Flo, Thanks for the jack suggestion. I definetly do need to spend some
time working with jack on my system. Also, thanks for the great low
latency documentation on tapas - ugh!. After getting 2.6 setup with |
Preemptible Kernel (Low-Latency Desktop), fiddeling with my hardware to
get the soundcard irq's right, and some tweeking with chrt all I can say
is "wow. WOW!".
I don't think there is anything you can do on a 2.4 kernel to get close
to this level of performance and control. Sweet! I almost giggle as the
entire system temporarily hangs while linuxsampler loads gig files at
outrageous speed. Also, I wanted to let you know that after a bit more
hacking I was able to get dshare working with the ice1712. It's awesome.
I can send linuxsampler out of channels 1-4 and have fluidsynth playing
out of channels 5 & 6, with 2 channels to spare for ecasound or
something... all simultaneously, at very low latency, & hardly any
processor load. I paid for the hardware... might as well take advantage
of it. Besides, this way I can give my processor a break, under-clock
it, & keep fan volume very low. Here is a snippet from asound.conf. -Garett
|pcm_slave.66_slave {
pcm "hw:1,0"
channels 8
rate 44100
buffer_size 256
period_size 128
}
pcm.66ch1234_dshare {
type dshare
ipc_key 18273645
slave 66_slave
bindings.0 0
bindings.1 1
bindings.2 2
gindings.3 3
}
pcm.66ch1234 {
type plug
slave.pcm "66ch1234_dshare"
}
pcm.66ch56_dshare {
type dshare
ipc_key 18273645
slave 66_slave
bindings.0 4
bindings.1 5
}
pcm.66ch56
type plug
slave.pcm "66ch56_dshare"
}
Florian Schmidt wrote:
> On Thu, 15 Sep 2005 22:40:32 -0600
> Garett Shulman <shulmang(a)colorado.edu> wrote:
>
>> Hello, I have been fooling around with my alsa asound.conf in an attempt
>> to take advantage of the hardware mixer in my ice1712 and have multiple
>> apps output to it.
>
>
> The hw mixer on the ice1712 is not doing what one normally refers to as
> hardware mixing with consumer grade cards (i.e. allowing several apps
> concurrent access at the same time). It mixes and routes channels from
> its single 10 in/12 out channel device.
>
> The way to get concurrent access with an ice1712 based card is _software
> mixing_. That is, ALSA needs to do it (or whatever sounddriver you use).
>
> I would urge you to take a very long look at JACK though. LinuxSampler
> has excellent jack support and most apps made for peolpe creating music
> usually have jack support, too. You'll save yourself lots of hassles
> (setting up jackd isn't so tough when you read some docs). It is very
> simple to route specific LS patches to specific output channels on your
> ice based card when using jack.
>
>> In order to acomplish this I need linuxsampler to be
>> able to access devices I setup in asound.conf instead of just hw:x,x
>> devices. This was easy enough to acomplish by nuking all of the '"hw:"
>> +' code from AudioOutputDeviceAlsa.cpp. The next issue relates to the
>> fact that the alsa dshare plugin requires that all of the "virtual
>> devices" that I create from the ice1712 card share the same buffer_size,
>> period_size, periods, & period_time settings. So, linuxsampler crashes
>> when it tries to set buffersize and periods. I figured I would just find
>> out what linuxsampler was trying to use for those values, setup the same
>> values in the asound.conf and then comment the code that sets them in
>> linuxsampler. This seems to work except that when linuxsampler connects
>> to the device I notice click and pops from the device & the output from
>> linuxsampler is rather distorted. I suspect that I'm just not setting
>> the values quite right in asound.conf. However, I guess I also may be
>> just mangleing linuxsampler beyond proper function. :) I was wondering
>> if anywone has any suggestions. When linuxsampler tries to set the
>> buffersize it is using FragmentSize=128 and Fragments=2. So I am
>> interpreting this as period_size=128, periods=2, & buffer_size=256. The
>> slave device in asound.conf looks like this:
>> pcm_slave.66_slave {
>> pcm "hw:0,0'
>> channels 8
>> rate 48000
>> buffer_size 256
>> period_size 128
>> periods 2 #I also tried 1 here... counting from 0...
>> period_time 0
>> }
>> Then I create some dshare devices from this and some plug devices for
>> each dshare device and connect linuxsampler to one of the plug devices.
>> Any suggestions would be greatly appreciated. -Garett
>
>
> dshare still won't allow you to have multiple apps use your ice1712
> card. You'll need dmix for that. Have a look at alsa.opensrc.org if you
> want to go this route. dmix will probably kill latency though.
>
> But better take a look at jackit.sf.net.
>
> Regards,
> Flo
>
On Sat, 17 Sep 2005 at 11:20 +0200, Adrian Prantl wrote:
> i'm afraid this is slightly offtopic, but does anyone know of an ogg/
> vorbis plugin for the new iTunes running on 10.4?
XMMS through darwin ports works, and that is what I currently use. I've
heard good things about VLC, too.
While we're on the subject, I've done a little bit of research on the
problem with quicktime. The qtcomponents project that worked before qt7
did things as a component, when they arguably should have made a codec
from the start. Apple hadn't (and hasn't) solidified the API, and so
things stopped working in Tiger and also with QT7 on Panther.
I'm not sure, but if it had been done as a codec to begin with it might
still be working. In any case it looks like doing it as a codec is the
way to go at this stage. I think this will require basically taking
Apple's AudioCodec example and wiring it up to libvorbisfile. I and at
least one other person intend to do this "when I get time." If anyone
out there is good with Apple codecs or good with libvorbisfile, help
would be appreciated, and speed up the process.
See this discussion for more information: http://tinyurl.com/8z6rb
Also this bug reporter got some good info from Apple:
http://tinyurl.com/cy35h
--
Hans Fugal | If more of us valued food and cheer and
http://hans.fugal.net/ | song above hoarded gold, it would be a
http://gdmxml.fugal.net/ | merrier world.
| -- J.R.R. Tolkien
---------------------------------------------------------------------
GnuPG Fingerprint: 6940 87C5 6610 567F 1E95 CB5E FC98 E8CD E0AA D460
On Sat, Sep 17, 2005 at 08:49:27AM +1000, Erik de Castro Lopo wrote:
> Richard Spindler wrote:
>
> > Hi,
> >
> > I'm looking for a convenient wrapper library for ogg vorbis, because
> > what I've found on the xiph.org pages looks a little overengineered to
> > me.
> >
> > I've used libsndfile most of the time, so this is the API Style that
> > I'd prefer, basically I need 4 functions I believe:
>
> Conrad Parker has been working on adding Ogg Vorbis and Speex support
> to libsndfile. Its been very close to working for some time now :-).
aye, if you'd like to play with it, it's in an arch repository at:
http://www.metadecks.org/arch/
The library dependencies etc. were described in this email (which
references Erik's libsndfile--hack branch):
http://music.columbia.edu/pipermail/linux-audio-dev/2005-July/013288.html
cheers,
Conrad.
Hi,
I'm looking for a convenient wrapper library for ogg vorbis, because
what I've found on the xiph.org pages looks a little overengineered to
me.
I've used libsndfile most of the time, so this is the API Style that
I'd prefer, basically I need 4 functions I believe:
what_samplerate_are_you( vorbis_file );
how_long_are_you( vorbis_file );
seek_to_position( vorbis_file, 23456 );
read_data( vorbis_file, buffer );
libvorbis and libogg however have to interact in some obscure way and
I honestly don't want to waste my time reinventing the wheel, while I
could be hacking on a new asskicking application ;)
-Richard
Hello administrators, hello list members,
for my research for a lecture I will be holding on opensource and audio
in October in Stuttgart the following informations would be very
interesting:
How many people are subscribed to the lau / lad -lists. How many
participate actively ? Is it possible to roughly estimate, where they
are from (Amerika, US, Asia, part of Europe, Germany).
Is anybody capable of providing those informations? Thank you very much,
Michael
2.6.13.1-rt6, rlimits-patched pam and configured thus:
# in /etc/security/limits.conf
* - rt_priority 0
* - nice 0
@audio - rt_priority 50
@audio - nice -10
jack starts happily with realtime enabled, and the xruns are very few.
But no jack application will start, they all give this error:
jack_create_thread: error -1 switching current thread to rt for
inheritance: Unknown error 4294967295
It's like the ol' jack is root apps can't connect days, but I thought
rtlimits was supposed to make it possible for a user to get rt_priority.
Am I missing something?
Ah! It came to me just now - I set the priority field in qjackctl to 0
instead of 1 (where it was) and now jack apps can start. Heads up there.
Maybe applications need a way to recognize what rt_priority level to ask
for based on what jack is running at? On a tangent, how exactly does
that work? Is rt_priority=0 sufficiently prioritized? (because it is the
only thing running realtime)
--
Hans Fugal | If more of us valued food and cheer and
http://hans.fugal.net/ | song above hoarded gold, it would be a
http://gdmxml.fugal.net/ | merrier world.
| -- J.R.R. Tolkien
---------------------------------------------------------------------
GnuPG Fingerprint: 6940 87C5 6610 567F 1E95 CB5E FC98 E8CD E0AA D460
>From: Anders Torger <torger(a)ludd.luth.se>
>
>Does anyone know of a tool especially useful for visualizing room
>impulse response measurements? Magnitude responses, waterfall plots and
>such.
Did you find anything?
I just started browsing
http://www.opendx.org
which should be open source. It is a modular visualization
system. Don't know if it helps but the gallery looks advanced.
I have a problem in checking out the CVS, though.
Juhana
--
http://music.columbia.edu/mailman/listinfo/linux-graphics-dev
for developers of open source graphics software
Hi!
I wonder if someone could provide me with the right hint, for I feel it is
not much to get me where I want.
I have an old suse system (gcc3.2 glibc 2.2.5), now I wanted to install a
newer gcc (3.4.3). It compiled fine and I can compile other programs with it.
Yet there is - apparently - some configuration problem. For if I start
linuxsampler or ecasound compiled with new gcc it says, it has a problem with
libgcc_s.so.1. It can't find some symbol VERSION_3_3 or something to that
effect. I tried ldd on those programs and it told me that they contain both
libgcc_s.so.1 from /lib and /usr/local/gcc343/lib. What am I doing wrong. For
I know this can work fine.
I recompiled my libstdc++ (from the 3.4.3 package) with my already installed
compiler to prevent it from including the old libgcc_s, but no effect.
What can I do? Should I put the new library-paths at a specific point in my
/etc/ld.so.conf? Set some special environment variable?
Any help is GREATLY APPRECIATED!
Kindest regards
Julien
--------
Music was my first love and it will be my last (John Miles)
======== FIND MY WEB-PROJECT AT: ========
http://ltsb.sourceforge.net - the Linux TextBased Studio guide
I am having issues getting my FC4 - x86-64 - GCC4.0 box to compile
swh-plugins cleanly. I have applyed the patches that I have seen on
this list but I still get errors ranging from not determining the proper
-march value to simply failing with the following error:
/bin/sh ./libtool --tag=CC --mode=link gcc -march=k8 -O3 -module
-avoid-version -Wc,-nostartfiles -o hermes_filter_1200.la -rpath
/usr/local/lib/ladspa hermes_filter_1200.lo util/libblo.a -lrt -lm -lm -lm
*** Warning: Linking the shared library hermes_filter_1200.la against the
*** static library util/libblo.a is not portable!
gcc -shared .libs/hermes_filter_1200.o util/libblo.a -lrt -lm
-march=k8 -nostartfiles -Wl,-soname -Wl,hermes_filter_1200.so -o
.libs/hermes_filter_1200.so
/usr/bin/ld: util/libblo.a(libblo_a-blo.o): relocation R_X86_64_32
against `a local symbol' can not be used when making a shared object;
recompile with -fPIC
util/libblo.a: could not read symbols: Bad value
collect2: ld returned 1 exit status
make[2]: *** [hermes_filter_1200.la] Error 1
make[2]: Leaving directory `/usr/local/src/audio_apps/swh-plugins-0.4.13'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/usr/local/src/audio_apps/swh-plugins-0.4.13'
make: *** [all] Error 2
Any help would be appreciated.
Does anyone know of a tool especially useful for visualizing room
impulse response measurements? Magnitude responses, waterfall plots and
such.
I'd prefer something more efficient and quicker to work with than octave
and gnuplot.
/Anders Torger