Over the last few weeks I've been going through Yoshimi with a fine-tooth comb
bug hunting, with some success (and much frustration). I'm doing this on
my 'office' computer which is running debian testing on a 64bit AMD - fairly
similar to my music machine.
During the week, I did an update via synaptic and now suddenly when I try to
compile Yoshimi it fails to link, giving the message:
Linking CXX executable yoshimi
/usr/bin/ld: /usr/lib/x86_64-linux-gnu/libfltk.a(Fl_x.o): undefined reference
to symbol 'dlsym@@GLIBC_2.2.5' /lib/x86_64-linux-gnu/libdl.so.2: error adding
symbols: DSO missing from command line collect2: error: ld returned 1 exit
status make[2]: *** [yoshimi] Error 1
make[1]: *** [CMakeFiles/yoshimi.dir/all] Error 2
make: *** [all] Error 2
will@debian:~/yoshimi_20130926/src$
The investigating I've done suggests that something has changed in cmake so
that yoshimi now needs a more 'correct' identification of fltk. I sort of
gathered that cmake.txt was the place to look, but doing so tells me nothing.
fltk is there but none of the information really makes sense to me.
Although I don't like doing anything experimental on my music machine, I can do
that for the time being as it is running an earlier version of the distro but
presumably the new requirement will eventually hit the stable releases.
Can someone help me sort this out please.
P.S.
A current version of ZynAddSubFX doesn't have this problem, but the build
structure is now very different so I couldn't find any points of comparison.
--
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.
Radium 1.9.31 is a big release with many new features and bug fixes.
1.9.31 is the first release with Pd embedded.
Pd is a " Real-time graphical dataflow programming environment for
audio, video, and graphical processing." (http://puredata.info/)
Pd embedded in Radium has got approximately the same features as Max
for Ableton Live. (https://www.ableton.com/en/live/max-for-live/)
Screenshot: http://folk.uio.no/ksvalast/radiumpd.png
Video 1: http://folk.uio.no/ksvalast/radium_pd.ogv
Video 2: http://folk.uio.no/ksvalast/radium-pd-invertnote.ogv
Radium homepage: http://users.notam02.no/~kjetism/radium/
Most important changes 1.9.30 -> 1.9.31:
* New demo song: BlowFish! Made by www.magnetophon.nl
* Save hashmap elements in sorted order so that songs can more easily
be compared textually
* Menu entry to show name of all included pd externals
* Help menu options to edit keybindings and menues
* Song comment dialog
* Fix "Switch Window Configuration" menu option
* Removed "Error. y2>=window->height: ..." error. Just print to stderr instead.
* Removed the "Something strange just happened in the function
Blt_markVisible" warning (print to stderr instead)
* Option to set number of scrolls per second. Scrolling too often can
be tiresome for the eyes.
* Make it easier to connect objects and see connections in the mixer,
plus adjust object sizes
* Patchbay sound object
* Fix crash loading Soundfonts in the Fluidsynth and Sampler instrument
* Show stars around filename if theres unsaved data
* When quitting or loading, only ask sure/yes/no if edited since last save.
* Change "Set Patch For Track" to "Set Instrument For Track" in the
instruments menu.
The word "patch" should not be exposed to the user anymore.
* Be able to load files with DOS char set
* Changed internal radium block size to 64 (similar to Pd)
* Sending note events between sound objects (green lines)
* Enable undo for on/off effect controllers
* Pd extended is included as a sound object. 921 externals are
included. GUI is working.
Several instances is working. Can be used to write both audio
effects and note effects.
* Fix qt paths on Archlinux (Javafant/archlinux)
* Many minor bug fixes
Hi
I'm working on some midi thing in python. I had a hard time finding a
decent library. In the past I used PySeq alot, but that's tricky to
install and seems un-maintained. So I settled for pygame, and it's going
ok. However:
1) In PySeq I created inports and outports that were visible for
instance in qjackctl. Is that possible in pygame (currently I just
connect to the right client from python)? Why would I prefer one over
way over the other?
2) If I unplug the connected client or press ctrl-c (my preferred way of
exiting my programs) I get:
PortMidi call failed...
PortMidi: `Bad pointer'
type ENTER...
How can I exit a pygame program cleanly without getting errors?
3) One of the things I'd like to do I automatically connect to any
client that shows up while the program is running and *gracefully handle
it* if clients disappear while the program is running. I imagine the
first is a matter of scanning for clients every now and then, but right
now I get the error mentioned in 2) if I pull the plug on a device that
my program is connected to. Any ideas how to handle this?
Any feedback is greatly appreciated!
--
Atte
http://atte.dkhttp://modlys.dk
Hi,
I'm running stock Debian Wheezy (7.1) --- except for a kernel upgrade
(3.10-0.bpo.2-amd64) from Debian backports --- as well as XFCE, on a
couple different hardware configs. However, I'm struggling to get
audio to work properly on my new Intel Haswell board.
(1) On a Thinkpad x220: Audio works great, out of the box
(2) On a Supermicro X10SLQ (Q87 chipset) with Haswell i4770S: Audio
doesn't work at all --- no audio at all comes out of standard
rear headphone jack. (NOTE: Supermicro website identifies audio
capabilities as provided by onboard RealTek ALC888S chipset, but
lspci [below] instead identifies Intel HDA audio, courtesy of
Haswell. I worry that this might be an issue.)
Could this be an issue with the kernel, or should I be looking
elsewhere. I was under the belief that 3.10 (unlike the 3.2 kernel
that shipped with Wheezy) had fairly comprehensive support for Haswell
in general!
Some more information is attached below...
I don't have much experience trouble-shooting audio problems, so I
appreciate any suggestions / pointers.
Thanks!
----------------------------------------------------------------------
guest@haswell:~$ lspci -v | grep -i -A8 audio
00:03.0 Audio device: Intel Corporation Haswell HD Audio Controller (rev 06)
Subsystem: Intel Corporation Device 2010
Flags: bus master, fast devsel, latency 0, IRQ 52
Memory at f0534000 (64-bit, non-prefetchable) [size=16K]
Capabilities: [50] Power Management version 2
Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit-
Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00
Kernel driver in use: snd_hda_intel
--
00:1b.0 Audio device: Intel Corporation Lynx Point High Definition Audio Controller (rev 04)
Subsystem: Super Micro Computer Inc Device 0654
Flags: bus master, fast devsel, latency 0, IRQ 53
Memory at f0530000 (64-bit, non-prefetchable) [size=16K]
Capabilities: [50] Power Management version 2
Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit+
Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00
Capabilities: [100] Virtual Channel
Kernel driver in use: snd_hda_intel
guest@haswell:~$ lsmod | grep snd
snd_hda_codec_realtek 32712 1
snd_hda_codec_hdmi 31720 1
snd_hda_intel 35718 5
snd_hda_codec 122850 3 snd_hda_codec_realtek,snd_hda_codec_hdmi,snd_hda_intel
snd_hwdep 13189 1 snd_hda_codec
snd_pcm 68525 3 snd_hda_codec_hdmi,snd_hda_codec,snd_hda_intel
snd_page_alloc 13018 2 snd_pcm,snd_hda_intel
snd_seq 45186 0
snd_seq_device 13176 1 snd_seq
snd_timer 22773 2 snd_pcm,snd_seq
snd 53068 19 snd_hda_codec_realtek,snd_hwdep,snd_timer,snd_hda_codec_hdmi,snd_pcm,snd_seq,snd_hda_codec,snd_hda_intel,snd_seq_device
soundcore 13026 1 snd
guest@haswell:~$ dmesg | grep -i sound
[ 2.007991] input: HDA Intel MID HDMI/DP,pcm=8 as /devices/pci0000:00/0000:00:03.0/sound/card0/input6
[ 2.008079] input: HDA Intel MID HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:03.0/sound/card0/input7
[ 2.008131] input: HDA Intel MID HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:03.0/sound/card0/input8
[ 2.034368] input: HDA Intel PCH Front Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card1/input9
[ 2.034620] input: HDA Intel PCH Line Out CLFE as /devices/pci0000:00/0000:00:1b.0/sound/card1/input10
[ 2.034834] input: HDA Intel PCH Line Out Surround as /devices/pci0000:00/0000:00:1b.0/sound/card1/input11
[ 2.035018] input: HDA Intel PCH Line Out Front as /devices/pci0000:00/0000:00:1b.0/sound/card1/input12
[ 2.035199] input: HDA Intel PCH Line as /devices/pci0000:00/0000:00:1b.0/sound/card1/input13
[ 2.035383] input: HDA Intel PCH Rear Mic as /devices/pci0000:00/0000:00:1b.0/sound/card1/input14
[ 2.035565] input: HDA Intel PCH Front Mic as /devices/pci0000:00/0000:00:1b.0/sound/card1/input15
guest@haswell:~$
I go all he way back to when music came over the AM radio, clipped,
attenuated, stomped, squashed, and premasticated.
The "stereo" of my day wasn't stereo (yet) and we called it Hi-Fi (for high
fidelity) and we built our own amps or assembled Heathkits.
So now (according to Steve Guttenberg) we are back to stomped and
pre-chewed, except around LAU where we all still roll our own one way or
another.
Pete
http://news.cnet.com/8301-13645_3-57605536-47/an-inconvenient-truth-why-mus…
Hi friends,
I have tryied everything what you mentioned about. But.. can it be a jack
samplerate problem?? When I put samplerate lower - 44.1khz, the glitches
go away.. it is possibile?
Fero
---------- Forwarded message ----------
From: salsaman <salsaman(a)gmail.com>
Date: 2013/10/1
Subject: Re: [estudiolivre] Fwd: [LAU] Lives: some beginner questions
To: estudiolivre(a)lists.riseup.net
Hi Rosea,
the best solution is not to encode to any format, just leave the clips as a
"set". You can do this either by File -> Close/Save all Clips, or when
exiting the application (LiVES will prompt you for a set name).
When you save the set there is a checkbox "auto load this set when LiVES is
started". If you check this, then the next time you start LiVES it will ask
you if you want to reload the set.
You can have any number of sets on the disk, so any time you can just load
the one you want to use. (File -> Load set). You have to close the current
set before you load a different one though.
You can also start up LiVES with any set, by using: lives -set <name_of_set>
See also in the manual:
http://lives.sourceforge.net/manual/LiVES_manual.html#section4.7
Hope this answers your question. Good luck !
Salsaman.
http://lives.sourceforge.nethttps://www.ohloh.net/accounts/salsaman
On Tue, Oct 1, 2013 at 11:26 AM, Daniel Roviriego <danifernando(a)gmail.com>wrote:
> Aeh Salsaman, não sei se você está na LAU..
>
> abs
>
> ---------- Forwarded message ----------
> From: rosea.grammostola <rosea.grammostola(a)gmail.com>
> Date: 2013/10/1
> Subject: [LAU] Lives: some beginner questions
> To: linux-audio-user(a)lists.linuxaudio.org
>
>
> Hi,
>
> Trying out Lives video/VJ tool.
>
> I want to make video art for a live gig. Its mostly making a clip from a
> picture and adding some effects to it, realtime effects and normal effects
> too. Im not planning to add effects live, but I want to apply the live
> realtime effects before the show. Im working in the clip editor. I will use
> Lives to switch to the next clip for the next song.
>
> What is a right format, size and resolution to export to, so I can use the
> clips during the gig? Audio is not needed.
>
> Tips for this workflow is welcome!
>
> Thanks in advance,
>
> \r
>
>
> ______________________________**_________________
> Linux-audio-user mailing list
> Linux-audio-user(a)lists.**linuxaudio.org<Linux-audio-user(a)lists.linuxaudio.org>
> http://lists.linuxaudio.org/**listinfo/linux-audio-user<http://lists.linuxaudio.org/listinfo/linux-audio-user>
>
>
>
> --
> Daniel Roviriego
> (21) 35920701
> (21) 99561654
>
>
--
Daniel Roviriego
(21) 35920701
(21) 99561654
My dear Linux Audio Users
I upgraded the firmware in my Fireface UCX yesterday.
In the Release Notes I saw this...:
"Fireface UCX:
V 42/220/23/4: Class Compliant: full 18 channels, choice of 2 playback
routings (C8/CA)"
I thought I should give it another try in debian Linux wheezy as they changed
something in the class-compliant mode.
And what can I say ... its now recording in Audacity in stereo (channel 1+2).
So if I plug in two mics, I can record! yippieh
So heres what I did: (I hope I remember all steps)
1.) I followed some Howto, I cant remember where I found it, but I created a
.pulse folder in my ~-Folder
Then I pasted these files in the ~/.pulse Folder:
File default.pa:
load-module module-native-protocol-unix
load-module module-jack-sink channels=2
load-module module-jack-source channels=2
load-module module-null-sink
load-module module-stream-restore
load-module module-rescue-streams
load-module module-always-sink
load-module module-suspend-on-idle
set-default-sink jack_out
set-default-source jack_in
#I think I will play with module-jack-source channels=2 to get more input
# channels
File daemon.conf:
sample-format = float32le
default-sample-rate = 48000
realtime-scheduling = yes
exit-idle-time = -1
2.) I logged out of KDE and logged back in
3.) Then I started qjackctl
In the options I chose Input and Output Device hw1 Fireface UCX
4.) Next I started jackd, by clicking start.
5.) I chose a CC setup in the Fireface UCX I created before in Totalmix to
have Phantom Power on both Inputs, by searching SU and then choosing the right
seetup number (refer to the Firefox-manual).
6.) Finally I started Audacity and ..... it was recording both inputs
Yeah!
If I check 96000 Hz in jackd-options, audacity shows 96000 Hz, too, and if I
export it as a wav-file the metadata tells me its 96 kHz, wow! And it sounds
like it, too ;)
greetz & have fun
drz
Hi all!
A quick e-mail to annonce the latest version of the lv2 set avw.lv2.
As a reminder, this set of plugins is designed for Ingen to create
modular synths. Inside you can find VCOs, LFO, VCA and all the main
modules to create Modular Synthesizers.
Parts of these plugins are a port of the internal modules of the great
Alsa Modular Synth.
In this specific version, I focused on bug fixing and usability.
I added as a well an Amplitude Modifier and a plugin to transform VC
to Controls.
This way, you can for example use the LFO (VC amplitude -1/1), use the
Amplitude Modifier to transform the amplitude of the LFO to 1000/3000,
use the plugin VC to Controls and plug all that into a Low Pass
Filter.
Finally, with the source code comes two example that you can load
straight away into Ingen to try the plugins out.
To download:
http://sourceforge.net/projects/avwlv2/files/avw.lv2-0.1.5.tar.gz/download
I really hope you will enjoy those, and I would be really interrested
to get any feedback!
enjoy :)
Aurélien
Thanks, Atte
I had it:
== GUI-enabled checks ==
Checking if you are root... no - good
Checking filesystem 'noatime' parameter... 3.6.11 kernel - good
(relatime is default since 2.6.30)
Checking CPU Governors... CPU 0: 'performance' CPU 1: 'performance' CPU 2:
'performance' CPU 3: 'performance' - good
Checking swappiness... 10 - good
Checking for resource-intensive background processes... none found - good
Checking checking sysctl inotify max_user_watches... >= 524288 - good
Checking access to the high precision event timer... readable - good
Checking access to the real-time clock... readable - good
Checking whether you're in the 'audio' group... yes - good
Checking for multiple 'audio' groups... no - good
Checking the ability to prioritize processes with chrt... yes - good
Checking kernel support for high resolution timers... found - good
Kernel with Real-Time Preemption... found - good
Checking if kernel system timer is set to 1000 hz... found - good
Checking kernel support for tickless timer... found - good
== Other checks ==
Checking filesystem types... ok.
so everytjing seems good... any other ideas ?
--
Fero