There was a comment a while ago about not listing rtc in rtirq a while
ago, that there are other better timers to use on modern systems anyway.
Does audio SW generally use these other timers by default then? Is it
worth while removing rtc from my rtirq config?
The default rtirq list is:
rtc snd usb i8042
I have found that in the case of two sound cards it is best to be more
specific:
rtc snd_ice1712 snd i8042
(I have removed usb because I don't use it for anything)
The other question is why is i8042 included? Is it used for more than
kb/mouse? It seems odd that the kb/mouse has higher priority than disk
writes for example.
For use of things like netjack (or other AoIP sw), should eth0 (or
whatever eth is used for audio) be added to this list? (if one eth is used
for audio and the other for Inet, then separating them into hi and lo
priority makes sense)
Asking questions because I don't know, maybe disk writes are buffered well
enough it doesn't matter.
I have not had problems with xruns with the setup I have, but would like
to know anyway.
--
Len Ovens
www.ovenwerks.net
This would not be portable across os... and require a config file. I am
wondering if it is possible.
I am thinking it would be nice to have an lv2 alsa trimpot. The idea is
that it would not affect the audio passing through, but provide a control
that varies the level of an alsa control. This is so a mixer strip that
was composed of plugins would be able to also control the alsa preamp
level if it was available. I realize figuring out which channel goes to
which strip at any one time (the same input channel might be routed to a
number of channels at the same time) would be _difficult_. Knowing which
control changes the level one wants might be hard to automate too. The
user interface would require the ability for the user to set it (them). If
there was more than one for the same control they would have to track as
well (alsa mixers seem to do this already though).
The easy way to do this would be to throw away incoming audio and select
our own. Some sort of middleware that gave a common interface for all AIs
would make this much easier too. Mudita24 has MIDI control for this... but
alas the midi control does not extend to the pre levels (ADC levels) where
it would be most useful (or at least moving these controls outputs no midi
the way the multimix controls do).
So far as I know, nobody does this, even in OSX/windows land
though of course it is common on both analog and digital mixboards even
when the pre is at the other end of a snake or on a foreign interface.
--
Len Ovens
www.ovenwerks.net
(Please excuse if this is a stupid question...)
I am learning Ardour doing some practice on my system (Debian stable, RT
kernel, RT tuning, MAudio Audiophile 2496 PCI, 3GHz 4core CPU, 8GB RAM,
JACK buffer 512). Working with a 3min stereo audio (24bit, 88.1kHz) I
have noticed DSP goes to 100% when exporting (WAV)...
Is it normal (simply Ardour is using all the available resources during
the export) or something is going wrong (and I will have glithces/pops
in the output file)?
Thanks
--
al3xu5 / dotcommon
Support free software! Join FSF: http://www.fsf.org/jf?referrer=7535
Are there any real benefits to building software on your local machine vs installing binaries, in this case all largely Ubuntu based?
I've finally got this old mackbook 1,1 (Nov 2006 white 13") running linux audio with comparable results to its native OSx 10.6. Whatevah Snow LeopRd was. Set up dual boot.
Basically installed "Linux-lite", pulled in kxstudio goodies and kernel. 3.13 seems to work best on T2500 dual core @ 2ghz. I can run 12 live tracks recording at 44.1/24 with 128/2 & almost zero xruns. Lots of neat hardware in the Mac for a 'puter of this vintage ... Only 32 bit. Processor is 64bit but some EFI issue I believe, kills ability to boot 64bit systems.
I've removed lots of software I won't use on it like cups, samba, misc daemons nibbling away at memory n cycles. Xfce desktop.
Can I gain any performance from building certain software on the machine? Mostly, I guess low or more efficient resource usage. Kernel included. Right now running SMP lowtatency kernel. I'm still if the belief that a RT kernel would be better but obviously in the deb world, not so many available.
Thanks for any input.
~ Russell
Hi!
When QJackCtl starts up now, it always displays the message window with
this showing in it:
21:30:58.980 Patchbay deactivated.
21:30:58.999 Statistics reset.
21:30:59.007 Could not open ALSA sequencer as a client. ALSA MIDI
patchbay will be not available.
ALSA lib seq_hw.c:457:(snd_seq_hw_open) open /dev/snd/seq failed: No
such file or directory
Cannot connect to server socket err = No such file or directory
Cannot connect to server request channel
jack server is not running or cannot be started
Happens whether I've got it set to use RAW or SEQ midi.
I don't have QJackCtl set to start JACK on startup, I start it manually.
When I start it manually, it starts up and everything audio works fine.
But if I set it to use SEQ, there are no entries in MIDI or ALSA tab. If
I set it to use RAW, it finds my E-MU XMidiX1 and adds 1 in and 1 out
port. That puts only 1 midi_capture_1 port on the MIDI tab, though. It
apparently no longer finds it as an outport (I can tell because the OUT
port light on my interface doesn't come on.
These are the snd kernel modules lsmod shows:
snd_usb_audio 132946 2
snd_usbmidi_lib 19427 1 snd_usb_audio
snd_rawmidi 18422 1 snd_usbmidi_lib
snd_seq_device 5371 1 snd_rawmidi
snd_hda_codec_via 19628 1
snd_hda_codec_hdmi 36654 1
snd_hda_codec_generic 51198 1 snd_hda_codec_via
snd_hda_intel 21867 4
snd_hda_controller 17418 1 snd_hda_intel
snd_hda_codec 89695 5
snd_hda_codec_hdmi,snd_hda_codec_via,snd_hda_codec_generic,snd_hda_intel,snd_hda_controller
snd_hwdep 6228 2 snd_usb_audio,snd_hda_codec
snd_pcm 78539 5
snd_usb_audio,snd_hda_codec_hdmi,snd_hda_codec,snd_hda_intel,snd_hda_controller
snd_timer 18102 1 snd_pcm
snd 58113 24
snd_usb_audio,snd_hwdep,snd_timer,snd_hda_codec_hdmi,snd_hda_codec_via,snd_pcm,snd_rawmidi,snd_hda_codec_generic,snd_usbmidi_lib,snd_hda_codec,snd_hda_intel,snd_seq_device
soundcore 5359 2 snd,snd_hda_codec
usbcore 165281 11
uas,btusb,snd_usb_audio,uvcvideo,usb_storage,snd_usbmidi_lib,ehci_hcd,ehci_pci,usbhid,xhci_hcd,xhci_pci
When I start Rosegarden and go to Manage MIDI Devices, it reports no
MIDI input or output port.
Using QJackCtl 0.3.12 (QT 4.8.6) on 64-bit Aptosid (Debian Sid) kernel
3.19.0-1.slh.4-aptosid-amd64. And apparently the old ALSA package has
been turned into a dummy and replaced with kmod?
Any ideas? Thanks.
See what happens when you upgrade things and *everything just seems to
work* except for weird little things like this? Everything before
upgrade was working with no problems. Stupid computers. ;)
--
David W. Jones
gnome(a)hawaii.rr.com
authenticity, honesty, community
http://dancingtreefrog.com
I've just pushed a jcm800 simulation lv2 plugin to the guitarix git
repository.
This one is for those who are more behind the lead sounds. Guitarix
users know it already as gx plugin, but for the lv2 format I've added
tone, presence and master controls to the plug, so that the full jcm800
is simulated.
Check it out and have fun.
|git clone git://git.code.sf.net/p/guitarix/git guitarix-git|
Hi Athanasios,
> i wonder if I can redirect audio from my android mobile phone to a jack
> server (preferably) or pulse audio server (not so preferably) through
wifi,
> in much the same way this can be done through bluetooth.
Linux-Audio and pulse is not a good partnership and for using jack you
have to build yourself the Android and for sure it has to be rooted.
Not everyone can do this or wants to root his phone. An Android 4.4.2
device is needed. The code and instruction you can find on GitHub.
https://github.com/KimJeongYeon/jack2_android
The next problem will be wireless transfer for Jack. As far as I know
netjack
is not designed for wireless connections. So I'm still looking for a
solution.
Google has a long history in the Android latency bug, many developers
left to IOs. Since Android 4.4 and the efforts by Google in 2013/14 audio
programming starts to make sense in Android, but in C++ and not in Java
less then Android5.
Yosef
DrumGizmo version 0.9.8.1-hotfix released!
After releasing 0.9.8 we discovered a rather serious bug in the
resampling code that would cause sample skewing over the channels when
resampling was enabled. This release fixes that.
Download it from http://www.drumgizmo.org
Visit us at the official irc channel at the Freenode network. Channel
name #DrumGizmo. We would love to hear from you!
// The DrumGizmo team