[LAD] Re: Linux Audio Conference (is back) 2025

[Jörn Nettingsmeier]

On 6/13/24 11:29 PM, Kevin Cole wrote:



On Mon, Jun 10, 2024 at 5:19 PM Fons Adriaensen <mailto:f...@linuxaudio.org>> wrote:


When LAC 2020 was announced I wrote:

   I expect some decent wine.


Belgian beers are more my "go-to". 😁


As in, considered harmful by some, but very nice if used sparingly?

Some experiences I've had with Belgian beers were more akin to memory 
barriers... stupid me was enjoying Duvel without parsing the metadata 
(the taste is so light and summer-y), matching my friends' cadence (who 
were drinking pilsener). One of the few evenings where I don't remember 
how I got home.


Great news to hear LAC is back, and as I'm clawing back control of my 
work-life-hack balance, I've rammed it into the bedrock of my calendar 
with a solid concrete base and some barbed wire around it. See if any 
client can move it :o) Very much looking forward! And then, wine (or 
French beers) it will be.


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Re: [LAD] Is Piperware a successor to Jack/Pulseaudio?

[Jörn Nettingsmeier]

On 7/2/21 1:32 AM, M. Edward (Ed) Borasky wrote:

The particular incident that relates to Pipewire arose from the latter
category - I saw some interesting writing about Pipewire and wanted to
experiment with it on the NVIDIA Jetsons. They ship with an
NVIDIA-supported operating system called Linux for Tegra (L4T), which
is arm64 Ubuntu 18.04 LTS "Bionic Beaver" with some modifications and
enhancements for the hardware platform. When I downloaded Pipewire and
tried to install it from source, it did not build because some
libraries on 18.04 are too old.


I assume the Jetsons are not your everyday machines, probably even headless.
I would argue that the main reason to run pipewire is seamless 
integration of pro-audio needs with pulseaudio convenience on your 
everyday office machine.


So if you "just" want to integrate the jetsons into you audio production 
workflow, install jack and zita-njbridge and never look back. Also makes 
for a lot more deterministic system.


As to backporting: that is a burden on the developers that takes 
resources away from developing. Pipewire is a fast-moving, very new 
project. You are on a customized embedded (and thus a little slower 
moving) platform. That is a problem, but if you want to combine embedded 
with cutting edge, you have to find a platform where the vendor tracks 
the latest stuff. The only community big enough to warrant that expense 
right now and deliver something close to "latest" is the Raspberry Pi, 
and, to a lesser degree, Armbian-supported boards. I know it doesn't 
help you, but I guess it's a fact of life.


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Re: [LAD] update of zita-njbridge

[Jörn Nettingsmeier]

On 4/16/21 10:03 AM, Fons Adriaensen wrote:

Hello all,

zita-njbridge 0.4.8 is now available at the usual place.

Changes:

* added --ipv4 and --ipv6 options
* removed some unused code
* some minor fixes


Sigh. There goes my last excuse to not learn about IPv6 multicasting. :o)

Thanks Fons for your amazing stack of free audio software!


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Re: [LAD] some updated RT programming wisdom

[Jörn Nettingsmeier]

On 11/18/20 6:04 PM, Jörn Nettingsmeier wrote:
It may be old news for seasoned programmers, but I found this to be a 
trove of information that goes a bit beyond of the traditional wisdom of 
"what's safe in a jack process() thread", so I thought I'd share it here 
(courtesy of LWN's free subscriber link feature, please check out LWN!)


https://lwn.net/SubscriberLink/837019/e323ab1009054668/

https://ogness.net/ese2020/ese2020_johnogness_rtchecklist.pdf



And I have a question: since nobody seems to pre-fault their stacks and 
heaps: is that the reason for xruns when adding new clients to the JACK 
graph? We all know not to malloc() in the process thread, but of course 
we may be using malloc'ed memory for the first time...





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[LAD] some updated RT programming wisdom

[Jörn Nettingsmeier]

It may be old news for seasoned programmers, but I found this to be a 
trove of information that goes a bit beyond of the traditional wisdom of 
"what's safe in a jack process() thread", so I thought I'd share it here 
(courtesy of LWN's free subscriber link feature, please check out LWN!)


https://lwn.net/SubscriberLink/837019/e323ab1009054668/

https://ogness.net/ese2020/ese2020_johnogness_rtchecklist.pdf


All best,


Jörn




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[LAD] IRT gets it: open-source libraries for object-based audio

[Jörn Nettingsmeier]

Hi *!


Haven't checked this out in any detail, but the German Institut für 
Rundfunktechnik (a sort of science outsourcing provider for public 
broadcast over here, with an impressive track record of ground-breaking 
work) have released a substantial body of code on object-based audio 
under an Apache open-source license:


https://lab.irt.de/more-open-source-for-open-object-based-audio-workflows/

If anyone gets interested and finds it useful, please give them a little 
shout-out - I guess our fellow open-source enthusiasts within those 
institutions can use some support!



All best,


Jörn



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Re: [LAD] jack meterbridge 0.9.2 update / maintenance

[Jörn Nettingsmeier]

On 7/29/19 8:44 PM, Robin Gareus wrote:

On 7/29/19 8:21 PM, Stephan Bourgeois wrote:


Should meterbridge be maintained or should another metering app de bundled
with jack?


Given that jack is cross-platform and meterbridge is a X11 application
it's not very likely to be bundled with jack itself.

The main use-case for meterbridge is the dpm mode with many channels to
check input activity.

You may also want to have a look at
https://lists.linuxaudio.org/archives/linux-audio-dev/2012-June/032475.html
- the debian package applied this, but it never reached upstream, it's
likely not relevant after your re-work either.


A better candidate to be bundled with JACK would be 
https://github.com/njh/jackmeter (although I tend to prefer jack_capture 
/dev/null, b/c it does multiple channels). A simple command line meter 
for troubleshooting would be great. But ballistics are of secondary 
importance, this is more to check if it's alive. That also means it 
should go down to -100dB or so.


That said, a "signal present" indicator would help in all patchbay 
applications such as patchage and qjackctl (switchable of course to not 
waste resources when not needed).



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Re: [LAD] USB audio on RPi3B+

[Jörn Nettingsmeier]

On 07/07/2018 05:47 PM, Fons Adriaensen wrote:

Hello all,

Has anyone tried using multichannel USB audio on a Raspberry 3B+ ?


Oh, and what distro are you running? I found that on raspbian I had to 
do "rpi-update" at some point (which updates the firmware, some core 
libraries and the kernel to a version newer than Debian stable)...







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Re: [LAD] USB audio on RPi3B+

[Jörn Nettingsmeier]

On 07/07/2018 05:47 PM, Fons Adriaensen wrote:

Hello all,

Has anyone tried using multichannel USB audio on a Raspberry 3B+ ?

It seems to work perfectly (using zita-alsa-pcmi) with stereo cards.

When I try my RME Babyface (12 in, 12 out) in CC mode, the device
opens without problems, but then Alsa_pcmi::pcm_wait() times out
waiting for the poll fd to become ready. Timeout in pcm_wait() is
1000 milliseconds.

The same seems to happen with Jackd, which uses similar code.

Is there anything in the Pi's system or configuration that
excludes multichannel cards ?


Not that I'm aware of. I've used an older Pi2 with a Gigaport HD+ USB 
device (eight RCA outputs at 44k1) with usable results for 5.1, and my 
KODI-based media center plays 5.1 movie content over (albeit over HDMI) 
flawlessly without breaking a sweat, ever.


If your quality requirements are modest you might want to look at the 
AudioInjector Octo I²S sound card for the Pi. I have two from an early 
series with known manufacturing defects, but if the developer (a very 
open-sourcey hardware hacker guy that is generally very helpful) has 
managed to get his chinese manufacturer to ramp up the quality control, 
it might be an option.

I did a few measurements with this card a while ago:
http://forum.audioinjector.net/viewtopic.php?t=3090

I have never used a professional USB multichannel card on the Pi since I 
don't own one (still stuck with aging ExpressCard Multi- and MADIfaces). 
But I'm trying to borrow one for testing becaus I have a similar usecase 
in the near future...



All best,

Jörn


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[LAD] Forgive me, for I have sinned, or: toss your Macintosh, as fast and wide as you can.

[Jörn Nettingsmeier]

Here's me, having to deal with a 48 channel live recording over the 
course of six shows. Since my MADI gear is kinda heavy and the rental 
company had a Dante system on offer, I dusted off the 2013 Macbook pro I 
bought used, purchased a Dante virtual soundcard license from Audinate 
and happily tracked the first three shows with Ardour.


On the train on an off-day, I started a rough mix-down so that the 
client can begin the selection process. In the middle of exporting, my 
Mac shuts down and boots into a PIN unlock screen, telling me it has 
been locked via "Find-my-Mac".


For the record, this Macbook had been purchased from a reputable large 
online dealer, and it had been factory-reset and completely installed 
from scratch.


The first thing I find as I frantically research this issue (on my 
proper laptop, that is controlled by me, not by some iFuckwits), is that 
this iFeature even contains the option of a remote data wipe.
My excuses to my fellow passengers who got in the way of a stream of 
expletives suddenly bursting forth from an otherwise unobtrusive 
business traveller, as said traveller notices he doesn't have a 
screwdriver to yank his data drive out of this ransomware machine, and 
cannot even be sure it's off when it says it's off because of course the 
battery cannot be removed without major surgery, either.


Was able to salvage the data at home using a real operating system on 
real hardware, and today I'm going to find an authorized mac reseller 
and give the guy at the guru bar a day he will remember and testify 
about at the next Apple employee incentive day.


Long story short: don't.



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Re: [LAD] OSC-generating plugin?

[Jörn Nettingsmeier]

On 06/19/2017 10:07 PM, Diemo Schwarz wrote:


ToscA!  http://forumnet.ircam.fr/shop/en/forumnet/84-tosca.html

"ToscA plugin can be inserted into a DAW (Digital Audio Workstation, 
such as ProTools, Apple Logic, Digital Performer, Ableton Live, etc.) in 
order to send/receive parameters’ automation.


The ToscA plugin is available in the following formats: AU, VST, VST3, 
AAX."


Thanks Diemo! I would have preferred an open-source tool, but since this 
whole scheme is evil anyways and you can only lose your soul once, it'll 
be fine :-D
I will admit to having bought a used Macbook recently (the customer is 
always right, and he uses QLAB), so I will use that for testing. I can't 
believe I will be remote-controlling Ardour from Reaper :-D An exercise 
in futility for the sake of maintaining the client's workflow of choice...



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Re: [LAD] OSC-generating plugin?

[Jörn Nettingsmeier]

Robin, Julius,

thanks for your replies. Julius, I wasn't aware of OSC magic in faust - 
I recently used it to decode a mic prototype array I was testing, and it 
worked like a charm (lazy me just used the online compiler), so I'll 
definitely check it out. Will need to understand the realtime issues 
that Robin pointed out, though.


On 06/19/2017 09:20 PM, Robin Gareus wrote:

On 06/19/2017 08:07 PM, Jörn Nettingsmeier wrote:

Hi *!

Does anybody know of a decent free plugin that generates arbitrary OSC
command streams from plugin automation data in the DAW? Preferrably
(gasp!) VST? Idea is to use SomeEvilDAW to send and control smart things
on a box running a friendly OS and a FriendlyDAW.


Sending OSC is not rt-safe and VST parameters are rather limited.
"arbitrary messages" are no fun and need all kinds of hacks (eg sending
them from the UI thread). There are a couple of single-parameter VSTs
though.


Clarification: "arbitrary" in the sense of "freely configurable", like 
so: data type, lower limit, upper limit, default value, OSC string, OSC 
target URL.
So the plugin will be pretty static once it's set up. I could even live 
with this being compile-time options.



Along those lines there's an ancient LADSPA plugin, too:
https://code.google.com/archive/p/noisesmith-linux-audio/downloads
needs some CFLAGS=-fPIC but otherwise still compiles, but it probably
won't run in SomeEvilDAW.

A more generic solution: https://github.com/x42/jackmidi2osc
Receive MIDI from someplace, generate fancy OSC based on rules.


I see. So this would also circumvent the realtime issue, since the 
plugin would be using the host's MIDI sending mechanism?



Spencer wrote a similar tool https://github.com/ssj71/OSC2MIDI/ which
despite its name can also turn MIDI into OSC.


Thanks for the pointers, I'll look at them.

Best,

Jörn


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[LAD] OSC-generating plugin?

[Jörn Nettingsmeier]

Hi *!

Does anybody know of a decent free plugin that generates arbitrary OSC 
command streams from plugin automation data in the DAW? Preferrably 
(gasp!) VST? Idea is to use SomeEvilDAW to send and control smart things 
on a box running a friendly OS and a FriendlyDAW.


Best,

Jörn


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[LAD] FLAC as open standard: help wanted.

[Jörn Nettingsmeier]

Hi *!


A friend and fellow free software enthusiast is trying to get FLAC 
standardized with the IETF to foster its use as a long-term archive 
format. If anyone here can see themselves drafting RFCs and helping with 
the tedium of such a process, please write to me off-list.


All best,


Jörn




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Re: [LAD] RME madi latency

[Jörn Nettingsmeier]

On 02/03/2016 02:54 PM, Fokke de Jong wrote:

Hi Adrian,

I could’ve guessed you were on this list as well :-)
I’m going to try this tomorrow when back in the studio, or maybe I can even do 
it remote if I have left everything patched up correctly :-)

I will come back with the results.
Thanks for your input!

fokke


watching this with interest, keep us posted about your results and 
longer-term experience with this card!



best,


jörn




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Re: [LAD] cpu spikes

[Jörn Nettingsmeier]

hi *!


sorry to hijack this thread, but: when enquiring about latency tuning, 
one frequently encounters hints like "disable cron", "disable indexing 
services", "disable this, disable that".


however, none of those alleged culprits run with real-time privileges or 
access driver or kernel code which does. so how can they be a problem 
(and disabling them part of the solution)? i'm asking because i've got 
my own anecdotal evidence that it *does* make a difference...


i understand how device drivers can be nasty (graphics cards locking up 
the pci bus, wifi chips hogging the kernel for milliseconds at a time or 
worse...) but it seems that a) either kernel preemption and real-time 
scheduling is terribly buggy or hand-wavey, or b) we're feeding each 
other snake-oil in recommending to disable userspace things that is 
running without rt privs.


i'd love to be educated on this.


best,


jörn




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Re: [LAD] ambix vs JUCE, segfault

[Jörn Nettingsmeier]

On 04/14/2015 01:42 AM, Fernando Lopez-Lezcano wrote:

On 04/13/2015 11:24 AM, Paul Davis wrote:

definitely caused by use of X / GUI toolkit calls from the wrong thread.
Not legal.


Ok, thanks, staring at code - no idea what to look for (Ambix uses the
JUCE LV2 wrapper) ...

On my laptop (Fedora 21 instead of Fedora 20, different video chipset)
the GUI starts fine but it can randomly crash with the same message.
Race condition that is sometimes triggered?

Anyone our there running the Ambix LV2 plugins successfully? (in, for
example, Ardour3?)



gave them a quick try in a3 4 weeks ago. i could get them to work by 
always using the generic UI. not ideal, but enough to get an idea what 
they can do for you.



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[LAD] Fwd: AES69-2015 standard for file exchange - Spatial acoustic data file format

[Jörn Nettingsmeier]

This was just announced on sursound. Might be worthwhile to adopt for 
anyone working with spatial audio datasets and files...



 Forwarded Message 
Subject: [Sursound] AES69-2015 standard for file exchange - Spatial 
acoustic data file format

Date: Sun, 15 Mar 2015 16:13:47 +0100
From: Markus Noisternig 
Reply-To: Surround Sound discussion group 
To: Surround Sound discussion group 

Dear Sursounders,

We are pleased to announce the recent publication of the AES69-2015 
standard for file exchange - Spatial acoustic data file format. See also 
the AES press release at http://www.aes.org/press/?ID=293 



The new AES69-2015 standard defines a file format to exchange 
space-related acoustic data in various forms. These include HRTF, as 
well as directional room impulse responses (DRIR). The format is 
designed to be scalable to match the available rendering process and to 
be sufficiently flexible to include source materials from different 
databases.


This project was developed in AES Standards Working Group SC-02-08 and 
standardizes the Spatially-oriented format for acoustics (SOFA), which 
aims at storing and transmitting any transfer-function data measured 
with microphone arrays and loudspeaker arrays. See 
http://www.sofaconventions.org/  for 
further information and ongoing format discussions.


Open source application-programming interfaces (API) for Matlab, Octave, 
and C++ are available online at 
http://sourceforge.net/projects/sofacoustics/ 



All the best,

Markus and Piotr


--
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Acoustics and Cognition Research Group
IRCAM, CNRS, Sorbonne Universities, UPMC
Paris, France

Piotr Majdak
Psychoacoustics and Experimental Audiology
Acoustics Research Institute
Austrian Academy of Sciences
Vienna, Austria
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[LAD] build problem with ambix plugins + LV2

[Jörn Nettingsmeier]

Hi Matthias,


I'm trying to build your ambix plugin suite on Linux using the LV2 
wrapper, which fails with the following problem during makefile generation:


CMake Error at CMakeLists_subprojects.txt.inc:104 (ADD_LIBRARY):
  Cannot find source file:


/local/build/ambix/JUCE/modules/juce_audio_plugin_client/LV2/juce_LV2_Wrapper.cpp

  Tried extensions .c .C .c++ .cc .cpp .cxx .m .M .mm .h .hh .h++ .hm .hpp
  .hxx .in .txx
Call Stack (most recent call first):
  ambix_binaural/ambix_binaural/CMakeLists.txt:20 (INCLUDE)

It seems that the JUCE tree in your repo does not have an LV2 directory 
at all, and neither has the official JUCE repo - has it been dropped?


Any hints much appreciated, best greetings from Essen to Vilnius,


Jörn





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Re: [LAD] JACK on Gigaport HD+ at 4800kHz (reduced channel count)

[Jörn Nettingsmeier]

hi clemens, thanks for your help.

On 02/25/2014 02:23 PM, Clemens Ladisch wrote:

Jörn Nettingsmeier wrote:

I'm trying to get my Gigaport HD+ to run at 48000kHz. The specs say it is 
capable of 8ch @ 44k1/16, 6ch at 44k1/24 and 48k/24.


Please show the output of "lsusb -v" for this device.


see below. i'd be interested to hear what you find.


When I try to start it at 48k, it comes up ok but ends up running at 44k1. It 
shows 8 channels, so that's expected.
Now, how do I tell it to use only 6, so that I can get to 48k?
I've tried setting -o6, which gives the usual "cannot set playback channel 
count".


In theory, any supported combination of parameters should work with Jack.

Does it work with "speaker-test -D hw:3 -c 6 -r 48000"?


~ # speaker-test -D hw:1 -c 6 -r 48000

speaker-test 1.0.27.2

Playback device is hw:1
Stream parameters are 48000Hz, S16_LE, 6 channels
Using 16 octaves of pink noise
Channels count (6) not available for playbacks: Invalid argument
Setting of hwparams failed: Invalid argument

so the issue appears to be at the ALSA level, not with JACK.
btw, the same happens with 48k/8ch (which is out of spec according to 
ESI), but also with 96k/2ch, which should be supported.


btw, the card claims 24bit support (whatever that is worth in this price 
range), but i find i cannot specify 24bit int, only S16_*, float, and 
S32_* - am i misunderstanding this option, or does it not represent 
24bit fixed-point?


what is the best way to pursue this further?


thanks,


jörn





Bus 002 Device 011: ID 2573:0009
Device Descriptor:
  bLength18
  bDescriptorType 1
  bcdUSB   1.00
  bDeviceClass0 (Defined at Interface level)
  bDeviceSubClass 0
  bDeviceProtocol 0
  bMaxPacketSize0 8
  idVendor   0x2573
  idProduct  0x0009
  bcdDevice1.00
  iManufacturer   1 ESI Audiotechnik GmbH
  iProduct2 GIGAPort HD+
  iSerial 0
  bNumConfigurations  1
  Configuration Descriptor:
bLength 9
bDescriptorType 2
wTotalLength  116
bNumInterfaces  2
bConfigurationValue 1
iConfiguration  0
bmAttributes 0x80
  (Bus Powered)
MaxPower  250mA
Interface Descriptor:
  bLength 9
  bDescriptorType 4
  bInterfaceNumber0
  bAlternateSetting   0
  bNumEndpoints   0
  bInterfaceClass 1 Audio
  bInterfaceSubClass  1 Control Device
  bInterfaceProtocol  0
  iInterface  0
  AudioControl Interface Descriptor:
bLength 9
bDescriptorType36
bDescriptorSubtype  1 (HEADER)
bcdADC   1.00
wTotalLength   46
bInCollection   1
baInterfaceNr( 0)   1
  AudioControl Interface Descriptor:
bLength12
bDescriptorType36
bDescriptorSubtype  2 (INPUT_TERMINAL)
bTerminalID 1
wTerminalType  0x0101 USB Streaming
bAssocTerminal  0
bNrChannels 8
wChannelConfig 0x00ff
  Left Front (L)
  Right Front (R)
  Center Front (C)
  Low Freqency Enhancement (LFE)
  Left Surround (LS)
  Right Surround (RS)
  Left of Center (LC)
  Right of Center (RC)
iChannelNames   0
iTerminal   0
  AudioControl Interface Descriptor:
bLength16
bDescriptorType36
bDescriptorSubtype  6 (FEATURE_UNIT)
bUnitID 2
bSourceID   1
bControlSize1
bmaControls( 0)  0x00
bmaControls( 1)  0x03
  Mute Control
  Volume Control
bmaControls( 2)  0x03
  Mute Control
  Volume Control
bmaControls( 3)  0x03
  Mute Control
  Volume Control
bmaControls( 4)  0x03
  Mute Control
  Volume Control
bmaControls( 5)  0x03
  Mute Control
  Volume Control
bmaControls( 6)  0x03
  Mute Control
  Volume Control
bmaControls( 7)  0x03
  Mute Control
  Volume Control
bmaControls( 8)  0x03
  Mute Control
  Volume Control
iFeature0
  AudioControl Interface Descriptor:
bLength 9
bDescriptorType36
bDescriptorSubtype  3 (OUTPUT_TERMINAL)
bTerminalID 3
wTerminalType  0x0301 Speaker
bAssocTerminal  0
bSourceID   2
iTerminal   0
Interface Descriptor:
   

Re: [LAD] JACK on Gigaport HD+ at 4800kHz (reduced channel count)

[Jörn Nettingsmeier]

On 02/25/2014 12:54 PM, Paul Davis wrote:




On Tue, Feb 25, 2014 at 6:48 AM, Fons Adriaensen mailto:f...@linuxaudio.org>> wrote:



Wrong, at least for snd_hdspm (and I did my homework before posting this
message and the previous one).


Things appear to have changed (for the better) then, since back in the
days of legend.


Thanks Fons and Paul for the enlightenment. Is the behaviour that Fons 
described supposed to be consistent between all ALSA drivers that 
support variable channel counts? Then what I'm seeing is clearly a bug, 
and I should take it to alsa-dev.
Otherwise, anything other than 44k1/16bits/8ch might just be unsupported 
by the driver.


If the maximum channel count is always fixed by the hardware, what is 
the point of the corresponding jackd option? As I said, I don't remember 
if ever doing something other than throwing the "cannot set channel 
count" error...



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[LAD] JACK on Gigaport HD+ at 4800kHz (reduced channel count)

[Jörn Nettingsmeier]

Hi *!

I'm trying to get my Gigaport HD+ to run at 48000kHz. The specs say it 
is capable of 8ch @ 44k1/16, 6ch at 44k1/24 and 48k/24.


When I try to start it at 48k, it comes up ok but ends up running at 
44k1. It shows 8 channels, so that's expected.

Now, how do I tell it to use only 6, so that I can get to 48k?
I've tried setting -o6, which gives the usual "cannot set playback 
channel count".
Next I looked at amixer -D hw:3 controls, and there is a playback switch 
map that is set to "on,on,on,on,on,on,on,on". I set it to 
"on,on,on,on,on,on,off,off" and retry. No luck. Still fails to set 
playback channel count, JACK still comes up with 8 outs and falls back 
to 44k1.


What am I missing? Is there some .asoundrc magic that might come to the 
rescue? Maybe define a device that has only 6 channels to begin with? 
But how do I do that?


Any hints much appreciated.


Best,


Jörn



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Re: [LAD] JACK latency API clarifications

[Jörn Nettingsmeier]

On 02/21/2014 07:52 PM, Lieven Moors wrote:

it was part of the API very early on, then we decided we didn't want to
impose the possibility of change on clients. as time goes on, it becomes
clear (to me at least) that we should have implemented it.


What would be use cases for changing the sample rate dynamically?



having wired up a complex signal graph, which for the most part depends 
on the studio, not on the project at hand, and then having to deal with 
different projects in different sample rates.


say your studio involves three monitoring setups, one main stereo, one 
nearfield, and one surround, you are using jack to do EQ on those 
things, in my case there's an ambisonic decoder in the loop as well. 
that means the jack graph is already quite elaborated. in that case, it 
would be nice to leave it running while switching from, say, a cd 
project at 44k1 to a tv thing at 48k.


as it is now, i have decided to do _everything_ at 48k (i have no second 
thoughts about a final resampling step), but if a client brings material 
at, say, 96k, i have to downsample first. sometimes i wish for an easy 
way to reclock a graph. obviously, nobody expects this to be gapless. 
fading everthing down and then taking a few seconds to reclock 
everything would be fine.


but then, many pieces of software in my chain would need changes. for 
instance, an important piece of dsp for me is jconvolver, as it sits in 
front of all my speakers.
of course, the impulse responses i use for EQ and room correction only 
make sense for a given sample rate - it would have to be changed to swap 
one set of IRs for another during a reclocking call, and of course that 
needs to be configured and the user actually needs to provide those 
different IRs.







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Re: [LAD] Zita Resampler unexpected output

[Jörn Nettingsmeier]

On 02/12/2014 10:25 PM, Fons Adriaensen wrote:

On Wed, Feb 12, 2014 at 09:37:01PM +0100, Robin Gareus wrote:


/me crawls back under his stone.


(C) stackingdwarves.net IIRC :-)


you mean i own the rights to being stupid in public?

wow, i'll be rich as standard oil :-D

cue evil laugh with cheesy reverb




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Re: [LAD] [ANN] A fifth of a Jubilee with a late Mike November release

[Jörn Nettingsmeier]

On 12/31/2013 08:49 PM, Rui Nuno Capela wrote:

Happy year-ending to y'all.

Not a fiscal report I'm afraid but the biggest load of Q-stuff released
ever ;)

<..>

Enjoy && have a very happy new year!


looks like you just _defined_ the beginning new year }:o)
/me goes svn up


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Re: [LAD] RME TCO option

[Jörn Nettingsmeier]

On 08/25/2013 10:33 PM, Adrian Knoth wrote:

On Fri, Aug 23, 2013 at 12:48:22PM +0100, Chris Goddard wrote:


Hi Adrian


Hi!

Let me CC Robin and LAD on this one.


Hope you don't mind me contacting you directly, but I have a
question regarding the RME TCO option and how it is supposed to
work. I saw your name on some ALSA commits relating to the TCO
recently and thought you would at least be the right person to start
by asking.

The TCO can read ltc, but once it has done that how does it
communicate that time information tothe user space ? If I was, say,
running Ardour and wanted it to chase this ltc what would Ardour
need to do to make use of it ?


can't say how you get at the TCO input, but thanks to robin's recent 
work, ardour can chase an incoming smpte ltc signal. it is just another 
audio signal, connected to ardour's ltc input port.


fwiw, you don't need a TCO to chase LTC at all. the TCO is interesting 
if you want to generate LTC in hardware, or if you want to _sync_ to 
LTC, i.e. recover a clock signal from it.



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Re: [LAD] Problem with recent hdspm, alsactl and systemd

[Jörn Nettingsmeier]

On 06/14/2013 12:02 AM, Fons Adriaensen wrote:

Hello all,

I wonder if any other users have experienced this problem and
how they handled it.

This has occured three times when doing an fresh Archlinux install
on a system using the RME MADI cards.

There seems to be something in the combination of recent versions
of the driver and alsactl that leads to alsactl freezing when the
configured (external) clock source for the card is not available.
The 'freeze' seems to be quite deep: it's impossible to kill the
process (even while that process is still a child of e.g. the
xterm from which it was launched, and not of PID 1). Any other
process trying to access the sound card (e.g. jackd) hangs in
the same way. This also means that when doing a poweroff or reboot
systemd will hang on the 'alsactl store' service, and the only
option is a power cycle.


i know i tend to over-estimate the power of strace, but you could try to 
run the offending process manually and watch it very closely:


root:~> strace /usr/sbin/alsactl store

maybe this gives you an idea where stuff goes wrong.




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Re: [LAD] Xiph.org - Video:Digital Show and Tell - No difference between analog and digitally processed sound.

[Jörn Nettingsmeier]

On 05/23/2013 02:18 PM, Fons Adriaensen wrote:

On Thu, May 23, 2013 at 09:38:40AM +0200, Jörn Nettingsmeier wrote:


moreover, i'd expect src circuits with only -12dB at fs/s to be
unusable in practise, because the aliasing artefacts would be
obvious. it means the top octave from 10-20hkz would be polluted
with junk at -24 to -12dB,


It doesn't have to mean that. If the filter gain is -12 dB at
0.5 Fs that doesn't imply it will be -24 dB at Fs - 10 kHz.


yes, sorry. i misread that one - for some reason, i had a 12db/oct 
filter in mind...




The potential intermodulation effects referred to in the paper
by Julian Dunn are real, but not realistic. If such signals
(high energy well above 20 kHz) are present you'll have some
serious problems even in a completely analog system.


probably :)

but i found the link between passband ripple and pre-echo really 
fascinating - this fact was completely new to me.



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Re: [LAD] Xiph.org - Video:Digital Show and Tell - No difference between analog and digitally processed sound.

[Jörn Nettingsmeier]

hi john,

On 05/22/2013 09:44 PM, John Rigg wrote:

On Wed, May 22, 2013 at 10:38:37AM -0400, Bill Gribble wrote:

There are real effects due to clock jitter on
both the A/D and D/A end that can cause small but measurable
distortions.


Not to mention audible if it's severe enough. Decimation filters
that only give 6 or 12dB attenuation at fs/2 (typical in many pro
audio ADC chips) can allow audible aliasing too. I wouldn't expect
an oscilloscope to have enough resolution to detect these effects,
but a good spectrum analyser and/or a good pair of ears often can.


this comment raised my eyebrows a little bit. can you explain what you 
mean by "decimation filter"? the way i understand it, decimation means 
chopping off bits, usually by shifting the data words, and possibly 
adding dither. how can this be a problem at fs/2? no new frequency 
components are introduced (apart from additional quantisation noise, 
which must necessarily be band-limited to fs/2), and the input of the 
decimation stage will already be band-limited as well.


otoh, if you mean sample rate down-conversion, i understand your 
comment, but then you picked an unfortunate term.
moreover, i'd expect src circuits with only -12dB at fs/s to be unusable 
in practise, because the aliasing artefacts would be obvious. it means 
the top octave from 10-20hkz would be polluted with junk at -24 to 
-12dB, unless of course there are some oversampling tricks going on and 
the effective fs is higher during down conversion. although i must 
confess i don't know anything about DAC and SRC design - if someone can 
explain this in more detail, i'm all ears.


best,


jörn



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[LAD] photos from lac 2013

[Jörn Nettingsmeier]

hi *!


here's a bunch of photos from lac 2013, enjoy:

http://www.stackingdwarves.net/public_stuff/linux_audio/lac2013/photos/


best,


jörn


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Re: [LAD] jack ringbuffer usage

[Jörn Nettingsmeier]

On 04/08/2013 05:27 PM, Raphaël BOLLEN wrote:

On 04/08/2013 03:29 PM, Paul Davis wrote:


all the xmms-based players do this (xmms, beep, beep2, audacious)

gstreamer based players also do this across files, but not above
play/stop.


Thanks Paul, these are really fine programs but mostly for playing
stereo music. I need to play mono, stereo and 4 or more channels files
on a headless system so it's a bit of a corner case application...


please count me in as an alpha tester for the "or more" scenario :)


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Re: [LAD] jack ringbuffer usage

[Jörn Nettingsmeier]

On 04/08/2013 01:51 PM, Harry van Haaren wrote:

On Mon, Apr 8, 2013 at 12:45 PM, Raphaël BOLLEN
mailto:raphael.bol...@mobistar.be>> wrote:
 > error: invalid conversion from 'void*' to 'char*' [-fpermissive]


jack_ringbuffer_read() expects the buffer pointer to be of type  char*
not void*.
The "char*" should just be interpreted as "pointer", as the data is
data, which is not necessarily a char.

Change your case from (void*) to (char*) and you won't need -fpermissive :)
HTH, -H


hmmm. isn't that a really outdated way of saying "generic pointer", like 
C89 or so? maybe the ringbuffer implementation should be changed to 
(void *)?

or am i misunderstanding something?




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Re: [LAD] build troubles with gtklick

[Jörn Nettingsmeier]

On 04/08/2013 01:25 PM, Dominique Michel wrote:

Le Mon, 08 Apr 2013 01:16:37 +0200,
Jörn Nettingsmeier  a écrit :


hi dominic!


first of all, thanks for sharing your tools - klick has saved the day
by adding a much-needed jack-transport aware metronome to a
sooperlooper setup.

now i'm a lazy bastard and want to use gtklick, but even though it
compiles and installs fine, it barfs when i start is, like so:

nettings@kleineronkel:/usr/lib/python2.7/site-packages/gtklick>
gtklick Traceback (most recent call last):
File "/usr/bin/gtklick", line 14, in 
  from gtklick.gtklick import GTKlick
File "/usr/lib/python2.7/site-packages/gtklick/gtklick.py", line
30, in 
  import klick_backend
File "/usr/lib/python2.7/site-packages/gtklick/klick_backend.py",
line 12, in 
  import liblo
ImportError: /usr/lib64/python2.7/site-packages/liblo.so: undefined
symbol: lo_address_new_with_proto


i have tried both liblo-0.26 and current liblo svn, no luck.

now the python paths in this openSUSE tumbleweed install are a
horrible mess, with three different python versions and libs
in /usr/lib, /usr/lib64, and /usr/local/lib64. but they all seem to
be found, and i made sure that the liblo.so mentioned in the error
message is actually the one from your pyliblo package (by copying it
manually). i removed the build directory of pyliblo for each try, and
also recreated liblo.c via cython.

how do i proceed to fix this?


Here on gentoo, gtklick depend on klick with osc support. And liblo
(liblo-0.26 here) is for osc support.
Are you sure klick have osc support enabled?


yes. the problem might have been two different versions of liblo (one 
from the distro (0.25 iirc), and latest svn in /usr/local. not normally 
a problem, but apparently one was picked up during building and the 
other at runtime, leading to that symbol error above. getting rid of the 
duplicate liblo.so fixes the problem for me.


thanks to all who replied with insightful python tips,


jörn



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Re: [LAD] build troubles with gtklick

[Jörn Nettingsmeier]

dominic, ralf, thanks for this hint, however...

On 04/08/2013 02:39 AM, Dominique Michel wrote:

Le Mon, 08 Apr 2013 01:16:37 +0200,
Jörn Nettingsmeier  a écrit :


File "/usr/lib/python2.7/site-packages/gtklick/klick_backend.py",
line 12, in 
  import liblo
ImportError: /usr/lib64/python2.7/site-packages/liblo.so: undefined
symbol: lo_address_new_with_proto

i have tried both liblo-0.26 and current liblo svn, no luck.

now the python paths in this openSUSE tumbleweed install are a
horrible mess, with three different python versions and libs
in /usr/lib, /usr/lib64, and /usr/local/lib64. but they all seem to
be found, and i made sure that the liblo.so mentioned in the error
message is actually the one from your pyliblo package (by copying it
manually). i removed the build directory of pyliblo for each try, and
also recreated liblo.c via cython.

how do i proceed to fix this?


python is a mess in itself because we have python 2 and 3, and the
shebang of many python programs is set as #!/usr/bin/python, which
doesn't tell the system which version to use.
You have to adjust the shebangs so that the scripts will use the
correct version. Something like

#!/usr/bin/python2.7


i've tried both python2 and python2.7 in the hashbang. but the error 
remains the same. to me it looks like a pyliblo/liblo version 
mismismatch, but i'm not sure....


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[LAD] build troubles with gtklick

[Jörn Nettingsmeier]

hi dominic!


first of all, thanks for sharing your tools - klick has saved the day by 
adding a much-needed jack-transport aware metronome to a sooperlooper setup.


now i'm a lazy bastard and want to use gtklick, but even though it 
compiles and installs fine, it barfs when i start is, like so:


nettings@kleineronkel:/usr/lib/python2.7/site-packages/gtklick> gtklick
Traceback (most recent call last):
  File "/usr/bin/gtklick", line 14, in 
from gtklick.gtklick import GTKlick
  File "/usr/lib/python2.7/site-packages/gtklick/gtklick.py", line 30, 
in 

import klick_backend
  File "/usr/lib/python2.7/site-packages/gtklick/klick_backend.py", 
line 12, in 

import liblo
ImportError: /usr/lib64/python2.7/site-packages/liblo.so: undefined 
symbol: lo_address_new_with_proto



i have tried both liblo-0.26 and current liblo svn, no luck.

now the python paths in this openSUSE tumbleweed install are a horrible 
mess, with three different python versions and libs in /usr/lib, 
/usr/lib64, and /usr/local/lib64. but they all seem to be found, and i 
made sure that the liblo.so mentioned in the error message is actually 
the one from your pyliblo package (by copying it manually). i removed 
the build directory of pyliblo for each try, and also recreated liblo.c 
via cython.


how do i proceed to fix this?

best,


jörn



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Re: [LAD] Mixing audio: Implementing pan and balance

[Jörn Nettingsmeier]

On 03/23/2013 06:05 PM, Raphaël BOLLEN wrote:

On 03/16/2013 07:23 PM, Fons Adriaensen wrote:

On Sat, Mar 16, 2013 at 09:25:46AM -0400, Paul Davis wrote:

On Sat, Mar 16, 2013 at 9:25 AM, John Rigg  wrote:



A lot of mixing consoles don't provide a mono switch, but it's usually
possible to work around it with sub groups. I still have to work out a
convenient method for mono checking in Ardour 3.



which means i need to write up a page or two for
http://manual.ardour.org/


And which reminds me of an app I've been using for years but
never released.

Zita-mu1 is a simple Jack client used to organise stereo monitoring
during recording and mixing. More here:
<http://kokkinizita.linuxaudio.org/linuxaudio/zita-mu1-doc/quickguide.html>

and download from
<http://kokkinizita.linuxaudio.org/linuxaudio/downloads/zita-mu1-0.2.0.tar.bz2>


Ciao,



Awesome, however cpu usage is ~12% on my system when playing silence and
drops to ~2% with some audio going through.


smells like a denormals issue.


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Tonmeister VDT

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Re: [LAD] Mixing audio: Implementing pan and balance

[Jörn Nettingsmeier]

On 03/14/2013 12:37 PM, Paul Davis wrote:



On Thu, Mar 14, 2013 at 6:42 AM, Jörn Nettingsmeier
mailto:netti...@stackingdwarves.net>> wrote:

On 03/12/2013 08:08 PM, Tim E. Real wrote:

But having said that, yes I'm wondering about a true 'stereo
pan' feature.
How would such a feature work?


there is no one true stereo pan.

a pan law for intensity stereo (i.e. a panned image or an XY
coincident microphone pair) would increase one channel and decrease
another such that the total energy remains constant. a cosine/sine
law is usually used, because

cos^2 + sin^2 = 1

ardour3 attempts to do this, by allowing you to reduce the width (by
introducing crosstalk), and then letting you move the compressed
image left or right. sort of works, but only for pan-potted stuff.

a pan law for run-time stereo (i.e. spaced omnis) would have to use
delays, leaving the original level intact.

the ardour3 panner gets this type of signal horribly wrong, because
you _never_ want to introduce crosstalk in spaced omnis - instant
comb-filtering hell.

for stereo techniques that incorporate both run-time and intensity,
such as ORTF, NOS, EBS, you-name-it, you need different amounts of
gain change _and_ delay.

that's why nobody wants to use a ready-made stereo balance control -
it is almost guaranteed to do the wrong thing for the source
material at hand.


git add libs/panners/spaced_omni_panner
git commit
git push



point taken :)

the problem is, you usually have a mixture of the above. hence, no way 
to get the stereo panner right. unless the user knows exactly what s/he 
is doing, and then s/he doesn't really need a stereo panner :-D


btw, sorry for the pot shot at ardour specifically - in fact, most if 
not all DAWs get it wrong. iirc, some DAWs with a focus on classical 
(sequoia, pyramix) have some code to mogrify complex stereo, but i 
haven't used it.


my approach to complex stereo re-panning is: split the signal into two 
mono busses, add a delay, add eqs if you have to, move faders.
and while we're at it, the same approach works for processing in the MS 
domain. it's painful enough to only use it when you really need it, and 
hard enough to discourage casual users :)


for those who read german, here is a discussion we had about this in the 
VDT forum: http://www.tonmeister.de/forum/viewtopic.php?f=6&t=238&p=817



best,


jörn



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Re: [LAD] Mixing audio: Implementing pan and balance

[Jörn Nettingsmeier]

On 03/12/2013 08:08 PM, Tim E. Real wrote:


But having said that, yes I'm wondering about a true 'stereo pan' feature.
How would such a feature work?


there is no one true stereo pan.

a pan law for intensity stereo (i.e. a panned image or an XY coincident 
microphone pair) would increase one channel and decrease another such 
that the total energy remains constant. a cosine/sine law is usually 
used, because


cos^2 + sin^2 = 1

ardour3 attempts to do this, by allowing you to reduce the width (by 
introducing crosstalk), and then letting you move the compressed image 
left or right. sort of works, but only for pan-potted stuff.


a pan law for run-time stereo (i.e. spaced omnis) would have to use 
delays, leaving the original level intact.


the ardour3 panner gets this type of signal horribly wrong, because you 
_never_ want to introduce crosstalk in spaced omnis - instant 
comb-filtering hell.


for stereo techniques that incorporate both run-time and intensity, such 
as ORTF, NOS, EBS, you-name-it, you need different amounts of gain 
change _and_ delay.


that's why nobody wants to use a ready-made stereo balance control - it 
is almost guaranteed to do the wrong thing for the source material at hand.


best,


jörn

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Re: [LAD] So what do you think sucks about Linux audio ?

[Jörn Nettingsmeier]

On 02/05/2013 03:58 PM, Dave Phillips wrote:

Greetings,



I'm not so interested in comments on the commentary, I have my own, but
say what you will about the list. I figure that most denizens of these
lists already have ready replies and responses to these and other
criticisms, many of which have been voiced here previously. What I'm
more interested in is what *you* think is missing most or just plain
wrong about the situation. Please, try to speak your piece without
flames or dissing other developers and/or their work.


[drum roll, the following with at least 15% THD and the distinct sound 
of a 50s ribbon mike after long abuse:]


ask not what free software can do for you, ask what you can do for free 
software!


[enter brass band with some heroic yet totally cheesy hymn arrangement, 
think charles ives stealing frank zappa's reggae horn arrangement of the 
stairway to heaven solo.]




free software, my friends, is a natural resource. complaining about the 
lack of this or that is just about as clever or useful as complaining 
about the utter lack of oil or rare earth metals on your home turf, 
which unjustly prevents you from becoming the next rockefeller.
making linux audio more approachable to people who have not grasped this 
basic fact has no benefits at all, neither to developers nor to users.


personally, i find my days of linux audio evangelism are over. it suits 
my needs better than ever before, and i make very sure that people i 
talk to are made aware of the treasure trove of linux audio tools.
and of course i assume the lotus position and put on my most radiating 
smile when people who have just figured that i'm a sort of computer 
person then start complaining about their problems with operating system 
$FOO and how proprietary tool $BAR is just a millstone around their 
necks. but that's it.
if they need guitar rig or protools or garageband, we can't give it to 
them, so obviously they are better off on other platforms. that is good. 
it's even better than turning them into frustrated converts who then 
keep complaining how they can't run TDM or RTAS plugins or their VSTs 
keep crashing or whatever.


if somebody decides to take the plunge (which also implies some other 
basic skills, such as being able to use email in a constructive manner, 
learning what IRC is, aiming at learning to compile one's own software, 
and so on), i will try to share tricks and help out as best as i can.


but why press-gang perfectly happy users of proprietary software into 
linux, or put up with jerks who think the world has to support their 
personal way of composing? that's just the lamest thing i can imagine.


don't get me wrong, i think it's perfectly ok for non-programmers to try 
and nudge developers gently towards what you think are good ideas. i do 
it myself all the time, but i try to do it 
_from_the_inside_of_a_project. that means i try to make myself a little 
useful, subscribe to the mailing list, learn the software, build from 
the latest dev tree, lose some productive time dealing with crashes and 
try to provide useful feedback. only then do i sound off about what new 
stuff i'd like to see.


anything else is just bikeshedding.


my 200 €.


jörn


--
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Re: [LAD] [ANN] CAPS 0.9.1

[Jörn Nettingsmeier]

On 12/31/2012 06:01 PM, Tim Goetze wrote:

CAPS 0.9.1
==

The C* Audio Plugin Suite is a selection of popular effects, unique
filters and generators.  For the digital guitarist, CAPS offers a
range of processors recreating the formation of tone in traditional
instrument amplification.  Beyond sound quality, central design
considerations are latency-free realtime operation, modest resource
demands and meaningful control interfaces.

http://quitte.de/dsp/caps.html


that's excellent documentation! i'm looking forward to listen to the 
saturation functions in particular, looks like there is much to learn. 
thanks for creating and sharing this!





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Re: [LAD] Announcing PHASEX-0.14.96

[Jörn Nettingsmeier]

On 12/30/2012 05:42 AM, William Weston wrote:

Happy New Year!

Yes, your eyes are working correctly.  This is v0.14.96.  Some things
are worth the wait.  I know it's been a while, but I haven't forgotton
about PHASEX...


whoohooo!


just a quick feedback: distros that are moving to a unified /usr/bin 
will need this patch:


diff --git a/configure.ac b/configure.ac
index 5fb0368..3b2496c 100644
--- a/configure.ac
+++ b/configure.ac
@@ -334,7 +334,8 @@ AC_DEFINE_UNQUOTED(PHASEX_GCC_MINOR, [$gccminor], 
[Built with gcc minor version]

 CPU_POWER_LEVEL=2
 ARCH_OPT_CFLAGS=""
 ARCH_MATH_CFLAGS=""
-ARCH_BITS=`( file /bin/true | grep 'ELF 64-bit' > /dev/null && echo 64 
) || echo 32`

+ARCH_TRUE=`which true`
+ARCH_BITS=`( file $ARCH_TRUE | grep 'ELF 64-bit' > /dev/null && echo 64 
) || echo 32`


because `file /bin/true` will return "/bin/true: symbolic link to 
`/usr/bin/true'", which in turn will force the bitness to 32, which then 
fails unless a full 32-bit environment is installed.


now i'm hunting a couple errors wrt jack headers - my guess is that 
PHASEX is being tested with JACK2 exclusively - it seems to rely on a 
couple of types and methods which don't seem to be present in my jack1 
environment (more or less fresh from svn).


best,


jörn




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[LAD] warning: seedy journal possibly targetting LAC authors

[Jörn Nettingsmeier]

hi everybody!


i just received a personalized spam mail from a very seedy publishing 
company which apparently has harvested the linux audio conference 
backlogs. so here's a friendly warning since this might happen to others 
who have contributed to LACs in the past.


this company doesn't check out at all as a relevant publisher, and a 
number of people have reported that their business model is to bait and 
hook scholars desperate for publications and then milking them for 
publication fees. they will also ask for a transfer of copyright.


needless to say, david publishing's journals are practically unheard of 
in the world of serious research, and being in one is very likely to 
hinder a career rather than foster it.


meanwhile, a happy holiday season and may santa bring good publishing 
opportunities to all scholars who have been well-behaved during the year!



jörn



 Original Message 
Subject:Call for Papers or Books from Journal of Literature and Art
Studies (ISSN 2159-5836)
Date:   Fri, 21 Dec 2012 13:58:59 +0800
From:   literature.art 
To: nettings 



  From Knowledge to Wisdom

*Journal of Literature and Art Studies, USA*

International Standard Serial Number:

ISSN2159-5836 (Print),ISSN 2159-5844 (Online)**

*Call for Papers and Books*

**

Dear Jörn NETTINGSMEIER,

This is a journal entitled */Journal of Literature and Art Studies
/(ISSN 2159-5836)*published across the United States by David Publishing
Company, EL MONTE, CA, USA.We are glad to know you have submitted a
paper
named:*_General­purpose Ambisonic playback systems for electroacoustic 
concerts -

a practical approach _*at Linux Audio Conference 2010.Weare very
interested in your research. If the paper mentioned has not been
published in other journals and you have the idea of making our journal
as a vehicle for your research interests,please send us the English
electronic version of your paper in MS word format. And all your
original and unpublished papers are welcome.

Hope to keep in touch with you by email. If you have other original and
unpublished papers or books at hand which have not been published yet,
please feel free to send them to us too. As an American academic
publishing group, we wish to become your friends if necessary. At
present, we also want to invite some people to be our reviewers or
become our editorial board members. If you are interested in our
journal, Please send your CV to us too. Expect to get your reply soon.

*Description*

**

*/Journal of Literature and Art Studies /*is an international academic
journal (print and online), published monthly by David Publishing
Company, USA, which was founded in 2001, and is striving to provide the
best platform for artists and scholars worldwide to exchange their
latest findings and results. Our journal is an English version and your
contribution to our journal would be very much welcome!

*Current columns involve:*

**

Literature studies, art theory, culture and history of arts,
appreciation of arts, etc..

*Some renowned databases our journal can be retrieved:*



*/Journal of Literature and Art Studies/*is to be indexed in following
databases:

★ Database of EBSCO, Massachusetts, USA (Humanities Abstracts (H.W. Wilson))
★ LLBA Database of CSA
★ Universe Digital Library S/B, Proquest
★ Proquest, USA
★ Chinese Database of CEPS, American Federal Computer Library center 
(OCLC), USA
★ Chinese Scientific Journals Database, VIP Corporation, Chongqing, P.R. 
China

★ Ulrich’s Periodicals Direcory
★ Summon Serials Solutions

*Information for author(s)*

**

1. The manuscript should be original, and has not been published
previously. Please do not submit material that is currently being
considered by another journal.

2. Manuscripts may be 3000-8000 words or longer if approved by the
editor, including an abstract, texts, tables, footnotes, appendixes, and
references. The title should not exceed 15 words, and abstract should
not exceed 400 words. 3-8 keywords or key phrases are required.

3. The manuscript should be in MS Word format, submitted as an email
attachment to our email address.

4. Author(s) of the articles being accepted are required to sign up the
Transfer of Copyright Agreement form.

5. The author(s) will receive 2 copies of the issue of our journal
containing his/her (their) article(s).

6. It is not our policy to pay authors.

*Submission**of manuscript*

**

All manuscripts submitted will be considered for publication. Please
visit our website at _www.davidpublishing.com_ for our automatic paper
submission systems or send an email attachment to:
art.literat...@yahoo.com<mailto:art.literature...@yahoo.com>,
_literature.a...@davidpublishing.org<mailto:philost...@gmail.com>.

Best Regards!

Wayne

Editorial Office


Journal of Literature and Art Studies (ISSN 2159-5836)
David Publishing Company
9460 TELSTAR AVE SUITE 5,EL MONTE,CA 91713,USA
Website: http://www.davidpublis

Re: [LAD] [OT] Real-time threads on openSuSE 11.4

[Jörn Nettingsmeier]

On 10/19/2012 03:02 PM, Fons Adriaensen wrote:

Hello all,

My ex-collegues at Alcatel are screaming for help. They want to run
an app (as root, debatable but that's another story) using SCHED_FIFO
threads on an openSuSE 11.4 system.

Using the 'default' kernel (which has CONFIG_PREEMPT not set), this
works. Using the 'desktop' kernel (CONFIG_PREEMPT=y) they get an
EPERM when trying to start a RT thread, even as root.

As I haven't used SuSE for ages, has anyone an idea of what is
happening here ?

TIA,


god's way of telling them oS 11.4 has just been deprecated and will not 
receive further mainline support? :)


i'm not sure 11.4 had cgroups enabled already, but if so, then robin is 
probably right on the mark.



--
Jörn Nettingsmeier
Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487

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Tonmeister VDT

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Re: [LAD] Is Linux audio moving forward - some very personal notes

[Jörn Nettingsmeier]

On 10/11/2012 01:14 PM, Adrian Knoth wrote:

On 10/11/2012 01:09 AM, Fons Adriaensen wrote:


The HW situation has been mentioned. Honestly, I wouldn't know
where to go if RME went away. Almost everything I've been doing
the last years has not only used their HW, but depended on it -
no alternatives.


JFTR, I've seen ardour running on an Audinate Dante PCIe card last
summer at musikmesse. They have made a proprietary Linux driver to drive
their WFS systems. Unfortunately, I forgot the company's name, all I know
is it wasn't IOSONO.


the wfs market isn't too big - could have been sonic emotion, a swiss 
company?



Last not least, the RAVENNA camp has stuff ready that might benefit from
Intel's recent kernel changes. Florian Faber has (unreleased) RAVENNA-jackd
integration, and you can buy RAVENNA-enabled audio gear from
directout.eu.


yup. i'm really excited about ravenna, although there seemed to have 
been a hiatus for quite some time. but afaict, it's quite elegant and 
open-source friendly, certainly friendlier than most of the AVB-related 
stuff out there...



--
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Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487

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Tonmeister VDT

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Re: [LAD] [LAU] Linux Audio 2012: Is Linux Audio moving forward?

[Jörn Nettingsmeier]

On 10/10/2012 11:00 PM, Patrick Shirkey wrote:


On Thu, October 11, 2012 7:25 am, Louigi Verona wrote:

@Patrick:

"The problem with that approach is that it tends to feed the negative
attitude towards Linux and that is exactly what the "competition" want."

There is no competition, Patrick. Windows Audio does not compete with
Linux Audio. Only if in our minds. And thus they do not want anything.



There are plenty of competitors to Linux Audio as a platform. AVID is the
most obvious competitor.


that's a bit like saying NASA is competing with the RC model helicopter 
community. i'm pretty sure the whole professional *non-embedded* linux 
audio market is a fraction of the size of AVID's _marketing_ budget.


now under the hood, things look quite different, but that doesn't have 
much impact on the public opinion towards or perception of linux.



There is no Windows Audio community, there is a Linux Audio community.
We try to compete with them. They do not compete with us.



Look at things from a professional business point of view and try again
please. I'm not talking about Linux Multimedia for amateur users or even
necessarily for artists/producers. I'm talking about businesses that use
Linux as their revenue generating platform.


i'm one such business, and despite my healthy illusions of grandeur i 
don't consider myself part of a relevant market for any major equipment 
or software manufacturer.


besides the obvious technical benefits of using linux (for my particular 
kind of workflow), the main advantage to me is to be able to _ignore_ 
the rat race of the mainstream pro audio software market.



Don't you mean that because "insert favorite application/plugin" is not
ported they will have to learn how to do something differently and that is
too much to ask?


that's not how marketing works, and that's not how the market works. the 
goal is to get kids to buy dsp cards with emulations of old UREIs that 
are great for snares and female vocals, and another emulation of an old 
fairchild which is great for male voices and kick drums, and the way to 
do it is to get fat old mixing gurus to advertise that kind of gear on 
youtube.


the linux community doesn't have those dsp cards to sell, our plugins 
don't have the kind of bling, and people who give their stuff away are 
less inclined to bullshit kids out of their money. we have a few 
limiters with a bunch of parameters that give useful results on all 
kinds of program material, all they lack is the instant rocknroll 
credibility thing of a fat bearded guy with a metallica t-shirt at a 
96-channel ssl who compares them to his obsolete analog treasures and 
praises them to high heaven.


hence, in my view, the absence of a "market" like this is a good thing.

the only time it hurts is when i cannot get hardware support for gear 
that i need. but these days, i can get linux drivers for everything from 
2 to 128 channels of i/o (more if i'm prepared to gang cards), so what's 
the problem?


intel and amd thankfully make dsp cards that will also deal with my 
email and run my browser (word processing on a sharc, anyone?), and they 
are well-supported by linux :)



I like it and I am
doing it,
but I would not advertise Linux Audio as comparable to Windows Audio since
it is
simply not true.


And it's a good thing too.


here i whole-heartedly agree!



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Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487

Meister für Veranstaltungstechnik (Bühne/Studio)
Tonmeister VDT

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[LAD] LAA list policy

[Jörn Nettingsmeier]

hi *!

this unfortunate announcement from nedko seems to have spawned a 
discussion on LAA. LAA list policy used to be no follow-ups except for 
factual corrections. the idea was to keep the traffic low for people who 
want to stay informed but not have to wade through too much mail. (think 
lwn.net among others.)


can i suggest that all further contributions to this LAA thread be NAKed 
by the moderator, please? the discussion is certainly important, but it 
should continue on LAD or LAU.


that said, while i can understand nedko's POV, the initial message never 
really belonged on LAA either. obviously, such announcements _will_ 
cause heated discussion, so they should be made in a forum which allows 
them.


best,


jörn


 Original Message 
Subject: [LAA] my lv2-related projects
Date: Tue, 21 Aug 2012 06:34:49 +0300
From: Nedko Arnaudov 
To: linux-audio-annou...@lists.linuxaudio.org

I'm abandoning all lv2 related projects that I currently maintain.
Here is a list:
 * zynjacku/lv2rack
 * lv2fil
 * ssg
 * lv2vocoder
 * lv2dynparam
 * external ui extension
 * lv2zynadd [partially, see below]
 * maybe something else I dont recall right now

The zyn-rewrite project that produced lv2zynadd stays but will be
cleared from all lv2 code. If anyone wants to take over the
maintainership of any project of mine, contact me. I'll wait a month
before wiping out all online lv2-related resources I control.

I don't want to participate in the lv2 madness anymore. I admit I cannot
communicate rationally with David Robillard. If contributing is not
pleasure, then a one doesn't belong to the community. I wish everyone
inloved more luck than I had.

--
Nedko Arnaudov 



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[LAD] CueFrog.pd: theatre cue player framework initial release

[Jörn Nettingsmeier]

hi everyone!


thanks to the excellent pd documentation out there and lots of hand 
holding by friendly pd gurus on this list and elsewhere, here's my 
humble take at creating a theatre cue player with pd that does what i 
need... all the heavy lifting is done by august black's excellent 
readanysf~, thanks for making this tool available!


CueFrog is designed to be multi-instance capable, so you can create as 
many decks as your machine can handle, and makes use of lots of 
send/receive ports to simulate some kind of object-oriented 
encapsulation stuff, based on my (limited) understanding of a 
model/view/controller paradigm.


grab it: 
http://stackingdwarves.net/public_stuff/software/CueFrog/CueFrog-0.0.2.tar.gz


it's documented, so you should get it going in no time. i'm sure there 
are many quirks there, and i found out it's very easy to create race 
conditions in pd, so no warranties :)

comments and suggestions for improvements are most welcome.

i have a vbap-based panning automation in the works (which has already 
been used live at a theatre festival), but the code is in 
oh-my-good-tomorrow-is-dress-rehearsal shape, so forgive me for 
withholding it another month or so.


and before you ask: frogs are cute. and when the director makes me jump, 
i need tools that jump along :-D



best,


jörn


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[LAD] Linux Audio Conference 2012 at CCRMA - proceedings and videos now available

[Jörn Nettingsmeier]

On 04/11/2012 07:55 PM, Jörn Nettingsmeier wrote:

Hi *!


On behalf of the conference organizers, we would like to invite you to
join the Linux Audio Conference 2012, kindly hosted by the Center for
Computer Research in Music and Acoustics (CCRMA) at Stanford University.

The conference will start tomorrow, Thursday April 12, at 10:00 PST
(that's UTC - 0700). Please refer to the schedule at

http://lac.linuxaudio.org/2012/program

for detailed information.


thanks to robin gareus, who not only designed and implemented the most 
kick-ass video workflow we ever had and integrated it with the sweetest 
conference database system we ever had, but also ran his machines day 
and nite to re-encode our video dumps, you can now surf to


  http://lac.linuxaudio.org/2012/program

and enjoy the results of a very intense four days at ccrma: slides, 
papers, and videos of all talks and workshops.


best,


jörn



--
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Tonmeister VDT

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[LAD] Linux Audio Conference 2012 at CCRMA - live stream coverage starting tomorrow

[Jörn Nettingsmeier]

Hi *!


On behalf of the conference organizers, we would like to invite you to 
join the Linux Audio Conference 2012, kindly hosted by the Center for 
Computer Research in Music and Acoustics (CCRMA) at Stanford University.


The conference will start tomorrow, Thursday April 12, at 10:00 PST 
(that's UTC - 0700). Please refer to the schedule at


   http://lac.linuxaudio.org/2012/program

for detailed information.

We will be streaming all paper presentations live in Ogg Theora/Ogg 
Vorbis format. Users of the Firefox browser should be able to watch this 
natively without any plugins. For users of other browsers, we recommend 
VLC, a cross-platform media player which you can download from 
http://videolan.org.


You are invited to join us on IRC while you're watching the streams, the 
conference channel is #lac2012 on freenode.net, to be accessed with the 
chat client of your choice, or via 
http://webchat.freenode.net/?channels=lac2012
Remote participants can post their questions or remarks on this channel, 
and a local chat operator here in Stanford will then relay them to the 
presenters and the local audience. You can also use this channel to get 
help in case of viewing problems.


All presentations will be recorded and uploaded for off-line watching 
within a day or so.


Needless to say, access to all streams is free of charge. This is all 
about open source after all :)


The primary stream relay is available at

   http://ccrma.stanford.edu:8080 (located on the west coast of the US).

A secondary relay which is preferrable for European users is at

   http://streamer.stackingdwarves.net (located in Germany).


Best regards,


the LAC stream team.

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Re: [LAD] [ot] rme fireface: weird balanced output measurements

[Jörn Nettingsmeier]

fons, gene,


thanks for your insights, particularly about the problems of AC rms 
meters at audio frequencies - i was totally ignorant of this, and i'd 
have shot myself in the foot with this eventually.


fons, after i read your remark i asked around, and yes, the ff800 does 
indeed use servo-driven outputs, so the voltage difference is 
intentional and no cause for concern. i'll definitely investigate the 
level differences between channels, but as you said, i'd better do that 
with a sound card rather than a meter.


gene, would love to follow your scope advice (particularly for the 
beneficial effects on gonads), and i'm looking out for one, but i'm not 
really prepared to pay much. in any case, it's way sexier than one of 
those usb sampling gadgets sold for a scope these days... maybe i'll 
find a nice, simple one in a dumpster somewhere.


nice to get you two guys into another discussion about signal 
transmission - very instructive!


best,


jörn


btw: yes, the folkwang address seems to have expired at last. the la* 
lists were the last things i used it for, and i have now resubscribed 
from my own mail domain. so please ditch that old one from your address 
books if you want to get in touch off-list.



--
Jörn Nettingsmeier
Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487

Meister für Veranstaltungstechnik (Bühne/Studio)
Tonmeister (VDT)

http://stackingdwarves.net
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[LAD] [ot] rme fireface: weird balanced output measurements

[Jörn Nettingsmeier]

hi everyone!


yesterday, i visited my friends' new studio, to help them shake out some 
bugs in the patch and fix a monitor problem. there i came across a 
really weird issue with the line outputs on two rme fireface 800s:


we put a test tone out in logic (yeah, they run a mac shop), and i went 
to measure the outputs. in addition to really high fluctuations from one 
output to the next (with identical digital input signals), i measured 
huge differences in voltage on the hot and cold side, such as


hot to ground: 2.00 V
cold to ground: 1.82 V

despite the fluctuations across several channels, this trend was pretty 
constant. so i figured, maybe this box has a problem with its negative 
voltage rail. they had another ff 800 in there, which we measured for 
comparison. same issue.


i figured, maybe apple wrote an oscillator with a dc offset, so we 
applied a known-good test tone .wav file. same result.


is my thinking flawed, or are rme really delivering such crappy output 
stages on such a pricey box?


my multimeter is not a calibrated one, but it's in the 100+€ range, and 
i used AC true rms measurement mode.


i understand my meter has a very high input impedance, and that's what 
line level connections should have, right? operating more or less 
open-loop, without significant currents flowing. or should i use a shunt 
resistor and measure across that? if so, what value is recommended?



thanks for any insights, best,


jörn

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Re: [LAD] LV2 and parameter interpolation

[Jörn Nettingsmeier]

On 02/06/2012 02:23 PM, Gabriel M. Beddingfield wrote:

On 02/06/2012 03:33 AM, Jörn Nettingsmeier wrote:

can i rely on control port data being available during activate() so
that i can initialize my current values to the control port values?


No. Ports are only valid during the run() method.


thanks for explaining.


now i could set the current values to NULL during instantiate and
deactivate, but then i'd have an extra conditional in run(), which i'd
like to avoid.


If your engine has different "states," just make this first-run
condition one of your states and use a switch() statement.

You should be able to do all this initialisation with a single compare
or switch... which is cheap. I don't see a way to avoid having it, though.


ok, did that now.

cheers,


jörn

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[LAD] LV2 input buffer problem in ardour

[Jörn Nettingsmeier]

hi *!


whatever i do, i can't seem to get any data from an audio input port in 
my lv2 plugin. i made local copies of all buffer pointers (even though 
that shouldn't be necessary iirc), and in case ardour does in-place 
processing, i always cache sample N before writing it to the first 
output, so that i still have it available for the other outputs.


i even set lv2:inPlaceBroken, but no luck. now i was beginning to 
suspect i'm very very stupid, but then i installed the eg-amp.lv2 plugin 
(the simple amplifier example from the lv2core repository, which i've 
used as a template for my own code), and hey presto: no signal either. 
what the fu..ndamental problem might be, i can't say.


any insights most welcome.


thanks,


jörn

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Re: [LAD] LV2 and parameter interpolation

[Jörn Nettingsmeier]

On 02/06/2012 12:04 PM, Fons Adriaensen wrote:

On Mon, Feb 06, 2012 at 10:33:31AM +0100, Jörn Nettingsmeier wrote:


what i want to avoid is a ramp from default or previous values whenever
the plugin is run for the first time. in that case, i want to have no
ramp and start cold with the control port value.


For general purpose processing (as opposed to synthesis) that doesn't
matter in practice. Certainly not if the ramp is just one period.


it might if a DAW deactivates plugins on transport stop, then locates, 
then re-activates on play - can i be sure this won't happen with any host?
i certainly don't want my first note to go through a sweep of all plugin 
controls each time i press "play".



can i rely on control port data being available during activate() so
that i can initialize my current values to the control port values?

now i could set the current values to NULL during instantiate and
deactivate, but then i'd have an extra conditional in run(), which i'd
like to avoid.


This would be outside the main loop, so harmless. You may want something
similar to switch between interpolated or constant runs as well, so it
would just add one boolean.


true.


Note that making parameter smoothing dependent on the period size
is not really recommended, except again for synthesis. For the zita
series it is Verboten.


well, i've considered carrying loads of state with me and maybe even do 
cubic interpolation, but that's for later. the problem is i don't know 
in advance how many control points i need to cache for constant 
smoothing over time, because LV2 doesn't guarantee a constant number of 
samples per run().


best,


jörn


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[LAD] LV2 and parameter interpolation

[Jörn Nettingsmeier]

hi *!


total lv2 newbie trying to get my feet wet, so bear with me :)

i'm trying to dezip my control ports by using simple linear 
interpolation. to that end, i'm storing the current values of all 
control ports in extra fields in the LV2_Handle.
they get ramped to the desired value during run(), so that the current 
values will equal the ones set by the control ports at the end of each 
run().


now i'm stuck trying to understand where to initialize my current value 
fields.
what i want to avoid is a ramp from default or previous values whenever 
the plugin is run for the first time. in that case, i want to have no 
ramp and start cold with the control port value.


can i rely on control port data being available during activate() so 
that i can initialize my current values to the control port values?


now i could set the current values to NULL during instantiate and 
deactivate, but then i'd have an extra conditional in run(), which i'd 
like to avoid.


what is the recommended procedure to deal with this?


jörn


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Re: [LAD] First release of zita-lrx

[Jörn Nettingsmeier]

On 02/05/2012 07:02 PM, Fons Adriaensen wrote:

Hello all,

Now available on




sweet! builds and runs just fine, but it'll be a while before i'll have 
a proper test case.


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Re: [LAD] JACK Timemachine patch that allows OSC port to be specified at start up

[Jörn Nettingsmeier]

On 01/03/2012 08:17 PM, Tristan Strange wrote:

Hi all,

I've been trying to get the attached patch to steve at plugin dot org
dot uk unfortunately his email address has been unavailable for 5
days now.

The attached patch allows the OSC port a timemachine instance is
listening on to be specified at startup with a -o flag.

Can anyone else commit this to the project's git repository whilst
he's unavailable (providing it's up to scratch of course!) or forward
this on to Steve for me please?

This is the first patch I'll have submitted to a well used open source
project so I'm super keen to see it in timemachines code base.


in case steve doesn't catch this mail via the list, he can sometimes be 
found as "swh" on #lad @ irc.freenode.net, if you hang out there for a 
couple of days...

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Re: [LAD] Replaygain for video?

[Jörn Nettingsmeier]

On 12/09/2011 09:20 PM, Philipp wrote:


Question 1: Is there anything better than replaygain that should be used
instead?


not better as such, but carrying more weight, for sure: the EBU R128 
recommendation. fons has presented an implementation at lac 2011:


http://lac.linuxaudio.org/2011/papers/26.pdf

this is what all european radio stations will implement over the next 
few years, and similar plans are being discussed among tv broadcasters.


there is a german forum on loudness at 
http://www.tonmeister.de/forum/viewforum.php?f=34&sid=41ae6829b326641eed358a4b0d55a602 
, which includes some useful links. you'll have to register to post, though.



best,


jörn


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Re: [LAD] New stuff: zita-dpl

[Jörn Nettingsmeier]

On 12/05/2011 02:50 PM, Fons Adriaensen wrote:

On Mon, Dec 05, 2011 at 12:20:24PM +0100, Jörn Nettingsmeier wrote:



* i'm assuming you are looking at the maximum level of all channels, and
then apply the same amount of gain reduction to each of them.


Yes.


certainly
the way to go in speaker-based mixes. but as i already mentioned over
that coffee at ICSA, do you think it could be useful to add an ambisonic
mode which would apply the gain reduction only to the component that's
actually over? my hope is that the result is more subtle, because only
the source sharpness would change slightly, and with b-format, there is
no danger of irritating jumps of the source...


Actually that is not true - sources would move. Imagine a simple WXY
system. You have a source at 45 degrees and one at 90. The one at
90 (say some percussion) makes Y go into limiting. This means the
one at 45 will move forward.


right, now i see the problem..


So any gain change would have to affect equally at least all components
of the same degree. And even that can only be done for a short time,
as modifying the gain of one degree makes a complete mess of the decoding
- rE will drop sharply and rV can take on any value, even go 'negative'.
After at most a few tens of milliseconds the other components would
have to follow.

So that would require separating transient gain changes from longer
ones. This would be possible in e.g. a compressor, but for a peak
limiter (which has to ensure that no samples are over 0dB whatever
happens, and operates in feed-forward mode) it can become quite
difficult.

I tried something similar: to make transient gain changes affect only
the mid and high frequency part of the signal, but I had to abandon
that idea (at least for now) - it really gets very hairy.

Assuming it could be done, the limiter would have to know which
channels belong to the same degree. No problem if you only have
complete sets, but I want to support horizontal only and mixed
order sets as well. The plugin system in AmbMixer can handle
this. Others could mayby do it with a few more extensions, but
those would always imply explicit AMB support by the host.


thanks for elaborating. i agree, it's too hairy to be worth it, and i 
dislike tools that try to be too clever - even if it could be done, i'm 
sure it will confuse some users in corner cases, and the final stage of 
a production is not a good place for interesting surprises...


best,


jörn


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Re: [LAD] Ardour3 - strange problem

[Jörn Nettingsmeier]

On 12/05/2011 03:22 PM, Fons Adriaensen wrote:

Hello all,


Trying out Ardour3 but I'm blocked...

I deep-copied an existing A2 session, paradiso-2, to /audio/ardour3-sessions.
But A3 complains that it can't find the audio files in

/audio/ardour3-sessions/paradiso-2/interchange/paradiso-2/audiofiles

which is the right place, and the files are there.


for some reason, a3 handles the directory names in a different way when 
it comes to underscores and hyphens, i think. you might find that your 
directory is actually named interchange/paradiso_2 or something. just 
add the correct name to the search path when a3 prompts you, and you 
should be fine.
i have one session which i want to use with both a2 and a3 (works 
because a3 will always leave an a2-compatible backup session named 
$FOO-1.ardour when it imports), and somehow i had to add a symlink in 
interchange/ to make it work (ln -s NAME_AS_A2_THINKS_IT_SHOULD_BE 
NAME_AS_A3_SEEMS_TO_EXPECT_IT)...


haven't gotten around to reporting it yet...


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Re: [LAD] New stuff: zita-dpl

[Jörn Nettingsmeier]

On 12/05/2011 12:22 AM, Fons Adriaensen wrote:

Hello all,

First release of zita-dpl1. Look-ahead digital peak limiter.
1-16 channels (highest one determines gain reduction).

More at.


excellent, thanks!

builds and runs fine, will probably be using it for real before the end 
of the week. can i suggest two small extensions?


* it would be great to have the input gain displayed numerically in the 
display as well - i need to document my sessions so that i can always 
revisit them in case the client requests changes. now i could just make 
a screenshot (as i do for zita-rev1, which works because small 
deviations in setup don't have dramatic effects), but for limiting, i 
need absolute precision and 100% reproducible settings. trivial patch is 
attached.


* i'm assuming you are looking at the maximum level of all channels, and 
then apply the same amount of gain reduction to each of them. certainly 
the way to go in speaker-based mixes. but as i already mentioned over 
that coffee at ICSA, do you think it could be useful to add an ambisonic 
mode which would apply the gain reduction only to the component that's 
actually over? my hope is that the result is more subtle, because only 
the source sharpness would change slightly, and with b-format, there is 
no danger of irritating jumps of the source...



best,


jörn


diff -urN zita-dpl1-0.0.1/source/mainwin.cc zita-dpl1-0.0.1-new//source/mainwin.cc
--- zita-dpl1-0.0.1/source/mainwin.cc	2011-12-04 22:18:55.0 +0100
+++ zita-dpl1-0.0.1-new//source/mainwin.cc	2011-12-05 12:12:28.135265876 +0100
@@ -62,6 +62,7 @@
 _dispmap = XCreatePixmap (dpy (), _dispwin->win (), 315, 72, disp ()->depth ());
 _dispgct = XCreateGC (dpy (), _dispmap, 0, NULL);
 init_disp ();
+print_param (R_INPGAIN);
 print_param (R_THRESHD);
 print_param (R_RELTIME);
 _dispwin->x_map ();
@@ -227,14 +228,20 @@
 
 switch (k)
 {
+case R_INPGAIN:
+y = 0;
+v = _inpgain->value ();
+sprintf(s, "%2.1lf dB", v);
+C = XftColors [C_INPGAIN];
+break;
 case R_THRESHD:
-	y = 5;
+	y = 18;
 	v = _threshd->value ();
 sprintf (s, "%5.1lf dB", v);
 	C = XftColors [C_THRESHD];
 	break;
 case R_RELTIME:
-	y = 25;
+	y = 36;
 	v = _reltime->value ();
 if  (v >= 0.30f)  sprintf (s, "%4.2lf s", v);
 else if (v >= 0.03f)  sprintf (s, "%3.0lf ms", 1e3f * v);
diff -urN zita-dpl1-0.0.1/source/styles.cc zita-dpl1-0.0.1-new//source/styles.cc
--- zita-dpl1-0.0.1/source/styles.cc	2011-12-04 22:29:36.0 +0100
+++ zita-dpl1-0.0.1-new//source/styles.cc	2011-12-05 12:16:31.955871428 +0100
@@ -42,6 +42,7 @@
 XftColors [C_DISP_BG] = disp->alloc_xftcolor (0.1f, 0.1f, 0.1f, 1.0f);
 XftColors [C_TEXT_BG] = disp->alloc_xftcolor (0.9f, 0.9f, 0.9f, 1.0f);
 XftColors [C_TEXT_FG] = disp->alloc_xftcolor (0.0f, 0.0f, 0.0f, 1.0f);
+XftColors [C_INPGAIN] = disp->alloc_xftcolor (0.9f, 0.9f, 0.2f, 1.0f);
 XftColors [C_THRESHD] = disp->alloc_xftcolor (1.0f, 0.2f, 0.2f, 1.0f);
 XftColors [C_RELTIME] = disp->alloc_xftcolor (0.6f, 0.6f, 1.0f, 1.0f);
  
diff -urN zita-dpl1-0.0.1/source/styles.h zita-dpl1-0.0.1-new//source/styles.h
--- zita-dpl1-0.0.1/source/styles.h	2011-12-04 22:32:35.0 +0100
+++ zita-dpl1-0.0.1-new//source/styles.h	2011-12-05 12:11:12.092205429 +0100
@@ -30,7 +30,7 @@
 {
 C_MAIN_BG, C_MAIN_FG, C_DISP_BG,
 C_TEXT_BG, C_TEXT_FG,
-C_THRESHD, C_RELTIME,
+C_INPGAIN, C_THRESHD, C_RELTIME,
 NXFTCOLORS
 };
 
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Re: [LAD] Kernel RT patches ?

[Jörn Nettingsmeier]

On 11/01/2011 10:37 PM, Parisson wrote:

Hi all !

Do anyone know where are the RT kernel patches since the kernel.org reinstall ?

They used to be there:
http://www.kernel.org/pub/linux/kernel/projects/rt/


thomas gleixner is hosting them temporarily at https://tglx.de/~tglx/rt/


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Re: [LAD] New release of zita-convolver / jconvolver / fconvolver

[Jörn Nettingsmeier]

On 10/25/2011 11:55 PM, Fons Adriaensen wrote:

Meanwhile both zita-convolver and jconvolver have been updated,
so please update the AUR packages as well.


works now, thanks a lot!

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Re: [LAD] lv2plug.in server down?

[Jörn Nettingsmeier]

On 10/25/2011 11:51 PM, David Robillard wrote:

On Tue, 2011-10-25 at 17:20 -0400, David Robillard wrote:

On Tue, 2011-10-25 at 11:06 +0200, Jörn Nettingsmeier wrote:

hi *!


it seems that http://lv2plug.in is unhappy. it throws a 500 error while
redirecting to trac. does anyone know who is running that machine is so
i can drop them a note?


That would be me.

Sigh, I just finally put some effort into cleaning up the Wiki...

I'll look into it, thanks for the heads up.


Fixed.  Dreamhost updated python behind my back...


cheers!

turns out the svn repo was unaffected after all, so thanks to google 
cache i was able to check it out even while trac was down...


i've just updated from drobilla-svn as well - looks like you've been busy :)

chapeau, and best regards,


jörn

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Re: [LAD] New release of zita-convolver / jconvolver / fconvolver

[Jörn Nettingsmeier]

On 10/25/2011 08:00 PM, Jörn Nettingsmeier wrote:

On 10/18/2011 09:06 PM, Fons Adriaensen wrote:

Hello all,

New releases on<http://kokkinizita.linuxaudio.org:/linuxaudio/downloads>:


zita-convolver-3.0.2



jconvolver/fconvolver-0.9.1


i'm having trouble with the linker. when i install the new
zita-convolver, /usr/local/lib64/libzita-convolver.so still points to
the old version. jconvolver 0.9.1 then complains about

jconvolver.o: In function `main':
jconvolver.cc:(.text+0x15e): undefined reference to
`zita_convolver_major_version()'
jconvolver.cc:(.text+0x5b4): undefined reference to
`Convproc::print(_IO_FILE*)'
...
and so on.

even when i manually remove the old libraries and re-run ldconfig, no
unversioned libzita-convolver.so link will be regenerated (which makes
sense, given that the library is explicitly not backwards-compatible).
but then i'd need some way to specify the new version explicitly in the
jconvolver makefile, don't i? -lzita-convolver seems to always find the
old one only, or none at all if the generic link is removed.

any hints appreciated.


manually creating a link from libzita-convolver.so to 
libzita-convolver.so.3.0.2 fixes the jconvolver build issue, but it will 
also break backwards compatibility i guess.


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Re: [LAD] New release of zita-convolver / jconvolver / fconvolver

[Jörn Nettingsmeier]

On 10/18/2011 09:06 PM, Fons Adriaensen wrote:

Hello all,

New releases on:


zita-convolver-3.0.2



jconvolver/fconvolver-0.9.1


i'm having trouble with the linker. when i install the new 
zita-convolver, /usr/local/lib64/libzita-convolver.so still points to 
the old version. jconvolver 0.9.1 then complains about


jconvolver.o: In function `main':
jconvolver.cc:(.text+0x15e): undefined reference to 
`zita_convolver_major_version()'
jconvolver.cc:(.text+0x5b4): undefined reference to 
`Convproc::print(_IO_FILE*)'

...
and so on.

even when i manually remove the old libraries and re-run ldconfig, no 
unversioned libzita-convolver.so link will be regenerated (which makes 
sense, given that the library is explicitly not backwards-compatible). 
but then i'd need some way to specify the new version explicitly in the 
jconvolver makefile, don't i? -lzita-convolver seems to always find the 
old one only, or none at all if the generic link is removed.


any hints appreciated.

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Re: [LAD] [OT] Another giant gone: RIP John McCarthy

[Jörn Nettingsmeier]

On 10/25/2011 12:27 PM, Dave Phillips wrote:

Greetings,

A sad day yesterday:

http://www.wired.com/wiredenterprise/2011/10/john-mccarthy-father-of-ai-and-lisp-dies-at-84/


(rest(in(peace)))


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[LAD] lv2plug.in server down?

[Jörn Nettingsmeier]

hi *!


it seems that http://lv2plug.in is unhappy. it throws a 500 error while 
redirecting to trac. does anyone know who is running that machine is so 
i can drop them a note?




thanks,


jörn

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Re: [LAD] LAC 2012: the Linux Audio Conference - Call for Participation

[Jörn Nettingsmeier]

hey fernando, this:

http://lac.linuxaudio.org/2012/img/lac2012.png

is utterly charming :-D

thanks for all your work, looking forward to seeing you all in stanford 
next year!



best,

jörn


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[LAD] [OT] Program now online - International Conference on Spatial Audio, Nov 10-13 2011, in Detmold, Germany

[Jörn Nettingsmeier]

Hi *!


The timetable for the International Conference on Spatial Audio 2011 in 
Detmold/Germany is now online:


http://www.icsa2011.org/vdt/webdownloads/icsa2011/ICSA_Program_Schedule_2011-10-02b.pdf

The conference emphasizes the scientific and technical aspects of 
spatial audio, but a number of workshops and presentations will be 
devoted to the artistic and esthetic possibilities and challenges as well.


In addition to five structured sessions on various spatial audio topics, 
the conference will feature installations of WFS + Height, Auro-3D, 
Binaural and Higher-order Ambisonic systems. During a three-day 
recording session last week totalling more than 140 microphone signals, 
we have recorded several chamber music ensembles simultaneously for all 
those methods and will offer conference participants a first-hand 
opportunity for comparison.


Complementing the two fixed WFS installations available in Detmold, an 
additional temporary 40-speaker setup will be rigged for WFS and 
Higher-order Ambisonics reproduction, and one seminar room will be 
equipped for 9+2 channel Auro-3D playback.



Registration is now open.


Best regards,


Jörn

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Re: [LAD] Fwd: NYT Science: Sound, the Way the Brain Prefers to Hear It

[Jörn Nettingsmeier]

On 09/10/2011 10:57 PM, Fons Adriaensen wrote:

On Sat, Sep 10, 2011 at 01:33:19PM -0700, Niels Mayer wrote:

http://www.nytimes.com/2011/09/06/science/06sound.html?_r=1&pagewanted=print
-->  Dr. Chris Kyriakakis’s audio lab at the University of Southern California
-->  http://www.audyssey.com/audio-technology/multeq
-->  http://en.wikipedia.org/wiki/Audyssey_Laboratories


The usual hype and misinformation designed to impress those
who don't even understand the basics.

<..>

Creating a Wikipedia page to advertise a business is really
in bad taste and illustrative of the mindset of those who do
it - completely ignoring and showing a complete lack of respect
for the fundamental choices made by the creators of Wikipedia.



aw, come on fons, don't be so harsh. this guy invented the loudness 
button, that sure warrants some encyclopedic fame:


"A video of the Eagles singing “Hotel California” sounded nice to a 
visitor until Audyssey’s hardware director, Andrew Turner, pointed out 
that there was no bass when the volume was low. He flicked a switch and 
the bass returned, enriching the music with startling effect."


in case anyone's wondering what the blabber on their website is all 
about, here is a bullshit-free tool that does exactly the same (although 
not fully automatically) and the author even explains to you how it all 
works: http://drc-fir.sf.net


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Re: [LAD] [ANN] International Conference on Spatial Audio, Nov 10-13 2011, in Detmold, Germany

[Jörn Nettingsmeier]

quick reminder:
the deadline for submissions to ICSA 2011 has been extended to August 15.
sorry i won't be able to address any questions you might have, since i'm 
about to go on vacation, but you can direct inquiries to the mail 
address given below.

i'd be happy to meet some fellow linux users and devs this autumn in
detmold! apologies for the hobbyist and freelancer unfriendly admission
fee - we have tried to cut costs wherever possible and we're still
comparably cheap, but i can see how it's a serious deterrent for some
people. however (plug!): there is a substantial discount for accepted
papers, so get busy :-D


here's the official message from our chair, prof. dr. malte kob:
>
> Dear colleagues,
>
> due to many requests the deadline for submission of contributions to
> the International Congress on Spatial Audio - ICSA 2011 has been
> extended until August 15, 2011.
>
> The congress offers scientific contributions and practical
> demonstrations for spatial recording and reproduction methods such as
> wave field synthesis, higher-order ambisonics, multichannel
> surround/stereo, binaural technique and 3D systems. Music
> performances and listening test will be performed in the frame of the
> conference.
>
> You are welcome to submit proposals for oral presentations, posters,
> workshops or product presentations via the congress webpage:
>
> http://www.icsa2011.org
>
> Contact address: icsa-2011-t...@tonmeister.de

best regards,


jörn





--
Jörn Nettingsmeier
Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487

Meister für Veranstaltungstechnik (Bühne/Studio)
Tonmeister VDT

http://stackingdwarves.net

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Re: [LAD] Feature requests: add JackSession support

[Jörn Nettingsmeier]

On 07/04/2011 03:23 AM, Paul Davis wrote:

2011/7/3 Dave Phillips:

Jörn Nettingsmeier wrote:


... none of the audio stuff i routinely do everyday would be possible
without jack.


Amen to that.


I disagree with both of you. I think what you really mean is "none of
this would be possible without some system for interconnecting
processing elements together in flexible, creative,possibly
unanticipated ways that also leaves the developers of those elements
free to do things in their own way".

that much i'd agree with. but this is not a description that requires
that the solution be at the process level.


well, like ladspa, jack has its time and place, and it has triggered an 
amazing surge of activity. to me, it's the best infrastructure in the 
market.
when it was conceived, there was no way to do what it does except at the 
process level. and now that we know how to combine multiple gui toolkits 
in a single process, it still looks very good.


and as long as jack keeps scaling over multiple cpu cores, we have 
cycles in abundance - so where's the harm in some inter-process 
communication overhead?


rather than lumping all audio into a single process, i'd say jack should 
even strengthen the boundaries between processes, such that it doesn't 
die as easily when a node in the graph goes bad, to make it even better 
suited for live use than it already is.

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Re: [LAD] Feature requests: add JackSession support

[Jörn Nettingsmeier]

On 07/03/2011 10:14 PM, Folderol wrote:


So (excusing my ignorance) are we approaching a brick wall or is there a way
out?


i wouldn't know. if it's a brick wall, i'm perfectly happy banging my 
head into it day after day, and so are my customers.



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Re: [LAD] Feature requests: add JackSession support

[Jörn Nettingsmeier]

On 07/03/2011 07:32 PM, Paul Davis wrote:


i feel that if you spend too long reasoning about this, you will
conclude, as I have, that JACK was actually a mistake (at least in
terms of the basic framework in which to glue together different
things processing data streams). the absence of a plugin API that was
likely to be adopted by all/most developers back in 2000 is what gave
rise to this situation. there's a limit to how far you can push the
usability of a "DAW" built out of N independent processes, each one
running code developed by different developers with no awareness of
the others.


true if you are arguing from the "streamlined workflow" pov only.

but the unix way is "make simple things easy" (= use one daw and 
everything else is a plugin) _and_ "make hard things possible". jack 
doesn't hurt the former and excels at the latter. none of the audio 
stuff i routinely do everyday would be possible without jack.
it's great to be able to combine stuff in ways the developers did not 
(need to) anticipate. that usually means combining separate processes.


jack is really unixy, and it certainly deserves the highest praise i 
could think of: it's totally |y.

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Re: [LAD] Determining Phase

[Jörn Nettingsmeier]

On 06/27/2011 02:35 PM, pshir...@boosthardware.com wrote:

pshir...@boosthardware.com wrote:



I'm looking for a way to adjust the phase of a signal rather than the
amplitude. Does such a plugin already exist? If not which ladspa plugin
would be the most suitable to start from?


check out the available allpass plugins - iirc, there's one in the CMT 
package distributed by richard furse.


if you need a 90° phase shift, you could use jconvolver with the hilbert 
transform kernel that comes with it, or check out fons uhj encoder in 
the AMB plugin package - it implements a 90° phase shift as an IIR filter.


what exactly are you trying to do?

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[LAD] [OT] Comparison recordings of CoreSound Tetramic and Soundfield ST450

[Jörn Nettingsmeier]

Hi *!


For those interested in Ambisonic surround sound: finally I've managed 
to upload some side-by-side recordings of a Tetramic and the new ST450 
which have been sitting on my harddisk for way too long.


http://stackingdwarves.net/download/TetraMic_vs_ST450/

Hopefully the recordings are worth your time even if you're not 
currently shopping for a new surround microphone.


Attached is a README that goes with the audio files.


Enjoy,


Jörn



A comparison of the CoreSound Tetramic (CS 2050) 
  and a pre-series prototype of the Soundfield ST450.


  Jörn Nettingsmeier 
January 2011

Thanks to Soundfield Ltd. and S.E.A. Vertrieb & Consulting GmbH for
providing me with an ST450 for testing.

The Tetramic was recorded in A-Format and processed with Fons Adriaensen's
Tetraproc using the custom filter coefficients provided by CoreSound. 
The ST450 went through its preamp/converter box, which produces B-Format.
Both mics were co-incident, the ST450 mounted upright in its cradle, and
the Tetramic on top, in end-fire mode. They were recorded through an RME 
Micstasy with digitally controlled and matched gain, via ADAT through a 
Focusrite Saffire PRO26, to a Linux Audio workstation running Ardour.
The tracks have been roughly matched for equal loudness and rotated for
congruent localisation by ear.


All files are classical first-order B-format 24bit WAV at 48khz.
If you want to listen to them on stereo speakers, you can either use a UHJ
encoder, or a virtual stereo microphone. Both are available as LADSPA
plugins for Linux and Mac users as part of Fons' AMB plugin package. For
users of that other operating system, Google is your friend.
For serious A/B comparison, you should probably import each pair of files
into some DAW and play it back in sync while switching between the mics. 
For recreational listening under Linux, try starting JACK and AmbDec with
a suitable first-order decoder, and then use

mplayer -channels 4 -ao jack:port=ambdec 

.

*.*

First, two brief excerpts from a concert by "The Kites", a recorder ensemble
from Germany. The concert was part of the Montag-Tontag series at the
Kunsthaus Essen, a very small location that holds about 40 guests if you
squeeze them a bit. Way too close-miked for recorders, directly in front of
the stage, about 1.5m away from the musicians, at standing ear-height. 
And of course the acoustics are quite hostile to recorders, so Fons'
zita-rev1 had to help a bit. Still too direct for my taste, but good for 
comparing mic performance.

Ye olde Englishe musicke, with some very neat ad-libbing over "Pastime with
Good Company":

The_Kites-Three_Pieces_from_the_Court_of_Henry_VIII-CoreSound_TetraMic-24bit-48k.amb
The_Kites-Three_Pieces_from_the_Court_of_Henry_VIII-Soundfield_ST450-24bit-48k.amb

Dave Holland's "Conference of the Birds". The Petzold bass flute was gently
amplified using an AER acousticube amp on stage. Something apparently bumped 
into the mic stand during the performance, thus a GLAME 5-pole highpass at 100Hz
was applied to both mics. Same reverb settings as before.

The_Kites-Conference_of_the_Birds-CoreSound_TetraMic-24bit-48k.amb
The_Kites-Conference_of_the_Birds-Soundfield_ST450-24bit-48k.amb

You will notice that the applause is louder on the rear left - that's how
the audience was seated.

*.*

Next, a 12-minute free improvisation by Vincent Royer on viola, Stefan Werni on
double bass and Thorsten Töpp on classical guitar. Same location, roughly
the same microphone position. No amplifiers - yes, that _is_ a weapons-grade
double bass. Beautiful collaboration with a keen sense for space and tone - 
give it a spin even if the word "free jazz" gives you the creeps, chances
are you'll like it. In any case, it's a great test signal.

No artificial reverb this time, the Kunsthaus works quite well for this kind
of intimate music.

Royer-Werni-Töpp_Improvisation_#3-CoreSound_TetraMic-24bit-48k.amb
Royer-Werni-Töpp_Improvisation_#3-Soundfield_ST450-24bit-48k.amb

You should hear the viola to the right, the double bass in the center, and
the guitar to the left.

*.*

My private conclusion:

The ST450 is quite bassy and warm, which can be nice but it's not how I like
my microphones. However, it's easily tuned to taste with some gentle EQ, and
I can see how the basic sound would appeal to most musicians and location
recordists.
The Tetramic does an amazing job for the price, if and only if the music is 
loud enough.
I have no clear preference for the sound of either, although the ST450 is
nicer to tricky instruments such as strings, or voices.
But if you need better signal-to-noise ratio, there's no way around the
ST450. Of all the Soundfields I have used so far (a DSF-1, a Mark IV, an
ST350), the ST450 has the best localisation.
Unfortunately, it will set you back by about 4k pounds, which makes the
Tetramic look really good in comparison. But that price includes a very good
beltpack-sized matched preamp wit

Re: [LAD] Determining Phase

[Jörn Nettingsmeier]

On 06/26/2011 05:47 PM, Arnold Krille wrote:

On Sunday 26 June 2011 11:58:54 Jörn Nettingsmeier wrote:

On 06/26/2011 10:50 AM, pshir...@boosthardware.com wrote:

Other than that, I'd make a really cool spectrum analyzer that ran the
Fourier analysis on two channels, correlated their phases then made a
+/- line vs. frequency for all to see so that the phase of the
components of the spectrum could be watched for phase relationships.

that is precisely what dual-fft tools for p.a. system calibration do,
and they're extremely useful.
they allow you to constantly monitor the system with _program_ material,
without having to use MLS noise or any other specific measurement signal.
one channel is used for the direct signal from the mixer, the other is
fed by a measurement microphone with delay compensation. you get instant
phase and amplitude response. good systems also give you an additional
"confidence" curve that tells you how much you can trust which parts of
the spectrum. for instance, if your program material is a boy soprano,
the confidence of the measurement in the low end is practically zero.


I use(d) japa for this :-) Altough that lacks phase display...


it will do the job nicely in its difference setting, but you have to 
find (and compensate for) the delay manually, like a real man.
modern p.a. tools allow sissies like yours truly to just click on the 
"find delay" button and be done :-D

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Re: [LAD] Determining Phase

[Jörn Nettingsmeier]

On 06/26/2011 10:50 AM, pshir...@boosthardware.com wrote:

So, perhaps any Haas-effect plugin would satisfy Patrick's needs.


so this is about panning? that's actually pretty easily done with just a 
time delay in addition to level difference. unless you want to spread 
complex sounds out in space, in which case you replace the delay with a 
frequency-dependent allpass.


careful with the term "haas effect". all the poor guy did was check the 
time window in which an echo would not be perceived as a separate event. 
basically, the haas effect says you can get hide a P.A. delay system 
that's 10dB louder than the main P.A. if it hits between 10 and 30ms 
later, and still maintain good "on-stage" localisation, with the delay 
being practically inaudible.

sometimes also called the "law of the first wavefront".

this has nothing to do with stereo localisation. time delays relevant 
for left-right localisation are in the 0 - 1ms range, some authors give 
a bit less, others a bit more.



Other than that, I'd make a really cool spectrum analyzer that ran the
Fourier analysis on two channels, correlated their phases then made a
+/- line vs. frequency for all to see so that the phase of the
components of the spectrum could be watched for phase relationships.


that is precisely what dual-fft tools for p.a. system calibration do, 
and they're extremely useful.
they allow you to constantly monitor the system with _program_ material, 
without having to use MLS noise or any other specific measurement signal.
one channel is used for the direct signal from the mixer, the other is 
fed by a measurement microphone with delay compensation. you get instant 
phase and amplitude response. good systems also give you an additional 
"confidence" curve that tells you how much you can trust which parts of 
the spectrum. for instance, if your program material is a boy soprano, 
the confidence of the measurement in the low end is practically zero.

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Re: [LAD] Determining Phase

[Jörn Nettingsmeier]

On 06/26/2011 04:17 AM, Fons Adriaensen wrote:

On Sun, Jun 26, 2011 at 12:22:58AM +0200, Jörn Nettingsmeier wrote:



- Phase is related to delay but it is not the same thing.
Group delay is again something different. Mixing up all
these is not going to help anyone understand things any
better.


well, i was trying to connect all those buzzwords... but you are right, 
it should be done more carefully. let me try again.


*delay* makes the *phase* response curve steeper. it doesn't introduce 
any non-linearities in the phase response.


amplitude response over frequency can be interpreted "as-is", but phase 
response needs to be looked at with your first-derivative glasses on: a 
system comprising a perfect speaker and your perfect ear only has zero 
phase when you stick your head into the speaker.

as soon as you move away, the phase drops, the steeper the further you go.
morale: constant amplitude response is what we want. constant phase 
response almost never happens, because of delays that creep in. instead, 
we want _linear_ phase response.


*group* *delay* is a *time* *delay* for a specific frequency. if you 
have a linear-phase system, the group delay is a _constant_: high 
frequencies may be phase-shifted by more cycles, but the time it takes 
them to arrive is the same as for low frequencies.
i think you get the group delay when you differentiate the phase 
response wrt frequency (but don't believe me when i talk calculus...)


it's important not to confuse phase delay with group delay: phase delay 
talks about a number of cycles of delay, whereas group delay is about 
time. when you want to assess how well a system responds to transients, 
you don't care how often the high frequencies have been wiggling around 
on the way to your ear drum - you want them to arrive at the same time 
as the low frequencies. hence, you care about the group delay, not the 
phase delay.


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Re: [LAD] Determining Phase

[Jörn Nettingsmeier]

On 06/26/2011 12:04 AM, Emanuel Rumpf wrote:

2011/6/25 Fons Adriaensen:

On Sat, Jun 25, 2011 at 01:55:05PM -0500, Gabriel M. Beddingfield wrote:


Do you mean... for a very simple sine wave?

Assuming yes:

   p = asin( x / A )

Where:

   A is the amplitude of the sine wave


you mean the maximal amplitude (-MAX<= x<= +MAX) , I guess ?


   x is the value of the sample (-A<= x<= A)
   p is the phase of the wave in radians (-pi/2<= p<= pi/2)


And what if the phase is<  -pi/2 or>  +pi/2 ?


since x<= A (always), that result is not possible



?!

it seems you have just proven that the maximum duration of any pure tone 
is 1/f. that is quite extraordinary. might it even be the explanation of 
the almost mythical 1/f noise? all those tones suddenly realizing they 
have to stop or violate rumpf's lemma :-D


sorry, couldn't resist...

but seriously, it does make a lot of sense to talk about arbitrarily 
large phase angles. take a look at a real-life speaker system: it's not 
uncommon for the HF to lag behind the subs several complete cycles after 
passing through the crossover.


even a perfectly phase-linear theoretical speaker exhibits them:
in fact, if you stand 3.4m away from a speaker, the phase angle of a 
100hz tone at your ear will be 360° relative to the membrane, while a 
200hz tone will be at 720°, and so on.
that's where delay becomes "group delay", i.e. the same constant time 
delay implies different phase angles depending on frequency, pretty much 
arbitrarily large as the frequency rises.








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Re: [LAD] Determining Phase

[Jörn Nettingsmeier]

On 06/25/2011 04:23 PM, pshir...@boosthardware.com wrote:

Hi,

Can anyone point me to a simple code example for how to determine the
phase at a specific time in a waveform?

ex. if I have a sample that is 5 seconds long and want to know the phase
at 2.5 seconds


talking about the "phase" at some particular point in time doesn't make 
sense. first you would have to do an fft (whose resolution is bound by 
the window size), and then you'd have to look at each frequency bin 
separately, because they will all be at different relative phase.


when you talk directly into my ear, the voice/ear system will be roughly 
zero phase. when i move away even a few inches, there will be a linear 
phase "distortion" due to the fact that i'm now one wavelength away at 
some mid frequency, but several wavelengths at high frequencies.


so "phase" is not really a meaningful thing unless you are talking about 
recombination of strongly correlated signals.


another example: if your sample consists of a sine at 100hz and one at 
200hz, after 10ms the first component will be at 0° while the second is 
at 180°, iff both started at 0° in the first place. but why would anyone 
care? with real-life sounds, this whole exercise becomes meaningless.


what are you trying to accomplish?

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Re: [LAD] Updates of zita-rev1 and zita-at1

[Jörn Nettingsmeier]

On 06/15/2011 12:26 AM, Fons Adriaensen wrote:

Hello all,

Some updates now available at




yay. yummy software updates :)


zita-rev1-0.2.1
zita-rev1-0.2.1 and zita-at1-0.2.2


it seems that, despite the ending ".bz2", both files are actually gzipped...
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Re: [LAD] LAC 2011

[Jörn Nettingsmeier]

On 06/02/2011 02:46 PM, Robin Gareus wrote:


We'd like to thank all speakers and everyone who volunteered to make
this an enjoyable event; in particular Frank Neumann, John Lato, Victor
Lazzarini and special thanks to Jörn Nettingsmeier.


after taking a humble bow and adding my thanks to viktor, john, and 
frank, i'd like to point out that robin did all the website grunt work. 
that means if you see him, buy him a beer.


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Re: [LAD] mp3-player needs sendmail - really?

[Jörn Nettingsmeier]

On 05/29/2011 09:11 AM, Jens M Andreasen wrote:

Anybody got any idea why the realplayer insist on installing sendmail as
well? Is it a rootkit, intending to turn me into a spam-node?


it's realplayer's way of telling you that _you_ don't need realplayer.

honestly, there are metric tons of great players out there, why this one?
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Re: [LAD] RME FIREFACE 400? RME MULTIFACE II?

[Jörn Nettingsmeier]

On 05/22/2011 11:43 AM, Ralf wrote:

Hi :)

I only watched pictures and read texts but didn't hear one of those RME
devices, anyway, until now I tend to order the RME FIREFACE 400 or RME
MULTIFACE II if they shouldn't cause issues with Linux. The two HDSP be
possible too.

Any experiences, information?


ralf, as usual, your level of disinformation is astonishing.

the fireface is an ieee1394 device, and the ffado drivers are somewhat 
experimental. so yes, this would be an "issue with linux". i suggest you 
check the ffado.org website and look through the ffado mailing list 
archive to see if the current level of support is sufficient for what 
you want to do.


the multiface ii is a break-out and converter box. it doesn't work on 
its own. consequently, there are no driver issues associated with its 
use, other than the firmware upload which the host computer has to take 
care of when you boot the device.
what you want to look into is the corresponding pci(e) card with the 
connector that looks like firewire but isn't (proprietary rme protocol).

the quality of the converters on the multiface leaves nothing to be desired.

the 9652 and 9636 cards are digital-only, with two resp. three adat i/o 
connectors. consequently, their "sound quality" is perfect.

you will have to combine them with some external adat ad/da converter.

most if not all rme cards come pci and pci express flavours. as you 
mentioned in a previous posting, the pci prices have dropped a lot, but 
when you consider one, factor in the cost for mainboards with pci slots 
- they will become quite rare in mass market in the near future, and 
then you would have to pay extra to get some "industry"-type product 
that still has them. for an idea of the extra cost, try shopping for an 
industry board with ISA slots today (still needed to run legacy process 
control cards and whatnot).


all rme cards i've come across will happily run at 64 frames, and some 
of the newer ones let you go down to 32 or 16, although i have not tried 
this yet.


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[LAD] Linux Audio Conference 2011 starts today with live video stream coverage

[Jörn Nettingsmeier]

hi everyone!


just a quick reminder that lac 2011 is starting today at 10:00 utc+1, 
hosted by NUI maynooth in ireland.


find the program at http://lac.linuxaudio.org/2011.

live streams will be available from streamer.stackingdwarves.net in ogg 
theora format.


remote participants are invited to join us on irc for conference-related 
chitchat and real-time feedback. at the end of each session, you will 
get the chance to ask questions to the presenter, which will be relayed 
to the crowd in maynooth by a local chat operator.



best,


jörn

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[LAD] LWN feature: Gleixner on the inclusion of the PREEMPT_RT patches into mainline

[Jörn Nettingsmeier]

hi!

guess this will be of interest to LAD patrons:

http://lwn.net/SubscriberLink/440064/dff63f71e1002605/

best,

jörn

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Re: [LAD] [ANN] International Conference on Spatial Audio, Nov 10-13 2011, in Detmold, Germany

[Jörn Nettingsmeier]

On 04/14/2011 07:26 PM, Arnold Krille wrote:

On Thursday 14 April 2011 19:08:42 Jörn Nettingsmeier wrote:

sorry for the slightly off-topic post, but since spatial audio has been
a frequent topic lately, i think some people here might be interested.


Neither my eyes nor my link-highlighting mail-client found a webaddress
pointing to the conference website for details. Can you post that too?
Wanna find out about the actual fee before I wet my eyes...


woops, the link got eaten. it's http://www.tonmeister.de/icsa2011. but 
there is no additional information there yet, just what i posted.

the fee has not been determined yet, that will take another couple of weeks.

cheers,

jörn


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[LAD] [ANN] International Conference on Spatial Audio, Nov 10-13 2011, in Detmold, Germany

[Jörn Nettingsmeier]

hi *!


sorry for the slightly off-topic post, but since spatial audio has been 
a frequent topic lately, i think some people here might be interested.
linux or FLOSS won't be exactly in the limelight, but yours truly will 
make sure there are at least 2-3 boxes with your favourite OS and audio 
tools humming along in various places. oh, and you might come early and 
watch a few high-end mixing consoles boot - the startup screen will 
bring tears to your eyes (as will the price tag, unfortunately :)


unfortunately, there will have to be an admission fee, which we haven't 
decided on yet. but we're trying to keep it reasonable. don't shout at 
me when it turns out to be a bit more costly than LAC, though...



jörn


*.*


ICSA 2011 - International Conference on Spatial Audio
November 10 - 13, Hochschule für Musik, Detmold

Organizers:

Verband Deutscher Tonmeister (VDT), in cooperation with
Deutsche Gesellschaft für Akustik e.V. (DEGA), and
European Acoustics Association (EAA).

Contact/Chair:

Prof. Dr.-Ing. Malte Kob
Erich-Thienhaus-Institut
Neustadt 22, 52756 Detmold
Mail: icsa2011attonmeister.de
Phone: +49-(0)5231-975-644
Fax: +49-(0)5231-975-689

Summary:

The International Conference on Spatial Audio 2011 takes place from 
November 10 to 13 at Detmold University of Music.
This expert‘s summit will examine current systems for multichannel audio 
reproduction and complementing recording techniques, and discuss their 
respective strengths and weaknesses.
Wavefield synthesis systems, a higher-order Ambisonics array, as well as 
5.1/7.1 installations in diverse acoustic environments will be available 
for comparative listening tests during the conference.
Structured plenary talks, paper and poster sessions will revisit 
fundamentals and present latest research.
A series of workshops will be dedicated to practical implementations of 
spatial sound capture and playback methods, and their esthetic and 
psychoacoustical implications for music perception.
Concerts that include music specially arranged for the conference will 
let you experience various spatial sound systems in "live" conditions.


Call for papers and music:

Your contributions are welcome, either as presentations, posters, or 
workshops. Submissions will undergo a review process, and accepted 
contributions will be published in the conference proceedings.

The conference language is English.

We are planning structured sessions on the following topics:

* Multichannel stereo
* Wave field synthesis
* Higher-order Ambisonics / spherical acoustics
* 3D systems
* Binaural techniques

An additional session will be dedicated to related miscellaneous 
contributions, such as hybrid systems and perception/evaluation of 
spatial music reproduction.



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Re: [LAD] Writing an IEEE 802.1BA (AVB) compliant network backend for Jack2

[Jörn Nettingsmeier]

On 04/05/2011 11:53 AM, Christoph Kuhr wrote:

Again its me just another list,

apparently the ASUS 890FX mainboard has a yukon 88e8059 nic, which i
have not verified yet.
the marvell yukon 88e8059 specs say this nic is avb ready.

is there anyone who can verify this?


during a seminar of $BIG_AMERICAN_SPEAKER_MANUFACTURER last month, the 
presenter talked about AVB and mentioned that marvell (among other 
manufacturers) has had full avb capability on their silicon for quite a 
while, and it was just a matter of activating it (which they wouldn't do 
on low-end gear of obvious reasons). so if they say their product does 
it, i'd say the chances are good. whether the implementation is complete 
and correct though, i can't say.


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Re: [LAD] Good Mixer Library

[Jörn Nettingsmeier]

On 04/01/2011 08:17 AM, Patrick Shirkey wrote:

On 04/01/2011 04:03 PM, Devin Anderson wrote:

I'm confused. Are you trolling about not trolling?



Not at all! I am genuinely concerned about the possibility of offending
the fervant believers in the righteous power of the App that shall
remain nameless and the sanctity of that hallowed institution leading
the way for us mere mortals in the eternal quest for realtime
performance and enlightenment of the dance massive.

It is clear that discussion of the Taboo could lead to a fatal turn of
events that might leave us completely at the mercy of Government
corruption and ineptitude while we endeavour to shield ourselves from
the ensuing radioactive fallout of a massive nuclear event caused by the
shifting techtonic plates due to the impact of global warming from the
excessive consumption of fossil fuels brought about by the need to
provide a rock solid environment for Audio posers and Professionals
alike to achieve their goals of world domination while at the same time
enjoying all the benefits of a successful career in the music industry
and the wealth, status and success that goes with selling out to the
highest bidder to provide corporate sponsored entertainment of the mass
market driving the entertainment market into fits of passion at the mere
sound of the latest autotuned sample while overlayed with the darkest
philosophical perspective due to the Mass Media Industrial Military
complexes desire to manipulate our thoughts with their pervasive agenda
of death and destruction on a global scale in order to confuse into
ignoring the depth of corruption and negligence inherent to the self
fulfilling system of Political and Corporate greed enhanced by the
merger of the fiat currency system with multimedia devices running
competing mobile operating systems to ensure vendor lock in while we
give away our rights in order to make sure we can enhance the facial
recognition technology of the Elite Societies that require total
dominance of the international global market for divine intervention in
the nationalised institutions that combine technologies to create a
broad based award winning platform for teleportation of our quantum
entangled states while we injest our daily dose of vitamins to counter
the effects of long term exposure to radionuclides emitted into our
environment for with a full lifetime of hundreds of thousands of years.


this might be the intellectual equivalent of zero point energy.
it's pretty random, and nobody will be able to extract anything useful 
from it.


forgive me for pouring diesel on the flames, but it seemed the safest 
option - water has been reported to combust spontaneously around here.


back under my stone,


jörn
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Re: [LAD] [ANN] IR: LV2 Convolution Reverb

[Jörn Nettingsmeier]

On 02/27/2011 01:05 AM, Thomas Vecchione wrote:

Fons

Being someone that tracks recordings live constantly, I am curious, if
the singer only wanted to overdub one section of their vocals with
another, and you are not touching the remainder of the recorded tracks,
exactly what stops you from doing a standard punch in/out in your example?


in classical recording sessions, overdubs happen rarely if ever.
i guess the situation here is that multiple full or partial takes were 
recorded with the full ensemble, and the editing happens afterwards, 
when all musicians are gone.
iiuc, the soloist requested one section to be replaced with another 
take. since there is no "click", this usually means that the part after 
the new spliced-in section will move in time, slightly.
which is a bit of a problem in ardour while you haven't consolidated 
region fragments (which often you don't want to do until the very end), 
because you have to be very careful to move all subsequent regions.
easy in the vertical thanks to edit groups, but quite hard in the 
horizontal. or maybe i'm overlooking yet another feature?


best,

jörn



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Re: [LAD] RDF libraries, was Re: [ANN] IR: LV2 Convolution Reverb

[Jörn Nettingsmeier]

On 02/26/2011 05:12 AM, Paul Davis wrote:

2) Is Richard contactable? If not, what was his contribution to


his company in london is active and in the audio software business
(ambisonics for the masses, IIRC). i can't remember its name right
now, but i bumped into by accident a month or two ago.


richard is reachable as first name at muse440 dot com.
his company is http://blueripplesound.com

is there a problem wrt ladspa unique ids?
last time i tried, he responded within 1-2 days, but granted that's been 
a while.


best,

jörn

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Re: [LAD] [ANN] IR: LV2 Convolution Reverb

[Jörn Nettingsmeier]

On 02/24/2011 06:39 PM, Paul Davis wrote:

well, panners in a3 are now plugins, of a fashion (they are a bit
different from normal plugin APIs for a variety of reasons, primarily
the fact that they never do in-place processing). its quite likely
that at least the simplest of your ambi LADSPA plugins will show up in
that fashion in the not too distant future, with a GUI very similar to
the one in that screen shot.


nice :)


when i was in berlin attending "serious" concerts of electro-acoustic
music, there seemed to be quite a range of opinion about ambisonics
versus VBAP. although i tend to take your word for it on such matters,
there were quite several smart people with years of experience in
multi-speaker setups who had real issues with ambisonics and felt that
VBAP was generally preferable. i don't want to make people choose.


actually, i didn't like what i heard of vbap so far, but it does work as 
advertised, even under very adverse conditions. unless we have franz 
zotter's partial sphere magic firmly in place for all workflows, *and* 
there are kick-ass, fool-proof automatic ambisonic decoder generators 
for even the most pathological speaker layouts, vbap sure has its place.


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Re: [LAD] LAD Activity (WAS: [ANN] IR: LV2 Convolution Reverb)

[Jörn Nettingsmeier]

On 02/23/2011 11:16 PM, David Robillard wrote:

On Wed, 2011-02-23 at 20:05 +0100, Robin Gareus wrote:

Note: all posts to LAA are being moderated. Once they make it through
moderation, the message will get on the linuxaudio.org front-page and is
automatically added to planet LAD. If you prefer to blog and would like
the blog to be included in planet.linuxaudio.org: read the sidebar of
planet LAD.


For the record, I have found it frustrating that if you announce a
release on LAA, and blog it, it ends up on Planet LAD twice.

I suppose I just shouldn't push those announcements to the RSS feed
picked up by Planet LAD, but.. well, it /is/ LAD :)


i don't really mind, unless the blog title is such that i can't readily 
identify it as a duplicate of the LAA announcement in my simplistic rss 
reader (a firefox live bookmark).

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Re: [LAD] [ANN] IR: LV2 Convolution Reverb

[Jörn Nettingsmeier]

On 02/24/2011 12:47 PM, Fons Adriaensen wrote:

On Wed, Feb 23, 2011 at 05:11:33PM -0500, David Robillard wrote:


Entirely Redland free. I hand-wrote a Turtle parser and serialiser.

In short, it's been a PITA for everyone in numerous ways since day one.


Congratulations. I mean it.


seconded. it used to be quite tricky to chase the correct versions of 
redland and its various dependencies, and i recall that at least one 
library down dependency hell road would require hefty manual massaging 
to even compile, at least on the distros i tried it on.


> This is probably the best move in the entire history of LV2.

now that's a bit nasty :)
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Re: [LAD] [ANN] IR: LV2 Convolution Reverb

[Jörn Nettingsmeier]

On 02/24/2011 12:43 PM, Fons Adriaensen wrote:


Spending some money on Protools is not really different to doing
the same for a kilometer of microphone cable or some XLR plugs.


imnsho, this simile only holds if you intend to hang yourself with the 
microphone cable.


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Re: [LAD] [OT] IR: LV2 Convolution Reverb

[Jörn Nettingsmeier]

On 02/22/2011 10:12 PM, David Robillard wrote:

As far as I am concerned, this is all about Libre audio software anyway,
and I disagree with the name of this list/site (who actually cares about
the specific kernel?). Getting e.g. OSX people on board is a part of
making the LAD 'platorm' a success. If people on proprietary platforms
start using free plugins, and they start using free hosts, eventually
they're using free everything (e.g. a Jack/LV2 based music platform) and
that's when they can switch to Lignux. Otherwise, they simply won't, and
that is obviously not a win for LAD, Linux, Open Source, GNU, Free
Software, or whatever label you prefer to rally behind.


agreed.


Maybe you don't care. Fine. You're obviously not the person to be
designing our plugin API, then.

Old persnickety grey-bearded UNIX administrators aren't exactly a
significant or compelling market for music software. Perhaps for you and
me, using Lignux is a given, and doing music stuff is something you may
want to tinker with. For the overwhelmingly vast majority of people who
use music software, it is the other way around.

Put simply:

"I don't care about portability" == "Nobody cares about my software".


compelling argument, but not totally true. i'm not really disagreeing 
with your earlier statements, but i think there are some interesting 
aspects to the old greybearded unix wizard approach that fons apparently 
stands for.


here's a bunch of software that uses static, totally non-cross-platform 
makefiles that won't work out-of-the-box on 90% of all architectures.

but they are dead easy to fix.
it uses a custom x11 toolkit, custom thread library wrapper, and other 
idiosyncrasies. but it doesn't depend on sixteen other packages. which 
actually makes the stuff quite portable to osx, if you are willing to 
run x11 on top of it, without going through dependency hell.
it has one heck of a large userbase, and some parts are considered 
reference implementations in their respective fields.

it also tends to just work.

i guess the argument is grand-unified-abstraction-meta-api vs. 
potentially limited but _focused_ software.
you can use a rack full of kickass midi gear with crossbars, mappers, 
generic controllers, whatnot. or you can have a hot soldering iron at 
the ready on top of your organ at all times and just rewire it as needed :)


the former approach will impose fewer limitations. but the latter allows 
you to make some noise right now.

both are very valid imho.




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Re: [LAD] [ANN] IR: LV2 Convolution Reverb

[Jörn Nettingsmeier]

On 02/22/2011 01:45 PM, Nick Copeland wrote:

 > ATM it doesn't even provide network transparency. Which means you can't
 > even do the equivalent of ssh -X.

Does anybody even use this feature anymore?



fwiw, 50% of my audio work happens on a laptop that i use to ssh into my 
audio workstation. (i find a laptop with wlan to be the tranzport as god 
intended it to be :-D).


about 80% of the unix systems administration i have done in the past 
happened over ssh, and i always had xforwarding enabled to be able to 
quickly start xosview or other metering tools.


x bashing is all very cool, but it's what we have and it works.

i wonder:  you are obviously willing to design for future graphical 
abstraction layers which are not yet available. good, but possibly a lot 
of extra (guess-)work. will it be that much worse to just design for x11 
today, and invest some extra work to port to future graphics layers in a 
few years? well-designed software should be easy to port to new 
graphical paradigms, and being lazy today prevents over-engineering.
heck, if your software kicks ass, chances are other people will do the 
porting ;)


fons is obviously being grumpy, but i can't help noticing that we've had 
an awful lot of "innovation" in linux lately that may or may not prove 
useful in the long term, but it definitely did invalidate a lot of 
sysadmin expertise. nobody seems to be factoring that into the equation.
and it's definitely true that most of the innovative replacements to 
traditional unix mechanisms we have seen are focusing on single-user 
desktop or mobile computing. if your usecase differs, you get many 
headaches without much tangible benefit.


so hooray for getting rid of decades of x11 cruft, but let's not throw 
out the baby with the bathwater (even though it's biodegradable).

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