[Sursound] The birth of Ambisonics & the Soundfield Mike

[Richard Lee]

https://www.youtube.com/watch?v=X23hZNoSkUs

The Calrec Soundfield mike, the practical implementation of Michael Gerzon & 
Peter Craven's invention, was how I got involved in microphone design.

It was incredible just being with Michael .. a guy with a brain truly the size 
of a planet.

The Mk4 was, IM not so HO, the best microphone of the 20th Century.
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[Sursound] practical HOA encoding

[Richard Lee]

.. a reply to stuff on the "Anyone had experience with AllRAD decoding to 
physical loudspeakers?" thread where Aaron brought up the topic of NFC.

" A practical problem is that forward NFC filters have infinite gain at DC.
Jerome Daniel's solution is to have a reference decoding distance for
Ambisonic program material, say 1 meter -- In the encoder, you encode and
then decode for speakers at 1 meter (which produces finite gains at DC).

[1] J. Daniel, "Spatial Sound Encoding Including Near Field Effect:
Introducing Distance Coding Filters and a Viable, New Ambisonic Format
," *Preprints
23rd AES International Conference, Copenhagen*, 2003.

[2] J. Daniel and S. Moreau, "Further Study of Sound Field Coding with
Higher Order Ambisonics," *Preprints from 116th AES Convention, Berlin*,
no. 6017, 2004. "

Daniel introduced this into a shark feeding frenzy of gurus & pseudo-gurus 
debating HOA encoding.  I wanted to join in and introduce Yet Another 
Encoding but beach bum issues intervened.  Besides I was seriously 
intimidated as I neber wen 2 skul.
___

MY SOLUTION TO LOADSA LF GAIN FOR HOA ENCODING

Ignore it.
___

Making a HOA soundfield, we deal with 2 type of sources/objects.

The first is naturally recorded stuff from a well calibrated soundfield 
mike and IMHO, these are the most important.  It was the incredible sense 
of REALITY playing an original Calrec Soundfield recording on a properly 
set up Ambisonic system that pulled many of us into the arcane world of 
Ambisonia.

As a(n Ambisonic) mike designer, you immediately come up with a serious 
problem with correct Ambisonic encoding .. the huge LF boost at close 
range.

Even the First Order Ambisonic (FOA) Calrecs & TetraMics have 30dB/8ve LF 
filters to alleviate their facility as seismographs, air-con, slamming 
doors, distant thunder  detectors.  And you will find even more LF 
filtering required at times.  The pretenders w/o such filters WILL have 
serious problems.

A practical HOA mike has additional problems as it must use cancellation & 
mixing of 1st & 2nd order microphone capsules + LF boost to get the high 
orders which give serious S/N problems at LF.  You see this with full force 
in Eigenmike which uses omnis which need even more LF boost.  Angelo Farina 
has several reports & papers showing how useful (??) this is.

The solution is to drastically HP filter the higher orders AND MATCH THE 
PHASE OF THE UNFILTERED CHANNELS.  You can do this with the method in the 
appendix of BLaH3

http://www.aes.org/e-lib/browse.cfm?elib=14705

In essence, a practical HOA microphone will give you as much HOA signal AS 
IS USABLE (mostly determined by S/N) while keeping the correct relationship 
between the components.
__

But you will argue, "But that means the Ambisonic order will be frequency 
dependent???"

Why is this a bad thing?  Ambisonic playback is also frequency dependent.

An Ambisonic recording is the DEFINITION OF SOUND AT A POINT.  But this 
POINT is frequency dependent!

For a FOA recording, at 20Hz, an IEC listening room is a 'point'.  But at 
1kHz, the 'point' is about the size of a Mk1 Human Head.

What HOA does is define the 'point' to be larger at any frequency.  We 
don't need a larger 'point' at LF so FOA is usually sufficient at LF.  At 
mid & higher frequencies, HOA encoding allows a larger 'point' to be 
defined.

So a practical HOA mike as described above degrades in exactly the correct 
fashion to be USEFUL.

A PRACTICAL decoder can make use of as much info as is available in the 
recording without worrying about EXACTLY what is available at each 
frequency.  (Actually this isn't strictly true but that's a whole new field 
of research in itself which may lead to Yet Another HOA Encoding and/or 
Decoders :) )


Back to HOA encoding.

The 2nd type of source/object are artificial.  HOA is most useful for 
shoot-them-up games and other virtual reality stuff.  You pan the (mono) 
bad guy to someplace in your HOA space

BUT THIS STUFF CAN USUALLY BE SET AT AN INFINITE DISTANCE SO NO NFC IS 
REQUIRED

It's very rare that proximity adds anything to the virtual reality gaming 
experience.  (In practice, 'infinite distance' can be considered anywhere 
outside the 'circle' of speakers.)

If da bad guys do come closer, you can use the same drastic HP filters to 
keep the LF boost within limits.  ie treat the sources as though they were 
being picked up by a REAL LIFE HOA MIKE.

There is NO NEED FOR SET DISTANCES/FREQUENCIES etc.  You only need to MATCH 
THE PHASE.

And usually you don't need to add NFC to the HOA signals for a realistic 
effect.

FOA (with NFC) is certainly good enough for the Spitfire flying low 
overhead .. or even Peter Lennox attempting to run over Grand Vizier 
Malham's Mk3A Soundfield with his motorcycle.  (one of the very first 
Soundfields with the correct NFC encoding)

So in most cases, just use LF boost on 

Re: [Sursound] Hardest wearing outdoor speakers

[Richard Lee]

> Behringer ms10s that withstood months of tropical storms in the Amazon - 
so

Have you got that model number right?  I still think the cheapo solution is 
the way to go .. but it would be good to know what has worked for you .. 
even if it isn't 35 yrs.  I get MS20 and loads of other stuff but not MS10 
or MS10s

And what is an Extreme 5 ?
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Re: [Sursound] Ambisonic Mic Comparison

[Richard Lee]

There's a number of issues brought up in this thread which Core Sound have 
been aware of for some time and have been attempting to address.  But its 
difficult for a small company to make major changes on the small turnover.

For what its worth, the 'new' PPAc will give around 1dB improvement in 
perceived S/N regardless of your favourite weighting (more if you are not 
using a Metric Halo or similar).  This has been achieved mainly by a bit 
more than 1dB more output. 8>D

The MOTU Traveler has near SOTA noise performance but the design is flawed 
and they often become very noisy over time.  If you have one which has been 
OK for more than 12 mths, you are probably OK

I'm on my 3rd Traveler.  The 2nd one developed the noise almost exactly 12 
mths after I received it.  As the 1st took 3 mths to be 'repaired', those 
of us in Oz are not happy bunnies.  (The Traveler is actually Angelo 
Farina's who kindly lent it to Cooktown Recording and Ambisonic 
Productions.)

Its pretty obvious MOTU don't have in-house design expertise and they deny 
there is a problem.  I've not looked inside a MOTU 4pre ... but so far, 
those we know of haven't developed this problem so it has our cautious 
recommendation.

In terms of noise with TetraMic, I'd expect a 'good' Traveler to be on par 
with Sound Devices and Metric Halo (sadly Mac only) and you would notice 
the noise difference between these and the DR680 which is our 
recommendation for an inexpensive portable device.

My experience is if you are not recording bird song in the Norfolk Broads, 
the noise performance of TetraMic wth the above good preamps is not a 
problem.  There are some excellent recordings on Ambisonia from John 
Leonard & Paul Hodges .. some of which were made in a very quiet studio.

That's not to say we aren't working on even better performance ... 8>D
___

(There are problems with noise on the P48V on early DR680s and Paul Hodges 
has a mod for these on Channels 1-4.  I believe, TASCAM, Europe were 
modifying Mk1 DR680s and the new one has sorted this out.

If you have an old DR680, it is worth doing Paul's mod as it affects some 
mikes, both $$$ & inexpensive.  TetraMic is actually pretty immune to P48V 
noise.)
__

If you have a good A/D without preamps, you can build a 4 channel preamp 
using THAT chips with near SOTA performance.

Bear in mind you need to match THAT 1510s & 1512s for gain.  The internal 
resistors are laser trimmed for CMR but the absolute values differ from 
chip to chip.  Thanks to David Pickett for this tip.

If you prefer to use SSM2019 or TI INA163 chips, use them with the latest 
THAT circuits for more reliable long term performance.  All three are 
capable of excellent performance in the right circuit.

It's the protection scheme that is flawed on the SSM, TI & (very early) 
THAT circuits.  The correct protection is cheapo 1n4004 diodes, preferable 
Glass Passivated 1n4004GP.


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Re: [Sursound] The BBC & Quadrophony in 1973

[Richard Lee]

If anyone has recordings of any of this Matrix H, HJ, UHJ stuff, please 
post a copy on

http://ambisonia.com/

with a good description of what it is and the circumstances.

I don't think UHJ is dead yet as the biggest present market for music is 2 
channel stuff for headphones and UHJ gives excellent results for this and 
other playback methods too.

> I still have some off-air recordings of the BBC's Matrix H system. Never 
tried to decode them but they have a pleasing surround sensation when 
listened to on headphones. Must have been the mic. placements ??

> One broadcast I recall was a play based upon Alice in Wonderland entitled 

"Alice's Adventures In Wonderland".

> I can remember listening to this play on BBC Radio 3/4 several times, 
great
stuff.

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Re: [Sursound] External phantom battery source for TetraMic PPAc

[Richard Lee]

> Most days I would agree with you: but there is always Murphy's Law to 
contend with, and the specific case of this states that a PP3-powered 
recording device will fail only when a unique event is to be recorded.

Ha!  In my limited experience, Murphy is MUCH more likely to strike at 
rechargeable batteries.  8>D

You find, far too late, that you haven't charged them properly or they no 
longer retain charge properly.  Battery tech is improving day by day but 
Alkalines are still near the top of the list for convenient reliable power.

If you use rechargeables, I suggest you have a new alkaline PP3 or two in 
your spares kit with the correct connector.

I've never had new Alkalines from the big makers, Duracell, Energiser, GP 
etc fail on me.  I'd be wary of OEM batteries though .. especially 
supermarket own name brands.  In Oz, the Jaycar Eclipse brand seems OK.

DO NOT USE ZINC CARBON CELLS FOR ANYTHING.  Alkalines ALWAYS work out 
cheaper regardless of application.

BTW, the 10 hr life I mentioned is for an end voltage of 8V but PPAc will 
work fine to lower voltages.  I mention 10 hrs as that should see most 
recording events & concerts with ample safety factor.  It's cheap insurance 
to always use a new one.
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Re: [Sursound] Sursound Digest, Vol 96, Issue 3

[Richard Lee]

Can you give us more detail about these tests and perhaps put some of these 
natural recordings on ambisonia.com?

The type of soundfield microphone used .. and particularly the accuracy of 
its calibration ... makes a HUGE difference to the 'naturalness' of a 
soundfield recording.

Some good examples of 'natural' soundfield recordings with loadsa stuff 
happening from all round are Paul Doombusch's Hampi, JH Roy's schoolyard & 
John Leonard's Aran music.  Musical examples include John Leonards Orfeo 
Trio, Paul Hodges "It was a lover and his lass" and Aaron Heller's (AJH) 
"Pulcinella".  The latter has individual soloists popping up in the 
soundfield .. not pasted on, but in a very natural and delicious fashion 
... as Stravinsky intended.

> Also to my experience, and that doesn?t seem to be a very popular view 
yet in ambisonic community, these parametric methods do not only upsample 
or sharpen the image compared to direct first-order decoding, but they 
actually reproduce the natural recording in a way that is closer 
perceptually to how the original sounded, both spatially and in timbre.

> Or at least that?s what our listening tests have shown in a number of 
cases and recordings. And the directional sharpening is one effect, but 
also the higher spatial decorrelation that they achieve (or lower 
inter-aural coherence) in reverberant recordings is equally important.

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Re: [Sursound] Using Ambisonic for a live streaming VR project

[Richard Lee]

> The main mechanisms for disambiguating 'cones of confusion' (and this 
includes front-back reversals) are: pinnae effects (Batteau) and 
head-movements (Wallach) - so, without either of these mechanisms at play, 
one would expect directional ambiguity.

You can test the relative importance of these for YOURSELF with the famous 
Malham / Van-Gogh Experiment

http://www.ambisonia.com/Members/ricardo/PermAmbi.htm/#VanGogh.

I still have some Diamond encrusted caps with optional Golden Pinnae but 
you need to pay in used bank notes.  No Confederate money please.

Michael came up with his rE & rV theories ... not by considering how to 
best replicate HRTFs bla bla .. but by asking ... "what information could 
the Mk1 Human Head (+ torso + processing inside + bla bla) possibly have 
available to determine localisation?"

If youi perform the above experiment, you'll find the Moving Head  cues are 
FAR more important than the Fixed Head cues (HRTFs bla bla).

Where the HRTFs have the most significance is in the vertical plane.  It's 
the different frequency response as a source moves off the horizontal plane 
that allows the Mk1 HH to process 'height'.  But even then, Moving Head 
cues are far more unambiguous .. and don't require a priori knowledge of 
the source.

If the HRTF cues break down completely (eg simulating a pair of coincident 
back to back cardioids as the crudest possible binaural decode), simulating 
the Moving Head cues (head tracking) lets the Mk1 HH decode all this 
without any problem, fuss or discomfort.

> I would like a little more information on ?head movements?.  I suspect 
all head movements are being treated as equal, and I have a theory that 
short rapid movements (like shaking the head) should be treated separately 
from movements that include the shoulders, or even the whole body. Short 
rapid movements of the eyeball have been studied and are well understood; 
without these small movements the visual field collapses completely. Does 
something similar happen for the aural field ?

One of the more surprising things that Michael worked out is that the 
Moving Head localisation models gave the "same answers" regardless of 
whether they assumed you turned your whole body to face the source (eg 
Makita) .. or those that only allowed small involuntary head movements (eg 
Clark, Dutton & Vanderlyn IIRC)

It's all there in his "General Metatheory  " if you are prepared to 
study it and follow up the references.  See especially the 'stereo' 
appendix.

http://www.aes.org/e-lib/browse.cfm?elib=6827

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[Sursound] OT Stereo stage width - Was: Static stereo source in rotating soundfield, possible?

[Richard Lee]

Aaron Heller wrote
> Marc Lavall, Eric Benjamin, and I put together a Trifield (three speaker 
> stereo) plugin and demo'ed it a Burning Amp last fall. It is hosted at
https://bitbucket.org/ajheller/trifield/overview
> There are also some plots that use Gerzon velocity and energy localization 
> vectors (rV and rE) to analyze, +/-45 deg stereo vs Trifield  vs +/- 30 deg 
> stereo that shed some light on why "the +/- 30 deg stereo triangle" works 
> well.
Thanks for this Aaron, Eric & Marc.
There's a small error in the last pic "Magnitude of Localisation Vectors +/- 
30".
The x-axis should be from -30 to +30.  You can extend the info to -45 to +45 
virtual sources by panning beyond the speaker positions and show +/- 30 is 
still 'better' than +/- 45 speakers.
You can also show the DIRECTIONS of the rV & rE vectors are more in synch for 
the +/- 30 case
I did a lot of this stuff in my mispent youth trying to apply Gerzons theories 
to stereo.
Another big effect of wide speaker spacing is the frequency response of the 
virtual source varies a LOT more with position.  It's bad enough with +/- 30 
speakers.  You can test this for yourself by comparing a mono source panned to 
the centre (or near centre) with it panned to L or R.
The exact centre is also prone to HUGE comb filter effects.
All these effects (image stability via rE & rV, frequency response, comb 
filtering) are reduced if the speaker spacing is reduced (with proper 
compensation via more L-R bla bla ... to maintain image width).  The various 
Transaural / Stereo Dipole / Ambiophonic systems may be considered 
sophisticated variants of this.
Stefan Schreiber wrote :
How can rV > 1 be true??! (second last image...)
Gerzon's infamous vector magnitudes rV & rE are a measure of image stability 
with head movement.  Small magnitudes mean the image moves LESS than you'd 
expect in 'real life'.  In theoretical 'real life' (?!?) rV = rE = 1 .. but 
actually the magnitudes are usually slightly less than 1.0  An rV > 1 means the 
image moves MORE than in 'real life'.  Very large rV s are unnatural & 
disturbing as they are VERY uncommon in 'real life'.  You can get a taste of 
this by listening to normal +/- 30 stereo with L-R turned up.  Also some of the 
more naive early surround sound systems.
All explained in the Appendix of Gerzon's "General Metatheory of Auditory 
Localisation," Preprints from the 92nd Convention, Vienna, no. 3306, 1992.  He 
shows how his rV & rE theories model (give the same answers as) all models of 
auditory localisation except for the impulsive HF and pinnae colouration 
models.  This Appendix is required reading for anyone interested in auditory 
localisation. please reply to rica...@justnet.com.au NOT to the yahoo address 
which is riddled with spam. I never look at it.
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Re: [Sursound] Soundfield MK IV Disassembly

[Richard Lee]

> I have a question for the keepers of ancient wisdom: how does one remove 
the capsule assembly from a Soundfield MK IV microphone? I have one on 
which the 1 kilohm "capsule heater" resistor has gone open circuit. The 
cone on the MK V and newer is split and comes apart pretty readily, but the 
MK IV cone is solid and it's not apparent how to reach the resistor.

Rudy, It's more than 30 yrs since I took a Mk4 capsule assembly apart so 
bear with me I'm pontificating from the wrong orifice.

1   Unsolder the leads from capsule to PCB.
Can't remember if you can reach this from inside as the connections may 
be 
in the cone section.
If so (!!#*??), you'll have to disassemble the capsule assembly, remove 
at 
least 2 capsules on their tetrahedron sections and unsolder from the tags 
behind the capsule.

Keep the capsules in their little holder to provide some measure of 
protection but this is still a brain surgery type operation.

2   The tetrahedron stalk is screwed to a brass block on the PCB.  The 1k 
resistor is sandwiched to the block/stalk in some way which I've long 
forgotten.  IIRC, there was another small PCB whose sole purpose was to 
clamp the resistor.

3   Removing the tetrahedron allows the cone to come off.

I you have the set of dwgs which were supplied with early Mk4s, there 
should be one that makes this clearer (or not).

My apologies for this Heath Robinson design from my mispent youth.
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Re: [Sursound] Never do math in public, or my take on explaining B-format to binaural

[Richard Lee]

Just to bring everyone down to earth ..

There are two easily reproduced experiments first carried out by prominent 
members of this group which put these effects into perspective.  They are 
the

Greene/Lee Neckbrace
and
Malham/Van Gogh Experiment

The first shows 'real life' Fixed Head Localisation (which matched HRTFs 
address) is TERRIBLE.  Many people can't even distinguish back/front with 
perfect (measured on their own noggin) HRTFs ... or even in 'real life' 
with a Greene/Lee neckbrace.  Anyone who has done fixed head localisation 
experiments finds this out real quick.

The second shows that even the tiniest amount of head movement improves 
localisation immensely and any ambiguity due to mismatched Pinnae etc (and 
YES, the pinnae colouration effects are chaotic) are INSTANTLY resolved. 
 No 'training' is necessary with head movement.

Even vertical localisation, for which Fixed Head HRTFs have the most 
benefit, require a priori knowledge of the source spectrum.  I've done a 
small amount of work involving victims ... I mean subjects ... blindfolded 
and tied up face down on anechoic chamber floors which show the first pin 
drop is impossible to localise.  Second and subsequent pin drops are much 
easier.

I'll point out that Gerzon had Fixed and Moving Head versions of all his 
Localisation Theories.  The infamous Energy and Velocity vector 
'magnitudes', rE & rV, are in fact a measure of the correspondence of Fixed 
with Moving.

His Energy and Velocity models incorporate (give the same results as) ALL 
the existing Localisation models except for the HF interaural delay 
(Transient) and Colouration ('HRTF') models.

One of Gerzon's most important contributions is that he shows the 
equivalence of the full Moving Head models like Makita, which assume the 
listener will fully turn to face the sound ... with the models that only 
assume small involuntary head movements.  See "General Metatheory ... " for 
the nitty gritty.


If you make some B-format recordings with a properly aligned Ambisonic mike 
like TetraMic, you can test some of this for yourself.  Have lots of things 
happening all around including up & down.  Use headphones and the crudest 
possible Binaural decoding ... slightly hyper cardioids at about 150.

You will find about 10% of the population have difficulty with front/back. 
 But let these guys twiddle the Azimuth & Elevation controls on VVMic 
themselves and they immediately become happy with the scene ... even before 
they work out VVMic's slightly quirky interface.


CONCLUSIONS

If you have Head Tracking (ie Moving Head Localisation), don't bother with 
fancy HRTFs.

Eric Benjamin found that you get most of the benefits from just getting 
head size right but even this isn't necessary if you have Head Tracking. 
 Blumlein shuffle probably worth doing as you essentially get it free with 
your simple IIR implementation.

Fancier HRTFs will need EVIL FIRs to be of use.  You will lose any chance 
of 'real time' and muck up the experience for 'mismatched' listeners. 
 Expect only small (if any) 'improvement' for the huge extra.computing load 
to interpolate between HRTFs.

If you haven't got Head Tracking, GOTO Head Tracking

This covers all the Virtual Reality applications.  The Video Game people 
like Simon Goodwin of Codeworks have been doing it for at leas a decade 
with 3rd Order HOA IIRC.

If you insist on fancy HRTFs and Fixed Head ... do you seriously think you 
can improve on the listening experience of present & past generations of 
listeners, who have listened to 'music' over ear buds for more than a 
decade ... with fancy HRTFs ? GOTO Head Tracking.


SPECIAL OFFER

Send $500 in used bank notes to me at Cooktown Recording and Ambisonic 
Productions mentioning Sursound, for a sample Greene/Lee Neckbrace and 
Diamond Encrusted Malham/Van Gogh cap. Golden Pinnae are an extra cost 
option on the last item.  No Confederate money please.
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Re: [Sursound] WFS in a small box?

[Richard Lee]

This is likely an implementation of Ambiodipole / Stereo Dipole / Transaural
http://ambisonia.com/Members/ricardo/StereDpl.htm/
There's pretty pictures on the ISVR website that show 'some' measure of 
wavefield (soundfield?) synthesis takes place .. certainly more than with the 
usual +/-30 stereo.
Note the LF limitations which are detailed in the ISVR OSD descriptions.
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Re: [Sursound] Modified SoundField Mk IV

[Richard Lee]

#641 & #650 refer to posts on the Gearslutz page

https://www.gearslutz.com/board/remote-possibilities-acoustic-music-loca  
tion-recording/547304-soundfield-mic-stereo-application-22.html

#641 describes his initial mods to the stick .. very similar (??!) to Ken's 
Mk5

#650 describes putting the FETs on the capsule backplate.  This gives 
probably the most improvement.

When I emerged from the bush, I toyed with the idea of offering a similar 
souping up service for Soundfields.  But the risk of damage in transit due 
to the flimsy tetrahedron is very high and I have no access to matched 
capsule sets.

I strengthened the tetrahedron while I was at Calrec ... but not enough :(

--
From:   Dave Malham[SMTP:dave.mal...@york.ac.uk]
Sent:   Mon, 19 October 2015 17:48
To: rica...@justnet.com.au; Surround Sound discussion group
Subject:Re: [Sursound] Modified SoundField Mk IV

What's the difference between #641 and #650??

   Dave

On 19 October 2015 at 06:38, Richard Lee <rica...@justnet.com.au> wrote:

> > Interesting: https://www.gearslutz.com/board/11410162-post650.html
>
> I endorse Rudy's mods described in #641 and #650
>
> Ken Farrar did something similar to his #641 mods in the Mk5
>
> ... but I think his post should be titled "Where Beach Bums & Angels fear
> to Tread" :)

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Re: [Sursound] Modified SoundField Mk IV

[Richard Lee]

> Interesting: https://www.gearslutz.com/board/11410162-post650.html

I endorse Rudy's mods described in #641 and #650

Ken Farrar did something similar to his #641 mods in the Mk5

... but I think his post should be titled "Where Beach Bums & Angels fear to 
Tread" :)
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Re: [Sursound] SoundField rental in Spain

[Richard Lee]

 A known floor noise of the dr-680 in high gain is from the phantom power, 
and the?Busman mod changes some components on the preamps, like some 
condensers and the opamps, and those condensers could be the solution on 
that kind of noise, but not all.

The biggest noise on dr-680 mk1 is poor phantom power.  Our own Paul Hodges 
found a cure which I believe is now in the Busman mod as well as the dr-689 
mk2 (Tascam Europe were modifying mk1s at one time when Tascam Japan were 
denying the problem)  This affected Soundfield SP200s and early Brahmas 
badly.  It should not affect TetraMic and mikes properly designed to be 
resistant to Common Mode noise.

http://outrecording.com/tascam-dr-680-noise-test/
DR680 have poor P48V noise.
http://www.audiomastersforum.net/amforum/index.php/topic,8654.msg77163.h  
tml#msg77163
Paul Hodges has a cure.  Increase P48V decoupling from 0u5 to 47u

The TASCAM specs are meagre and only suggest the Mk2 is still some way from 
MH/SD/RME/Motu noise.

Busman isn't much better when he says lowers the noise floor in the high 
gain setting by over -8db so I dunno what improvements his supa dupa OPAs 
bring.

 Because the Tetramic have a known noise floor highest than others mics 
like a SF ST350, I thought: less noise always is better, then I decided for 
this option.

TetraMic is inherently very quiet.  Your venue noise will be more intrusive 
even in a quiet studio.  Try John Leonard  Paul Hodges TetraMic recordings 
from Ambisonia.com  to check this out.

http://ambisonic.info/audio.html

ST350 is badly flawed but ST450 sorta returns to the performance of a Mk5 
Soundfield.  ST450 is quieter than TetraMic .. but only if you recording 
quiet Nature stuff.

What TetraMic does need is the very best preamps.  If you have a Metric 
Halo, Sound Devices, RME, Motu 4pre (or Traveler when its working properly 
8D) you'll get TetraMic's full noise potential which, as the Ambisonia 
excepts show, is rather good.

For lesser preamps, the new PPAc from Core-Sound, which simplifies life if 
you have to run long (or short) cables, gets you nearly 2dB nearer that 
potential.  It doesn't make TetraMic inherently quieter but 2dB more output 
means the preamps will sound 2dB quieter.

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Re: [Sursound] Directional confusion between different B-format players

[Richard Lee]

How is the Oktava Tetrahedral mike calibrated?

In what form are the calibration files?

Can you post a copy of the User Manual for it?

http://www.oktavausa.com/ProductsPages/Ambient4DMic.html has no info at all.
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Re: [Sursound] Converting 16 mic array recording to B format

[Richard Lee]

Mathias, can you please post these recordings on ambisonia.com

The Furse-Malham *.AMB format allows up to 3rd order

These would be the first publically available live HOA recordings of music from 
a HOA mike and may re-surrect the discussion of HOA decoders

Which Soundfield did you use?

What was the Playback Speaker Rig?  How many, how many levels?

What was the Decoder?

Is the L?sler/Zotter decoder in the Public Domain?  Or even a description?

From: Matthias Kronlachner m.kronlach...@gmail.com
To: sursound@music.vt.edu
Subject: Re: [Sursound] Converting 16 mic array recording to B format

 FWIW, I have Eigenmike recordings from the Essen Philharmonic (a 
 contemporary piece performed by musikFabrik K?ln) - they have been 
 sitting on my hard drive for more than a year because I couldn't get 
 studio time anywhere to do anything with it. Matthias Kronlachner was 
 part of that project, he might have done something with them.

I did not do any mixing of the material, but the IEM Graz played 
excerpts of the Eigenmike recording on several occasions eg. at the 
Ambisonics Symposium in Berlin and people did like it very much.
The 360? video recording is not so useful from this recording as it was 
very dark in the hall and the resolution is not that great. But who 
needs video if the sound is great ;-)

I did quite a number of recordings since then with the EM. Compared to a 
first order recording you can use a larger number of loudspeakers for 
playback and achieve a bigger sweet spot. The immersiveness is really 
nice! At the IEM we did several informal comparisons with switching 
between EM, a Soundfield (decoded 1st order and 3rd order with Harpex) 
and a Schoeps Omni as reference. All of these configurations have their 
use cases, but the representation of the space was always best with the EM.

The problem of limited bandwith for each order though makes it difficult 
to achieve the same tonal balance when changing the decoder/loudspeaker 
configuration.

But using good radial filters eg. by Mr. L?sler/Zotter (which is the 
only thing needed to process EM recordings that are currently not 
available to the public as far as I know) gives you something to start 
with and only little EQ is needed to get a convincing playback.


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Re: [Sursound] Converting 16 mic array recording to B format

[Richard Lee]

Fons, have you heard any music recordings with EigenMike?  No one seems to 
have tried.

Elko has a standing invitation to bring EM to any venue that Aaron  Eric 
are recording world class orchestras in good halls ... I mean obscure Mid 
West bands in non-descript halls :)  They can reliably record a zillion 
channels.

I've been trying to follow Angelo and his Merry Mens' efforts with EM but 
IIRC ... the best result so far, shows 1st order Ambi with an absolute 
(nearly brick wall) top limit of 10kHz which sorta ties in with your 
observations.

This was with Angelo's beloved funky Kirkeby FIRs :D

It's so nearly there but not quite.

Date: Thu, 14 May 2015 09:52:59 +
From: Fons Adriaensen f...@linuxaudio.org
To: sursound@music.vt.edu
Subject: Re: [Sursound] Converting 16 mic array recording to B format
 ...
The limits pointed out above are something that Gary Elko has clearly
understood very well (and some others apparently have not).
The beamforming sofware that comes with the Eigenmike will in general
not let you do things that depend on unrealistic accuracy of the mic
gain calibration.

On the upper end the limit for the EM is around 8 kHz. Above that,
the SW will just give you the signal of the single capsule that
is closest to the intended direction of the beam. The polar pattern
above the upper limit will be the one resulting from diffraction
caused by the solid spherical body (this becomes quite directional
in the frequency range considered). This at least produces a clean
signal in the upper octave which is better than the chaotic pattern
a beamformer would produce.

On the lower end, the limits that can be achieved assuming +/- 0.5 dB
gain errors are roughly

  1st order: 50 Hz
  2nd order: 630 Hz
  3rd order: 1.6 kHz
  4th order  2.5 kHz

The latter has such a limited frequency range that it's probably
better to forget about it. The EM software wisely doesn't claim
anything above third order.

The result of this, in particular of the lower frequency limits,
is that any higher order directional pattern will have to be a
compromise between on-axis and diffuse-field frequency response.
The requirements for this will depend on the application: a spot
mic or a set of beams intended for surround reproduction. The
tradeoff can be made partly by EQ.

So when using the EM for e.g. orchestral recording using a number
of beams pointed at the various sections of the orchestra, you will
need some rather unconventional EQ for the best results. This will
probably surprise most sound engineers used to the more traditional
way of using a set of normal mics to cover the sections. It may
also put them off. But it is certainly possible to make very good
recordings with the EM.

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Re: [Sursound] Converting 16 mic array recording to B format

[Richard Lee]

Duu.uuh!!  http://parole.loria.fr/DEMAND/DEMAND.pdf states

the microphones of the array ... are not calibrated with respect to each 
other, and so gain variations are to be expected: we found that the energy 
in some channels is consistently higher than in other channels. Algorithms 
working on this data should compensate for this variation

ie they haven't a clue what each capsule is doing.

This precludes any attempt at conversion to B-format and also of 
beamforming.

I was hoping this might lead to a discussion about EigenMike and how it 
might be made good enough to record music but this is certainly NOT the 
vehicle.

I can't help feeling they should beg borrow or steal a TetraMic and repeat 
their recordings.

Presently, about all you can say is they have a close bunch of unspecified 
mikes in some sort of horizontal pattern.

Curtiss, if you are after some 'realistic' atmospheric background (and this 
is something TetraMic and properly aligned Soundfields do better than an  
ything else), try ambisonia.com and recordings by John Leonard 
(soundmanjohn), Paul Doombusch, JH Roy  others.

John Leonard's specialty is WW2 aircraft flyovers but he includes a lot of 
airfield noise too :)  He's also got some very realistic street scenes, 
audience noise, applause etc too.

Aaron, you are right about SVD not being much use here as we have multiple 
solutions but I was hoping to dream up something to help S/N at LF.

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Re: [Sursound] Sursound Digest, Vol 82, Issue 2

[Richard Lee]

The strict answer to your question is in Gerzon's

The Design of Precisely Coincident Microphone Arrays for Stereo and 
Surround Sound

http://www.aes.org/e-lib/browse.cfm?elib=2466

There's a corrected copy at 
https://groups.yahoo.com/neo/groups/SoundfieldMic/files/Ricardo/.  You 
might have to join.

I used to be able to translate da maths to practical stuff but this 
Millenium my single remaining brain cell has given up.

Aaron Heller  Fons Adrieansen are your best bets.

You need to know the polar directivity patterns of your 16 microphones and 
their 'exact' postion.

Bet yus guys didn't know that paper was about more than Tetrahedral mikes 
:)

It's just that in da old days, the computing power required was thought 
impossible eg the impossible task of 'tweaking' 16 complex (in both 
senses) frequency responses to get the (unknown) best match to the desired 
polar diagrams.

Today, computing power is (usually) never a constraint and the much greater 
problem is knowing how to use it.

 I am wondering how I go about converting a recording made with a 16 
microphone array into B format?
...
 The recordings were made using a microphone array of 16 microphones 
arranged in 4 staggered rows, spaced such there is a 5 cm distance form 
each microphone to its immediate neighbors. The array is in a plane which 
in all recordings is parallel to the ground.

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Re: [Sursound] Infra sound Sub bass

[Richard Lee]

The question isn't whether 'music' has frequencies below 1/zillion Hz. 
 It's whether such content adds to the MUSIC.

This is the case ONLY with organs reproduced properly.

You can check this with DBLTs.  The speaker that has come out top in every 
single DBLT it has been in nearly 2 decades (some dozen tests in all) was a 
small 6.5 ltr reflex with 70Hz cutoff.  It was usually up against MUCH 
larger  expensive speakers.  No one has complained about its bass though 
some remark it's a bit down.  As we did ABC (33% chance if guessing) rather 
than ABX (50% chance) tests, the probability that this result is chance is 
extremely small.

There's also a lot of myth about how to design speakers for better defined 
 more energetic beat.  None of this comes out in DBLTs and this little 
ported box with terrible LF ringing always has comments like tuneful, 
well defined bla bla bass attributed to it.

Going from 70Hz to 40Hz cutoff will more than triple the size and cost of 
the speaker .. and to go to 20Hz will more than double that again.  You 
need to decide if you are getting any MUSICAL gains.

The organ example ... differentiating 'pressure in the head'  and 
'velocity' (trouser flapping) sensations is an important part of the MUSIC.

For percussion in modern music (??!), Bruce Willis destroying the universe 
 other 21st century stuff, single subs are acceptable as they are all 
'pressure in the head (and other parts of the anatomy)' sensations. 
 Distributed subs fed with a mono signal allow this over a larger area.

There's another myth that it's difficult to produce LF in small rooms.  If 
all you want to do is dinosaur footsteps et al, this isn't the case.  It's 
easier to pressurize a small room than a large one as you have to stuff 
less air in  out.

But a similar sensation is obtained by having someone beat you over parts 
of your anatomy with a blunt instrument.  In a large venue this is cost 
effective because the required staff are already there.  They are called 
bouncers.

What IS difficult is to produce Velocity (trouser flapping) at LF.  You 
have to MOVE all the air in the room.

I have no comment on mixes with loadsa stuff below 30Hz that overload the 
majority of playback systems without any MUSICAL benefit ... whether its to 
cheat a A-weighted meter or not.
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Re: [Sursound] ENVELOP - 3D Sound, on Kickstarter.com - SUB frequency range

[Richard Lee]

There is an important reason to maintain full Ambisonic capability to below 
20Hz.

It is a MUSICAL one.

Go to a service at a large cathedral with good organ.  At the end of the 
service, the organist will play something to show off while everyone is 
leaving.  You can walk around and listen at various positions without 
causing offense.

On SOME pedal notes at SOME positions, you will experience 'pressure in the 
head' sensations .. on other notes and positions the experience is 'trouser 
flapping'   Pressure  Velocity nodes in cathedral space

A mono recording or single high quality Sub converts the pleasant 
'trouser flapping' to unpleasant 'pressure in the head'.

You need at least stereo, preferably at least 4 subs in a classic Ambi 
system (and a TetraMic recording) to reproduce these important aspects of 
an organ performance.  This isn't about LF 'localisation'.  It's about 
pleasant musical sounds.

If the desired effect is Bruce Willis blowing up the universe, dinosaur 
footsteps or modern music (??), your single high quality Sub is 
sufficient.  You can also achieve the same effect by having someone stand 
behind you wielding a blunt instrument at the appropriate moment.
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Re: [Sursound] Brendan Hearne

[Richard Lee]

I'm so sorry to hear that.

In my younger days, I always thought of him as the handsome dashing young 
Prince of Ambisonia, a foil for Grand Vizier Malham.

Ken Farrar was Chief of the Dwarves with Clem Beaumont his resident 
Sorceror.

Michael was a Yoda type character with Geoffrey Barton sorta like Gandalf.
_

GV Malham, when are you going to put your true title(s), Grand Vizier  
Most High Poo Bah of the Grand Duchy of Ambisonia, Plenipotentiary 
Extraordinaire of said Grand Duchy to us unwashed Earthlings etc etc. .. in 
your signature?

Or is your mission on Earth still secret?
_

What is to become of Music From York?
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Re: [Sursound] New Ambisonic VST Plugins

[Richard Lee]

 It's certainly true that some decoding methods can become highly unstable 
when provided with an inadequate number of speakers and/or an uneven 
distribution.

 However, the Rapture3D decoders (including the TOA ones) do NOT have this 
problem. One of the core features of the Rapture3D decoder generator is its 
handling of arbitrary/irregular speaker layouts - a LOT of work went into 
this a few years ago. For the best quality output, you should feed these
decoders with the highest order actual signal you have, regardless of how 
many speakers are present - the more accurate information these decoders 
have, the better they can perform. They will make some use of higher order 
components for almost all speaker layouts, although the amounts vary (the 
only exception I can think of now is mono, where only the W/omni component 
is used).

Mr. Furse, are the Rapture3D TOA decoders and their development described 
in detail anywhere?

I have long thought that HOA should be a seamless step up from FOA but 
alas, losing my ability this Millenium to reed rite en kunt, I have had 
difficulties investigating this.

BLaH would like to steal ... I mean improve on your work.


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[Sursound] Core Sound TetraMic

[Richard Lee]

Just a short note about noise.

While the ST450 is slightly quieter than TetraMic, in practice, TetraMic 
noise has never been a problem when Recording Music .. even in very quiet 
studios.

Paul Hodges recorded his son in the Warehouse, a well known studio in 
London, so you can judge for yourself.

http://ambisonic.info/audio/paulhodges/others.html
http://ambisonic.info/tetramic/samples.html
http://ambisonic.info/tetramic/results.html

I wouldn't use TetraMic for recording distant bird calls in the Norfolk 
salt marshes .. but I wouldn't use a ST450, or even a Mk4, for that either 
8D

In case it isn't obvious, my completely biased opinion is that the Mk4 
Soundfield is the best microphone of the last Millenium while TetraMic 
holds that position in this Millenium.
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Re: [Sursound] Eigenmike

[Richard Lee]

 that's the main problem with the eigenmike: the higher the order, the 
 narrower the bandwidth and the higher the lower boundary. which means that an 
 eigenmike signal set will not usually be downwards-compatible - if you aim 
 for a correct amplitude response for the full 4th order set, it will sound 
 duller and duller the more orders you truncate.

Alas, the problem exists even at 1st order which is why Eigenmike is still some 
way from matching TetraMic for recording music.

The relevant paper from Angelo et al is 252-AES-40.pdf which shows their 
improved software compared to the MH offering.

A SPHERICAL MICROPHONE ARRAY FOR SYNTHESIZING VIRTUAL DIRECTIVE MICROPHONES IN 
LIVE BROADCASTING AND IN POST PRODUCTION - Farina, Capra, Chiesi e Scopece

AES 40th Conference, Tokyo oct2010

You'll note poor S/N (especially at LF)  response above 10kHz.
___

The use of omnis has some advantages (which Eigenmike doesn't take full 
advantage of) but also severe disadvantages.

For Sound Intensity, Angelo has some papers comparing various methods to the 
time honoured BK Sound Intensity wand  the new fangled Microflown (??).  A 
properly calibrated soundfield like TetraMic has many PRACTICAL advantages 
including PRACTICAL accuracy and allows 3D measurement too.  One might argue 
the directivity patterns of TetraMic are not 'perfect' at HF ... but neither 
are the those for the wands and other methods too.  And those for TetraMic are 
predictable and can be compensated for a measurement application.

The BK wand is practically useless below 100Hz even with the largest spacer 
cos S/N.  Accuracy at HF is limited, even with the smallest spacer.  You'll 
have to dig up Angelo's papers for chapter  verse.

I don't think I'm letting the cat out of the bag by revealing at least 1 
company is planning a Sound Intensity instrument based on TetraMic.


While trying to dig up that paper, I came across some typical Angelo gems. In 
...

RECORDING, SIMULATION AND REPRODUCTION OF SPATIAL 
SOUNDFIELDS BY PCM SPATIAL SAMPLING - Farina et al
International Seminar on Virtual Acoustics VALENCIA, November 24-25, 2011

.. he calls the system at 

Casa della Musica, Parma as a '.. much worst playback system ..' 

preferring the 

duodecahedral (maybe even higher order) system of Nelson  Fazi at ISVR. 

... the shape of things to come ... thought the wife might have something to 
say about the furnishings. 8D
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Re: [Sursound] Eigenmike

[Richard Lee]

I must agree with Fons on Eigenmike.

IMHO, Angelo's software for Eigenmike is loads better than what's supplied by 
the makers.  Its nea...aarly good enough for recording music  but certainly 
not there yet.  There's a paper on the new software on Angelo's website.

I'm not sure if there is a sensible (in Ambi mike terms) calibration procedure 
for it.  The makers claim you just stick a mike calibrator on each capsule.  
Excuse me if I ROTFL.
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Re: [Sursound] Patent application: Data structure for HOA

[Richard Lee]

 Unless things have changed a lot, last I checked lossy compression messes up 
 phase relationships, and that would be an issue for things like UHJ, which as 
 long as portable stereo players with limited battery life (and thus limited 
 CPUs), is the only viable, because stereo compatible, distribution format.

 At this point in time, not only is most music listened on mobile devices, 
 most music is even purchased on mobile devices, and that's strictly a stereo 
 (or maybe binaural) world.

Try this simple experiment.  Take your favourite Nimbus UHJ CD and rip it using 
the most evil MP3 encoder you can find .. probably the one built into the 
latest Windoz Media Player.

Do this at 256kB/s and also (shock!  horror!) at 128kB/s.  Now listen to the 
resultant files on a mobile device.  Then you can pontificate to us on how the 
musicality has all escaped and no one is going to find these acceptable.

You can also rip to a WAV file if your mobile device will play these and 
compare the MP3s with the 'original'.

This is just testing Ronald's assertion about compressed UHJ on stereo mobile 
devices.  I dunno about full UHJ surround decode cos there don't seem to be any 
good ones in the public domain.

PS  I expect you to hear ve.eery slight differenes with one MP3 and 
probably none with the other.  I won't insist on Double Blind bla bla but you 
might find that educational.
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Re: [Sursound] Patent application: Data structure for HOA

[Richard Lee]

As I've said ad nauseum, the guy who first integrates an Ambi decoder into VLC, 
getting around the evil Windoz mixer etc. gets to choose the data structure for 
next important Ambi format.

This will be a lossy compressed format probably based on the public domain 
Vorbis.

Ambisonia was the 1st major breakthrough, source material.

I hope everyone is aware and thankful for Etienne's huge amount of effort  
work.  We are lucky that York have taken over this and hope it will continue to 
increase and prosper.

The 2nd is the ongoing decoder work by BLaH and others on this forum.  At least 
the theory of how to design a good decoder is available.

Sadly, only Fons' Happy Days decoder is flexible enough to take advantage of 
the new work.  This isn't descrying the decoders available from D McGriffy, 
Richard Dobson, the York Mafia  others; just pointing out that they are 
'fixed'.

But none of these will conveniently play MP3s ripped from CDs or youTube, 
surround videos etc  .. no nice database for music .. so will remain niche 
interests.  Happy Days doesn't even run on evil Windoz and probably never will 
if its inventor has any say in it .. 8D

I finally managed to compile VLC this year but can't seem to do it again.

8(

When Ambi VLC happens, I predict the re-surrection of UHJ.  Simple 2 channels 
will remain the most important distribution format in the forseable future.

But it will pave the way for HOA and other exotics.

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Re: [Sursound] Spherical speakers ?

[Richard Lee]

 I mean that instead of a cone there is a sphere, and able to emit sound in 
 all directions I'm guessing if it hasn't been done its because it presents 
 considerable technical difficulties - still ...

There are no serious difficulties as long as size  $$ are not constraints.

But apart from acousticians measuring hall acoustics, why do you want such a 
beast?  Are you one of those who believe omni speakers give better stereo, 
sound etc ?
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[Sursound] CF spectrum vs L R in stereo etc

[Richard Lee]

.. was Somting for the Weekend - Commerisal 3D sound

 Sure, the center phantom image generated as a sum of two identical L/R 
 signals sounds a little different. But little is the operative word. ... Most 
 people do not notice this at all.

It's not that we don't notice but that we are so used to it that we think it 
'natural'.

Most people who hear a Classic Ambi rectangle rig for the first time with the 
Super Stereo decodes, suddenly become aware of this and it is irritating.

That's not to say, Ambi decode is faultless; just significantly better than 
stereo in this respect.  The senior members of BLaH are investigating this.

 I say most people because a small number of people hear individual 
speakers no matter what you do.

Michael told me that up to 10% of the population 'suffer' from this.  I didn't 
believe him but looked up his reference which confirmed it.

I've forgotten the reference so would appreciate anyone who knows this.

From nearly 2 decades of Blind Listening Tests, I can confirm this is about 
right for one small esoteric population ... UK HiFi reviewers.

These people are not deaf but in fact, more discriminating than the unwashed 
masses.  They are not easily fooled.

 I think the speakers are audible in stereo because of the detent effect. If 
 you keep the images between the speakers where they belong, most people do 
 not hear the speakers at all.

Actually keeping images ONLY between the speakers is what draws attention to 
them.  This is one advantage of Blumlein fig-8s.  The soundstage and reverb 
extends seamlessly beyond the speakers.

The most unstable phantom image is CF and sources close to it .. especially 
with wide speaker spacing.

Easy to test all this for yourself if you have a mixer with panpots .. or even 
a preamp with mono button  balance control.  Try it with noise as well as 
music.
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Re: [Sursound] Somting for the Weekend - Commerisal 3D sound

[Richard Lee]

 well, depends. iirc, theile's argument is that a two-speaker phantom source 
 should be a mess in terms of spectrum, but isn't (as two-speaker stereophony 
 demonstrates). so for some reason, the brain is able to sort it out. more 
 than two correlated sources, and things go awry, e.g. L/C/R 
with too much crosstalk is a pitiful mess.

Err.rrh!  Actually two speaker stereo IS a mess in terms of spectrum.  Just 
compare a mono signal panned to CF with it panned to hard left or right.  It's 
one of the things which draws attention to the speakers  spoils the illusion.

One reason for the seamless performance of 1st order Ambi is that, even with 
just 4 rather unevenly spaced speakers, it alleviates this effect and helps 
make the speakers disappear.
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Re: [Sursound] Question about plane waves, and the precedence effect

[Richard Lee]

 except its not quite the same effect. If I hear a plan(ish) wave in nature
such as in thunder or a very distant giant waterfall

Distant thunder is often an extreme case of Proximity effect. Velocity
components are boosted at LF compared omni components.

You encounter this if you are using TetraMic or properly aligned Soundield and
there is a distant thunderstorm.

Sometimes you will see XYZ, the Velocity Components go wild when nothing is
happening on W, the Omni bit. Usually you can't hear anything either but can
time the 'happenings' many seconds after the lightning.

This is in spite of built in 24dB/8ve electrical filtering and effectively
36dB/8ve acoustic HP filtering.

On TetraMic, the variable HP filter is usually good to get rid of this effect.
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Re: [Sursound] Super Stereo emulation (UHJ)

[Richard Lee]

No.  Ambi Super Stereo isn't just feeding the stereo into a UHJ decoder.

This is a post from 2006.  I'd recommend a Width control so you can decide 
how wide you want your stereo.

All the decodes here are basically the same with different Width settings.  
Only minor differences, with preference in Barton 82.

I distinguish between WXY which is strict FuMa and W' X' Y' which already have 
Shelf Filters applied.

Stereo, SuperStereo use the same Shelfs as FuMa. ie normal B-format Shelfs as 
described in BLaH3.

UHJ will need different ones which are in Fig 3.3 SHELF FILTERS for Ambisonic 
Decoders
_

Michael's solution circa late 80's and used in most Ambisonic decoders of that 
period.  MINIM

 W  = 0.3667 M + j 0.3586S  M = L + R   S = L - R
 X  = 0.529 M   - j 0.404 S
 Y  = S

From examination of the simple Minim circuit on RKRs 
http://www.geocities.com/ambinutter.  Strict B-format

The next 2 are from 
http://www.geocities.com/ambinutter/UHJ_and_Ambisonic_equations.html

Two approaches, from WW 1977INTEGREX
 
 W' = 0.717S - 0.291jD  1.23 M -  j 0.4991 S Stereo
 X' = 0.717S + 0.291jD  1.23 M + j 0.4991 S
 Y' = 0.583DS
 
W' X' Y' are feeds to the Speaker Matrix at HF after Shelf filtering as noted 
in the 1977 Integrex article.  I also show this normalised to Y' = S for 
comparison with the Minim.  NO FURTHER SCALING REQUIRED.

 W = 0.65S - 0.27jD(a /180) 0.8667M - j 0.36 S   Super Stereo   KEATING
 X = 0.98S + 0.40jD(a /180) 1.307M   - j 0.5333 S
 Y = 0.75D(a /180)  S

 where:
 S = Left + Right
 D = Left - Right
 a = Width control, varying between 0 and 180°
 j = 90° phase shift

Yes probably from The generation of virtual acoustic environments for blind 
people by D A Keating at the University of Reading.  I took this straight off 
RKRs web page so don't know if it's Strict or HF.  Probably Strict.

I presented these 3 cos they were different and seemed to represent Michael's 
thinking over time, 1977 Integrex, Superstereo Keating?, then 1980s Minim.

 Geoff Barton 1982
 
 W = 0.6098637*S - 0.6896511*j*w*D  0.36253 M  - j 0.40996 S w
 X  = 0.8624776*S + 0.7626955*j*w*D 0.51270 M + j 0.45338 S w
 Y = 1.6822415*w*D - 0.2156194*j*S  S w  - j 0.12817 M
 
 where 'w' is a width setting, the optimal (in some senses) value of which is 
about
 0.593, S  D are as defined above. NB, in MAG's notation W'' etc is the signal
 after the shelf filter, W before.

Strict B-format.  For w = 1, this is very similar to the Minim, except for 
the j 0.12817 M term which is a bit of Forward Preference.  But adding this 
means it doesn't drop to strict mono when w = 0.  You could consider the 
Minim a simplified version of 1982.

Applying w = 0.593, moves it nearly towards the Keating and 1977 Integrex 
decoders.

Correct value for w, a, width etc. depends on what angle you think a stereo 
stage should extend.  Opinions directly to Robert Greene please.

Can anyone report on listening tests for any of these?  eg like Steve Thompson. 
 I played a bit 20+ yrs ago but seem to remember the 1977 Integrex was as good 
as anything fancier.  Work really well on Blumlein.  Show spaced omnis as crap.

Richard L

PS  I've taken the liberty of editing the above to use W X Y ONLY for 
strict B-format.

PPS The above encodes (except for the Integrex) are suitable ONLY if the 
decoder uses full Shelf Filters.  If not, you need to use Energy playback.  
see SHELF FILTERS for Ambisonic Decoders from ambisonicbootlegs.com


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Re: [Sursound] any idea's on the background of the stereo equations in

[Richard Lee]

You can't make naive predictions on what ambi stereo decode will sound like 
that; especially when you have j terms.  You have to apply the full General 
Metatheory ...

http://www.aes.org/e-lib/browse.cfm?elib=6827   It should be in Sampo's 
motherlode too.

- Decode the W' X' Y' signals to a speaker layout.  A square is convenient for 
horizontal super stereo predictions.
- Apply Michael's equations to get what the listener perceives in the centre.

- Do the same at HF according to the Shelf filters.  These are particularly 
important for 2 - stuff.

Only then will everything make sense according to the theories of Bauer, 
Blumlein ..Franssen ... Strutt, Vanderlyn etc.

Note General Meta doesn't use WXYZ in exactly the same way as traditional Ambi 
theory so be warned.

I did a lot of this stuff in Jurassic days, trying to find a better stereo 
decode but had to admit the WW 77 equations were as good as any and better than 
most.  There are small differences in some Geoffrey Barton stereo decodes to do 
with some added Forward Preference.  This improves 'phasiness' performance for 
front centre sources but doesn't change the perceived directions.

I've applied General Meta .. to 'normal' 2 speaker stereo in the design of my 
Wharfedale Option 1  2 speakers.

 I have a question concerning the conversion of a stereo recording to
 ambisonics. In the Wireless World of 1977 articles there are equations for
 the
 conversion of stereo to W,X,Y equivalent signals. The use of j*Diff term
 doesn't seem to make sense. As I found no background info on them, maybe
 someone in the group has.
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Re: [Sursound] OT: Spatial music

[Richard Lee]

 can a tetrahedral mic be used to create a room (correction) impulse response 
 in B format? and how?

Yes.

I can make a sensible attempt today for an Ambi rig spaced away from the walls 
as the HiFi pundits and other gurus have mandated for years.  This however has 
near zero Wife Acceptance Factor.

What i can't figure out is how to EQ for speakers mounted on or close to a 
wall.  (Unless the speakers have been designed to work well in such positions.  
eg from the Unobtainium Speaker Co.)

This is necessary to move towards Jeff's integrate in-wall loudspeakers at 
pre-specified locations (including ceiling) except the Supa Ambi Decoder 
doesn't need pre-specified locations.  It measures the speaker positions using 
the TetraMike.

I think Angelo has tried the 1st method using some naive strategies; just EQing 
WXYZ to get matching WXYZ from a Soundfield.  This doesn't give very good 
results cos in speaker / room EQ, what you DON'T EQ is probably more important 
than what you do.

Perhaps Fons knows more.

You need some strategy like what's used in Dennis' Digital Room Correction but 
taking into account multiple speakers and B-format.

I'd dearly love details of what Trinnov do.

If I ever get to grips with 21st century programming tools, I intend to do some 
work on this so expect results before the end of the millenium.
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Re: [Sursound] question about simple equation

[Richard Lee]

 cardioid = 1/2 ( sqrt2 W + X )

 but for Mid-Side decoding from B-Format, where M is the same cardioid 
 component, I often find this equation:

 M = sqrt2 W + X

This is an instance of a patented Microsoft software invention called a Volume 
Control.  There are also ancient hardware implementations of this useful device 
which can be used to avoid clipping by those versed in the art.  

Also to generate clipping where this is a desired effect.  It is sometimes 
called a Gain Control.  Both positive  negative gain is possible.

 writing the equation inside max6 I had several clipping problems with the
decoding of a Soundfield ST450 signal, that has four matched line under single 
knob.

The ST450 uses such a hardware implementation I designed circa 1980 for the Mk4 
Soundfield.  Experiment to find the best settings for you.

It is possible to have several instances (both hardware  software) of these 
devices to optimise the dynamic range and clipping performance of a complex 
signal processing chain at various points.

 I record the b-format in a single quadro file.

If you do all recording and processing in 64b Floating Point, you may be able 
to avoid infringing Microsoft patents on the Volume Control.
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Re: [Sursound] ST450 capsule assembly

[Richard Lee]

Svein Berge shows the ST350 capsule assembly at

http://harpex.net/

My thanks to him.  I want to know if this new stronger assembly, with the 
capsules secured by screw on rings, is used in the ST450 or if they have gone 
back to the old assembly using screws as in the Mk4, Mk5, ST250, SP422.

David, your Mk3A may have an even more skeletal version of the Mk4.

http://www.soundfield.com/soundfield/soundfield.php
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[Sursound] Gilbert Briggs Wharfedale

[Richard Lee]

http://www.impublications.com/content/a-pair-of-wharfedales

I never met the Old Man but still cherish a letter in reply to a query of mine 
on how to enter the British loudspeaker industry.

Like all who had even the slightest contact with him, his warmth, generosity 
and encouragement were never forgotten.

I was RD Manager for more than a decade and kept coming across old letters 
which showed the high regard in which he was held throughout the world.
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Re: [Sursound] UHJ decoding and shelf filters

[Richard Lee]

http://ambisonic.info/info/ricardo/shelfs.html

This is now only of historical interest.  My latest thinking on Shelf Filters 
is encapsulated in BLaH3 Is my decoder Ambisonic? where Aaron's clarity puts 
me to shame and explains why I'm still a beach bum.

The only useful info there which isn't in BLaH3 are the UHJ Shelf Filters.  
(There's loadsa now useless info)

Fig 3.3 shows them in the form that Ambdec uses; matched gain at LF.

So 'order gain' at LF is 1.0 for both.

At HF, 'order gain' is 

3.7dB for W
and
-2.1dB for X  Y

This is exact for regular horizontal speaker layouts.  (Actually I'm just 
reading off my Fig 3.3)

In theory, rectangular layouts should have slightly different values but these 
are known to work well subjectively.  I daren't ponitificate on Guru Fons' 
individually matched Shelfs for his 5.1 layout [1] but suggest these might work 
well too.

You'll excuse me if I dodge the Shelf Filter for Forward Preference question. 
 My brain hurts.

[1] I think he derives another matrix at HF by an iterative process.  BLaH4 is 
our attempt at this for non-regular / non-diametric opposite layouts.

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Re: [Sursound] UHJ decoding and shelf filters

[Richard Lee]

 The actual shelf gains (the difference between LF and HF gain for any 
 component) don't tell the real story. Changing the LF or HF gain for all 
 components by the same amount doesn't modify the decoding, it just results in 
 some EQ.

Du...uh!  I stand corrected.  Shelf filters change the ratio of Pressure (W) vs 
Velocity (XY) so the different gains must be incorporated in the matrix.[1]

Does anyone have Fons' 5.1 matrices handy that they could send me?  I can't 
promise anything quick as its 35C in the shade and 100% RH in Cooktown and my 
small brain is boiling.  It's also 3 decades since I did this stuff.

 Ambdec doesn't use shelf filters, it uses crossover filters and two 
 independent matrices. So all the above is assuming that the two matrices are 
 the same. They usually are for regular layouts, but not in the case of the 
 5.1 one. 

Shelfs.zip describe how to use crossover filters for Shelfs and attribute this 
to Fons.  BLaH3 does the same but attributes this to Geoffrey Barton cos he 
first suggested it.  We are going to use the term Shelf filters for any phase 
matched filters, whether IIR or FIR, used to adjust the ratio of Pressure vs 
Velocity in an Ambi decoder.  I can't see any point in using FIRs these days 
unless you have a soft spot for them like Angelo.  8D

Fons' approach having separate LF  HF matrices is purer.  You have to do 
this for irregular layouts like the evil ITU 5.1  The ideal HF matrix for 5.1 
is still open to debate.  There is at least Fons, Bruce Wiggins, the 
Huddersfield Choral Society[2] and BLaH4.

BLaH4, Design of Ambisonic Decoders for Irregular Arrays of Loudspeakers by 
Non- Linear Optimization Heller et al, preprint 8243 AES San Francisco nov10, 
discusses the various trade-offs and presents our favoured method.

Roger, you might want to use a rectangle, other diametric-opposite, or regular 
(pentagon, hexagon, octogan etc) layout and sit in the centre.  The LF  HF 
matrices will then be very similar and it will be easy to work out what needs 
to be done to get the 5.8dB difference.


IMHO, the surround community needs to have another look at UHJ, especially 
simple 2 channel BHJ.  The expanding music market is simple 2 channel stereo 
driven by iShops  other iThings, not multi channel surround.  Any dedicated 
compressed Ambi format needs to make efficient provision for UHJ.  My Ambisonic 
Surround Decoder (due out before the end of the millenium) will decode UHJ 
properly as well as Dolby Digital zillion.1 better than their native decoders.


[1] Double Du...uh!   order_gain DOES change the matrix.

 If you look in the preset file, there are two matrices defined, one for LF 
 and one for HF. There are also 'order_gain' values for each matrix. Only the 
 first two of these matter (the others are for higher order). The first is 
 applied to the W column of the matrix (the first column), the second to X and 
 Y (the second and third column).

If I read this correctly, you CAN use order_gain for a Shelf filter effect.  
I'll suggest trying an extra 1.95dB for the 1st (W) order gain and an extra 
-0.85dB for X  Y with Fons' 5.1 decode at HF

[2] Moore D, Wakefield J The design of ambisonic decoders for the ITU 5.1 
layout with even performance characteristics  124th AES  Convention May08

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Re: [Sursound] SoundField: from Calrec to MBHO, what year?

[Richard Lee]

1989 was when Steve Jagger, Ken Farrar and some of the other original 
Calrec mob bought the company back from AMS.  They left the Soundfield with 
AMS.

AMS had problems making the mike consistently.

When Soundfield Research took over, they dug Lynne out of retirement.  She 
used to make the Soundfield capsules for me.  Though no longer a Yorkshire 
virgin, she brought at least the capsule production back to its former, 
Calrec, standard.

When she decided to retire for good, Soundfield Research transferred the 
production jigs to MBHO.  This happened quite recently; 2006?

MBHO do a decent job with the capsules but final assembly  test is still 
done in Wakefield.  MBHO capsules are the only ones with gold flashed mylar 
diaphragms.  Apart from that, the capsule is exactly as I left it circa 
1983 warts  all.

The SP200 uses a MBHO subcardioid capsule.

Info via YorkshireLeaks ..

Why do you ask, Daniel?

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Re: [Sursound] Questions regarding a very early 'Ambisonic' LP

[Richard Lee]

The change to UHJ at that time is very small and was made just to get rid 
of the need for another phase shifter in the decoder (IIRC).

Any of the Classic Decoders, the Integrex, Minims, AD etc will decode this 
properly.

I for one would be most interested in a copy of the recording as it 
represents Calrec before I was a padawan.  The people who recorded it were 
my mentors at Calrec and have passed on.

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[Sursound] the recent 2-channel 3D sound formats and their viability for actual 360 degree sound

[Richard Lee]

There's loadsa good stuff being discussed here.  If I can comment on just 
one or two 

 When listening to this through a speaker rig, we hear this boost and tend 
to interpret it as meaning the sound is close especially in a dry acoustic 
with a Greene-Lee head brace etc., etc.,. However, surely (unless I am 
being more dense than usual tonight) this is a learnt response based on the 
behaviour we have heard from directional mics? After all, taken 
individually, at those sort of frequencies our ears are essentially 
omnidirectional and not subject to bass boost (to anything like the same 
degree).

You can test this.  Blindfolded your victim, creep up to him silently  
whisper in one ear.  He will report significant proximity effect.  Eric, 
Duda  Marten will explain this happens cos our lugholes are spaced and 
form a crude (?) left/right velocity sensor.


Robert, Guru Fons has explained with his usual precision some of the 
limitations of the soundfield.  The important question, however, is, what 
effect does all this have on the listener in the middle of a Classic Ambi 
system?.  The answer to both the whisper in your TetraMic/Soundfield and 
motorcycle examples is closeness is exaggerated.

If you've got any Soundfield or TetraMic recordings where you've had to 
place the mike near or surrounded by the audience, the chap unwrapping 
sweets, rustling his programme or coughing always sounds as though he is 
directly next to the mike.


Mr. Hunt, I hope Sampo  Fons have been sufficiently enlightening.  A 
Classic Ambi rig or soundfield mike has no concept of a unit circle. 
 They record  present distance as presented to them.  The mike cos 
Helmholtz etc and the Classic Ambi rig with tricks like NFC.  Even simple 
1st order Classic Ambi rigs with NFC do a good job at plane wave 
reconstruction at LF.

For a synthetic source to replicate this serendipitious situation, you have 
to

1)  Add proximity for close sources or motorcyles as the Encoding Eqns in 
Appendix of Is my Decoder Ambisonic?  This is the most important (only) 
cue available for close sources in anechoic conditions
2)  Add a suitable reverb pattern as MAG's Distance Panners.  You need to 
do this not cos 1) dun wuk but cos real life distance perception is 
TERRIBLE under anechoic conditions.  Ambisonics is probably the best I am 
there system cos it's isotropic nature reproduces reverb and other diffuse 
fields 'accurately'.  This was one of MAG's obsessions, even with stereo.

3)  For very far souces, you might want to add HF absorption etc but this 
is 
probably out of the realm of the sources you want to simulate.


On heretical NFC HOA, there is a far simpler encoding system which avoids 
overload at LF and doesn't assume any speaker rig.  ie pure encoding.  This 
is to treat encoding as you would if you were designing a HOA mike.

For this, you MUST roll-off the LF at higher orders cos S/N.  This 
conveniently avoids the LF overload too.  But in any real playback 
situation, you only need HOA at HF so you don't actually lose much useful 
info.  All you have to do is to ensure the phase response of your LF 
roll-off is replicated by appropriate all-pass networks in the lower 
orders.  There's no need to even specify the LF roll-offs.

Building NFC into encoding (ie NFC HOA) should be a de-pinnaeable offense. 
 Keep encoding  decoding separate  clean please..

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[Sursound] Book on Michael Gerzon, audio pioneer and music recordist

[Richard Lee]

Steve Thornton is selling a coffee table book based on his encyclopediac 
photographic record of OUTRS and other stuff.

Be the first to brag a limited edition hardcover link which goes back to A. 
Blumlein   Loadsa stuff on the rise  fall of quad, surround and the 
early history of Ambisonics, the Soundfield Mike ...


Book on Michael Gerzon, audio pioneer and music recordist

A personal account of early work and experiments, with words and pictures 
by Stephen Thornton.

Michael was a remarkable character whose deep contributions to audio are 
now becoming more widely appreciated. This is a non-technical account aimed 
at bringing his early work to a wider audience. It contains descriptions of 
his published articles and a blow-by-blow account of experimental sessions, 
illustrated by dozens of contemporary photographs. These give a fascinating 
insight into a unique individual.

There are no royalties to the author: the price you pay is the cost charged 
by the publisher. Order the book from Blurb at 
http://www.blurb.com/bookstore/detail/1356012. You can turn sample pages to 
see what it looks like.

Chapters: Introduction; Early Days; Surround Sound from Two-Channel Stereo; 
The Dolby Experience; Microphones; Tetrahedral Recording; Commercial 
Quadraphonics; BBC Quadraphonics; Electronic Music Concert; Ambisonics; 
Michael the Man.

Large format: 12 inches (30 cm) square, 84 pages, 146 black-and-white and 
colour photographs.

I hope you will want to have this permanent record of Michael's work.

Stephen Thornton

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[Sursound] HD Harwood: Stereophonic Image Sharpness, WW vol74 1968 p207-211

[Richard Lee]

Anyone got a copy of

HD Harwood: Stereophonic Image Sharpness
(Wireless World, vol 74, 1968, pp.207-211)  ?

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Re: [Sursound] Question regarding UHJ Encoding - Decoding

[Richard Lee]

 This sqrt(2) factor is an endless source of confusion. It seems silly that W 
 should be divided by sqrt(2) in recoding to restore its value, which was 
 multiplied by this in the encode to UHJ.

This sqrt(2) factor is WRONG.  Where do people come up with such myths?

The Encoding eqns on http://en.wikipedia.org/wiki/Ambisonic_UHJ_format are 
correct and come from Gerzon himself.

Ambisonics in Multichannel Broadcasting and Video. JAES nov85, 33 (11): 
859-871.

The Decoding eqns however, though supposedly coming from the same source, add 
an erroneous 1/sqrt(2) factor.  They are wrong.  If you take this abomination 
out, you get ...
___

Formal UHJ Decode for 3/4 channels.

S = (Left + Right)/2.0
D = (Left - Right)/2.0

W = 0.982*S + j*0.197(0.828*D + 0.768*T)
X = 0.419*S - j(0.828*D + 0.768*T)
Y = 0.796*D - 0.676*T + j*0.187*S
Z = 1.023*Q
___

.. a formal decode, the Inverse Matrix of the Encoder which recovers WXYZ 
exactly.  This is what's shown in Gerzon85

Richard, the signs of j come out from the mathematics of the complex Matrix 
Inversion.  The test is that if you encode  decode, you should recover the 
original WXYZ (apart from small roundoff errors due to limited precision of the 
coefficients)

For 2 channel UHJ, because WXY cannot be recovered exactly, there is a range of 
good decodes with different properties.[1]

The one shown on wikipedia is supposedly from Gerzon85 but, because wiki has 
added a erroneous 1/sqrt(2) factor, will give very poor results.

Richard, will you reveal the source of your heresy.  You are granted immunity 
from de-pinnaeing if you talk.

[1]   The optimum 2 channel decode is tied in with the design of the speaker 
decoder which derives the signals to feed the speakers.

BLaH3 ...  Is my decoder Ambisonic? Heller et al, AES San Francisco 2008  
 is the definitive reference for design of Classic B-format decoders.

However, 2 channel UHJ decode has other complexities which are yet to be 
explained clearly and accurately.  Expect a BLaH encyclical on the subject 
before the end of the millenium.

In the meantime, let's stamp out incorrect, inaccurate and misleading info.
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[Sursound] Question regarding UHJ Encoding - Decoding

[Richard Lee]

I have questions about UHJ decoding too.

Richard, how did you get your decoding eqns?

My question is about 

http://en.wikipedia.org/wiki/Ambisonic_UHJ_format

It has separate decoding eqns for 2  3 channel UHJ.  They are consistent 
except for Y

Y = 0.763*D   + j*0.385*S   2-channel
Y = 0.796*D - 0.676*T + j*0.187*S   3 channel

It's been 30+ yrs since I pretended to unnerstan dis stuff but if IIRC, 3 
channel decoding was supposed to be robust to the T channel. ie if the T 
channel faded out, decode would revert seamlessly to the 2 channel eqns.

The idea was that the Velocity  Energy Vector directions would remain the 
same but T would increase the Vector Magnitudes.

I thought I had this scribbled somewhere but all I can find is stuff for 45J 
which is UHJ's immediate predecessor and shows how long ago this was.  I worked 
on the Calrec version of the Transcoder which was UHJ but I can't remember 
doing any UHJ work on the Calrec Ghetto Blasters which was Ken Farrar/Ellis's 
Periphonic System(s).  I did their speaker matrix stuff though.

And where do these eqns come from?  I can't remember a formal 
document/published paper where MAG or Geoffrey Barton state the UHJ Encoding  
Decoding Eqns.

One coudl take the attitude that only Encoding needs to be specified and the 
user can do what he likes on Decode.  Simple Matrix Inversion would give you a 
formal 3 channel Decode but that doesn't help me cos kunt kont or reed en rite 
.. 8D

Martin, when we resolve this issue, or even if we don't, we should put 
something in wikipedia stating the source or else something like

there is no formal source but we have this from Dr. Geoffrey Barton 
or 
 we got these from reverse engineering the Audio  Design Trancoder  Decoder.

I have been unsuccesful in my application to become a wiki editor cos kunt 

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[Sursound] Mark Decker The Bells!

[Richard Lee]

IIRC, Mark Decker recorded a BBC radio drama where a man is killed by being 
tied up in a bellfry with the bells pealing away.

Can't remember the title but I remember reading a web page where he describes 
this experience which was his first use of a Mk4 Calrec Soundfield.

I'd appreciate a link to this page if anyone is better at searching.

Also if anyone has a recording of the actual thing 
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[Sursound] distance perception in virtual environments

[Richard Lee]

You must simulate at least 2 things.

At close range, you must simulate the curvature of the soundfield.  This is 
simply proximity for 1st order and the effect is, if anything, exaggerated.  
see the Appendix of

Is My Decoder Ambisonic, Heller et al, AES San Francisco 1980 aka BLaH3

See Daniel for HOA

You have to simulate early reflections and a reverb pattern appropriate to 
source distance.  MAG has a paper on this under Distance Panners from an idea 
by Peter Craven.

Real Life Distance Perception is TERRIBLE under (near) anechoic conditions.  I 
recorded Paul Robinson's band at the IMAX theatre in Bradford.  They were 
providing music for a festival of silent movies.  Even after 5 days, we still 
found it disconcerting in that very dead environment.  Someone would call you 
from the door 20m away and you thought they were beside you.


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[Sursound] distance perception in virtual environments

[Richard Lee]

I hope you have a control where you measure real distance perception too.

Not having a real control is a fault in many localisation experiments.
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Re: [Sursound] horizontal-only decoder design for line sources

[Richard Lee]

 [1] Loudspeakers and the Stereo Seat - G Millward, HFN, 1981?

 It's 

 Millward, G; Loudspeakers and the Stereo Image, Hi-Fi News 29, nov84, 

 according to 

http://decoy.iki.fi/dsound/ambisonic/motherlode/source/AmbiLoc2.pdf

 by a certain R.L and A.H. ;-)

Thanks to Grand Vizier, Don Malham, for this correction. I'm in the throes of 
yet another computer hard disk recovery  that's my excuse.

 ...  to donate a bunch of good quality line sources ...

Write a begging letter to the MD of the Unobtainium Speaker Co. Ltd.   ...   8D
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Re: [Sursound] horizontal-only decoder design for line sources

[Richard Lee]

I think this is a virgin field of research.

On the practical design of line sources for large venues, it may be that we 
can only achieve ..more even coverage, as you increase the distance from a 
speaker system to the nearest listener. This also makes it rather more 
comfortable for the nearest listener.

Having had some success with designing stereo speaker for a larger listening 
area[1], my initial thoughts on speakers for domestic ambi were -- design for 
mounting near the ceiling with directivity such that maximum loudness would be 
on the opposite side of the room.  All combined with (at the time, 
non-existent?) digital room  speaker EQ.

But this discussion seems about much larger areas and apart from my crude 
efforts, no one else seems prepared to report their experiences.

I note a deafening silence from the York  Derby Ambisonic Mafia who probably 
have more experience with large area playback than the rest of the world put 
together.

 that's not really true. the sweet spot in the strict ambisonic sense is a 
 function of order and wavelength only.

That's only strictly true for LF where Ambi is a (crude?) form or WFS ie when 
it attempts to recreate the soundfield at a point.  An IEC Listening Room,  2.5 
x 6 x 4m is a point at 20Hz.

While MAG's rE appears to predict what happens when this no longer holds true 
on a domestic scale , we don't know how listeners react to large area Ambi 
soundfields.

Please do pontificate on this subject but lets hear from those who have tried 
stuff out so we can test these prophecies.

You probably do need special decoders for horizontal ambi with line sources, 
but what these need to do is a new field.  I can only report from my small 
experiments, that a normal Classic Decoder seems to work well.

 that's another thing worth exploring: the funktion one guys have reported 
 that ambi rigs have an advantage in this respect, because the ratio of useful 
 loudness inside to leaked emission outside of the array is better than with 
 stereo (or maybe even conventional four-point) playback systems.

This is very obvious even for domestic ambi systems.

As well as the very real sense of something real happening within the circle of 
speakers when viewed from outside.  Peter Lennox thinks it's just glorified 
stereo but the effect is so startling that it must be more than that.

[1] Loudspeakers and the Stereo Seat - G Millward, HFN, 1981?
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[Sursound] horizontal-only decoder design for line sources

[Richard Lee]

When I was RD tea-boy at Wharfedale, I spent time trailing around the country 
with Barry Fox our Promotions Manager, putting on lectures/demos  shows.

Part of the stuff we carted around was the Wharfedale Ambi rig which was

- 12 x TSR110 speakers 
http://www.gramophone.net/Issue/Page/October%201980/126/792093/Wharfedale+Laser+200+and+TSR+110+Loudspeakers
- A souped up Integrex decoder optimised to run a hex system using only 4 
channels of decode and amps
- Calrec Soundfield Mk3a controller
- Rotel RB5000 amp. 100W/channel and still my all time favourite.
- ANOther high powered amp which I can't remember.
- Teac 3440 + stack of Dolby Bs

all to play my Soundfield recordings and some UHJ stuff.  Horizontal only.

The speakers were arranged as top  bottom units with the top sitting upside 
down on the bottom; each precarious pair mounted on quite a tall stand to get 
the speakers above the seated audience.  The idea was to create a line source, 
short at HF where the treble units were close to each other while the 4 bass 
units spanned nearly 1.4m vertically. Our biggest crowd was more than 150 in 
Guildford (IIRC), a foreign city South of Watford.

The huge amp was to implement the system MAG described in the Integrex article 
to use 4 amps to drive 6 speakers in a regular hex.  'W' needs to be nearly 4x 
the power of the other Amps.  I describe a low level version for multichannel 
soundcards at http://ambisonic.info/info/ricardo/decoder.html at the end of the 
section on Classic Ambisonic Decoders.  MAG told me (to my surprise) that it 
was the first practical trial of this method.

The most successful part of the demo was making the soundstage revolve round 
the audience with the Mk3a controller.  Oh.  And the whole she-bang sounded 
good too especially the stuff recorded in Bradford's St. Georges Hall; Sir 
Thomas Beecham's favourite venue.

The biggest problem was making sure all speakers were connected up properly and 
in phase. The supa dupa Ambisonic Surround Decoder MUST have built in 
facilities to facilitate this.
___

The idea behind line sources was to increase the sweet spot.  To some extent, 
the Wharfedale rig achieved this; good sound and localisation even on the 
fringe of the large audience area. A line source avoids (?) the 1/r attenuation 
you get with a point source.  But we also put the speakers as far from the 
audience as possible which might have been a bigger affect.

Conclusions

- I don't think you can get proper line sources unless they span from floor to 
ceiling.
- Even then, all you are doing is putting most of the audience in the near 
field.

In BLaH2, we used tall Revel Studio Monitors (2 x 8 units arranged vertically 
like the TSR110s) and spent some time trying to determine if we had 1/r with 
them in the listening room.  In most rooms, at frequencies where ceiling  
floor reflections might help lengthen the line source, you are well into the 
area of reflections so how do you measure 1/r?  In fact measuring 1/r is a 
standard technique to determine how anechoic a chamber is.  I don't think we 
had anything approaching a line source for BLaH2

- However I do think even the above poor approximations help with a large 
audience.
- I think standard Classic Ambi Decoder with NFC at the correct frequency for 
the size of the array is appropriate even with these line sources.
- Probably good to move the transition from rV to rE lower down.


The York decoders for large area work are either rE with no Shelfs or the Furse 
Controlled Opposites which guarantee no sound from speakers opposite the 
source direction.

I should point out that Controlled Opposites require a fairly even distribution 
of speakers.  If the array is at all wonky, you get some really strange 
decoders.

If anyone else has experience with line sources in a horizontal Ambi rig, 
please show  tell.

Or even pseudo pontificating ...
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Re: [Sursound] Death of Ambisonia.com

[Richard Lee]

I hope all of you will join BLaH in thanking Etienne for the immense efforts 
and huge amount of time money that resulted in Ambisonia.  It must have seemed 
thankless at times.

I am certain it has been a major factor in the resurgence of interest and much 
new work on Ambisonics and surround sound in general.  Certainly, the BLaH 
papers were inspired by and would not have taken place without the motherlode 
of high quality recordings in Ambisonia.

Much remains to be done.  The next stage is to 

1)  create a general AV player (probably based on VideoLAN) which can play 
B-format Ambisonic recordings.

2)  In parallel with this, a good compressed format (probably based on 
Vorbis) and translating the Ambisonia treasure trove to this.

I believe these will be the catalyst for widespread acceptance of Ambisonics as 
people with a good universal AV player will be able to play B-format and there 
will be pressure to apply BLaH4 and other new technologies to create the 
Ambisonic Surround Decoder which will finally allow arbitrary speaker layouts.

Many thanks, Etienne.
We are forever in your debt.
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