Hi
I have a little confusion with some real time kernel settings, sure will be
hardware dependent, but ... I really interested to discuse of some settings
like the following and others:
--- Enable the block layer
[ ] Support for Large Block Devices
[ ] Support for tracing block io actions
[ ] Support for Large Single Files
[ ] Block layer SG support v4 (EXPERIMENTAL)
IO Schedulers --->
Processor type and features --->
HPET Timer Support
Device Drivers --->
Character devices --->
<*> Enhanced Real Time Clock Support
[ ] Real Time Clock Histogram Support
<*> Priority Inheritance Debugging (Blocker) Device Support
[ ] HPET - High Precision Event Timer
Sound --->
Advanced Linux Sound Architecture --->
<M> RTC Timer support
[*] Use RTC as default sequencer timer
I have this settings ..... but .... ALSA don't load snd_rtctimer
running lsmod I can only see snd-timer .... and I need to load snd-rtctimer by
hand ... but in fact I don't know what module is used.
Thanks
Josep
I have some general questions about sound in linux; the interplay of
pulse/(esd) and gstreamer,jackd and alsa.
Let me state what I think I know.
Alsa provides the drivers that allow sound to be produced and also the
midi subsystem. You can use just the alsa system if you want to use
audio or midi, but if you want to use (possibly) more than one program
that uses sound eg the system sounds that gnome and kde make when alerts
happen and say a music program, you need to use something like pulse or
esd, or dmix in alsa. This is because the program captures the whole use
of the sound card and can't release it for other programmes. If you want
to use midi and audio together as in rosegarden, you need jackd because
that allows more than one program to access the soundcard at a time. It
is also useful because if you have the right kernel and privileges your
audio programme can override the right of other programmes to access the
kernel as well meaning that you don't get glitches when you are
recording a masterpiece. Gstreamer fits in the normal esd/pulse picture
by providing the access to codecs that files come in. Now this is where
my understanding fails: gstreamer has a jack module (so I hear rumoured,
but I can't find it in the ubuntu reps) and so does pulse. How do these
fit in with the jackd stuff? Obviously they both make it so that jack
can work with both or either of the sound servers.
I welcome corrections in my interpretation of what's happening.
Shelagh
chris beagles wrote:
> Is there any where you can recommend that I can get free samples from?
From time to time some suggestions pop up here, but I can't really
recommend any site in particular since I rarely shop there. It also
depends heavily on your taste and style of music...
> Google brings up loads of results. Have always used soundfonts and live
> synths to create music.
Soundfonts wouldn't bee too taxing on the cpu either AFAIK. Try
fluidsynth or one of it's guis, most notably Qsynth and
fluidsynth-dssi.so (the later for instance with ghostess). Soundfonts
are of course much easier to deal with than single samples when we're
talking about multi sampled instruments like piano.
BTW: The best (free) piano I heard is
ftp://musix.ourproject.org/pub/musix/sf2/Steinway_IMIS2.2/Steinway_IMIS2.2.…
For drums, I always design a kit for each song, and specimen is
super-super fast at doing that.
> Will have a read through the online pd help manual, never managed to
> grasp it before but i shall try and persevere this time!
It's not that tricky. And the community is very helpful.
--
peace, love & harmony
Atte
http://atte.dk | http://myspace.com/attejensenhttp://anagrammer.dk | http://modlys.dk
Hi People,
I have two questions, or maybe we can call it, surveys.
First: Is there anyone here who uses LASH? I tried it but I didn't found
it useful, am I missing something?
Second: Yesterday I was told to try the OSS V4.0, because it is -- or so
I was told -- better than ALSA. It goes against what I think to be the
common opinion around here.
Thanks,
Lobo
Afternoon list,
Just curious about the possibilities of making music with my lowend laptop.
It is a Compaq Armada E500 with a Pentium 3 800MHz and 192MB of RAM.
Im running ubuntu gutsy (rc1) with kernel 2.6.22-14-rt and openbox.
Im wondering about what programs I can use to compose that wont completely
overload my laptop.
I normally use ardour, hydrogen, qjackctl, qsynth, seq24 and zyn in my
studio at home. But this laptop definitely wont handle all of that (or even
much of that).
Is it feasible to be able to do any sort of composition with this sort of
setup?
Cheers,
Chris
Geoff Beasley wrote:
> You are not running a RT kernel; that's your problem. For any serious
> audio work you MUST run a RT kernel. You can roll your own quite easily
> if you wish, or use one of a number of specially prepared audiocentric
> distro's for linux; eg: Planet CCRMA, 64 Studio,Musix and many others.
Aha! It turns out that there is no official Arch RT kernel (for whatever
reason). However, some nice person has a feed with an RT patched kernel
for Arch- I've installed that and now I have no xruns- woohoo!
Thanks
Simon
In my attempt to stop the flipping xruns I get even from a really simple
jack/fluidsynth (with 6.5mb soundfont) setup, I'm going to see if I can
get it to run without xruns at all- so I'm going to stop the X server
(which takes up quite a lot of cpu and memory).
Before I can do this, I need to work out how on earth to connect
fluidsynth to alsa_pcm without qjackctl. Why are there no man pages or
--help or anything for jack_connect, etc? I've tried the obvious:
"jack_connect fluidsynth alsa_pcm", but that doesn't work and without
any usage information whatsoever I'm completely stuffed. Also, a command
line list of port names that can be connected would be nice.
Thanks
Simon
It's probably hardwired and dependent on the mainboard but I
thought I'd ask this anyway, just in case someone has an idea.
I notice Chuckks' /proc/interupts is like this below whereas
my Abit AN-M2 (nForce4) setup follows... is there anyway I can
manipulate my interupts so the USB devices line up on a single
IRQ like Chuckks ?
CPU0
0: 7451384 IO-APIC-edge timer
1: 16910 IO-APIC-edge i8042
8: 0 IO-APIC-edge rtc
12: 91809 IO-APIC-edge i8042
14: 55194 IO-APIC-edge ide0
16: 17785 IO-APIC-fasteoi eth1, HDA Intel
17: 124946 IO-APIC-fasteoi bcm43xx
19: 41988 IO-APIC-fasteoi ohci_hcd:usb1, ohci_hcd:usb2, ehci_hcd:usb3
20: 1 IO-APIC-fasteoi yenta, tifm_7xx1
21: 3228 IO-APIC-fasteoi acpi
CPU0 CPU1
0: 136 561 IO-APIC-edge timer
1: 49 27195 IO-APIC-edge i8042
8: 0 0 IO-APIC-edge rtc0
9: 0 30 IO-APIC-fasteoi acpi
14: 0 0 IO-APIC-edge libata
15: 0 0 IO-APIC-edge libata
16: 13465 13271804 IO-APIC-fasteoi ohci_hcd:usb1, ahci
17: 13523 11359306 IO-APIC-fasteoi ohci_hcd:usb2, eth0
18: 4 2906 IO-APIC-fasteoi ehci_hcd:usb3, HDA Intel
19: 5332 4029220 IO-APIC-fasteoi ehci_hcd:usb4, nvidia
I have a problem with distorted USB sound (Tascam US-122)
but no matter what USB port I use it is always doubled up
with another device. If I could "force" all the USB busses
onto one IRQ (or ideally ohci_hcd on one and ehci_hcd on
another), and made sure my US-122 was the only USB device,
then that might make a difference.
--markc
This is first release of the LV2 vocoder.
Code is based on version 0.3 of LADSPA plugin created by Josh Green.
LADSPA plugin created by Josh Green is basically an adaption of
Achim Settelmeier's Vocoder program to LADSPA.
Code and screenshots can be found here:
http://nedko.arnaudov.name/soft/lv2vocoder/
If you get DNS resolve problems, you can try this URL instead (may not
work in future):
http://triton.atia.com/nedko/soft/lv2vocoder/
--
Nedko Arnaudov <GnuPG KeyID: DE1716B0>