On 13 Feb 2003 01:38:13 +1100
Son of Zev <sonofzev(a)labyrinth.net.au> wrote:
> However I have been involved in the ardour lists for over 2 years. I
> have spent much time reading about potential problems and responded to
> all those that have had the courtesy to respond. I have also learnt and
> spent much time investigating problems that have nothing to do with the
> production of music.. simply to try and help developers iron out
> problems..
Pretty much everyone into linux audio at this point does that.
> If you look back into archives a couple of years back alot of
> development into inclusion of MIDI sync was added to some softwares
> (especially MusE) to accomodate users like myself who depend on this
> sync to make software compatible with real studios.
>
> BUT. In the last few weeks I have spent much time into trying to
> configure and compile ardour (the potentially greatest audio recording
> software available to us)
Heh, I've been up to the same task. It seems to depend on fairly recent development tools, and upgrading to them all can mess things up. But slowly working towards it.
> but all my responses after I sent an opinion
> to the string "ahem" concerning Steinberg having produced (but not
> released) versions of their commercial softwares under our much loved
> platform, have been completely ignored.. except this one..
>
> My responses have not been ignorant nor lacking in information. As I
> mentioned I have spent much time learning about the Linux platform and
> contributing when I can ...
Well, position yourself with a working ardour, and then think how much time you would spend on helping others, when all that time you could be making music :)
> My annoyance is when it was asked of me to provdie more information and
> I did .. no response came.. then when I tried to resolve it myself and
> no further response came that's when I could only put 2 & 2 together and
> see that a response I made to the string "ahem" could have been related
Sometimes more information does ring a bell to someone, sometimes it doesn't. That's pretty much about it from my point of view. There are many bits in cyberspace.
janne
(or is there? ;)
> cheers
>
> Allan
I run Linux 2.4 / SUSE 8.1 pro on a Duron 1300 - PC with a Terratec EWX 24/96
- PCI - Audiocard.
Yast2 installs Kysela`s Envy24-Control - Mixer for this Card and it works well
for playing Wav/Ogg/MPEG-Audiostreams etc.
BUT: to set the Inputlevel properly is nearly impossible.
If a soundmessage from Bash occurs the input is distorted and set to SW-Radio
- "Quality", same with other Systemmessages.
To get an undistorted Signal i have to set the Input extremely low, if it is
set to the same volume as the internal output (from XMMS etc) it is extremely
distorted and/or i get the SW-Radiosound as described above.
here my questions:
1) is there an alternative to alsa/envy24-control that works ok with EWX ?
2) is there a patch for envy24 as RPM ?
thanks
--
HZN/Berlin/Fh
http:www.linuxuse.de
> ...But even 30 MB/sec should be far enough to get
> dropout-free recording.
> 44100 * 16 * 2 (bit/sec) = 1411200 bit/s
> 44100*16*2/8 = 176400 byte/s = around 200 kB/sec.
> See? Even if you play at the same time (duplex), your
> 30 MB/sec is far more than that.
> One more thing, but be aware, I heard switching the
> dma-transfer-rate might be dangerous
> if you don't have the new 80pin cables. (right?)
Yes, I definatly have the cables.
> you might want to check. "hdparm -i" which gives you
> some "*" on the options
> that are set. see the "dma1 dma2 dma3 dma4 dma5" line,
> which (in my case has the "*" at the dma4=100MB/secautomatically set.)
Here's what doesn't make sense to me. The Web Site that you referred
me to says:
"Ultra DMA modes, such as UDMA2, 4, and 5, are also referred to as
ATA/33, ATA/66, and ATA/100, respectively."
Now, the term "respectivly" means that mode 5 refers to ATA/100.
When I look at the output of "dmesg" I see the line:
"SiS730 ATA 100 controller"
And yet the output of "hdparm -i /dev/hda" lists"
Model=Maxtor 6Y080P0, FwRev=YAR41VW0, SerialNo=Y30HJLPE
Config={ Fixed }
RawCHS=16383/16/63, TrkSize=0, SectSize=0, ECCbytes=57
BuffType=DualPortCache, BuffSize=7936kB, MaxMultSect=16, MultSect=16
CurCHS=16383/16/63, CurSects=16514064, LBA=yes, LBAsects=160086528
IORDY=on/off, tPIO={min:120,w/IORDY:120}, tDMA={min:120,rec:120}
PIO modes: pio0 pio1 pio2 pio3 pio4
DMA modes: mdma0 mdma1 mdma2
UDMA modes: udma0 udma1 udma2
AdvancedPM=yes: disabled (255) WriteCache=enabled
Drive conforms to: (null): 1 2 3 4 5 6 7
There's not even a listing for udma5.
Since I keep reading all these warnings
about not trying settings that aren't supported,
should I try udma5?
Just in case I missed something (and I've been reading everything
can get my hands on for weeks now) here some info from my Mother
Board Manual:
The mainboard delivers high-level performance with support for...
<SNIP>
... and two PCI Bus Master Ultra DMA (UDMA) ports that support
up to four ATAPI devices. The PCI IDE also supports PIO Modes
3 and 4, UDMA33/66/100 IDE and an ATAPI CD-ROM
And further into the Mother Board Manual it says:
The built-in fast PCI IDE controller supports ATA Programmed
Input/Output (PIO) and Direct Memory Access (DMA) mode.
The Ultra DMA33/66/100 feature delivers data transfer rates
up to 100 MB/s by providing a separate data path for two
IDE channels and improving multi-tasking performance
Thanks
Rocco
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Hi all,
(Tried getting some help at the LAD list but so far I got no response,
hopefully I can get some input from you guys :-) -- for those who are
subscribed to both lists, my apologies for cross-posting)
I am in the process of revamping my app as well as adding more features.
The problem my app currently has is that the main window is not
resizable, which may potentially cause problems on displays with limited
resolution. I've decided to fix this issue once and for all but am not
sure which path to take. The app in question is RTMix and its current
layout is split into two parts:
Top with timers and buttons
Bottom with the main notification widget and bunch of tabs for editor
and other functionalities.
The app looks something like this:
_________
|buttons|
| timer |
_________
| tabs |
|widgets|
_________
Now I am ready to begin working on a resizable version (which should
mostly not be a problem, other than being time-consuming) and am
wondering whether I should separate the two widgets all together and
have them therefore fully customizable (size-wise). While this option
seems very easy to implement and also attractive, I am concerned with
the issue that the app will easily lose its default looks and may
quickly become overwhelming for the first-time user (since it is
designed to provide performer-computer interface with the least amount
of familiarization required). I would especially appreciate comments
from people who got to use RTMix as to what would they like to see, but
of course I would appreciate anyone else's opinions on this matter as
well.
The other option I've been thinking is to move the two widgets
side-by-side, rather than having one over the other, which would
certainly make maximizing window easier and would utilize the desktop
space more efficiently. However, the issue is also that the app has a
bunch of other deployable floating sub-windows and as such this may not
prove to be the best solution, and would probably look pretty useless
when having a relatively small window.
Finally, I can simply leave the app as it is but then the notification
interface (the bottom part) as well as other tabbed widgets of the
bottom part (such as the script editor) will not be as efficient in
utilizing the desktop space as they could possibly be.
I also thought of having the two widgets designed in such a way that
they can be "glued" together (i.e. xmms-like) or separated as needed,
but I am just afraid that this option may take too much effort since
then I would have to get rid of the default window decorations (in order
to make them "gluable") and make my own custom-designed window handles,
which seems too much of a nuisance to implement.
I am sure someone might have yet another idea regarding this issue that
I am simply unable to think of so I would greatly appreciate your input
on this matter.
The bottom line is that I am looking to make this app as flexible in
terms of desktop space utilization as possible while sacrificing the
minimum amounts of the "standardization" it currently has to offer.
Your help is greatly appreciated! BTW I am using Qt (if that becomes
relevant at some point).
P.S. I have a bunch of screenshots on my site, but the network is
currently screwed up on the University campus so it may end up not being
available for the next 8-24 hrs. Hence, I am unable to give you the
screenshots. However, you may wanna try the URL in my sig -- who knows,
maybe the network will be up by the time this hits your inboxes :-).
Again thanks for your help! Sincerely,
Ivica Ico Bukvic, composer, multimedia sculptor,
programmer, webmaster & computer consultant
http://meowing.ccm.uc.edu/~ico
============================
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"To be is to do" - Socrates
"To do is to be" - Sartre
"Do be do be do" - Sinatra
"2b || ! 2b" - ?
"I am" - God
I have an onboard cmipci without a midi port. However, aconnect always
shows the mpu401 that the chipset supports. Is there a module option
to disable this so i don't see it?
--ant
Here is a new release of jMax (4.0.1)
This release includes a lots of bugfixes and several new features.
You can download it from:
http://sourceforge.net/project/showfiles.php?group_id=26076&release_id=1391…
Changes:
- new audio configuration panel (Project->Edit Current Configuration)
- "Save As Default" menu for project and Audio/MIDI configuration
- templates are supported
- in~/out~ take an audiolabel name as argument
- C++ and Python client library under development
- clean sequence package
- detection of available audio devices
- several bug fixes
- remove obsolete documentation
- help patches are not complete
- new input~ object
- new fvec/ivec editors
Linux:
- user project is $HOME/.jmax/default-project.jrpj
- user configuration is $HOME/.jmax/default-config.jcfg
- detection of available Jack Audio Port (in Jack Audio Connection Kit
support package)
- connection with Jack Audio Port inside jMax (new audio configuration
panel)
Installation on Linux:
- the rpm have been built on RedHat 7.3
- we have split jMax into several RPMs:
-- jmax-4.0.1-1.i386.rpm : contains FTS server, packages and Java FTS
client
-- jmax-client-4.0.1-1.i386.rpm : contains only C++ and Python FTS
client
-- jmax-common-4.0.1-1.i386.rpm : contains only FTS server and packages
-- jmax-devel-4.0.1-1.i386.rpm : contains only FTS header and packages
header
-- jmax-jack-4.0.1-1.i386.rpm : contains only Jack Audio Connection Kit
support package
-- jmax-java-4.0.1-1.i386.rpm : contains only Java FTS client
Linux source compilation:
- do the following shell commands:
$ tar xvzf jmax-4.0.1.tar.gz
$ cd jmax-4.0.1
$ ./configure
$ make
$ make install
- if you want to compile with Jack Audio Connection Kit support, add --enable-jack when calling configure
$./configure --enable-jack for compilation with Jack Audio Connection Kit support
jMax sourceforge home page:
http://sourceforge.net/projects/jmax/
--
Patrice Tisserand <Patrice.Tisserand(a)ircam.fr>
IRCAM
Hi,
Maybe someone could hit me with the clue stick.
Please.
I have generated a data disc, comprised of audio
files, using; 'mkisofs -o $FILE.iso $FILE | cdrecord
-v fs=6m speed=4 dev=0,6,0 /home/tmp/$FILE.iso'.
When these disks are inserted into the CD drives on
Macintosh computers, they either fail to be recognized
immediately or they'll be partially read but
ultimately fail.
Could someone please give me a clue as to what I need
to do to produce a data disk/file system that
Macintosh computers can read. Preferably using mkisofs
and cdrecord.
Thanks,
ron
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Hi all,
New stuff. Good.
* you can now use plugins with unequal numbers of input and output ports.
any excess ports get exposed as extra jack ports with names like
"amplif_1-1_i2", of the form
"<shortened plugin name>_<rack position>-<plugin copy>_{i,o}<port>"
* wet/dry controls for the output of each plugin
* logarithmic controls
* much more resilient file loader
http://pkl.net/~node/jack-rack.html
Bob
--
Bob Ham <rah(a)bash.sh>
Hello!
I connect my active guitar directly to my soundcard line in plug
and get a great clean guitar sound. This way I can easily apply
nice effects to the guitar sound.
I haven't found a really great distortion effect
though. Does anybody know where to get one? On
http://plugin.org.uk/ladspa-swh/docs/ladspa-swh.html and
http://www.ladspa.org/cmt there are some distortion effects, but
I can't manage to get a great sound with them. Perhaps somebody
knows the proper plugin parameters to use?
The sound may not be really great because I don't use any amplifier
or box simulation. How exactly do these simulations work? Where can
I get them for Linux? If at all possible in form of a LADSPA plugin.
Thanks for your help.
ra
When my system first boots, alls the sound modules show (from lsmod) status
of (autoclean). What causes this? I can run 'alsasound stop' and then
'alsasound start' and they are loaded fine and sound works. Also, the mixer
module alsways seems to show as (autoclean) and 'unused', and I cannot use
the graphical mixer to set levels etc. Any suggestions? Thanks!
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