumber in the level showed in the sublevel track.
Its a flexible way to zoom, and has nothing to do with looping.
The number "15" shows the linenumber at sublevel 0.
sublevel 0 is colored black
sublevel 1 is colored white
sublevel 2 is colored brown
etc..
Its not that confusing when you actually use it yourself, and
you don't have to use it if you don't want to.
There are two ways to zoom, the one I have been talking about
now, which I have called local zooming, and the other way
which is global zooming. The global zooming is just a plain
graphical effect. You do a local zoom by pressing
left shift+<arrow down/up> and global zooming by pressing
left meta+, and left meta+. and left meta+.
> Does the relative tempo column effect the pitch of the
> files being played?
>
> Anyway, it looks interesting.
>
Thanks. :)
--
1. A short summary of changes
Minor bugs in JACK support have been fixed. Now Ecamegapedal
makes sure it won't launch the JACK daemon by accident
when probing for available devices on startup. The manual
pages have been updated with some new sections.
---
2. What is Ecamegapedal?
Ecamegapedal is a real-time effect processor software with
a graphical user interface for controlling the effect
parameters. It is meant to be used as a virtual guitar-fx
or studio effect box. In addition to real-time operation,
Ecamegapedal also supports reading from and writing to audio
files. All audio object and effect plugin types provided by the
Ecasound libraries are supported. This includes ALSA, JACK,
OSS, aRts, over 20 file formats, over 30 effect types, LADSPA
plugins and multi-operator effect presets. Ecamegapedal's
implementation is based on Ecasound and Qt libraries.
Ecamegapedal is licensed under the GPL.
---
3. Contributors
Patches
Kai Vehmanen (various)
---
4. Links and files
http://www.eca.cx/ecamegapedalhttp://ecasound.seul.org/download/ecamegapedal-0.4.4.tar.gz
---
http://www.eca.cx
Audio software for Linux!
Hi folks!
I'm looking for someone to accompany me to Karlsruhe by train. I'm coming
from Paderborn. But I will go through Dortmund, essen, duisburg, Koeln. To
those places I can get on my own, from there I need help. For those of you who
don't know it: I'm blind. So I get a free companion. That menas half the
price. I want to go there on wednesday and leave on sunday. On wednesday Joern
can come with me, but the way back is unsure yet. If anyone goes that way:
just tell me...
Kindest regards
Julien
http://ltsb.sourceforge.net - the Linux TextBased Studio guide
---------------------------------------------------------
SBS C-LAB
Fuerstenallee 11
33102 Paderborn
Phone: (+49) 5251 60 6060
Fax: (+49) 5251 60 6065
www.c-lab.de/~wegge
Hi,
As I've gone through my most challanging mastering
project, I've developed more questions than answers.
There's been reference, on this list, to documentation
that explains file formats-- I don't recall the
document title or where to find it. Ultimately, I need
a resouce that explains things like; the number of
available samples for the different bit depths (16bit
range from -ABC to +XYZ, 24bit...), DC offset is, peak
amplitude is, RMS is, etc.
I need to know if samples are syncronous with decibel
level, is maximum samples equal to 0db?
In the following sndfile-info report, what are Length
and Block Align?
Version : libsndfile-1.0.6
========================================
File : guajira-jam.wav
Length : 30396228
RIFF : 30396220
WAVE
fmt : 16
Format : 0x1 => WAVE_FORMAT_PCM
Channels : 2
Sample Rate : 44100
Block Align : 4
Bit Width : 16
Bytes/sec : 176400
data : 30396184
End
----------------------------------------
Sample Rate : 44100
Frames : 7599046
Channels : 2
Format : 0x00010002
Sections : 1
Seekable : TRUE
Duration : 00:02:52.313
Signal Max : 25922
Is Signal Max a measurement of used samples?
One of the problems I've confronted during mastering
is JAMin, Ardour and hardware meters tell me that a
track is peaking around -0.5db but the hardware mixer
indicates overload. Of course metering balistics being
what they are this is understandable. The mixer
documentation doesn't tell me at what level the
overloads are set to go off at and I haven't found a
configuration interface.
With this metering and overload discrepancy I'd like
to read a sndfile-info report that tells me; of the
available samples, this file uses a minimum of X and
maximum of X.
I wonder if some of qualities any of us should know
about our mastered files includes:
Min Sample Value
Max Sample Value
Peak Amplitude
Possibly Clipped
DC Offset
Minimum RMS Power
Maximum RMS Power
Average RMS Power
Total RMS Power
Of course another challange is tools like sndfile-info
assume that a file exists. This is not always the case
and in my situation it's almost never true. I return
JAMin output to an Ardour return bus and don't produce
a file until the return bus is exported. Printing a
track to the file system and then analyzing it is no
way to save time.
Anyway, I appreciate all the responses to my past
questions and am hopeful that someone can look at the
current mumbo jumbo and prescribe some effective
medications; coffee, sleep, black bear gallbladders,
urls to useful documents, etc.
ron
__________________________________
Do you Yahoo!?
Yahoo! Tax Center - File online by April 15th
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Ron,
I think Erik gave the most accurate answers to your questions; however,
I do have a couple of comments:
1) Erik said that 0db is the maximum sample value. Just a clarification,
he means the maximum possible sample value, not the maximum sample
value in a particular file. For example, for 16 bits, 32767. It's
not a relative measurement, but an absolute number, despite the db label.
2) Erik also said that the Block Align is an internal detail. Well,
yes that's correct, but it isn't anything complicated or secret. It
is the number of BYTES for a sample (2 for 16 bits, 3 for 24 bits)
multiplied by the number of channels. 16-bit stereo should give
you 4; 24-bit stereo should give you 6. This information is
redundant in WAV headers, and perhaps this is why Erik said not to
worry about it. It would be better if it wasn't there because
it's primarily an opportunity to screw up a program (or a posting!).
Hi,
I have HTPC entertainment with home automation control integrated (Based on
Misterhouse). I also have couple of sound cards for 2 independent stereo
outputs that go to separate rooms. Now I start 2 instances of alsaplayer -
playing music to those rooms.
Now I have also some other processes on PC (Apache, PVR 350 video recording,
media serving to LAN) - so I get occasional hickups on music output. But I'd
like to play music smoothly probably at the price of resposivness of other
applications.
With what settings/configuration/additions could I achieve that. I assume
that assigning higher priorities may be not enough ?
How do you tweak your machines to get smooth audio play ?
Regards,
Robert.
Hi
I'm confused about how to setup alsa. Originally I managed to get things
working, but after buying an usb sound card, I'm even so confused that
I'm not exactly sure what to ask, but I'll try.
I run debian/unstable on a laptop with my own 2.6.4 kernel and alsa 1.0.
I have the following hardware that I believe to be related to alsa:
*Onboard i810 soundcard
*USB Edirol UA-1A sound card
*4 port usb hub
*2 evolution usb-keyboards connected to the hub
*I need one virtual midi port for each keyboard, so 2 for now
The problem is that right now the cards get assigned a card number that
reflects in what order they were inserted. I'd rather have them show up
on fixed locations. Below is my childish attempt on a /etc/modutils/alsa
file. I'm pretty sure it's not correct to refer to snd-usb-midi, since I
don't have a module by that name. Could anyone please give me a hand +
some general bckground information?
Thanks in advance!!!!
alias char-major-116 snd
alias char-major-14 soundcore
# i810
alias sound-service-0-0 snd-mixer-oss
alias sound-service-0-1 snd-seq-oss
alias sound-service-0-3 snd-pcm-oss
alias sound-service-0-8 snd-seq-oss
alias sound-service-0-12 snd-pcm-oss
alias snd-card-0 snd-intel8x0
alias sound-slot-0 snd-card-0
# ua-1a
alias sound-service-1-0 snd-mixer-oss
alias sound-service-1-1 snd-seq-oss
alias sound-service-1-3 snd-pcm-oss
alias sound-service-1-8 snd-seq-oss
alias sound-service-1-12 snd-pcm-oss
alias snd-card-1 snd-usb-audio
alias sound-slot-1 snd-card-1
# virtual midi 1
alias snd-card-2 snd-virmidi
alias sound-slot-2 snd-card-2
# virtual midi 2
alias snd-card-3 snd-virmidi
alias sound-slot-3 snd-card-3
#evolution 1
alias snd-card-4 snd-usb-midi
alias sound-slot-5 snd-card-5
#evolution 2
alias snd-card-5 snd-usb-midi
alias sound-slot-5 snd-card-5
--
peace, love & harmony
Atte
http://www.atte.dk
http://www.notam02.no/arkiv/src/radium-0.63.tar.bz2http://www.notam02.no/radium/
(Note, the CVS is very outdated)
Radium V0.63 Alpha Linux Port
Released 15.4.2004
HOW TO MAKE IT RUN WITHOUT READING THE REST OF THE README FILE
make
./start.sh
INTRODUCTION
This is the second public, and the first official, release of
the linux port of the music editor Radium. The Amiga version
has been stable since January 2002.
Radium is a new type of graphical (currently only) notebased
music editor.
For more info about the program, check out the online manual placed
at http://www.notam02.no/radium/ somewhere.
COMPILE
The following packages are currently used to build and run Radium:
*libgc - http://www.hpl.hp.com/personal/Hans_Boehm/gc/
*Python - www.python.org (at least V2.0 and tkinter)
*Python development package
*pyqt - http://www.gnu.org/directory/devel/specific/PyQT.html
*pygtk - http://www.daa.com.au/~james/software/pygtk/ (pygtk1, use pygtk-0.6.11)
*xterm - Yupp.
Most of these things are allready included in the radium/ directory (mainly because
of compatibility problems between various versions of gtk, pygtk and
libglade), so you only need to get hold of "pyqt".
"pyqt" is available as package for most linux distributions.
CURRENT STATE
The biggest problem is that saving does not work. Loading does though.
Other things that needs to be fixed is that the playing behaves
strange when reaching the end of the blocks, and that the alsa-seq code
needs some care (midi/alsaseq/).
ABOUT THE CODE
The code is a mix between C and Python. The state of some of the code is
quite horrible and ununderstandable, but relatively bug-free, eh
hopefully. :)
CONTACT
k.s.matheussen(a)notam02.no
http://www.notam02.no/radium/
--