Thank you, thank you, a million thank you's!!! I just tried this and can
actually control elements for my needs.
I'm only really using AMS as a "host" for various LADSPA plugins like
Freeverb and delays for live vocal processing on a small PA. I have a
multi fx guitar pedal with midi capabilities (limited) and need to be
able to turn the FX of between songs as I talk to the audience.
As simple as this was to do (when I knew where to look! And not to
mention it did speak to it somewhat in the "manual" on Agnula), I was
not finding how to get it done.
On Mon, 2004-03-08 at 09:00, Matthias Nagorni wrote:
> On Sun, 7 Mar 2004, Russell Hanaghan wrote:
>
> > I would like to know if anyone has been able to set AMS up to see
> > Program change / Patch change midi messages? I can get the CC bound to
> > stuff easily...bit I dont see patch change messages appearing in the
> > "Control" window like the CC does. Is this possible?
>
> Program Change messages are used to control the presets in the
> "Parameter View" dialog. See the examples in the "instruments" folder for
> this.
>
> Matthias
Hi All,
I have been pulling my hair out trying to compile timidity...
I configure thusly...
./configure --enable-audio-oss --enable-interface=gtk
--enable-audio=alsa --enable-server --enable-alsaseq
When I 'make' I get the following error.
make[2]: Entering directory `/usr/src/MUSIC/TiMidity++-2.9.5/timidity'
gcc -DHAVE_CONFIG_H -I. -I. -I.. -I.. -I.. -I../libarc
-I../libunimod -I../interface -I../utils
-I/usr/local/include -I/usr/X11R6/include -I/usr/include/gtk-1.2
-I/usr/include/glib-1.2 -I/usr/lib/glib/include -DAU_ALSA
-DPKGDATADIR=\"/usr/local/share/timidity\"
-DPKGLIBDIR=\"/usr/local/lib/timidity\"
-DDEFAULT_PATH=\"/usr/local/share/timidity\" -DTIMIDITY_OUTPUT_ID=\"s\"
-O2 -c alsa_a.c
alsa_a.c:52: conflicting types for `snd_pcm_t'
/usr/local/include/alsa/pcm.h:300: previous declaration of `snd_pcm_t'
alsa_a.c: In function `check_sound_cards':
alsa_a.c:116: storage size of `ctl_hw_info' isn't known
alsa_a.c:117: storage size of `pcm_info' isn't known
alsa_a.c:152: warning: passing arg 2 of `snd_ctl_open' makes pointer
from integer without a cast
alsa_a.c:152: too few arguments to function `snd_ctl_open'
alsa_a.c:186: warning: passing arg 2 of `snd_ctl_pcm_info' makes pointer
from integer without a cast
alsa_a.c:186: too many arguments to function `snd_ctl_pcm_info'
alsa_a.c:194: `SND_PCM_INFO_PLAYBACK' undeclared (first use in this
function)
alsa_a.c:194: (Each undeclared identifier is reported only once
alsa_a.c:194: for each function it appears in.)
alsa_a.c: In function `set_playback_info':
alsa_a.c:235: `snd_pcm_playback_info_t' undeclared (first use in this
function)
alsa_a.c:235: parse error before "pinfo"
alsa_a.c:237: storage size of `pparams' isn't known
alsa_a.c:238: storage size of `pstatus' isn't known
alsa_a.c:248: `pinfo' undeclared (first use in this function)
alsa_a.c:263: `SND_PCM_PINFO_8BITONLY' undeclared (first use in this
function)
alsa_a.c:265: `SND_PCM_PINFO_16BITONLY' undeclared (first use in this
function)
alsa_a.c:277: request for member `rate' in something not a structure or
union
alsa_a.c:298: request for member `channels' in something not a structure
or union
alsa_a.c:300: request for member `channels' in something not a structure
or union
alsa_a.c:306: `SND_PCM_FMT_S16_LE' undeclared (first use in this function)
alsa_a.c:311: request for member `format' in something not a structure
or union
alsa_a.c:311: `SND_PCM_SFMT_S16_LE' undeclared (first use in this function)
alsa_a.c:342: `SND_PCM_FMT_U8' undeclared (first use in this function)
alsa_a.c:347: request for member `format' in something not a structure
or union
alsa_a.c:347: `SND_PCM_SFMT_U8' undeclared (first use in this function)
alsa_a.c:383: request for member `channels' in something not a structure
or union
alsa_a.c: In function `open_output':
alsa_a.c:541: `SND_PCM_OPEN_PLAYBACK' undeclared (first use in this
function)
alsa_a.c:541: warning: passing arg 1 of `snd_pcm_open' from incompatible
pointer type
alsa_a.c:541: warning: passing arg 2 of `snd_pcm_open' makes pointer
from integer without a cast
alsa_a.c: In function `acntl':
alsa_a.c:673: storage size of `pstatus' isn't known
make[2]: *** [alsa_a.o] Error 1
make[2]: Leaving directory `/usr/src/MUSIC/TiMidity++-2.9.5/timidity'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/usr/src/MUSIC/TiMidity++-2.9.5'
make: *** [all-recursive-am] Error 2
[root@innanu /usr/src/MUSIC/TiMidity++-2.9.5]#
I don't get it! My /usr/include/sound|sys|alsa are all synced (to the
alsa includes), but those defines dont exist in any of the includes anyway
I can;t ust a compiled RPM as I need the alsa-sequencer part, which
isn;t compiled into any rpm I have tried (about 10!)
However, in the pcm.h there IS a var called SND_PCM_FORMAT_U8 vs
SND_PCM_FMT_U8. I can't believe this... there must be a version of the
pch.h that uses the FMT other than FORMAT. This seems so obvious an
error I can;t believe I see nothing on the net about it. In addition
snd_pcm_t is defined twice... the type in pcm.h conflicting with the
instance in timidity's alsa_a.c
I have search thru every pcm.h I have.. with no luck.
I found one refernce in google
(http://cvs.sourceforge.net/viewcvs.py/alsa/alsa-driver/cards/Attic/card-dum…)
which refers to the card_dummy.c, but I don;t have (want) and dummy as I
have a real card.
Does anyone have a clue as to why this is and how to fix it?
Thanks...
---------
info
---------
Version: TiMidity++-2.9.5
ALSA: 1.0.3
Linux: 2.6.4-rc2
/etc/modules.conf
---BEGIN---
alias parport_lowlevel parport_pc
alias usb-controller usb-uhci
alias char-major-116 snd
options snd major=116 cards_limit=1
alias sound-slot-0 snd-intel8x0
alias snd-card-0 snd-intel8x0
options snd-intel8x0 index=0 id="ICH" mpu_port=0x330
alias sound-slot-1 snd-card-1
alias snd-card-1 snd-virmidi
alias char-major-14 soundcore
alias sound-service-0-0 snd-mixer-oss
alias sound-service-0-1 snd-seq-oss
alias sound-service-0-3 snd-pcm-oss
alias sound-service-0-8 snd-seq-oss
alias sound-service-0-12 snd-pcm-oss
alias /dev/mixer snd-mixer-oss
alias /dev/dsp snd-pcm-oss
alias /dev/midi snd-seq-oss
install sound-slot-0 /sbin/modprobe --ignore-install sound-slot-0 && {
/bin/aumix-minimal -f
remove sound-slot-0 { /bin/aumix-minimal -f /etc/.aumixrc -S >/dev/null
2>&1; /bin/true; }; /
alias eth0 8139too
alias char-major-89 i2c-dev
---END---
hi
at last, the oss things are workin on my AMiTY CN
with has Slackware installed
and has ES1878 for sound things
planning to do SAOL things on this machine
but til things start workin
would like to enjoy listening to mp3s
the mpg123 works fine
but on my machine
it is too loud
read trough the man
but my poor english readings couldn't find things
to meet my demand
is there any option flag to make mpg123 play more carmly
or
i should tune something more master-volume oriented
..maybe something @ /etc directory ?
thanks in advance
I just sent this e-mail to Edirol. It may be useful to someone if
they're contacting them or any other company for Linux support.
Dan
-----Forwarded Message-----
From: Dan Harper <tech(a)danharper.org>
To: sales(a)edirol.com
Cc: artists(a)edirol.com
Subject: Edirol FA-101 Linux Support
Date: Wed, 10 Mar 2004 16:11:58 +1100
To Whom It May Concern,
I'm a recording musician with a home studio and I am looking to upgrade
and go mobile with some of my gear. I will be purchasing a laptop in
the next few weeks along with a new pair of studio monitors and a
multi-channel audio interface upgrade.
I happened to stumble across your new product, the Edirol FA-101
Firewire interface that seems to fit all the criteria I had for the new
interface. The final and deciding requirement I have for the hardware
however, is not mentioned on your web site, and that is Linux
compatibility.
My existing studio setup consists of using the DAW software Ardour
(http://www.ardour.org) along side the JACK realtime audio server
(http://jackit.sf.net) and the mastering suite Jamin
(http://jamin.sf.net). I currently have no problems using my PCI audio
card with my desktop machine, but it seems that FireWire interfaces are
not fully supported as yet.
Currently ALSA (An audio system for Linux that Jack utilises,
http://www.alsa-project.org) supports many audio interfaces, including
several Edirol USB products. I haven't checked with the Linux Audio
community yet as to what they know of your product, but are there any
plans to, or would you consider:
- releasing the product data, specifications and other technical
documentation of the FA-101 unit?
- assist in writing drivers, or supporting the developers (by e-mail)
in writing drivers for ALSA?
One positive aspect of companies such as Edirol releasing product data,
specifications, and other technical documentation regarding their
products to the Linux community, is that there is no need to invest time
or money in supporting and continually developing drivers. The
Linux/ALSA community is happy to develop and support drivers for your
hardware at no cost.
I realise that Linux has not been a traditional market for professional
audio, but I have successfully and happily made the switch from
Microsoft Windows to Linux and I know of many other musicians that have
also switched over, or are looking to do so. This would be a great
opportunity to increase the compatibility and user base of your hardware
with minimal effort on your part.
For me personally, since I have fully made the switch to Linux, my next
audio interface upgrade would have to have (or plans to have) Linux/ALSA
drivers.
If you require any further information, please con't hesitate to contact
me. I look forward to hearing from you soon.
Thanks,
Dan Harper
--
Dan Harper
Songwriter and Musician
Melbourne, Australia
ZynAddSubFX is using XML files
Hi.
I am Paul, the author of zynaddsubfx.
For those who you don't know about it: is a software
synthsizer for linux and windows and is located at
http://zynaddsubfx.sourceforge.net
I added many things from 1.4.3. The most notable is
the XML support.
I have the intention to drop the compatibility to the
old formats (mas_zyn) since 2.0. Anyway, the code for
using this format is still inside zynaddsubfx to make
possible to write a file converter.
Now, zynaddsubfx requires 2 new libraries: an excelent
library called Mini-XML
(http://www.easysw.com/~mike/mxml/) and zlib.
Also, the banks are not longer stored into a single
file; now, the banks are stored as instruments into a
directory. This make very easy to manage the banks,
with any file manager.
You can checkout the cvs version; see
http://sourceforge.net/cvs/?group_id=62934 for
details.
Also, if you don't want to use cvs you can download
the version from 3 March 2004 from:
http://zynaddsubfx.sourceforge.net/zynaddsubfx_cvs.tar.bz2
(Please notice that this link is temporary)
There are other features that will be documented into
2.0. Until then, there is more do be done :)
Please test it and report if happens to save a file to
XML and if is not loaded all parameters (It is
possible that I missed some parameters from XML, but I
hope not ;) )
Paul
P.S. There is a ChangeLog inside the arhive, but is
Romanian ;) (few parts are English)
__________________________________
Do you Yahoo!?
Yahoo! Search - Find what you�re looking for faster
http://search.yahoo.com
Hello,
is there anybody who uses the Evolution EVS-1 (an old 1 RU FM-, AM-,
PM-synth). My Problem is, that it is only editable using a DOS- or
Atari-ST-software. Is there any linux-editor-software which supports this
synth or a possibility to run the atari-software (e.g. an emulator with
MIDI-support)?
- Thomas
Hello,
Ill be brief as your time is surely valuable. Thank you for your time.
Im trying to set up my new RME hammerfall DSP multiface on my dell inspiron 8000 running linux-2.6.3 with alsa software(drivers) 1.3(a);
Question one: I have (make install)-ed alsa 1.3a drivers, do I have to rebuild my kernel, as alsa is now built into the kernel in 2.6?
Question two: I get this error with pd and the multiface
using non-interleaved audio input
Sample width set to 4 bytes
snd_pcm_hw_params_set_access (output): Invalid argument
using non-interleaved audio
pd: pcm.c:1133: snd_pcm_readi: Assertion `pcm->access == SND_PCM_ACCESS_RW_INTERLEAVED\\\'
failed.
Pd: signal 6
and xmms sees the card (proabably from my .asoundrc file) but wont play through it. Ive run
hdsploader and the little red host error light is gone, so whats up?
Thanks again,
-thewade
Hello,
I would like to use JACKD with two SBLives! soundcards in Surround40
mode (8 output channels and 4 input channels). I have seen on internet
examples of "alsa.conf" how to use SBLive! in Surround40 mode and
also how to use two normal sound cards for jack.
Problem is that I am new to this problematic and I have no idea how to
achieve my desires. Can you please help me with this? I hope it is
possible.
Petr
galan-0.3.0_beta5 is released.
For those who know it already:
One of the new features include cloning. you can now select a bunch of
components and clone them. (This will be changed to a full blown
copy/paste mechanism soon) And the polyphony option will be added soon.
galan is a modular synthesizer similar to SSM and AMS.
The difference is that you build the schematics in one window and
have the controls in another window. So your synth UI is not cluttered
with the schematics.
galan also supports subpatches. You can add patches from the library to
your current patch with 2 mouseclicks.
the LADSPA adsr can be used for real note-on note-off behaviour.
And with the voice allocator it is already possible to build polyphonic
synths. I will refine that process somwhat. But it already works.
When you have setup your schematic and placed the controlling knobs and
sliders on your control panel, you can draw a custom background for the
ControlPanel and import the painted picture into galan. The knobs now
support alpha transparency, so they should integrate onto every
background.
galan supports jack and alsa sequencer. You can open any number of ports
you would like. The internal sequencers can sync to jack-transport.
galan also supports LADSPA. You will have all LADSPA effects handy you
normally use. And you will be able to setup a nice Interface for your
Effects. You would then save them to your sheet library, making them 2
clicks away.
So wiring a jack-inport and another jack-outport to your favorite
plugin, is done in a few seconds.
It even supports the first jack-midi patch (although only as a synth and
as the patch changed recently, there will be no sequencer around :)
perhaps the next patch.
get it at http://galan.sf.net
and look at
http://galan.sf.net/anti-aliased-knobs.png
cause this is how it looks like.
--
torben Hohn
http://galan.sourceforge.net -- The graphical Audio language
hmm.. the M-Audio Delta 1010 is looking like a great deal in the end.
Matthew Polashek
Associate Editor, Silver Burdett Ginn - Music
Scott Foresman/Pearson Education
299 Jefferson Road
Parsippany, NJ 07054-0480
office: 973.739.8709
fax: 973.739.8098
Matthew.Polashek(a)scottforesman.com
> ----------
> From: Mark Knecht
> Reply To: A list for linux audio users
> Sent: Monday, March 8, 2004 2:29 PM
> To: A list for linux audio users
> Subject: Re: [linux-audio-user] Good Sound Card for Pro Multitracking
>
> Polashek, Matthew wrote:
> > I would simply prefer to be able to hit 96K. I definitely need an
> external
> > A/D/D/A unit for sound quality reasons. I can probably use external mic
> > pre-amps. I have been considering using a Presonus DIGImaxLt with a RME
> > Hammerfall 9652 but I think it ends up being cheaper to get an outboard
> box
> > with balanced ins and outs and throwing some kind of 8 channel mic pre
> in
> > front of it, unless there was a less expensive ALSA compatible option
> that
> > had 8 mic pres and 8 outs.
> >
>
> OK. Well, 8-10 inputs of 96K A/D is going to cost you, but I'd guess
> your best bet under Linux is probably an RME card of some type, with one
> of their external 96KHz D/As.
>
> Look into the RME HDSP line. The low end of what might work for you
> would be the HDSP MultiFace. I think the drivers for that work pretty
> well.
>
> Higher end I'd look at maybe a Hammerfall Light (Digi9636) With an ADI-8
> Pro maybe. Good sound. Please double check that the ADI-8 Pro stills
> supports the full 8 channels at 96KHz. It has enough ADAT interfaces to
> do this but I'm finding the web page lacking on this fact.
>
> You'd certainly have more choices sticking with 44.1/48K for lower cost
> converters, but you sound interested in 96K so this limits your options
> with RME cards anyway.
>
> Don't get the stand alone HDSP9652 or HDSP9636 until you make sure the
> Linux drivers will do what you need.
>
> Good luck,
> Mark
>
>
>
>
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