Hello,
Here's one I just made. I am stuck when it comes to find a genre for
this. Rock ? Progressive rock ? Can you suggest something ?
Comments welcomed !
https://soundcloud.com/nominal6/buta
Cheers.
Anyone else unable to reach this?
I've been trying though the afternoon, but each time, after a very long delay
end up with just a blank page - not even an error message :(
Not seeing problems anywhere else.
--
Will J Godfrey
http://www.musically.me.uk
Say you have a poem and I have a tune.
Exchange them and we can both have a poem, a tune, and a song.
Hi, there!
I've searched for this topic and the only thing I got were some obscure
methods to keep some usb-devices from being occupied by pulseaudio...
So, I thought, I'll ask the pros...
Don't want to cause any fight between pro- and con-pulseaudio-users. I
like pulseaudio very much on my laptop, but there are some
circumstances, where I'd rather have more control about pulseaudios
behaviour.
I have a nice firewire device named ESI QuataFire 610, which is
supported by FFADO _and_ ALSA.
Problem is, when I plug in the QuataFire, pulseaudio grabs it. But this
device is not made for desktop audio. After some seconds doing nothing,
pulse sends it to sleep. Waking up, f.e. when the desktop makes some
noise, or - whatever - some jabber-message comes in, does take some
time, so, the event is really over, already, when a tail of a sample
comes through the speakers...
Another one: pulseaudio grabs the QuataFire with a certain samplerate
I'm not supposed to change and I want to start jackd to make some
recording in _another_ samperate, jackd won't start or the QuataFire
gets mad or another unedifying thing happens. That sucks.
Long story short: If I could tell pulseaudio not to care about the
QuataFire (or any other certain device), everything would be easy to
handle. Does pulseaudio has such a feature and how can I use it? (I've
tried the terminal, but I haven't found it.)
And if not - and some pulseaudio-dev is reading this - then put this
feature on the wishlist, please!
Greets!
Mitsch
> Hi,
> I'm having trouble setting up Jack to interface between my digital piano
> application (pianoteq) and the ecasound audio processing app. I'm using
> ecasound with Ladspa plugins to create a crossover network. Ecasound
> splits the 2 pianoteq channels into six (woofer, mid & tweeter), and sends
> them to my analog outputs through alsa. I have it working, with jack &
> qjackctl, but the buffer size that jack is presenting to the digital piano
> app is 1024, about 10x bigger than I want. I would like pianoteq and
> ecasound to run synchronously with minimal latency. At first I thought I
> could just call jackd -d ecasound -p 128, but of course ecasound is not
> one of jack's supported backends. So I've been using the "dummy" backend,
> and using qjackctl to connect ports from pianoteq to ecasound. That works
> fine, but I can't manage to configure the buffer size down to 128, even
> though I start up ecasound with -i jack -b:128, and I also go to setup in
> qjackctl and specify buffer size 128 for "dummy". When qjackctl brings up
> the jack server, the buffer size gets overridden to 1024; I see the message
> in the log. What am I doing wrong? Is Jack the wrong approach, when it is
> ecasound, not jack, that writes to alsa?
> Thanks very much!
> John
> Msi mb with i5 3ghz, AVS Linux
>
>
>
> --
Sent from Gmail Mobile
Hi
I like to announce the first release of GxPlugins.lv2
GxPlugins.lv2 is a set of mostly analogue guitar pedal simulations as
LV2 plugins, simulated with the guitarix ampsim toolkit.
They contain the following plugs:
GxBottleRocket.lv2 - -> tube based preamp pedal
GxHotBox.lv2 - -> tube based preamp pedal
GxVBassPreAmp.lv2 - -> transistor based Bass preamp
GxSuppaToneBender.lv2 - -> transistor based preamp
GxHyperion.lv2 - -> transistor based Fuzz pedal
GxVoodoFuzz.lv2 - -> transistor based Fuzz pedal
GxSaturator.lv2 - -> saturation plugin
GxVintageFuzzMaster.lv2 - -> transistor based Fuzz pedal
GxSuperFuzz.lv2 - -> transistor based Fuzz pedal
GxVmk2.lv2 - -> transistor based solid stage preamp
GxUVox720k.lv2 - -> transistor based solid stage preamp
GxSlowGear.lv2 - -> volume swell plugin
GxGuvnor.lv2 - -> transistor based overdrive pedal
GxToneMachine.lv2 - -> transistor based Fuzz pedal
I hope they may be useful for the one or the other.
Build instruction and screenshots may be found here:
https://github.com/brummer10/GxPlugins.lv2
the release zip file is located here:
https://github.com/brummer10/GxPlugins.lv2/releases
regards
hermann
No problem, Edgar. I think there is a lot of good information in this
thread, I wonder if it is searchable in some way because it could help
others. I really appreciate all the help and all the inputs are helping me
learn more things.
On Jan 21, 2017 4:45 AM, "edogawa" <edogawa(a)aon.at> wrote:
Hi John,
Am 21.01.2017 um 08:26 schrieb john gibby:
> I booted my LiveCD and saw that the dpkg output there is just the same as
> for my regular system (on solid-state drive). I'm glad to know that I have
> jack1 and not jack2, though I'm still not really sure I understand
> completely, since the jack process I get is named jackdbus. For now, I
> stopped using the jack_control shell script and am just using qjackctl to
> bring jack up, and still, the process name is jackdbus. I thought jackdbus
> was jack2...
>
As this has become more and more of a mystery, I have investigated a little
more. reading the AVLinux user manual, particularly the section on software
management, I was able to locate where the jackd packages come from, and
that is in fact the kxstudio repos. once I found that out I was able to
inspect the build logs for the jack package here:
https://launchpadlibrarian.net/229691222/buildlog_ubuntu-xen
ial-amd64.jack-audio-connection-kit_2%3A0.124.2~20151211-2~
xenial1_BUILDING.txt.gz
and it turns out that it has the jackdbus patch applied.
So this part of the mystery is resolved, and your system seems to be in
good shape.
I have to apologize for adding to confusion and leading you on the wrong
track, just remember I'm not familiar with AV Linux (have looked at it
several times in the past 15 years but never regularly used it...) and this
2016 version is significantly different from previous ones in that it
incorporates the kxstudio repos as a core part, which i didn't realize at
first. I'm really sorry for that.
As for your crossover filter network, I've never tried to do such a thig in
software, so I have to pass that on to more knwledgable folks...
Cheers, Edgar
_______________________________________________
Linux-audio-user mailing list
Linux-audio-user(a)lists.linuxaudio.org
http://lists.linuxaudio.org/listinfo/linux-audio-user
Hello,List!
This isn't directly a Linux topic, but many users on this list have
experience with pipe organs. A friend of mine forwarded me this piece:
https://www.youtube.com/watch?v=ii3l44_xD2s
The stop in question, the Vox Humana, can be heard at 1:27. Apparently,
it's unique. Can anyone please tell me a little about how this one works?
i hear a very early formant filter and am wondering just how this might be
mechanically accomplished.
Thanks!
Tom
Hello, Jeannette--
Thank you for the reply.
it's not a formant filter. It's a reed pipe with a resonator. Wikipedia
> has this to say - and probably to show on the matter:
> https://en.wikipedia.org/w/index.php?title=Vox_humana
>
My apologies, Jeannette. "Formant filter" was a little misleading on
purpose, more like bait. But now that a couple people have bit, i'll be
more accurate. Ralf's link to organ stops contains a good demonstration of
a vox humana:
http://www.organstops.org/_sounds/CulverAcademy/VoxHumanaArpeg.mp3
It does change formants slightly, but nowhere near as dramatically as the
basilica organ. So then, how did the boys from Weingarten accomplish such
a drastic change in vowels?
i've found a highly technical article on reed pipes. It may be above my
head, but i'll try to read it before forwarding the link.
Thank you, again, Jeannette! Please take care.
Tom
Hi,
I don't know what solved the issue, but actually the output sound is
ok. It's still not in the same class as RME, but from good hifi quality.
For the cable plugged into channel 2 of the Focusrite only tip
and sleeve are soldered.
For the cable plugged into channel 1 of the Focusrite only tip
and sleeve are soldered, too, but by a crocodile clip cable I connected
and disconnected ring with sleeve [1].
There's neither a difference in volume, nor in tone, with ring and
sleeve short circuit or not.
A big surprise was, that comparing the output of the Focusrite, with
the output of a CD player, everything was ok yesterday, when the
testing was done. I couldn't reproduce the bad sound I experienced
before. The output level of the Focusrite was ok as well. No other
analog audio cables were connected to the other channels. Perhaps some
of the other audio cables, connected to other outputs caused problems by
the tests, I made a while ago.
However, this morning I checked channels 3, 5, 7 and 9 against channel
2, again with a short circuit and without it and after that I compared
channels 4, 6, 8 and 10 with channel 9, but only without a short
circuit. And finally I repeated to compare channel 1 and 2 with a CD
player. Everything sounded very well.
With linux 4.9.0-rt1 it's possible to go down to 32 frames at 48 KHz,
getting lots of inaudible xruns, at 16 frames as well as 64 frames the
interface is completely unusable. At 128 frames it's the same as for a
vanilla kernel with threadirqs, just scrolling with the USB mouse wheel
in roxterm usually does cause issues, very seldom a random xrun could
happen, when switching to another window. At 256 frames it seems to be
absolutely stable, however, it seems to be usable at 128 frames, with
linux-rt as well as a vanilla linux with threadirqs. FWIW using another
USB cable at 128 frames doesn't make a difference. Actually it even
seems to be usable at 32 frames, the xruns were inaudible. I don't know
why it's completely broken at 64 frames. 16 and 64 frames cause long
interrupted audio signals.
Regards,
Ralf
[1] http://picpaste.com/pics/IMG_2250_unbalanced.1484895640.jpg
Hi lista :)
Using an i3 <https://i3wm.org/docs/userguide.html#exec> shortcut, I can
control my jack transport from everywhere, even when no jack application
has focus :
bindsym $mod+p exec echo play | jack_transport
bindsym $mod+Shift+p exec echo stop | jack_transport
Thanks to FalkTX for the pipe trick BTW, I saw that on linuxmusicians ;
This is very cool if you have some sort of wireless keyboard, but as you
can see it's two different keystrokes, how do I make it so <command>
toggles play/pause? Alternatively, how do I know when jack transport is
rolling, so I can hack my way into making my own toggle script? I did my
homework and read the inline --help of all the available jack commands
on my machine :
âš¡ jack_
jack_alias jack_freewheel jack_monitor_client
jack_simple_client
jack_bufsize jack_iodelay jack_multiple_metro
jack_simple_session_client
jack_capture jack_latent_client
jack_net_master jack_test
jack_capture_gui jack_load jack_net_slave jack_thru
jack_connect jack_lsp jack_netsource
jack_transport
jack_control jack_metro jack_rec
jack_unload
jack_cpu jack_midi_dump jack_samplerate
jack_wait
jack_cpu_load jack_midi_latency_test
jack_server_control jack_zombie
jack_disconnect jack_midiseq jack_session_notify
jack_evmon jack_midisine jack_showtime
But found no way to detect the transport status ; Is there a way to know
it?
yPhil
--
Yassin Philip - New album out NOW
http://yassinphilip.bitbucket.org