I just tried taking a stereo track in ardour that was recorded with a X/Y pair and add two pugins to the track: a Stereo to Mid-Side(MS) matrix and a Mid-Side to Stereo matrix. The idea was that the MS to Stereo matrix gives me a nice simple control to control the width of the stereo image.
What I discovered, to my surprise, was that with the width set to 2.0, the widest setting, the stereo image was still narrower than with the pair of plugins bypassed.
Can anyone think why this should be?
Steve
Hi,
Since I upgraded from Suse 8.2 to Suse 9.1, I get a very weak
recording signal with my Audigy SoundBlaster Card. Alsa package is
alsa-1.0.3-37.
In Audacity I can see that it is 1/3 of what it was formerly although
with Alsamixer all possible controls are set to maximum.
What else can I do?
There has been a thread in the Web but it didn`t turn out to meet my
problem.
So help is very much appreciated.
Thanks Hartmut
Hi folks. I've made this icon for Soundtracker using Kiconedit. I've contacted
Michael Krause as to using this icon, and got the reply, "Feel free to use
the icon, no problem there". Nigel.
Icon attached: Soundtracker.png
Dear guys,
I am writing here because I have some problem with my setup.
It is composed by:
-Linux based audio workstation (Fedora Core 3 with PlanetCCRMA, RME
HDSP9652)
-Z-sys z-8.8a lightpipe digital detangler
(http://www.z-sys.com/pp_routing.html#z8)
-RME ADI-8 Pro
-A modified SACD/DVD-A/DVD/CD player which is capable of outputting 6 (5.1)
discrete PCM channel (for SACD a DSD->PCM conversion is performed,
converting DSD to 24bit/88.2Khz PCM signal)
On the workstation AlmusVCU is running, which is used as a real-time
convolver.
The system is connected as follows:
modified SACD/DVD-A/DVD/CD player-->(3 s/pdif cables)-->Z-sys z-8.8a-->(1
ADAT cable)-->RME HDSP 2496 ADAT 1 input-->(1 ADAT cable from ADAT 1
output)-->RME ADI-8 Pro ADAT input
the clock signal is generated internally by RME ADI-8 Pro (48Khz), which is
connected by word clock BNC cable to RME HDSP9652, which is setted in slave
mode.
Z-sys z-8.8s does not have an internal clock, so this is provided through
its input 1 by ADI-8 Pro ADAT output; the Z-sys unit is also setted with SRC
ON, so it resample all incoming signal to 24bit/48Khz (48Khz is provided by
ADI-8 Pro) on output.
Now the problem is the sound is very crappy, similar to what you can hear
when DAC is not synched with source signal: switching SRC to OFF on Z-sys
unit provides an even more crappy sound.
I tried all different clock routings I can think to, but never solved the
problem: does anyone can provide some general (I know it will be hard to
"get into" this setup) rules for clock signal routing in setups like mine?
I also tried to play music directly in the audio workstation sending it to
RME ADI-8 Pro (AlmusVCU allows this) and everything sounds fine, so I think
the problem is given by Z-Sys unit.
Thank you very much in advance to everyone who will reply!
Michele
Well... Do you guys accept the two cents from a novice user?
In fact I think I am not exactly a complete novice, but I am not a
developer and, although I have a lot of ideas and suggestions, I just
don´t have the skill and the time for getting my hands dirty of code.
This said, I would point some things I think could be better:
1) Realtime - Man, having to recompile the kernel for having realtime
support is not a trivial task for the average computer user. I hope that
new distros will have this module enabled by default, or at least available.
2) JACK is wonderful and I hope it keeps in evolution. I wish it can be
less machine-hungry (with fewer xruns) and that the JACK transport
feature support tempo changes someday soon.
3) I had a LOT of problems until I could record audio in my Audigy 2ZS
card, and I think it was a problem with the Linux mixers. None of them
had proper support for the EMU10K1 ALSA driver, until I could get the
guts with Alsamixer. And, in fact, I still have not found a really good
way for controlling sound output for all of my 5.1 speakers. I think
it´s not fair to blame the Linux developers for that. But, indeed, big
hardware companies, like Creative Labs, which still insist in giving
only Windoze support for their products. Just look at their open-source
webpage and you will know how much they collaborate with the open-source
community.
4) There are already a good number of interesting music apps for
GNU-Linux, like Ardour, Rosegarden and Hydrogen; but they surely have
some steps to walk yet. I wish I can help their development in some way.
Regards,
Fabricio Rocha
Brasilia, Brasil
_______________________________________________________
Yahoo! Acesso Grátis - Internet rápida e grátis.
Instale o discador agora! http://br.acesso.yahoo.com/
by Kjetil Svalastog Matheussen <k.s.matheussen@notam02.no>
Dave Phillips:
>
> Greetings:
>
> The subject says it all.
>
> My own "Linux audio sucks" hobbyhorse:
>
> No way to recall a complex configuration of apps and plugins with
> all settings intact. If I use a complicated setup with multiple synths
> and plugins I have no way to recall these applications to their previous
> settings. LASH/LADCCA was supposed to address this situation but I don't
> know where that project stands at this point.
>
> And your favorite is... ?
>
Right now: artsd and flashplayer
artsd because it currently on the machine I'm working with right now
continouisly tries to start. When I kill it, it starts again, for
mysterious reasons, only seconds later. Sometimes, it is even owned by
root, and I have to kill it as root. Even that doesnt help, it starts
again and again and again.
flashplayer because it lives inside a browser and should not have been
allowed to occupy the soundcard. Sometimes, the only way to get sound (in
proper programs, like jack), is to exit the browser.
The first one is because of some crazy bug or design or whatever. The
second one can probably be fixed somehow. However, it probably all goes
down to the current long discussion about finding a common sound-server
thing, which is definitely needed. I don't understand what happens with
this crazy artsd-madness-program, and I think I'm some kind of power-user.
Linux audio sucks because of this, right now, for me. esound, artsd,
polyaudio, jackd; theire all the wrong solution for this problem. The
correct solution is that alsa needs to be redesigned so that it can switch
from direct hardware access (hw:0, 1, etc.) to dmix or jackplug and back
again on the fly without the user noticing it. The second thing that needs
to be done is to remove artsd from all linux audio machines around the
world so that noone ever will have crazy problems with it again. At least,
I wont to get rid of it on the machine I'm currently working on.
Unfortunately, I'm not the administrator of it, so I can't do that.
(No, I don't want any help in fixing the artsd-thing, right now I just
want to hate it)
Any recommendations for laptops? I'm looking at Toshiba and
Fujitsu-Siemens models around EUR1100. I'll mostly be doing synthesis,
eg ZynAddSubFX, and programming. I'll be using the built-in soundcard,
at least for now. I don't really care about battery life.
The main question is: Celeron v Pentium 4 v Pentium-M? I read where
Steve Harris wrote:
> the Pentium M's are better for audio work (IMHO) than
pentium 4's, due to better denormal handling, and can do more
clock-for-clock.
Is this true for Celeron-M as well (if it really exists)? Or should I
definitely, definitely avoid Celeron?
Howdy peeps.
Has anyone got jack and an sblive (value) to work for recording?
I get output from the microphone (actually a guitar, but I don't think
that makes much difference) through my speakers, but jack hears
nothing.
I have MIC set to capture in alsamixer and ac97 capture is turned up.
Is there anything else I need to do?
If I haven't given enough info, please just tell me.
Cheers,
James
--
"I'd crawl over an acre of 'Visual This++' and 'Integrated Development
That' to get to gcc, Emacs, and gdb. Thank you."
(By Vance Petree, Virginia Power)
Lee Revell:
>
> > So I gather that my best bet is to avoid mainboards that have VIA
> > chipsets.
>
> VIA chipsets work fine. Their bad reputation is due to poor quality Windows
> drivers.
>
Agreed. VIA is probably _the_ chipset to use for linux, at least for
AMD-based systems.
--
hi list
first of all, thanks for all those who sent answers.
some more questions:
> you should install one a the last kernel-multimedia : apt-get install
> kernel-multimedia or use synaptic to find one, Alsa will be already
> configured.
is there such a kernel for ppc? i wrote to the demudi list and
apparently they are still working on a ppc distribution, but with no
release date in sight.
is there a so-called multimedia kernel for ppc available somewhere else?
i now have sound working with alsa drivers. well, at least it works with
pd. i still can't listen to cds.
another problem is, the system sound configuration doesn't seem to get
saved. that is, when i re-start i have to run
mopprobe snd-powermac
alsaconf
then i can use the alsa mixer and pd audio runs. there seems to be no
way now to control sound, aside from using the alsa mixer.
where should i go from here?
thanks again.
best regards
jason kahn