I've been trying to use Kover 0.8.4 to create CD tray cards,
and, more importantly, inserts. This is the latest version of
Kover I could find that does not require KDE 3.0. I'm using 2.2.1
under Mandrake 8.1.
The problem I haven't been able to work around is creating the
INSIDE of a CD booklet, i.e., pages 2-3. I can get it to put the
title on the RH panel and a contents list or whatever I want on
the LH panel. However, I can't start on the LH panel with the
inside of the booklet and have the text wrap around to the RH
panel.
Does anyone have ideas? Is there a better program out there for
doing this? FWIW, none of this stuff can be downloaded from an
Internet database; it's all recordings of live opera
performances.
Thanks for all suggestions. I realize this is not the main
emphasis of the list, so please feel free to reply privately.
Howard Sanner
flagstad(a)mindspring.com
I am trying to user USB-AUDIO product with a TI laptop for a portable
digital recorder.
I have followed the instructions at
http://www.alsa-project.org/alsa-doc/doc-php/template.php3
?company=Midiman&card=USB+Audio+Duo&chip=Cypress+AN2131XX&module=usb-audio
The compiles for building alsa-driver-0.9.0rc6, alsa-lib-0.9.0rc6, and
alsa-utils-0.9.0rc6 went smoothly. The utility snddevices ran with no
errors.
When running the commad alsamixer, I get the error message
"No mixer elems found."
Proc/asound appear to have the correct entries
dr-xr-xr-x 4 root root 0 Dec 7 16:55 card0
-r--r--r-- 1 root root 0 Dec 7 16:55 cards
dr-xr-xr-x 2 root root 0 Dec 7 16:55 dev
-r--r--r-- 1 root root 0 Dec 7 16:55 devices
lrwxrwxrwx 1 root root 0 Dec 7 16:55 MidimanDuo -> card0
dr-xr-xr-x 2 root root 0 Dec 7 16:55 oss
-r--r--r-- 1 root root 0 Dec 7 16:55 pcm
dr-xr-xr-x 2 root root 0 Dec 7 16:55 seq
-r--r--r-- 1 root root 0 Dec 7 16:55 timers
-r--r--r-- 1 root root 0 Dec 7 16:55 version
With Proc/asound/card0 having
-r--r--r-- 1 root root 0 Dec 7 17:00 id
-rw-r--r-- 1 root root 0 Dec 7 17:00 oss_mixer
dr-xr-xr-x 3 root root 0 Dec 7 17:00 pcm0c
dr-xr-xr-x 3 root root 0 Dec 7 17:00 pcm0p
-rw-r--r-- 1 root root 0 Dec 7 17:00 stream0
I found the contributed utility qinit and it runs
correctly with the hw name = "hw:0,0".
With the "dmasound_pmac" module,
dd if=/dev/dsp of=/tmp/sound.out bs=1024 count=1
would produce a file filled with unsigned bytes.
The rest of the day was spent trashing about in config
files to no effect.
Questions:
Should alsamixer work with the M-Audio Duo?
After enabling with qinit should dd dump data?
Any help, or fellow travellers would be appreciated.
John Phillips
I have forwareded this on to LAU also.
From reading your description (nicely worded too, have you considered
writing docs?) It seems like you are mising something either very simple
when trying to record from JACKed apps or your driver is not working
correctly.
Have you tried doing the basic arecord test?
Set the capture channel with alsamixer then
arecord -f cd -D hw:0,0 -d 10 somefile.wav
This should record a cd quality stereo track from the first channel on
your first sound device.
If that doesn't work then none of the other apps could be expected to
either. Personally I still use arecord for recording and edit with other
apps once I have file to work with.
Ryan Beisner (home) wrote:
> Hi Patrick
>
> I'm just having a heck of a time getting all my things in order here.
> Any help you can provide would really be appreciated. Here's what I
> have:
>
> System:
> Redhat 8 / Kernel 2.4.19-1.ll
> PIII 800 / 512MB RAM / 60GB
> SBLive Value PCI
>
>>From CCRMA:
> Kernel 2.4.19-1.ll
> alsa (alsaplayer alsamixer amixer etc)
> jack
> ecasound (and ecawave)
> audacity
> rosegarden
> muse
> meterbridge
> qjackconnect
>
> I have followed the jack and ecasound documents, and successfully set up
> a scenario like this to test:
> alsaplayer -o jack -> meterbridge -> alsa pcm out
> ... and it works. = )
>
> (these are not the exact commands issued)
>
> I have also done this:
> ecasound -i:somewave -o jack -> and that works too.
>
> I've come to the conclusion that all these audio apps should be run as
> root in order for them to play together ... so that's what I do. I
> realize they could be suid etc.
>
>
> My problem is: I CANT RECORD = (
>
> In alsamixer, with the Line set to CAPTURE, I set up the following:
>
> ecasound -i:jack -o:somewave -> it launches without error, and records
> to a wav file. qJackconnect reflects the proper routing connections.
>
> But the wav is dead air... silence for as long as I record.
>
> OK so I thought I might have missed something with the whole jack setup
> so I kill jack and try it with ARecord then again with Audacity. Same
> story. Silence.
>
>
>
> ---DUAL CAPTURE REQUIRED? Weird.
> OK now I notice a "capture" item in alsamixer. So I simultaneously
> hilight LINE and CAPTURE to do capturing. With the LINE muted, audio is
> still monitored (it shouldn't be) unless I de-select the CAPTURE item
> from capturing, but then it won't record. Hope that made sense.
>
> So I give it a go... ignoring the "I can't mute it" problem. arecord
> still silent, same with the jack setup. But now Audacity records ...
> and it's garbled (sounds like playing a 32bit file at 16bit or
> something, plus it's not a fluid recording). And I see that when you
> record in Audacity, it forces 32-bit float for the format. I think
> maybe if I could convince Audacity to operate strictly in 16-bit, it
> might work.
>
This may have already been fixed in cvs. I recall something similar
being discussed recently.
>
>
> I've tried also to capture CD Audio in the same fashion. I've seen
> claims of the same problem dating back to 2000 at geocrawler etc... but
> no info to fix it?
>
> The card's capture functions work in a windoz box.
>
>
>
> What would you suggest? I'm at a loss.
>
>
>
>
> Thanks in advance,
>
> -Ryan Beisner
>
>
>
> ps. System is tuned nicely. Jack looks like this when it's running:
>
> load = 0.3496 max usecs: 43.216, spare = 11565.784
> load = 0.4025 max usecs: 52.861, spare = 11556.139
> load = 0.3142 max usecs: 26.222, spare = 11582.777
> load = 0.2733 max usecs: 26.975, spare = 11582.025
> load = 0.2133 max usecs: 17.801, spare = 11591.199
> load = 0.1753 max usecs: 15.932, spare = 11593.067
> load = 0.1941 max usecs: 24.719, spare = 11584.281
> load = 0.1540 max usecs: 13.230, spare = 11595.770
> load = 0.1292 max usecs: 12.119, spare = 11596.881
> load = 0.2109 max usecs: 33.976, spare = 11575.023
> load = 0.2310 max usecs: 29.149, spare = 11579.852
>
>
>
>
--
Patrick Shirkey - Boost Hardware Ltd.
For the discerning hardware connoisseur
Http://www.boosthardware.comHttp://www.djcj.org - The Linux Audio Users guide
========================================
Being on stage with the band in front of crowds shouting, "Get off! No!
We want normal music!", I think that was more like acting than anything
I've ever done.
Goldie, 8 Nov, 2002
The Scotsman
Hello everybody,
I did a small presentation about the state of Linux audio & music
apps on 23rd of November, for the Italian LinuxDay promoted by the
Italian Linux Society. I was asked for by some members of FerraraLUG
(Ferrara Linux User Group), and now they have put some material on the
web.
It's in Italian, but if you want to take a look, or reuse the slides...
The slides:
http://linuxday.ferrara.linux.it/files/linuxday2002-grilli.sxi
The audio stream (ogg format):
http://linuxday.ferrara.linux.it/audio/32kbps/emilio.ogg (11MB)
I had only 45 minutes, so I'm sorry for any software that I left
out. I hope to have done a good advocacy for Linux audio for the
audience.
Best regards,
--
.------------------------------------.
| Emiliano Grilli - emillo(a)libero.it |
'------------------------------------'
Hi all,
Does anybody know where i can get some good/clear/concise explanations
about how sounds works under linux?
I'm a programmer and a musician myself.
I'm so confused between midi, alsa, oss, oss emulation, arts, sound servers
artsd, esd, etc...etc...
midi ports "a la alsa" or "a la oss" /dev/midiXXX /dev/sound /dev/sequencer
or 72:0 73:0 or whatever
softsynth, hardsynth...........................
All this gets me CONFUSED!
I want to develop some kind of band-in-a-box-like stuff with TSE3
The app won't need a midi card (i want to use a softsynth)
But first I want to have a clear, global, vision of sound architecture on
linux (if not so much precise)
I've abandoned window$ years ago and the only thing i really miss is band in a
box and i don't want to fall back into microsoft!
But i must admit i can't find a unique place that explains
the way sound works in linux.
I find loads of stuff but can't have a satisfactory global vision of the whole
stuff.
Has anybody written somewhere a real digest i could use as good starting
point?
For instance, what REALLY makes the difference between alsa and oss and arts?
What are midi ports in each of those architecture?
How to adress them?
How to to build something for ANY sound architecture?
Things about sound deamonds also (what are they for, and how to use them)?
Well have been working too hard today , i think my brain is about to burst
out ;-)
I think sound programming under linux is very complicated because of the
different architectures.
But developing sound apps like you guys do is
a necessary step to convince people to use linux!
Maybe my goal is too hard for me but anyway i want to give it a try...
But what a MESS!!
So any hint on where i can get a thorough, clear, concise explanation
of the whole stuff is welcome!
Thanks
pma <Armstrong.Peter.DMA(a)aya.yale.edu> writes:
> Dear List --
>
> With Debian sources and Dan McCombs's "Installing and Configuring ALSA"
> (from http://www.LinuxOrbit.com) I have compiled a 2.4.19 kernel, customized
> the /etc/alsa/ config files for snd-ice1712, and installed ALSA via
> make-kpkg.
>
> However, the created /lib/modules/2.4.19/alsa/*.o files include none
> for my card.
Debian's alsa-source package has a config file named
/etc/alsa/alsa-source.conf. Debconf should have prompted you for the
name(s) of your card(s) when you installed this package. If that
wasn't done, you can edit that file to define something like this:
ALSA_CARDS="ice1712, rme9652, via82xx"
Or, you can run (as root):
dpkg-reconfigure alsa-source
and select the ice1712 driver.
--
Jack O'Quin
Austin, Texas, USA
Hi.
Today I relased a new version of ZynAddSubFX.
News: - corrected a bug that made ZynAddSubFX to
crash(sometimes) if you disable a part
- wrote Resonance (This produces natural sounds,
listen demo07.ogg from homepage)
- added the BandPass filter
- added the recording feature (as raw files)
- added "New instrument" menuitem
ZynAddSubFX is a free (GPL v.2) software synthesizer.
The homepage is:
http://zynaddsubfx.sourceforge.net/
You can download it from:
http://sourceforge.net/projects/zynaddsubfx
Hope you like it.
Paul.
__________________________________________________
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Does anyone know of a free software SoundDiver clone?
SoundDiver is an E-magic synth patch editor with a plug in
architecture and support for a huge number of synths. One of my
favorite toys - the Yamaha FS1R - is near impossible to program
without it. Is anyone working on a clone for Linux?
A.
Mark Knecht wrote
>Have you tried running Sound Diver under Wine? It's on my to do
>list.
Hi Mark,
No I haven't tried it under Wine, for this reason...
SD needs to authorise itself against the CD periodically.
Precisely what it does during this procedure is anyone's guess.
All I know is that it takes around 3 minutes on my W98 PC. The
machine frezes druing the authorisation and some very very
scary noises come form the HD and CDROM drive.
I remember in the 80s there were once some odd copy protection
schemes that stored key data in illegal sectors on the HD. The
odd behaviour during authorisation make me think of that. I
suspect the chances of authorisation working on a foreign file
system, under an emulator are about 0. I have visions of it
cutting a hole in my HD if I tried it under Linux.
>Have you tried running Sound Diver under Wine? It's on my to do
>list
You are a brave man than I. Let me know how you get on.
A.