I have an older box that I've been toying around with as a means to play a new
M-Audio Axiom 49 I just picked up. With some tinkering I was able to get
timidity installed as server and used jack to get the thing playing sounds but
still noticed an ever so slight bit of latency on this old beast of a machine.
So the other day I saw the local second-hand store selling a PCI 16 soundblaster
card and I picked it up for $5. I have it working now for audio, but still seem
to have trouble getting anything to come out the sequencer.
My motherboard has a VIA/realtek 8233 compatible chip on it and this new one
runs under the ES1371 driver. My M-Audio shows up as USB-Audio
# lspci | fgrep audio
00:0b.0 Multimedia audio controller: Ensoniq 5880B [AudioPCI] (rev 02)
00:11.5 Multimedia audio controller: VIA Technologies, Inc. VT8233/A/8235/8237
AC97 Audio Controller (rev 50)
# cat /proc/asound/cards
0 [AudioPCI ]: ENS1371 - Ensoniq AudioPCI
Ensoniq AudioPCI ENS1371 at 0xec00, irq 10
1 [U49 ]: USB-Audio - USB Axiom 49
M-Audio USB Axiom 49 at usb-0000:00:10.0-2, full speed
2 [V8235 ]: VIA8233 - VIA 8235
VIA 8235 with ALC101 at 0xd800, irq 5
Of course the Timidity stuff shows up at 128 for ports 0-3 and alsa is saying
that port 0 on the AudioPCI is showing up at 16. But no matter what I seem to
do, I can't get any audio to come out:
# aplaymidi -l
Port Client name Port name
14:0 Midi Through Midi Through Port-0
16:0 Ensoniq AudioPCI ES1371
20:0 USB Axiom 49 USB Axiom 49 MIDI 1
128:0 TiMidity TiMidity port 0
128:1 TiMidity TiMidity port 1
128:2 TiMidity TiMidity port 2
128:3 TiMidity TiMidity port 3
# aplaymidi -p 128:0 sample.mid # this works
# aplaymidi -p 16:0 sample.mid # this acts like it's doing something
# but no sound comes out
Does anyone have any suggestions to help me get this thing working? The reason
I bought this thing (besides the fact it was only $5) was because of the latency
running the timidity soft-synth with my Axiom. I was hoping that since it had a
hardware-based wavetable that it would have less latency. I'd love to get it
working! Thanks for any help in advance.
SW
----
Barack-O-phobia: The fear of politicians who think (more) government is the solution to every problem.
> Message: 3
> Date: Sat, 25 Jun 2011 06:36:04 -1000
> From: david <gnome(a)hawaii.rr.com>
> Subject: Re: [LAU] Gnome-shell | Unity
> To: linux-audio-user <linux-audio-user(a)lists.linuxaudio.org>
> Message-ID: <4E060E74.80709(a)hawaii.rr.com>
> Content-Type: text/plain; charset=us-ascii; format=flowed
>
> rosea grammostola wrote:
>
> > So guys, just upgrade Gnome2 to Gnome3? ... Is it true that both
> > Gnome-shell and Unity needs Pulseaudio?
>
> I don't know about that. I know that Gnome3 uses 1GB of memory ... My
> wife's new netbook has Ubuntu 11.04 on it, haven't noticed if it's using
> PA or not, but sound works.
I kept GNOME2 and I'm unwilling even to test GNOME3, but I asked
somebody from the Ubunut Studio lists, regarding to the claim that
GNOME3 should use 1GB of RAM. It might be true on your machine, for your
install, but the one I asked claims that you're (or who ever it was who
claimed this before) mistaken. IIRC his GNOME3 does need 42MB only.
Since I didn't need to install PA for GNOME2 on Debian, I suspect hat
it's just another multimedia unfriendly issue by your distro (note,
Debian doesn't install PA, but it ships with other annoying multimedia
unfriendly issue ;). Perhaps GNOME3 for Debian doesn't need PA too.
Hi,
I use Ubuntu-studio 10.04 with Ardour 2.8.11 on my PackardBell mh36
laptop (intel 2core t4200, 3 gb ram).
When i stop playback ,during my mixing session, dsp increases to 100 %.
This it causes me a lot of xrun in jack...
If i disable the eq10q in many tracks everything come back to normal.
Any suggestion about that ?
I'd like to donate something to this important project (eq10q) but i
can't find any donate button here:
http://eq10q.sourceforge.net/
Can you show me a way to do that ?
Thanks in advance
Nicola
I installed lv2core a couple of months ago on a Slackware 13.1 system,
and I created my own SlackBuild script to compile and install it, which
configured waf with "--prefix=/usr --configdir=/etc --libdir=/usr/lib64
--mandir=/usr/man". This put a copy of lv2.h in /usr/include/, and
another copy in /usr/lib64/lv2/lv2core.lv2/. I also installed SLV2
with no problems.
Since lilv came out (along with a couple of other packages to replace
SLV2), I decided to install it. I installed serd and sord first, but
when I try to compile lilv, it bombs out because it can't find
lv2/lv2plug.in/ns/lv2core/lv2.h (presumably, this path would be appended
to /usr/include/ for the complete path), and I can't figure out any way
to get lv2.h there without just creating each directory in the path
myself and copying it there. There is no mention of that path anywhere
in the waf scripts. The only mention I can find of that path is in the
README and INSTALL files, both of which suggest that programmers use that
path in their build scripts to determine whether lv2core is installed.
Why is that being recommended when the lv2core scripts don't install it
there? Am I missing something? This is driving me crazy. Any help
in preserving my sanity would be greatly appreciated.
Chuck
P.S. I've already tried putting this in my SlackBuild script:
sed -i -e "s,lv2/lv2plug.in/ns/lv2core/,," wscript lilv/lilv.h
and that allows it to compile and install alright, but it means I'll
also have to do something similar to every other program that looks
for lv2.h in that location, and that doesn't seem to be a be a very
nice solution.
On Wed, 2011-06-22 at 11:58 +0000,
linux-audio-user-request(a)lists.linuxaudio.org wrote:
> Message: 28
> Date: Wed, 22 Jun 2011 13:04:39 +0200 (CEST)
> From: Julien Claassen <julien(a)c-lab.de>
> Subject: Re: [LAU] Text-based sound visualisation?
> To: Fons Adriaensen <fons(a)linuxaudio.org>
> Cc: linux-audio-user(a)lists.linuxaudio.org
> Message-ID: <alpine.DEB.2.00.1106221301350.23405(a)britney.spears.net>
> Content-Type: TEXT/PLAIN; format=flowed; charset=US-ASCII
>
> Hello Fons!
> It's good to hear, that people are more often fooled by spectral
> analysers
> than their own ears. I suppose I'll just continue the way I have and
> if I'm
> just curious I will use songanalysis and have some fun. :-)
> Thanks for clarifying and putting me at ease. I hope Massy is more
> at ease
> as well. You so often hear about graphic displays of waveforms,
> analysis with
> graphs and what not and think, that it's one of the important tools to
> get
> professional mixes, especially if you think about "radio
> compatibility" and
> "pop standards".
> Warm regards
> Julien
You might be able to see low band frequencies inaudible by your
speakers, that could stress other speakers. A little bit far-fetched,
but I could imagine that someone might use it for this or similar
issues. I don't know anybody using spectral analysis for the mastering.
Message: 16
> Date: Fri, 24 Jun 2011 12:02:14 +0200
> From: Philipp ?berbacher <hollunder(a)lavabit.com>
> Subject: Re: [LAU] msuci of the ages :-)
> To: linux-audio-user <linux-audio-user(a)lists.linuxaudio.org>
> Message-ID: <1308909303-sup-9628@eris>
> Content-Type: text/plain; charset=UTF-8
> Place one was a Lady Gaga
> song, and while I liked some of her songs, this one was really nothing
> special at all.
Randomly I read this digest mail, until now no others.
I agree that even Lady Gaga made some good songs. I once heard a song
from Kesha, performed live, that had less to do with the radio recording
of the same song she performed, it also was very good.
As soon as they stop loudness war and don't use auto-tune, even some of
very commercial chart music is ok, but not all ...
In Oberhausen we've got a very big funfair
( http://www.fronleichnamskirmes.de/index.php ) at the moment. They play
'audio material' that loud, that it's dangerous to health and with
overkill boost for the bass. This 'audio material' is just garbage,
boom, boom, boom, boom.
Jets are less loud! The regulatory agency isn't active! Money is more
important, than the health of children :(.
I have an older box that I've been toying around with as a means to play a new
M-Audio Axiom 49 I just picked up. With some tinkering I was able to get
timidity installed as server and used jack to get the thing playing sounds but
still noticed an ever so slight bit of latency on this old beast of a machine.
So the other day I saw the local second-hand store selling a PCI 16 soundblaster
card and I picked it up for $5. I have it working now for audio, but still seem
to have trouble getting anything to come out the sequencer.
My motherboard has a VIA/realtek 8233 compatible chip on it and this new one
runs under the ES1371 driver. My M-Audio shows up as USB-Audio
# lspci | fgrep audio
00:0b.0 Multimedia audio controller: Ensoniq 5880B [AudioPCI] (rev 02)
00:11.5 Multimedia audio controller: VIA Technologies, Inc. VT8233/A/8235/8237
AC97 Audio Controller (rev 50)
# cat /proc/asound/cards
0 [AudioPCI ]: ENS1371 - Ensoniq AudioPCI
Ensoniq AudioPCI ENS1371 at 0xec00, irq 10
1 [U49 ]: USB-Audio - USB Axiom 49
M-Audio USB Axiom 49 at usb-0000:00:10.0-2, full speed
2 [V8235 ]: VIA8233 - VIA 8235
VIA 8235 with ALC101 at 0xd800, irq 5
Of course the Timidity stuff shows up at 128 for ports 0-3 and alsa is saying
that port 0 on the AudioPCI is showing up at 16. But no matter what I seem to
do, I can't get any audio to come out:
# aplaymidi -l
Port Client name Port name
14:0 Midi Through Midi Through Port-0
16:0 Ensoniq AudioPCI ES1371
20:0 USB Axiom 49 USB Axiom 49 MIDI 1
128:0 TiMidity TiMidity port 0
128:1 TiMidity TiMidity port 1
128:2 TiMidity TiMidity port 2
128:3 TiMidity TiMidity port 3
# aplaymidi -p 128:0 sample.mid # this works
# aplaymidi -p 16:0 sample.mid # this acts like it's doing something
# but no sound comes out
Does anyone have any suggestions to help me get this thing working? The reason
I bought this thing (besides the fact it was only $5) was because of the latency
running the timidity soft-synth with my Axiom. I was hoping that since it had a
hardware-based wavetable that it would have less latency. I'd love to get it
working! Thanks for any help in advance.
SW
----
Barack-O-phobia: The fear of politicians who think (more) government is the solution to every problem.
Hi there,
in a discussion today someone asked me where those 60 degrees necessary
for the production of phantom images come from and I couldn't deliver a
satisfactory answer. Someone tried to explain to me that it has
something to do with wavelengths or whatever but couldn't explain it in
a way that anyone would understand.
My best guess is that with a larger angle the head gets in the way and
the ears have an easier time telling the signals apart. Also, I guess 60
degrees is a rough estimate and chosen because this leads to a nice
Equilateral triangle.
So, what's the real reason behind those 60 degrees?
Regards,
Philipp
Hi all,
just want to share my happiness with the whole community because Pat Metheny
just left my studio.
He was here for an interview for the radio station i work with.
It was just an interview but i can say that Pat Metheny was recorded with
free software :)
Cheers
http://trexstudio.com/wp-content/gallery/gallery/trex_ben_pat_metheny.jpg
--
Giorgio Baù
*Sound engineer*
T.Rex Studio
www.trexstudio.com