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
Dear list,
the following problem is giving me headache since a very long time:
My RME Multiface connected via the ExpressCard to my Lenovo X201s
running Debian/GNU testing is 20% of the time detected as Digiface when
inserting the card into the computer. Dmesg tells me that
Hammerfall-DSP: Digiface found
I have to eject the card multiple times and power cycle the Multiface
until it is eventually randomly detected as the multiface it is.
Does anyone have any idea why this could happen? Does anyone have
similar experiences?
thank you so much!
Peter
Some folks are still contributing by writing RT patches for current
kernel version. Several people who aren't able to contribute by
programming are willing to help by testing. So far, so good. How can we
provide a useful bug report, when the distros we're using comenmwith
systemd? IOW journalctl rules.
I send requests to archaudio-discuss(a)archaudio.org and to
linux-rt-users(a)vger.kernel.org , but journalctl is an issue.
For more than a decade I'm using Linux only, but I'm unable to find a
way to get useful journalctl output.
Any ideas?
I apologize in advance for what seems like a trivial and often asked
question. But I can't seem to figure out even the tools to diagnose my
problem. I've been reading web pages and alsa and jackd documentation for
two weeks, with no luck.
I'm trying to configure my ubuntu 14.04 system to use my Akai EWI USB midi
controller. I am trying to use Ted's Linux MIDI Guide
<http://tedfelix.com/linux/linux-midi.html> to configure things. But before
I get midi working I need to test that jackd can route audio to alsa, and I
can't get audio ouput from jackd.
I have verified that alsa works with pulseaudio, when pa is running. I
disabled pa and then made sure that pa did not autospawn (autospawn = no in
/etc/pulseaudio/client.conf.)
I also verified that my hardware device is hw:0. It's the only HW device in
my machine. The EWI is not currently plugged in, and it's a MIDI device
anyway.
When I run qjackctl and start the server I get this:
19:04:28.770 D-BUS: JACK server was started (org.jackaudio.service aka
jackdbus).
Sat Mar 7 19:04:28 2015: Starting jack server...
Sat Mar 7 19:04:28 2015: JACK server starting in realtime mode with
priority 10
Sat Mar 7 19:04:28 2015: Acquired audio card Audio0
Sat Mar 7 19:04:28 2015: creating alsa driver ...
hw:0|hw:0|128|2|44100|0|0|nomon|swmeter|-|32bit
Sat Mar 7 19:04:28 2015: configuring for 44100Hz, period = 128 frames (2.9
ms), buffer = 2 periods
Sat Mar 7 19:04:28 2015: ALSA: final selected sample format for capture:
32bit integer little-endian
Sat Mar 7 19:04:28 2015: ALSA: use 2 periods for capture
Sat Mar 7 19:04:28 2015: ALSA: final selected sample format for playback:
32bit integer little-endian
Sat Mar 7 19:04:28 2015: ALSA: use 2 periods for playback
Sat Mar 7 19:04:28 2015: graph reorder: new port 'system:capture_1'
Sat Mar 7 19:04:28 2015: New client 'system' with PID 0
Sat Mar 7 19:04:28 2015: graph reorder: new port 'system:capture_2'
Sat Mar 7 19:04:28 2015: graph reorder: new port 'system:playback_1'
Sat Mar 7 19:04:28 2015: graph reorder: new port 'system:playback_2'
Sat Mar 7 19:04:28 2015: graph reorder: new port 'system:playback_3'
Sat Mar 7 19:04:28 2015: graph reorder: new port 'system:playback_4'
Sat Mar 7 19:04:28 2015: graph reorder: new port 'system:playback_5'
Sat Mar 7 19:04:28 2015: graph reorder: new port 'system:playback_6'
Sat Mar 7 19:04:28 2015: graph reorder: new port 'system:playback_7'
Sat Mar 7 19:04:28 2015: graph reorder: new port 'system:playback_8'
Sat Mar 7 19:04:30 2015: Saving settings to
"/home/poppa/.config/jack/conf.xml" ...
19:04:31.000 JACK connection change.
19:04:31.031 Server configuration saved to "/home/poppa/.jackdrc".
19:04:31.032 Statistics reset.
19:04:31.038 Client activated.
19:04:31.043 JACK connection graph change.
Sat Mar 7 19:04:30 2015: New client 'qjackctl' with PID 27579
(It complained initially that the server was not running, but that's
because it wasn't.)
export JACK_PLAY_CONNECT_TO=system:playback_%d
jack.play test.wav
I have played this .wav file before with aplay, so I know it plays, but no
sound comes out of my speakers. I think jack thinks it's playing, since
qjackctl tells me this when I run jack.play:
19:04:31.043 JACK connection graph change.
Sat Mar 7 19:04:30 2015: New client 'qjackctl' with PID 27579
19:07:09.840 JACK connection graph change.
19:07:09.955 JACK connection change.
Sat Mar 7 19:07:09 2015: New client 'jack.play-27607' with PID 27607
Sat Mar 7 19:07:09 2015: Connecting 'jack.play-27607:out_1' to
'system:playback_1'
Sat Mar 7 19:07:09 2015: Connecting 'jack.play-27607:out_2' to
'system:playback_2'
19:07:10.772 JACK connection graph change.
19:07:10.959 JACK connection change.
Sat Mar 7 19:07:10 2015: Disconnecting 'jack.play-27607:out_1' from
'system:playback_1'
Sat Mar 7 19:07:10 2015: Disconnecting 'jack.play-27607:out_2' from
'system:playback_2'
Sat Mar 7 19:07:10 2015: Client 'jack.play-27607' with PID 27607 is out
I know it's some simple configuration error I have made. I have no
.asoundrc nor an
/etc/asound.conf file, so I have the default configuration. I also have
the most common
kind of sound card in the world, so it can't be that hard to configure it.
Can anybody give me a clue? Thanks.
P.S.
FWIW, alsa is working fine, so I don't think that's the problem. But
catting /proc/asound/cards gives me:
Code:
poppa@ossian:~/.config$ cat /proc/asound/cards
0 [SB ]: HDA-Intel - HDA ATI SB
HDA ATI SB at 0xfdff4000 irq 16
Also, alsamixer tells me the levels are fine and not muted.
P.P.S. The EWI works with alsa and fluidsynth alone, but there seems to be
a noticable latency between fingering notes and hearing the notes. I'm
hoping jack will fix some or all of that.