from:"Jörn Nettingsmeier"


On 04/08/2016 08:00 PM, Marc Lavallee wrote:


Finally, a 3D camera with an integrated FOA microphone?
https://www.orah.co/tech-specs/

https://www.orah.co/about/ :
"In addition, the ambisonic 3D sound capture capabilities of Orah
enable the viewer of the content to locate the origin of the sound
source with a VR headset, bringing the feeling of immersion to a
stunning new level."

https://www.orah.co/ :
Looking at the pictures and captions, it says: "Immersive Sound, Four
high dynamic-range microphones for ambisonic 3D sound". The capsules
(or mems?) look soldered on the sides of PC boards, listening through
holes.


very far apart, with body resonances that i don't want to imagine.
hate to be negative about so much exciting new tech, but "stick n>=4 
cheap capsules somewhere and call it ambisonic" seems questionable to 
me. simple truth of the matter is that as long as video techs add mics 
as an afterthought, the results are going to suck, badly.



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Re: [Sursound] Anyone know anything about this?


On 04/08/2016 04:26 PM, Marc Lavallee wrote:

On Fri, 8 Apr 2016 15:13:16 +0200
Jörn Nettingsmeier <netti...@stackingdwarves.net> wrote:



They might have very pragmatic reasons: if they know their equivalent
input noise is at 30 dB SPL and their capsules barf at 120, then
restricting the word length to 96 dB is a perfectly reasonable
decision, given the extremely cramped space and the thermal
challenges inside the sphere. It does not leave much room for error
though, so they better get their analog gains right.


It makes sense. Thanks.

So what 96kHz refers to?


96 kHz is probably due to the marketing department looking for something 
that suggests "hi-resolution"...


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Re: [Sursound] Anyone know anything about this?


On 04/08/2016 02:10 PM, Marc Lavallée wrote:

On Fri, 8 Apr 2016 13:00:20 +0200,
Jörn Nettingsmeier <netti...@stackingdwarves.net> wrote :


On 04/07/2016 08:13 PM, Marc Lavallee wrote:


The FAQ says:
"audio is recorded in 96 KHz/16 bit quality"

I would prefer 48 KHz/24 bit.


let's not discuss matters of taste, but rather keep things scientific.

96/16 = 6
48/24 = 2

so theirs is clearly 3x better than yours!


Of course! Hertz per bit... Or is it bit per hertz?
What's better for marketing? More bits of more hertz?


More per bit per second!

They might have very pragmatic reasons: if they know their equivalent 
input noise is at 30 dB SPL and their capsules barf at 120, then 
restricting the word length to 96 dB is a perfectly reasonable decision, 
given the extremely cramped space and the thermal challenges inside the 
sphere. It does not leave much room for error though, so they better get 
their analog gains right.




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Re: [Sursound] Anyone know anything about this?


On 04/07/2016 08:13 PM, Marc Lavallee wrote:


The FAQ says:
"audio is recorded in 96 KHz/16 bit quality"

I would prefer 48 KHz/24 bit.


let's not discuss matters of taste, but rather keep things scientific.

96/16 = 6
48/24 = 2

so theirs is clearly 3x better than yours!

(i held it during the april foolery, but now it must out :)


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[Sursound] Invitation to present a technical demo / a workshop / a product presentation at the AES International Conference on Headphone Technologies, 2016, Aalborg (DK)

2016-04-06 Thread Jörn Nettingsmeier


Sent on behalf of Dr. Alexander Lindau:
--

Dear Colleagues,

the organizational committee of the International Conference on 
Headphone Technologies invites you to propose demonstrations, workshops, 
or thematically suited product presentations. The conference will focus 
on technologies for headphones with a special emphasis on the emerging 
fields of Mobile Spatial Audio, Personal Assistive Listening, and 
Augmented Reality and will be held from August 24-26 in Aalborg, Denmark 
(http://www.aes.org/conferences/2016/headphones). At the time of writing 
we have received 42 proposals for scientific contributions which are 
currently in the process of being double peer-reviewed. The conference 
program will be complemented by a poster exhibition, a social event and 
a number of distinguished plenary speakers.


For this purpose we can offer a 140m² demo space to be shared by 
multiple parties. Places for presentation of table top demos (i.e. 
including only a Laptop, sound card & Headphones) are free of charge. 
Space for setting up small booths of 4-10 m² may be requested for fees 
between 400-1000U$. Spaces for larger exclusive booths may be made 
available on special request. Demo presenters which do not take part in 
the scientific program will have to register as “Presenters” (Early 
Bird: 380 U$, see 
http://www.aes.org/conferences/2016/headphones/registration.cfm).


We would like to emphasize that the AES Int. Conf. on Headphone 
Technologies is a scientific conference in the first place; therefore, 
it does not host a product fair per se. Proposed product presentations 
will be reviewed with respect to in how far they feature a thematic 
bonus to the scientific program. Written proposals (~ 1 page) should 
include a short description, and a statement about technical 
requirements and be sent to 2016hp_de...@aes.org by May 31, 2016. 
Notifications of acceptance will be emailed by July 15, 2016


A Best Demonstration will be elected based on decision of the conference 
audience. A Best Demo Award including a certificate will be presented at 
the conference closing ceremony.


With best regards

Dr. Alexander Lindau
Paper’s Chair
AES International Conference on Headphone Technologies, Aalborg (DK), 2016

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Re: [Sursound] Decay times and frequency response for control rooms/studios

2016-03-30 Thread Jörn Nettingsmeier


On 03/30/2016 03:49 PM, Augustine Leudar wrote:

Can anyone point me to any kind of standard (BBC/AES etc) for control room
room/equipment responses as a whole transfer function.
  Id like to know what decay times are acceptable. I read somewhere that ,
within reason, how dead or live a room is not the main issue - its more
that the decay times are uniform across frequency spectrum - so you don't
have the bass lingering on 100 ms after the mids have dissipated etc ,
Also in terms of frequency response - the acceptable +/- dB specifications
for a room response cheers,
Gus


EBU Tech 3276 is quite helpful there.



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Re: [Sursound] Flac for FOA or amb files?

2016-03-27 Thread Jörn Nettingsmeier


On 03/26/2016 09:15 PM, Bo-Erik Sandholm wrote:

I will use foa in b-format, as far as i remember ambix is Only Chanel order
different in  for Higher order b-format ..


No. AmbiX has a different channel ordering altogether, and its weighting 
coefficients are different. First-order AmbiX is fundamentally 
incompatible to POA B-format, although trivially converted.


https://en.wikipedia.org/wiki/Ambisonic_data_exchange_formats#Reference_table_of_layouts_and_normalisations

Note that Bruce Wiggins pointed out an inconsistency between the 
weighing coefficients of SN3D (which are taken from Chapman) and the 
formulation of SN3D by Nachbar et al. (which, if combined as-is, will 
yield a result that is off by 1/4pi or, as Franz Zotter helpfully 
pointed out, ~11dB). Not terribly disturbing since it's constant, but 
something that needs to be fixed for internal consistency.



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Re: [Sursound] Static stereo source in rotating soundfield, possible?

2016-03-27 Thread Jörn Nettingsmeier


On 03/26/2016 05:22 PM, Albert Leusink wrote:

Jörn Nettingsmeier <nettings@...> writes:



i don't see why you would want to do that. the effect will be quite
strange... why would any part of the sound mix stay constant wrt head
position?
the effect would be a bit like rotating the music bed in the cinema
every time the camera pans - funny, but certainly irritating.


It would be for off-camera audio (voice over, music etc.) that don't have
any relation to the camera position/rotation.


To quote:

"What are you doing?", asked Minsky.
"I am training a randomly wired neural net to play Tic-tac-toe", 
Sussman replied.

"Why is the net wired randomly?", asked Minsky.
"I do not want it to have any preconceptions of how to play", 
Sussman said.


Minsky then shut his eyes.
"Why do you close your eyes?" Sussman asked his teacher.
"So that the room will be empty."
At that moment, Sussman was enlightened.


Imagine a video with an on-camera actor (dialog), a voice over and a music
track. You would want the on-camera dialog to match the video position (so
counter-rotate) , but the VO and music track will not rotate.


I still don't think that makes sense. Certainly it will be detrimental 
to the proper externalisation of the VR content.


If you want to create two disjunct acoustic spaces, do it with _space_, 
i.e. (lack of) reverb. A close-miked voice-over will be perceptually 
clearly separated from any surround location recording at all times.


If you really want to freak out your users, you could route VO and music 
to W only - the result will be a source inside your head. But my guess 
is your brain will just flip you a birdie at that point and hitherto 
refuse to externalise anything you throw at it, not just the voice-over.



if you absolutely have to do it, the only way is to deliver two streams,
one head-tracked and counter-rotated, the other not. which means you'd
have to have control over the listener's player software.


That's what I was afraid of...so I would need 6 channels instead of 4.


If and only if you find it actually benefits your production.


the only way to get two rotationally invariant signals into the stream
is a cardioid pointing up and another one pointing down. if your player
ignores head tilt, the result is like summing to mono and mixing into W.
if it supports head tilt, the result is likely even worse :-


Would that be the same as rotating the encoded stereo stream (set to 0º
spread)  by 90º vertically?


That was not meant to be an actual solution, I was merely stating that 
with this technique, you could insert two signals into your B-format 
stream that would be rotationally invariant and could be extracted in 
the player after the head tracking stage. But they would each spill into 
your horizontal signal at -6dB, because there is only really room for 
one extra signal... But if you can do DSP after headtracking, you could 
also do it right and use two extra channels, iff it actually made sense 
to do it from a perceptual point of view. So this whole paragraph is 
hypo³thetical :-D





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Re: [Sursound] Static stereo source in rotating soundfield, possible?

2016-03-26 Thread Jörn Nettingsmeier


On 03/26/2016 05:58 AM, Albert Leusink wrote:

Hello,

Is it possible to have a non-rotating stereo source in the ambisonic
soundfield, while all the other sources rotate?

Let's say I have a stereo music bed in a spherical video that needs to stay
in position, while the other elements (dialog, sfx etc.) respond to rotation.


are you talking about a head-tracked VR movie?

i don't see why you would want to do that. the effect will be quite 
strange... why would any part of the sound mix stay constant wrt head 
position?
the effect would be a bit like rotating the music bed in the cinema 
every time the camera pans - funny, but certainly irritating.


if you absolutely have to do it, the only way is to deliver two streams, 
one head-tracked and counter-rotated, the other not. which means you'd 
have to have control over the listener's player software.


the only way to get two rotationally invariant signals into the stream 
is a cardioid pointing up and another one pointing down. if your player 
ignores head tilt, the result is like summing to mono and mixing into W. 
if it supports head tilt, the result is likely even worse :-D




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Re: [Sursound] Furse-Malham to ACN conversion

2016-03-23 Thread Jörn Nettingsmeier


On 03/23/2016 10:30 PM, Dave Malham wrote:

Hi Martin (and Eric!),
  One very simple thing I would do, before doing anything else, with any
system that's playing, as we say, silly bu..ers, is just to play a well
localisable sound out of each speaker (on its own) in turn and check that
(a) it's coming out of the speaker it should (all connections are correct)
and that it sounds like it's coming from the direction you think it should
(acoustics not too disruptive). If you really want to be picky, stick a
soundfield type mic at the nominal centre point and check correct B format
signals are produced for each speaker location at the same time. Only then
start worrying about decoders, plugin connections and the rest. I once
worked out that in a simple 1st order system driving a cube of speakers,
there are 16 million ways of it going wrong, without counting individual
component failures in amps, etc. Of course, lots of these ways of going
wrong are self cancelling (*both* ends of speaker cable can be connected
wrongly, cancelling out the polarity inversion, for instance) which is a
darn good job otherwise our job would be near impossible. So, checking the
simple things first is a good way to avoid delving around the complex..


in good old steve ballmer tradition: "polarity, polarity, polarity, 
polarity, polarity, polarity..."
if you can't measure it directly, i found it useful to use a stereo 
phantom source between a reference speaker and the speaker under test.



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Re: [Sursound] Furse-Malham to ACN conversion

2016-03-22 Thread Jörn Nettingsmeier


On 03/22/2016 07:49 PM, Martin Dupras wrote:

Today I tried playback sources in third order Ambisonics on a 8+6+1
hemispheric speaker array using Reaper. It didn't quite work as
intended so I'm trying to figure out where I've gone wrong.

I was using the Blue Ripple TOA-Core panner plugin to position the
sound. I understand that Blue Rippler plugins use the Furse-Malham
convention.

The only decoders that I could find to decode to my specific array
(using coefficients that I calculated using the Ambisonics Decoder
Toolkit) were the Ambix Plug-ins and AmbDec.

I tried Ambix first, which I understand uses the ACN ordering
convention. I tried re-ordering the channels based on information that
I found here: https://en.wikipedia.org/wiki/Ambisonic_data_exchange_formats#ACN.
But that didn't really work.

I then tried to run 16 outputs out of Reaper into Jack, and from Jack
into AmbDec, again using my ADT-calculated coefficients. I understand
that AmbDec uses the Furse-Malham convention, so I would have thought
it was compatible with the output of the Blue Rippler plugins. But
again, that didn't really work well at all.

In both cases the sound was coming from seemingly random places, and a
number of positions went practically silent.


To debug erratic panning behaviour, start with first order, verify, and 
work your way up from there. To make sure the error is not in your 
calculated coefficients, try to use a known-good decoding matrix that 
approximates what you have, before feeding in your optimized one.


With Ambdec, weird things can happen if you connect several ins at the 
same time using some graphical client, because the order shown in for 
example qjackctl is lexical, whereas the internal order is different. So 
you will end up with garbled connections. You can feed an ACN signal 
into ambdec succesfully if you choose SN3D input scaling _and_ manually 
connect the inputs correctly.

This wikipedia article
https://en.wikipedia.org/wiki/Ambisonic_data_exchange_formats
has some information on that, and other pitfalls when interfacing 
different formats.



All best,


Jörn



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[Sursound] expressing HRTFs in spherical harmonics

2016-02-25 Thread Jörn Nettingsmeier


On 01/27/2016 01:56 PM, Jörn Nettingsmeier wrote:

On 01/26/2016 11:05 PM, Politis Archontis wrote:

Hi Jorn,

yes that is correct. I think however that the virtual loudspeaker
stage is unnecessary. It is equivalent if you expand the left and
right HRTFs into spherical harmonics and multiply their coefficients
(in the frequency domain) directly with the coefficients of the sound
scene (which in the 1st-order case is the B-format recording). This
is simpler and more elegant I think. Taking the IFFT of each
coefficient of the HRTFs, you end up with an FIR filter that maps the
respective HOA signal to its binaural output, hence as you said it's
always 2*(HOA channels) no matter what. Arbitrary rotations can be
done on the HOA signals before the HOA-to-binaural filters, so
head-tracking is perfectly possible.


Wow. That sounds intriguing, thanks! I'll try to wrap my head around the
SH expression of an HRTF set in the coming months, hopefully with the
help of Rozenn Nicol's book.


Sorry to revive such an old thread, but the AES monograph on binaural 
technology has arrived, and I've begun to study it. Definitely a great 
resource, recommended:


http://www.aes.org/publications/monographs/

Archontis, I'm still trying to understand how to express a set of HRTFS 
as a SH series.
If I understand correctly, all HRTFS for a given ear can be expressed as 
a function on the sphere, but it would be frequency dependent. So we'd 
need an extra degree of freedom there, how does that tie in with 
Ambisonics? One HRTF "balloon" per frequency bin?
Also, how do you express the inter-aural time delay conveniently (which, 
as I've learned from Rozenn Nicol, depends not only on direction, but 
also on frequency)?


Are there papers out there that describe this in detail?

Best,


Jörn



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Re: [Sursound] Inventory of Ambisonics plug-ins/tools on OS X

2016-02-25 Thread Jörn Nettingsmeier


On 02/25/2016 02:17 PM, Courville, Daniel wrote:

I'm making a list of the Ambisonics plug-ins/tools available on OS X.



So far, I have this: am I missing anything obvious?

AAX
SoundField: http://www.tslproducts.com/soundfield/soundfield-surroundzone2/
Harpex: http://harpex.net/download.html

Audio Unit
B2X: http://www.radio.uqam.ca/ambisonic/b2x.html
Harpex: http://harpex.net/download.html
SoundField: http://www.tslproducts.com/soundfield/soundfield-surroundzone2/

VST
AAT: http://www.ironbridge-elt.com/products/aat.html
ambix: http://www.matthiaskronlachner.com/?p=2015
B2X: http://www.radio.uqam.ca/ambisonic/b2x.html
Blue Ripple Sound: http://www.blueripplesound.com/products/toa-core-vst
Harpex: http://harpex.net/download.html
SoundField: http://www.tslproducts.com/soundfield/soundfield-surroundzone2/
VVAudio: http://www.vvaudio.com/downloads
WigWare: http://www.brucewiggins.co.uk/?page_id=78

Reaper
ATK: 
http://www.ambisonictoolkit.net/wiki/tiki-index.php?page=Downloads#JS_FX_plugins_for_Reaper

Max
Graham Wakefield: http://www.grahamwakefield.net/soft/ambi~/
HoaLibrary: http://www.mshparisnord.fr/hoalibrary/
ICST: https://www.zhdk.ch/index.php?id=icst_ambisonicsexternals

Pure Data
HoaLibrary: http://www.mshparisnord.fr/hoalibrary/

Plogue Bidule
Aristotel Digenis: http://www.digenis.co.uk/?page_id=59


i've added AAT to the list on wikipedia, i think the rest is there.

https://en.wikipedia.org/wiki/List_of_Ambisonic_Software


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Re: [Sursound] AmbiExplorer: no sensors on CM13.0 (Android 6.0 Marshmallow)

2016-02-24 Thread Jörn Nettingsmeier


On 01/08/2016 11:25 PM, Jörn Nettingsmeier wrote:

Hi Hector, hi everone,


after a botched mobile phone upgrade (I'm using CM on my Samsung S4 and
update frequently), I was forced to redo the phone from scratch, taking
the opportunity to move to CM13.0, which is based on Android 6.0 aka
Marshmallow. I reinstalled AmbiExplorer from Google Play and had a good
look at all the promising new features that I had somehow neglected to
play with before :)

The only problem is that the sensors have stopped working. I can move
the sound field by dragging the head icon, but it does not react to
phone orientation anymore. Hadn't used it in a month or so, so I can't
be 100% sure it's due to the OS upgrade, but it seems likely.

Other apps can access all relevant sensors (I'm using the Physics
Toolbox Suite by Vieyra), and I don't have any privacy settings that
might interfere with AmbiExplorer. Under Android
Settings/Apps/AmbiExplorer, I see it has the permissions to use Location
services and storage. I wonder if it is missing an extra permission to
access the sensors, but the Vieyra suite doesn't have anything like it
either, so I guess permissions can be ruled out.

Have you had the chance to try AE on 6.0 on a Samsung? I know this is
quite bleeding edge and CM is pretty shaky still, too. So no ill
feelings if there's no immediate fix. Maybe I should just go into
version junkie detox and stick with what works :-D


sorry to revive such an old thread, but i just wanted to remark that 
sensors in AmbiExplorer have been working again in CM13 nightlies since 
the last couple of weeks. back to head tracking fun!



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Re: [Sursound] Ambisonic Decoder Design Resources

2016-02-24 Thread Jörn Nettingsmeier


On 02/24/2016 11:37 AM, /dav/random wrote:

Thanks Archontis for mentioning also my little project!
Since I moved out from Barcelona Media, I'm developing the project in my
freetime and the updated repository changed to:
https://github.com/davrandom/idhoa

I don't want to SPAM more... so if anyone is interested, just drop an email.


can't speak for anyone else here, but personally i would very much like 
to be informed about this project, and it seems perfectly on topic for 
sursound.



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Re: [Sursound] Different usages, different spaces, different decoders?

2016-02-23 Thread Jörn Nettingsmeier


On 02/22/2016 09:28 PM, Martin Leese wrote:

Peter Lennox wrote:


  Following on from discussions of decoder solutions: Forgive me if I've
missed this (I've been watching sursound for about 20 years, or so - but I
just may have missed the odd discussion!)

Has anyone systematically studied the interactions between decoders, speaker
layouts and particular rooms?


Dermot Furlong looked at the last two in the
early 1990s.  He made a lengthy post to
"sursound" in June 1996 describing his work.
This post used to be available in my area on
the Ambisonia.com site, but it seems to have
been deleted.  I still have the files, but am not
sure of the best way for making them available.


I would very much like to have that post. It could be made available on 
the internet somewhere, and if it is still relevant in the light of 
recent decoding techniques, used as a reference on the wikipedia page on 
ambisonic listening rigs.



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Re: [Sursound] Different usages, different spaces, different decoders?

2016-02-22 Thread Jörn Nettingsmeier


On 02/22/2016 11:24 AM, Peter Lennox wrote:

Following on from discussions of decoder solutions: Forgive me if
I've missed this (I've been watching sursound for about 20 years, or
so - but I just may have missed the odd discussion!)

Has anyone systematically studied the interactions between decoders,
speaker layouts and particular rooms?

I ask because, it seems to me that interactions between room
acoustics and speaker positioning are known to have significant
psychoacoustic (and aesthetic) effects. Informally, I've observed
ambisonics working better than it had any right to, in particularly
difficult rooms (big reflective empty shoebox, for example).

But in respect of particular speaker layouts, (as per the discussion
on avoiding too many speaker in the horizontal plane), it seems to me
that there could be non-trivial interactions, so that (for arguments'
sake) a particular room might benefit from 'this' speaker layout as
against 'that' speaker layout.

I would seem a monster task to test a wide variety of rooms each with
a wide variety of speaker layouts (and I haven't even mentioned the
possible variety of speaker dispersion characteristics!) - but in the
long run, it needs doing - and sufficient testing might reveal
'families' of layout-room acoustic relationships that can point to
underlying causal rules.

If it's been done, I'd like to read it, and if it hasn't - sounds
like I've just knocked together a precis proposal for a PhD project!


to me, the single most obnoxious effect is the phasiness. next is 
localisation precision, but a long way down. absolutely nobody cares 
about localisation accuracy, so optimising for minmum angular error 
seems undesirable to me (unless there is a very specific use case).


i've been thinking about modelling the interference patterns to minimize 
phasiness in a systematic way. haven't done that yet, but now that I 
learned how to work with the SFS toolkit, i will look at it.

after setting up many ambi rigs, these are my working hypotheses:

* don't be precise. measure to centimetres, but then add random delay 
errors in for subjectively nicer reproduction at low orders.
this is what needs to be tackled systematically. we need a rigorous 
technique to dither spatial aliasing optimally.
i guess we want the peaks and dips to be smeared out uniformly over a 
large area.


* reverberation is your friend. it smoothes away the phasiness. unless 
your content has subtle reverb which would be drowned in the room response.


* 3D rigs seem to localise worse but sound better for low orders, my 
guess is it's because of smoother interference patterns. their path 
length variation is different to that of horizontal speakers across the 
listening area, which might be helpful. that's why i tend to recommend 
3d rigs over higher-order horizontal-only ones, unless you're sure that 
all your content will be at maximum order. even then, some height is 
nice. at ICSA 2011, i heard a small IOSONO wfs rig being augmented 
rather haphazardly with just four small height speakers (which, to the 
best of my knowledge, were used "to taste" and not in any systematic 
way), and the improvement was absolutely striking, not for localisation, 
but for tone color and plausibility of space.


* i have a hunch that stacked rings, for all their wastefulness, seem to 
have very nice interference patterns. for example, the SPIRAL in 
huddersfield (triple octogon with zenith) is in a ridiculously dry room, 
but i was quite surprised about its first-order performance, especially 
since it has way to many speakers for that, in theory.


if somebody can suggest any measurements that are feasible in the field, 
i will gladly obtain them from any future rig that i get to set up. i 
guess room impulse responses would be the most important piece of the 
puzzle. maybe we should just sweep each speaker into a tetramic in the 
sweet spot as a start. with careful analysis, that should contain a lot 
of information about the speakers and the room. we get free-field 
response above a few hundred hertz, and below the schroeder frequency, 
we're in mode land anyways...


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Re: [Sursound] Wireless Solutions for Binaural Event

2016-02-22 Thread Jörn Nettingsmeier


On 02/21/2016 09:47 PM, Justin Bennett wrote:


The solution used in tunnels, caves, mines etc. is to use a
radiating cable instead of a normal antenna. This is a coax
cable that is designed to 'leak' part of the energy that
passes through it, usually by having some holes in the
shield (a standard coax won't work).


yes, that’s what I meant. There was for a long time a sound art piece
on the Afsluitdijk in the Netherlands by Moniek Toebosch that used
a leaky coax cable all along the dijk. Drivers could tune into the frequency
and listen to Angels.

very local and linear coverage!


thanks for pointing this out. i had heard about this technique before, 
but i was assuming that it works because the cable is like a line 
radiator and it's straight.
i'm no radio guy, so i'm applying acoustic principles here and may be 
totally wrong, but my reasoning is that at uhf, say 600 MHz, you end up 
with a wavelength of about half a meter. that means the distance between 
the leaks would have to be small compared to that. fine. should give you 
a uniform cylindrical field. but what if you follow a u-shaped or zig 
zag path because that's what the building is like, with a distance 
between the legs of the u that are a lot longer than the wavelength?
i would expect pretty fancy interference patters with loads of complete 
nulls all over the place. or is there some sort of near-field effect 
that makes the effect of parallel lengths of wire negligible?




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Re: [Sursound] Wireless Solutions for Binaural Event

2016-02-21 Thread Jörn Nettingsmeier


On 02/21/2016 03:32 AM, Chris Timpson wrote:

Hi all

Wondering if anyone has suggestions for a wireless headphone solution? I'm
working on a live event that will be a 30mins binaural sound experience in
a medieval prison for 24 audience members at a time. We need the audio to
begin simultaneously for all audience members and they will be walking
around between 3 locations. The distances aren't huge but quite a few walls
etc.

I've been looking at silent disco type headphones but have concerns about
the quality and also that the signal apparently is converted to mono then
back to stereo during RF transmission. Anyone tested these?

It could be that we use wired headphones with some kind of small playback
device that can somehow be remotely triggered to play. There will just be a
single audio file that plays from start to finish. Wondering if anyone has
tried to build something similar, or perhaps theres an existing solution
i've completely overlooked !?


I guess the most straightforward approach would be little portable 
players started at the same time. If you splice a test tone and then 
five minutes of silence before the program material, you can have 
attendants start them before handing them to the audience.


Another solution might be IR-based systems as used by interpreters. They 
are quite resilient and easily handle multiple emitters. I don't know 
what their audio bandwidth is, however.



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Re: [Sursound] Wireless Solutions for Binaural Event

2016-02-21 Thread Jörn Nettingsmeier


On 02/21/2016 12:54 PM, Justin Bennett wrote:


thick walls would be a problem though. You could try running antenna wires 
throughout the space?
Don’t know if that would help.


i think that would actually make things a lot worse.
multiple senders interfere. you can only ever have one transmitter 
working on a given frequency. if you wanted to hand over to another, it 
would have to work on a different frequency, and the receivers would 
have to support that kind of feature. I don't know any headphones that 
can do it.


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Re: [Sursound] Ambisonic Decoder Design Resources

2016-02-21 Thread Jörn Nettingsmeier

On 02/20/2016 08:31 PM, David Pickett wrote:

At 19:53 20-02-16, Bo-Erik Sandholm wrote:

 >It has been verified by listening test that for FOA it is optimal to not
 >have too many speakers in the horizontal ring. Look at old mails in the
 >list.

This is a tall order: could you specify approx dates, subject lines, or
keywords to search on?  Alternatively, please repeat the information
here, as it could be of interest.

IIRC, one of the BLaH papers also cites listening tests that have found 
the hexagon to be preferred over square or rectangular setups for first 
order.

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Re: [Sursound] Ambisonic Decoder Design Resources

2016-02-20 Thread Jörn Nettingsmeier


On 02/20/2016 06:22 PM, Richard Graham wrote:

Hi Archontis,

I would like to design decoders for 2d and 3d arrays, 1st through
3rd order (at least), both regular and irregular arrays. C code
examples would be incredibly helpful as I plan to develop decoders
for Pd and Max.


Fons' ambdec is GPL, and it comes with a nice set of example setups.
It's C++, but the way Fons uses it, it reads pretty much like plain C.
After all, a dsp loop is a dsp loop...


Most importantly, I’d like to figure out how to calculate these
coefficients myself and I am having trouble finding literature on
how to do that. I have reached out to a few folks who used their own
programs to calculate coefficients. Essentially, I’d like to build
my own program in the C programming language.


Aaron Heller has a Matlab/Octave toolkit out that will generate matrices
for you, and it's completely open. But it relies on quite complex 
functions of the framework... His solutions are used at CCRMA, to

great effect. Probably your best starting point.

Richard has one but keeps it proprietary, Fons has one but also doesn't
like to part with it (although he has been very generous about
generating custom Ambdec setups for people, me included).

For the nitty-gritty, check out the papers from recent Ambisonics
symposia and the ICSA conferences. Talk to Thomas Musil from Graz for
the old-school, lovingly hand-optimized matrix approach, or to Zotter et
al. for the All-Rad approach that works for arbitrary setups but is
quite complex and kind of brute-forceish. I can dig them up for you if 
you can't find them.



Shortly, I will have access to a 16-channel ring on the horizontal
plane and a b-format cube. This system will be modular and
configurable into irregular setups, too.


nice! but unless you really need extremely high horizontal resolution 
for research purposes or a truly humongous listening area, a better use 
for all those speakers would be to make a more or less uniform 3D rig.

gets you a nice dodecahedron for full third-order all around.


best,


jörn



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[Sursound] Would you like to see Ambisonics at the VDT International Convention 2016?

2016-02-18 Thread Jörn Nettingsmeier


Hi everyone,


following up on the recent thread about (the lack of) Ambisonics at 
upcoming conventions, I'd like to inform you about this year's VDT 
International Convention (aka the "Tonmeistertagung"), to take place in 
Cologne, Germany, Nov. 17-20 2016:


http://www.tonmeister.de/index.php?p=tonmeistertagung/2016

The program is not completely in English, but many of the cutting-edge 
research papers are.


I'm a little bit proud to say that through many years of continuous 
grass-roots efforts, Ambisonics has finally made it onto the radar of 
the European professional audio community, not the least since the VDT 
founded and co-hosted the International Conferences on Spatial Audio, 
which have worked little miracles in bringing the research community and 
the professional audio market in touch.
Consequently, the last VDT convention in 2014 has featured an 
Ambisonics-capable lecture hall for the first time (3h2v only, but it's 
a start, and it was for more than 100 people), and we're looking for an 
excuse to provide one again this year.


So please save the date, and keep your workshop ideas and paper 
presentations coming. It's still a while before the call for papers, but 
I'd welcome some signs on- or off-list about the kind of interest we can 
expect to kick some spherical harmonic butt, at a conference that has 
had a strong reputation for 3D audio topics and drawn thousands of 
highly skilled visitors, 180 exhibitors and more than 200 paper 
presentations in the past.
And the more interest we see, the more resources we can justify to put 
into a nice Ambi rig.


Disclaimer: I'm a member of VDT and work for them as technical director 
of the conference.



Best,


Jörn


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Re: [Sursound] Advice on practicalities of 16-speaker half-spherical arrangement

2016-02-12 Thread Jörn Nettingsmeier


On 02/09/2016 10:51 PM, Martin Dupras wrote:

Thanks for all the responses. Much appreciated.

I'll re-phrase the question in light of some of the answers I've been given.

I will be using third-order Ambisonics. My aim mostly is to experiment
to get a good sense of what is possible with Ambisonics with height. I
have experimented successfully with 8-channel planar Ambisonics some
time ago. My primary intent is to spatialise multiple monophonic
(synthesised) sources using 3rd-order Ambisonics spatialisation, and
the playback of mixed sources (spatialised monphonic and stereophonic
sources as well as B-format 4-channel recordings.)

At this moment in time, I have the opportunity to deploy (next week) a
16-channel array, so I would like some advice on a configuration that
would be a good start to experiment with Ambisonics with height.
Someone suggested that I consult the wikipedia page on Ambisonics.
That is indeed where I got the idea that an "upper hemisphere" setup
might be suitable, since I only have on this occasion 16 speakers.
There is however no suggestion as to what a suitable hemispherical
configuration might be for a 16-speaker array, which is why I asked my
original question.


yeah, that page is pretty much still in the somewhat incomplete stage i 
started it in, we should add to it. my advice would be to just use 15 
speakers in an 8-6-1 configuration if you want to go for a hemisphere, 
and to try a icosahedron (with 12 speakers on the vertices) if you want 
full-sphere, but then only for 2nd order. 3-6-3 can also be nice and 
simpler to set up.
the big advantage of an icosahedron is that most decoders have a 
suitable setup by default. for hemispheres, you'd need to compute it 
yourself.



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Re: [Sursound] Soundfield ST350

2016-02-08 Thread Jörn Nettingsmeier


On 02/08/2016 02:03 PM, Steven Boardman wrote:

Anyone know what compatible Rycote parts can be used with a Soundfield
ST350 / ST450.
I have quite a few Rycote sets and am reluctant to purchase another full
set from soundfield.
They and their distributors have been less than help full in telling me
what parts are the same.  (they use different part numbers than Rycote).
Rycote also can't tell me,  as they only make them for soundfield, and say
I need to contact them!


i don't have the blimp for the st450, but the lyre suspension that came 
with it is red (in the hope that it'c color-coded for weight).
the length of the microphone body including connector and curved cable 
is about 30 cm (see photo sent in private that's probably not going to 
make it to the list).


best,


jörn





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Re: [Sursound] Never do electronic in public.

2016-02-03 Thread Jörn Nettingsmeier


On 02/03/2016 02:48 PM, florian.came...@orf.at wrote:


(But we shouldn't divert from the original topic.)


no, that is frowned upon here :-D
thanks for the insightful comments, i was wondering the same...




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Re: [Sursound] Never do electronic in public.

2016-02-01 Thread Jörn Nettingsmeier


On 02/01/2016 04:25 PM, David McGriffy wrote:


The standard I hear in the VR world is 20ms "motion to photons". Minimum
frame rate of 60Hz, preferably 90-120Hz. These faster rates do not allow
convolution with the full block size of the listen database, though careful
truncation should be OK.


? the latency of a convolver has nothing to do with the length of the 
kernel, only with its own blocksize. with a standard pc, it should be 
easily possible to get under 3 ms latency, even less if you're prepared 
to upsample your hrtfs and content to 96k (and burn twice as much cpu, 
and waste storage). in any case, you can make use of the full length of 
the hrtfs.


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Re: [Sursound] Never do math in public, or my take on explaining B-format to binaural

2016-01-28 Thread Jörn Nettingsmeier


On 01/28/2016 10:12 PM, Bo-Erik Sandholm wrote:

I do understand that HOA can represent resolution of directivity in the
mathematic domain better than FOA.
But I am starting to suspect we are overworking something when we are
talking of order 8 to 15?


talking about using order 15 to retain the fine details of hrtfs doesn't 
mean we have to use 384 loudspeakers. nor will your average smart phone 
break a sweat when dealing with a very-high-order intermittent stage.


but given the care that goes into obtaining precise hrtf sets and the 
sensitivity of listeners towards coloration, it seems appropriate to 
represent hrtfs in a way that retains their precision.



Is it realistic to even think of measuring individual HRTF response with
that angle resolution?


i have a set of custom hrtfs measured by ITA in aachen that (off the top 
of my head, can't check right now) has 3° horizontal and 5° vertical 
resolution. the whole measuring process took less than 7 minutes, and 
i'm told they have refined their procedure a lot since then.



And is it even neccessary when we know the
adaptability of the auditory system?


given that even with my own hrtfs, externalisation is still a problem 
because i can't seem to get the headphone/ear canal equalisation right: 
yes! yes!! yes!!!
that is somewhat orthogonal to localisation precision, but it shows that 
adaptability has its limits: when the illusion breaks down, it breaks 
down completely.



As stereo works good enough over 45 degrees with 2 speakers and correct
psycho acoustic setup and a good recording are we not aiming for a overkill
system?


if you still think about signal processing in terms of circuit boards 
and wires or old-school economic elegance, yes. but if the difference is 
in typing create_HOA_convolver(3) vs. create_HOA_convolver(15) and my 
phone isn't even getting warm, who cares?


i guess it's understood that we're on the very flat end of the 
diminishing returns curve at order 15. but the cost is getting 
negligible, _and_ binaural has this habit of failing completely for some 
people in the presence of very subtle errors.



As a normal guy without training in listening for direction of sound
sources I suspect I cannot really pinpoint many things in more than +-10
degrees without visual cues.


if it was only about directional precision, then i agree, unless we're 
talking augmented reality: if you're flying this fighter airplane, i'm 
sure you'd want the "INCOMING!" warning voice to be positioned a lot 
more precisely than your ability to localize it.



I remember old discussion results about ideal number of loudspeakers for
horizontal FOA replay being 6 speakers.

My goal is to have a device that can play through headphones a stereo or
FOA recording and give me a minimum experience of listening to a stereo
system or FOA setup with out of head sound and a stable position of the
soundstage.

I am not certain this is relevant in this discussion thread as we probably
have different views of the goals and the path to the goals.


well, what do you care about the order in which the hrtfs are 
represented? you feed FOA in, you get binaural out, and unless your 
processor starts burning up, you'll never know that inside the black 
box, some very-high-order stuff might be going on.


if by quadrupling the processing power, i can get a robust 5% 
improvement in the rendering of first-order material, i wouldn't 
hesitate a second, except in very special cases.


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Re: [Sursound] Never do math in public, or my take on explaining B-format to binaural

2016-01-27 Thread Jörn Nettingsmeier


On 01/27/2016 10:02 AM, florian.came...@orf.at wrote:

Hello,

may I point you to the AES Monograph on Binaural Technology by Rozenn Nicol,
published on 2010. Rozenn has nicely summarised most of the issues which have 
been discussed
here lately, and she provides an extensive list of references (more than 200!). 
Well worth reading
(35$ for AES members).


i've been eyeing this one for a while (it's advertised in every new 
issue of the AES journal...), but your recommendation finally made me 
order it. thanks!



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Re: [Sursound] Never do math in public, or my take on explaining B-format to binaural

2016-01-27 Thread Jörn Nettingsmeier


On 01/26/2016 11:05 PM, Politis Archontis wrote:

Hi Jorn,

yes that is correct. I think however that the virtual loudspeaker
stage is unnecessary. It is equivalent if you expand the left and
right HRTFs into spherical harmonics and multiply their coefficients
(in the frequency domain) directly with the coefficients of the sound
scene (which in the 1st-order case is the B-format recording). This
is simpler and more elegant I think. Taking the IFFT of each
coefficient of the HRTFs, you end up with an FIR filter that maps the
respective HOA signal to its binaural output, hence as you said it's
always 2*(HOA channels) no matter what. Arbitrary rotations can be
done on the HOA signals before the HOA-to-binaural filters, so
head-tracking is perfectly possible.


Wow. That sounds intriguing, thanks! I'll try to wrap my head around the 
SH expression of an HRTF set in the coming months, hopefully with the 
help of Rozenn Nicol's book.


Meanwhile, forgive the man without the heavy ion accelerator that this 
problem did indeed look like a nail :-D



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[Sursound] Never do math in public, or my take on explaining B-format to binaural

2016-01-26 Thread Jörn Nettingsmeier


On 01/26/2016 06:36 PM, Stefan Schreiber wrote:


2. < 8 > impulses (for 4 virtual speakers) implies that you don't
support 3D decoders (?). If not, why this? (Immersive/3D audio is on the
requirement list for VR. It wouldn't make a lot of sense if all sound
sources will follow your gaze - looking upwards or downwards.)


I think the 8 impulses are used differently. I'm scared of trying to 
explain something of which my own understanding is somewhat hazy, but 
here it goes: please correct me ruthlessly. Even if in the end I wish 
I'd never been born, there might be something to learn from the 
resulting discussion :)


W goes to loudspeaker LS1, LS2, ..., LSn.
Same for X, Y, and Z.

Each LSn then goes both to left ear and right ear.

So you start with a 4 to n matrix, feeding into an n to 2 matrix. The 
component-to-speaker convolutions and the speaker-to-ear convolutions 
(the HRTFs) are constant.


Convolution and mixing are both linear, time-invariant operations. That 
means they can be performed in any order and the result will be 
identical. I guess in math terms they are transitive and associative, so 
that (a # X) + (b # X) is the same as (a + b) # X, and a # b # c is the 
same as a # (b # c), where "#" means convolution.


So the convolution steps can be pre-computed as follows, where DEC(N,m) 
is the decoding coefficient of component N to loudspeaker m, expressed 
as convolution with a dirac pulse of the appropriate value:


L = W # DEC(W,LS1) # HRTF(L,LS1) + ... + W # DEC(W,LSn) # HRTF(L,LSn)
  + X # DEC(X,LS1) # HRTF(L,LS1) + ... + X # DEC(X,LSn) # HRTF(L,LSn)
  + Y # ...
  + Z # ...

(same for R)

which can be expressed as

L = W # ( (DEC(W,LS1) # HRTF(L,LS1) + ... + DEC(W,LSn) # HRTF(L,LSn) )
  + X # ...
  + Y # ...
  + Z # ...

(same for R).

Note that everything in brackets is now constant and can be folded into 
a single convolution kernel.


That means you can, for first order, reduce the problem to 8 
convolutions, going from {WXYZ} to {LR} directly. The complexity is 
constant no matter how many virtual loudspeakers you use.


Of course, that does not take into account dual-band decoding. But if we 
express the cross-over filters as another convolution and split the 
decoding matrix into a hf and lf part, we can also throw both halves of 
the decoder together and do everything in one go.


For nth order, you have (n-1)² * 2 convolutions to handle.

For head-tracking, the virtual loudspeakers would move with the head (so 
that we don't have to swap HRTFs), and the Ambisonic signal would be 
counter-rotated accordingly. Of course that gets the torso reflections 
slightly wrong as it assumes the whole upper body moves, rather than 
just the neck, but I guess it's a start.




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[Sursound] Submissions now accepted for 2016 AES International Conference on Headphone Technology

2016-01-17 Thread Jörn Nettingsmeier


[sent on behalf of Dr Alexander Lindau]

Dear colleagues!

(sorry for cross-posting)

As Papers Co-chairs for the 2016 AES International Conference on
Headphone Technology in Aalborg (DK), this summer, we, Alexander Lindau
and Jürgen Peissig, are pleased to announce that we are now accepting
proposals for full technical papers, poster presentations and technical
demos for our conference which is to be held in Aalborg (DK) this summer.

For more information and author's kits, please visit the web site

http://www.aes.org/conferences/2016/headphones/

and navigate to the 'Paper Submission' subpage.

*Conference Topics*(non-exlcusive)

*Headphone Design*

Headphone Transducers: Technology, Measurement,

Microdrivers, Smart Control, Simulation

Personalization

Additional Sensors

Wearing and Listening Comfort

*Applications for Mobile Audio*

Augmented Reality

Mobile Spatial Audio

Personal and Assistive Listening

Binaural Techniques

Up-Mixing

Noise Control

Monitoring and Analytic Listening

*Evaluation*

Headphone Quality

Quality Standards

Automatic Quality Evaluation

Perceptual Audio Evaluation

Perceptual Targets

We invite the submission of full papers between 4 and 8 pages by March
16, 2016.  All submitted papers will be double-peer-reviewed before
selection. The conference review committee will decide which papers are
accepted, and authors will be informed of the decision by May 16, 2016.
After revision, final versions of papers must be submitted by July 15, 2016.

Additionally, the committee invites interested parties to propose
demonstrations, workshops, or thematically suited product presentations.
Email proposals should be sent to 2016hp_de...@aes.org
<mailto:2016hp_de...@aes.org> by May 31, 2016.  Acceptance notifications
for demonstrations will be emailed by June 30, 2016.

*Awards*

A scientific jury will decide about Best Paper Award. Awards for Best
Poster and Best Demonstration will be decided about with the help of all
attendees and presented at the conference.

*Dissemination *

No later than 1 month after the conference the proceedings will be made
permanently available online from the AES Electronic Library to
subscribers. Further on, AES has a pay-to-publish policy in Open Access.
If an author wants to make his/her paper openly available to everyone,
this is possible by paying a specific Open Access fee. Please see
details in http://www.aes.org/openaccess/

Finally, we ask you to share this information with interested colleagues
and are hoping to welcome you in friendly Aalborg this summer.

Sincerely,

Alexander Lindau - Max-Planck Institute for Empirical Aesthetics,
Frankfurt/Main

Jürgen Peissig - Leibnitz University, Hannover

Papers Co-Chairs

2016hp_pap...@aes.org <mailto:2016hp_pap...@aes.org>




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Re: [Sursound] YouTube adds ambisonics support

2016-01-15 Thread Jörn Nettingsmeier


On 01/15/2016 10:00 PM, Martin Leese wrote:

Andres Cabrera wrote:

Very interesting.

I'm wondering if it's worth considering separating the order for horizontal
vs. vertical (instead of a single unified order).


This mixed order scheme, specifying #H and
#P, has the disadvantage that as a source
leaves the horizontal, its sharpness degrades
rapidly to that of the height-order.  An
alternative scheme, which does not have this
problem, is "Complete mixed-order sets".  This
would also require two numbers to be specified,
#H and #V, and is described at:
https://en.wikipedia.org/wiki/Mixed-order_Ambisonics#Complete_mixed-order_sets_.28.23H.23V.29

However, I don't know of anybody who has
experience with decoding such sets.


Fons's decoder can handle that, and iirc it comes with examples. This 
scheme is particularly useful for stacked rings.



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[Sursound] AmbiExplorer: no sensors on CM13.0 (Android 6.0 Marshmallow)

2016-01-08 Thread Jörn Nettingsmeier


Hi Hector, hi everone,


after a botched mobile phone upgrade (I'm using CM on my Samsung S4 and 
update frequently), I was forced to redo the phone from scratch, taking 
the opportunity to move to CM13.0, which is based on Android 6.0 aka 
Marshmallow. I reinstalled AmbiExplorer from Google Play and had a good 
look at all the promising new features that I had somehow neglected to 
play with before :)


The only problem is that the sensors have stopped working. I can move 
the sound field by dragging the head icon, but it does not react to 
phone orientation anymore. Hadn't used it in a month or so, so I can't 
be 100% sure it's due to the OS upgrade, but it seems likely.


Other apps can access all relevant sensors (I'm using the Physics 
Toolbox Suite by Vieyra), and I don't have any privacy settings that 
might interfere with AmbiExplorer. Under Android 
Settings/Apps/AmbiExplorer, I see it has the permissions to use Location 
services and storage. I wonder if it is missing an extra permission to 
access the sensors, but the Vieyra suite doesn't have anything like it 
either, so I guess permissions can be ruled out.


Have you had the chance to try AE on 6.0 on a Samsung? I know this is 
quite bleeding edge and CM is pretty shaky still, too. So no ill 
feelings if there's no immediate fix. Maybe I should just go into 
version junkie detox and stick with what works :-D



All best,


Jörn




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Re: [Sursound] Sennheiser Easy 3D Recording and Modeling

2016-01-06 Thread Jörn Nettingsmeier


On 01/06/2016 06:36 PM, Courville, Daniel wrote:

« Sennheiser realizes that content is king and that for any new
technology to gain traction, it must be easy to develop in.
Sennheiser is therefore looking to help content creators take
advantage of their 3D audio platform with easy-to-use recording and
modeling tools.

On the capture front, Sennheiser will feature a virtual-reality
microphone. Unlike traditional microphone designs, the new mic will
capture high-quality audio in four quadrants. Sennheiser says the mic
was designed in coordination with VR content producers. The VR mic is
scheduled to launch in the third quarter of 2016. In 2017, Sennheiser
will ship a software plug-in that will be bundled with the same mic
for VR content post-production. »

From
http://www.techhive.com/article/3019706/home-audio/sennheiser-launches-new-flagship-headphones-the-hd-800-s-and-moves-into-the-3d-audio-space-with-amb.html



Too bad they kicked the Eigenmike guys out :-D
Those who do not understand spherical harmonics have to reinvent it...

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Re: [Sursound] The Mike Skeet Collection

2015-12-20 Thread Jörn Nettingsmeier


On 12/20/2015 12:00 PM, Richard G Elen wrote:


I wonder if any of you can think of anyone who might be able to give
this landmark collection a home, or if you can pass this information on
to someone who might have thoughts on the subject.


How much of it is digitized? I would be willing to host any digital 
material provided that the copyright situation is clear. I could also 
propose to the VDT to designate some webspace. If the legal heirs are 
willing to release this under some reasonably open license (CC-BY?), it 
might be a great resource.
If digitizing is required, I know people who could do it, but then the 
question is who's going to dig through the archives and ship the stuff?


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Re: [Sursound] Wireless speakers

2015-12-17 Thread Jörn Nettingsmeier

On 12/17/2015 01:27 PM, Augustine Leudar wrote:

I may have posted it before but maybe things have changed. I do a lot of
multichannel site specific work sometimes I use speakers up to a hundred
metres away from the control center. I currently use Mackie SRM450s and
miles and miles of xlr cable which is inconvenient to say the least (think
cherry pickers and risk assessments that resemble Phd theses etc etc) . Is
there currently any wireless pa or even smaller speakers that can work
wireless (I would love to go wireless on both small and large
isntallations) . I know I can use seinheizer headphone units but I was
wondering if there was a speaker with build in wireless that is relatively
compression free (ie not bluetooth!) plus 32 Seinheizer units plus speakers
is too costly at the moment. All the wirless speakers Ive seen (eg Alto)
only allow stereo - there does not seem to be a system that can deal with
multiple discrete channels. Jorn did you mention such a system ? Anyone
else care to chime in ?
I look forward to the day when I hear the question "Grandad - what's an
audio cable ?"

oh yes :)

when you say "multichannel", and at the same time mention such huge
distances, it would be interesting to learn whether you need fixed phase
relationships between the speakers (as in an ambisonic or stereophonic
playback situation), or have singular sound objects that do not need to
maintain strict timing with respect to their neighbours.

if it's mostly singular speakers each doing their own stuff, i guess
wifi streaming as suggested by marc would be a good option. although
i've never tried multicast outside the lab, and certainly never over
wifi. maybe it doesn't even have to be multicast then, just synchronized
streams on the sender side, and keep the receiver buffers within the
maximum allowable time difference between streams. icecast in
combination with some jack stream source (ezstream or ices2) should do
the job.

when you want to get fixed-phase playback over several receivers, things
get difficult. but maybe just one receiver for each group of speakers
requiring fixed-phase sync would help already?

i've just begun playing with the new raspberry pi 2 in combination with
an edimax 7711-UW wireless usb dongle (each one will cost you less than
50€ including psu, case, and sd card).

there's the problem that the onboard sound is crap and there's no usb
2.0, so i'm using the HDMI audio out (eight channels) in combination
with something like
https://www.ligawo.eu/ligawo-6518770-hdmi-audio-extractor-dac-7-1-5-1-2-0-analog-audio/a-6518770/
haven't measured this thing yet (i'm not using it for anything
critical), and it sure isn't great but usable, and for the price who'd
complain.

one of those would enable you to run third-order horizontal ambisonics
over wireless without jumping through fiery hoops.

all best,

jörn

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Re: [Sursound] vertical precendence and summing localisation (wallis and lee 2015)

2015-12-15 Thread Jörn Nettingsmeier


On 12/15/2015 03:32 AM, Stefan Schreiber wrote:


In our discussion before we have found convincing evidence and arguments
that head motion should be relevant even to obtain improved vertical
localization.


Just to set the record straight again: evidence is not to be found on 
sursound. Evidence is found in the lab. :-]


Hat-tip to those who actually do the grunt work that forms the basis of 
sursound sermons.



If sound sources are immovable, their positions can't be determined
precisely, because the brain needs them moving (movement of the source
or subconscious micro-movements in the listener's head), which helps
to determine a sound source position in the geometrical space.


(?!)


Why quote such questionable statements?


Modern systems of reproduction of positioned 3D sound utilize HRTF
functions forming virtual sound sources, but these synthetic virtual
sources are spot. In the real life the sound mostly comes from large
sources or composite ones which can consist of several individual
sound generators. Large and composite sound sources allow for more
realistic effects in comparison with spot sources.
A spot source can be successfully applied to large but distant
objects, for example, a moving train. But in the real life when the
train is approaching the listener it's no more a spot source.


(See


One of our postgrads (Dan Peterson
<https://dxarts.washington.edu/people/daniel-peterson>) has been
working on
a doppler-panner that includes diffusion filtering and the proximity
effect.

)


These two are orthogonal. The first quote talks about sources being 
physically spread out (e.g. composed of multiple point sources along a 
line or area), while the second talks about what happens if a point 
source approaches the listener.



The third group consists of the sound tone parameters. This can help
the player define what the walls are made of, what is the air density
in the environment etc. Every material reflects and absorbs certain
frequencies. These parameters emulate such absorption and reflection.
They are relative frequencies (LF - Low Frequency and HF - High
Frequency) within which changes can be made. For example, metallic
walls reflect more frequencies than wooden ones, and the HF level will
be lower for them than for emulation of wood. For example, the
workshop has the following parameters: 362Hz LF and 3762 Hz HF; a
wooden room has the LF at 99 Hz and the HF at 4900 Hz. Finally, there
are parameters controlling the effect of Room LF and HF frequencies
(in dB). This subgroup also contains  Decay factor for LF and HF, and
Air Absorption HF factors.


This is games design, not acoustics.
Done properly, it should look something like http://www.audioborn.com/. 
A new company spun off from a research effort at ITA/RWTH Aachen, and 
their demo at ICSA 2015 was mighty sweet.



It is a safe bet that specifically AR/VR will require a solid
understanding of acoustics and human audio perception. They will have to
find improved ways to reproduce surround sound (including 3D audio) via
headphones and loudspeakers.


Thanks for pointing this out. :-]

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Re: [Sursound] vertical precendence and summing localisation (wallis and lee 2015)

2015-12-10 Thread Jörn Nettingsmeier


On 12/10/2015 04:59 PM, Peter Lennox wrote:

It does imply that an ambisonic panner plugin that incorporates spectral 
manipulation would be more efficacious


noo!

if it's an ambisonic panner, it doesn't change the spectrum. if it 
changes the spectrum, it's not an ambisonic panner :)



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Re: [Sursound] vertical precendence and summing localisation (wallis and lee 2015)

2015-12-10 Thread Jörn Nettingsmeier


On 12/09/2015 03:00 PM, Jörn Nettingsmeier wrote:


i've attached the paper, since it is open access.


well, i meant to, but apparently the attachment got eaten. here it is:
http://www.aes.org/e-lib/browse.cfm?elib=18040




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[Sursound] vertical precendence and summing localisation (wallis and lee 2015)

2015-12-09 Thread Jörn Nettingsmeier

On 12/08/2015 09:07 PM, Peter Lennox wrote:

no -percedence effects include a range of phenomena. But precedence
in the median plane isn't quite as effective as in the azimuthal
plane, according to Litovsky, Rakerd, Hartmann et al, but is still
quite effective and so not negligible. So I'd like to understand what
Lee (Huddersfield) was saying, to compare.

i've attached the paper, since it is open access.
i guess i misrepresented it a bit, because i was being sloppy about
distinguishing between precedence effect and summing localisation.

however, wallis and lee conclude:

"Additionally, no evidence could be found to support
the operation of the precedence effect in median plane
stereophony. In the present study the only occasions
whereby stimuli were localized at the position of the ear-
lier emitting loudspeaker were due to the pitch height ef-
fect. There was also no consistent effect of time panning
observed, with localization judgments for the broadband
source becoming more biased towards the upper loud-
speaker as ICTD increased, as opposed to the lower."

[the upper speaker was always lagging behind the lower in this experiment.]

in comparing the results with litovsky et al, it should be pointed out
that while both were conducted under anechoic conditions, the stimuli
used by wallis and lee were long noise snippets with 1s fade-ins and
fade-outs rather than clicks, with no transient information at all
(which seem designed to test the presence of summing localisation), so i
guess they are not in direct contradiction.

it just shows that the musical reality will be somewhere in between...

Certainly, in respect of producing phantom imagery in the vertical,
I've found this to be quite effective (though often slightly more
vague than in horizontal) which would explain why periphonic
ambisonics works at all - and this seems to be a related issue to the
precedence one

i found that vbap/stereophonic vertical localisation is excellent on
speaker positions (because it gets the spectral cues right), and
unusable anywhere else.
3rd-order ambisonic vertical localisation seems uniformly so-so
throughout the elevation range, which to me is preferrable...

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Re: [Sursound] OZO? vertical precedence

2015-12-08 Thread Jörn Nettingsmeier


On 12/08/2015 01:47 PM, Peter Lennox wrote:

Couldn't find the full paper again - but there's this one in full: 
https://www.pa.msu.edu/acoustics/litovsky.pdf

The abstract ends "...models that attribute the precedecence effect entirely to 
processes that involve binaural differences are no longer viable"

The researchers are known as excellent contributors to the corpus of 
psychophysics (Ruth Litovsky did the defninitive review of precedence effects).

So I would be interested to examine the differences in their findings and 
Huddersfield's


thanks, very interesting! a quick glance makes me very curious, i'm 
looking forward to reading this tonight.



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Re: [Sursound] OZO? vertical precedence

2015-12-08 Thread Jörn Nettingsmeier


On 12/07/2015 02:31 PM, Peter Lennox wrote:

But see: Localization dominance in the median-sagittal plane: Effect
of stimulus duration Roberto M. Dizon and Ruth Y. Litovsky Received
19 June 2003; accepted for publication 22 March 2004


interesting!

i wonder:


Lead-lag pairs of   noise
bursts   were   presented   from   locations   spaced   in   15°
increments   in   the   frontal, median-sagittal plane, with a 2-ms
delay in their onsets, for source durations of 1, 10, 25, and 50-ms.


does this mean they used the same noise source, where one channel was 
actually delayed, or the same noise source and one channel was just 
faded up later ("onset delay" could be read this way), or different 
noise sources altogether?


in order to investigate "phantom source" mechanisms, it should be the 
same noise source, delayed, which is likely what they did, but i can't 
check this paper unfortunately.



Intermixed  with  these  trials  were  single-speaker  trials,  in
which  lead  and  lag  were  summed  and presented from one speaker.



Listeners identified the speaker that was nearest to the perceived
source location.


so this is a simple "either/or" decision, not a continuum of possible 
phantom source locations. or put differently: not summing localisation, 
but something like a precendence effect. ok.


i could hypothesize that the initial phase of 2ms from one speaker only 
is enough information to localize the source, and that the lagging 
signal is not contributing any more cues. if so, that would not really 
contradict lee et al.


they go on to say


With   single-speaker   stimuli,   localization
improves   as   signal   duration   is   increased.


the single speaker case is not relevant to the discussion really 
(although it's a nice touch to add this to the experiment). it just 
means that if get more time to pinpoint a single source, localisation 
performance improves. very well.


but this could be read as implying "in two speaker stimuli, there was 
_no_ improvement of localisation as the signal duration is increased". 
which seems to suggest that indeed, the localisation process is over and 
done with during the initial 2ms of only a single speaker playing.


to test this, one would need to use a coherent signal in both speakers 
that starts at the same time, but one is delayed relative to the other. 
maybe by delaying a noise source and fading it in at the same time in 
both speakers. otherwise, we're really only looking at onset transients.


> Furthermore,

evidence of elevation compression was found with a dependence on
duration. With lead-lag pairs, localization dominance occurs in the
median plane, and becomes more robust with increased signal duration.


this general statement would contradict my interpretation above. is this 
paper available somewhere?


this one however leaves me scratching my head:


These results suggest that accurate localization of a co-located
lead-lag pair is necessary for localization dominance to occur when
the lag is spatially separated from the lead.


i can't imagine what this means.



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Re: [Sursound] OZO?

2015-12-07 Thread Jörn Nettingsmeier


On 12/07/2015 02:55 AM, Stefan Schreiber wrote:


Your plots refer to an empty sphere, don't they?


no, fons' plots are about a rigid sphere with omni mikes in the surface.


Take for example the 8kHz response, which is the most directive one in
the set. It's something like 3.5 dB
down at 90 degrees. Apart from the narrow peak at the
back, that's subcardioid.


See above. We currently don't know the exact frequency responses...


the point is that you are dealing with omni microphones, and the only 
directivity that you can work with other than matrixing several capsules 
is the acoustic shadow of the sphere.




And? This is exactly the case here. (IRT cross in two dimensions and the
8-channel hedgehog in the described cuboid form are isotropic. All 12
microphone stereo angles are 90º. A cube is a Platonic solid, "as
everybody knows".)


stefan, i think you are confused about the fact that the hedgehog or any 
other open microphone arrays employ directional microphones. the OZO 
cannot, because it is a rigid sphere with no way for a second acoustic 
path to each capsule.



The 'hedgehog' doesn't produce anything like that, nor
is it meant to.


It does and is meant to!


this is not the case. the eight-supercards hedgehog is not meant to be 
rotated ever. it has a 1:1 mapping from mics to speakers. rotating it 
will give massive shifts both in source width and timbre.
the design goal of Theile's arrays is to minimize crosstalk and ensure 
well-defined decorrelation between speaker channels, not isotropy. it is 
intended to produce ambience only.



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Re: [Sursound] OZO?

2015-12-07 Thread Jörn Nettingsmeier


On 12/07/2015 12:18 AM, Steven Boardman wrote:

Just a little point here. I thought the amount of movement ones head can
cover vertically is more than it can cover in any other plane. This is
obviously not including moving the rest of the body at the same time. This
being so, and generally having the ground for reflections, it allows us to
work out any confusion. Especially coupled with a slight tilt.
Surely hearing what is below is very important, probably more than above,
and as such there must be an important mechanism to determine it.


the natural vertical motion is to tilt one's neck. this does not change 
the orientation of the ear spacing, only the angle of pinnae and the 
direction of the torso/shoulder reflections.


the discussion was about inter-channel time differences in vertically 
spaced speakers not resulting in inter-aural time difference cues at the 
ears, unless you bend your neck to the sides and then up/down. this is a 
very unnatural movement. i do use it from time to time to check for 
errors in complex loudspeaker systems, but it usually results in 
bystanders asking if i'm ok.






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Re: [Sursound] OZO?


On 12/05/2015 05:26 PM, Stefan Schreiber wrote:


I wrote: "8-channel ... hedgehog", which is/was already some form of
educated guess.

See:

http://www.hauptmikrofon.de/HW/TMT2012_3DNaturalRecording_Theile_Wittek_2012_11.pdf,

pg. 19.

This hedgehog layout really fits to the microphone openings of the Ozo
camera...


btw, since you're quoting this very interesting article, it has been 
partly superseded by recent research of lee at al. at huddersfield (see 
latest JAES), who found that there is _no_ vertical precendence effect 
and that interchannel time differences in vertically spaced loudspeakers 
do not contribute to localisation in any way. helmut is aware of this 
and has presented a much more compact 8-channel mic array at ICSA 2015 
in graz, where the top and bottom mics are practically coincident.



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Re: [Sursound] OZO?


On 12/05/2015 07:06 PM, Stefan Schreiber wrote:

Fons Adriaensen wrote:


On Sat, Dec 05, 2015 at 04:26:05PM +, Stefan Schreiber wrote:




See:

http://www.hauptmikrofon.de/HW/TMT2012_3DNaturalRecording_Theile_Wittek_2012_11.pdf,

pg. 19.

This hedgehog layout really fits to the microphone openings of the
Ozo camera...



* The hedgehog is at least twenty times as big.



Hardly...


Theile's and Wittek's arrays rely on directional microphones. It is not 
practically possible to achieve directionality from capsules 
flush-mounted into a sphere, because the rear acoustic path that effects 
the cancelleation is missing. So any directional effect is either due to 
matrixing à la Eigenmike or due to baffle effects as described by Fons.


Now that I see a hi-resolution picture for the first time, I can't say 
I'm much excited by the audio capabilities this thing can possibly have. 
For that price, let's hope they see the light and partner with mh or 
Duraiswami and deliver something that really kicks butt in the audio 
domain...


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Re: [Sursound] OZO?


On 12/06/2015 07:44 PM, Kees de Visser wrote:

On 6 Dec 2015, at 11:46, Jörn Nettingsmeier wrote:


btw, since you're quoting this very interesting article, it has
been partly superseded by recent research of lee at al. at
huddersfield (see latest JAES), who found that there is _no_
vertical precendence effect and that interchannel time differences
in vertically spaced loudspeakers do not contribute to localisation
in any way. helmut is aware of this and has presented a much more
compact 8-channel mic array at ICSA 2015 in graz, where the top and
bottom mics are practically coincident.


Perhaps I've missed it but no-one seems to have mentioned head
movements.


head movements are certainly important, but mostly for horizontal 
lateral localisation. you would have to stretch your neck to very odd 
angles in order to make use of vertical ICTD, and at the same time lose 
horizontal ITD (although the latter might not be a problem since 
localisation perception seems to be constant once a suitable set of cues 
has been "collected", as long as no contradictory cues show up).




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Re: [Sursound] OZO?


On 12/06/2015 06:14 PM, Stefan Schreiber wrote:

Jörn Nettingsmeier wrote:



btw, since you're quoting this very interesting article, it has been
partly superseded by recent research of lee at al. at huddersfield
(see latest JAES), who found that there is _no_ vertical precendence
effect and that interchannel time differences in vertically spaced
loudspeakers do not contribute to localisation in any way.


? This is supposed to be new? Methinks there can be only ILD and
spectrum cues to enable height perception (but no ITD cues), because the
ears are positioned in the same plane.
No vertical precedence: Could be very much related to this simple
observation. (I guess IC cues will also not matter.)


well, obvious in theory, but nobody in the 3d audio world seemed to be 
aware of it, and now lee has done the deed and pulled the rug from under 
all those vertically spaced arrays...
the "new" part is that somebody actually got around to disprove it, in a 
rigorous listening test.



Do I miss s.th.?


no. but there is great value in moving something from the "everybody 
knows that..." into the "rigorous listening tests have unambiguously 
proven that..." realm.



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Re: [Sursound] OT: Opportunities for Study and Funding at the University of Birmingham / BEAST

2015-11-21 Thread Jörn Nettingsmeier


On 11/21/2015 11:15 AM, Dave Malham wrote:

Not quite sure how we got from defining acousmatic music to film sound,
but...


For the record, we got here from Scott announcing study opportunities (a 
hazy after-image of the original intention of this thread is still 
visible in the subject line).


:-D



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Re: [Sursound] Advice on new loudspeaker array... Genelec 8010 speakers?

2015-10-22 Thread Jörn Nettingsmeier


On 10/22/2015 08:27 AM, Augustine Leudar wrote:

Bah... (; hello Charlie hope you're well ! I just realised the x32 has 32
ins but only 22 outs you can expand it but probably end up paying more than
madi or adats...


FWIW, there even is a MADI I/O card for the X32/M32. The IEM Graz has 
one, I got to play with it at ICSA. Works.



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Re: [Sursound] SQ QUAD

2015-10-21 Thread Jörn Nettingsmeier


On 10/21/2015 03:14 PM, Richard wrote:

That is very true, and there never will be.



I have a marvellous algorithm that will restore old shellacs to their 
original 10-octaves full surround beauty, but since the world is what it 
is, I'm not going to show it to you.



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Re: [Sursound] Advice on new loudspeaker array... Genelec 8010 speakers?

2015-10-17 Thread Jörn Nettingsmeier


On 10/16/2015 08:34 PM, Fons Adriaensen wrote:

Next Ambdec (already in use here, and to be released soon)
can do this.

<..>

1. Bandsplitting, four options:

   - single band
   - single band with sub xover
   - dual band
   - dual band with sub xover

   so in the latter case you'd have 3 bands. The
   sub filter is 4th order.

2. For each band you can add as many matrices
as you want, each of them handling user
defined subsets of inputs and outputs.

3. Matrix outputs are added, near-field compensation,
delay and gain control are done for each output.

Processing can be multi-threaded on SMP hardware.

The 'sub' band you could use to drive subs, or to
crossover to a lower order decode using the full-
range speakers. Of course if you want to do both
and dual band as well, you'd need four bands.
I'll consider that if there is some press^H^H^H^H^H
interest.


Consider this interest :-D Where do I sign up as alpha tester? (a lousy 
one since I don't have too many speakers at the moment...)


I don't think four bands are strictly necessary... If there are subs, it 
seems an odd choice to additionally spread the low-mids across mid-hi 
speakers. Better to move the xover a bit higher if the mid-hi speakers 
need some more help.



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[Sursound] spherical mic array order balancing

2015-10-17 Thread Jörn Nettingsmeier


On 10/16/2015 06:27 PM, David McGriffy wrote:

I've been thinking about how this discussion might apply to a couple of
things I'm working on and it seems to me there are two different problems
here.

First, there is the issue of higher order mics often not really being
higher order at low frequencies.


Nor at high frequencies because of spatial aliasing...


But isn't this really a problem of
encoding and not decoding?  It seems like we shouldn't have to know
anything about the mic once we are in B-format.  And such considerations
would not apply to synthetically panned higher order signals, right?


Yes, that is the sad thing about B-format from spherical microphone 
arrays. They are not truncatable, since the upper orders are 
increasingly band-limited.
You have to balance the spectrum by equalizing the lower orders 
accordingly, so that if you use for example the full fourth-order output 
of the Eigenmike, you arrive at a reasonably flat energy spectrum.
But if your listener assumes she or he can truncate the signal set to 
third or second order due to lack of speakers, they will lose more 
treble than bass by doing this.


I wonder if it's possible to put some fake LF borrowed from the lower 
orders into the higher ones, so that the spectrum is flat for each 
order, with the understanding that the directional information will be 
wrong... But I'm not sure I'm seeing all the implications of this...



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Re: [Sursound] ADAT vs. MADI

2015-10-17 Thread Jörn Nettingsmeier


On 10/16/2015 11:37 PM, Augustine Leudar wrote:

I've been using them for four years and not experienced those problems
(apart from original a da 8000's PSU problems) . The four adat set to slave
are controlled by the raydat master clock no need for Wordclock.


Agreed, this particular setup works well in my experience, too.

The problem with ADAT for pro environments is that you cannot do 
distance and you cannot have patchbays (well, you can try with Toslink 
connectors, but it's very unreliable).



In fact
many people seem to regard external master clocks as snake oil.  Im not
going there.


If you have really crappy clocks, an external master helps. But if you 
have any RME equipment, you already have a very good word clock 
generator, no need to pay for an extra one.


What I'm saying is that some implementations of ADAT are notoriously bad 
at locking and syncing (Yamaha desks come to mind - I've had to 
wordclock them time and again because they were unable to sync to ADAT 
reliably). So I would prefer having wordclock sync everywhere, which in 
the case of the ADA8200s requires an extra wordclock splitter since they 
don't have WC outs.



Cable length is not a problem as I have computer and adats in
the same rack so cable length is not an issue.


Well, if you can wire the ADAT once, inside a single rack, and never 
touch it again, you're probably fine.



As for signal to noise ratio well I've seen
this discussed quite a lot with pre amps but rarely these days with dacs.
Thd and "noise " is well below audible levels in both units - I can point
you to measurements of ada800 if your interested certainly it held its own
against more expensive units.


It's true that the ADA8k is good enough for most applications, but the 
extra price of a better converter does get you another 10-15dB less 
THD+N, plus the THD at least in the old 8000s was quite nasty. I've just 
replaced an 8000 with an Andiamo in my mixing environment, and the 
difference is quite audible there. Probably not in your average live 
venue, though.
Haven't had the chance to hear the new 8200s yet, from what I heard 
they're a lot better.



The one place I do concede is the cables are
a pain but if you set it up right and dont move it there shouldn't be any
problems . My rig is mobile do the cables are annoying but not 2 thousand
pounds worth of annoying. I guess this is why large multichannel
  installations such as the wfs systems at Salford university or game of
life because the extra cost of madi is not worth the gain in performance
  (which audibly is none). Still i like the idea of one box out of curiosity
does the Andiamo supply balanced outs ?


Of course. Due to space restrictions (it's 32 AD/DA in one rack unit), 
the balanced ins and outs are D-SUB-25. A bit of extra hassle - I made 
my own breakouts for it. You can get them from RME or Directout, but 
they are quite expensive.
But since I'm also using it in the studio, the D-SUBs actually quicker 
to connect than individual XLRs.
And the new firmware comes with a complete matrix router that spans all 
analog and MADI ins and outs. Quite handy, and you can run it 
MIDI-over-MADI, so you don't need an extra cable there.



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Re: [Sursound] Advice on new loudspeaker array... Genelec 8010 speakers?

2015-10-16 Thread Jörn Nettingsmeier


On 10/15/2015 10:51 PM, Dave Malham wrote:

One of the things that should be investigated in conjunction with higher
order Ambisonics material would be to "fade down" the higher order
components as the frequency drops, thus spreading the bass over more
speakers, reducing the strain on the individual speakers whilst maintaining
the spectral balance - hey, wasn't that Richard Lee's Powered Integrated
Sub concept from several years ago?? Doesn't help with first order materiel
but





Intriguing idea, that. So we would apply zero-phase high-pass filters to 
the second and higher components?
Should be nice for a test run, but how to keep latency down for live 
electronics and A/V sync? How would we phase-align an IIR filter? 
Allpasses on the lower components?


The spectral balance would be maintained despite the filters, since 
we're in LF, where each new order "takes away" as much as it "adds", so 
to say. Unlike at HF, where we have to add energies and any such 
filtering throws the spectral balance of kilter, as Eigenmike users will 
know...


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[Sursound] ADAT vs. MADI

2015-10-16 Thread Jörn Nettingsmeier


On 10/16/2015 01:02 PM, Augustine Leudar wrote:


I am curious also about your comment that adat was unreliable - is this
your experience in general or just the original ada8000 ? The ada8200 has
no reports of unreliability since the.changed PSU. Unless he needs
his.computer more than ten metres from adat it puts the cost of 32 channels
from around 1500 euros to andiamo and madi around 5000 euros .


I wasn't talking about the Behringer converter, specifically. Yes, the 
old one was notorious for its badly designed power supply - it died on 
me several times. The new one is supposed to be much better.


But ADAT itself is a problem in fixed installations. The connectors are 
toys (I usually apply a drop of hot glue, but that has its downsides :), 
th plastic fibers are toys (get the right length, signal will drop a lot 
when you coil them too much), and ADAT sync is one of these technologies 
that are working most of the time. Which in my book is worse than 
something that never works :(
Plus ADAT interfaces will pop loudly when you switch them in the wrong 
order or hotplug a connector.


I already talked about ADAT cable length restrictions. Can be overcome, 
but then you should factor in the costs of a dual-screen remote console 
in the venue that connects to the rendering computer in the machine 
room. Also, factor in the (small) cost of Wordclock sync to every ADAT 
device, just to be safe.


Just providing a single MADI connector for visiting artists to hook up 
to is often easier that re-routing a bunch of ADAT lines. Plus you don't 
want to take your ADAT plugs through too many mating cycles, they will 
become even more unreliable after a short while. BNC or SC are bullet-proof.


So yes, you can cut the budget a little, but I don't think it's the best 
option. I feel I should be recommending something Audio-over-IP-based 
rather than MADI, but with its strange renaissance, MADI gear is now 
quite affordable and will be for another few years. Plus it's usually 
built to broadcast standards, which means it's an investment that will 
last you a while. And in terms of SNR, the Andiamo is a lot better than 
the Behringer (not to say the latter is bad).




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Re: [Sursound] Advice on new loudspeaker array... Genelec 8010 speakers?


On 10/15/2015 11:31 AM, Sebastian Gabler wrote:

That's specifically true for the 8010s. Their max SPL @ 1 m is app. 105
dB per pair, but the long term SPL is only 91 dB, expressly stated that
it is because of the protection circuit in the manual.
That being said, with 32 speakers, the SPL is 4 time higher than for 2.
That's far beyond comfort zone for any listener for the specified values.


Careful there. No program material ever uses all speakers for the oomph 
passages. Usually it's a very sharply located source that shouts out at 
you, which basically means every single speaker has to be able to 
deliver that oomph, single-handedly. Curse of multichannel. Content 
doesn't scale :(


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Re: [Sursound] Advice on new loudspeaker array... Genelec 8010 speakers?


On 10/15/2015 03:30 PM, Augustine Leudar wrote:

I admit there use of the word "ultrasound" is ridiculous - but they do work

On 15 October 2015 at 14:27, Augustine Leudar <augustineleu...@gmail.com>
wrote:


Here you go Jorn - a speaker with a narrow dispersion pattern - (they call
it parametric or sound lazer) :

http://www.soundlazer.com/what-is-a-parametric-speaker/


Hmmm. That website completely fails to amaze me.  ;)
Show me your polar pattern (per frequency), show me your amplitude 
response, let me look at the phase, then we talk.


Their selling point on page one seems to be "works great in advertising" :-D

Not to say this thing can't be amazing fun to play with. But I'm not 
considering it a serious tool unless I can see the specs.


Which will be sobering, unless their other project on kickstarter took 
off as well and they have the PhysicsIsLookingElsewhere™ field working 
by now. ;)



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Re: [Sursound] Advice on new loudspeaker array... Genelec 8010 speakers?


On 10/15/2015 03:41 PM, Augustine Leudar wrote:

What can I say Jorn - we have some and they work - not as well as
adverstised obviously but strip away the hype they have something thta mor
eor less does as described -  you'll have to wait a while for those
measurements ;)


For now, the low cutoff frequency would be enough.

Mind you, I'm not saying that these things cannot be great creative 
tools. But I'm a sound engineer. I need to deal with the artistic output 
of other people, in such a way that they are not going to ram the 
speakers down my throat because half of the spectrum is missing. I need 
general-purpose speakers, which on a bad day means ten octaves and 110dB 
peak SPL.


We've seen all those outlandish claims of magical waveguides that are 
just fractions of the wavelength in diameter and yet shape the sound so 
wonderfully that a 20Hz beam will travel all the way to the moon (using 
the revolutionary VacuProof™ technology that will finally bring 
cinema-friendly space battles). The problem is, this waveshaping is not 
physically possible. You can make a plane wave, but unless it's a huge 
plane wave with respect to frequency, the edge dispersion will make it 
fall apart. In the end, it's just a point source with a little dent in 
it. I don't even need to wait for measurements.


Even if you consider ultrasonic systems: at some point, the ultrasound 
has to be demodulated (it is actually demodulated _everywhere_ in the 
beam), and then you again have a frequency-dependent radiation pattern. 
So yes, some years in the future maybe we have an ultrasonic projector 
that is actually capable of putting a kick drum right in the middle of 
the room without frying anything in its path. But as soon as the 
baseband sound materializes, it will be (almost) omnidirectional again.



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Re: [Sursound] Advice on new loudspeaker array... Genelec 8010 speakers?


On 10/15/2015 03:59 PM, Jörn Nettingsmeier wrote:

On 10/15/2015 03:41 PM, Augustine Leudar wrote:

What can I say Jorn - we have some and they work - not as well as
adverstised obviously but strip away the hype they have something thta
mor
eor less does as described -  you'll have to wait a while for those
measurements ;)


For now, the low cutoff frequency would be enough.

Mind you, I'm not saying that these things cannot be great creative
tools. But I'm a sound engineer. I need to deal with the artistic output
of other people, in such a way that they are not going to ram the
speakers down my throat because half of the spectrum is missing. I need
general-purpose speakers, which on a bad day means ten octaves and 110dB
peak SPL.

We've seen all those outlandish claims of magical waveguides that are
just fractions of the wavelength in diameter and yet shape the sound so
wonderfully that a 20Hz beam will travel all the way to the moon (using
the revolutionary VacuProof™ technology that will finally bring
cinema-friendly space battles). The problem is, this waveshaping is not
physically possible. You can make a plane wave, but unless it's a huge
plane wave with respect to frequency, the edge dispersion will make it
fall apart. In the end, it's just a point source with a little dent in
it. I don't even need to wait for measurements.

Even if you consider ultrasonic systems: at some point, the ultrasound
has to be demodulated (it is actually demodulated _everywhere_ in the
beam), and then you again have a frequency-dependent radiation pattern.
So yes, some years in the future maybe we have an ultrasonic projector
that is actually capable of putting a kick drum right in the middle of
the room without frying anything in its path. But as soon as the
baseband sound materializes, it will be (almost) omnidirectional again.




Here's an interesting datapoint:

http://www.ultrasonic-audio.com/images/Acouspade_vs_Audio_Spotlight.png

Consider the Acouspade, to the right. The polar pattern is truly nice, 
unless you look at the fact that we are not told what happens below 500 
Hz (because very likely, nothing much is happening there), and we take 
note of the tiny, tiny dotted line that denotes 4 kHz, which is down by 
around 18 dB on-axis, indicating that in the treble range, not much is 
happening either.


This is a wonderful tool if you want to whisper textual information at 
your visitor. But what you are actually perceiving is your amazing 
ability to suspend disbelief in the presence of a friendly voice from 
your own species.


--
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: [Sursound] Advice on new loudspeaker array... Genelec 8010 speakers?