Hi all,
I seem to have put myself in a brilliant position... My flight lands at
22:30 in Milan Malpensa airport, meaning that I can't make the last
train from Milan to Parma.
I have two possibilities: either stay a night in Milan and take the
morning train, or rent a car in Malpensa, drive it to Parma and drop of
it there the next morning. I've done the accounting for both scenario's,
and the first one would come down to 100eur, while the car rental would
set me back about 120 eur. Therefore I'm probably going to rent the car
as it saves me a lot of hassle.
But before booking it, I'm wondering if there are other people that
might have the same issue. If so, a split fee would make things even
more interesting (at least for me ;).
Greets,
Pieter
On Sunday 12 April 2009 01:10:09 you wrote:
> Andrew,
>
> > > > To have some reserve in volume I set amplifier's volume regulator
> > > > higher than, say, is needed at current moment. As a result I have
> > > > software level noticeably below 100%. I suppose such software volume
> > > > regulating eats samples resolution (and, at least theoretically,
> > > > sometimes I have, say, 12-13 bits rather 16).
> > >
> > > If you use the hdsp's matrix mixer, it will render the volume with
> > > higher precision than 16 bits, so play back unattenuated and control
> > > the output level in the matrix mixer.
> >
> > How to deal with the mixer (output channels) via CLI?
>
> amixer.
>
>
> Flo
Florian,
Yes, I see amixer must be used. The proble is, hdsp driver maps card's mixer
controls to alsa mixer controls in not-obvious way.There was somewhere some
information about accessing hdsp mixer controls (how to relate channels and
appropriate controls, write (and read?) values and such), but I can not found
it now (probably it was at old alsa site).
The card has
numid=5,iface=HWDEP,name='Mixer'
control, and, probably, something like
amixer cset numid=5 x,y,z
must be used. What are those x, y, z? :-)
OTOH, it seems like I can not get current hdsp mixer settings: 'amixer cget
numid=5' just returns a tuple with three zero rather full 3dim matrix. Can I?
Has anybody elaborated information about all these issues?
Andrew
-------------------------------------------------------
Hi!
For "lirc-ized" playing back I use mpd CLI client. As far as hdsp driver
hasn't common "Master Volume" (or such) alsa control, I use mpd's "software"
volume leveling, so lirc commands are directed to CLI mpd client.
To have some reserve in volume I set amplifier's volume regulator higher than,
say, is needed at current moment. As a result I have software level noticeably
below 100%. I suppose such software volume regulating eats samples resolution
(and, at least theoretically, sometimes I have, say, 12-13 bits rather 16).
Am I right?
If "yes" - are there ways to use lirc to set a volume level more directly
communicating with hdsp driver via CLI? Will such way keep sample accuracy?
Chris, no problem for you. I'm talking and asking about performance
timing at data level across multiple apps, not the port structure
itself. Jackmidi does show alsa ports when it's in seq, so no problem
there aside from the renaming challenge, in qjackctl at least.
So you can keep right on enjoying yourself. :)
Alex.
On Sat, Apr 11, 2009 at 11:28 PM, Christopher Cherrett
<stuff(a)trackingsolutions.ca> wrote:
> I thought that the newest jack will create jack midi ports out of alsa midi
> ports for you if the driver was set to seq.
>
> I really hope this does not pose a problem for me as I am almost finished
> putting in the interface for program and bank changes on the fly. The only
> thing left to do is finish the new ruler that will visually indicate where
> the program changes are.
>
> Are you running program changes on multiple tracks? Or is this just
> happening on one track?
>
> -------- Original Message --------
> Subject: [Rosegarden-user] timing penalty?
> From: alex stone <compose59(a)gmail.com>
> To: rosegarden-user <rosegarden-user(a)lists.sourceforge.net>, LAU Mail List
> <linux-audio-user(a)lists.linuxaudio.org>
> Date: 04/11/09 12:30
>>
>> I've been conducting a couple of experiments related to timing, and in
>> particular a rapid program change sequence, between Rosegarden, using
>> Alsamidi, through A2jmidid, to Linuxsampler.
>>
>> 1.) When i use alsamidi in both RG and LS, timing is reasonable, but i
>> get the occasional blip if too many PChanges happen in quick
>> succession. (And as a relative scale, we're talking about a sequence
>> of 1/64 notes, with a PC for each one, and additional volume CC
>> changes as well, at about 140 bpm.)
>>
>> 2.) When i use alsamidi in RG, and a2jmidid through to LS jackmidi,
>> timing has improved, but still the occasional missed PC.
>>
>>
>> So using a2jmidid seems to be a step forward, but my question is, is
>> there a 'penalty' of some sort for using it in bridging format, and if
>> so, what is it?
>>
>>
>> 3.) When i use jackmidi native apps, running Muse and LS, for
>> example, the timing is good, and i don't seem to miss fewer PChanges.
>>
>> (Which doesn't mean i'm giving up on RG. It has too many essential
>> tools that help me workflow wise.)
>>
>>
>> Final Question.
>> If we use an RT kernel (which i am, being 2.6.29-RT in 64studio 3.0),
>> with 1024HZ (or higher) for timing, what can we do after that, in
>> jack, a2jmidid, RG, LS, etc.., to ensure that midi timing (Alsa or
>> Jack) is pristine, and the combination of apps has the best change of
>> coping with rapid, and high volume, midi data being sent?
>>
>> I'm trying to finesse the midi timing and midi data management across
>> the board, and get it as good as possible. (It's already good. This is
>> about the....extra 10%.)
>>
>> I'm definitely in favour of using jackmidi based on my modest and
>> subjective experience, but i'd like to stick with RG for the power of
>> it's toolset for workflow. (Plus a thousand other good reasons to use
>> it.)
>>
>> So if anyone has ideas, and tips, on raising the midi data flow
>> precision in transistion between the above apps, and tips or compile
>> flags to do this at build time, or runtime, i'd appreciate some
>> insight.
>>
>> I've applied the usual tweaks across all the apps, and have Jack
>> running at 128/2/48000, with no xruns during normal operation.
>>
>> Alex.
>>
>> 64studio 64bit 3.0 (beta 3), on an AMD64 5600+ dual core, with 4GB ram.
>> M-Audio Delta 44 soundcard.
>>
>>
>> Parchment Studios (It started as a joke...)
>>
>>
>> ------------------------------------------------------------------------------
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>> https://lists.sourceforge.net/lists/listinfo/rosegarden-user
>>
>
>
>
--
Parchment Studios (It started as a joke...)
I've been conducting a couple of experiments related to timing, and in
particular a rapid program change sequence, between Rosegarden, using
Alsamidi, through A2jmidid, to Linuxsampler.
1.) When i use alsamidi in both RG and LS, timing is reasonable, but i
get the occasional blip if too many PChanges happen in quick
succession. (And as a relative scale, we're talking about a sequence
of 1/64 notes, with a PC for each one, and additional volume CC
changes as well, at about 140 bpm.)
2.) When i use alsamidi in RG, and a2jmidid through to LS jackmidi,
timing has improved, but still the occasional missed PC.
So using a2jmidid seems to be a step forward, but my question is, is
there a 'penalty' of some sort for using it in bridging format, and if
so, what is it?
3.) When i use jackmidi native apps, running Muse and LS, for
example, the timing is good, and i don't seem to miss fewer PChanges.
(Which doesn't mean i'm giving up on RG. It has too many essential
tools that help me workflow wise.)
Final Question.
If we use an RT kernel (which i am, being 2.6.29-RT in 64studio 3.0),
with 1024HZ (or higher) for timing, what can we do after that, in
jack, a2jmidid, RG, LS, etc.., to ensure that midi timing (Alsa or
Jack) is pristine, and the combination of apps has the best change of
coping with rapid, and high volume, midi data being sent?
I'm trying to finesse the midi timing and midi data management across
the board, and get it as good as possible. (It's already good. This is
about the....extra 10%.)
I'm definitely in favour of using jackmidi based on my modest and
subjective experience, but i'd like to stick with RG for the power of
it's toolset for workflow. (Plus a thousand other good reasons to use
it.)
So if anyone has ideas, and tips, on raising the midi data flow
precision in transistion between the above apps, and tips or compile
flags to do this at build time, or runtime, i'd appreciate some
insight.
I've applied the usual tweaks across all the apps, and have Jack
running at 128/2/48000, with no xruns during normal operation.
Alex.
64studio 64bit 3.0 (beta 3), on an AMD64 5600+ dual core, with 4GB ram.
M-Audio Delta 44 soundcard.
Parchment Studios (It started as a joke...)
Hi,
I tried to compile Xjadeo on Debian, but I got:
xjadeo 0.4.6 - Configuration Report
------------------------------------------------------------------------
prefix: /usr/local
exec-prefix: ${prefix}
install path: ${exec_prefix}/bin
System wide config: ${prefix}/etc/xjadeorc
Video Displays
- libxv: ok
- libx11/imlib2: not avail.
MIDI: alsa
LASH: ok
POSIX msg queue: ok
On screen display:
- freetype: ok
- default font: -
GUI: QT4
build contib/*: no.
now type 'make'; followed by 'make install' as root.
d@debianstudio32:~/2home/xjadeo-0.4.6$ make
make all-recursive
make[1]: Map '/home/d/2home/xjadeo-0.4.6' wordt binnengegaan
Making all in src
make[2]: Map '/home/d/2home/xjadeo-0.4.6/src' wordt binnengegaan
Making all in xjadeo
make[3]: Map '/home/d/2home/xjadeo-0.4.6/src/xjadeo' wordt binnengegaan
make all-am
make[4]: Map '/home/d/2home/xjadeo-0.4.6/src/xjadeo' wordt binnengegaan
gcc -DHAVE_CONFIG_H -I. -I../.. -Wall -g -O3 -I/usr/include/ffmpeg
-I/usr/include/ffmpeg//libavcodec -I/usr/include/ffmpeg//libavformat
-I/usr/include/ffmpeg//libavutil -I/usr/include/ffmpeg//libswscale
-I/usr/include/freetype2 -I/usr/include/lash-1.0 -I/usr/include/alsa
"-DSUBVERSION=\"exported\"" -g -O2 -MT xjadeo-xjadeo.o -MD -MP -MF
.deps/xjadeo-xjadeo.Tpo -c -o xjadeo-xjadeo.o `test -f 'xjadeo.c' ||
echo './'`xjadeo.c
xjadeo.c: In function ‘render_empty_frame’:
xjadeo.c:451: error: ‘PIX_FMT_RGBA32’ undeclared (first use in this
function)
xjadeo.c:451: error: (Each undeclared identifier is reported only once
xjadeo.c:451: error: for each function it appears in.)
xjadeo.c: In function ‘display_frame’:
xjadeo.c:672: warning: implicit declaration of function ‘img_convert’
make[4]: *** [xjadeo-xjadeo.o] Fout 1
make[4]: Map '/home/d/2home/xjadeo-0.4.6/src/xjadeo' wordt verlaten
make[3]: *** [all] Fout 2
make[3]: Map '/home/d/2home/xjadeo-0.4.6/src/xjadeo' wordt verlaten
make[2]: *** [all-recursive] Fout 1
make[2]: Map '/home/d/2home/xjadeo-0.4.6/src' wordt verlaten
make[1]: *** [all-recursive] Fout 1
make[1]: Map '/home/d/2home/xjadeo-0.4.6' wordt verlaten
make: *** [all] Fout 2
Help would be nice.
\r
Hi There,
Here in Turkey, music technology education is mostly associated with
recording,
mixing and other studio applications. As I get from those mail lists,
outside Turkey
music technology (at the universities of course) is mostly interested in
creating the
technology rather using it.
There are 4 or 5 music technology departments at different universities here
in Turkey.
It seems like the philosophy is not grabbed in a correct way because all has
dissimilar course programs.
Only my department gives courses like Csound, Programming Languages, Sound
Synthesis
etc. along with studio applications. Rest of the departments just focus on
studio stuff.
So, I need to publish a paper about music technology formation all around
the world which
I can adapt it here in Turkey.
I just need your help in that way:
1) Personal ideas about my problem above.
2) The course programs (or just the names of the courses) of the departments
that you're working at.
3) The qualification requirements of the student candidates.
Although it looks like this mail mostly concerns the university men,
everybody is welcome.
If any document files need to be attached you can mail me to
ardaeden(a)gmail.com
Thanks for your help.
--
Arda EDEN
Cumhuriyet University
Faculty of Fine Arts
Department of Music Technology
Sivas/TURKEY
Hi,
I want to try kenvy24control, http://kenvy24.wiki.sourceforge.net/
but I can't install it:
kenvy24-1.2-src/build$ cmake
..
CMake Error at /usr/share/cmake-2.6/Modules/FindKDE4.cmake:84
(MESSAGE):
ERROR: cmake/modules/FindKDE4Internal.cmake not found
in
/home/d/.kde/share/apps;/usr/share/kde4/apps
Call Stack (most recent call
first):
CMakeLists.txt:5
(find_package)
CMake Warning (dev) in CMakeLists.txt:
No cmake_minimum_required command is present. A line of code such as
cmake_minimum_required(VERSION 2.6)
should be added at the top of the file. The version specified may be
lower
if you wish to support older CMake versions for this project. For more
information run "cmake --help-policy CMP0000".
This warning is for project developers. Use -Wno-dev to suppress it.
-- Configuring incomplete, errors occurred!
I've cmake 2.6 installed..
\r
Hi,
I want to record electric guitar chord progression. I've an:
- epiphone les paul
- little amp, orange 30r
- maudio dmp 3 preamp
- shure 58a beta
- mic stand
What do you prefer, preamp of mic in front of amp?
How do you manage the gain and eq?
- guitar
- amp
- preamp
- ardour
How do you mix it?
I've read that it is useful to use an low-pass filter, some compression
and EQ.
What are good plugins on GNU/Linux for this?
What goes pre- and what postfader?
Thanks in advance,
\r