Maluvia spoke in tongues:
> One of our heros - and actually, we do that too.
> We started doing that when we switched to Zaolla cables since they are
> designed to have a preferred direction for the signal flow, (but they no
> longer put the arrows on the cables).
How, then, does the sound know which direction to go?
You're not worried about rectification or picking up AM radio, are you?
If I had cables with the arrows, how much would it cost to upgrade to
cables without the arrows? Is there a kit? How would you characterise
the difference between the sound of the older and newer cables?
If I were to build a suspension bridge out of such cables, provided they
were aligned correctly, would anyone be interested in purchasing one? (I
bet it would sound great!)
Just asking, of course.
Bristol Audio has been re-released, version 0.9.1 and is available on
http://sourceforge.net/projects/bristol
Bristol emulates several synthesisers and organs:
Moog Mini
Moog Voyager
Yamaha DX-7
Roland Juno 6
Rhodes Stage-73
Hammond B3
Vox Continental
Sequential Circuits Prophet-5/52
Sequential Circuits Prophet-10
Oberheim OBX
Another Oberheim (OBX-a) and a mixer are under development. The application
emulates the synthesiser algorithms of the original instruments and gives a
representation of their layout. It is currently standalone but work is
underway for Jack integration. All emulations can be played simultaneously
by the multitimbral engine and can be driven from a midi keyboard or ALSA
midi sequencer.
Bugs and feature requests to the author or via sourceforge.
There are known issues with the Oberheims so mileage may vary. There are
known issues with note off in layered emulations on a single midi channel.
The mixer only implements the graphical user interface, there is no engine
algorithm behind it and it can best be started with the '-libtest' option.
Nick Copeland
nickycopeland(a)hotmail.com
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Well, it's not much, as I seem to be moving from Sonar to Linux-based
stuff at a glacial pace - by no means due to problems with Linux - all
that's gone swimmingly - I just have too many unfinished products
laying around on the windows system - and for both aesthetic and
technical reasons i refuse to switch horses mid-record.
But anyway, here's a minute and a half of music I made for use in a
short film by a friend. Equipment used: a Shure SM58 and a Kawai K4r
run via cheap Behringer mixer into an M-Audio Delta 44 connected to a
p2.4g/512mb CCRMA box running Freewheeling - output to a windows box
running Sonar (gratuitous windows use due to increasingly antiquated
workflow), which I used only for a fade-out at the end (two hours
practice with Ardour would likely reduce my need for windows to nil.
Unfortunately, I'm spending those two hours learning how to play
drums.)
http://towndowner.com/audio/etc/theme-from-some-assembly-required.mp3
comments either on/off-list appreciated.
--
dan(a)towndowner.com dan(a)burntpossum.com daneasley(a)gmail.com
http://towndowner.comhttp://burntpossum.com
Hi all,
After a long while, I am proud to announce that there is a new release of
WONDER -
Wave field synthesis Of New Dimensions of Electronic music in
Realtime
It is designed to provide an interface between a wave field synthesis
system and the software or hardware composers and performers of
electronic music are used to.
The current version (v2.1.0) allows the user to either create a
composition for the movement of the sound sources in his composition or
to control the movements in realtime, either via the gui or via open
sound control.
More information and download is available on:
http://www.kgw.tu-berlin.de/~baalman/
or
http://swonder.sourceforge.net
This release finally concludes the restructuring of the program, I
started well over a year ago (and talked about at the last LAC), and the
program should run stable now, having done several tests myself and
thoroughly using valgrind to check for memory leaks. :)
I created an email list for users of the program, and invite anyone who
is interested to subcribe at, to keep up-to-date on developments or to
ask questions about the use of the program:
http://lists.sourceforge.net/lists/listinfo/swonder-list
Yours sincerely,
Marije Baalman
>good grief, i never suggested that you *leave*. stop being so absurdly
>personal and . . .
Telling someone that they are either 'completely ignorant or deliberately
ridiculous' is getting *personal*, Paul.
You are being disingenous.
>your complaints are not specific to open source digital audio, but are
>about about digital audio in general.
> . . . . . . .
>if you want to discuss audio
>fidelity and how to improve it, you would **be better off** in a forum
>focused on such matters.
And why is it I would be ''better off' in such a forum?
I see all kinds of topics discussed here that are not specific to Linux or
open-source (or even music).
The thread about Quiet PCs, or guitar levels are good examples recently.
I didn't see you telling any of those people that they would be 'better
off' in some more 'general' forum to do with computer recording, audio
tweaks, or whatever.
You are being hypocritical here.
In fact, this is a good place to mention something I have noticed for a
long time on these lists:
There seems to be an attitude prevalent among some of the long-time members
here of - shall we say - 'impatience' with newcomers - especially those who
are not of the programmer variety.
I have seen newcomers post on these lists (I mean here all 4 of them), with
great enthusiasm and excitement - happy to have discovered Linux Audio or
Ardour, full of eager, albeit 'newbie', questions or problems, and have
seen a number of them treated with irritation, sometimes bordering on open
contempt and disdain.
I honestly don't think you realize how you often sound to others - and
since I am already in disfavor, I have nothing to lose by bringing it up
now.
This is no way to build a healthy and happy community.
I would think you would be anxious to bring more people into this community
- both as testers, and also potential code contributors, and would endeavor
to make them feel more welcome, and not subject them to ridicule for their
inexperience or lack of knowledge.
Bringing more people into the community brings with it the potential for
more donations and more help with development, as well as more
word-of-mouth.
I can understand why you programmer veterans don't welcome those kind of
posts on the developer lists - in which case it would be best to tactfully
and good-naturedly redirect them to the user lists.
But I have also seen people treated with unneccessary brusqueness on the
user lists - and that should not be happening.
At least it was my impression that the user lists were intended to be much
more open both in topics, and in skill level of participants.
I think it would be extremely beneficial for all concerned if the veterans
here tried to create a more friendly and casual atmosphere - a more 'open'
atmosphere - perhaps try working on your interpersonal communication skills
a bit.
If you are going to tell me that you are all paragons of graciousness, I
can only say you don't have much objectivity.
Some of you are - some are definitely not.
I have seen newcomers come here happy and eager, treated brusquely - and
they have not returned.
That is a real shame - unless you just want the linux audio community to be
a closed clique of code-hackers and engineers and are not interested in
reaching out to the greater musician community.
>why is that *nobody* who makes these kinds of audiophile-esque claims is
>ever willing to do the leg work? when the rest of the measurement
>industry switched to double blind testing 40-50 years ago for everything
>else, why is it that hearing is somehow exempt?
First, I do not subscribe to any "audiophile-esqe" philosophies - that is
just more tired dogma.
In fact I find some of it to be silly, extreme and often irrational.
When I have come across 'audiophile' ideas, or *any* suggestions in the way
of audio tweaking that I find interesting or plausible - I simply test them
out when possible.
In fact, one recent attempt to implement one of these ideas, gleaned from
tweaks suggested by the Mapleshade Records site, involved buying and
applying a special silver paste contact enhancer (a lot like the thermal
paste you use on your heatsink but optimized for audio rather than thermal
conductivity).
(Called SilClear: http://mapleshaderecords.com/audioproducts/silclear.php,
if any of you are actually interested in trying it out.)
Suffice it to say this experiment did not go well for us - we think we
trashed some very expensive cables and a good preamp trying to use it.
Maybe we just applied it wrong - maybe it didn't agree with our equipment -
maybe it just doesn't work.
I don't believe we will be repeating this experiment however, as we cannot
afford to keep replacing expensive equipment.
I cannot say whether this stuff works or not, but it didn't work out for
us.
Apparently, though there are many people who swear by it. Are they wrong?
If it works for them, and they are happy with it - great!
[Incidently this site:
http://mapleshaderecords.com/audioproducts/index.php, has tons of ideas and
products w/re to audio tweaking in the form of vibration control devices,
racks, and stands, high-quality cabling, contact enhancers and CD
treatments.
Some of you might actually find it interesting or useful.]
I find it quite plausible that they may yield improvements in audio
quality, but as they are quite expensive we can't afford to try them out.
I am just keeping an open mind concerning their ideas.
>why not try the experience of a double blind test? you haven't suggested
>any test, any experience that any of us can try that would test your
>claims. several of us have suggested to you that a double blind test
>would test them and would provide you with a new experience that in your
>words might be inconsistent with your current belief system.
I would ask you: why you are so insistent that I need to 'prove my claim'?
How on earth did my simple comments about what *I* hear stir up so many
bees in so many bonnets?
I merely mentioned in passing that I hear those subtle differences on the
way to trying to explain why I think bit-depth matters - this was after
being told that I was not really hearing differences in bit-depth, but
rather just differences in quality of electronics, or basing my perceptions
of quality on some subjective bias rather than what I heard - which I even
acknowledged was quite possible.
I am not out to convince you or anyone else that I hear what I do.
I regret now even mentioning it - since it has elicited such an outraged
response.
Do you think I have some desire to convince you all of what I hear - that I
have some agenda to convert people to agreeing with my perceptions, or some
big theory to prove?
I do not.
>and why do they put so
>much effort into avoiding this simple test (which could be done in your
>home with perhaps 30 minutes of set up time) .
Perhaps for the same reason you have not subjected your preference for
Apogee converters to a double-blind test.
You trust what you hear, you like your equipment and are happy to continue
using it.
You apparently feel no compulsion to prove to yourself that your
perceptions are correct.
I am already satisfied that I am hearing what I hear - and my husband hears
it as well, btw.
We do conduct single-blind experiments all the time both with regard to the
issue being debated here, and with regard to different mixes we are
evaluating.
(I even try to do it sort-of double-blind by trying to hide or mix up which
mixes are which, and we test each other out as well - I'll play the tracks
while he listens - not telling him which is which, then we rearrange their
order, and he plays them back for me.)
And amazingly - to me - more often than not we hear the same things and
agree.
But I have also come to recognize a behavior we both have - which may be
universal - where there is an *initial* tendency to think that anything
that sounds different sounds better, or whatever sounds louder sounds
'better'. (Kind of like the Pepsi challenge thing, I guess.)
But we are well aware of this tendency, and you can get past it and train
yourself to listen more carefully - to specific frequency bands, for
example, or specific effect paramaters, etc. etc. and eventually get to a
more objective headspace with your listening.
Certainly part of the challenge of the art and science of being an audio
engineer.
(I hope the above comment is not going to be taken as another 'claim' which
I need to 'prove'.)
Seriously, can I make an observation regarding my views and perceptions
without being bombarded with a fusillade of indignant demands that I must
'prove my claim'?
There *is* a difference between making an observation or voicing an
opinion, versus claiming something to be an indisputable fact which then
needs to be subjected to rigorous scientific or legal inquiry.
Perhaps we will try the double-blind test someday, I don't know.
Right now we're much more interested in getting a good-sounding mix on the
guitar, fine-tuning our patches, and on the purely creative and
compositional aspects of the music.
We have very little free-time to work on our audio-related projects, and
your double-blind test suggestion falls low-to-nonexistant in our list of
priorities.
In fact, I am finding that trying to keep up with this list, let alone
participate on it, seems to be taking far more time and energy than I can
reasonably justify.
I was interested in trying to become a more active participant in the
community, but it is feeling rather hostile and counterproductive.
I am working on putting up a blog site (linuxaudyssey.com) - relating our
experiences trying to set up our Linux DAW, and making the transition to
Linux and open-source tools for audio recording.
I hope that eventually I can develop it into something of more general use
to the community - at least the newbies/non-programmers like ourselves.
(Not really anything there yet as I am just now setting it up and have
found I need to switch web hosts.)
Perhaps I'd best confine the bulk of my observations to that site.
Paul, you and I seem to rub each other the wrong way - and I regret that.
Just happens sometimes.
Regardless of our differences, I very much respect and very much appreciate
all your hard work (and the work of everyon else) on making Ardour a
reality.
I think it is already fantastic.
I wish very much that we could send you a big chunk of money in support of
this project and to show our appreciation - and we will definitely do so
when circumstances permit.
I admire your steadfastness in continuing to work on this project despite
virtually non-existent financial support.
I'm afraid that, for the most part, musicians and others likely to be
attracted to this project, are not 'well-heeled'.
Carlo - what can I say?
Keep being yourself.
Be cool.
Make beautiful music.
Pax,
Maluvia
I finally bought an audio card to provide digital i/o to my other equipment.
Thanks to advice last year from subscribers to this list, I managed to get an
RME Digi9652 card on the used market within my budget.
Now to the configuration. I am looking for a working .asoundrc file for this
card. The closest I could find was in a Linux Journal article discussing the
Hammerfall HDSP cards, which are not the same as the Digi9652. I adapted the
configuration found in the article somewhat, but wasn't able to produce
output. The kernel module is loaded and identifies the card:
0000:00:0c.0 Multimedia audio controller: Xilinx Corporation RME Digi9652
(Hammerfall) (rev 03)
and I have been able to modify the amixer settings (these are appended to this
message for reference).
My attempt at an .asoundrc file is as follows. I don't understand what the
ttable values mean and couldn't find an explanation on the alsa.org site or
elsewhere.
pcm.rme9652 {
type hw
card 1
}
lct.rme9652 {
type hw
card 1
}
pcm.adat1 {
type plug
ttable.0.0 1
ttable.1.1 1
ttable.2.2 1
ttable.3.3 1
ttable.4.4 1
ttable.5.5 1
ttable.6.6 1
ttable.7.7 1
slave.pcm rme9652
}
Note: for testing purposes this is only supposed to cover the first ADAT
interface.
aplay -D adat1
runs without error.
One possible cause of trouble is a hardware conflict I know to exist with an
EEPro 100 network card on my system which doesn't work now that the RME
Digi9652 is installed. I am working on this.
Configuration experiences/suggestions and any .asoundrc file from a working
system would be much appreciated.
Jason.
Amixer settings:
numid=5,iface=MIXER,name='IEC958 Input Connector'
; type=ENUMERATED,access=rw---,values=1,items=3
; Item #0 'ADAT1'
; Item #1 'Coaxial'
; Item #2 'Internal'
: values=1
numid=6,iface=MIXER,name='IEC958 Output also on ADAT1'
; type=BOOLEAN,access=rw---,values=1
: values=on
numid=10,iface=MIXER,name='IEC958 Sample Rate'
; type=INTEGER,access=r----,values=1,min=0,max=96000,step=0
: values=-1
numid=16,iface=MIXER,name='ADAT1 Input Source'
; type=ENUMERATED,access=rw---,values=1,items=2
; Item #0 'ADAT1'
; Item #1 'Internal'
: values=0
numid=11,iface=MIXER,name='ADAT1 Sync Check'
; type=ENUMERATED,access=r----,values=1,items=4
; Item #0 'No Lock'
; Item #1 'Lock'
; Item #2 'No Lock Sync'
; Item #3 'Lock Sync'
: values=0
numid=12,iface=MIXER,name='ADAT2 Sync Check'
; type=ENUMERATED,access=r----,values=1,items=4
; Item #0 'No Lock'
; Item #1 'Lock'
; Item #2 'No Lock Sync'
; Item #3 'Lock Sync'
: values=0
numid=15,iface=MIXER,name='ADAT3 Sync Check'
; type=ENUMERATED,access=r----,values=1,items=4
; Item #0 'No Lock'
; Item #1 'Lock'
; Item #2 'No Lock Sync'
; Item #3 'Lock Sync'
: values=0
numid=9,iface=MIXER,name='Channels Thru'
; type=BOOLEAN,access=rw---,values=26
: values=on,off,off,off,off,off,off,off,off,off,off,off,off,off,off,off,off,off,off,off,off,off,off,off,off,off
numid=14,iface=MIXER,name='Passthru'
; type=BOOLEAN,access=rw---,values=1
: values=off
numid=8,iface=MIXER,name='Preferred Sync Source'
; type=ENUMERATED,access=rw---,values=1,items=4
; Item #0 'IEC958 In'
; Item #1 'ADAT1 In'
; Item #2 'ADAT2 In'
; Item #3 'ADAT3 In'
: values=1
numid=7,iface=MIXER,name='Sync Mode'
; type=ENUMERATED,access=rw---,values=1,items=3
; Item #0 'AutoSync'
; Item #1 'Master'
; Item #2 'Word Clock'
: values=1
numid=13,iface=MIXER,name='Timecode Valid'
; type=BOOLEAN,access=r----,values=1
: values=off
numid=3,iface=PCM,name='IEC958 Playback Con Mask'
; type=IEC958,access=r----,values=1
: values=?
numid=1,iface=PCM,name='IEC958 Playback Default'
; type=IEC958,access=rw---,values=1
: values=?
numid=4,iface=PCM,name='IEC958 Playback Pro Mask'
; type=IEC958,access=r----,values=1
: values=?
Hi I am new here trying to get help on Linux.
I am having 3 different problems with Rosegarden 4 on 3 different debian
distros.
Xubuntu 32 bit
When I start Rosegarden I get the splash screen then after a while I get an
error message flashing up that's too fast to read. Rosegarden does not
start up.
Xubuntu 64bit
I get the RG splashscreen and then a crash sound output (breaking glass). I
do not get any error message.
aGNUla/deMudi
Nothing works when I start it up, no keyboard, no mouse, nothing. The only
way out is to hit the system reset button whcih is obvously no good for the
system. I also takes longer to recover from sucha position becasue Demudi
does a system disk check.
I really would like to get Rosegarden up an running as I have ben trying to
do so for about a year now (on and off)!.
Please let me know what info I need to post to get help on this.
Thanks for any help.
Bal